am 5b299d2f: am 18643e70: am 6bc542e0: am 5aff259c: Merge "stagefright aacenc: Don\'t address a Word32 with a Word16 pointer"
* commit '5b299d2f9038452aa09a97d4f2ce57ec20bbc06a':
stagefright aacenc: Don't address a Word32 with a Word16 pointer
diff --git a/camera/Camera.cpp b/camera/Camera.cpp
index ee458f1..d43cb0b 100644
--- a/camera/Camera.cpp
+++ b/camera/Camera.cpp
@@ -27,7 +27,8 @@
#include <camera/ICameraRecordingProxyListener.h>
#include <camera/ICameraService.h>
-#include <surfaceflinger/Surface.h>
+#include <gui/ISurfaceTexture.h>
+#include <gui/Surface.h>
namespace android {
diff --git a/camera/ICamera.cpp b/camera/ICamera.cpp
index 70f5dbc..8d8408c 100644
--- a/camera/ICamera.cpp
+++ b/camera/ICamera.cpp
@@ -22,6 +22,8 @@
#include <sys/types.h>
#include <binder/Parcel.h>
#include <camera/ICamera.h>
+#include <gui/ISurfaceTexture.h>
+#include <gui/Surface.h>
namespace android {
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index e9642f7..f26747b 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -35,7 +35,7 @@
record.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright liblog libutils libbinder
+ libstagefright liblog libutils libbinder libstagefright_foundation
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
@@ -59,7 +59,7 @@
recordvideo.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright liblog libutils libbinder
+ libstagefright liblog libutils libbinder libstagefright_foundation
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
@@ -84,7 +84,7 @@
audioloop.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright liblog libutils libbinder
+ libstagefright liblog libutils libbinder libstagefright_foundation
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
@@ -147,4 +147,28 @@
include $(BUILD_EXECUTABLE)
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES:= \
+ codec.cpp \
+ SimplePlayer.cpp \
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright liblog libutils libbinder libstagefright_foundation \
+ libmedia libgui libcutils libui
+
+LOCAL_C_INCLUDES:= \
+ $(JNI_H_INCLUDE) \
+ frameworks/base/media/libstagefright \
+ $(TOP)/frameworks/base/include/media/stagefright/openmax
+
+LOCAL_CFLAGS += -Wno-multichar
+
+LOCAL_MODULE_TAGS := debug
+
+LOCAL_MODULE:= codec
+
+include $(BUILD_EXECUTABLE)
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
new file mode 100644
index 0000000..f269e80
--- /dev/null
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -0,0 +1,646 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SimplePlayer"
+#include <utils/Log.h>
+
+#include "SimplePlayer.h"
+
+#include <gui/SurfaceTextureClient.h>
+#include <media/AudioTrack.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/NativeWindowWrapper.h>
+#include <media/stagefright/NuMediaExtractor.h>
+
+namespace android {
+
+SimplePlayer::SimplePlayer()
+ : mState(UNINITIALIZED),
+ mDoMoreStuffGeneration(0),
+ mStartTimeRealUs(-1ll) {
+}
+
+SimplePlayer::~SimplePlayer() {
+}
+
+// static
+status_t PostAndAwaitResponse(
+ const sp<AMessage> &msg, sp<AMessage> *response) {
+ status_t err = msg->postAndAwaitResponse(response);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (!(*response)->findInt32("err", &err)) {
+ err = OK;
+ }
+
+ return err;
+}
+status_t SimplePlayer::setDataSource(const char *path) {
+ sp<AMessage> msg = new AMessage(kWhatSetDataSource, id());
+ msg->setString("path", path);
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::setSurface(const sp<ISurfaceTexture> &surfaceTexture) {
+ sp<AMessage> msg = new AMessage(kWhatSetSurface, id());
+
+ sp<SurfaceTextureClient> surfaceTextureClient;
+ if (surfaceTexture != NULL) {
+ surfaceTextureClient = new SurfaceTextureClient(surfaceTexture);
+ }
+
+ msg->setObject(
+ "native-window", new NativeWindowWrapper(surfaceTextureClient));
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::prepare() {
+ sp<AMessage> msg = new AMessage(kWhatPrepare, id());
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::start() {
+ sp<AMessage> msg = new AMessage(kWhatStart, id());
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::stop() {
+ sp<AMessage> msg = new AMessage(kWhatStop, id());
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t SimplePlayer::reset() {
+ sp<AMessage> msg = new AMessage(kWhatReset, id());
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+void SimplePlayer::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatSetDataSource:
+ {
+ status_t err;
+ if (mState != UNINITIALIZED) {
+ err = INVALID_OPERATION;
+ } else {
+ CHECK(msg->findString("path", &mPath));
+ mState = UNPREPARED;
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatSetSurface:
+ {
+ status_t err;
+ if (mState != UNPREPARED) {
+ err = INVALID_OPERATION;
+ } else {
+ sp<RefBase> obj;
+ CHECK(msg->findObject("native-window", &obj));
+
+ mNativeWindow = static_cast<NativeWindowWrapper *>(obj.get());
+
+ err = OK;
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatPrepare:
+ {
+ status_t err;
+ if (mState != UNPREPARED) {
+ err = INVALID_OPERATION;
+ } else {
+ err = onPrepare();
+
+ if (err == OK) {
+ mState = STOPPED;
+ }
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatStart:
+ {
+ status_t err = OK;
+
+ if (mState == UNPREPARED) {
+ err = onPrepare();
+
+ if (err == OK) {
+ mState = STOPPED;
+ }
+ }
+
+ if (err == OK) {
+ if (mState != STOPPED) {
+ err = INVALID_OPERATION;
+ } else {
+ err = onStart();
+
+ if (err == OK) {
+ mState = STARTED;
+ }
+ }
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatStop:
+ {
+ status_t err;
+
+ if (mState != STARTED) {
+ err = INVALID_OPERATION;
+ } else {
+ err = onStop();
+
+ if (err == OK) {
+ mState = STOPPED;
+ }
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatReset:
+ {
+ status_t err = OK;
+
+ if (mState == STARTED) {
+ CHECK_EQ(onStop(), (status_t)OK);
+ mState = STOPPED;
+ }
+
+ if (mState == STOPPED) {
+ err = onReset();
+ mState = UNINITIALIZED;
+ }
+
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatDoMoreStuff:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mDoMoreStuffGeneration) {
+ break;
+ }
+
+ status_t err = onDoMoreStuff();
+
+ if (err == OK) {
+ msg->post(10000ll);
+ }
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+status_t SimplePlayer::onPrepare() {
+ CHECK_EQ(mState, UNPREPARED);
+
+ mExtractor = new NuMediaExtractor;
+
+ status_t err = mExtractor->setDataSource(mPath.c_str());
+
+ if (err != OK) {
+ mExtractor.clear();
+ return err;
+ }
+
+ if (mCodecLooper == NULL) {
+ mCodecLooper = new ALooper;
+ mCodecLooper->start();
+ }
+
+ bool haveAudio = false;
+ bool haveVideo = false;
+ for (size_t i = 0; i < mExtractor->countTracks(); ++i) {
+ sp<AMessage> format;
+ status_t err = mExtractor->getTrackFormat(i, &format);
+ CHECK_EQ(err, (status_t)OK);
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+
+ if (!haveAudio && !strncasecmp(mime.c_str(), "audio/", 6)) {
+ haveAudio = true;
+ } else if (!haveVideo && !strncasecmp(mime.c_str(), "video/", 6)) {
+ haveVideo = true;
+ } else {
+ continue;
+ }
+
+ err = mExtractor->selectTrack(i);
+ CHECK_EQ(err, (status_t)OK);
+
+ CodecState *state =
+ &mStateByTrackIndex.editValueAt(
+ mStateByTrackIndex.add(i, CodecState()));
+
+ state->mNumFramesWritten = 0;
+ state->mCodec = MediaCodec::CreateByType(
+ mCodecLooper, mime.c_str(), false /* encoder */);
+
+ CHECK(state->mCodec != NULL);
+
+ err = state->mCodec->configure(
+ format, mNativeWindow->getSurfaceTextureClient(),
+ 0 /* flags */);
+
+ CHECK_EQ(err, (status_t)OK);
+
+ size_t j = 0;
+ sp<ABuffer> buffer;
+ while (format->findBuffer(StringPrintf("csd-%d", j).c_str(), &buffer)) {
+ state->mCSD.push_back(buffer);
+
+ ++j;
+ }
+ }
+
+ for (size_t i = 0; i < mStateByTrackIndex.size(); ++i) {
+ CodecState *state = &mStateByTrackIndex.editValueAt(i);
+
+ status_t err = state->mCodec->start();
+ CHECK_EQ(err, (status_t)OK);
+
+ err = state->mCodec->getInputBuffers(&state->mBuffers[0]);
+ CHECK_EQ(err, (status_t)OK);
+
+ err = state->mCodec->getOutputBuffers(&state->mBuffers[1]);
+ CHECK_EQ(err, (status_t)OK);
+
+ for (size_t j = 0; j < state->mCSD.size(); ++j) {
+ const sp<ABuffer> &srcBuffer = state->mCSD.itemAt(j);
+
+ size_t index;
+ err = state->mCodec->dequeueInputBuffer(&index, -1ll);
+ CHECK_EQ(err, (status_t)OK);
+
+ const sp<ABuffer> &dstBuffer = state->mBuffers[0].itemAt(index);
+
+ CHECK_LE(srcBuffer->size(), dstBuffer->capacity());
+ dstBuffer->setRange(0, srcBuffer->size());
+ memcpy(dstBuffer->data(), srcBuffer->data(), srcBuffer->size());
+
+ err = state->mCodec->queueInputBuffer(
+ index,
+ 0,
+ dstBuffer->size(),
+ 0ll,
+ MediaCodec::BUFFER_FLAG_CODECCONFIG);
+ CHECK_EQ(err, (status_t)OK);
+ }
+ }
+
+ return OK;
+}
+
+status_t SimplePlayer::onStart() {
+ CHECK_EQ(mState, STOPPED);
+
+ mStartTimeRealUs = -1ll;
+
+ sp<AMessage> msg = new AMessage(kWhatDoMoreStuff, id());
+ msg->setInt32("generation", ++mDoMoreStuffGeneration);
+ msg->post();
+
+ return OK;
+}
+
+status_t SimplePlayer::onStop() {
+ CHECK_EQ(mState, STARTED);
+
+ ++mDoMoreStuffGeneration;
+
+ return OK;
+}
+
+status_t SimplePlayer::onReset() {
+ CHECK_EQ(mState, STOPPED);
+
+ for (size_t i = 0; i < mStateByTrackIndex.size(); ++i) {
+ CodecState *state = &mStateByTrackIndex.editValueAt(i);
+
+ CHECK_EQ(state->mCodec->stop(), (status_t)OK);
+ }
+
+ mStartTimeRealUs = -1ll;
+
+ mStateByTrackIndex.clear();
+ mCodecLooper.clear();
+ mExtractor.clear();
+ mNativeWindow.clear();
+ mPath.clear();
+
+ return OK;
+}
+
+status_t SimplePlayer::onDoMoreStuff() {
+ ALOGV("onDoMoreStuff");
+ for (size_t i = 0; i < mStateByTrackIndex.size(); ++i) {
+ CodecState *state = &mStateByTrackIndex.editValueAt(i);
+
+ status_t err;
+ do {
+ size_t index;
+ err = state->mCodec->dequeueInputBuffer(&index);
+
+ if (err == OK) {
+ ALOGV("dequeued input buffer on track %d",
+ mStateByTrackIndex.keyAt(i));
+
+ state->mAvailInputBufferIndices.push_back(index);
+ } else {
+ ALOGV("dequeueInputBuffer on track %d returned %d",
+ mStateByTrackIndex.keyAt(i), err);
+ }
+ } while (err == OK);
+
+ do {
+ BufferInfo info;
+ err = state->mCodec->dequeueOutputBuffer(
+ &info.mIndex,
+ &info.mOffset,
+ &info.mSize,
+ &info.mPresentationTimeUs,
+ &info.mFlags);
+
+ if (err == OK) {
+ ALOGV("dequeued output buffer on track %d",
+ mStateByTrackIndex.keyAt(i));
+
+ state->mAvailOutputBufferInfos.push_back(info);
+ } else if (err == INFO_FORMAT_CHANGED) {
+ err = onOutputFormatChanged(mStateByTrackIndex.keyAt(i), state);
+ CHECK_EQ(err, (status_t)OK);
+ } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+ err = state->mCodec->getOutputBuffers(&state->mBuffers[1]);
+ CHECK_EQ(err, (status_t)OK);
+ } else {
+ ALOGV("dequeueOutputBuffer on track %d returned %d",
+ mStateByTrackIndex.keyAt(i), err);
+ }
+ } while (err == OK
+ || err == INFO_FORMAT_CHANGED
+ || err == INFO_OUTPUT_BUFFERS_CHANGED);
+ }
+
+ for (;;) {
+ size_t trackIndex;
+ status_t err = mExtractor->getSampleTrackIndex(&trackIndex);
+
+ if (err != OK) {
+ ALOGI("encountered input EOS.");
+ break;
+ } else {
+ CodecState *state = &mStateByTrackIndex.editValueFor(trackIndex);
+
+ if (state->mAvailInputBufferIndices.empty()) {
+ break;
+ }
+
+ size_t index = *state->mAvailInputBufferIndices.begin();
+ state->mAvailInputBufferIndices.erase(
+ state->mAvailInputBufferIndices.begin());
+
+ const sp<ABuffer> &dstBuffer =
+ state->mBuffers[0].itemAt(index);
+
+ err = mExtractor->readSampleData(dstBuffer);
+ CHECK_EQ(err, (status_t)OK);
+
+ int64_t timeUs;
+ CHECK_EQ(mExtractor->getSampleTime(&timeUs), (status_t)OK);
+
+ err = state->mCodec->queueInputBuffer(
+ index,
+ dstBuffer->offset(),
+ dstBuffer->size(),
+ timeUs,
+ 0);
+ CHECK_EQ(err, (status_t)OK);
+
+ ALOGV("enqueued input data on track %d", trackIndex);
+
+ err = mExtractor->advance();
+ CHECK_EQ(err, (status_t)OK);
+ }
+ }
+
+ int64_t nowUs = ALooper::GetNowUs();
+
+ if (mStartTimeRealUs < 0ll) {
+ mStartTimeRealUs = nowUs + 1000000ll;
+ }
+
+ for (size_t i = 0; i < mStateByTrackIndex.size(); ++i) {
+ CodecState *state = &mStateByTrackIndex.editValueAt(i);
+
+ while (!state->mAvailOutputBufferInfos.empty()) {
+ BufferInfo *info = &*state->mAvailOutputBufferInfos.begin();
+
+ int64_t whenRealUs = info->mPresentationTimeUs + mStartTimeRealUs;
+ int64_t lateByUs = nowUs - whenRealUs;
+
+ if (lateByUs > -10000ll) {
+ bool release = true;
+
+ if (lateByUs > 30000ll) {
+ ALOGI("track %d buffer late by %lld us, dropping.",
+ mStateByTrackIndex.keyAt(i), lateByUs);
+ state->mCodec->releaseOutputBuffer(info->mIndex);
+ } else {
+ if (state->mAudioTrack != NULL) {
+ const sp<ABuffer> &srcBuffer =
+ state->mBuffers[1].itemAt(info->mIndex);
+
+ renderAudio(state, info, srcBuffer);
+
+ if (info->mSize > 0) {
+ release = false;
+ }
+ }
+
+ if (release) {
+ state->mCodec->renderOutputBufferAndRelease(
+ info->mIndex);
+ }
+ }
+
+ if (release) {
+ state->mAvailOutputBufferInfos.erase(
+ state->mAvailOutputBufferInfos.begin());
+
+ info = NULL;
+ } else {
+ break;
+ }
+ } else {
+ ALOGV("track %d buffer early by %lld us.",
+ mStateByTrackIndex.keyAt(i), -lateByUs);
+ break;
+ }
+ }
+ }
+
+ return OK;
+}
+
+status_t SimplePlayer::onOutputFormatChanged(
+ size_t trackIndex, CodecState *state) {
+ sp<AMessage> format;
+ status_t err = state->mCodec->getOutputFormat(&format);
+
+ if (err != OK) {
+ return err;
+ }
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+
+ if (!strncasecmp(mime.c_str(), "audio/", 6)) {
+ int32_t channelCount;
+ int32_t sampleRate;
+ CHECK(format->findInt32("channel-count", &channelCount));
+ CHECK(format->findInt32("sample-rate", &sampleRate));
+
+ state->mAudioTrack = new AudioTrack(
+ AUDIO_STREAM_MUSIC,
+ sampleRate,
+ AUDIO_FORMAT_PCM_16_BIT,
+ (channelCount == 1)
+ ? AUDIO_CHANNEL_OUT_MONO : AUDIO_CHANNEL_OUT_STEREO,
+ 0);
+
+ state->mNumFramesWritten = 0;
+ }
+
+ return OK;
+}
+
+void SimplePlayer::renderAudio(
+ CodecState *state, BufferInfo *info, const sp<ABuffer> &buffer) {
+ CHECK(state->mAudioTrack != NULL);
+
+ if (state->mAudioTrack->stopped()) {
+ state->mAudioTrack->start();
+ }
+
+ uint32_t numFramesPlayed;
+ CHECK_EQ(state->mAudioTrack->getPosition(&numFramesPlayed), (status_t)OK);
+
+ uint32_t numFramesAvailableToWrite =
+ state->mAudioTrack->frameCount()
+ - (state->mNumFramesWritten - numFramesPlayed);
+
+ size_t numBytesAvailableToWrite =
+ numFramesAvailableToWrite * state->mAudioTrack->frameSize();
+
+ size_t copy = info->mSize;
+ if (copy > numBytesAvailableToWrite) {
+ copy = numBytesAvailableToWrite;
+ }
+
+ if (copy == 0) {
+ return;
+ }
+
+ int64_t startTimeUs = ALooper::GetNowUs();
+
+ ssize_t nbytes = state->mAudioTrack->write(
+ buffer->base() + info->mOffset, copy);
+
+ CHECK_EQ(nbytes, (ssize_t)copy);
+
+ int64_t delayUs = ALooper::GetNowUs() - startTimeUs;
+
+ uint32_t numFramesWritten = nbytes / state->mAudioTrack->frameSize();
+
+ if (delayUs > 2000ll) {
+ ALOGW("AudioTrack::write took %lld us, numFramesAvailableToWrite=%u, "
+ "numFramesWritten=%u",
+ delayUs, numFramesAvailableToWrite, numFramesWritten);
+ }
+
+ info->mOffset += nbytes;
+ info->mSize -= nbytes;
+
+ state->mNumFramesWritten += numFramesWritten;
+}
+
+} // namespace android
diff --git a/cmds/stagefright/SimplePlayer.h b/cmds/stagefright/SimplePlayer.h
new file mode 100644
index 0000000..2548252
--- /dev/null
+++ b/cmds/stagefright/SimplePlayer.h
@@ -0,0 +1,109 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/foundation/AString.h>
+#include <utils/KeyedVector.h>
+
+namespace android {
+
+struct ABuffer;
+struct ALooper;
+struct AudioTrack;
+struct ISurfaceTexture;
+struct MediaCodec;
+struct NativeWindowWrapper;
+struct NuMediaExtractor;
+
+struct SimplePlayer : public AHandler {
+ SimplePlayer();
+
+ status_t setDataSource(const char *path);
+ status_t setSurface(const sp<ISurfaceTexture> &surfaceTexture);
+ status_t prepare();
+ status_t start();
+ status_t stop();
+ status_t reset();
+
+protected:
+ virtual ~SimplePlayer();
+
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum State {
+ UNINITIALIZED,
+ UNPREPARED,
+ STOPPED,
+ STARTED
+ };
+
+ enum {
+ kWhatSetDataSource,
+ kWhatSetSurface,
+ kWhatPrepare,
+ kWhatStart,
+ kWhatStop,
+ kWhatReset,
+ kWhatDoMoreStuff,
+ };
+
+ struct BufferInfo {
+ size_t mIndex;
+ size_t mOffset;
+ size_t mSize;
+ int64_t mPresentationTimeUs;
+ uint32_t mFlags;
+ };
+
+ struct CodecState
+ {
+ sp<MediaCodec> mCodec;
+ Vector<sp<ABuffer> > mCSD;
+ Vector<sp<ABuffer> > mBuffers[2];
+
+ List<size_t> mAvailInputBufferIndices;
+ List<BufferInfo> mAvailOutputBufferInfos;
+
+ sp<AudioTrack> mAudioTrack;
+ uint32_t mNumFramesWritten;
+ };
+
+ State mState;
+ AString mPath;
+ sp<NativeWindowWrapper> mNativeWindow;
+
+ sp<NuMediaExtractor> mExtractor;
+ sp<ALooper> mCodecLooper;
+ KeyedVector<size_t, CodecState> mStateByTrackIndex;
+ int32_t mDoMoreStuffGeneration;
+
+ int64_t mStartTimeRealUs;
+
+ status_t onPrepare();
+ status_t onStart();
+ status_t onStop();
+ status_t onReset();
+ status_t onDoMoreStuff();
+ status_t onOutputFormatChanged(size_t trackIndex, CodecState *state);
+
+ void renderAudio(
+ CodecState *state, BufferInfo *info, const sp<ABuffer> &buffer);
+
+ DISALLOW_EVIL_CONSTRUCTORS(SimplePlayer);
+};
+
+} // namespace android
diff --git a/cmds/stagefright/SineSource.cpp b/cmds/stagefright/SineSource.cpp
index 021f636..14b4306 100644
--- a/cmds/stagefright/SineSource.cpp
+++ b/cmds/stagefright/SineSource.cpp
@@ -3,7 +3,7 @@
#include <math.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index 858681f..a6362a4 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -2,10 +2,10 @@
#include <binder/ProcessState.h>
#include <media/mediarecorder.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/AudioSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXClient.h>
@@ -24,7 +24,7 @@
android::ProcessState::self()->startThreadPool();
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
#if 0
sp<MediaSource> source = new SineSource(kSampleRate, kNumChannels);
@@ -82,7 +82,7 @@
delete player;
player = NULL;
#elif 0
- CHECK_EQ(decoder->start(), OK);
+ CHECK_EQ(decoder->start(), (status_t)OK);
MediaBuffer *buffer;
while (decoder->read(&buffer) == OK) {
@@ -95,7 +95,7 @@
buffer = NULL;
}
- CHECK_EQ(decoder->stop(), OK);
+ CHECK_EQ(decoder->stop(), (status_t)OK);
#endif
#endif
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
new file mode 100644
index 0000000..b850190
--- /dev/null
+++ b/cmds/stagefright/codec.cpp
@@ -0,0 +1,408 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "codec"
+#include <utils/Log.h>
+
+#include "SimplePlayer.h"
+
+#include <binder/ProcessState.h>
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaCodec.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/NuMediaExtractor.h>
+#include <gui/SurfaceComposerClient.h>
+
+static void usage(const char *me) {
+ fprintf(stderr, "usage: %s [-a] use audio\n"
+ "\t\t[-v] use video\n"
+ "\t\t[-p] playback\n", me);
+
+ exit(1);
+}
+
+namespace android {
+
+struct CodecState {
+ sp<MediaCodec> mCodec;
+ Vector<sp<ABuffer> > mCSD;
+ size_t mCSDIndex;
+ Vector<sp<ABuffer> > mInBuffers;
+ Vector<sp<ABuffer> > mOutBuffers;
+ bool mSawOutputEOS;
+};
+
+} // namespace android
+
+static int decode(
+ const android::sp<android::ALooper> &looper,
+ const char *path,
+ bool useAudio,
+ bool useVideo) {
+ using namespace android;
+
+ sp<NuMediaExtractor> extractor = new NuMediaExtractor;
+ if (extractor->setDataSource(path) != OK) {
+ fprintf(stderr, "unable to instantiate extractor.\n");
+ return 1;
+ }
+
+ KeyedVector<size_t, CodecState> stateByTrack;
+
+ bool haveAudio = false;
+ bool haveVideo = false;
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<AMessage> format;
+ status_t err = extractor->getTrackFormat(i, &format);
+ CHECK_EQ(err, (status_t)OK);
+
+ AString mime;
+ CHECK(format->findString("mime", &mime));
+
+ if (useAudio && !haveAudio
+ && !strncasecmp(mime.c_str(), "audio/", 6)) {
+ haveAudio = true;
+ } else if (useVideo && !haveVideo
+ && !strncasecmp(mime.c_str(), "video/", 6)) {
+ haveVideo = true;
+ } else {
+ continue;
+ }
+
+ ALOGV("selecting track %d", i);
+
+ err = extractor->selectTrack(i);
+ CHECK_EQ(err, (status_t)OK);
+
+ CodecState *state =
+ &stateByTrack.editValueAt(stateByTrack.add(i, CodecState()));
+
+ state->mCodec = MediaCodec::CreateByType(
+ looper, mime.c_str(), false /* encoder */);
+
+ CHECK(state->mCodec != NULL);
+
+ err = state->mCodec->configure(
+ format, NULL /* surfaceTexture */, 0 /* flags */);
+
+ CHECK_EQ(err, (status_t)OK);
+
+ size_t j = 0;
+ sp<ABuffer> buffer;
+ while (format->findBuffer(StringPrintf("csd-%d", j).c_str(), &buffer)) {
+ state->mCSD.push_back(buffer);
+
+ ++j;
+ }
+
+ state->mCSDIndex = 0;
+ state->mSawOutputEOS = false;
+
+ ALOGV("got %d pieces of codec specific data.", state->mCSD.size());
+ }
+
+ CHECK(!stateByTrack.isEmpty());
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ sp<MediaCodec> codec = state->mCodec;
+
+ CHECK_EQ((status_t)OK, codec->start());
+
+ CHECK_EQ((status_t)OK, codec->getInputBuffers(&state->mInBuffers));
+ CHECK_EQ((status_t)OK, codec->getOutputBuffers(&state->mOutBuffers));
+
+ ALOGV("got %d input and %d output buffers",
+ state->mInBuffers.size(), state->mOutBuffers.size());
+
+ while (state->mCSDIndex < state->mCSD.size()) {
+ size_t index;
+ status_t err = codec->dequeueInputBuffer(&index);
+
+ if (err == -EAGAIN) {
+ usleep(10000);
+ continue;
+ }
+
+ CHECK_EQ(err, (status_t)OK);
+
+ const sp<ABuffer> &srcBuffer =
+ state->mCSD.itemAt(state->mCSDIndex++);
+
+ const sp<ABuffer> &buffer = state->mInBuffers.itemAt(index);
+
+ memcpy(buffer->data(), srcBuffer->data(), srcBuffer->size());
+
+ err = codec->queueInputBuffer(
+ index,
+ 0 /* offset */,
+ srcBuffer->size(),
+ 0ll /* timeUs */,
+ MediaCodec::BUFFER_FLAG_CODECCONFIG);
+
+ CHECK_EQ(err, (status_t)OK);
+ }
+ }
+
+ bool sawInputEOS = false;
+
+ for (;;) {
+ if (!sawInputEOS) {
+ size_t trackIndex;
+ status_t err = extractor->getSampleTrackIndex(&trackIndex);
+
+ if (err != OK) {
+ ALOGV("signalling EOS.");
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ for (;;) {
+ size_t index;
+ err = state->mCodec->dequeueInputBuffer(&index);
+
+ if (err == -EAGAIN) {
+ continue;
+ }
+
+ CHECK_EQ(err, (status_t)OK);
+
+ err = state->mCodec->queueInputBuffer(
+ index,
+ 0 /* offset */,
+ 0 /* size */,
+ 0ll /* timeUs */,
+ MediaCodec::BUFFER_FLAG_EOS);
+
+ CHECK_EQ(err, (status_t)OK);
+ break;
+ }
+ }
+
+ sawInputEOS = true;
+ } else {
+ CodecState *state = &stateByTrack.editValueFor(trackIndex);
+
+ size_t index;
+ err = state->mCodec->dequeueInputBuffer(&index);
+
+ if (err == OK) {
+ ALOGV("filling input buffer %d", index);
+
+ const sp<ABuffer> &buffer = state->mInBuffers.itemAt(index);
+
+ err = extractor->readSampleData(buffer);
+ CHECK_EQ(err, (status_t)OK);
+
+ int64_t timeUs;
+ err = extractor->getSampleTime(&timeUs);
+ CHECK_EQ(err, (status_t)OK);
+
+ err = state->mCodec->queueInputBuffer(
+ index,
+ 0 /* offset */,
+ buffer->size(),
+ timeUs,
+ 0 /* flags */);
+
+ CHECK_EQ(err, (status_t)OK);
+
+ extractor->advance();
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ }
+
+ bool sawOutputEOSOnAllTracks = true;
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+ if (!state->mSawOutputEOS) {
+ sawOutputEOSOnAllTracks = false;
+ break;
+ }
+ }
+
+ if (sawOutputEOSOnAllTracks) {
+ break;
+ }
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ if (state->mSawOutputEOS) {
+ continue;
+ }
+
+ size_t index;
+ size_t offset;
+ size_t size;
+ int64_t presentationTimeUs;
+ uint32_t flags;
+ status_t err = state->mCodec->dequeueOutputBuffer(
+ &index, &offset, &size, &presentationTimeUs, &flags,
+ 10000ll);
+
+ if (err == OK) {
+ ALOGV("draining output buffer %d, time = %lld us",
+ index, presentationTimeUs);
+
+ err = state->mCodec->releaseOutputBuffer(index);
+ CHECK_EQ(err, (status_t)OK);
+
+ if (flags & MediaCodec::BUFFER_FLAG_EOS) {
+ ALOGV("reached EOS on output.");
+
+ state->mSawOutputEOS = true;
+ }
+ } else if (err == INFO_OUTPUT_BUFFERS_CHANGED) {
+ ALOGV("INFO_OUTPUT_BUFFERS_CHANGED");
+ CHECK_EQ((status_t)OK,
+ state->mCodec->getOutputBuffers(&state->mOutBuffers));
+
+ ALOGV("got %d output buffers", state->mOutBuffers.size());
+ } else if (err == INFO_FORMAT_CHANGED) {
+ sp<AMessage> format;
+ CHECK_EQ((status_t)OK, state->mCodec->getOutputFormat(&format));
+
+ ALOGV("INFO_FORMAT_CHANGED: %s", format->debugString().c_str());
+ } else {
+ CHECK_EQ(err, -EAGAIN);
+ }
+ }
+ }
+
+ for (size_t i = 0; i < stateByTrack.size(); ++i) {
+ CodecState *state = &stateByTrack.editValueAt(i);
+
+ CHECK_EQ((status_t)OK, state->mCodec->stop());
+ }
+
+ return 0;
+}
+
+int main(int argc, char **argv) {
+ using namespace android;
+
+ const char *me = argv[0];
+
+ bool useAudio = false;
+ bool useVideo = false;
+ bool playback = false;
+
+ int res;
+ while ((res = getopt(argc, argv, "havp")) >= 0) {
+ switch (res) {
+ case 'a':
+ {
+ useAudio = true;
+ break;
+ }
+
+ case 'v':
+ {
+ useVideo = true;
+ break;
+ }
+
+ case 'p':
+ {
+ playback = true;
+ break;
+ }
+
+ case '?':
+ case 'h':
+ default:
+ {
+ usage(me);
+ }
+ }
+ }
+
+ argc -= optind;
+ argv += optind;
+
+ if (argc != 1) {
+ usage(me);
+ }
+
+ if (!useAudio && !useVideo) {
+ useAudio = useVideo = true;
+ }
+
+ ProcessState::self()->startThreadPool();
+
+ DataSource::RegisterDefaultSniffers();
+
+ sp<ALooper> looper = new ALooper;
+ looper->start();
+
+ if (playback) {
+ sp<SurfaceComposerClient> composerClient = new SurfaceComposerClient;
+ CHECK_EQ(composerClient->initCheck(), (status_t)OK);
+
+ ssize_t displayWidth = composerClient->getDisplayWidth(0);
+ ssize_t displayHeight = composerClient->getDisplayHeight(0);
+
+ ALOGV("display is %ld x %ld\n", displayWidth, displayHeight);
+
+ sp<SurfaceControl> control =
+ composerClient->createSurface(
+ String8("A Surface"),
+ 0,
+ displayWidth,
+ displayHeight,
+ PIXEL_FORMAT_RGB_565,
+ 0);
+
+ CHECK(control != NULL);
+ CHECK(control->isValid());
+
+ SurfaceComposerClient::openGlobalTransaction();
+ CHECK_EQ(control->setLayer(INT_MAX), (status_t)OK);
+ CHECK_EQ(control->show(), (status_t)OK);
+ SurfaceComposerClient::closeGlobalTransaction();
+
+ sp<Surface> surface = control->getSurface();
+ CHECK(surface != NULL);
+
+ sp<SimplePlayer> player = new SimplePlayer;
+ looper->registerHandler(player);
+
+ player->setDataSource(argv[0]);
+ player->setSurface(surface->getSurfaceTexture());
+ player->start();
+ sleep(60);
+ player->stop();
+ player->reset();
+
+ composerClient->dispose();
+ } else {
+ decode(looper, argv[0], useAudio, useVideo);
+ }
+
+ looper->stop();
+
+ return 0;
+}
diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index 613435d..45c3f7b 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -17,11 +17,11 @@
#include "SineSource.h"
#include <binder/ProcessState.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/CameraSource.h>
#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaExtractor.h>
@@ -38,7 +38,7 @@
static const int32_t kAudioBitRate = 12200;
static const int64_t kDurationUs = 10000000LL; // 10 seconds
-#if 1
+#if 0
class DummySource : public MediaSource {
public:
@@ -183,7 +183,7 @@
return 1;
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
status_t err = OK;
@@ -231,14 +231,14 @@
sp<MPEG4Writer> writer = new MPEG4Writer("/sdcard/output.mp4");
writer->addSource(encoder);
writer->setMaxFileDuration(kDurationUs);
- CHECK_EQ(OK, writer->start());
+ CHECK_EQ((status_t)OK, writer->start());
while (!writer->reachedEOS()) {
fprintf(stderr, ".");
usleep(100000);
}
err = writer->stop();
#else
- CHECK_EQ(OK, encoder->start());
+ CHECK_EQ((status_t)OK, encoder->start());
MediaBuffer *buffer;
while (encoder->read(&buffer) == OK) {
@@ -272,7 +272,7 @@
for (int i = 0; i < 100; ++i) {
MediaBuffer *buffer;
status_t err = source->read(&buffer);
- CHECK_EQ(err, OK);
+ CHECK_EQ(err, (status_t)OK);
printf("got a frame, data=%p, size=%d\n",
buffer->data(), buffer->range_length());
@@ -299,7 +299,7 @@
android::ProcessState::self()->startThreadPool();
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
const int32_t kSampleRate = 22050;
const int32_t kNumChannels = 2;
@@ -318,7 +318,7 @@
sp<MetaData> encMeta = new MetaData;
encMeta->setCString(kKeyMIMEType,
- 1 ? MEDIA_MIMETYPE_AUDIO_AMR_WB : MEDIA_MIMETYPE_AUDIO_AAC);
+ 0 ? MEDIA_MIMETYPE_AUDIO_AMR_WB : MEDIA_MIMETYPE_AUDIO_AAC);
encMeta->setInt32(kKeySampleRate, kSampleRate);
encMeta->setInt32(kKeyChannelCount, kNumChannels);
encMeta->setInt32(kKeyMaxInputSize, 8192);
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index c402286..3bd1fe2 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -17,9 +17,9 @@
#include "SineSource.h"
#include <binder/ProcessState.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MPEG4Writer.h>
@@ -243,7 +243,7 @@
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
status_t err = OK;
sp<MediaSource> source =
@@ -283,7 +283,7 @@
sp<MPEG4Writer> writer = new MPEG4Writer(fileName);
writer->addSource(encoder);
int64_t start = systemTime();
- CHECK_EQ(OK, writer->start());
+ CHECK_EQ((status_t)OK, writer->start());
while (!writer->reachedEOS()) {
}
err = writer->stop();
diff --git a/cmds/stagefright/sf2.cpp b/cmds/stagefright/sf2.cpp
index ae80f88..1d28793 100644
--- a/cmds/stagefright/sf2.cpp
+++ b/cmds/stagefright/sf2.cpp
@@ -32,8 +32,7 @@
#include <media/stagefright/NativeWindowWrapper.h>
#include <media/stagefright/Utils.h>
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
+#include <gui/SurfaceComposerClient.h>
#include "include/ESDS.h"
@@ -198,9 +197,7 @@
(new AMessage(kWhatSeek, id()))->post(5000000ll);
} else if (what == ACodec::kWhatOutputFormatChanged) {
- } else {
- CHECK_EQ(what, (int32_t)ACodec::kWhatShutdownCompleted);
-
+ } else if (what == ACodec::kWhatShutdownCompleted) {
mDecodeLooper->unregisterHandler(mCodec->id());
if (mDecodeLooper != looper()) {
@@ -360,7 +357,7 @@
buffer->meta()->setInt32("csd", true);
mCSD.push(buffer);
- msg->setObject("csd", buffer);
+ msg->setBuffer("csd", buffer);
} else if (meta->findData(kKeyESDS, &type, &data, &size)) {
ESDS esds((const char *)data, size);
CHECK_EQ(esds.InitCheck(), (status_t)OK);
@@ -410,9 +407,8 @@
return;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<ABuffer> outBuffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> outBuffer;
+ CHECK(msg->findBuffer("buffer", &outBuffer));
if (mCSDIndex < mCSD.size()) {
outBuffer = mCSD.editItemAt(mCSDIndex++);
@@ -511,15 +507,14 @@
}
}
- reply->setObject("buffer", outBuffer);
+ reply->setBuffer("buffer", outBuffer);
reply->post();
}
void onDrainThisBuffer(const sp<AMessage> &msg) {
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
mTotalBytesReceived += buffer->size();
sp<AMessage> reply;
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 7cb8f62..dab2e0f 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -55,9 +55,7 @@
#include <fcntl.h>
#include <gui/SurfaceTextureClient.h>
-
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
+#include <gui/SurfaceComposerClient.h>
using namespace android;
diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index 0d6c738..efa1445 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp
@@ -32,8 +32,7 @@
#include <binder/IServiceManager.h>
#include <media/IMediaPlayerService.h>
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
+#include <gui/SurfaceComposerClient.h>
#include <fcntl.h>
diff --git a/drm/common/DrmEngineBase.cpp b/drm/common/DrmEngineBase.cpp
index 9b16c36..1c345a2 100644
--- a/drm/common/DrmEngineBase.cpp
+++ b/drm/common/DrmEngineBase.cpp
@@ -120,13 +120,23 @@
}
status_t DrmEngineBase::openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, int fd, off64_t offset, off64_t length) {
- return onOpenDecryptSession(uniqueId, decryptHandle, fd, offset, length);
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length, const char* mime) {
+
+ if (!mime || mime[0] == '\0') {
+ return onOpenDecryptSession(uniqueId, decryptHandle, fd, offset, length);
+ }
+
+ return onOpenDecryptSession(uniqueId, decryptHandle, fd, offset, length, mime);
}
status_t DrmEngineBase::openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, const char* uri) {
- return onOpenDecryptSession(uniqueId, decryptHandle, uri);
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri, const char* mime) {
+ if (!mime || mime[0] == '\0') {
+ return onOpenDecryptSession(uniqueId, decryptHandle, uri);
+ }
+ return onOpenDecryptSession(uniqueId, decryptHandle, uri, mime);
}
status_t DrmEngineBase::closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle) {
diff --git a/drm/common/IDrmManagerService.cpp b/drm/common/IDrmManagerService.cpp
index 3ed8ade..43f64f2 100644
--- a/drm/common/IDrmManagerService.cpp
+++ b/drm/common/IDrmManagerService.cpp
@@ -600,7 +600,7 @@
}
DecryptHandle* BpDrmManagerService::openDecryptSession(
- int uniqueId, int fd, off64_t offset, off64_t length) {
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime) {
ALOGV("Entering BpDrmManagerService::openDecryptSession");
Parcel data, reply;
@@ -609,6 +609,11 @@
data.writeFileDescriptor(fd);
data.writeInt64(offset);
data.writeInt64(length);
+ String8 mimeType;
+ if (mime) {
+ mimeType = mime;
+ }
+ data.writeString8(mimeType);
remote()->transact(OPEN_DECRYPT_SESSION, data, &reply);
@@ -620,13 +625,20 @@
return handle;
}
-DecryptHandle* BpDrmManagerService::openDecryptSession(int uniqueId, const char* uri) {
- ALOGV("Entering BpDrmManagerService::openDecryptSession");
+DecryptHandle* BpDrmManagerService::openDecryptSession(
+ int uniqueId, const char* uri, const char* mime) {
+
+ ALOGV("Entering BpDrmManagerService::openDecryptSession: mime=%s", mime? mime: "NULL");
Parcel data, reply;
data.writeInterfaceToken(IDrmManagerService::getInterfaceDescriptor());
data.writeInt32(uniqueId);
data.writeString8(String8(uri));
+ String8 mimeType;
+ if (mime) {
+ mimeType = mime;
+ }
+ data.writeString8(mimeType);
remote()->transact(OPEN_DECRYPT_SESSION_FROM_URI, data, &reply);
@@ -1265,8 +1277,10 @@
const off64_t offset = data.readInt64();
const off64_t length = data.readInt64();
+ const String8 mime = data.readString8();
+
DecryptHandle* handle
- = openDecryptSession(uniqueId, fd, offset, length);
+ = openDecryptSession(uniqueId, fd, offset, length, mime.string());
if (NULL != handle) {
writeDecryptHandleToParcelData(handle, reply);
@@ -1283,8 +1297,9 @@
const int uniqueId = data.readInt32();
const String8 uri = data.readString8();
+ const String8 mime = data.readString8();
- DecryptHandle* handle = openDecryptSession(uniqueId, uri.string());
+ DecryptHandle* handle = openDecryptSession(uniqueId, uri.string(), mime.string());
if (NULL != handle) {
writeDecryptHandleToParcelData(handle, reply);
diff --git a/drm/drmserver/DrmManager.cpp b/drm/drmserver/DrmManager.cpp
index 3abf3d3..999295a 100644
--- a/drm/drmserver/DrmManager.cpp
+++ b/drm/drmserver/DrmManager.cpp
@@ -426,7 +426,9 @@
return DRM_NO_ERROR;
}
-DecryptHandle* DrmManager::openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length) {
+DecryptHandle* DrmManager::openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime) {
+
Mutex::Autolock _l(mDecryptLock);
status_t result = DRM_ERROR_CANNOT_HANDLE;
Vector<String8> plugInIdList = mPlugInManager.getPlugInIdList();
@@ -438,7 +440,7 @@
for (unsigned int index = 0; index < plugInIdList.size(); index++) {
String8 plugInId = plugInIdList.itemAt(index);
IDrmEngine& rDrmEngine = mPlugInManager.getPlugIn(plugInId);
- result = rDrmEngine.openDecryptSession(uniqueId, handle, fd, offset, length);
+ result = rDrmEngine.openDecryptSession(uniqueId, handle, fd, offset, length, mime);
if (DRM_NO_ERROR == result) {
++mDecryptSessionId;
@@ -453,7 +455,8 @@
return handle;
}
-DecryptHandle* DrmManager::openDecryptSession(int uniqueId, const char* uri) {
+DecryptHandle* DrmManager::openDecryptSession(
+ int uniqueId, const char* uri, const char* mime) {
Mutex::Autolock _l(mDecryptLock);
status_t result = DRM_ERROR_CANNOT_HANDLE;
Vector<String8> plugInIdList = mPlugInManager.getPlugInIdList();
@@ -465,7 +468,7 @@
for (unsigned int index = 0; index < plugInIdList.size(); index++) {
String8 plugInId = plugInIdList.itemAt(index);
IDrmEngine& rDrmEngine = mPlugInManager.getPlugIn(plugInId);
- result = rDrmEngine.openDecryptSession(uniqueId, handle, uri);
+ result = rDrmEngine.openDecryptSession(uniqueId, handle, uri, mime);
if (DRM_NO_ERROR == result) {
++mDecryptSessionId;
diff --git a/drm/drmserver/DrmManagerService.cpp b/drm/drmserver/DrmManagerService.cpp
index df17ac5..caeb026 100644
--- a/drm/drmserver/DrmManagerService.cpp
+++ b/drm/drmserver/DrmManagerService.cpp
@@ -208,20 +208,20 @@
}
DecryptHandle* DrmManagerService::openDecryptSession(
- int uniqueId, int fd, off64_t offset, off64_t length) {
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime) {
ALOGV("Entering DrmManagerService::openDecryptSession");
if (isProtectedCallAllowed()) {
- return mDrmManager->openDecryptSession(uniqueId, fd, offset, length);
+ return mDrmManager->openDecryptSession(uniqueId, fd, offset, length, mime);
}
return NULL;
}
DecryptHandle* DrmManagerService::openDecryptSession(
- int uniqueId, const char* uri) {
+ int uniqueId, const char* uri, const char* mime) {
ALOGV("Entering DrmManagerService::openDecryptSession with uri");
if (isProtectedCallAllowed()) {
- return mDrmManager->openDecryptSession(uniqueId, uri);
+ return mDrmManager->openDecryptSession(uniqueId, uri, mime);
}
return NULL;
diff --git a/drm/drmserver/main_drmserver.cpp b/drm/drmserver/main_drmserver.cpp
index ed42818..e61b269 100644
--- a/drm/drmserver/main_drmserver.cpp
+++ b/drm/drmserver/main_drmserver.cpp
@@ -14,6 +14,9 @@
* limitations under the License.
*/
+#define LOG_TAG "drmserver"
+//#define LOG_NDEBUG 0
+
#include <sys/types.h>
#include <unistd.h>
#include <grp.h>
@@ -32,7 +35,7 @@
{
sp<ProcessState> proc(ProcessState::self());
sp<IServiceManager> sm = defaultServiceManager();
- ALOGI("ServiceManager: %p", sm.get());
+ ALOGV("ServiceManager: %p", sm.get());
DrmManagerService::instantiate();
ProcessState::self()->startThreadPool();
IPCThreadState::self()->joinThreadPool();
diff --git a/drm/libdrmframework/DrmManagerClient.cpp b/drm/libdrmframework/DrmManagerClient.cpp
index c9c0d57..8768c08 100644
--- a/drm/libdrmframework/DrmManagerClient.cpp
+++ b/drm/libdrmframework/DrmManagerClient.cpp
@@ -116,12 +116,18 @@
return mDrmManagerClientImpl->getAllSupportInfo(mUniqueId, length, drmSupportInfoArray);
}
-sp<DecryptHandle> DrmManagerClient::openDecryptSession(int fd, off64_t offset, off64_t length) {
- return mDrmManagerClientImpl->openDecryptSession(mUniqueId, fd, offset, length);
+sp<DecryptHandle> DrmManagerClient::openDecryptSession(
+ int fd, off64_t offset, off64_t length, const char* mime) {
+
+ return mDrmManagerClientImpl->openDecryptSession(
+ mUniqueId, fd, offset, length, mime);
}
-sp<DecryptHandle> DrmManagerClient::openDecryptSession(const char* uri) {
- return mDrmManagerClientImpl->openDecryptSession(mUniqueId, uri);
+sp<DecryptHandle> DrmManagerClient::openDecryptSession(
+ const char* uri, const char* mime) {
+
+ return mDrmManagerClientImpl->openDecryptSession(
+ mUniqueId, uri, mime);
}
status_t DrmManagerClient::closeDecryptSession(sp<DecryptHandle> &decryptHandle) {
diff --git a/drm/libdrmframework/DrmManagerClientImpl.cpp b/drm/libdrmframework/DrmManagerClientImpl.cpp
index b222b8f..fb0439e 100644
--- a/drm/libdrmframework/DrmManagerClientImpl.cpp
+++ b/drm/libdrmframework/DrmManagerClientImpl.cpp
@@ -255,15 +255,19 @@
}
sp<DecryptHandle> DrmManagerClientImpl::openDecryptSession(
- int uniqueId, int fd, off64_t offset, off64_t length) {
- return getDrmManagerService()->openDecryptSession(uniqueId, fd, offset, length);
+ int uniqueId, int fd, off64_t offset,
+ off64_t length, const char* mime) {
+
+ return getDrmManagerService()->openDecryptSession(
+ uniqueId, fd, offset, length, mime);
}
sp<DecryptHandle> DrmManagerClientImpl::openDecryptSession(
- int uniqueId, const char* uri) {
+ int uniqueId, const char* uri, const char* mime) {
+
DecryptHandle* handle = NULL;
if (NULL != uri) {
- handle = getDrmManagerService()->openDecryptSession(uniqueId, uri);
+ handle = getDrmManagerService()->openDecryptSession(uniqueId, uri, mime);
}
return handle;
}
diff --git a/drm/libdrmframework/include/DrmManager.h b/drm/libdrmframework/include/DrmManager.h
index ac2b946..c9167d4 100644
--- a/drm/libdrmframework/include/DrmManager.h
+++ b/drm/libdrmframework/include/DrmManager.h
@@ -111,9 +111,10 @@
status_t getAllSupportInfo(int uniqueId, int* length, DrmSupportInfo** drmSupportInfoArray);
- DecryptHandle* openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length);
+ DecryptHandle* openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime);
- DecryptHandle* openDecryptSession(int uniqueId, const char* uri);
+ DecryptHandle* openDecryptSession(int uniqueId, const char* uri, const char* mime);
status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle);
diff --git a/drm/libdrmframework/include/DrmManagerClientImpl.h b/drm/libdrmframework/include/DrmManagerClientImpl.h
index e3338d9..2aa493f 100644
--- a/drm/libdrmframework/include/DrmManagerClientImpl.h
+++ b/drm/libdrmframework/include/DrmManagerClientImpl.h
@@ -300,20 +300,24 @@
* @param[in] fd File descriptor of the protected content to be decrypted
* @param[in] offset Start position of the content
* @param[in] length The length of the protected content
+ * @param[in] mime The mime type of the protected content if it is not NULL or empty
* @return
* Handle for the decryption session
*/
- sp<DecryptHandle> openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length);
+ sp<DecryptHandle> openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length, const char* mime);
/**
* Open the decrypt session to decrypt the given protected content
*
* @param[in] uniqueId Unique identifier for a session
* @param[in] uri Path of the protected content to be decrypted
+ * @param[in] mime The mime type of the protected content if it is not NULL or empty
* @return
* Handle for the decryption session
*/
- sp<DecryptHandle> openDecryptSession(int uniqueId, const char* uri);
+ sp<DecryptHandle> openDecryptSession(
+ int uniqueId, const char* uri, const char* mime);
/**
* Close the decrypt session for the given handle
diff --git a/drm/libdrmframework/include/DrmManagerService.h b/drm/libdrmframework/include/DrmManagerService.h
index 9cb5804..1a8c2ae 100644
--- a/drm/libdrmframework/include/DrmManagerService.h
+++ b/drm/libdrmframework/include/DrmManagerService.h
@@ -98,9 +98,11 @@
status_t getAllSupportInfo(int uniqueId, int* length, DrmSupportInfo** drmSupportInfoArray);
- DecryptHandle* openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length);
+ DecryptHandle* openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length, const char *mime);
- DecryptHandle* openDecryptSession(int uniqueId, const char* uri);
+ DecryptHandle* openDecryptSession(
+ int uniqueId, const char* uri, const char* mime);
status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle);
diff --git a/drm/libdrmframework/include/IDrmManagerService.h b/drm/libdrmframework/include/IDrmManagerService.h
index b9618bb..a7d21c5 100644
--- a/drm/libdrmframework/include/IDrmManagerService.h
+++ b/drm/libdrmframework/include/IDrmManagerService.h
@@ -139,9 +139,12 @@
virtual status_t getAllSupportInfo(
int uniqueId, int* length, DrmSupportInfo** drmSupportInfoArray) = 0;
- virtual DecryptHandle* openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length) = 0;
+ virtual DecryptHandle* openDecryptSession(
+ int uniqueId, int fd, off64_t offset,
+ off64_t length, const char* mime) = 0;
- virtual DecryptHandle* openDecryptSession(int uniqueId, const char* uri) = 0;
+ virtual DecryptHandle* openDecryptSession(
+ int uniqueId, const char* uri, const char* mime) = 0;
virtual status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle) = 0;
@@ -222,9 +225,12 @@
virtual status_t getAllSupportInfo(
int uniqueId, int* length, DrmSupportInfo** drmSupportInfoArray);
- virtual DecryptHandle* openDecryptSession(int uniqueId, int fd, off64_t offset, off64_t length);
+ virtual DecryptHandle* openDecryptSession(
+ int uniqueId, int fd, off64_t offset, off64_t length,
+ const char* mime);
- virtual DecryptHandle* openDecryptSession(int uniqueId, const char* uri);
+ virtual DecryptHandle* openDecryptSession(
+ int uniqueId, const char* uri, const char* mime);
virtual status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle);
diff --git a/drm/libdrmframework/plugins/common/include/DrmEngineBase.h b/drm/libdrmframework/plugins/common/include/DrmEngineBase.h
index 4a5afcf..08f6e6d 100644
--- a/drm/libdrmframework/plugins/common/include/DrmEngineBase.h
+++ b/drm/libdrmframework/plugins/common/include/DrmEngineBase.h
@@ -80,10 +80,12 @@
DrmSupportInfo* getSupportInfo(int uniqueId);
status_t openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, int fd, off64_t offset, off64_t length);
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length, const char* mime);
status_t openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, const char* uri);
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri, const char* mime);
status_t closeDecryptSession(int uniqueId, DecryptHandle* decryptHandle);
@@ -375,7 +377,29 @@
* DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
*/
virtual status_t onOpenDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, int fd, off64_t offset, off64_t length) = 0;
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length) = 0;
+
+ /**
+ * Open the decrypt session to decrypt the given protected content
+ *
+ * @param[in] uniqueId Unique identifier for a session
+ * @param[in] decryptHandle Handle for the current decryption session
+ * @param[in] fd File descriptor of the protected content to be decrypted
+ * @param[in] offset Start position of the content
+ * @param[in] length The length of the protected content
+ * @param[in] mime Mime type of the protected content
+ * drm plugin may do some optimization since the mime type is known.
+ * @return
+ * DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
+ */
+ virtual status_t onOpenDecryptSession(
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length,
+ const char* mime) {
+
+ return DRM_ERROR_CANNOT_HANDLE;
+ }
/**
* Open the decrypt session to decrypt the given protected content
@@ -387,7 +411,26 @@
* DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
*/
virtual status_t onOpenDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, const char* uri) = 0;
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri) = 0;
+
+ /**
+ * Open the decrypt session to decrypt the given protected content
+ *
+ * @param[in] uniqueId Unique identifier for a session
+ * @param[in] decryptHandle Handle for the current decryption session
+ * @param[in] uri Path of the protected content to be decrypted
+ * @param[in] mime Mime type of the protected content. The corresponding
+ * drm plugin may do some optimization since the mime type is known.
+ * @return
+ * DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
+ */
+ virtual status_t onOpenDecryptSession(
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri, const char* mime) {
+
+ return DRM_ERROR_CANNOT_HANDLE;
+ }
/**
* Close the decrypt session for the given handle
diff --git a/drm/libdrmframework/plugins/common/include/IDrmEngine.h b/drm/libdrmframework/plugins/common/include/IDrmEngine.h
index 77460f6..dcf5977 100644
--- a/drm/libdrmframework/plugins/common/include/IDrmEngine.h
+++ b/drm/libdrmframework/plugins/common/include/IDrmEngine.h
@@ -320,11 +320,14 @@
* @param[in] fd File descriptor of the protected content to be decrypted
* @param[in] offset Start position of the content
* @param[in] length The length of the protected content
+ * @param[in] mime Mime type of the protected content if it is
+ * not NULL or empty
* @return
* DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
*/
virtual status_t openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, int fd, off64_t offset, off64_t length) = 0;
+ int uniqueId, DecryptHandle* decryptHandle,
+ int fd, off64_t offset, off64_t length, const char* mime) = 0;
/**
* Open the decrypt session to decrypt the given protected content
@@ -332,11 +335,14 @@
* @param[in] uniqueId Unique identifier for a session
* @param[in] decryptHandle Handle for the current decryption session
* @param[in] uri Path of the protected content to be decrypted
+ * @param[in] mime Mime type of the protected content if it is
+ * not NULL or empty
* @return
* DRM_ERROR_CANNOT_HANDLE for failure and DRM_NO_ERROR for success
*/
virtual status_t openDecryptSession(
- int uniqueId, DecryptHandle* decryptHandle, const char* uri) = 0;
+ int uniqueId, DecryptHandle* decryptHandle,
+ const char* uri, const char* mime) = 0;
/**
* Close the decrypt session for the given handle
diff --git a/include/camera/ICamera.h b/include/camera/ICamera.h
index 400d7f4..3d18837 100644
--- a/include/camera/ICamera.h
+++ b/include/camera/ICamera.h
@@ -20,15 +20,15 @@
#include <utils/RefBase.h>
#include <binder/IInterface.h>
#include <binder/Parcel.h>
-#include <surfaceflinger/Surface.h>
#include <binder/IMemory.h>
#include <utils/String8.h>
#include <camera/Camera.h>
-#include <gui/ISurfaceTexture.h>
namespace android {
class ICameraClient;
+class ISurfaceTexture;
+class Surface;
class ICamera: public IInterface
{
diff --git a/include/common_time/ICommonClock.h b/include/common_time/ICommonClock.h
new file mode 100644
index 0000000..d7073f1
--- /dev/null
+++ b/include/common_time/ICommonClock.h
@@ -0,0 +1,108 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_ICOMMONCLOCK_H
+#define ANDROID_ICOMMONCLOCK_H
+
+#include <stdint.h>
+#include <linux/socket.h>
+
+#include <binder/IInterface.h>
+#include <binder/IServiceManager.h>
+
+namespace android {
+
+class ICommonClockListener : public IInterface {
+ public:
+ DECLARE_META_INTERFACE(CommonClockListener);
+
+ virtual void onTimelineChanged(uint64_t timelineID) = 0;
+};
+
+class BnCommonClockListener : public BnInterface<ICommonClockListener> {
+ public:
+ virtual status_t onTransact(uint32_t code, const Parcel& data,
+ Parcel* reply, uint32_t flags = 0);
+};
+
+class ICommonClock : public IInterface {
+ public:
+ DECLARE_META_INTERFACE(CommonClock);
+
+ // Name of the ICommonClock service registered with the service manager.
+ static const String16 kServiceName;
+
+ // a reserved invalid timeline ID
+ static const uint64_t kInvalidTimelineID;
+
+ // a reserved invalid error estimate
+ static const int32_t kErrorEstimateUnknown;
+
+ enum State {
+ // the device just came up and is trying to discover the master
+ STATE_INITIAL,
+
+ // the device is a client of a master
+ STATE_CLIENT,
+
+ // the device is acting as master
+ STATE_MASTER,
+
+ // the device has lost contact with its master and needs to participate
+ // in the election of a new master
+ STATE_RONIN,
+
+ // the device is waiting for announcement of the newly elected master
+ STATE_WAIT_FOR_ELECTION,
+ };
+
+ virtual status_t isCommonTimeValid(bool* valid, uint32_t* timelineID) = 0;
+ virtual status_t commonTimeToLocalTime(int64_t commonTime,
+ int64_t* localTime) = 0;
+ virtual status_t localTimeToCommonTime(int64_t localTime,
+ int64_t* commonTime) = 0;
+ virtual status_t getCommonTime(int64_t* commonTime) = 0;
+ virtual status_t getCommonFreq(uint64_t* freq) = 0;
+ virtual status_t getLocalTime(int64_t* localTime) = 0;
+ virtual status_t getLocalFreq(uint64_t* freq) = 0;
+ virtual status_t getEstimatedError(int32_t* estimate) = 0;
+ virtual status_t getTimelineID(uint64_t* id) = 0;
+ virtual status_t getState(State* state) = 0;
+ virtual status_t getMasterAddr(struct sockaddr_storage* addr) = 0;
+
+ virtual status_t registerListener(
+ const sp<ICommonClockListener>& listener) = 0;
+ virtual status_t unregisterListener(
+ const sp<ICommonClockListener>& listener) = 0;
+
+ // Simple helper to make it easier to connect to the CommonClock service.
+ static inline sp<ICommonClock> getInstance() {
+ sp<IBinder> binder = defaultServiceManager()->checkService(
+ ICommonClock::kServiceName);
+ sp<ICommonClock> clk = interface_cast<ICommonClock>(binder);
+ return clk;
+ }
+};
+
+class BnCommonClock : public BnInterface<ICommonClock> {
+ public:
+ virtual status_t onTransact(uint32_t code, const Parcel& data,
+ Parcel* reply, uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif // ANDROID_ICOMMONCLOCK_H
diff --git a/include/common_time/ICommonTimeConfig.h b/include/common_time/ICommonTimeConfig.h
new file mode 100644
index 0000000..497b666
--- /dev/null
+++ b/include/common_time/ICommonTimeConfig.h
@@ -0,0 +1,73 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_ICOMMONTIMECONFIG_H
+#define ANDROID_ICOMMONTIMECONFIG_H
+
+#include <stdint.h>
+#include <linux/socket.h>
+
+#include <binder/IInterface.h>
+#include <binder/IServiceManager.h>
+
+namespace android {
+
+class String16;
+
+class ICommonTimeConfig : public IInterface {
+ public:
+ DECLARE_META_INTERFACE(CommonTimeConfig);
+
+ // Name of the ICommonTimeConfig service registered with the service
+ // manager.
+ static const String16 kServiceName;
+
+ virtual status_t getMasterElectionPriority(uint8_t *priority) = 0;
+ virtual status_t setMasterElectionPriority(uint8_t priority) = 0;
+ virtual status_t getMasterElectionEndpoint(struct sockaddr_storage *addr) = 0;
+ virtual status_t setMasterElectionEndpoint(const struct sockaddr_storage *addr) = 0;
+ virtual status_t getMasterElectionGroupId(uint64_t *id) = 0;
+ virtual status_t setMasterElectionGroupId(uint64_t id) = 0;
+ virtual status_t getInterfaceBinding(String16& ifaceName) = 0;
+ virtual status_t setInterfaceBinding(const String16& ifaceName) = 0;
+ virtual status_t getMasterAnnounceInterval(int *interval) = 0;
+ virtual status_t setMasterAnnounceInterval(int interval) = 0;
+ virtual status_t getClientSyncInterval(int *interval) = 0;
+ virtual status_t setClientSyncInterval(int interval) = 0;
+ virtual status_t getPanicThreshold(int *threshold) = 0;
+ virtual status_t setPanicThreshold(int threshold) = 0;
+ virtual status_t getAutoDisable(bool *autoDisable) = 0;
+ virtual status_t setAutoDisable(bool autoDisable) = 0;
+ virtual status_t forceNetworklessMasterMode() = 0;
+
+ // Simple helper to make it easier to connect to the CommonTimeConfig service.
+ static inline sp<ICommonTimeConfig> getInstance() {
+ sp<IBinder> binder = defaultServiceManager()->checkService(
+ ICommonTimeConfig::kServiceName);
+ sp<ICommonTimeConfig> clk = interface_cast<ICommonTimeConfig>(binder);
+ return clk;
+ }
+};
+
+class BnCommonTimeConfig : public BnInterface<ICommonTimeConfig> {
+ public:
+ virtual status_t onTransact(uint32_t code, const Parcel& data,
+ Parcel* reply, uint32_t flags = 0);
+};
+
+}; // namespace android
+
+#endif // ANDROID_ICOMMONTIMECONFIG_H
diff --git a/include/common_time/cc_helper.h b/include/common_time/cc_helper.h
new file mode 100644
index 0000000..8c4d5c0
--- /dev/null
+++ b/include/common_time/cc_helper.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __CC_HELPER_H__
+#define __CC_HELPER_H__
+
+#include <stdint.h>
+#include <common_time/ICommonClock.h>
+#include <utils/threads.h>
+
+namespace android {
+
+// CCHelper is a simple wrapper class to help with centralizing access to the
+// Common Clock service and implementing lifetime managment, as well as to
+// implement a simple policy of making a basic attempt to reconnect to the
+// common clock service when things go wrong.
+//
+// On platforms which run the native common_time service in auto-disable mode,
+// the service will go into networkless mode whenever it has no active clients.
+// It tracks active clients using registered CommonClockListeners (the callback
+// interface for onTimelineChanged) since this provides a convienent death
+// handler notification for when the service's clients die unexpectedly. This
+// means that users of the common time service should really always have a
+// CommonClockListener, unless they know that the time service is not running in
+// auto disabled mode, or that there is at least one other registered listener
+// active in the system. The CCHelper makes this a little easier by sharing a
+// ref counted ICommonClock interface across all clients and automatically
+// registering and unregistering a listener whenever there are CCHelper
+// instances active in the process.
+class CCHelper {
+ public:
+ CCHelper();
+ ~CCHelper();
+
+ status_t isCommonTimeValid(bool* valid, uint32_t* timelineID);
+ status_t commonTimeToLocalTime(int64_t commonTime, int64_t* localTime);
+ status_t localTimeToCommonTime(int64_t localTime, int64_t* commonTime);
+ status_t getCommonTime(int64_t* commonTime);
+ status_t getCommonFreq(uint64_t* freq);
+ status_t getLocalTime(int64_t* localTime);
+ status_t getLocalFreq(uint64_t* freq);
+
+ private:
+ class CommonClockListener : public BnCommonClockListener {
+ public:
+ void onTimelineChanged(uint64_t timelineID);
+ };
+
+ static bool verifyClock_l();
+
+ static Mutex lock_;
+ static sp<ICommonClock> common_clock_;
+ static sp<ICommonClockListener> common_clock_listener_;
+ static uint32_t ref_count_;
+};
+
+
+} // namespace android
+#endif // __CC_HELPER_H__
diff --git a/include/common_time/local_clock.h b/include/common_time/local_clock.h
new file mode 100644
index 0000000..845d1c2
--- /dev/null
+++ b/include/common_time/local_clock.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#ifndef __LOCAL_CLOCK_H__
+#define __LOCAL_CLOCK_H__
+
+#include <stdint.h>
+
+#include <hardware/local_time_hal.h>
+#include <utils/Errors.h>
+#include <utils/threads.h>
+
+namespace android {
+
+class LocalClock {
+ public:
+ LocalClock();
+
+ bool initCheck();
+
+ int64_t getLocalTime();
+ uint64_t getLocalFreq();
+ status_t setLocalSlew(int16_t rate);
+ int32_t getDebugLog(struct local_time_debug_event* records,
+ int max_records);
+
+ private:
+ static Mutex dev_lock_;
+ static local_time_hw_device_t* dev_;
+};
+
+} // namespace android
+#endif // __LOCAL_CLOCK_H__
diff --git a/include/drm/DrmManagerClient.h b/include/drm/DrmManagerClient.h
index b8fe46d..c47bbfb 100644
--- a/include/drm/DrmManagerClient.h
+++ b/include/drm/DrmManagerClient.h
@@ -66,19 +66,21 @@
* @param[in] fd File descriptor of the protected content to be decrypted
* @param[in] offset Start position of the content
* @param[in] length The length of the protected content
+ * @param[in] mime Mime type of the protected content if it is not NULL or empty
* @return
* Handle for the decryption session
*/
- sp<DecryptHandle> openDecryptSession(int fd, off64_t offset, off64_t length);
+ sp<DecryptHandle> openDecryptSession(int fd, off64_t offset, off64_t length, const char* mime);
/**
* Open the decrypt session to decrypt the given protected content
*
* @param[in] uri Path of the protected content to be decrypted
+ * @param[in] mime Mime type of the protected content if it is not NULL or empty
* @return
* Handle for the decryption session
*/
- sp<DecryptHandle> openDecryptSession(const char* uri);
+ sp<DecryptHandle> openDecryptSession(const char* uri, const char* mime);
/**
* Close the decrypt session for the given handle
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
index 1417416..02dfc1b 100644
--- a/include/media/AudioEffect.h
+++ b/include/media/AudioEffect.h
@@ -108,7 +108,8 @@
* Returned value
* *descriptor updated with effect descriptor
*/
- static status_t getEffectDescriptor(effect_uuid_t *uuid, effect_descriptor_t *descriptor);
+ static status_t getEffectDescriptor(const effect_uuid_t *uuid,
+ effect_descriptor_t *descriptor) /*const*/;
/*
@@ -226,8 +227,8 @@
AudioEffect(const effect_uuid_t *type,
const effect_uuid_t *uuid = NULL,
int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0,
audio_io_handle_t io = 0
);
@@ -238,8 +239,8 @@
AudioEffect(const char *typeStr,
const char *uuidStr = NULL,
int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0,
audio_io_handle_t io = 0
);
@@ -260,8 +261,8 @@
status_t set(const effect_uuid_t *type,
const effect_uuid_t *uuid = NULL,
int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0,
audio_io_handle_t io = 0
);
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
index 605680a..437a89c 100644
--- a/include/media/AudioRecord.h
+++ b/include/media/AudioRecord.h
@@ -22,7 +22,6 @@
#include <media/IAudioFlinger.h>
#include <media/IAudioRecord.h>
-#include <media/AudioTrack.h>
#include <utils/RefBase.h>
#include <utils/Errors.h>
@@ -34,6 +33,8 @@
namespace android {
+class audio_track_cblk_t;
+
// ----------------------------------------------------------------------------
class AudioRecord
@@ -67,7 +68,7 @@
};
uint32_t flags;
int channelCount;
- int format;
+ audio_format_t format;
size_t frameCount;
size_t size;
union {
@@ -111,7 +112,7 @@
static status_t getMinFrameCount(int* frameCount,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelCount);
/* Constructs an uninitialized AudioRecord. No connection with
@@ -148,14 +149,14 @@
RECORD_IIR_ENABLE = AUDIO_IN_ACOUSTICS_TX_IIR_ENABLE,
};
- AudioRecord(int inputSource,
+ AudioRecord(audio_source_t inputSource,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
int sessionId = 0);
@@ -174,14 +175,14 @@
* - NO_INIT: audio server or audio hardware not initialized
* - PERMISSION_DENIED: recording is not allowed for the requesting process
* */
- status_t set(int inputSource = 0,
+ status_t set(audio_source_t inputSource = AUDIO_SOURCE_DEFAULT,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channelMask = AUDIO_CHANNEL_IN_MONO,
int frameCount = 0,
uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
bool threadCanCallJava = false,
int sessionId = 0);
@@ -202,12 +203,12 @@
/* getters, see constructor */
- int format() const;
+ audio_format_t format() const;
int channelCount() const;
int channels() const;
uint32_t frameCount() const;
- int frameSize() const;
- int inputSource() const;
+ size_t frameSize() const;
+ audio_source_t inputSource() const;
/* After it's created the track is not active. Call start() to
@@ -340,15 +341,14 @@
private:
friend class AudioRecord;
virtual bool threadLoop();
- virtual status_t readyToRun() { return NO_ERROR; }
+ virtual status_t readyToRun();
virtual void onFirstRef() {}
AudioRecord& mReceiver;
- Mutex mLock;
};
bool processAudioBuffer(const sp<ClientRecordThread>& thread);
status_t openRecord_l(uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -359,15 +359,16 @@
sp<IAudioRecord> mAudioRecord;
sp<IMemory> mCblkMemory;
sp<ClientRecordThread> mClientRecordThread;
+ status_t mReadyToRun;
Mutex mLock;
+ Condition mCondition;
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
- uint32_t mFormat;
+ audio_format_t mFormat;
uint8_t mChannelCount;
- uint8_t mInputSource;
- uint8_t mReserved[2];
+ audio_source_t mInputSource;
status_t mStatus;
uint32_t mLatency;
@@ -385,6 +386,8 @@
uint32_t mChannelMask;
audio_io_handle_t mInput;
int mSessionId;
+ int mPreviousPriority; // before start()
+ int mPreviousSchedulingGroup;
};
}; // namespace android
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
index 6a15f6e..1916ac5 100644
--- a/include/media/AudioSystem.h
+++ b/include/media/AudioSystem.h
@@ -55,19 +55,21 @@
static status_t getMasterMute(bool* mute);
// set/get stream volume on specified output
- static status_t setStreamVolume(int stream, float value, int output);
- static status_t getStreamVolume(int stream, float* volume, int output);
+ static status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output);
+ static status_t getStreamVolume(audio_stream_type_t stream, float* volume,
+ audio_io_handle_t output);
// mute/unmute stream
- static status_t setStreamMute(int stream, bool mute);
- static status_t getStreamMute(int stream, bool* mute);
+ static status_t setStreamMute(audio_stream_type_t stream, bool mute);
+ static status_t getStreamMute(audio_stream_type_t stream, bool* mute);
- // set audio mode in audio hardware (see audio_mode_t)
- static status_t setMode(int mode);
+ // set audio mode in audio hardware
+ static status_t setMode(audio_mode_t mode);
// returns true in *state if tracks are active on the specified stream or has been active
// in the past inPastMs milliseconds
- static status_t isStreamActive(int stream, bool *state, uint32_t inPastMs = 0);
+ static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs = 0);
// set/get audio hardware parameters. The function accepts a list of parameters
// key value pairs in the form: key1=value1;key2=value2;...
@@ -83,13 +85,19 @@
static float linearToLog(int volume);
static int logToLinear(float volume);
+ static status_t getOutputSamplingRate(int* samplingRate, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ static status_t getOutputFrameCount(int* frameCount, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+ static status_t getOutputLatency(uint32_t* latency, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
+
+ // DEPRECATED
static status_t getOutputSamplingRate(int* samplingRate, int stream = AUDIO_STREAM_DEFAULT);
+
+ // DEPRECATED
static status_t getOutputFrameCount(int* frameCount, int stream = AUDIO_STREAM_DEFAULT);
- static status_t getOutputLatency(uint32_t* latency, int stream = AUDIO_STREAM_DEFAULT);
- static bool routedToA2dpOutput(int streamType);
+ static bool routedToA2dpOutput(audio_stream_type_t streamType);
- static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
+ static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
size_t* buffSize);
static status_t setVoiceVolume(float volume);
@@ -103,7 +111,7 @@
// - BAD_VALUE: invalid parameter
// NOTE: this feature is not supported on all hardware platforms and it is
// necessary to check returned status before using the returned values.
- static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = AUDIO_STREAM_DEFAULT);
+ static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT);
static unsigned int getInputFramesLost(audio_io_handle_t ioHandle);
@@ -123,12 +131,12 @@
NUM_CONFIG_EVENTS
};
- // audio output descritor used to cache output configurations in client process to avoid frequent calls
+ // audio output descriptor used to cache output configurations in client process to avoid frequent calls
// through IAudioFlinger
class OutputDescriptor {
public:
OutputDescriptor()
- : samplingRate(0), format(0), channels(0), frameCount(0), latency(0) {}
+ : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channels(0), frameCount(0), latency(0) {}
uint32_t samplingRate;
int32_t format;
@@ -142,13 +150,12 @@
//
static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address);
static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address);
- static status_t setPhoneState(int state);
- static status_t setRingerMode(uint32_t mode, uint32_t mask);
+ static status_t setPhoneState(audio_mode_t state);
static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
static audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = AUDIO_CHANNEL_OUT_STEREO,
audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT);
static status_t startOutput(audio_io_handle_t output,
@@ -158,9 +165,9 @@
audio_stream_type_t stream,
int session = 0);
static void releaseOutput(audio_io_handle_t output);
- static audio_io_handle_t getInput(int inputSource,
+ static audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = AUDIO_CHANNEL_IN_MONO,
audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0,
int sessionId = 0);
@@ -170,8 +177,12 @@
static status_t initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax);
- static status_t setStreamVolumeIndex(audio_stream_type_t stream, int index);
- static status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index);
+ static status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ static status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
static uint32_t getStrategyForStream(audio_stream_type_t stream);
static uint32_t getDevicesForStream(audio_stream_type_t stream);
@@ -208,7 +219,7 @@
// indicate a change in the configuration of an output or input: keeps the cached
// values for output/input parameters upto date in client process
- virtual void ioConfigChanged(int event, int ioHandle, void *param2);
+ virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, void *param2);
};
class AudioPolicyServiceClient: public IBinder::DeathRecipient
@@ -233,13 +244,13 @@
static size_t gInBuffSize;
// previous parameters for recording buffer size queries
static uint32_t gPrevInSamplingRate;
- static int gPrevInFormat;
+ static audio_format_t gPrevInFormat;
static int gPrevInChannelCount;
static sp<IAudioPolicyService> gAudioPolicyService;
// mapping between stream types and outputs
- static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap;
+ static DefaultKeyedVector<audio_stream_type_t, audio_io_handle_t> gStreamOutputMap;
// list of output descriptors containing cached parameters
// (sampling rate, framecount, channel count...)
static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs;
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index d1a8105..9f2bd3a 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -38,7 +38,7 @@
// ----------------------------------------------------------------------------
-class AudioTrack
+class AudioTrack : virtual public RefBase
{
public:
enum channel_index {
@@ -58,8 +58,8 @@
EVENT_BUFFER_END = 5 // Playback head is at the end of the buffer.
};
- /* Create Buffer on the stack and pass it to obtainBuffer()
- * and releaseBuffer().
+ /* Client should declare Buffer on the stack and pass address to obtainBuffer()
+ * and releaseBuffer(). See also callback_t for EVENT_MORE_DATA.
*/
class Buffer
@@ -68,34 +68,39 @@
enum {
MUTE = 0x00000001
};
- uint32_t flags;
- int format;
+ uint32_t flags; // 0 or MUTE
+ audio_format_t format; // but AUDIO_FORMAT_PCM_8_BIT -> AUDIO_FORMAT_PCM_16_BIT
+ // accessed directly by WebKit ANP callback
int channelCount; // will be removed in the future, do not use
- size_t frameCount;
- size_t size;
+
+ size_t frameCount; // number of sample frames corresponding to size;
+ // on input it is the number of frames desired,
+ // on output is the number of frames actually filled
+
+ size_t size; // input/output in byte units
union {
void* raw;
- short* i16;
- int8_t* i8;
+ short* i16; // signed 16-bit
+ int8_t* i8; // unsigned 8-bit, offset by 0x80
};
};
/* As a convenience, if a callback is supplied, a handler thread
* is automatically created with the appropriate priority. This thread
- * invokes the callback when a new buffer becomes availlable or an underrun condition occurs.
+ * invokes the callback when a new buffer becomes available or various conditions occur.
* Parameters:
*
* event: type of event notified (see enum AudioTrack::event_type).
* user: Pointer to context for use by the callback receiver.
* info: Pointer to optional parameter according to event type:
* - EVENT_MORE_DATA: pointer to AudioTrack::Buffer struct. The callback must not write
- * more bytes than indicated by 'size' field and update 'size' if less bytes are
- * written.
+ * more bytes than indicated by 'size' field and update 'size' if fewer bytes are
+ * written.
* - EVENT_UNDERRUN: unused.
* - EVENT_LOOP_END: pointer to an int indicating the number of loops remaining.
- * - EVENT_MARKER: pointer to an uin32_t containing the marker position in frames.
- * - EVENT_NEW_POS: pointer to an uin32_t containing the new position in frames.
+ * - EVENT_MARKER: pointer to an uint32_t containing the marker position in frames.
+ * - EVENT_NEW_POS: pointer to an uint32_t containing the new position in frames.
* - EVENT_BUFFER_END: unused.
*/
@@ -109,7 +114,7 @@
*/
static status_t getMinFrameCount(int* frameCount,
- int streamType =-1,
+ audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
uint32_t sampleRate = 0);
/* Constructs an uninitialized AudioTrack. No connection with
@@ -135,14 +140,27 @@
* flags: Reserved for future use.
* cbf: Callback function. If not null, this function is called periodically
* to request new PCM data.
+ * user: Context for use by the callback receiver.
* notificationFrames: The callback function is called each time notificationFrames PCM
- * frames have been comsumed from track input buffer.
- * user Context for use by the callback receiver.
+ * frames have been consumed from track input buffer.
+ * sessionId: Specific session ID, or zero to use default.
*/
- AudioTrack( int streamType,
+ AudioTrack( audio_stream_type_t streamType,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
+ int channelMask = 0,
+ int frameCount = 0,
+ uint32_t flags = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
+ int notificationFrames = 0,
+ int sessionId = 0);
+
+ // DEPRECATED
+ explicit AudioTrack( int streamType,
+ uint32_t sampleRate = 0,
+ int format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
uint32_t flags = 0,
@@ -152,46 +170,46 @@
int sessionId = 0);
/* Creates an audio track and registers it with AudioFlinger. With this constructor,
- * The PCM data to be rendered by AudioTrack is passed in a shared memory buffer
+ * the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
* identified by the argument sharedBuffer. This prototype is for static buffer playback.
- * PCM data must be present into memory before the AudioTrack is started.
- * The Write() and Flush() methods are not supported in this case.
- * It is recommented to pass a callback function to be notified of playback end by an
+ * PCM data must be present in memory before the AudioTrack is started.
+ * The write() and flush() methods are not supported in this case.
+ * It is recommended to pass a callback function to be notified of playback end by an
* EVENT_UNDERRUN event.
*/
- AudioTrack( int streamType,
+ AudioTrack( audio_stream_type_t streamType,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
const sp<IMemory>& sharedBuffer = 0,
uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
int sessionId = 0);
/* Terminates the AudioTrack and unregisters it from AudioFlinger.
- * Also destroys all resources assotiated with the AudioTrack.
+ * Also destroys all resources associated with the AudioTrack.
*/
~AudioTrack();
/* Initialize an uninitialized AudioTrack.
* Returned status (from utils/Errors.h) can be:
- * - NO_ERROR: successful intialization
- * - INVALID_OPERATION: AudioTrack is already intitialized
+ * - NO_ERROR: successful initialization
+ * - INVALID_OPERATION: AudioTrack is already initialized
* - BAD_VALUE: invalid parameter (channels, format, sampleRate...)
* - NO_INIT: audio server or audio hardware not initialized
* */
- status_t set(int streamType =-1,
+ status_t set(audio_stream_type_t streamType = AUDIO_STREAM_DEFAULT,
uint32_t sampleRate = 0,
- int format = 0,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
int channelMask = 0,
int frameCount = 0,
uint32_t flags = 0,
- callback_t cbf = 0,
- void* user = 0,
+ callback_t cbf = NULL,
+ void* user = NULL,
int notificationFrames = 0,
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
@@ -199,25 +217,30 @@
/* Result of constructing the AudioTrack. This must be checked
- * before using any AudioTrack API (except for set()), using
+ * before using any AudioTrack API (except for set()), because using
* an uninitialized AudioTrack produces undefined results.
* See set() method above for possible return codes.
*/
status_t initCheck() const;
- /* Returns this track's latency in milliseconds.
+ /* Returns this track's estimated latency in milliseconds.
* This includes the latency due to AudioTrack buffer size, AudioMixer (if any)
* and audio hardware driver.
*/
uint32_t latency() const;
- /* getters, see constructor */
+ /* getters, see constructors and set() */
- int streamType() const;
- int format() const;
+ audio_stream_type_t streamType() const;
+ audio_format_t format() const;
int channelCount() const;
uint32_t frameCount() const;
- int frameSize() const;
+
+ /* Return channelCount * (bit depth per channel / 8).
+ * channelCount is determined from channelMask, and bit depth comes from format.
+ */
+ size_t frameSize() const;
+
sp<IMemory>& sharedBuffer();
@@ -233,8 +256,8 @@
void stop();
bool stopped() const;
- /* flush a stopped track. All pending buffers are discarded.
- * This function has no effect if the track is not stoped.
+ /* Flush a stopped track. All pending buffers are discarded.
+ * This function has no effect if the track is not stopped.
*/
void flush();
@@ -244,29 +267,28 @@
*/
void pause();
- /* mute or unmutes this track.
- * While mutted, the callback, if set, is still called.
+ /* Mute or unmute this track.
+ * While muted, the callback, if set, is still called.
*/
void mute(bool);
bool muted() const;
-
- /* set volume for this track, mostly used for games' sound effects
- * left and right volumes. Levels must be <= 1.0.
+ /* Set volume for this track, mostly used for games' sound effects
+ * left and right volumes. Levels must be >= 0.0 and <= 1.0.
*/
status_t setVolume(float left, float right);
- void getVolume(float* left, float* right);
+ void getVolume(float* left, float* right) const;
- /* set the send level for this track. An auxiliary effect should be attached
- * to the track with attachEffect(). Level must be <= 1.0.
+ /* Set the send level for this track. An auxiliary effect should be attached
+ * to the track with attachEffect(). Level must be >= 0.0 and <= 1.0.
*/
status_t setAuxEffectSendLevel(float level);
- void getAuxEffectSendLevel(float* level);
+ void getAuxEffectSendLevel(float* level) const;
- /* set sample rate for this track, mostly used for games' sound effects
+ /* Set sample rate for this track, mostly used for games' sound effects
*/
status_t setSampleRate(int sampleRate);
- uint32_t getSampleRate();
+ uint32_t getSampleRate() const;
/* Enables looping and sets the start and end points of looping.
*
@@ -274,19 +296,17 @@
*
* loopStart: loop start expressed as the number of PCM frames played since AudioTrack start.
* loopEnd: loop end expressed as the number of PCM frames played since AudioTrack start.
- * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any pending or
- * active loop. loopCount = -1 means infinite looping.
+ * loopCount: number of loops to execute. Calling setLoop() with loopCount == 0 cancels any
+ * pending or active loop. loopCount = -1 means infinite looping.
*
* For proper operation the following condition must be respected:
* (loopEnd-loopStart) <= framecount()
*/
status_t setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount);
- status_t getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount);
-
- /* Sets marker position. When playback reaches the number of frames specified, a callback with event
- * type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker notification
- * callback.
+ /* Sets marker position. When playback reaches the number of frames specified, a callback with
+ * event type EVENT_MARKER is called. Calling setMarkerPosition with marker == 0 cancels marker
+ * notification callback.
* If the AudioTrack has been opened with no callback function associated, the operation will fail.
*
* Parameters:
@@ -298,13 +318,13 @@
* - INVALID_OPERATION: the AudioTrack has no callback installed.
*/
status_t setMarkerPosition(uint32_t marker);
- status_t getMarkerPosition(uint32_t *marker);
+ status_t getMarkerPosition(uint32_t *marker) const;
- /* Sets position update period. Every time the number of frames specified has been played,
- * a callback with event type EVENT_NEW_POS is called.
- * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
- * callback.
+ /* Sets position update period. Every time the number of frames specified has been played,
+ * a callback with event type EVENT_NEW_POS is called.
+ * Calling setPositionUpdatePeriod with updatePeriod == 0 cancels new position notification
+ * callback.
* If the AudioTrack has been opened with no callback function associated, the operation will fail.
*
* Parameters:
@@ -316,14 +336,13 @@
* - INVALID_OPERATION: the AudioTrack has no callback installed.
*/
status_t setPositionUpdatePeriod(uint32_t updatePeriod);
- status_t getPositionUpdatePeriod(uint32_t *updatePeriod);
-
+ status_t getPositionUpdatePeriod(uint32_t *updatePeriod) const;
/* Sets playback head position within AudioTrack buffer. The new position is specified
- * in number of frames.
+ * in number of frames.
* This method must be called with the AudioTrack in paused or stopped state.
- * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
- * Therefore using this method makes sense only when playing a "static" audio buffer
+ * Note that the actual position set is <position> modulo the AudioTrack buffer size in frames.
+ * Therefore using this method makes sense only when playing a "static" audio buffer
* as opposed to streaming.
* The getPosition() method on the other hand returns the total number of frames played since
* playback start.
@@ -335,12 +354,12 @@
* Returned status (from utils/Errors.h) can be:
* - NO_ERROR: successful operation
* - INVALID_OPERATION: the AudioTrack is not stopped.
- * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer
+ * - BAD_VALUE: The specified position is beyond the number of frames present in AudioTrack buffer
*/
status_t setPosition(uint32_t position);
status_t getPosition(uint32_t *position);
- /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
+ /* Forces AudioTrack buffer full condition. When playing a static buffer, this method avoids
* rewriting the buffer before restarting playback after a stop.
* This method must be called with the AudioTrack in paused or stopped state.
*
@@ -350,7 +369,7 @@
*/
status_t reload();
- /* returns a handle on the audio output used by this AudioTrack.
+ /* Returns a handle on the audio output used by this AudioTrack.
*
* Parameters:
* none.
@@ -360,18 +379,17 @@
*/
audio_io_handle_t getOutput();
- /* returns the unique ID associated to this track.
+ /* Returns the unique session ID associated with this track.
*
* Parameters:
* none.
*
* Returned value:
- * AudioTrack ID.
+ * AudioTrack session ID.
*/
- int getSessionId();
+ int getSessionId() const;
-
- /* Attach track auxiliary output to specified effect. Used effectId = 0
+ /* Attach track auxiliary output to specified effect. Use effectId = 0
* to detach track from effect.
*
* Parameters:
@@ -385,29 +403,41 @@
*/
status_t attachAuxEffect(int effectId);
- /* obtains a buffer of "frameCount" frames. The buffer must be
- * filled entirely. If the track is stopped, obtainBuffer() returns
- * STOPPED instead of NO_ERROR as long as there are buffers availlable,
+ /* Obtains a buffer of "frameCount" frames. The buffer must be
+ * filled entirely, and then released with releaseBuffer().
+ * If the track is stopped, obtainBuffer() returns
+ * STOPPED instead of NO_ERROR as long as there are buffers available,
* at which point NO_MORE_BUFFERS is returned.
* Buffers will be returned until the pool (buffercount())
* is exhausted, at which point obtainBuffer() will either block
* or return WOULD_BLOCK depending on the value of the "blocking"
* parameter.
+ *
+ * Interpretation of waitCount:
+ * +n limits wait time to n * WAIT_PERIOD_MS,
+ * -1 causes an (almost) infinite wait time,
+ * 0 non-blocking.
*/
enum {
- NO_MORE_BUFFERS = 0x80000001,
+ NO_MORE_BUFFERS = 0x80000001, // same name in AudioFlinger.h, ok to be different value
STOPPED = 1
};
status_t obtainBuffer(Buffer* audioBuffer, int32_t waitCount);
+
+ /* Release a filled buffer of "frameCount" frames for AudioFlinger to process. */
void releaseBuffer(Buffer* audioBuffer);
-
/* As a convenience we provide a write() interface to the audio buffer.
- * This is implemented on top of lockBuffer/unlockBuffer. For best
- * performance
- *
+ * This is implemented on top of obtainBuffer/releaseBuffer. For best
+ * performance use callbacks. Returns actual number of bytes written >= 0,
+ * or one of the following negative status codes:
+ * INVALID_OPERATION AudioTrack is configured for shared buffer mode
+ * BAD_VALUE size is invalid
+ * STOPPED AudioTrack was stopped during the write
+ * NO_MORE_BUFFERS when obtainBuffer() returns same
+ * or any other error code returned by IAudioTrack::start() or restoreTrack_l().
*/
ssize_t write(const void* buffer, size_t size);
@@ -416,7 +446,7 @@
*/
status_t dump(int fd, const Vector<String16>& args) const;
-private:
+protected:
/* copying audio tracks is not allowed */
AudioTrack(const AudioTrack& other);
AudioTrack& operator = (const AudioTrack& other);
@@ -432,13 +462,13 @@
virtual status_t readyToRun();
virtual void onFirstRef();
AudioTrack& mReceiver;
- Mutex mLock;
};
+ // body of AudioTrackThread::threadLoop()
bool processAudioBuffer(const sp<AudioTrackThread>& thread);
- status_t createTrack_l(int streamType,
+ status_t createTrack_l(audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -449,6 +479,7 @@
status_t setLoop_l(uint32_t loopStart, uint32_t loopEnd, int loopCount);
audio_io_handle_t getOutput_l();
status_t restoreTrack_l(audio_track_cblk_t*& cblk, bool fromStart);
+ bool stopped_l() const { return !mActive; }
sp<IAudioTrack> mAudioTrack;
sp<IMemory> mCblkMemory;
@@ -459,8 +490,8 @@
uint32_t mFrameCount;
audio_track_cblk_t* mCblk;
- uint32_t mFormat;
- uint8_t mStreamType;
+ audio_format_t mFormat;
+ audio_stream_type_t mStreamType;
uint8_t mChannelCount;
uint8_t mMuted;
uint8_t mReserved;
@@ -468,9 +499,9 @@
status_t mStatus;
uint32_t mLatency;
- volatile int32_t mActive;
+ bool mActive; // protected by mLock
- callback_t mCbf;
+ callback_t mCbf; // callback handler for events, or NULL
void* mUserData;
uint32_t mNotificationFramesReq; // requested number of frames between each notification callback
uint32_t mNotificationFramesAct; // actual number of frames between each notification callback
@@ -485,10 +516,35 @@
uint32_t mFlags;
int mSessionId;
int mAuxEffectId;
- Mutex mLock;
+ mutable Mutex mLock;
status_t mRestoreStatus;
+ bool mIsTimed;
+ int mPreviousPriority; // before start()
+ int mPreviousSchedulingGroup;
};
+class TimedAudioTrack : public AudioTrack
+{
+public:
+ TimedAudioTrack();
+
+ /* allocate a shared memory buffer that can be passed to queueTimedBuffer */
+ status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer);
+
+ /* queue a buffer obtained via allocateTimedBuffer for playback at the
+ given timestamp. PTS units a microseconds on the media time timeline.
+ The media time transform (set with setMediaTimeTransform) set by the
+ audio producer will handle converting from media time to local time
+ (perhaps going through the common time timeline in the case of
+ synchronized multiroom audio case) */
+ status_t queueTimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+
+ /* define a transform between media time and either common time or
+ local time */
+ enum TargetTimeline {LOCAL_TIME, COMMON_TIME};
+ status_t setMediaTimeTransform(const LinearTransform& xform,
+ TargetTimeline target);
+};
}; // namespace android
diff --git a/include/media/EffectsFactoryApi.h b/include/media/EffectsFactoryApi.h
index 8ae13cc..65c26f4 100644
--- a/include/media/EffectsFactoryApi.h
+++ b/include/media/EffectsFactoryApi.h
@@ -87,7 +87,7 @@
// Description: Creates an effect engine of the specified type and returns an
// effect control interface on this engine. The function will allocate the
// resources for an instance of the requested effect engine and return
-// a handler on the effect control interface.
+// a handle on the effect control interface.
//
// Input:
// pEffectUuid: pointer to the effect uuid.
@@ -109,23 +109,23 @@
// *pHandle: updated with the effect handle.
//
////////////////////////////////////////////////////////////////////////////////
-int EffectCreate(effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle);
+int EffectCreate(const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle);
////////////////////////////////////////////////////////////////////////////////
//
// Function: EffectRelease
//
-// Description: Releases the effect engine whose handler is given as argument.
+// Description: Releases the effect engine whose handle is given as argument.
// All resources allocated to this particular instance of the effect are
// released.
//
// Input:
-// handle: handler on the effect interface to be released.
+// handle: handle on the effect interface to be released.
//
// Output:
// returned value: 0 successful operation.
// -ENODEV factory failed to initialize
-// -EINVAL invalid interface handler
+// -EINVAL invalid interface handle
//
////////////////////////////////////////////////////////////////////////////////
int EffectRelease(effect_handle_t handle);
@@ -151,7 +151,7 @@
// *pDescriptor: updated with the effect descriptor.
//
////////////////////////////////////////////////////////////////////////////////
-int EffectGetDescriptor(effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor);
+int EffectGetDescriptor(const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor);
////////////////////////////////////////////////////////////////////////////////
//
@@ -167,7 +167,7 @@
// 1 if uuid is equal to EFFECT_UUID_NULL.
//
////////////////////////////////////////////////////////////////////////////////
-int EffectIsNullUuid(effect_uuid_t *pEffectUuid);
+int EffectIsNullUuid(const effect_uuid_t *pEffectUuid);
#if __cplusplus
} // extern "C"
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
index 9e3cb7f..7a2ada0 100644
--- a/include/media/IAudioFlinger.h
+++ b/include/media/IAudioFlinger.h
@@ -27,6 +27,7 @@
#include <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
#include <media/IAudioFlingerClient.h>
+#include <system/audio.h>
#include <hardware/audio_effect.h>
#include <media/IEffect.h>
#include <media/IEffectClient.h>
@@ -46,22 +47,23 @@
*/
virtual sp<IAudioTrack> createTrack(
pid_t pid,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
+ bool isTimed,
int *sessionId,
status_t *status) = 0;
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -71,11 +73,11 @@
/* query the audio hardware state. This state never changes,
* and therefore can be cached.
*/
- virtual uint32_t sampleRate(int output) const = 0;
- virtual int channelCount(int output) const = 0;
- virtual uint32_t format(int output) const = 0;
- virtual size_t frameCount(int output) const = 0;
- virtual uint32_t latency(int output) const = 0;
+ virtual uint32_t sampleRate(audio_io_handle_t output) const = 0;
+ virtual int channelCount(audio_io_handle_t output) const = 0;
+ virtual audio_format_t format(audio_io_handle_t output) const = 0;
+ virtual size_t frameCount(audio_io_handle_t output) const = 0;
+ virtual uint32_t latency(audio_io_handle_t output) const = 0;
/* set/get the audio hardware state. This will probably be used by
* the preference panel, mostly.
@@ -89,76 +91,83 @@
/* set/get stream type state. This will probably be used by
* the preference panel, mostly.
*/
- virtual status_t setStreamVolume(int stream, float value, int output) = 0;
- virtual status_t setStreamMute(int stream, bool muted) = 0;
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output) = 0;
+ virtual status_t setStreamMute(audio_stream_type_t stream, bool muted) = 0;
- virtual float streamVolume(int stream, int output) const = 0;
- virtual bool streamMute(int stream) const = 0;
+ virtual float streamVolume(audio_stream_type_t stream,
+ audio_io_handle_t output) const = 0;
+ virtual bool streamMute(audio_stream_type_t stream) const = 0;
// set audio mode
- virtual status_t setMode(int mode) = 0;
+ virtual status_t setMode(audio_mode_t mode) = 0;
// mic mute/state
virtual status_t setMicMute(bool state) = 0;
virtual bool getMicMute() const = 0;
- virtual status_t setParameters(int ioHandle, const String8& keyValuePairs) = 0;
- virtual String8 getParameters(int ioHandle, const String8& keys) = 0;
+ virtual status_t setParameters(audio_io_handle_t ioHandle,
+ const String8& keyValuePairs) = 0;
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const = 0;
// register a current process for audio output change notifications
virtual void registerClient(const sp<IAudioFlingerClient>& client) = 0;
// retrieve the audio recording buffer size
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const = 0;
- virtual int openOutput(uint32_t *pDevices,
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags) = 0;
- virtual int openDuplicateOutput(int output1, int output2) = 0;
- virtual status_t closeOutput(int output) = 0;
- virtual status_t suspendOutput(int output) = 0;
- virtual status_t restoreOutput(int output) = 0;
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2) = 0;
+ virtual status_t closeOutput(audio_io_handle_t output) = 0;
+ virtual status_t suspendOutput(audio_io_handle_t output) = 0;
+ virtual status_t restoreOutput(audio_io_handle_t output) = 0;
- virtual int openInput(uint32_t *pDevices,
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics) = 0;
- virtual status_t closeInput(int input) = 0;
+ audio_in_acoustics_t acoustics) = 0;
+ virtual status_t closeInput(audio_io_handle_t input) = 0;
- virtual status_t setStreamOutput(uint32_t stream, int output) = 0;
+ virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) = 0;
virtual status_t setVoiceVolume(float volume) = 0;
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) = 0;
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const = 0;
- virtual unsigned int getInputFramesLost(int ioHandle) = 0;
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const = 0;
virtual int newAudioSessionId() = 0;
virtual void acquireAudioSessionId(int audioSession) = 0;
virtual void releaseAudioSessionId(int audioSession) = 0;
- virtual status_t queryNumberEffects(uint32_t *numEffects) = 0;
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const = 0;
- virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) = 0;
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const = 0;
- virtual status_t getEffectDescriptor(effect_uuid_t *pEffectUUID, effect_descriptor_t *pDescriptor) = 0;
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pEffectUUID,
+ effect_descriptor_t *pDescriptor) const = 0;
virtual sp<IEffect> createEffect(pid_t pid,
effect_descriptor_t *pDesc,
const sp<IEffectClient>& client,
int32_t priority,
- int output,
+ audio_io_handle_t output,
int sessionId,
status_t *status,
int *id,
int *enabled) = 0;
- virtual status_t moveEffects(int session, int srcOutput, int dstOutput) = 0;
+ virtual status_t moveEffects(int session, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput) = 0;
};
diff --git a/include/media/IAudioFlingerClient.h b/include/media/IAudioFlingerClient.h
index aa0cdcf..f3b4df1 100644
--- a/include/media/IAudioFlingerClient.h
+++ b/include/media/IAudioFlingerClient.h
@@ -21,6 +21,7 @@
#include <utils/RefBase.h>
#include <binder/IInterface.h>
#include <utils/KeyedVector.h>
+#include <system/audio.h>
namespace android {
@@ -32,7 +33,7 @@
DECLARE_META_INTERFACE(AudioFlingerClient);
// Notifies a change of audio input/output configuration.
- virtual void ioConfigChanged(int event, int ioHandle, void *param2) = 0;
+ virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, void *param2) = 0;
};
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
index 9807cbe..4d88297 100644
--- a/include/media/IAudioPolicyService.h
+++ b/include/media/IAudioPolicyService.h
@@ -45,13 +45,12 @@
const char *device_address) = 0;
virtual audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device,
const char *device_address) = 0;
- virtual status_t setPhoneState(int state) = 0;
- virtual status_t setRingerMode(uint32_t mode, uint32_t mask) = 0;
+ virtual status_t setPhoneState(audio_mode_t state) = 0;
virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config) = 0;
virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage) = 0;
virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
audio_policy_output_flags_t flags = AUDIO_POLICY_OUTPUT_FLAG_INDIRECT) = 0;
virtual status_t startOutput(audio_io_handle_t output,
@@ -61,9 +60,9 @@
audio_stream_type_t stream,
int session = 0) = 0;
virtual void releaseOutput(audio_io_handle_t output) = 0;
- virtual audio_io_handle_t getInput(int inputSource,
+ virtual audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
audio_in_acoustics_t acoustics = (audio_in_acoustics_t)0,
int audioSession = 0) = 0;
@@ -73,8 +72,12 @@
virtual status_t initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax) = 0;
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index) = 0;
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index) = 0;
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device) = 0;
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device) = 0;
virtual uint32_t getStrategyForStream(audio_stream_type_t stream) = 0;
virtual uint32_t getDevicesForStream(audio_stream_type_t stream) = 0;
virtual audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc) = 0;
@@ -85,7 +88,7 @@
int id) = 0;
virtual status_t unregisterEffect(int id) = 0;
virtual status_t setEffectEnabled(int id, bool enabled) = 0;
- virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const = 0;
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const = 0;
virtual status_t queryDefaultPreProcessing(int audioSession,
effect_descriptor_t *descriptors,
uint32_t *count) = 0;
diff --git a/include/media/IAudioRecord.h b/include/media/IAudioRecord.h
index 46735de..7869020 100644
--- a/include/media/IAudioRecord.h
+++ b/include/media/IAudioRecord.h
@@ -37,8 +37,9 @@
/* After it's created the track is not active. Call start() to
* make it active. If set, the callback will start being called.
+ * tid identifies the client callback thread, or 0 if not needed.
*/
- virtual status_t start() = 0;
+ virtual status_t start(pid_t tid) = 0;
/* Stop a track. If set, the callback will cease being called and
* obtainBuffer will return an error. Buffers that are already released
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
index 47d530b..77f3e21 100644
--- a/include/media/IAudioTrack.h
+++ b/include/media/IAudioTrack.h
@@ -24,7 +24,7 @@
#include <utils/Errors.h>
#include <binder/IInterface.h>
#include <binder/IMemory.h>
-
+#include <utils/LinearTransform.h>
namespace android {
@@ -35,30 +35,34 @@
public:
DECLARE_META_INTERFACE(AudioTrack);
+ /* Get this track's control block */
+ virtual sp<IMemory> getCblk() const = 0;
+
/* After it's created the track is not active. Call start() to
* make it active. If set, the callback will start being called.
+ * tid identifies the client callback thread, or 0 if not needed.
*/
- virtual status_t start() = 0;
+ virtual status_t start(pid_t tid) = 0;
/* Stop a track. If set, the callback will cease being called and
* obtainBuffer will return an error. Buffers that are already released
- * will be processed, unless flush() is called.
+ * will continue to be processed, unless/until flush() is called.
*/
virtual void stop() = 0;
- /* flush a stopped track. All pending buffers are discarded.
- * This function has no effect if the track is not stoped.
+ /* Flush a stopped or paused track. All pending/released buffers are discarded.
+ * This function has no effect if the track is not stopped or paused.
*/
virtual void flush() = 0;
- /* mute or unmutes this track.
- * While mutted, the callback, if set, is still called.
+ /* Mute or unmute this track.
+ * While muted, the callback, if set, is still called.
*/
virtual void mute(bool) = 0;
/* Pause a track. If set, the callback will cease being called and
* obtainBuffer will return an error. Buffers that are already released
- * will be processed, unless flush() is called.
+ * will continue to be processed, unless/until flush() is called.
*/
virtual void pause() = 0;
@@ -67,8 +71,23 @@
*/
virtual status_t attachAuxEffect(int effectId) = 0;
- /* get this tracks control block */
- virtual sp<IMemory> getCblk() const = 0;
+
+ /* Allocate a shared memory buffer suitable for holding timed audio
+ samples */
+ virtual status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer) = 0;
+
+ /* Queue a buffer obtained via allocateTimedBuffer for playback at the given
+ timestamp */
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) = 0;
+
+ /* Define the linear transform that will be applied to the timestamps
+ given to queueTimedBuffer (which are expressed in media time).
+ Target specifies whether this transform converts media time to local time
+ or Tungsten time. The values for target are defined in AudioTrack.h */
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target) = 0;
};
// ----------------------------------------------------------------------------
diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
index e905903..6425886 100644
--- a/include/media/IMediaPlayer.h
+++ b/include/media/IMediaPlayer.h
@@ -21,6 +21,7 @@
#include <binder/IInterface.h>
#include <binder/Parcel.h>
#include <utils/KeyedVector.h>
+#include <system/audio.h>
namespace android {
@@ -51,7 +52,7 @@
virtual status_t getCurrentPosition(int* msec) = 0;
virtual status_t getDuration(int* msec) = 0;
virtual status_t reset() = 0;
- virtual status_t setAudioStreamType(int type) = 0;
+ virtual status_t setAudioStreamType(audio_stream_type_t type) = 0;
virtual status_t setLooping(int loop) = 0;
virtual status_t setVolume(float leftVolume, float rightVolume) = 0;
virtual status_t setAuxEffectSendLevel(float level) = 0;
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
index 93bbe13..4f46fcd 100644
--- a/include/media/IMediaPlayerService.h
+++ b/include/media/IMediaPlayerService.h
@@ -23,6 +23,7 @@
#include <utils/String8.h>
#include <binder/IInterface.h>
#include <binder/Parcel.h>
+#include <system/audio.h>
#include <media/IMediaPlayerClient.h>
#include <media/IMediaPlayer.h>
@@ -43,8 +44,8 @@
virtual sp<IMediaMetadataRetriever> createMetadataRetriever(pid_t pid) = 0;
virtual sp<IMediaPlayer> create(pid_t pid, const sp<IMediaPlayerClient>& client, int audioSessionId = 0) = 0;
- virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat) = 0;
- virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat) = 0;
+ virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat) = 0;
+ virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat) = 0;
virtual sp<IOMX> getOMX() = 0;
// codecs and audio devices usage tracking for the battery app
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
index c4cc947..a295e9a 100644
--- a/include/media/IOMX.h
+++ b/include/media/IOMX.h
@@ -42,10 +42,10 @@
typedef void *buffer_id;
typedef void *node_id;
- // Given the calling process' pid, returns true iff
+ // Given a node_id and the calling process' pid, returns true iff
// the implementation of the OMX interface lives in the same
// process.
- virtual bool livesLocally(pid_t pid) = 0;
+ virtual bool livesLocally(node_id node, pid_t pid) = 0;
struct ComponentInfo {
String8 mName;
diff --git a/include/media/JetPlayer.h b/include/media/JetPlayer.h
index 16764a9..491a950 100644
--- a/include/media/JetPlayer.h
+++ b/include/media/JetPlayer.h
@@ -65,7 +65,6 @@
private:
- static int renderThread(void*);
int render();
void fireUpdateOnStatusChange();
void fireEventsFromJetQueue();
@@ -95,8 +94,30 @@
S_JET_STATUS mJetStatus;
S_JET_STATUS mPreviousJetStatus;
- char mJetFilePath[256];
+ char mJetFilePath[PATH_MAX];
+ class JetPlayerThread : public Thread {
+ public:
+ JetPlayerThread(JetPlayer *player) : mPlayer(player) {
+ }
+
+ protected:
+ virtual ~JetPlayerThread() {}
+
+ private:
+ JetPlayer *mPlayer;
+
+ bool threadLoop() {
+ int result;
+ result = mPlayer->render();
+ return false;
+ }
+
+ JetPlayerThread(const JetPlayerThread &);
+ JetPlayerThread &operator=(const JetPlayerThread &);
+ };
+
+ sp<JetPlayerThread> mThread;
}; // end class JetPlayer
diff --git a/include/media/MediaPlayerInterface.h b/include/media/MediaPlayerInterface.h
index 80f43a3..23a3e49 100644
--- a/include/media/MediaPlayerInterface.h
+++ b/include/media/MediaPlayerInterface.h
@@ -46,6 +46,9 @@
// The shared library with the test player is passed passed as an
// argument to the 'test:' url in the setDataSource call.
TEST_PLAYER = 5,
+
+ AAH_RX_PLAYER = 100,
+ AAH_TX_PLAYER = 101,
};
@@ -85,7 +88,7 @@
// audio data.
virtual status_t open(
uint32_t sampleRate, int channelCount,
- int format=AUDIO_FORMAT_PCM_16_BIT,
+ audio_format_t format=AUDIO_FORMAT_PCM_16_BIT,
int bufferCount=DEFAULT_AUDIOSINK_BUFFERCOUNT,
AudioCallback cb = NULL,
void *cookie = NULL) = 0;
@@ -96,6 +99,8 @@
virtual void flush() = 0;
virtual void pause() = 0;
virtual void close() = 0;
+
+ virtual status_t setPlaybackRatePermille(int32_t rate) { return INVALID_OPERATION; }
};
MediaPlayerBase() : mCookie(0), mNotify(0) {}
@@ -199,7 +204,7 @@
virtual ~MediaPlayerHWInterface() {}
virtual bool hardwareOutput() { return true; }
virtual status_t setVolume(float leftVolume, float rightVolume) = 0;
- virtual status_t setAudioStreamType(int streamType) = 0;
+ virtual status_t setAudioStreamType(audio_stream_type_t streamType) = 0;
};
}; // namespace android
diff --git a/include/media/MemoryLeakTrackUtil.h b/include/media/MemoryLeakTrackUtil.h
index 290b748..ac0f6b2 100644
--- a/include/media/MemoryLeakTrackUtil.h
+++ b/include/media/MemoryLeakTrackUtil.h
@@ -19,7 +19,7 @@
namespace android {
/*
- * Dump the memory adddress of the calling process to the given fd.
+ * Dump the memory address of the calling process to the given fd.
*/
extern void dumpMemoryAddresses(int fd);
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
index 1ad1f26..df0c97e 100644
--- a/include/media/ToneGenerator.h
+++ b/include/media/ToneGenerator.h
@@ -151,10 +151,10 @@
NUM_SUP_TONES = LAST_SUP_TONE-FIRST_SUP_TONE+1
};
- ToneGenerator(int streamType, float volume, bool threadCanCallJava = false);
+ ToneGenerator(audio_stream_type_t streamType, float volume, bool threadCanCallJava = false);
~ToneGenerator();
- bool startTone(int toneType, int durationMs = -1);
+ bool startTone(tone_type toneType, int durationMs = -1);
void stopTone();
bool isInited() { return (mState == TONE_IDLE)?false:true;}
@@ -266,7 +266,7 @@
Mutex mCbkCondLock; // Mutex associated to mWaitCbkCond
Condition mWaitCbkCond; // condition enabling interface to wait for audio callback completion after a change is requested
float mVolume; // Volume applied to audio track
- int mStreamType; // Audio stream used for output
+ audio_stream_type_t mStreamType; // Audio stream used for output
unsigned int mProcessSize; // Size of audio blocks generated at a time by audioCallback() (in PCM frames).
bool initAudioTrack();
@@ -274,7 +274,7 @@
bool prepareWave();
unsigned int numWaves(unsigned int segmentIdx);
void clearWaveGens();
- int getToneForRegion(int toneType);
+ tone_type getToneForRegion(tone_type toneType);
// WaveGenerator generates a single sine wave
class WaveGenerator {
diff --git a/include/media/Visualizer.h b/include/media/Visualizer.h
index 5d2c874..60fa15b 100644
--- a/include/media/Visualizer.h
+++ b/include/media/Visualizer.h
@@ -66,8 +66,8 @@
* See AudioEffect constructor for details on parameters.
*/
Visualizer(int32_t priority = 0,
- effect_callback_t cbf = 0,
- void* user = 0,
+ effect_callback_t cbf = NULL,
+ void* user = NULL,
int sessionId = 0);
~Visualizer();
@@ -143,7 +143,7 @@
void periodicCapture();
uint32_t initCaptureSize();
- Mutex mLock;
+ Mutex mCaptureLock;
uint32_t mCaptureRate;
uint32_t mCaptureSize;
uint32_t mSampleRate;
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
index e6a0cc5..d0b87c8 100644
--- a/include/media/mediaplayer.h
+++ b/include/media/mediaplayer.h
@@ -145,6 +145,9 @@
// audio track, or zero for error (e.g. no audio track) or unknown.
KEY_PARAMETER_AUDIO_CHANNEL_COUNT = 1200, // get only
+ // Playback rate expressed in permille (1000 is normal speed), saved as int32_t, with negative
+ // values used for rewinding or reverse playback.
+ KEY_PARAMETER_PLAYBACK_RATE_PERMILLE = 1300, // set only
};
// ----------------------------------------------------------------------------
@@ -185,13 +188,13 @@
status_t getCurrentPosition(int *msec);
status_t getDuration(int *msec);
status_t reset();
- status_t setAudioStreamType(int type);
+ status_t setAudioStreamType(audio_stream_type_t type);
status_t setLooping(int loop);
bool isLooping();
status_t setVolume(float leftVolume, float rightVolume);
void notify(int msg, int ext1, int ext2, const Parcel *obj = NULL);
- static sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat);
- static sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat);
+ static sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
+ static sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
status_t invoke(const Parcel& request, Parcel *reply);
status_t setMetadataFilter(const Parcel& filter);
status_t getMetadata(bool update_only, bool apply_filter, Parcel *metadata);
@@ -223,7 +226,7 @@
int mSeekPosition;
bool mPrepareSync;
status_t mPrepareStatus;
- int mStreamType;
+ audio_stream_type_t mStreamType;
bool mLoop;
float mLeftVolume;
float mRightVolume;
diff --git a/include/media/stagefright/AACWriter.h b/include/media/stagefright/AACWriter.h
index fa3ab8a..49397ee 100644
--- a/include/media/stagefright/AACWriter.h
+++ b/include/media/stagefright/AACWriter.h
@@ -34,7 +34,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual bool reachedEOS();
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
protected:
@@ -66,6 +66,7 @@
bool exceedsFileSizeLimit();
bool exceedsFileDurationLimit();
status_t writeAdtsHeader(uint32_t frameLength);
+ status_t reset();
DISALLOW_EVIL_CONSTRUCTORS(AACWriter);
};
diff --git a/include/media/stagefright/ACodec.h b/include/media/stagefright/ACodec.h
index 3963d9c..70799a6 100644
--- a/include/media/stagefright/ACodec.h
+++ b/include/media/stagefright/ACodec.h
@@ -22,6 +22,7 @@
#include <android/native_window.h>
#include <media/IOMX.h>
#include <media/stagefright/foundation/AHierarchicalStateMachine.h>
+#include <OMX_Audio.h>
namespace android {
@@ -37,6 +38,9 @@
kWhatFlushCompleted = 'fcom',
kWhatOutputFormatChanged = 'outC',
kWhatError = 'erro',
+ kWhatComponentAllocated = 'cAll',
+ kWhatComponentConfigured = 'cCon',
+ kWhatBuffersAllocated = 'allc',
};
ACodec();
@@ -47,6 +51,10 @@
void signalResume();
void initiateShutdown();
+ void initiateAllocateComponent(const sp<AMessage> &msg);
+ void initiateConfigureComponent(const sp<AMessage> &msg);
+ void initiateStart();
+
protected:
virtual ~ACodec();
@@ -70,6 +78,9 @@
kWhatFlush = 'flus',
kWhatResume = 'resm',
kWhatDrainDeferredMessages = 'drai',
+ kWhatAllocateComponent = 'allo',
+ kWhatConfigureComponent = 'conf',
+ kWhatStart = 'star',
};
enum {
@@ -118,6 +129,7 @@
List<sp<AMessage> > mDeferredQueue;
bool mSentFormat;
+ bool mIsEncoder;
status_t allocateBuffersOnPort(OMX_U32 portIndex);
status_t freeBuffersOnPort(OMX_U32 portIndex);
@@ -132,8 +144,8 @@
uint32_t portIndex, IOMX::buffer_id bufferID,
ssize_t *index = NULL);
- void setComponentRole(bool isEncoder, const char *mime);
- void configureCodec(const char *mime, const sp<AMessage> &msg);
+ status_t setComponentRole(bool isEncoder, const char *mime);
+ status_t configureCodec(const char *mime, const sp<AMessage> &msg);
status_t setVideoPortFormatType(
OMX_U32 portIndex,
@@ -145,20 +157,37 @@
status_t setupVideoDecoder(
const char *mime, int32_t width, int32_t height);
+ status_t setupVideoEncoder(
+ const char *mime, const sp<AMessage> &msg);
+
status_t setVideoFormatOnPort(
OMX_U32 portIndex,
int32_t width, int32_t height,
OMX_VIDEO_CODINGTYPE compressionFormat);
- status_t setupAACDecoder(int32_t numChannels, int32_t sampleRate);
- status_t setupAMRDecoder(bool isWAMR);
- status_t setupG711Decoder(int32_t numChannels);
+ status_t setupAACCodec(
+ bool encoder,
+ int32_t numChannels, int32_t sampleRate, int32_t bitRate);
+
+ status_t selectAudioPortFormat(
+ OMX_U32 portIndex, OMX_AUDIO_CODINGTYPE desiredFormat);
+
+ status_t setupAMRCodec(bool encoder, bool isWAMR, int32_t bitRate);
+ status_t setupG711Codec(bool encoder, int32_t numChannels);
status_t setupRawAudioFormat(
OMX_U32 portIndex, int32_t sampleRate, int32_t numChannels);
status_t setMinBufferSize(OMX_U32 portIndex, size_t size);
+ status_t setupMPEG4EncoderParameters(const sp<AMessage> &msg);
+ status_t setupH263EncoderParameters(const sp<AMessage> &msg);
+ status_t setupAVCEncoderParameters(const sp<AMessage> &msg);
+
+ status_t verifySupportForProfileAndLevel(int32_t profile, int32_t level);
+ status_t configureBitrate(int32_t bitrate);
+ status_t setupErrorCorrectionParameters();
+
status_t initNativeWindow();
// Returns true iff all buffers on the given port have status OWNED_BY_US.
@@ -173,7 +202,9 @@
void sendFormatChange();
- void signalError(OMX_ERRORTYPE error = OMX_ErrorUndefined);
+ void signalError(
+ OMX_ERRORTYPE error = OMX_ErrorUndefined,
+ status_t internalError = UNKNOWN_ERROR);
DISALLOW_EVIL_CONSTRUCTORS(ACodec);
};
diff --git a/include/media/stagefright/AMRWriter.h b/include/media/stagefright/AMRWriter.h
index 62d57b4..392f968 100644
--- a/include/media/stagefright/AMRWriter.h
+++ b/include/media/stagefright/AMRWriter.h
@@ -37,7 +37,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual bool reachedEOS();
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
protected:
@@ -60,6 +60,7 @@
status_t threadFunc();
bool exceedsFileSizeLimit();
bool exceedsFileDurationLimit();
+ status_t reset();
AMRWriter(const AMRWriter &);
AMRWriter &operator=(const AMRWriter &);
diff --git a/include/media/stagefright/AudioPlayer.h b/include/media/stagefright/AudioPlayer.h
index 0b79324..70c47ae 100644
--- a/include/media/stagefright/AudioPlayer.h
+++ b/include/media/stagefright/AudioPlayer.h
@@ -64,6 +64,8 @@
bool isSeeking();
bool reachedEOS(status_t *finalStatus);
+ status_t setPlaybackRatePermille(int32_t ratePermille);
+
private:
friend class VideoEditorAudioPlayer;
sp<MediaSource> mSource;
diff --git a/include/media/stagefright/AudioSource.h b/include/media/stagefright/AudioSource.h
index 19bd31b..f5466e8 100644
--- a/include/media/stagefright/AudioSource.h
+++ b/include/media/stagefright/AudioSource.h
@@ -34,13 +34,13 @@
// Note that the "channels" parameter is _not_ the number of channels,
// but a bitmask of audio_channels_t constants.
AudioSource(
- int inputSource, uint32_t sampleRate,
+ audio_source_t inputSource, uint32_t sampleRate,
uint32_t channels = AUDIO_CHANNEL_IN_MONO);
status_t initCheck() const;
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual sp<MetaData> getFormat();
// Returns the maximum amplitude since last call.
@@ -95,8 +95,10 @@
int32_t startFrame, int32_t rampDurationFrames,
uint8_t *data, size_t bytes);
+ void queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs);
void releaseQueuedFrames_l();
void waitOutstandingEncodingFrames_l();
+ status_t reset();
AudioSource(const AudioSource &);
AudioSource &operator=(const AudioSource &);
diff --git a/include/media/stagefright/CameraSource.h b/include/media/stagefright/CameraSource.h
index 446720b..5a35358 100644
--- a/include/media/stagefright/CameraSource.h
+++ b/include/media/stagefright/CameraSource.h
@@ -79,7 +79,7 @@
virtual ~CameraSource();
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t read(
MediaBuffer **buffer, const ReadOptions *options = NULL);
@@ -163,7 +163,6 @@
bool storeMetaDataInVideoBuffers);
virtual void startCameraRecording();
- virtual void stopCameraRecording();
virtual void releaseRecordingFrame(const sp<IMemory>& frame);
// Returns true if need to skip the current frame.
@@ -220,7 +219,9 @@
status_t checkFrameRate(const CameraParameters& params,
int32_t frameRate);
+ void stopCameraRecording();
void releaseCamera();
+ status_t reset();
CameraSource(const CameraSource &);
CameraSource &operator=(const CameraSource &);
diff --git a/include/media/stagefright/CameraSourceTimeLapse.h b/include/media/stagefright/CameraSourceTimeLapse.h
index b060691..0936da2 100644
--- a/include/media/stagefright/CameraSourceTimeLapse.h
+++ b/include/media/stagefright/CameraSourceTimeLapse.h
@@ -121,9 +121,6 @@
// Wrapper over CameraSource::read() to implement quick stop.
virtual status_t read(MediaBuffer **buffer, const ReadOptions *options = NULL);
- // For video camera case, just stops the camera's video recording.
- virtual void stopCameraRecording();
-
// mSkipCurrentFrame is set to true in dataCallbackTimestamp() if the current
// frame needs to be skipped and this function just returns the value of mSkipCurrentFrame.
virtual bool skipCurrentFrame(int64_t timestampUs);
diff --git a/include/media/stagefright/DataSource.h b/include/media/stagefright/DataSource.h
index 713af92..00d583e 100644
--- a/include/media/stagefright/DataSource.h
+++ b/include/media/stagefright/DataSource.h
@@ -81,7 +81,7 @@
static void RegisterDefaultSniffers();
// for DRM
- virtual sp<DecryptHandle> DrmInitialization() {
+ virtual sp<DecryptHandle> DrmInitialization(const char *mime = NULL) {
return NULL;
}
virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client) {};
diff --git a/include/media/stagefright/FileSource.h b/include/media/stagefright/FileSource.h
index 6cf86dc..d994cb3 100644
--- a/include/media/stagefright/FileSource.h
+++ b/include/media/stagefright/FileSource.h
@@ -38,7 +38,7 @@
virtual status_t getSize(off64_t *size);
- virtual sp<DecryptHandle> DrmInitialization();
+ virtual sp<DecryptHandle> DrmInitialization(const char *mime);
virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
diff --git a/include/media/stagefright/HardwareAPI.h b/include/media/stagefright/HardwareAPI.h
index 32eed3f..17efd35 100644
--- a/include/media/stagefright/HardwareAPI.h
+++ b/include/media/stagefright/HardwareAPI.h
@@ -19,7 +19,7 @@
#define HARDWARE_API_H_
#include <media/stagefright/OMXPluginBase.h>
-#include <ui/android_native_buffer.h>
+#include <system/window.h>
#include <utils/RefBase.h>
#include <OMX_Component.h>
diff --git a/include/media/stagefright/MPEG2TSWriter.h b/include/media/stagefright/MPEG2TSWriter.h
index e4c1c49..a7c9ecf 100644
--- a/include/media/stagefright/MPEG2TSWriter.h
+++ b/include/media/stagefright/MPEG2TSWriter.h
@@ -37,7 +37,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual status_t start(MetaData *param = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
virtual bool reachedEOS();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -78,6 +78,7 @@
void writeAccessUnit(int32_t sourceIndex, const sp<ABuffer> &buffer);
ssize_t internalWrite(const void *data, size_t size);
+ status_t reset();
DISALLOW_EVIL_CONSTRUCTORS(MPEG2TSWriter);
};
diff --git a/include/media/stagefright/MPEG4Writer.h b/include/media/stagefright/MPEG4Writer.h
index 77166ed..0409b30 100644
--- a/include/media/stagefright/MPEG4Writer.h
+++ b/include/media/stagefright/MPEG4Writer.h
@@ -37,7 +37,7 @@
virtual status_t addSource(const sp<MediaSource> &source);
virtual status_t start(MetaData *param = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t pause();
virtual bool reachedEOS();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -184,6 +184,7 @@
void writeLongitude(int degreex10000);
void sendSessionSummary();
void release();
+ status_t reset();
MPEG4Writer(const MPEG4Writer &);
MPEG4Writer &operator=(const MPEG4Writer &);
diff --git a/include/media/stagefright/MediaCodec.h b/include/media/stagefright/MediaCodec.h
new file mode 100644
index 0000000..8c11c9c
--- /dev/null
+++ b/include/media/stagefright/MediaCodec.h
@@ -0,0 +1,183 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef MEDIA_CODEC_H_
+
+#define MEDIA_CODEC_H_
+
+#include <gui/ISurfaceTexture.h>
+#include <media/stagefright/foundation/AHandler.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+struct ABuffer;
+struct ACodec;
+struct AMessage;
+struct SoftwareRenderer;
+struct SurfaceTextureClient;
+
+struct MediaCodec : public AHandler {
+ enum ConfigureFlags {
+ CONFIGURE_FLAG_ENCODE = 1,
+ };
+
+ enum BufferFlags {
+ BUFFER_FLAG_SYNCFRAME = 1,
+ BUFFER_FLAG_CODECCONFIG = 2,
+ BUFFER_FLAG_EOS = 4,
+ };
+
+ static sp<MediaCodec> CreateByType(
+ const sp<ALooper> &looper, const char *mime, bool encoder);
+
+ static sp<MediaCodec> CreateByComponentName(
+ const sp<ALooper> &looper, const char *name);
+
+ status_t configure(
+ const sp<AMessage> &format,
+ const sp<SurfaceTextureClient> &nativeWindow,
+ uint32_t flags);
+
+ status_t start();
+ status_t stop();
+
+ status_t flush();
+
+ status_t queueInputBuffer(
+ size_t index,
+ size_t offset,
+ size_t size,
+ int64_t presentationTimeUs,
+ uint32_t flags);
+
+ status_t dequeueInputBuffer(size_t *index, int64_t timeoutUs = 0ll);
+
+ status_t dequeueOutputBuffer(
+ size_t *index,
+ size_t *offset,
+ size_t *size,
+ int64_t *presentationTimeUs,
+ uint32_t *flags,
+ int64_t timeoutUs = 0ll);
+
+ status_t renderOutputBufferAndRelease(size_t index);
+ status_t releaseOutputBuffer(size_t index);
+
+ status_t getOutputFormat(sp<AMessage> *format) const;
+
+ status_t getInputBuffers(Vector<sp<ABuffer> > *buffers) const;
+ status_t getOutputBuffers(Vector<sp<ABuffer> > *buffers) const;
+
+protected:
+ virtual ~MediaCodec();
+ virtual void onMessageReceived(const sp<AMessage> &msg);
+
+private:
+ enum State {
+ UNINITIALIZED,
+ INITIALIZING,
+ INITIALIZED,
+ CONFIGURING,
+ CONFIGURED,
+ STARTING,
+ STARTED,
+ FLUSHING,
+ STOPPING,
+ };
+
+ enum {
+ kPortIndexInput = 0,
+ kPortIndexOutput = 1,
+ };
+
+ enum {
+ kWhatInit = 'init',
+ kWhatConfigure = 'conf',
+ kWhatStart = 'strt',
+ kWhatStop = 'stop',
+ kWhatDequeueInputBuffer = 'deqI',
+ kWhatQueueInputBuffer = 'queI',
+ kWhatDequeueOutputBuffer = 'deqO',
+ kWhatReleaseOutputBuffer = 'relO',
+ kWhatGetBuffers = 'getB',
+ kWhatFlush = 'flus',
+ kWhatGetOutputFormat = 'getO',
+ kWhatDequeueInputTimedOut = 'dITO',
+ kWhatDequeueOutputTimedOut = 'dOTO',
+ kWhatCodecNotify = 'codc',
+ };
+
+ enum {
+ kFlagIsSoftwareCodec = 1,
+ kFlagOutputFormatChanged = 2,
+ kFlagOutputBuffersChanged = 4,
+ kFlagStickyError = 8,
+ kFlagDequeueInputPending = 16,
+ kFlagDequeueOutputPending = 32,
+ };
+
+ struct BufferInfo {
+ void *mBufferID;
+ sp<ABuffer> mData;
+ sp<AMessage> mNotify;
+ bool mOwnedByClient;
+ };
+
+ State mState;
+ sp<ALooper> mLooper;
+ sp<ALooper> mCodecLooper;
+ sp<ACodec> mCodec;
+ uint32_t mReplyID;
+ uint32_t mFlags;
+ sp<SurfaceTextureClient> mNativeWindow;
+ SoftwareRenderer *mSoftRenderer;
+ sp<AMessage> mOutputFormat;
+
+ List<size_t> mAvailPortBuffers[2];
+ Vector<BufferInfo> mPortBuffers[2];
+
+ int32_t mDequeueInputTimeoutGeneration;
+ uint32_t mDequeueInputReplyID;
+
+ int32_t mDequeueOutputTimeoutGeneration;
+ uint32_t mDequeueOutputReplyID;
+
+ MediaCodec(const sp<ALooper> &looper);
+
+ static status_t PostAndAwaitResponse(
+ const sp<AMessage> &msg, sp<AMessage> *response);
+
+ status_t init(const char *name, bool nameIsType, bool encoder);
+
+ void setState(State newState);
+ void returnBuffersToCodec();
+ void returnBuffersToCodecOnPort(int32_t portIndex);
+ size_t updateBuffers(int32_t portIndex, const sp<AMessage> &msg);
+ status_t onQueueInputBuffer(const sp<AMessage> &msg);
+ status_t onReleaseOutputBuffer(const sp<AMessage> &msg);
+ ssize_t dequeuePortBuffer(int32_t portIndex);
+
+ bool handleDequeueInputBuffer(uint32_t replyID, bool newRequest = false);
+ bool handleDequeueOutputBuffer(uint32_t replyID, bool newRequest = false);
+ void cancelPendingDequeueOperations();
+
+ DISALLOW_EVIL_CONSTRUCTORS(MediaCodec);
+};
+
+} // namespace android
+
+#endif // MEDIA_CODEC_H_
diff --git a/include/media/stagefright/MediaDebug.h b/include/media/stagefright/MediaDebug.h
deleted file mode 100644
index 2ca9667..0000000
--- a/include/media/stagefright/MediaDebug.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef MEDIA_DEBUG_H_
-
-#define MEDIA_DEBUG_H_
-
-#include <cutils/log.h>
-
-#define LITERAL_TO_STRING_INTERNAL(x) #x
-#define LITERAL_TO_STRING(x) LITERAL_TO_STRING_INTERNAL(x)
-
-#define CHECK_EQ(x,y) \
- LOG_ALWAYS_FATAL_IF( \
- (x) != (y), \
- __FILE__ ":" LITERAL_TO_STRING(__LINE__) " " #x " != " #y)
-
-#define CHECK(x) \
- LOG_ALWAYS_FATAL_IF( \
- !(x), \
- __FILE__ ":" LITERAL_TO_STRING(__LINE__) " " #x)
-
-#endif // MEDIA_DEBUG_H_
diff --git a/include/media/stagefright/MediaErrors.h b/include/media/stagefright/MediaErrors.h
index 21d00b8..dd3bf28 100644
--- a/include/media/stagefright/MediaErrors.h
+++ b/include/media/stagefright/MediaErrors.h
@@ -40,6 +40,7 @@
// Not technically an error.
INFO_FORMAT_CHANGED = MEDIA_ERROR_BASE - 12,
INFO_DISCONTINUITY = MEDIA_ERROR_BASE - 13,
+ INFO_OUTPUT_BUFFERS_CHANGED = MEDIA_ERROR_BASE - 14,
// The following constant values should be in sync with
// drm/drm_framework_common.h
diff --git a/include/media/stagefright/MediaExtractor.h b/include/media/stagefright/MediaExtractor.h
index eb45237..94090ee 100644
--- a/include/media/stagefright/MediaExtractor.h
+++ b/include/media/stagefright/MediaExtractor.h
@@ -56,10 +56,10 @@
virtual uint32_t flags() const;
// for DRM
- virtual void setDrmFlag(bool flag) {
+ void setDrmFlag(bool flag) {
mIsDrm = flag;
};
- virtual bool getDrmFlag() {
+ bool getDrmFlag() {
return mIsDrm;
}
virtual char* getDrmTrackInfo(size_t trackID, int *len) {
diff --git a/include/media/stagefright/NativeWindowWrapper.h b/include/media/stagefright/NativeWindowWrapper.h
index f323cbc..97cc0ce 100644
--- a/include/media/stagefright/NativeWindowWrapper.h
+++ b/include/media/stagefright/NativeWindowWrapper.h
@@ -18,40 +18,28 @@
#define NATIVE_WINDOW_WRAPPER_H_
-#include <surfaceflinger/Surface.h>
#include <gui/SurfaceTextureClient.h>
namespace android {
-// Both Surface and SurfaceTextureClient are RefBase that implement the
-// ANativeWindow interface, but at different addresses. ANativeWindow is not
-// a RefBase but acts like one for use with sp<>. This wrapper converts a
-// Surface or SurfaceTextureClient into a single reference-counted object
-// that holds an sp reference to the underlying Surface or SurfaceTextureClient,
-// It provides a method to get the ANativeWindow.
+// SurfaceTextureClient derives from ANativeWindow which derives from multiple
+// base classes, in order to carry it in AMessages, we'll temporarily wrap it
+// into a NativeWindowWrapper.
struct NativeWindowWrapper : RefBase {
NativeWindowWrapper(
- const sp<Surface> &surface) :
- mSurface(surface) { }
-
- NativeWindowWrapper(
const sp<SurfaceTextureClient> &surfaceTextureClient) :
mSurfaceTextureClient(surfaceTextureClient) { }
sp<ANativeWindow> getNativeWindow() const {
- if (mSurface != NULL) {
- return mSurface;
- } else {
- return mSurfaceTextureClient;
- }
+ return mSurfaceTextureClient;
}
- // If needed later we can provide a method to ask what kind of native window
+ sp<SurfaceTextureClient> getSurfaceTextureClient() const {
+ return mSurfaceTextureClient;
+ }
private:
- // At most one of mSurface and mSurfaceTextureClient will be non-NULL
- const sp<Surface> mSurface;
const sp<SurfaceTextureClient> mSurfaceTextureClient;
DISALLOW_EVIL_CONSTRUCTORS(NativeWindowWrapper);
diff --git a/include/media/stagefright/NuMediaExtractor.h b/include/media/stagefright/NuMediaExtractor.h
new file mode 100644
index 0000000..96efdff
--- /dev/null
+++ b/include/media/stagefright/NuMediaExtractor.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef NU_MEDIA_EXTRACTOR_H_
+#define NU_MEDIA_EXTRACTOR_H_
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+#include <utils/Vector.h>
+
+namespace android {
+
+struct ABuffer;
+struct AMessage;
+struct MediaBuffer;
+struct MediaExtractor;
+struct MediaSource;
+
+struct NuMediaExtractor : public RefBase {
+ NuMediaExtractor();
+
+ status_t setDataSource(const char *path);
+
+ size_t countTracks() const;
+ status_t getTrackFormat(size_t index, sp<AMessage> *format) const;
+
+ status_t selectTrack(size_t index);
+
+ status_t seekTo(int64_t timeUs);
+
+ status_t advance();
+ status_t readSampleData(const sp<ABuffer> &buffer);
+ status_t getSampleTrackIndex(size_t *trackIndex);
+ status_t getSampleTime(int64_t *sampleTimeUs);
+
+protected:
+ virtual ~NuMediaExtractor();
+
+private:
+ enum TrackFlags {
+ kIsVorbis = 1,
+ };
+
+ struct TrackInfo {
+ sp<MediaSource> mSource;
+ size_t mTrackIndex;
+ status_t mFinalResult;
+ MediaBuffer *mSample;
+ int64_t mSampleTimeUs;
+ uint32_t mFlags; // bitmask of "TrackFlags"
+ };
+
+ sp<MediaExtractor> mImpl;
+
+ Vector<TrackInfo> mSelectedTracks;
+
+ ssize_t fetchTrackSamples(int64_t seekTimeUs = -1ll);
+ void releaseTrackSamples();
+
+ DISALLOW_EVIL_CONSTRUCTORS(NuMediaExtractor);
+};
+
+} // namespace android
+
+#endif // NU_MEDIA_EXTRACTOR_H_
+
diff --git a/include/media/stagefright/OMXCodec.h b/include/media/stagefright/OMXCodec.h
index 84f8282..e541c18 100644
--- a/include/media/stagefright/OMXCodec.h
+++ b/include/media/stagefright/OMXCodec.h
@@ -172,6 +172,7 @@
uint32_t mFlags;
bool mIsEncoder;
+ bool mIsVideo;
char *mMIME;
char *mComponentName;
sp<MetaData> mOutputFormat;
@@ -334,7 +335,7 @@
status_t applyRotation();
status_t waitForBufferFilled_l();
- int64_t retrieveDecodingTimeUs(bool isCodecSpecific);
+ int64_t getDecodingTimeUs();
status_t parseAVCCodecSpecificData(
const void *data, size_t size,
diff --git a/include/media/stagefright/SurfaceMediaSource.h b/include/media/stagefright/SurfaceMediaSource.h
index d0940bb..54baab6 100644
--- a/include/media/stagefright/SurfaceMediaSource.h
+++ b/include/media/stagefright/SurfaceMediaSource.h
@@ -58,7 +58,7 @@
// For the MediaSource interface for use by StageFrightRecorder:
virtual status_t start(MetaData *params = NULL);
- virtual status_t stop();
+ virtual status_t stop() { return reset(); }
virtual status_t read(
MediaBuffer **buffer, const ReadOptions *options = NULL);
virtual sp<MetaData> getFormat();
@@ -359,6 +359,8 @@
Condition mFrameAvailableCondition;
Condition mFrameCompleteCondition;
+ status_t reset();
+
// Avoid copying and equating and default constructor
DISALLOW_IMPLICIT_CONSTRUCTORS(SurfaceMediaSource);
};
diff --git a/include/media/stagefright/foundation/AMessage.h b/include/media/stagefright/foundation/AMessage.h
index 7ec54aa..e5416e4 100644
--- a/include/media/stagefright/foundation/AMessage.h
+++ b/include/media/stagefright/foundation/AMessage.h
@@ -25,6 +25,7 @@
namespace android {
+struct ABuffer;
struct AString;
struct Parcel;
@@ -50,6 +51,7 @@
void setPointer(const char *name, void *value);
void setString(const char *name, const char *s, ssize_t len = -1);
void setObject(const char *name, const sp<RefBase> &obj);
+ void setBuffer(const char *name, const sp<ABuffer> &buffer);
void setMessage(const char *name, const sp<AMessage> &obj);
void setRect(
@@ -64,6 +66,7 @@
bool findPointer(const char *name, void **value) const;
bool findString(const char *name, AString *value) const;
bool findObject(const char *name, sp<RefBase> *obj) const;
+ bool findBuffer(const char *name, sp<ABuffer> *buffer) const;
bool findMessage(const char *name, sp<AMessage> *obj) const;
bool findRect(
@@ -90,10 +93,6 @@
AString debugString(int32_t indent = 0) const;
-protected:
- virtual ~AMessage();
-
-private:
enum Type {
kTypeInt32,
kTypeInt64,
@@ -105,8 +104,16 @@
kTypeObject,
kTypeMessage,
kTypeRect,
+ kTypeBuffer,
};
+ size_t countEntries() const;
+ const char *getEntryNameAt(size_t index, Type *type) const;
+
+protected:
+ virtual ~AMessage();
+
+private:
uint32_t mWhat;
ALooper::handler_id mTarget;
@@ -131,7 +138,7 @@
};
enum {
- kMaxNumItems = 16
+ kMaxNumItems = 32
};
Item mItems[kMaxNumItems];
size_t mNumItems;
@@ -140,6 +147,9 @@
void freeItem(Item *item);
const Item *findItem(const char *name, Type type) const;
+ void setObjectInternal(
+ const char *name, const sp<RefBase> &obj, Type type);
+
DISALLOW_EVIL_CONSTRUCTORS(AMessage);
};
diff --git a/include/private/gui/ComposerService.h b/include/private/gui/ComposerService.h
new file mode 100644
index 0000000..d04491a
--- /dev/null
+++ b/include/private/gui/ComposerService.h
@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_PRIVATE_GUI_COMPOSER_SERVICE_H
+#define ANDROID_PRIVATE_GUI_COMPOSER_SERVICE_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <utils/Singleton.h>
+#include <utils/StrongPointer.h>
+
+
+namespace android {
+
+// ---------------------------------------------------------------------------
+
+class IMemoryHeap;
+class ISurfaceComposer;
+class surface_flinger_cblk_t;
+
+// ---------------------------------------------------------------------------
+
+class ComposerService : public Singleton<ComposerService>
+{
+ // these are constants
+ sp<ISurfaceComposer> mComposerService;
+ sp<IMemoryHeap> mServerCblkMemory;
+ surface_flinger_cblk_t volatile* mServerCblk;
+ ComposerService();
+ friend class Singleton<ComposerService>;
+public:
+ static sp<ISurfaceComposer> getComposerService();
+ static surface_flinger_cblk_t const volatile * getControlBlock();
+};
+
+// ---------------------------------------------------------------------------
+}; // namespace android
+
+#endif // ANDROID_PRIVATE_GUI_COMPOSER_SERVICE_H
diff --git a/include/private/surfaceflinger/LayerState.h b/include/private/gui/LayerState.h
similarity index 97%
rename from include/private/surfaceflinger/LayerState.h
rename to include/private/gui/LayerState.h
index 3eb5c99..ca277e0 100644
--- a/include/private/surfaceflinger/LayerState.h
+++ b/include/private/gui/LayerState.h
@@ -23,8 +23,7 @@
#include <utils/Errors.h>
#include <ui/Region.h>
-
-#include <surfaceflinger/ISurface.h>
+#include <gui/ISurface.h>
namespace android {
diff --git a/include/private/surfaceflinger/SharedBufferStack.h b/include/private/gui/SharedBufferStack.h
similarity index 100%
rename from include/private/surfaceflinger/SharedBufferStack.h
rename to include/private/gui/SharedBufferStack.h
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 20abd51..af2db93 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -54,50 +54,99 @@
#define CBLK_RESTORED_ON 0x0040 // track has been restored after invalidation
#define CBLK_RESTORED_OFF 0x0040 // by AudioFlinger
+// Important: do not add any virtual methods, including ~
struct audio_track_cblk_t
{
// The data members are grouped so that members accessed frequently and in the same context
// are in the same line of data cache.
- Mutex lock;
- Condition cv;
+ Mutex lock; // sizeof(int)
+ Condition cv; // sizeof(int)
+
+ // next 4 are offsets within "buffers"
volatile uint32_t user;
volatile uint32_t server;
uint32_t userBase;
uint32_t serverBase;
+
+ // if there is a shared buffer, "buffers" is the value of pointer() for the shared
+ // buffer, otherwise "buffers" points immediately after the control block
void* buffers;
uint32_t frameCount;
+
// Cache line boundary
+
uint32_t loopStart;
- uint32_t loopEnd;
- int loopCount;
- volatile union {
- uint16_t volume[2];
- uint32_t volumeLR;
- };
+ uint32_t loopEnd; // read-only for server, read/write for client
+ int loopCount; // read/write for client
+
+ // Channel volumes are fixed point U4.12, so 0x1000 means 1.0.
+ // Left channel is in [0:15], right channel is in [16:31].
+ // Always read and write the combined pair atomically.
+ // For AudioTrack only, not used by AudioRecord.
+private:
+ uint32_t mVolumeLR;
+public:
+
uint32_t sampleRate;
+
// NOTE: audio_track_cblk_t::frameSize is not equal to AudioTrack::frameSize() for
// 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer
- uint8_t frameSize;
+ // read-only for client, server writes once at initialization and is then read-only
+ uint8_t frameSize; // would normally be size_t, but 8 bits is plenty
+
+ // never used
uint8_t pad1;
+
+ // used by client only
uint16_t bufferTimeoutMs; // Maximum cumulated timeout before restarting audioflinger
- uint16_t waitTimeMs; // Cumulated wait time
- uint16_t sendLevel;
+ uint16_t waitTimeMs; // Cumulated wait time, used by client only
+private:
+ // client write-only, server read-only
+ uint16_t mSendLevel; // Fixed point U4.12 so 0x1000 means 1.0
+public:
volatile int32_t flags;
// Cache line boundary (32 bytes)
+ // Since the control block is always located in shared memory, this constructor
+ // is only used for placement new(). It is never used for regular new() or stack.
audio_track_cblk_t();
- uint32_t stepUser(uint32_t frameCount);
- bool stepServer(uint32_t frameCount);
+ uint32_t stepUser(uint32_t frameCount); // called by client only, where
+ // client includes regular AudioTrack and AudioFlinger::PlaybackThread::OutputTrack
+ bool stepServer(uint32_t frameCount); // called by server only
void* buffer(uint32_t offset) const;
uint32_t framesAvailable();
uint32_t framesAvailable_l();
- uint32_t framesReady();
+ uint32_t framesReady(); // called by server only
bool tryLock();
+
+ // No barriers on the following operations, so the ordering of loads/stores
+ // with respect to other parameters is UNPREDICTABLE. That's considered safe.
+
+ // for AudioTrack client only, caller must limit to 0.0 <= sendLevel <= 1.0
+ void setSendLevel(float sendLevel) {
+ mSendLevel = uint16_t(sendLevel * 0x1000);
+ }
+
+ // for AudioFlinger only; the return value must be validated by the caller
+ uint16_t getSendLevel_U4_12() const {
+ return mSendLevel;
+ }
+
+ // for AudioTrack client only, caller must limit to 0 <= volumeLR <= 0x10001000
+ void setVolumeLR(uint32_t volumeLR) {
+ mVolumeLR = volumeLR;
+ }
+
+ // for AudioFlinger only; the return value must be validated by the caller
+ uint32_t getVolumeLR() const {
+ return mVolumeLR;
+ }
+
};
diff --git a/include/private/opengles/gl_context.h b/include/private/opengles/gl_context.h
deleted file mode 100644
index 6b1fa77..0000000
--- a/include/private/opengles/gl_context.h
+++ /dev/null
@@ -1,640 +0,0 @@
-/*
- * Copyright (C) 2006 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_OPENGLES_CONTEXT_H
-#define ANDROID_OPENGLES_CONTEXT_H
-
-#include <stdint.h>
-#include <stddef.h>
-#include <sys/types.h>
-#include <pthread.h>
-#ifdef HAVE_ANDROID_OS
-#include <bionic_tls.h>
-#endif
-
-#include <private/pixelflinger/ggl_context.h>
-#include <hardware/gralloc.h>
-
-#include <GLES/gl.h>
-#include <GLES/glext.h>
-
-namespace android {
-
-
-const unsigned int OGLES_NUM_COMPRESSED_TEXTURE_FORMATS = 10
-#ifdef GL_OES_compressed_ETC1_RGB8_texture
- + 1
-#endif
- ;
-
-class EGLTextureObject;
-class EGLSurfaceManager;
-class EGLBufferObjectManager;
-
-namespace gl {
-
-struct ogles_context_t;
-struct matrixx_t;
-struct transform_t;
-struct buffer_t;
-
-ogles_context_t* getGlContext();
-
-template<typename T>
-static inline void swap(T& a, T& b) {
- T t(a); a = b; b = t;
-}
-template<typename T>
-inline T max(T a, T b) {
- return a<b ? b : a;
-}
-template<typename T>
-inline T max(T a, T b, T c) {
- return max(a, max(b, c));
-}
-template<typename T>
-inline T min(T a, T b) {
- return a<b ? a : b;
-}
-template<typename T>
-inline T min(T a, T b, T c) {
- return min(a, min(b, c));
-}
-template<typename T>
-inline T min(T a, T b, T c, T d) {
- return min(min(a,b), min(c,d));
-}
-
-// ----------------------------------------------------------------------------
-// vertices
-// ----------------------------------------------------------------------------
-
-struct vec3_t {
- union {
- struct { GLfixed x, y, z; };
- struct { GLfixed r, g, b; };
- struct { GLfixed S, T, R; };
- GLfixed v[3];
- };
-};
-
-struct vec4_t {
- union {
- struct { GLfixed x, y, z, w; };
- struct { GLfixed r, g, b, a; };
- struct { GLfixed S, T, R, Q; };
- GLfixed v[4];
- };
-};
-
-struct vertex_t {
- enum {
- // these constant matter for our clipping
- CLIP_L = 0x0001, // clipping flags
- CLIP_R = 0x0002,
- CLIP_B = 0x0004,
- CLIP_T = 0x0008,
- CLIP_N = 0x0010,
- CLIP_F = 0x0020,
-
- EYE = 0x0040,
- RESERVED = 0x0080,
-
- USER_CLIP_0 = 0x0100, // user clipping flags
- USER_CLIP_1 = 0x0200,
- USER_CLIP_2 = 0x0400,
- USER_CLIP_3 = 0x0800,
- USER_CLIP_4 = 0x1000,
- USER_CLIP_5 = 0x2000,
-
- LIT = 0x4000, // lighting has been applied
- TT = 0x8000, // texture coords transformed
-
- FRUSTUM_CLIP_ALL= 0x003F,
- USER_CLIP_ALL = 0x3F00,
- CLIP_ALL = 0x3F3F,
- };
-
- // the fields below are arranged to minimize d-cache usage
- // we group together, by cache-line, the fields most likely to be used
-
- union {
- vec4_t obj;
- vec4_t eye;
- };
- vec4_t clip;
-
- uint32_t flags;
- size_t index; // cache tag, and vertex index
- GLfixed fog;
- uint8_t locked;
- uint8_t mru;
- uint8_t reserved[2];
- vec4_t window;
-
- vec4_t color;
- vec4_t texture[GGL_TEXTURE_UNIT_COUNT];
- uint32_t reserved1[4];
-
- inline void clear() {
- flags = index = locked = mru = 0;
- }
-};
-
-struct point_size_t {
- GGLcoord size;
- GLboolean smooth;
-};
-
-struct line_width_t {
- GGLcoord width;
- GLboolean smooth;
-};
-
-struct polygon_offset_t {
- GLfixed factor;
- GLfixed units;
- GLboolean enable;
-};
-
-// ----------------------------------------------------------------------------
-// arrays
-// ----------------------------------------------------------------------------
-
-struct array_t {
- typedef void (*fetcher_t)(ogles_context_t*, GLfixed*, const GLvoid*);
- fetcher_t fetch;
- GLvoid const* physical_pointer;
- GLint size;
- GLsizei stride;
- GLvoid const* pointer;
- buffer_t const* bo;
- uint16_t type;
- GLboolean enable;
- GLboolean pad;
- GLsizei bounds;
- void init(GLint, GLenum, GLsizei, const GLvoid *, const buffer_t*, GLsizei);
- inline void resolve();
- inline const GLubyte* element(GLint i) const {
- return (const GLubyte*)physical_pointer + i * stride;
- }
-};
-
-struct array_machine_t {
- array_t vertex;
- array_t normal;
- array_t color;
- array_t texture[GGL_TEXTURE_UNIT_COUNT];
- uint8_t activeTexture;
- uint8_t tmu;
- uint16_t cull;
- uint32_t flags;
- GLenum indicesType;
- buffer_t const* array_buffer;
- buffer_t const* element_array_buffer;
-
- void (*compileElements)(ogles_context_t*, vertex_t*, GLint, GLsizei);
- void (*compileElement)(ogles_context_t*, vertex_t*, GLint);
-
- void (*mvp_transform)(transform_t const*, vec4_t*, vec4_t const*);
- void (*mv_transform)(transform_t const*, vec4_t*, vec4_t const*);
- void (*tex_transform[2])(transform_t const*, vec4_t*, vec4_t const*);
- void (*perspective)(ogles_context_t*c, vertex_t* v);
- void (*clipVertex)(ogles_context_t* c, vertex_t* nv,
- GGLfixed t, const vertex_t* s, const vertex_t* p);
- void (*clipEye)(ogles_context_t* c, vertex_t* nv,
- GGLfixed t, const vertex_t* s, const vertex_t* p);
-};
-
-struct vertex_cache_t {
- enum {
- // must be at least 4
- // 3 vertice for triangles
- // or 2 + 2 for indexed triangles w/ cache contention
- VERTEX_BUFFER_SIZE = 8,
- // must be a power of two and at least 3
- VERTEX_CACHE_SIZE = 64, // 8 KB
-
- INDEX_BITS = 16,
- INDEX_MASK = ((1LU<<INDEX_BITS)-1),
- INDEX_SEQ = 1LU<<INDEX_BITS,
- };
- vertex_t* vBuffer;
- vertex_t* vCache;
- uint32_t sequence;
- void* base;
- uint32_t total;
- uint32_t misses;
- int64_t startTime;
- void init();
- void uninit();
- void clear();
- void dump_stats(GLenum mode);
-};
-
-// ----------------------------------------------------------------------------
-// fog
-// ----------------------------------------------------------------------------
-
-struct fog_t {
- GLfixed density;
- GLfixed start;
- GLfixed end;
- GLfixed invEndMinusStart;
- GLenum mode;
- GLfixed (*fog)(ogles_context_t* c, GLfixed z);
-};
-
-// ----------------------------------------------------------------------------
-// user clip planes
-// ----------------------------------------------------------------------------
-
-const unsigned int OGLES_MAX_CLIP_PLANES = 6;
-
-struct clip_plane_t {
- vec4_t equation;
-};
-
-struct user_clip_planes_t {
- clip_plane_t plane[OGLES_MAX_CLIP_PLANES];
- uint32_t enable;
-};
-
-// ----------------------------------------------------------------------------
-// lighting
-// ----------------------------------------------------------------------------
-
-const unsigned int OGLES_MAX_LIGHTS = 8;
-
-struct light_t {
- vec4_t ambient;
- vec4_t diffuse;
- vec4_t specular;
- vec4_t implicitAmbient;
- vec4_t implicitDiffuse;
- vec4_t implicitSpecular;
- vec4_t position; // position in eye space
- vec4_t objPosition;
- vec4_t normalizedObjPosition;
- vec4_t spotDir;
- vec4_t normalizedSpotDir;
- GLfixed spotExp;
- GLfixed spotCutoff;
- GLfixed spotCutoffCosine;
- GLfixed attenuation[3];
- GLfixed rConstAttenuation;
- GLboolean enable;
-};
-
-struct material_t {
- vec4_t ambient;
- vec4_t diffuse;
- vec4_t specular;
- vec4_t emission;
- GLfixed shininess;
-};
-
-struct light_model_t {
- vec4_t ambient;
- GLboolean twoSide;
-};
-
-struct color_material_t {
- GLenum face;
- GLenum mode;
- GLboolean enable;
-};
-
-struct lighting_t {
- light_t lights[OGLES_MAX_LIGHTS];
- material_t front;
- light_model_t lightModel;
- color_material_t colorMaterial;
- vec4_t implicitSceneEmissionAndAmbient;
- vec4_t objViewer;
- uint32_t enabledLights;
- GLboolean enable;
- GLenum shadeModel;
- typedef void (*light_fct_t)(ogles_context_t*, vertex_t*);
- void (*lightVertex)(ogles_context_t* c, vertex_t* v);
- void (*lightTriangle)(ogles_context_t* c,
- vertex_t* v0, vertex_t* v1, vertex_t* v2);
-};
-
-struct culling_t {
- GLenum cullFace;
- GLenum frontFace;
- GLboolean enable;
-};
-
-// ----------------------------------------------------------------------------
-// textures
-// ----------------------------------------------------------------------------
-
-struct texture_unit_t {
- GLuint name;
- EGLTextureObject* texture;
- uint8_t dirty;
-};
-
-struct texture_state_t
-{
- texture_unit_t tmu[GGL_TEXTURE_UNIT_COUNT];
- int active; // active tmu
- EGLTextureObject* defaultTexture;
- GGLContext* ggl;
- uint8_t packAlignment;
- uint8_t unpackAlignment;
-};
-
-// ----------------------------------------------------------------------------
-// transformation and matrices
-// ----------------------------------------------------------------------------
-
-struct matrixf_t;
-
-struct matrixx_t {
- GLfixed m[16];
- void load(const matrixf_t& rhs);
-};
-
-struct matrix_stack_t;
-
-
-struct matrixf_t {
- void loadIdentity();
- void load(const matrixf_t& rhs);
-
- inline GLfloat* editElements() { return m; }
- inline GLfloat const* elements() const { return m; }
-
- void set(const GLfixed* rhs);
- void set(const GLfloat* rhs);
-
- static void multiply(matrixf_t& r,
- const matrixf_t& lhs, const matrixf_t& rhs);
-
- void dump(const char* what);
-
-private:
- friend struct matrix_stack_t;
- GLfloat m[16];
- void load(const GLfixed* rhs);
- void load(const GLfloat* rhs);
- void multiply(const matrixf_t& rhs);
- void translate(GLfloat x, GLfloat y, GLfloat z);
- void scale(GLfloat x, GLfloat y, GLfloat z);
- void rotate(GLfloat a, GLfloat x, GLfloat y, GLfloat z);
-};
-
-enum {
- OP_IDENTITY = 0x00,
- OP_TRANSLATE = 0x01,
- OP_UNIFORM_SCALE = 0x02,
- OP_SCALE = 0x05,
- OP_ROTATE = 0x08,
- OP_SKEW = 0x10,
- OP_ALL = 0x1F
-};
-
-struct transform_t {
- enum {
- FLAGS_2D_PROJECTION = 0x1
- };
- matrixx_t matrix;
- uint32_t flags;
- uint32_t ops;
-
- union {
- struct {
- void (*point2)(transform_t const* t, vec4_t*, vec4_t const*);
- void (*point3)(transform_t const* t, vec4_t*, vec4_t const*);
- void (*point4)(transform_t const* t, vec4_t*, vec4_t const*);
- };
- void (*pointv[3])(transform_t const* t, vec4_t*, vec4_t const*);
- };
-
- void loadIdentity();
- void picker();
- void dump(const char* what);
-};
-
-struct mvui_transform_t : public transform_t
-{
- void picker();
-};
-
-struct matrix_stack_t {
- enum {
- DO_PICKER = 0x1,
- DO_FLOAT_TO_FIXED = 0x2
- };
- transform_t transform;
- uint8_t maxDepth;
- uint8_t depth;
- uint8_t dirty;
- uint8_t reserved;
- matrixf_t *stack;
- uint8_t *ops;
- void init(int depth);
- void uninit();
- void loadIdentity();
- void load(const GLfixed* rhs);
- void load(const GLfloat* rhs);
- void multiply(const matrixf_t& rhs);
- void translate(GLfloat x, GLfloat y, GLfloat z);
- void scale(GLfloat x, GLfloat y, GLfloat z);
- void rotate(GLfloat a, GLfloat x, GLfloat y, GLfloat z);
- GLint push();
- GLint pop();
- void validate();
- matrixf_t& top() { return stack[depth]; }
- const matrixf_t& top() const { return stack[depth]; }
- uint32_t top_ops() const { return ops[depth]; }
- inline bool isRigidBody() const {
- return !(ops[depth] & ~(OP_TRANSLATE|OP_UNIFORM_SCALE|OP_ROTATE));
- }
-};
-
-struct vp_transform_t {
- transform_t transform;
- matrixf_t matrix;
- GLfloat zNear;
- GLfloat zFar;
- void loadIdentity();
-};
-
-struct transform_state_t {
- enum {
- MODELVIEW = 0x01,
- PROJECTION = 0x02,
- VIEWPORT = 0x04,
- TEXTURE = 0x08,
- MVUI = 0x10,
- MVIT = 0x20,
- MVP = 0x40,
- };
- matrix_stack_t *current;
- matrix_stack_t modelview;
- matrix_stack_t projection;
- matrix_stack_t texture[GGL_TEXTURE_UNIT_COUNT];
-
- // modelview * projection
- transform_t mvp __attribute__((aligned(32)));
- // viewport transformation
- vp_transform_t vpt __attribute__((aligned(32)));
- // same for 4-D vertices
- transform_t mvp4;
- // full modelview inverse transpose
- transform_t mvit4;
- // upper 3x3 of mv-inverse-transpose (for normals)
- mvui_transform_t mvui;
-
- GLenum matrixMode;
- GLenum rescaleNormals;
- uint32_t dirty;
- void invalidate();
- void update_mvp();
- void update_mvit();
- void update_mvui();
-};
-
-struct viewport_t {
- GLint x;
- GLint y;
- GLsizei w;
- GLsizei h;
- struct {
- GLint x;
- GLint y;
- } surfaceport;
- struct {
- GLint x;
- GLint y;
- GLsizei w;
- GLsizei h;
- } scissor;
-};
-
-// ----------------------------------------------------------------------------
-// Lerping
-// ----------------------------------------------------------------------------
-
-struct compute_iterators_t
-{
- void initTriangle(
- vertex_t const* v0,
- vertex_t const* v1,
- vertex_t const* v2);
-
- void initLine(
- vertex_t const* v0,
- vertex_t const* v1);
-
- inline void initLerp(vertex_t const* v0, uint32_t enables);
-
- int iteratorsScale(int32_t it[3],
- int32_t c0, int32_t c1, int32_t c2) const;
-
- void iterators1616(GGLfixed it[3],
- GGLfixed c0, GGLfixed c1, GGLfixed c2) const;
-
- void iterators0032(int32_t it[3],
- int32_t c0, int32_t c1, int32_t c2) const;
-
- void iterators0032(int64_t it[3],
- int32_t c0, int32_t c1, int32_t c2) const;
-
- GGLcoord area() const { return m_area; }
-
-private:
- // don't change order of members here -- used by iterators.S
- GGLcoord m_dx01, m_dy10, m_dx20, m_dy02;
- GGLcoord m_x0, m_y0;
- GGLcoord m_area;
- uint8_t m_scale;
- uint8_t m_area_scale;
- uint8_t m_reserved[2];
-
-};
-
-// ----------------------------------------------------------------------------
-// state
-// ----------------------------------------------------------------------------
-
-#ifdef HAVE_ANDROID_OS
- // We have a dedicated TLS slot in bionic
- inline void setGlThreadSpecific(ogles_context_t *value) {
- ((uint32_t *)__get_tls())[TLS_SLOT_OPENGL] = (uint32_t)value;
- }
- inline ogles_context_t* getGlThreadSpecific() {
- return (ogles_context_t *)(((unsigned *)__get_tls())[TLS_SLOT_OPENGL]);
- }
-#else
- extern pthread_key_t gGLKey;
- inline void setGlThreadSpecific(ogles_context_t *value) {
- pthread_setspecific(gGLKey, value);
- }
- inline ogles_context_t* getGlThreadSpecific() {
- return static_cast<ogles_context_t*>(pthread_getspecific(gGLKey));
- }
-#endif
-
-
-struct prims_t {
- typedef ogles_context_t* GL;
- void (*renderPoint)(GL, vertex_t*);
- void (*renderLine)(GL, vertex_t*, vertex_t*);
- void (*renderTriangle)(GL, vertex_t*, vertex_t*, vertex_t*);
-};
-
-struct ogles_context_t {
- context_t rasterizer;
- array_machine_t arrays __attribute__((aligned(32)));
- texture_state_t textures;
- transform_state_t transforms;
- vertex_cache_t vc;
- prims_t prims;
- culling_t cull;
- lighting_t lighting;
- user_clip_planes_t clipPlanes;
- compute_iterators_t lerp; __attribute__((aligned(32)));
- vertex_t current;
- vec4_t currentColorClamped;
- vec3_t currentNormal;
- viewport_t viewport;
- point_size_t point;
- line_width_t line;
- polygon_offset_t polygonOffset;
- fog_t fog;
- uint32_t perspective : 1;
- uint32_t transformTextures : 1;
- EGLSurfaceManager* surfaceManager;
- EGLBufferObjectManager* bufferObjectManager;
-
- GLenum error;
-
- static inline ogles_context_t* get() {
- return getGlThreadSpecific();
- }
-
-};
-
-}; // namespace gl
-}; // namespace android
-
-#endif // ANDROID_OPENGLES_CONTEXT_H
-
diff --git a/media/common_time/Android.mk b/media/common_time/Android.mk
new file mode 100644
index 0000000..526f17b
--- /dev/null
+++ b/media/common_time/Android.mk
@@ -0,0 +1,21 @@
+LOCAL_PATH:= $(call my-dir)
+#
+# libcommon_time_client
+# (binder marshalers for ICommonClock as well as common clock and local clock
+# helper code)
+#
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := libcommon_time_client
+LOCAL_MODULE_TAGS := optional
+LOCAL_SRC_FILES := cc_helper.cpp \
+ local_clock.cpp \
+ ICommonClock.cpp \
+ ICommonTimeConfig.cpp \
+ utils.cpp
+LOCAL_SHARED_LIBRARIES := libbinder \
+ libhardware \
+ libutils
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/common_time/ICommonClock.cpp b/media/common_time/ICommonClock.cpp
new file mode 100644
index 0000000..28b43ac
--- /dev/null
+++ b/media/common_time/ICommonClock.cpp
@@ -0,0 +1,432 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <linux/socket.h>
+
+#include <common_time/ICommonClock.h>
+#include <binder/Parcel.h>
+
+#include "utils.h"
+
+namespace android {
+
+/***** ICommonClock *****/
+
+enum {
+ IS_COMMON_TIME_VALID = IBinder::FIRST_CALL_TRANSACTION,
+ COMMON_TIME_TO_LOCAL_TIME,
+ LOCAL_TIME_TO_COMMON_TIME,
+ GET_COMMON_TIME,
+ GET_COMMON_FREQ,
+ GET_LOCAL_TIME,
+ GET_LOCAL_FREQ,
+ GET_ESTIMATED_ERROR,
+ GET_TIMELINE_ID,
+ GET_STATE,
+ GET_MASTER_ADDRESS,
+ REGISTER_LISTENER,
+ UNREGISTER_LISTENER,
+};
+
+const String16 ICommonClock::kServiceName("common_time.clock");
+const uint64_t ICommonClock::kInvalidTimelineID = 0;
+const int32_t ICommonClock::kErrorEstimateUnknown = 0x7FFFFFFF;
+
+class BpCommonClock : public BpInterface<ICommonClock>
+{
+ public:
+ BpCommonClock(const sp<IBinder>& impl)
+ : BpInterface<ICommonClock>(impl) {}
+
+ virtual status_t isCommonTimeValid(bool* valid, uint32_t* timelineID) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(IS_COMMON_TIME_VALID,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *valid = reply.readInt32();
+ *timelineID = reply.readInt32();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t commonTimeToLocalTime(int64_t commonTime,
+ int64_t* localTime) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ data.writeInt64(commonTime);
+ status_t status = remote()->transact(COMMON_TIME_TO_LOCAL_TIME,
+ data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *localTime = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t localTimeToCommonTime(int64_t localTime,
+ int64_t* commonTime) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ data.writeInt64(localTime);
+ status_t status = remote()->transact(LOCAL_TIME_TO_COMMON_TIME,
+ data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *commonTime = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getCommonTime(int64_t* commonTime) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_COMMON_TIME, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *commonTime = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getCommonFreq(uint64_t* freq) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_COMMON_FREQ, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *freq = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getLocalTime(int64_t* localTime) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_LOCAL_TIME, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *localTime = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getLocalFreq(uint64_t* freq) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_LOCAL_FREQ, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *freq = reply.readInt64();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getEstimatedError(int32_t* estimate) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_ESTIMATED_ERROR, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *estimate = reply.readInt32();
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getTimelineID(uint64_t* id) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_TIMELINE_ID, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *id = static_cast<uint64_t>(reply.readInt64());
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getState(State* state) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_STATE, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *state = static_cast<State>(reply.readInt32());
+ }
+ }
+ return status;
+ }
+
+ virtual status_t getMasterAddr(struct sockaddr_storage* addr) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ADDRESS, data, &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK)
+ deserializeSockaddr(&reply, addr);
+ }
+ return status;
+ }
+
+ virtual status_t registerListener(
+ const sp<ICommonClockListener>& listener) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ data.writeStrongBinder(listener->asBinder());
+
+ status_t status = remote()->transact(REGISTER_LISTENER, data, &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t unregisterListener(
+ const sp<ICommonClockListener>& listener) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonClock::getInterfaceDescriptor());
+ data.writeStrongBinder(listener->asBinder());
+ status_t status = remote()->transact(UNREGISTER_LISTENER, data, &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+};
+
+IMPLEMENT_META_INTERFACE(CommonClock, "android.os.ICommonClock");
+
+status_t BnCommonClock::onTransact(uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags) {
+ switch(code) {
+ case IS_COMMON_TIME_VALID: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ bool valid;
+ uint32_t timelineID;
+ status_t status = isCommonTimeValid(&valid, &timelineID);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(valid);
+ reply->writeInt32(timelineID);
+ }
+ return OK;
+ } break;
+
+ case COMMON_TIME_TO_LOCAL_TIME: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int64_t commonTime = data.readInt64();
+ int64_t localTime;
+ status_t status = commonTimeToLocalTime(commonTime, &localTime);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(localTime);
+ }
+ return OK;
+ } break;
+
+ case LOCAL_TIME_TO_COMMON_TIME: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int64_t localTime = data.readInt64();
+ int64_t commonTime;
+ status_t status = localTimeToCommonTime(localTime, &commonTime);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(commonTime);
+ }
+ return OK;
+ } break;
+
+ case GET_COMMON_TIME: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int64_t commonTime;
+ status_t status = getCommonTime(&commonTime);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(commonTime);
+ }
+ return OK;
+ } break;
+
+ case GET_COMMON_FREQ: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ uint64_t freq;
+ status_t status = getCommonFreq(&freq);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(freq);
+ }
+ return OK;
+ } break;
+
+ case GET_LOCAL_TIME: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int64_t localTime;
+ status_t status = getLocalTime(&localTime);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(localTime);
+ }
+ return OK;
+ } break;
+
+ case GET_LOCAL_FREQ: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ uint64_t freq;
+ status_t status = getLocalFreq(&freq);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(freq);
+ }
+ return OK;
+ } break;
+
+ case GET_ESTIMATED_ERROR: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ int32_t error;
+ status_t status = getEstimatedError(&error);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(error);
+ }
+ return OK;
+ } break;
+
+ case GET_TIMELINE_ID: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ uint64_t id;
+ status_t status = getTimelineID(&id);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(static_cast<int64_t>(id));
+ }
+ return OK;
+ } break;
+
+ case GET_STATE: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ State state;
+ status_t status = getState(&state);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(static_cast<int32_t>(state));
+ }
+ return OK;
+ } break;
+
+ case GET_MASTER_ADDRESS: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ struct sockaddr_storage addr;
+ status_t status = getMasterAddr(&addr);
+
+ if ((status == OK) && !canSerializeSockaddr(&addr)) {
+ status = UNKNOWN_ERROR;
+ }
+
+ reply->writeInt32(status);
+
+ if (status == OK) {
+ serializeSockaddr(reply, &addr);
+ }
+
+ return OK;
+ } break;
+
+ case REGISTER_LISTENER: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ sp<ICommonClockListener> listener =
+ interface_cast<ICommonClockListener>(data.readStrongBinder());
+ status_t status = registerListener(listener);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case UNREGISTER_LISTENER: {
+ CHECK_INTERFACE(ICommonClock, data, reply);
+ sp<ICommonClockListener> listener =
+ interface_cast<ICommonClockListener>(data.readStrongBinder());
+ status_t status = unregisterListener(listener);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+ }
+ return BBinder::onTransact(code, data, reply, flags);
+}
+
+/***** ICommonClockListener *****/
+
+enum {
+ ON_TIMELINE_CHANGED = IBinder::FIRST_CALL_TRANSACTION,
+};
+
+class BpCommonClockListener : public BpInterface<ICommonClockListener>
+{
+ public:
+ BpCommonClockListener(const sp<IBinder>& impl)
+ : BpInterface<ICommonClockListener>(impl) {}
+
+ virtual void onTimelineChanged(uint64_t timelineID) {
+ Parcel data, reply;
+ data.writeInterfaceToken(
+ ICommonClockListener::getInterfaceDescriptor());
+ data.writeInt64(timelineID);
+ remote()->transact(ON_TIMELINE_CHANGED, data, &reply);
+ }
+};
+
+IMPLEMENT_META_INTERFACE(CommonClockListener,
+ "android.os.ICommonClockListener");
+
+status_t BnCommonClockListener::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) {
+ switch(code) {
+ case ON_TIMELINE_CHANGED: {
+ CHECK_INTERFACE(ICommonClockListener, data, reply);
+ uint32_t timelineID = data.readInt64();
+ onTimelineChanged(timelineID);
+ return NO_ERROR;
+ } break;
+ }
+
+ return BBinder::onTransact(code, data, reply, flags);
+}
+
+}; // namespace android
diff --git a/media/common_time/ICommonTimeConfig.cpp b/media/common_time/ICommonTimeConfig.cpp
new file mode 100644
index 0000000..8eb37cb
--- /dev/null
+++ b/media/common_time/ICommonTimeConfig.cpp
@@ -0,0 +1,508 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <linux/socket.h>
+
+#include <common_time/ICommonTimeConfig.h>
+#include <binder/Parcel.h>
+
+#include "utils.h"
+
+namespace android {
+
+/***** ICommonTimeConfig *****/
+
+enum {
+ GET_MASTER_ELECTION_PRIORITY = IBinder::FIRST_CALL_TRANSACTION,
+ SET_MASTER_ELECTION_PRIORITY,
+ GET_MASTER_ELECTION_ENDPOINT,
+ SET_MASTER_ELECTION_ENDPOINT,
+ GET_MASTER_ELECTION_GROUP_ID,
+ SET_MASTER_ELECTION_GROUP_ID,
+ GET_INTERFACE_BINDING,
+ SET_INTERFACE_BINDING,
+ GET_MASTER_ANNOUNCE_INTERVAL,
+ SET_MASTER_ANNOUNCE_INTERVAL,
+ GET_CLIENT_SYNC_INTERVAL,
+ SET_CLIENT_SYNC_INTERVAL,
+ GET_PANIC_THRESHOLD,
+ SET_PANIC_THRESHOLD,
+ GET_AUTO_DISABLE,
+ SET_AUTO_DISABLE,
+ FORCE_NETWORKLESS_MASTER_MODE,
+};
+
+const String16 ICommonTimeConfig::kServiceName("common_time.config");
+
+class BpCommonTimeConfig : public BpInterface<ICommonTimeConfig>
+{
+ public:
+ BpCommonTimeConfig(const sp<IBinder>& impl)
+ : BpInterface<ICommonTimeConfig>(impl) {}
+
+ virtual status_t getMasterElectionPriority(uint8_t *priority) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ELECTION_PRIORITY,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *priority = static_cast<uint8_t>(reply.readInt32());
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setMasterElectionPriority(uint8_t priority) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(static_cast<int32_t>(priority));
+ status_t status = remote()->transact(SET_MASTER_ELECTION_PRIORITY,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getMasterElectionEndpoint(struct sockaddr_storage *addr) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ELECTION_ENDPOINT,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ deserializeSockaddr(&reply, addr);
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setMasterElectionEndpoint(
+ const struct sockaddr_storage *addr) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ if (!canSerializeSockaddr(addr))
+ return BAD_VALUE;
+ if (NULL == addr) {
+ data.writeInt32(0);
+ } else {
+ data.writeInt32(1);
+ serializeSockaddr(&data, addr);
+ }
+ status_t status = remote()->transact(SET_MASTER_ELECTION_ENDPOINT,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getMasterElectionGroupId(uint64_t *id) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ELECTION_GROUP_ID,
+ data,
+ &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *id = static_cast<uint64_t>(reply.readInt64());
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setMasterElectionGroupId(uint64_t id) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt64(id);
+ status_t status = remote()->transact(SET_MASTER_ELECTION_GROUP_ID,
+ data,
+ &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getInterfaceBinding(String16& ifaceName) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_INTERFACE_BINDING,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ ifaceName = reply.readString16();
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setInterfaceBinding(const String16& ifaceName) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeString16(ifaceName);
+ status_t status = remote()->transact(SET_INTERFACE_BINDING,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getMasterAnnounceInterval(int *interval) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_MASTER_ANNOUNCE_INTERVAL,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *interval = reply.readInt32();
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setMasterAnnounceInterval(int interval) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(interval);
+ status_t status = remote()->transact(SET_MASTER_ANNOUNCE_INTERVAL,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getClientSyncInterval(int *interval) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_CLIENT_SYNC_INTERVAL,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *interval = reply.readInt32();
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setClientSyncInterval(int interval) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(interval);
+ status_t status = remote()->transact(SET_CLIENT_SYNC_INTERVAL,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getPanicThreshold(int *threshold) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_PANIC_THRESHOLD,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *threshold = reply.readInt32();
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setPanicThreshold(int threshold) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(threshold);
+ status_t status = remote()->transact(SET_PANIC_THRESHOLD,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t getAutoDisable(bool *autoDisable) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_AUTO_DISABLE,
+ data,
+ &reply);
+ if (status == OK) {
+ status = reply.readInt32();
+ if (status == OK) {
+ *autoDisable = (0 != reply.readInt32());
+ }
+ }
+
+ return status;
+ }
+
+ virtual status_t setAutoDisable(bool autoDisable) {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ data.writeInt32(autoDisable ? 1 : 0);
+ status_t status = remote()->transact(SET_AUTO_DISABLE,
+ data,
+ &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+
+ virtual status_t forceNetworklessMasterMode() {
+ Parcel data, reply;
+ data.writeInterfaceToken(ICommonTimeConfig::getInterfaceDescriptor());
+ status_t status = remote()->transact(FORCE_NETWORKLESS_MASTER_MODE,
+ data,
+ &reply);
+
+ if (status == OK) {
+ status = reply.readInt32();
+ }
+
+ return status;
+ }
+};
+
+IMPLEMENT_META_INTERFACE(CommonTimeConfig, "android.os.ICommonTimeConfig");
+
+status_t BnCommonTimeConfig::onTransact(uint32_t code,
+ const Parcel& data,
+ Parcel* reply,
+ uint32_t flags) {
+ switch(code) {
+ case GET_MASTER_ELECTION_PRIORITY: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ uint8_t priority;
+ status_t status = getMasterElectionPriority(&priority);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(static_cast<int32_t>(priority));
+ }
+ return OK;
+ } break;
+
+ case SET_MASTER_ELECTION_PRIORITY: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ uint8_t priority = static_cast<uint8_t>(data.readInt32());
+ status_t status = setMasterElectionPriority(priority);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_MASTER_ELECTION_ENDPOINT: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ struct sockaddr_storage addr;
+ status_t status = getMasterElectionEndpoint(&addr);
+
+ if ((status == OK) && !canSerializeSockaddr(&addr)) {
+ status = UNKNOWN_ERROR;
+ }
+
+ reply->writeInt32(status);
+
+ if (status == OK) {
+ serializeSockaddr(reply, &addr);
+ }
+
+ return OK;
+ } break;
+
+ case SET_MASTER_ELECTION_ENDPOINT: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ struct sockaddr_storage addr;
+ int hasAddr = data.readInt32();
+
+ status_t status;
+ if (hasAddr) {
+ deserializeSockaddr(&data, &addr);
+ status = setMasterElectionEndpoint(&addr);
+ } else {
+ status = setMasterElectionEndpoint(&addr);
+ }
+
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_MASTER_ELECTION_GROUP_ID: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ uint64_t id;
+ status_t status = getMasterElectionGroupId(&id);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt64(id);
+ }
+ return OK;
+ } break;
+
+ case SET_MASTER_ELECTION_GROUP_ID: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ uint64_t id = static_cast<uint64_t>(data.readInt64());
+ status_t status = setMasterElectionGroupId(id);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_INTERFACE_BINDING: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ String16 ret;
+ status_t status = getInterfaceBinding(ret);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeString16(ret);
+ }
+ return OK;
+ } break;
+
+ case SET_INTERFACE_BINDING: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ String16 ifaceName;
+ ifaceName = data.readString16();
+ status_t status = setInterfaceBinding(ifaceName);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_MASTER_ANNOUNCE_INTERVAL: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int interval;
+ status_t status = getMasterAnnounceInterval(&interval);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(interval);
+ }
+ return OK;
+ } break;
+
+ case SET_MASTER_ANNOUNCE_INTERVAL: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int interval = data.readInt32();
+ status_t status = setMasterAnnounceInterval(interval);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_CLIENT_SYNC_INTERVAL: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int interval;
+ status_t status = getClientSyncInterval(&interval);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(interval);
+ }
+ return OK;
+ } break;
+
+ case SET_CLIENT_SYNC_INTERVAL: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int interval = data.readInt32();
+ status_t status = setClientSyncInterval(interval);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_PANIC_THRESHOLD: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int threshold;
+ status_t status = getPanicThreshold(&threshold);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(threshold);
+ }
+ return OK;
+ } break;
+
+ case SET_PANIC_THRESHOLD: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ int threshold = data.readInt32();
+ status_t status = setPanicThreshold(threshold);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case GET_AUTO_DISABLE: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ bool autoDisable;
+ status_t status = getAutoDisable(&autoDisable);
+ reply->writeInt32(status);
+ if (status == OK) {
+ reply->writeInt32(autoDisable ? 1 : 0);
+ }
+ return OK;
+ } break;
+
+ case SET_AUTO_DISABLE: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ bool autoDisable = (0 != data.readInt32());
+ status_t status = setAutoDisable(autoDisable);
+ reply->writeInt32(status);
+ return OK;
+ } break;
+
+ case FORCE_NETWORKLESS_MASTER_MODE: {
+ CHECK_INTERFACE(ICommonTimeConfig, data, reply);
+ status_t status = forceNetworklessMasterMode();
+ reply->writeInt32(status);
+ return OK;
+ } break;
+ }
+ return BBinder::onTransact(code, data, reply, flags);
+}
+
+}; // namespace android
+
diff --git a/media/common_time/cc_helper.cpp b/media/common_time/cc_helper.cpp
new file mode 100644
index 0000000..8d8556c
--- /dev/null
+++ b/media/common_time/cc_helper.cpp
@@ -0,0 +1,129 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <stdint.h>
+
+#include <common_time/cc_helper.h>
+#include <common_time/ICommonClock.h>
+#include <utils/threads.h>
+
+namespace android {
+
+Mutex CCHelper::lock_;
+sp<ICommonClock> CCHelper::common_clock_;
+sp<ICommonClockListener> CCHelper::common_clock_listener_;
+uint32_t CCHelper::ref_count_ = 0;
+
+bool CCHelper::verifyClock_l() {
+ bool ret = false;
+
+ if (common_clock_ == NULL) {
+ common_clock_ = ICommonClock::getInstance();
+ if (common_clock_ == NULL)
+ goto bailout;
+ }
+
+ if (ref_count_ > 0) {
+ if (common_clock_listener_ == NULL) {
+ common_clock_listener_ = new CommonClockListener();
+ if (common_clock_listener_ == NULL)
+ goto bailout;
+
+ if (OK != common_clock_->registerListener(common_clock_listener_))
+ goto bailout;
+ }
+ }
+
+ ret = true;
+
+bailout:
+ if (!ret) {
+ common_clock_listener_ = NULL;
+ common_clock_ = NULL;
+ }
+ return ret;
+}
+
+CCHelper::CCHelper() {
+ Mutex::Autolock lock(&lock_);
+ ref_count_++;
+ verifyClock_l();
+}
+
+CCHelper::~CCHelper() {
+ Mutex::Autolock lock(&lock_);
+
+ assert(ref_count_ > 0);
+ ref_count_--;
+
+ // If we were the last CCHelper instance in the system, and we had
+ // previously register a listener, unregister it now so that the common time
+ // service has the chance to go into auto-disabled mode.
+ if (!ref_count_ &&
+ (common_clock_ != NULL) &&
+ (common_clock_listener_ != NULL)) {
+ common_clock_->unregisterListener(common_clock_listener_);
+ common_clock_listener_ = NULL;
+ }
+}
+
+void CCHelper::CommonClockListener::onTimelineChanged(uint64_t timelineID) {
+ // do nothing; listener is only really used as a token so the server can
+ // find out when clients die.
+}
+
+// Helper methods which attempts to make calls to the common time binder
+// service. If the first attempt fails with DEAD_OBJECT, the helpers will
+// attempt to make a connection to the service again (assuming that the process
+// hosting the service had crashed and the client proxy we are holding is dead)
+// If the second attempt fails, or no connection can be made, the we let the
+// error propagate up the stack and let the caller deal with the situation as
+// best they can.
+#define CCHELPER_METHOD(decl, call) \
+ status_t CCHelper::decl { \
+ Mutex::Autolock lock(&lock_); \
+ \
+ if (!verifyClock_l()) \
+ return DEAD_OBJECT; \
+ \
+ status_t status = common_clock_->call; \
+ if (DEAD_OBJECT == status) { \
+ if (!verifyClock_l()) \
+ return DEAD_OBJECT; \
+ status = common_clock_->call; \
+ } \
+ \
+ return status; \
+ }
+
+#define VERIFY_CLOCK()
+
+CCHELPER_METHOD(isCommonTimeValid(bool* valid, uint32_t* timelineID),
+ isCommonTimeValid(valid, timelineID))
+CCHELPER_METHOD(commonTimeToLocalTime(int64_t commonTime, int64_t* localTime),
+ commonTimeToLocalTime(commonTime, localTime))
+CCHELPER_METHOD(localTimeToCommonTime(int64_t localTime, int64_t* commonTime),
+ localTimeToCommonTime(localTime, commonTime))
+CCHELPER_METHOD(getCommonTime(int64_t* commonTime),
+ getCommonTime(commonTime))
+CCHELPER_METHOD(getCommonFreq(uint64_t* freq),
+ getCommonFreq(freq))
+CCHELPER_METHOD(getLocalTime(int64_t* localTime),
+ getLocalTime(localTime))
+CCHELPER_METHOD(getLocalFreq(uint64_t* freq),
+ getLocalFreq(freq))
+
+} // namespace android
diff --git a/media/common_time/local_clock.cpp b/media/common_time/local_clock.cpp
new file mode 100644
index 0000000..a7c61fc
--- /dev/null
+++ b/media/common_time/local_clock.cpp
@@ -0,0 +1,92 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "common_time"
+#include <utils/Log.h>
+
+#include <assert.h>
+#include <stdint.h>
+
+#include <common_time/local_clock.h>
+#include <hardware/hardware.h>
+#include <hardware/local_time_hal.h>
+#include <utils/Errors.h>
+#include <utils/threads.h>
+
+namespace android {
+
+Mutex LocalClock::dev_lock_;
+local_time_hw_device_t* LocalClock::dev_ = NULL;
+
+LocalClock::LocalClock() {
+ int res;
+ const hw_module_t* mod;
+
+ AutoMutex lock(&dev_lock_);
+
+ if (dev_ != NULL)
+ return;
+
+ res = hw_get_module_by_class(LOCAL_TIME_HARDWARE_MODULE_ID, NULL, &mod);
+ if (res) {
+ ALOGE("Failed to open local time HAL module (res = %d)", res);
+ } else {
+ res = local_time_hw_device_open(mod, &dev_);
+ if (res) {
+ ALOGE("Failed to open local time HAL device (res = %d)", res);
+ dev_ = NULL;
+ }
+ }
+}
+
+bool LocalClock::initCheck() {
+ return (NULL != dev_);
+}
+
+int64_t LocalClock::getLocalTime() {
+ assert(NULL != dev_);
+ assert(NULL != dev_->get_local_time);
+
+ return dev_->get_local_time(dev_);
+}
+
+uint64_t LocalClock::getLocalFreq() {
+ assert(NULL != dev_);
+ assert(NULL != dev_->get_local_freq);
+
+ return dev_->get_local_freq(dev_);
+}
+
+status_t LocalClock::setLocalSlew(int16_t rate) {
+ assert(NULL != dev_);
+
+ if (!dev_->set_local_slew)
+ return INVALID_OPERATION;
+
+ return static_cast<status_t>(dev_->set_local_slew(dev_, rate));
+}
+
+int32_t LocalClock::getDebugLog(struct local_time_debug_event* records,
+ int max_records) {
+ assert(NULL != dev_);
+
+ if (!dev_->get_debug_log)
+ return INVALID_OPERATION;
+
+ return dev_->get_debug_log(dev_, records, max_records);
+}
+
+} // namespace android
diff --git a/media/common_time/utils.cpp b/media/common_time/utils.cpp
new file mode 100644
index 0000000..6539171
--- /dev/null
+++ b/media/common_time/utils.cpp
@@ -0,0 +1,89 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <arpa/inet.h>
+#include <linux/socket.h>
+
+#include <binder/Parcel.h>
+
+namespace android {
+
+bool canSerializeSockaddr(const struct sockaddr_storage* addr) {
+ switch (addr->ss_family) {
+ case AF_INET:
+ case AF_INET6:
+ return true;
+ default:
+ return false;
+ }
+}
+
+void serializeSockaddr(Parcel* p, const struct sockaddr_storage* addr) {
+ switch (addr->ss_family) {
+ case AF_INET: {
+ const struct sockaddr_in* s =
+ reinterpret_cast<const struct sockaddr_in*>(addr);
+ p->writeInt32(AF_INET);
+ p->writeInt32(ntohl(s->sin_addr.s_addr));
+ p->writeInt32(static_cast<int32_t>(ntohs(s->sin_port)));
+ } break;
+
+ case AF_INET6: {
+ const struct sockaddr_in6* s =
+ reinterpret_cast<const struct sockaddr_in6*>(addr);
+ const int32_t* a =
+ reinterpret_cast<const int32_t*>(s->sin6_addr.s6_addr);
+ p->writeInt32(AF_INET6);
+ p->writeInt32(ntohl(a[0]));
+ p->writeInt32(ntohl(a[1]));
+ p->writeInt32(ntohl(a[2]));
+ p->writeInt32(ntohl(a[3]));
+ p->writeInt32(static_cast<int32_t>(ntohs(s->sin6_port)));
+ p->writeInt32(ntohl(s->sin6_flowinfo));
+ p->writeInt32(ntohl(s->sin6_scope_id));
+ } break;
+ }
+}
+
+void deserializeSockaddr(const Parcel* p, struct sockaddr_storage* addr) {
+ memset(addr, 0, sizeof(addr));
+
+ addr->ss_family = p->readInt32();
+ switch(addr->ss_family) {
+ case AF_INET: {
+ struct sockaddr_in* s =
+ reinterpret_cast<struct sockaddr_in*>(addr);
+ s->sin_addr.s_addr = htonl(p->readInt32());
+ s->sin_port = htons(static_cast<uint16_t>(p->readInt32()));
+ } break;
+
+ case AF_INET6: {
+ struct sockaddr_in6* s =
+ reinterpret_cast<struct sockaddr_in6*>(addr);
+ int32_t* a = reinterpret_cast<int32_t*>(s->sin6_addr.s6_addr);
+
+ a[0] = htonl(p->readInt32());
+ a[1] = htonl(p->readInt32());
+ a[2] = htonl(p->readInt32());
+ a[3] = htonl(p->readInt32());
+ s->sin6_port = htons(static_cast<uint16_t>(p->readInt32()));
+ s->sin6_flowinfo = htonl(p->readInt32());
+ s->sin6_scope_id = htonl(p->readInt32());
+ } break;
+ }
+}
+
+} // namespace android
diff --git a/media/common_time/utils.h b/media/common_time/utils.h
new file mode 100644
index 0000000..ce79d0d
--- /dev/null
+++ b/media/common_time/utils.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_LIBCOMMONCLOCK_UTILS_H
+#define ANDROID_LIBCOMMONCLOCK_UTILS_H
+
+#include <linux/socket.h>
+
+#include <binder/Parcel.h>
+#include <utils/Errors.h>
+
+namespace android {
+
+extern bool canSerializeSockaddr(const struct sockaddr_storage* addr);
+extern void serializeSockaddr(Parcel* p, const struct sockaddr_storage* addr);
+extern status_t deserializeSockaddr(const Parcel* p,
+ struct sockaddr_storage* addr);
+
+}; // namespace android
+
+#endif // ANDROID_LIBCOMMONCLOCK_UTILS_H
diff --git a/media/libaah_rtp/Android.mk b/media/libaah_rtp/Android.mk
new file mode 100644
index 0000000..54fd9ec
--- /dev/null
+++ b/media/libaah_rtp/Android.mk
@@ -0,0 +1,40 @@
+LOCAL_PATH:= $(call my-dir)
+#
+# libaah_rtp
+#
+
+include $(CLEAR_VARS)
+
+LOCAL_MODULE := libaah_rtp
+LOCAL_MODULE_TAGS := optional
+
+LOCAL_SRC_FILES := \
+ aah_decoder_pump.cpp \
+ aah_rx_player.cpp \
+ aah_rx_player_core.cpp \
+ aah_rx_player_ring_buffer.cpp \
+ aah_rx_player_substream.cpp \
+ aah_tx_packet.cpp \
+ aah_tx_player.cpp \
+ aah_tx_sender.cpp \
+ pipe_event.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/include \
+ frameworks/base/include/media/stagefright/openmax \
+ frameworks/base/media \
+ frameworks/base/media/libstagefright
+
+LOCAL_SHARED_LIBRARIES := \
+ libcommon_time_client \
+ libbinder \
+ libmedia \
+ libstagefright \
+ libstagefright_foundation \
+ libutils
+
+LOCAL_LDLIBS := \
+ -lpthread
+
+include $(BUILD_SHARED_LIBRARY)
+
diff --git a/media/libaah_rtp/aah_decoder_pump.cpp b/media/libaah_rtp/aah_decoder_pump.cpp
new file mode 100644
index 0000000..72fe43b
--- /dev/null
+++ b/media/libaah_rtp/aah_decoder_pump.cpp
@@ -0,0 +1,520 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <poll.h>
+#include <pthread.h>
+
+#include <common_time/cc_helper.h>
+#include <media/AudioSystem.h>
+#include <media/AudioTrack.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OMXClient.h>
+#include <media/stagefright/OMXCodec.h>
+#include <media/stagefright/Utils.h>
+#include <utils/Timers.h>
+#include <utils/threads.h>
+
+#include "aah_decoder_pump.h"
+
+namespace android {
+
+static const long long kLongDecodeErrorThreshold = 1000000ll;
+static const uint32_t kMaxLongErrorsBeforeFatal = 3;
+static const uint32_t kMaxErrorsBeforeFatal = 60;
+
+AAH_DecoderPump::AAH_DecoderPump(OMXClient& omx)
+ : omx_(omx)
+ , thread_status_(OK)
+ , renderer_(NULL)
+ , last_queued_pts_valid_(false)
+ , last_queued_pts_(0)
+ , last_ts_transform_valid_(false)
+ , last_volume_(0xFF) {
+ thread_ = new ThreadWrapper(this);
+}
+
+AAH_DecoderPump::~AAH_DecoderPump() {
+ shutdown();
+}
+
+status_t AAH_DecoderPump::initCheck() {
+ if (thread_ == NULL) {
+ ALOGE("Failed to allocate thread");
+ return NO_MEMORY;
+ }
+
+ return OK;
+}
+
+status_t AAH_DecoderPump::queueForDecode(MediaBuffer* buf) {
+ if (NULL == buf) {
+ return BAD_VALUE;
+ }
+
+ if (OK != thread_status_) {
+ return thread_status_;
+ }
+
+ { // Explicit scope for AutoMutex pattern.
+ AutoMutex lock(&thread_lock_);
+ in_queue_.push_back(buf);
+ }
+
+ thread_cond_.signal();
+
+ return OK;
+}
+
+void AAH_DecoderPump::queueToRenderer(MediaBuffer* decoded_sample) {
+ Mutex::Autolock lock(&render_lock_);
+ sp<MetaData> meta;
+ int64_t ts;
+ status_t res;
+
+ // Fetch the metadata and make sure the sample has a timestamp. We
+ // cannot render samples which are missing PTSs.
+ meta = decoded_sample->meta_data();
+ if ((meta == NULL) || (!meta->findInt64(kKeyTime, &ts))) {
+ ALOGV("Decoded sample missing timestamp, cannot render.");
+ CHECK(false);
+ } else {
+ // If we currently are not holding on to a renderer, go ahead and
+ // make one now.
+ if (NULL == renderer_) {
+ renderer_ = new TimedAudioTrack();
+ if (NULL != renderer_) {
+ int frameCount;
+ AudioTrack::getMinFrameCount(&frameCount,
+ AUDIO_STREAM_DEFAULT,
+ static_cast<int>(format_sample_rate_));
+ int ch_format = (format_channels_ == 1)
+ ? AUDIO_CHANNEL_OUT_MONO
+ : AUDIO_CHANNEL_OUT_STEREO;
+
+ res = renderer_->set(AUDIO_STREAM_DEFAULT,
+ format_sample_rate_,
+ AUDIO_FORMAT_PCM_16_BIT,
+ ch_format,
+ frameCount);
+ if (res != OK) {
+ ALOGE("Failed to setup audio renderer. (res = %d)", res);
+ delete renderer_;
+ renderer_ = NULL;
+ } else {
+ CHECK(last_ts_transform_valid_);
+
+ res = renderer_->setMediaTimeTransform(
+ last_ts_transform_, TimedAudioTrack::COMMON_TIME);
+ if (res != NO_ERROR) {
+ ALOGE("Failed to set media time transform on AudioTrack"
+ " (res = %d)", res);
+ delete renderer_;
+ renderer_ = NULL;
+ } else {
+ float volume = static_cast<float>(last_volume_)
+ / 255.0f;
+ if (renderer_->setVolume(volume, volume) != OK) {
+ ALOGW("%s: setVolume failed", __FUNCTION__);
+ }
+
+ renderer_->start();
+ }
+ }
+ } else {
+ ALOGE("Failed to allocate AudioTrack to use as a renderer.");
+ }
+ }
+
+ if (NULL != renderer_) {
+ uint8_t* decoded_data =
+ reinterpret_cast<uint8_t*>(decoded_sample->data());
+ uint32_t decoded_amt = decoded_sample->range_length();
+ decoded_data += decoded_sample->range_offset();
+
+ sp<IMemory> pcm_payload;
+ res = renderer_->allocateTimedBuffer(decoded_amt, &pcm_payload);
+ if (res != OK) {
+ ALOGE("Failed to allocate %d byte audio track buffer."
+ " (res = %d)", decoded_amt, res);
+ } else {
+ memcpy(pcm_payload->pointer(), decoded_data, decoded_amt);
+
+ res = renderer_->queueTimedBuffer(pcm_payload, ts);
+ if (res != OK) {
+ ALOGE("Failed to queue %d byte audio track buffer with media"
+ " PTS %lld. (res = %d)", decoded_amt, ts, res);
+ } else {
+ last_queued_pts_valid_ = true;
+ last_queued_pts_ = ts;
+ }
+ }
+
+ } else {
+ ALOGE("No renderer, dropping audio payload.");
+ }
+ }
+}
+
+void AAH_DecoderPump::stopAndCleanupRenderer() {
+ if (NULL == renderer_) {
+ return;
+ }
+
+ renderer_->stop();
+ delete renderer_;
+ renderer_ = NULL;
+}
+
+void AAH_DecoderPump::setRenderTSTransform(const LinearTransform& trans) {
+ Mutex::Autolock lock(&render_lock_);
+
+ if (last_ts_transform_valid_ && !memcmp(&trans,
+ &last_ts_transform_,
+ sizeof(trans))) {
+ return;
+ }
+
+ last_ts_transform_ = trans;
+ last_ts_transform_valid_ = true;
+
+ if (NULL != renderer_) {
+ status_t res = renderer_->setMediaTimeTransform(
+ last_ts_transform_, TimedAudioTrack::COMMON_TIME);
+ if (res != NO_ERROR) {
+ ALOGE("Failed to set media time transform on AudioTrack"
+ " (res = %d)", res);
+ }
+ }
+}
+
+void AAH_DecoderPump::setRenderVolume(uint8_t volume) {
+ Mutex::Autolock lock(&render_lock_);
+
+ if (volume == last_volume_) {
+ return;
+ }
+
+ last_volume_ = volume;
+ if (renderer_ != NULL) {
+ float volume = static_cast<float>(last_volume_) / 255.0f;
+ if (renderer_->setVolume(volume, volume) != OK) {
+ ALOGW("%s: setVolume failed", __FUNCTION__);
+ }
+ }
+}
+
+// isAboutToUnderflow is something of a hack used to figure out when it might be
+// time to give up on trying to fill in a gap in the RTP sequence and simply
+// move on with a discontinuity. If we had perfect knowledge of when we were
+// going to underflow, it would not be a hack, but unfortunately we do not.
+// Right now, we just take the PTS of the last sample queued, and check to see
+// if its presentation time is within kAboutToUnderflowThreshold from now. If
+// it is, then we say that we are about to underflow. This decision is based on
+// two (possibly invalid) assumptions.
+//
+// 1) The transmitter is leading the clock by more than
+// kAboutToUnderflowThreshold.
+// 2) The delta between the PTS of the last sample queued and the next sample
+// is less than the transmitter's clock lead amount.
+//
+// Right now, the default transmitter lead time is 1 second, which is a pretty
+// large number and greater than the 50mSec that kAboutToUnderflowThreshold is
+// currently set to. This should satisfy assumption #1 for now, but changes to
+// the transmitter clock lead time could effect this.
+//
+// For non-sparse streams with a homogeneous sample rate (the vast majority of
+// streams in the world), the delta between any two adjacent PTSs will always be
+// the homogeneous sample period. It is very uncommon to see a sample period
+// greater than the 1 second clock lead we are currently using, and you
+// certainly will not see it in an MP3 file which should satisfy assumption #2.
+// Sparse audio streams (where no audio is transmitted for long periods of
+// silence) and extremely low framerate video stream (like an MPEG-2 slideshow
+// or the video stream for a pay TV audio channel) are examples of streams which
+// might violate assumption #2.
+bool AAH_DecoderPump::isAboutToUnderflow(int64_t threshold) {
+ Mutex::Autolock lock(&render_lock_);
+
+ // If we have never queued anything to the decoder, we really don't know if
+ // we are going to underflow or not.
+ if (!last_queued_pts_valid_ || !last_ts_transform_valid_) {
+ return false;
+ }
+
+ // Don't have access to Common Time? If so, then things are Very Bad
+ // elsewhere in the system; it pretty much does not matter what we do here.
+ // Since we cannot really tell if we are about to underflow or not, its
+ // probably best to assume that we are not and proceed accordingly.
+ int64_t tt_now;
+ if (OK != cc_helper_.getCommonTime(&tt_now)) {
+ return false;
+ }
+
+ // Transform from media time to common time.
+ int64_t last_queued_pts_tt;
+ if (!last_ts_transform_.doForwardTransform(last_queued_pts_,
+ &last_queued_pts_tt)) {
+ return false;
+ }
+
+ // Check to see if we are underflowing.
+ return ((tt_now + threshold - last_queued_pts_tt) > 0);
+}
+
+void* AAH_DecoderPump::workThread() {
+ // No need to lock when accessing decoder_ from the thread. The
+ // implementation of init and shutdown ensure that other threads never touch
+ // decoder_ while the work thread is running.
+ CHECK(decoder_ != NULL);
+ CHECK(format_ != NULL);
+
+ // Start the decoder and note its result code. If something goes horribly
+ // wrong, callers of queueForDecode and getOutput will be able to detect
+ // that the thread encountered a fatal error and shut down by examining
+ // thread_status_.
+ thread_status_ = decoder_->start(format_.get());
+ if (OK != thread_status_) {
+ ALOGE("AAH_DecoderPump's work thread failed to start decoder (res = %d)",
+ thread_status_);
+ return NULL;
+ }
+
+ DurationTimer decode_timer;
+ uint32_t consecutive_long_errors = 0;
+ uint32_t consecutive_errors = 0;
+
+ while (!thread_->exitPending()) {
+ status_t res;
+ MediaBuffer* bufOut = NULL;
+
+ decode_timer.start();
+ res = decoder_->read(&bufOut);
+ decode_timer.stop();
+
+ if (res == INFO_FORMAT_CHANGED) {
+ // Format has changed. Destroy our current renderer so that a new
+ // one can be created during queueToRenderer with the proper format.
+ //
+ // TODO : In order to transition seamlessly, we should change this
+ // to put the old renderer in a queue to play out completely before
+ // we destroy it. We can still create a new renderer, the timed
+ // nature of the renderer should ensure a seamless splice.
+ stopAndCleanupRenderer();
+ res = OK;
+ }
+
+ // Try to be a little nuanced in our handling of actual decode errors.
+ // Errors could happen because of minor stream corruption or because of
+ // transient resource limitations. In these cases, we would rather drop
+ // a little bit of output and ride out the unpleasantness then throw up
+ // our hands and abort everything.
+ //
+ // OTOH - When things are really bad (like we have a non-transient
+ // resource or bookkeeping issue, or the stream being fed to us is just
+ // complete and total garbage) we really want to terminate playback and
+ // raise an error condition all the way up to the application level so
+ // they can deal with it.
+ //
+ // Unfortunately, the error codes returned by the decoder can be a
+ // little non-specific. For example, if an OMXCodec times out
+ // attempting to obtain an output buffer, the error we get back is a
+ // generic -1. Try to distinguish between this resource timeout error
+ // and ES corruption error by timing how long the decode operation
+ // takes. Maintain accounting for both errors and "long errors". If we
+ // get more than a certain number consecutive errors of either type,
+ // consider it fatal and shutdown (which will cause the error to
+ // propagate all of the way up to the application level). The threshold
+ // for "long errors" is deliberately much lower than that of normal
+ // decode errors, both because of how long they take to happen and
+ // because they generally indicate resource limitation errors which are
+ // unlikely to go away in pathologically bad cases (in contrast to
+ // stream corruption errors which might happen 20 times in a row and
+ // then be suddenly OK again)
+ if (res != OK) {
+ consecutive_errors++;
+ if (decode_timer.durationUsecs() >= kLongDecodeErrorThreshold)
+ consecutive_long_errors++;
+
+ CHECK(NULL == bufOut);
+
+ ALOGW("%s: Failed to decode data (res = %d)",
+ __PRETTY_FUNCTION__, res);
+
+ if ((consecutive_errors >= kMaxErrorsBeforeFatal) ||
+ (consecutive_long_errors >= kMaxLongErrorsBeforeFatal)) {
+ ALOGE("%s: Maximum decode error threshold has been reached."
+ " There have been %d consecutive decode errors, and %d"
+ " consecutive decode operations which resulted in errors"
+ " and took more than %lld uSec to process. The last"
+ " decode operation took %lld uSec.",
+ __PRETTY_FUNCTION__,
+ consecutive_errors, consecutive_long_errors,
+ kLongDecodeErrorThreshold, decode_timer.durationUsecs());
+ thread_status_ = res;
+ break;
+ }
+
+ continue;
+ }
+
+ if (NULL == bufOut) {
+ ALOGW("%s: Successful decode, but no buffer produced",
+ __PRETTY_FUNCTION__);
+ continue;
+ }
+
+ // Successful decode (with actual output produced). Clear the error
+ // counters.
+ consecutive_errors = 0;
+ consecutive_long_errors = 0;
+
+ queueToRenderer(bufOut);
+ bufOut->release();
+ }
+
+ decoder_->stop();
+ stopAndCleanupRenderer();
+
+ return NULL;
+}
+
+status_t AAH_DecoderPump::init(const sp<MetaData>& params) {
+ Mutex::Autolock lock(&init_lock_);
+
+ if (decoder_ != NULL) {
+ // already inited
+ return OK;
+ }
+
+ if (params == NULL) {
+ return BAD_VALUE;
+ }
+
+ if (!params->findInt32(kKeyChannelCount, &format_channels_)) {
+ return BAD_VALUE;
+ }
+
+ if (!params->findInt32(kKeySampleRate, &format_sample_rate_)) {
+ return BAD_VALUE;
+ }
+
+ CHECK(OK == thread_status_);
+ CHECK(decoder_ == NULL);
+
+ status_t ret_val = UNKNOWN_ERROR;
+
+ // Cache the format and attempt to create the decoder.
+ format_ = params;
+ decoder_ = OMXCodec::Create(
+ omx_.interface(), // IOMX Handle
+ format_, // Metadata for substream (indicates codec)
+ false, // Make a decoder, not an encoder
+ sp<MediaSource>(this)); // We will be the source for this codec.
+
+ if (decoder_ == NULL) {
+ ALOGE("Failed to allocate decoder in %s", __PRETTY_FUNCTION__);
+ goto bailout;
+ }
+
+ // Fire up the pump thread. It will take care of starting and stopping the
+ // decoder.
+ ret_val = thread_->run("aah_decode_pump", ANDROID_PRIORITY_AUDIO);
+ if (OK != ret_val) {
+ ALOGE("Failed to start work thread in %s (res = %d)",
+ __PRETTY_FUNCTION__, ret_val);
+ goto bailout;
+ }
+
+bailout:
+ if (OK != ret_val) {
+ decoder_ = NULL;
+ format_ = NULL;
+ }
+
+ return OK;
+}
+
+status_t AAH_DecoderPump::shutdown() {
+ Mutex::Autolock lock(&init_lock_);
+ return shutdown_l();
+}
+
+status_t AAH_DecoderPump::shutdown_l() {
+ thread_->requestExit();
+ thread_cond_.signal();
+ thread_->requestExitAndWait();
+
+ for (MBQueue::iterator iter = in_queue_.begin();
+ iter != in_queue_.end();
+ ++iter) {
+ (*iter)->release();
+ }
+ in_queue_.clear();
+
+ last_queued_pts_valid_ = false;
+ last_ts_transform_valid_ = false;
+ last_volume_ = 0xFF;
+ thread_status_ = OK;
+
+ decoder_ = NULL;
+ format_ = NULL;
+
+ return OK;
+}
+
+status_t AAH_DecoderPump::read(MediaBuffer **buffer,
+ const ReadOptions *options) {
+ if (!buffer) {
+ return BAD_VALUE;
+ }
+
+ *buffer = NULL;
+
+ // While its not time to shut down, and we have no data to process, wait.
+ AutoMutex lock(&thread_lock_);
+ while (!thread_->exitPending() && in_queue_.empty())
+ thread_cond_.wait(thread_lock_);
+
+ // At this point, if its not time to shutdown then we must have something to
+ // process. Go ahead and pop the front of the queue for processing.
+ if (!thread_->exitPending()) {
+ CHECK(!in_queue_.empty());
+
+ *buffer = *(in_queue_.begin());
+ in_queue_.erase(in_queue_.begin());
+ }
+
+ // If we managed to get a buffer, then everything must be OK. If not, then
+ // we must be shutting down.
+ return (NULL == *buffer) ? INVALID_OPERATION : OK;
+}
+
+AAH_DecoderPump::ThreadWrapper::ThreadWrapper(AAH_DecoderPump* owner)
+ : Thread(false /* canCallJava*/ )
+ , owner_(owner) {
+}
+
+bool AAH_DecoderPump::ThreadWrapper::threadLoop() {
+ CHECK(NULL != owner_);
+ owner_->workThread();
+ return false;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_decoder_pump.h b/media/libaah_rtp/aah_decoder_pump.h
new file mode 100644
index 0000000..f5a6529
--- /dev/null
+++ b/media/libaah_rtp/aah_decoder_pump.h
@@ -0,0 +1,107 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __DECODER_PUMP_H__
+#define __DECODER_PUMP_H__
+
+#include <pthread.h>
+
+#include <common_time/cc_helper.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/LinearTransform.h>
+#include <utils/List.h>
+#include <utils/threads.h>
+
+namespace android {
+
+class MetaData;
+class OMXClient;
+class TimedAudioTrack;
+
+class AAH_DecoderPump : public MediaSource {
+ public:
+ explicit AAH_DecoderPump(OMXClient& omx);
+ status_t initCheck();
+
+ status_t queueForDecode(MediaBuffer* buf);
+
+ status_t init(const sp<MetaData>& params);
+ status_t shutdown();
+
+ void setRenderTSTransform(const LinearTransform& trans);
+ void setRenderVolume(uint8_t volume);
+ bool isAboutToUnderflow(int64_t threshold);
+ bool getStatus() const { return thread_status_; }
+
+ // MediaSource methods
+ virtual status_t start(MetaData *params) { return OK; }
+ virtual sp<MetaData> getFormat() { return format_; }
+ virtual status_t stop() { return OK; }
+ virtual status_t read(MediaBuffer **buffer,
+ const ReadOptions *options);
+
+ protected:
+ virtual ~AAH_DecoderPump();
+
+ private:
+ class ThreadWrapper : public Thread {
+ public:
+ friend class AAH_DecoderPump;
+ explicit ThreadWrapper(AAH_DecoderPump* owner);
+
+ private:
+ virtual bool threadLoop();
+ AAH_DecoderPump* owner_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(ThreadWrapper);
+ };
+
+ void* workThread();
+ virtual status_t shutdown_l();
+ void queueToRenderer(MediaBuffer* decoded_sample);
+ void stopAndCleanupRenderer();
+
+ sp<MetaData> format_;
+ int32_t format_channels_;
+ int32_t format_sample_rate_;
+
+ sp<MediaSource> decoder_;
+ OMXClient& omx_;
+ Mutex init_lock_;
+
+ sp<ThreadWrapper> thread_;
+ Condition thread_cond_;
+ Mutex thread_lock_;
+ status_t thread_status_;
+
+ Mutex render_lock_;
+ TimedAudioTrack* renderer_;
+ bool last_queued_pts_valid_;
+ int64_t last_queued_pts_;
+ bool last_ts_transform_valid_;
+ LinearTransform last_ts_transform_;
+ uint8_t last_volume_;
+ CCHelper cc_helper_;
+
+ // protected by the thread_lock_
+ typedef List<MediaBuffer*> MBQueue;
+ MBQueue in_queue_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_DecoderPump);
+};
+
+} // namespace android
+#endif // __DECODER_PUMP_H__
diff --git a/media/libaah_rtp/aah_rx_player.cpp b/media/libaah_rtp/aah_rx_player.cpp
new file mode 100644
index 0000000..9dd79fd
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player.cpp
@@ -0,0 +1,288 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+
+#include <binder/IServiceManager.h>
+#include <media/MediaPlayerInterface.h>
+#include <utils/Log.h>
+
+#include "aah_rx_player.h"
+
+namespace android {
+
+const uint32_t AAH_RXPlayer::kRTPRingBufferSize = 1 << 10;
+
+sp<MediaPlayerBase> createAAH_RXPlayer() {
+ sp<MediaPlayerBase> ret = new AAH_RXPlayer();
+ return ret;
+}
+
+AAH_RXPlayer::AAH_RXPlayer()
+ : ring_buffer_(kRTPRingBufferSize)
+ , substreams_(NULL) {
+ thread_wrapper_ = new ThreadWrapper(*this);
+
+ is_playing_ = false;
+ multicast_joined_ = false;
+ transmitter_known_ = false;
+ current_epoch_known_ = false;
+ data_source_set_ = false;
+ sock_fd_ = -1;
+
+ substreams_.setCapacity(4);
+
+ memset(&listen_addr_, 0, sizeof(listen_addr_));
+ memset(&transmitter_addr_, 0, sizeof(transmitter_addr_));
+
+ fetchAudioFlinger();
+}
+
+AAH_RXPlayer::~AAH_RXPlayer() {
+ reset_l();
+ CHECK(substreams_.size() == 0);
+ omx_.disconnect();
+}
+
+status_t AAH_RXPlayer::initCheck() {
+ if (thread_wrapper_ == NULL) {
+ ALOGE("Failed to allocate thread wrapper!");
+ return NO_MEMORY;
+ }
+
+ if (!ring_buffer_.initCheck()) {
+ ALOGE("Failed to allocate reassembly ring buffer!");
+ return NO_MEMORY;
+ }
+
+ // Check for the presense of the common time service by attempting to query
+ // for CommonTime's frequency. If we get an error back, we cannot talk to
+ // the service at all and should abort now.
+ status_t res;
+ uint64_t freq;
+ res = cc_helper_.getCommonFreq(&freq);
+ if (OK != res) {
+ ALOGE("Failed to connect to common time service!");
+ return res;
+ }
+
+ return omx_.connect();
+}
+
+status_t AAH_RXPlayer::setDataSource(
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ AutoMutex api_lock(&api_lock_);
+ uint32_t a, b, c, d;
+ uint16_t port;
+
+ if (data_source_set_) {
+ return INVALID_OPERATION;
+ }
+
+ if (NULL == url) {
+ return BAD_VALUE;
+ }
+
+ if (5 != sscanf(url, "%*[^:/]://%u.%u.%u.%u:%hu", &a, &b, &c, &d, &port)) {
+ ALOGE("Failed to parse URL \"%s\"", url);
+ return BAD_VALUE;
+ }
+
+ if ((a > 255) || (b > 255) || (c > 255) || (d > 255) || (port == 0)) {
+ ALOGE("Bad multicast address \"%s\"", url);
+ return BAD_VALUE;
+ }
+
+ ALOGI("setDataSource :: %u.%u.%u.%u:%hu", a, b, c, d, port);
+
+ a = (a << 24) | (b << 16) | (c << 8) | d;
+
+ memset(&listen_addr_, 0, sizeof(listen_addr_));
+ listen_addr_.sin_family = AF_INET;
+ listen_addr_.sin_port = htons(port);
+ listen_addr_.sin_addr.s_addr = htonl(a);
+ data_source_set_ = true;
+
+ return OK;
+}
+
+status_t AAH_RXPlayer::setDataSource(int fd, int64_t offset, int64_t length) {
+ return INVALID_OPERATION;
+}
+
+status_t AAH_RXPlayer::setVideoSurface(const sp<Surface>& surface) {
+ return OK;
+}
+
+status_t AAH_RXPlayer::setVideoSurfaceTexture(
+ const sp<ISurfaceTexture>& surfaceTexture) {
+ return OK;
+}
+
+status_t AAH_RXPlayer::prepare() {
+ return OK;
+}
+
+status_t AAH_RXPlayer::prepareAsync() {
+ sendEvent(MEDIA_PREPARED);
+ return OK;
+}
+
+status_t AAH_RXPlayer::start() {
+ AutoMutex api_lock(&api_lock_);
+
+ if (is_playing_) {
+ return OK;
+ }
+
+ status_t res = startWorkThread();
+ is_playing_ = (res == OK);
+ return res;
+}
+
+status_t AAH_RXPlayer::stop() {
+ return pause();
+}
+
+status_t AAH_RXPlayer::pause() {
+ AutoMutex api_lock(&api_lock_);
+ stopWorkThread();
+ CHECK(sock_fd_ < 0);
+ is_playing_ = false;
+ return OK;
+}
+
+bool AAH_RXPlayer::isPlaying() {
+ AutoMutex api_lock(&api_lock_);
+ return is_playing_;
+}
+
+status_t AAH_RXPlayer::seekTo(int msec) {
+ sendEvent(MEDIA_SEEK_COMPLETE);
+ return OK;
+}
+
+status_t AAH_RXPlayer::getCurrentPosition(int *msec) {
+ if (NULL != msec) {
+ *msec = 0;
+ }
+ return OK;
+}
+
+status_t AAH_RXPlayer::getDuration(int *msec) {
+ if (NULL != msec) {
+ *msec = 1;
+ }
+ return OK;
+}
+
+status_t AAH_RXPlayer::reset() {
+ AutoMutex api_lock(&api_lock_);
+ reset_l();
+ return OK;
+}
+
+void AAH_RXPlayer::reset_l() {
+ stopWorkThread();
+ CHECK(sock_fd_ < 0);
+ CHECK(!multicast_joined_);
+ is_playing_ = false;
+ data_source_set_ = false;
+ transmitter_known_ = false;
+ memset(&listen_addr_, 0, sizeof(listen_addr_));
+}
+
+status_t AAH_RXPlayer::setLooping(int loop) {
+ return OK;
+}
+
+player_type AAH_RXPlayer::playerType() {
+ return AAH_RX_PLAYER;
+}
+
+status_t AAH_RXPlayer::setParameter(int key, const Parcel &request) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_RXPlayer::getParameter(int key, Parcel *reply) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_RXPlayer::invoke(const Parcel& request, Parcel *reply) {
+ if (!reply) {
+ return BAD_VALUE;
+ }
+
+ int32_t magic;
+ status_t err = request.readInt32(&magic);
+ if (err != OK) {
+ reply->writeInt32(err);
+ return OK;
+ }
+
+ if (magic != 0x12345) {
+ reply->writeInt32(BAD_VALUE);
+ return OK;
+ }
+
+ int32_t methodID;
+ err = request.readInt32(&methodID);
+ if (err != OK) {
+ reply->writeInt32(err);
+ return OK;
+ }
+
+ switch (methodID) {
+ // Get Volume
+ case INVOKE_GET_MASTER_VOLUME: {
+ if (audio_flinger_ != NULL) {
+ reply->writeInt32(OK);
+ reply->writeFloat(audio_flinger_->masterVolume());
+ } else {
+ reply->writeInt32(UNKNOWN_ERROR);
+ }
+ } break;
+
+ // Set Volume
+ case INVOKE_SET_MASTER_VOLUME: {
+ float targetVol = request.readFloat();
+ reply->writeInt32(audio_flinger_->setMasterVolume(targetVol));
+ } break;
+
+ default: return BAD_VALUE;
+ }
+
+ return OK;
+}
+
+void AAH_RXPlayer::fetchAudioFlinger() {
+ if (audio_flinger_ == NULL) {
+ sp<IServiceManager> sm = defaultServiceManager();
+ sp<IBinder> binder;
+ binder = sm->getService(String16("media.audio_flinger"));
+
+ if (binder == NULL) {
+ ALOGW("AAH_RXPlayer failed to fetch handle to audio flinger."
+ " Master volume control will not be possible.");
+ }
+
+ audio_flinger_ = interface_cast<IAudioFlinger>(binder);
+ }
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_rx_player.h b/media/libaah_rtp/aah_rx_player.h
new file mode 100644
index 0000000..7a1b6e3
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player.h
@@ -0,0 +1,313 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AAH_RX_PLAYER_H__
+#define __AAH_RX_PLAYER_H__
+
+#include <common_time/cc_helper.h>
+#include <media/MediaPlayerInterface.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OMXClient.h>
+#include <netinet/in.h>
+#include <utils/KeyedVector.h>
+#include <utils/LinearTransform.h>
+#include <utils/threads.h>
+
+#include "aah_decoder_pump.h"
+#include "pipe_event.h"
+
+namespace android {
+
+class AAH_RXPlayer : public MediaPlayerInterface {
+ public:
+ AAH_RXPlayer();
+
+ virtual status_t initCheck();
+ virtual status_t setDataSource(const char *url,
+ const KeyedVector<String8, String8>*
+ headers);
+ virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
+ virtual status_t setVideoSurface(const sp<Surface>& surface);
+ virtual status_t setVideoSurfaceTexture(const sp<ISurfaceTexture>&
+ surfaceTexture);
+ virtual status_t prepare();
+ virtual status_t prepareAsync();
+ virtual status_t start();
+ virtual status_t stop();
+ virtual status_t pause();
+ virtual bool isPlaying();
+ virtual status_t seekTo(int msec);
+ virtual status_t getCurrentPosition(int *msec);
+ virtual status_t getDuration(int *msec);
+ virtual status_t reset();
+ virtual status_t setLooping(int loop);
+ virtual player_type playerType();
+ virtual status_t setParameter(int key, const Parcel &request);
+ virtual status_t getParameter(int key, Parcel *reply);
+ virtual status_t invoke(const Parcel& request, Parcel *reply);
+
+ protected:
+ virtual ~AAH_RXPlayer();
+
+ private:
+ class ThreadWrapper : public Thread {
+ public:
+ friend class AAH_RXPlayer;
+ explicit ThreadWrapper(AAH_RXPlayer& player)
+ : Thread(false /* canCallJava */ )
+ , player_(player) { }
+
+ virtual bool threadLoop() { return player_.threadLoop(); }
+
+ private:
+ AAH_RXPlayer& player_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(ThreadWrapper);
+ };
+
+#pragma pack(push, 1)
+ // PacketBuffers are structures used by the RX ring buffer. The ring buffer
+ // is a ring of pointers to PacketBuffer structures which act as variable
+ // length byte arrays and hold the contents of received UDP packets. Rather
+ // than make this a structure which hold a length and a pointer to another
+ // allocated structure (which would require two allocations), this struct
+ // uses a structure overlay pattern where allocation for the byte array
+ // consists of allocating (arrayLen + sizeof(ssize_t)) bytes of data from
+ // whatever pool/heap the packet buffer pulls from, and then overlaying the
+ // packed PacketBuffer structure on top of the allocation. The one-byte
+ // array at the end of the structure serves as an offset to the the data
+ // portion of the allocation; packet buffers are never allocated on the
+ // stack or using the new operator. Instead, the static allocate-byte-array
+ // and destroy methods handle the allocate and overlay pattern. They also
+ // allow for a potential future optimization where instead of just
+ // allocating blocks from the process global heap and overlaying, the
+ // allocator is replaced with a different implementation (private heap,
+ // free-list, circular buffer, etc) which reduces potential heap
+ // fragmentation issues which might arise from the frequent allocation and
+ // destruction of the received UDP traffic.
+ struct PacketBuffer {
+ ssize_t length_;
+ uint8_t data_[1];
+
+ // TODO : consider changing this to be some form of ring buffer or free
+ // pool system instead of just using the heap in order to avoid heap
+ // fragmentation.
+ static PacketBuffer* allocate(ssize_t length);
+ static void destroy(PacketBuffer* pb);
+
+ private:
+ // Force people to use allocate/destroy instead of new/delete.
+ PacketBuffer() { }
+ ~PacketBuffer() { }
+ };
+
+ struct RetransRequest {
+ uint32_t magic_;
+ uint32_t mcast_ip_;
+ uint16_t mcast_port_;
+ uint16_t start_seq_;
+ uint16_t end_seq_;
+ };
+#pragma pack(pop)
+
+ enum GapStatus {
+ kGS_NoGap = 0,
+ kGS_NormalGap,
+ kGS_FastStartGap,
+ };
+
+ struct SeqNoGap {
+ uint16_t start_seq_;
+ uint16_t end_seq_;
+ };
+
+ class RXRingBuffer {
+ public:
+ explicit RXRingBuffer(uint32_t capacity);
+ ~RXRingBuffer();
+
+ bool initCheck() const { return (ring_ != NULL); }
+ void reset();
+
+ // Push a packet buffer with a given sequence number into the ring
+ // buffer. pushBuffer will always consume the buffer pushed to it,
+ // either destroying it because it was a duplicate or overflow, or
+ // holding on to it in the ring. Callers should not hold any references
+ // to PacketBuffers after they have been pushed to the ring. Returns
+ // false in the case of a serious error (such as ring overflow).
+ // Callers should consider resetting the pipeline entirely in the event
+ // of a serious error.
+ bool pushBuffer(PacketBuffer* buf, uint16_t seq);
+
+ // Fetch the next buffer in the RTP sequence. Returns NULL if there is
+ // no buffer to fetch. If a non-NULL PacketBuffer is returned,
+ // is_discon will be set to indicate whether or not this PacketBuffer is
+ // discontiuous with any previously returned packet buffers. Packet
+ // buffers returned by fetchBuffer are the caller's responsibility; they
+ // must be certain to destroy the buffers when they are done.
+ PacketBuffer* fetchBuffer(bool* is_discon);
+
+ // Returns true and fills out the gap structure if the read pointer of
+ // the ring buffer is currently pointing to a gap which would stall a
+ // fetchBuffer operation. Returns false if the read pointer is not
+ // pointing to a gap in the sequence currently.
+ GapStatus fetchCurrentGap(SeqNoGap* gap);
+
+ // Causes the read pointer to skip over any portion of a gap indicated
+ // by nak. If nak is NULL, any gap currently blocking the read pointer
+ // will be completely skipped. If any portion of a gap is skipped, the
+ // next successful read from fetch buffer will indicate a discontinuity.
+ void processNAK(const SeqNoGap* nak = NULL);
+
+ // Compute the number of milliseconds until the inactivity timer for
+ // this RTP stream. Returns -1 if there is no active timeout, or 0 if
+ // the system has already timed out.
+ int computeInactivityTimeout();
+
+ private:
+ Mutex lock_;
+ PacketBuffer** ring_;
+ uint32_t capacity_;
+ uint32_t rd_;
+ uint32_t wr_;
+
+ uint16_t rd_seq_;
+ bool rd_seq_known_;
+ bool waiting_for_fast_start_;
+ bool fetched_first_packet_;
+
+ uint64_t rtp_activity_timeout_;
+ bool rtp_activity_timeout_valid_;
+
+ DISALLOW_EVIL_CONSTRUCTORS(RXRingBuffer);
+ };
+
+ class Substream : public virtual RefBase {
+ public:
+ Substream(uint32_t ssrc, OMXClient& omx);
+
+ void cleanupBufferInProgress();
+ void shutdown();
+ void processPayloadStart(uint8_t* buf,
+ uint32_t amt,
+ int32_t ts_lower);
+ void processPayloadCont (uint8_t* buf,
+ uint32_t amt);
+ void processTSTransform(const LinearTransform& trans);
+
+ bool isAboutToUnderflow();
+ uint32_t getSSRC() const { return ssrc_; }
+ uint16_t getProgramID() const { return (ssrc_ >> 5) & 0x1F; }
+ status_t getStatus() const { return status_; }
+
+ protected:
+ virtual ~Substream() {
+ shutdown();
+ }
+
+ private:
+ void cleanupDecoder();
+ bool shouldAbort(const char* log_tag);
+ void processCompletedBuffer();
+ bool setupSubstreamType(uint8_t substream_type,
+ uint8_t codec_type);
+
+ uint32_t ssrc_;
+ bool waiting_for_rap_;
+ status_t status_;
+
+ bool substream_details_known_;
+ uint8_t substream_type_;
+ uint8_t codec_type_;
+ sp<MetaData> substream_meta_;
+
+ MediaBuffer* buffer_in_progress_;
+ uint32_t expected_buffer_size_;
+ uint32_t buffer_filled_;
+
+ sp<AAH_DecoderPump> decoder_;
+
+ static int64_t kAboutToUnderflowThreshold;
+
+ DISALLOW_EVIL_CONSTRUCTORS(Substream);
+ };
+
+ typedef DefaultKeyedVector< uint32_t, sp<Substream> > SubstreamVec;
+
+ status_t startWorkThread();
+ void stopWorkThread();
+ virtual bool threadLoop();
+ bool setupSocket();
+ void cleanupSocket();
+ void resetPipeline();
+ void reset_l();
+ bool processRX(PacketBuffer* pb);
+ void processRingBuffer();
+ void processCommandPacket(PacketBuffer* pb);
+ bool processGaps();
+ int computeNextGapRetransmitTimeout();
+ void fetchAudioFlinger();
+
+ PipeEvent wakeup_work_thread_evt_;
+ sp<ThreadWrapper> thread_wrapper_;
+ Mutex api_lock_;
+ bool is_playing_;
+ bool data_source_set_;
+
+ struct sockaddr_in listen_addr_;
+ int sock_fd_;
+ bool multicast_joined_;
+
+ struct sockaddr_in transmitter_addr_;
+ bool transmitter_known_;
+
+ uint32_t current_epoch_;
+ bool current_epoch_known_;
+
+ SeqNoGap current_gap_;
+ GapStatus current_gap_status_;
+ uint64_t next_retrans_req_time_;
+
+ RXRingBuffer ring_buffer_;
+ SubstreamVec substreams_;
+ OMXClient omx_;
+ CCHelper cc_helper_;
+
+ // Connection to audio flinger used to hack a path to setMasterVolume.
+ sp<IAudioFlinger> audio_flinger_;
+
+ static const uint32_t kRTPRingBufferSize;
+ static const uint32_t kRetransRequestMagic;
+ static const uint32_t kFastStartRequestMagic;
+ static const uint32_t kRetransNAKMagic;
+ static const uint32_t kGapRerequestTimeoutUSec;
+ static const uint32_t kFastStartTimeoutUSec;
+ static const uint32_t kRTPActivityTimeoutUSec;
+
+ static const uint32_t INVOKE_GET_MASTER_VOLUME = 3;
+ static const uint32_t INVOKE_SET_MASTER_VOLUME = 4;
+
+ static uint64_t monotonicUSecNow();
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_RXPlayer);
+};
+
+} // namespace android
+
+#endif // __AAH_RX_PLAYER_H__
diff --git a/media/libaah_rtp/aah_rx_player_core.cpp b/media/libaah_rtp/aah_rx_player_core.cpp
new file mode 100644
index 0000000..d2b3386
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_core.cpp
@@ -0,0 +1,807 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include <fcntl.h>
+#include <poll.h>
+#include <sys/socket.h>
+#include <time.h>
+#include <utils/misc.h>
+
+#include <media/stagefright/Utils.h>
+
+#include "aah_rx_player.h"
+#include "aah_tx_packet.h"
+
+namespace android {
+
+const uint32_t AAH_RXPlayer::kRetransRequestMagic =
+ FOURCC('T','r','e','q');
+const uint32_t AAH_RXPlayer::kRetransNAKMagic =
+ FOURCC('T','n','a','k');
+const uint32_t AAH_RXPlayer::kFastStartRequestMagic =
+ FOURCC('T','f','s','t');
+const uint32_t AAH_RXPlayer::kGapRerequestTimeoutUSec = 75000;
+const uint32_t AAH_RXPlayer::kFastStartTimeoutUSec = 800000;
+const uint32_t AAH_RXPlayer::kRTPActivityTimeoutUSec = 10000000;
+
+static inline int16_t fetchInt16(uint8_t* data) {
+ return static_cast<int16_t>(U16_AT(data));
+}
+
+static inline int32_t fetchInt32(uint8_t* data) {
+ return static_cast<int32_t>(U32_AT(data));
+}
+
+static inline int64_t fetchInt64(uint8_t* data) {
+ return static_cast<int64_t>(U64_AT(data));
+}
+
+uint64_t AAH_RXPlayer::monotonicUSecNow() {
+ struct timespec now;
+ int res = clock_gettime(CLOCK_MONOTONIC, &now);
+ CHECK(res >= 0);
+
+ uint64_t ret = static_cast<uint64_t>(now.tv_sec) * 1000000;
+ ret += now.tv_nsec / 1000;
+
+ return ret;
+}
+
+status_t AAH_RXPlayer::startWorkThread() {
+ status_t res;
+ stopWorkThread();
+ res = thread_wrapper_->run("TRX_Player", PRIORITY_AUDIO);
+
+ if (res != OK) {
+ ALOGE("Failed to start work thread (res = %d)", res);
+ }
+
+ return res;
+}
+
+void AAH_RXPlayer::stopWorkThread() {
+ thread_wrapper_->requestExit(); // set the exit pending flag
+ wakeup_work_thread_evt_.setEvent();
+
+ status_t res;
+ res = thread_wrapper_->requestExitAndWait(); // block until thread exit.
+ if (res != OK) {
+ ALOGE("Failed to stop work thread (res = %d)", res);
+ }
+
+ wakeup_work_thread_evt_.clearPendingEvents();
+}
+
+void AAH_RXPlayer::cleanupSocket() {
+ if (sock_fd_ >= 0) {
+ if (multicast_joined_) {
+ int res;
+ struct ip_mreq mreq;
+ mreq.imr_multiaddr = listen_addr_.sin_addr;
+ mreq.imr_interface.s_addr = htonl(INADDR_ANY);
+ res = setsockopt(sock_fd_,
+ IPPROTO_IP,
+ IP_DROP_MEMBERSHIP,
+ &mreq, sizeof(mreq));
+ if (res < 0) {
+ ALOGW("Failed to leave multicast group. (%d, %d)", res, errno);
+ }
+ multicast_joined_ = false;
+ }
+
+ close(sock_fd_);
+ sock_fd_ = -1;
+ }
+
+ resetPipeline();
+}
+
+void AAH_RXPlayer::resetPipeline() {
+ ring_buffer_.reset();
+
+ // Explicitly shudown all of the active substreams, then call clear out the
+ // collection. Failure to clear out a substream can result in its decoder
+ // holding a reference to itself and therefor not going away when the
+ // collection is cleared.
+ for (size_t i = 0; i < substreams_.size(); ++i)
+ substreams_.valueAt(i)->shutdown();
+
+ substreams_.clear();
+
+ current_gap_status_ = kGS_NoGap;
+}
+
+bool AAH_RXPlayer::setupSocket() {
+ long flags;
+ int res, buf_size;
+ socklen_t opt_size;
+
+ cleanupSocket();
+ CHECK(sock_fd_ < 0);
+
+ // Make the socket
+ sock_fd_ = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
+ if (sock_fd_ < 0) {
+ ALOGE("Failed to create listen socket (errno %d)", errno);
+ goto bailout;
+ }
+
+ // Set non-blocking operation
+ flags = fcntl(sock_fd_, F_GETFL);
+ res = fcntl(sock_fd_, F_SETFL, flags | O_NONBLOCK);
+ if (res < 0) {
+ ALOGE("Failed to set socket (%d) to non-blocking mode (errno %d)",
+ sock_fd_, errno);
+ goto bailout;
+ }
+
+ // Bind to our port
+ struct sockaddr_in bind_addr;
+ memset(&bind_addr, 0, sizeof(bind_addr));
+ bind_addr.sin_family = AF_INET;
+ bind_addr.sin_addr.s_addr = INADDR_ANY;
+ bind_addr.sin_port = listen_addr_.sin_port;
+ res = bind(sock_fd_,
+ reinterpret_cast<const sockaddr*>(&bind_addr),
+ sizeof(bind_addr));
+ if (res < 0) {
+ uint32_t a = ntohl(bind_addr.sin_addr.s_addr);
+ uint16_t p = ntohs(bind_addr.sin_port);
+ ALOGE("Failed to bind socket (%d) to %d.%d.%d.%d:%hd. (errno %d)",
+ sock_fd_,
+ (a >> 24) & 0xFF,
+ (a >> 16) & 0xFF,
+ (a >> 8) & 0xFF,
+ (a ) & 0xFF,
+ p,
+ errno);
+
+ goto bailout;
+ }
+
+ buf_size = 1 << 16; // 64k
+ res = setsockopt(sock_fd_,
+ SOL_SOCKET, SO_RCVBUF,
+ &buf_size, sizeof(buf_size));
+ if (res < 0) {
+ ALOGW("Failed to increase socket buffer size to %d. (errno %d)",
+ buf_size, errno);
+ }
+
+ buf_size = 0;
+ opt_size = sizeof(buf_size);
+ res = getsockopt(sock_fd_,
+ SOL_SOCKET, SO_RCVBUF,
+ &buf_size, &opt_size);
+ if (res < 0) {
+ ALOGW("Failed to fetch socket buffer size. (errno %d)", errno);
+ } else {
+ ALOGI("RX socket buffer size is now %d bytes", buf_size);
+ }
+
+ if (listen_addr_.sin_addr.s_addr) {
+ // Join the multicast group and we should be good to go.
+ struct ip_mreq mreq;
+ mreq.imr_multiaddr = listen_addr_.sin_addr;
+ mreq.imr_interface.s_addr = htonl(INADDR_ANY);
+ res = setsockopt(sock_fd_,
+ IPPROTO_IP,
+ IP_ADD_MEMBERSHIP,
+ &mreq, sizeof(mreq));
+ if (res < 0) {
+ ALOGE("Failed to join multicast group. (errno %d)", errno);
+ goto bailout;
+ }
+ multicast_joined_ = true;
+ }
+
+ return true;
+
+bailout:
+ cleanupSocket();
+ return false;
+}
+
+bool AAH_RXPlayer::threadLoop() {
+ struct pollfd poll_fds[2];
+ bool process_more_right_now = false;
+
+ if (!setupSocket()) {
+ sendEvent(MEDIA_ERROR);
+ goto bailout;
+ }
+
+ while (!thread_wrapper_->exitPending()) {
+ // Step 1: Wait until there is something to do.
+ int gap_timeout = computeNextGapRetransmitTimeout();
+ int ring_timeout = ring_buffer_.computeInactivityTimeout();
+ int timeout = -1;
+
+ if (!ring_timeout) {
+ ALOGW("RTP inactivity timeout reached, resetting pipeline.");
+ resetPipeline();
+ timeout = gap_timeout;
+ } else {
+ if (gap_timeout < 0) {
+ timeout = ring_timeout;
+ } else if (ring_timeout < 0) {
+ timeout = gap_timeout;
+ } else {
+ timeout = (gap_timeout < ring_timeout) ? gap_timeout
+ : ring_timeout;
+ }
+ }
+
+ if ((0 != timeout) && (!process_more_right_now)) {
+ // Set up the events to wait on. Start with the wakeup pipe.
+ memset(&poll_fds, 0, sizeof(poll_fds));
+ poll_fds[0].fd = wakeup_work_thread_evt_.getWakeupHandle();
+ poll_fds[0].events = POLLIN;
+
+ // Add the RX socket.
+ poll_fds[1].fd = sock_fd_;
+ poll_fds[1].events = POLLIN;
+
+ // Wait for something interesing to happen.
+ int poll_res = poll(poll_fds, NELEM(poll_fds), timeout);
+ if (poll_res < 0) {
+ ALOGE("Fatal error (%d,%d) while waiting on events",
+ poll_res, errno);
+ sendEvent(MEDIA_ERROR);
+ goto bailout;
+ }
+ }
+
+ if (thread_wrapper_->exitPending()) {
+ break;
+ }
+
+ wakeup_work_thread_evt_.clearPendingEvents();
+ process_more_right_now = false;
+
+ // Step 2: Do we have data waiting in the socket? If so, drain the
+ // socket moving valid RTP information into the ring buffer to be
+ // processed.
+ if (poll_fds[1].revents) {
+ struct sockaddr_in from;
+ socklen_t from_len;
+
+ ssize_t res = 0;
+ while (!thread_wrapper_->exitPending()) {
+ // Check the size of any pending packet.
+ res = recv(sock_fd_, NULL, 0, MSG_PEEK | MSG_TRUNC);
+
+ // Error?
+ if (res < 0) {
+ // If the error is anything other than would block,
+ // something has gone very wrong.
+ if ((errno != EAGAIN) && (errno != EWOULDBLOCK)) {
+ ALOGE("Fatal socket error during recvfrom (%d, %d)",
+ (int)res, errno);
+ goto bailout;
+ }
+
+ // Socket is out of data, just break out of processing and
+ // wait for more.
+ break;
+ }
+
+ // Allocate a payload.
+ PacketBuffer* pb = PacketBuffer::allocate(res);
+ if (NULL == pb) {
+ ALOGE("Fatal error, failed to allocate packet buffer of"
+ " length %u", static_cast<uint32_t>(res));
+ goto bailout;
+ }
+
+ // Fetch the data.
+ from_len = sizeof(from);
+ res = recvfrom(sock_fd_, pb->data_, pb->length_, 0,
+ reinterpret_cast<struct sockaddr*>(&from),
+ &from_len);
+ if (res != pb->length_) {
+ ALOGE("Fatal error, fetched packet length (%d) does not"
+ " match peeked packet length (%u). This should never"
+ " happen. (errno = %d)",
+ static_cast<int>(res),
+ static_cast<uint32_t>(pb->length_),
+ errno);
+ }
+
+ bool drop_packet = false;
+ if (transmitter_known_) {
+ if (from.sin_addr.s_addr !=
+ transmitter_addr_.sin_addr.s_addr) {
+ uint32_t a = ntohl(from.sin_addr.s_addr);
+ uint16_t p = ntohs(from.sin_port);
+ ALOGV("Dropping packet from unknown transmitter"
+ " %u.%u.%u.%u:%hu",
+ ((a >> 24) & 0xFF),
+ ((a >> 16) & 0xFF),
+ ((a >> 8) & 0xFF),
+ ( a & 0xFF),
+ p);
+
+ drop_packet = true;
+ } else {
+ transmitter_addr_.sin_port = from.sin_port;
+ }
+ } else {
+ memcpy(&transmitter_addr_, &from, sizeof(from));
+ transmitter_known_ = true;
+ }
+
+ if (!drop_packet) {
+ bool serious_error = !processRX(pb);
+
+ if (serious_error) {
+ // Something went "seriously wrong". Currently, the
+ // only trigger for this should be a ring buffer
+ // overflow. The current failsafe behavior for when
+ // something goes seriously wrong is to just reset the
+ // pipeline. The system should behave as if this
+ // AAH_RXPlayer was just set up for the first time.
+ ALOGE("Something just went seriously wrong with the"
+ " pipeline. Resetting.");
+ resetPipeline();
+ }
+ } else {
+ PacketBuffer::destroy(pb);
+ }
+ }
+ }
+
+ // Step 3: Process any data we mave have accumulated in the ring buffer
+ // so far.
+ if (!thread_wrapper_->exitPending()) {
+ processRingBuffer();
+ }
+
+ // Step 4: At this point in time, the ring buffer should either be
+ // empty, or stalled in front of a gap caused by some dropped packets.
+ // Check on the current gap situation and deal with it in an appropriate
+ // fashion. If processGaps returns true, it means that it has given up
+ // on a gap and that we should try to process some more data
+ // immediately.
+ if (!thread_wrapper_->exitPending()) {
+ process_more_right_now = processGaps();
+ }
+
+ // Step 5: Check for fatal errors. If any of our substreams has
+ // encountered a fatal, unrecoverable, error, then propagate the error
+ // up to user level and shut down.
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ status_t status;
+ CHECK(substreams_.valueAt(i) != NULL);
+
+ status = substreams_.valueAt(i)->getStatus();
+ if (OK != status) {
+ ALOGE("Substream index %d has encountered an unrecoverable"
+ " error (%d). Signalling application level and shutting"
+ " down.", i, status);
+ sendEvent(MEDIA_ERROR);
+ goto bailout;
+ }
+ }
+ }
+
+bailout:
+ cleanupSocket();
+ return false;
+}
+
+bool AAH_RXPlayer::processRX(PacketBuffer* pb) {
+ CHECK(NULL != pb);
+
+ uint8_t* data = pb->data_;
+ ssize_t amt = pb->length_;
+ uint32_t nak_magic;
+ uint16_t seq_no;
+ uint32_t epoch;
+
+ // Every packet either starts with an RTP header which is at least 12 bytes
+ // long or is a retry NAK which is 14 bytes long. If there are fewer than
+ // 12 bytes here, this cannot be a proper RTP packet.
+ if (amt < 12) {
+ ALOGV("Dropping packet, too short to contain RTP header (%u bytes)",
+ static_cast<uint32_t>(amt));
+ goto drop_packet;
+ }
+
+ // Check to see if this is the special case of a NAK packet.
+ nak_magic = ntohl(*(reinterpret_cast<uint32_t*>(data)));
+ if (nak_magic == kRetransNAKMagic) {
+ // Looks like a NAK packet; make sure its long enough.
+
+ if (amt < static_cast<ssize_t>(sizeof(RetransRequest))) {
+ ALOGV("Dropping packet, too short to contain NAK payload (%u bytes)",
+ static_cast<uint32_t>(amt));
+ goto drop_packet;
+ }
+
+ SeqNoGap gap;
+ RetransRequest* rtr = reinterpret_cast<RetransRequest*>(data);
+ gap.start_seq_ = ntohs(rtr->start_seq_);
+ gap.end_seq_ = ntohs(rtr->end_seq_);
+
+ ALOGV("Process NAK for gap at [%hu, %hu]", gap.start_seq_, gap.end_seq_);
+ ring_buffer_.processNAK(&gap);
+
+ return true;
+ }
+
+ // According to the TRTP spec, version should be 2, padding should be 0,
+ // extension should be 0 and CSRCCnt should be 0. If any of these tests
+ // fail, we chuck the packet.
+ if (data[0] != 0x80) {
+ ALOGV("Dropping packet, bad V/P/X/CSRCCnt field (0x%02x)",
+ data[0]);
+ goto drop_packet;
+ }
+
+ // Check the payload type. For TRTP, it should always be 100.
+ if ((data[1] & 0x7F) != 100) {
+ ALOGV("Dropping packet, bad payload type. (%u)",
+ data[1] & 0x7F);
+ goto drop_packet;
+ }
+
+ // Check whether the transmitter has begun a new epoch.
+ epoch = (U32_AT(data + 8) >> 10) & 0x3FFFFF;
+ if (current_epoch_known_) {
+ if (epoch != current_epoch_) {
+ ALOGV("%s: new epoch %u", __PRETTY_FUNCTION__, epoch);
+ current_epoch_ = epoch;
+ resetPipeline();
+ }
+ } else {
+ current_epoch_ = epoch;
+ current_epoch_known_ = true;
+ }
+
+ // Extract the sequence number and hand the packet off to the ring buffer
+ // for dropped packet detection and later processing.
+ seq_no = U16_AT(data + 2);
+ return ring_buffer_.pushBuffer(pb, seq_no);
+
+drop_packet:
+ PacketBuffer::destroy(pb);
+ return true;
+}
+
+void AAH_RXPlayer::processRingBuffer() {
+ PacketBuffer* pb;
+ bool is_discon;
+ sp<Substream> substream;
+ LinearTransform trans;
+ bool foundTrans = false;
+
+ while (NULL != (pb = ring_buffer_.fetchBuffer(&is_discon))) {
+ if (is_discon) {
+ // Abort all partially assembled payloads.
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ CHECK(substreams_.valueAt(i) != NULL);
+ substreams_.valueAt(i)->cleanupBufferInProgress();
+ }
+ }
+
+ uint8_t* data = pb->data_;
+ ssize_t amt = pb->length_;
+
+ // Should not have any non-RTP packets in the ring buffer. RTP packets
+ // must be at least 12 bytes long.
+ CHECK(amt >= 12);
+
+ // Extract the marker bit and the SSRC field.
+ bool marker = (data[1] & 0x80) != 0;
+ uint32_t ssrc = U32_AT(data + 8);
+
+ // Is this the start of a new TRTP payload? If so, the marker bit
+ // should be set and there are some things we should be checking for.
+ if (marker) {
+ // TRTP headers need to have at least a byte for version, a byte for
+ // payload type and flags, and 4 bytes for length.
+ if (amt < 18) {
+ ALOGV("Dropping packet, too short to contain TRTP header"
+ " (%u bytes)", static_cast<uint32_t>(amt));
+ goto process_next_packet;
+ }
+
+ // Check the TRTP version and extract the payload type/flags.
+ uint8_t trtp_version = data[12];
+ uint8_t payload_type = (data[13] >> 4) & 0xF;
+ uint8_t trtp_flags = data[13] & 0xF;
+
+ if (1 != trtp_version) {
+ ALOGV("Dropping packet, bad trtp version %hhu", trtp_version);
+ goto process_next_packet;
+ }
+
+ // Is there a timestamp transformation present on this packet? If
+ // so, extract it and pass it to the appropriate substreams.
+ if (trtp_flags & 0x02) {
+ ssize_t offset = 18 + ((trtp_flags & 0x01) ? 4 : 0);
+ if (amt < (offset + 24)) {
+ ALOGV("Dropping packet, too short to contain TRTP Timestamp"
+ " Transformation (%u bytes)",
+ static_cast<uint32_t>(amt));
+ goto process_next_packet;
+ }
+
+ trans.a_zero = fetchInt64(data + offset);
+ trans.b_zero = fetchInt64(data + offset + 16);
+ trans.a_to_b_numer = static_cast<int32_t>(
+ fetchInt32 (data + offset + 8));
+ trans.a_to_b_denom = U32_AT(data + offset + 12);
+ foundTrans = true;
+
+ uint32_t program_id = (ssrc >> 5) & 0x1F;
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ sp<Substream> iter = substreams_.valueAt(i);
+ CHECK(iter != NULL);
+
+ if (iter->getProgramID() == program_id) {
+ iter->processTSTransform(trans);
+ }
+ }
+ }
+
+ // Is this a command packet? If so, its not necessarily associate
+ // with one particular substream. Just give it to the command
+ // packet handler and then move on.
+ if (4 == payload_type) {
+ processCommandPacket(pb);
+ goto process_next_packet;
+ }
+ }
+
+ // If we got to here, then we are a normal packet. Find (or allocate)
+ // the substream we belong to and send the packet off to be processed.
+ substream = substreams_.valueFor(ssrc);
+ if (substream == NULL) {
+ substream = new Substream(ssrc, omx_);
+ if (substream == NULL) {
+ ALOGE("Failed to allocate substream for SSRC 0x%08x", ssrc);
+ goto process_next_packet;
+ }
+ substreams_.add(ssrc, substream);
+
+ if (foundTrans) {
+ substream->processTSTransform(trans);
+ }
+ }
+
+ CHECK(substream != NULL);
+
+ if (marker) {
+ // Start of a new TRTP payload for this substream. Extract the
+ // lower 32 bits of the timestamp and hand the buffer to the
+ // substream for processing.
+ uint32_t ts_lower = U32_AT(data + 4);
+ substream->processPayloadStart(data + 12, amt - 12, ts_lower);
+ } else {
+ // Continuation of an existing TRTP payload. Just hand it off to
+ // the substream for processing.
+ substream->processPayloadCont(data + 12, amt - 12);
+ }
+
+process_next_packet:
+ PacketBuffer::destroy(pb);
+ } // end of main processing while loop.
+}
+
+void AAH_RXPlayer::processCommandPacket(PacketBuffer* pb) {
+ CHECK(NULL != pb);
+
+ uint8_t* data = pb->data_;
+ ssize_t amt = pb->length_;
+
+ // verify that this packet meets the minimum length of a command packet
+ if (amt < 20) {
+ return;
+ }
+
+ uint8_t trtp_version = data[12];
+ uint8_t trtp_flags = data[13] & 0xF;
+
+ if (1 != trtp_version) {
+ ALOGV("Dropping packet, bad trtp version %hhu", trtp_version);
+ return;
+ }
+
+ // calculate the start of the command payload
+ ssize_t offset = 18;
+ if (trtp_flags & 0x01) {
+ // timestamp is present (4 bytes)
+ offset += 4;
+ }
+ if (trtp_flags & 0x02) {
+ // transform is present (24 bytes)
+ offset += 24;
+ }
+
+ // the packet must contain 2 bytes of command payload beyond the TRTP header
+ if (amt < offset + 2) {
+ return;
+ }
+
+ uint16_t command_id = U16_AT(data + offset);
+
+ switch (command_id) {
+ case TRTPControlPacket::kCommandNop:
+ break;
+
+ case TRTPControlPacket::kCommandEOS:
+ case TRTPControlPacket::kCommandFlush: {
+ uint16_t program_id = (U32_AT(data + 8) >> 5) & 0x1F;
+ ALOGI("*** %s flushing program_id=%d",
+ __PRETTY_FUNCTION__, program_id);
+
+ Vector<uint32_t> substreams_to_remove;
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ sp<Substream> iter = substreams_.valueAt(i);
+ if (iter->getProgramID() == program_id) {
+ iter->shutdown();
+ substreams_to_remove.add(iter->getSSRC());
+ }
+ }
+
+ for (size_t i = 0; i < substreams_to_remove.size(); ++i) {
+ substreams_.removeItem(substreams_to_remove[i]);
+ }
+ } break;
+ }
+}
+
+bool AAH_RXPlayer::processGaps() {
+ // Deal with the current gap situation. Specifically...
+ //
+ // 1) If a new gap has shown up, send a retransmit request to the
+ // transmitter.
+ // 2) If a gap we were working on has had a packet in the middle or at
+ // the end filled in, send another retransmit request for the begining
+ // portion of the gap. TRTP was designed for LANs where packet
+ // re-ordering is very unlikely; so see the middle or end of a gap
+ // filled in before the begining is an almost certain indication that
+ // a retransmission packet was also dropped.
+ // 3) If we have been working on a gap for a while and it still has not
+ // been filled in, send another retransmit request.
+ // 4) If the are no more gaps in the ring, clear the current_gap_status_
+ // flag to indicate that all is well again.
+
+ // Start by fetching the active gap status.
+ SeqNoGap gap;
+ bool send_retransmit_request = false;
+ bool ret_val = false;
+ GapStatus gap_status;
+ if (kGS_NoGap != (gap_status = ring_buffer_.fetchCurrentGap(&gap))) {
+ // Note: checking for a change in the end sequence number should cover
+ // moving on to an entirely new gap for case #1 as well as resending the
+ // begining of a gap range for case #2.
+ send_retransmit_request = (kGS_NoGap == current_gap_status_) ||
+ (current_gap_.end_seq_ != gap.end_seq_);
+
+ // If this is the same gap we have been working on, and it has timed
+ // out, then check to see if our substreams are about to underflow. If
+ // so, instead of sending another retransmit request, just give up on
+ // this gap and move on.
+ if (!send_retransmit_request &&
+ (kGS_NoGap != current_gap_status_) &&
+ (0 == computeNextGapRetransmitTimeout())) {
+
+ // If out current gap is the fast-start gap, don't bother to skip it
+ // because substreams look like the are about to underflow.
+ if ((kGS_FastStartGap != gap_status) ||
+ (current_gap_.end_seq_ != gap.end_seq_)) {
+ for (size_t i = 0; i < substreams_.size(); ++i) {
+ if (substreams_.valueAt(i)->isAboutToUnderflow()) {
+ ALOGV("About to underflow, giving up on gap [%hu, %hu]",
+ gap.start_seq_, gap.end_seq_);
+ ring_buffer_.processNAK();
+ current_gap_status_ = kGS_NoGap;
+ return true;
+ }
+ }
+ }
+
+ // Looks like no one is about to underflow. Just go ahead and send
+ // the request.
+ send_retransmit_request = true;
+ }
+ } else {
+ current_gap_status_ = kGS_NoGap;
+ }
+
+ if (send_retransmit_request) {
+ // If we have been working on a fast start, and it is still not filled
+ // in, even after the extended retransmit time out, give up and skip it.
+ // The system should fall back into its normal slow-start behavior.
+ if ((kGS_FastStartGap == current_gap_status_) &&
+ (current_gap_.end_seq_ == gap.end_seq_)) {
+ ALOGV("Fast start is taking forever; giving up.");
+ ring_buffer_.processNAK();
+ current_gap_status_ = kGS_NoGap;
+ return true;
+ }
+
+ // Send the request.
+ RetransRequest req;
+ uint32_t magic = (kGS_FastStartGap == gap_status)
+ ? kFastStartRequestMagic
+ : kRetransRequestMagic;
+ req.magic_ = htonl(magic);
+ req.mcast_ip_ = listen_addr_.sin_addr.s_addr;
+ req.mcast_port_ = listen_addr_.sin_port;
+ req.start_seq_ = htons(gap.start_seq_);
+ req.end_seq_ = htons(gap.end_seq_);
+
+ {
+ uint32_t a = ntohl(transmitter_addr_.sin_addr.s_addr);
+ uint16_t p = ntohs(transmitter_addr_.sin_port);
+ ALOGV("Sending to transmitter %u.%u.%u.%u:%hu",
+ ((a >> 24) & 0xFF),
+ ((a >> 16) & 0xFF),
+ ((a >> 8) & 0xFF),
+ ( a & 0xFF),
+ p);
+ }
+
+ int res = sendto(sock_fd_, &req, sizeof(req), 0,
+ reinterpret_cast<struct sockaddr*>(&transmitter_addr_),
+ sizeof(transmitter_addr_));
+ if (res < 0) {
+ ALOGE("Error when sending retransmit request (%d)", errno);
+ } else {
+ ALOGV("%s request for range [%hu, %hu] sent",
+ (kGS_FastStartGap == gap_status) ? "Fast Start" : "Retransmit",
+ gap.start_seq_, gap.end_seq_);
+ }
+
+ // Update the current gap info.
+ current_gap_ = gap;
+ current_gap_status_ = gap_status;
+ next_retrans_req_time_ = monotonicUSecNow() +
+ ((kGS_FastStartGap == current_gap_status_)
+ ? kFastStartTimeoutUSec
+ : kGapRerequestTimeoutUSec);
+ }
+
+ return false;
+}
+
+// Compute when its time to send the next gap retransmission in milliseconds.
+// Returns < 0 for an infinite timeout (no gap) and 0 if its time to retransmit
+// right now.
+int AAH_RXPlayer::computeNextGapRetransmitTimeout() {
+ if (kGS_NoGap == current_gap_status_) {
+ return -1;
+ }
+
+ int64_t timeout_delta = next_retrans_req_time_ - monotonicUSecNow();
+
+ timeout_delta /= 1000;
+ if (timeout_delta <= 0) {
+ return 0;
+ }
+
+ return static_cast<uint32_t>(timeout_delta);
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_rx_player_ring_buffer.cpp b/media/libaah_rtp/aah_rx_player_ring_buffer.cpp
new file mode 100644
index 0000000..0d8b31f
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_ring_buffer.cpp
@@ -0,0 +1,366 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+#include <utils/Log.h>
+
+#include "aah_rx_player.h"
+
+namespace android {
+
+AAH_RXPlayer::RXRingBuffer::RXRingBuffer(uint32_t capacity) {
+ capacity_ = capacity;
+ rd_ = wr_ = 0;
+ ring_ = new PacketBuffer*[capacity];
+ memset(ring_, 0, sizeof(PacketBuffer*) * capacity);
+ reset();
+}
+
+AAH_RXPlayer::RXRingBuffer::~RXRingBuffer() {
+ reset();
+ delete[] ring_;
+}
+
+void AAH_RXPlayer::RXRingBuffer::reset() {
+ AutoMutex lock(&lock_);
+
+ if (NULL != ring_) {
+ while (rd_ != wr_) {
+ CHECK(rd_ < capacity_);
+ if (NULL != ring_[rd_]) {
+ PacketBuffer::destroy(ring_[rd_]);
+ ring_[rd_] = NULL;
+ }
+ rd_ = (rd_ + 1) % capacity_;
+ }
+ }
+
+ rd_ = wr_ = 0;
+ rd_seq_known_ = false;
+ waiting_for_fast_start_ = true;
+ fetched_first_packet_ = false;
+ rtp_activity_timeout_valid_ = false;
+}
+
+bool AAH_RXPlayer::RXRingBuffer::pushBuffer(PacketBuffer* buf,
+ uint16_t seq) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+ CHECK(NULL != buf);
+
+ rtp_activity_timeout_valid_ = true;
+ rtp_activity_timeout_ = monotonicUSecNow() + kRTPActivityTimeoutUSec;
+
+ // If the ring buffer is totally reset (we have never received a single
+ // payload) then we don't know the rd sequence number and this should be
+ // simple. We just store the payload, advance the wr pointer and record the
+ // initial sequence number.
+ if (!rd_seq_known_) {
+ CHECK(rd_ == wr_);
+ CHECK(NULL == ring_[wr_]);
+ CHECK(wr_ < capacity_);
+
+ ring_[wr_] = buf;
+ wr_ = (wr_ + 1) % capacity_;
+ rd_seq_ = seq;
+ rd_seq_known_ = true;
+ return true;
+ }
+
+ // Compute the seqence number of this payload and of the write pointer,
+ // normalized around the read pointer. IOW - transform the payload seq no
+ // and the wr pointer seq no into a space where the rd pointer seq no is
+ // zero. This will define 4 cases we can consider...
+ //
+ // 1) norm_seq == norm_wr_seq
+ // This payload is contiguous with the last. All is good.
+ //
+ // 2) ((norm_seq < norm_wr_seq) && (norm_seq >= norm_rd_seq)
+ // aka ((norm_seq < norm_wr_seq) && (norm_seq >= 0)
+ // This payload is in the past, in the unprocessed region of the ring
+ // buffer. It is probably a retransmit intended to fill in a dropped
+ // payload; it may be a duplicate.
+ //
+ // 3) ((norm_seq - norm_wr_seq) & 0x8000) != 0
+ // This payload is in the past compared to the write pointer (or so very
+ // far in the future that it has wrapped the seq no space), but not in
+ // the unprocessed region of the ring buffer. This could be a duplicate
+ // retransmit; we just drop these payloads unless we are waiting for our
+ // first fast start packet. If we are waiting for fast start, than this
+ // packet is probably the first packet of the fast start retransmission.
+ // If it will fit in the buffer, back up the read pointer to its position
+ // and clear the fast start flag, otherwise just drop it.
+ //
+ // 4) ((norm_seq - norm_wr_seq) & 0x8000) == 0
+ // This payload which is ahead of the next write pointer. This indicates
+ // that we have missed some payloads and need to request a retransmit.
+ // If norm_seq >= (capacity - 1), then the gap is so large that it would
+ // overflow the ring buffer and we should probably start to panic.
+
+ uint16_t norm_wr_seq = ((wr_ + capacity_ - rd_) % capacity_);
+ uint16_t norm_seq = seq - rd_seq_;
+
+ // Check for overflow first.
+ if ((!(norm_seq & 0x8000)) && (norm_seq >= (capacity_ - 1))) {
+ ALOGW("Ring buffer overflow; cap = %u, [rd, wr] = [%hu, %hu], seq = %hu",
+ capacity_, rd_seq_, norm_wr_seq + rd_seq_, seq);
+ PacketBuffer::destroy(buf);
+ return false;
+ }
+
+ // Check for case #1
+ if (norm_seq == norm_wr_seq) {
+ CHECK(wr_ < capacity_);
+ CHECK(NULL == ring_[wr_]);
+
+ ring_[wr_] = buf;
+ wr_ = (wr_ + 1) % capacity_;
+
+ CHECK(wr_ != rd_);
+ return true;
+ }
+
+ // Check case #2
+ uint32_t ring_pos = (rd_ + norm_seq) % capacity_;
+ if ((norm_seq < norm_wr_seq) && (!(norm_seq & 0x8000))) {
+ // Do we already have a payload for this slot? If so, then this looks
+ // like a duplicate retransmit. Just ignore it.
+ if (NULL != ring_[ring_pos]) {
+ ALOGD("RXed duplicate retransmit, seq = %hu", seq);
+ PacketBuffer::destroy(buf);
+ } else {
+ // Looks like we were missing this payload. Go ahead and store it.
+ ring_[ring_pos] = buf;
+ }
+
+ return true;
+ }
+
+ // Check case #3
+ if ((norm_seq - norm_wr_seq) & 0x8000) {
+ if (!waiting_for_fast_start_) {
+ ALOGD("RXed duplicate retransmit from before rd pointer, seq = %hu",
+ seq);
+ PacketBuffer::destroy(buf);
+ } else {
+ // Looks like a fast start fill-in. Go ahead and store it, assuming
+ // that we can fit it in the buffer.
+ uint32_t implied_ring_size = static_cast<uint32_t>(norm_wr_seq)
+ + (rd_seq_ - seq);
+
+ if (implied_ring_size >= (capacity_ - 1)) {
+ ALOGD("RXed what looks like a fast start packet (seq = %hu),"
+ " but packet is too far in the past to fit into the ring"
+ " buffer. Dropping.", seq);
+ PacketBuffer::destroy(buf);
+ } else {
+ ring_pos = (rd_ + capacity_ + seq - rd_seq_) % capacity_;
+ rd_seq_ = seq;
+ rd_ = ring_pos;
+ waiting_for_fast_start_ = false;
+
+ CHECK(ring_pos < capacity_);
+ CHECK(NULL == ring_[ring_pos]);
+ ring_[ring_pos] = buf;
+ }
+
+ }
+ return true;
+ }
+
+ // Must be in case #4 with no overflow. This packet fits in the current
+ // ring buffer, but is discontiuguous. Advance the write pointer leaving a
+ // gap behind.
+ uint32_t gap_len = (ring_pos + capacity_ - wr_) % capacity_;
+ ALOGD("Drop detected; %u packets, seq_range [%hu, %hu]",
+ gap_len,
+ rd_seq_ + norm_wr_seq,
+ rd_seq_ + norm_wr_seq + gap_len - 1);
+
+ CHECK(NULL == ring_[ring_pos]);
+ ring_[ring_pos] = buf;
+ wr_ = (ring_pos + 1) % capacity_;
+ CHECK(wr_ != rd_);
+
+ return true;
+}
+
+AAH_RXPlayer::PacketBuffer*
+AAH_RXPlayer::RXRingBuffer::fetchBuffer(bool* is_discon) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+ CHECK(NULL != is_discon);
+
+ // If the read seqence number is not known, then this ring buffer has not
+ // received a packet since being reset and there cannot be any packets to
+ // return. If we are still waiting for the first fast start packet to show
+ // up, we don't want to let any buffer be consumed yet because we expect to
+ // see a packet before the initial read sequence number show up shortly.
+ if (!rd_seq_known_ || waiting_for_fast_start_) {
+ *is_discon = false;
+ return NULL;
+ }
+
+ PacketBuffer* ret = NULL;
+ *is_discon = !fetched_first_packet_;
+
+ while ((rd_ != wr_) && (NULL == ret)) {
+ CHECK(rd_ < capacity_);
+
+ // If we hit a gap, stall and do not advance the read pointer. Let the
+ // higher level code deal with requesting retries and/or deciding to
+ // skip the current gap.
+ ret = ring_[rd_];
+ if (NULL == ret) {
+ break;
+ }
+
+ ring_[rd_] = NULL;
+ rd_ = (rd_ + 1) % capacity_;
+ ++rd_seq_;
+ }
+
+ if (NULL != ret) {
+ fetched_first_packet_ = true;
+ }
+
+ return ret;
+}
+
+AAH_RXPlayer::GapStatus
+AAH_RXPlayer::RXRingBuffer::fetchCurrentGap(SeqNoGap* gap) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+ CHECK(NULL != gap);
+
+ // If the read seqence number is not known, then this ring buffer has not
+ // received a packet since being reset and there cannot be any gaps.
+ if (!rd_seq_known_) {
+ return kGS_NoGap;
+ }
+
+ // If we are waiting for fast start, then the current gap is a fast start
+ // gap and it includes all packets before the read sequence number.
+ if (waiting_for_fast_start_) {
+ gap->start_seq_ =
+ gap->end_seq_ = rd_seq_ - 1;
+ return kGS_FastStartGap;
+ }
+
+ // If rd == wr, then the buffer is empty and there cannot be any gaps.
+ if (rd_ == wr_) {
+ return kGS_NoGap;
+ }
+
+ // If rd_ is currently pointing at an unprocessed packet, then there is no
+ // current gap.
+ CHECK(rd_ < capacity_);
+ if (NULL != ring_[rd_]) {
+ return kGS_NoGap;
+ }
+
+ // Looks like there must be a gap here. The start of the gap is the current
+ // rd sequence number, all we need to do now is determine its length in
+ // order to compute the end sequence number.
+ gap->start_seq_ = rd_seq_;
+ uint16_t end = rd_seq_;
+ uint32_t tmp = (rd_ + 1) % capacity_;
+ while ((tmp != wr_) && (NULL == ring_[tmp])) {
+ ++end;
+ tmp = (tmp + 1) % capacity_;
+ }
+ gap->end_seq_ = end;
+
+ return kGS_NormalGap;
+}
+
+void AAH_RXPlayer::RXRingBuffer::processNAK(const SeqNoGap* nak) {
+ AutoMutex lock(&lock_);
+ CHECK(NULL != ring_);
+
+ // If we were waiting for our first fast start fill-in packet, and we
+ // received a NAK, then apparantly we are not getting our fast start. Just
+ // clear the waiting flag and go back to normal behavior.
+ if (waiting_for_fast_start_) {
+ waiting_for_fast_start_ = false;
+ }
+
+ // If we have not received a packet since last reset, or there is no data in
+ // the ring, then there is nothing to skip.
+ if ((!rd_seq_known_) || (rd_ == wr_)) {
+ return;
+ }
+
+ // If rd_ is currently pointing at an unprocessed packet, then there is no
+ // gap to skip.
+ CHECK(rd_ < capacity_);
+ if (NULL != ring_[rd_]) {
+ return;
+ }
+
+ // Looks like there must be a gap here. Advance rd until we have passed
+ // over the portion of it indicated by nak (or all of the gap if nak is
+ // NULL). Then reset fetched_first_packet_ so that the next read will show
+ // up as being discontiguous.
+ uint16_t seq_after_gap = (NULL == nak) ? 0 : nak->end_seq_ + 1;
+ while ((rd_ != wr_) &&
+ (NULL == ring_[rd_]) &&
+ ((NULL == nak) || (seq_after_gap != rd_seq_))) {
+ rd_ = (rd_ + 1) % capacity_;
+ ++rd_seq_;
+ }
+ fetched_first_packet_ = false;
+}
+
+int AAH_RXPlayer::RXRingBuffer::computeInactivityTimeout() {
+ AutoMutex lock(&lock_);
+
+ if (!rtp_activity_timeout_valid_) {
+ return -1;
+ }
+
+ uint64_t now = monotonicUSecNow();
+ if (rtp_activity_timeout_ <= now) {
+ return 0;
+ }
+
+ return (rtp_activity_timeout_ - now) / 1000;
+}
+
+AAH_RXPlayer::PacketBuffer*
+AAH_RXPlayer::PacketBuffer::allocate(ssize_t length) {
+ if (length <= 0) {
+ return NULL;
+ }
+
+ uint32_t alloc_len = sizeof(PacketBuffer) + length;
+ PacketBuffer* ret = reinterpret_cast<PacketBuffer*>(
+ new uint8_t[alloc_len]);
+
+ if (NULL != ret) {
+ ret->length_ = length;
+ }
+
+ return ret;
+}
+
+void AAH_RXPlayer::PacketBuffer::destroy(PacketBuffer* pb) {
+ uint8_t* kill_me = reinterpret_cast<uint8_t*>(pb);
+ delete[] kill_me;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_rx_player_substream.cpp b/media/libaah_rtp/aah_rx_player_substream.cpp
new file mode 100644
index 0000000..1e4c784
--- /dev/null
+++ b/media/libaah_rtp/aah_rx_player_substream.cpp
@@ -0,0 +1,498 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+
+#include <include/avc_utils.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/OMXCodec.h>
+#include <media/stagefright/Utils.h>
+
+#include "aah_rx_player.h"
+
+namespace android {
+
+int64_t AAH_RXPlayer::Substream::kAboutToUnderflowThreshold =
+ 50ull * 1000;
+
+AAH_RXPlayer::Substream::Substream(uint32_t ssrc, OMXClient& omx) {
+ ssrc_ = ssrc;
+ substream_details_known_ = false;
+ buffer_in_progress_ = NULL;
+ status_ = OK;
+
+ decoder_ = new AAH_DecoderPump(omx);
+ if (decoder_ == NULL) {
+ ALOGE("%s failed to allocate decoder pump!", __PRETTY_FUNCTION__);
+ }
+ if (OK != decoder_->initCheck()) {
+ ALOGE("%s failed to initialize decoder pump!", __PRETTY_FUNCTION__);
+ }
+
+ // cleanupBufferInProgress will reset most of the internal state variables.
+ // Just need to make sure that buffer_in_progress_ is NULL before calling.
+ cleanupBufferInProgress();
+}
+
+
+void AAH_RXPlayer::Substream::shutdown() {
+ substream_meta_ = NULL;
+ status_ = OK;
+ cleanupBufferInProgress();
+ cleanupDecoder();
+}
+
+void AAH_RXPlayer::Substream::cleanupBufferInProgress() {
+ if (NULL != buffer_in_progress_) {
+ buffer_in_progress_->release();
+ buffer_in_progress_ = NULL;
+ }
+
+ expected_buffer_size_ = 0;
+ buffer_filled_ = 0;
+ waiting_for_rap_ = true;
+}
+
+void AAH_RXPlayer::Substream::cleanupDecoder() {
+ if (decoder_ != NULL) {
+ decoder_->shutdown();
+ }
+}
+
+bool AAH_RXPlayer::Substream::shouldAbort(const char* log_tag) {
+ // If we have already encountered a fatal error, do nothing. We are just
+ // waiting for our owner to shut us down now.
+ if (OK != status_) {
+ ALOGV("Skipping %s, substream has encountered fatal error (%d).",
+ log_tag, status_);
+ return true;
+ }
+
+ return false;
+}
+
+void AAH_RXPlayer::Substream::processPayloadStart(uint8_t* buf,
+ uint32_t amt,
+ int32_t ts_lower) {
+ uint32_t min_length = 6;
+
+ if (shouldAbort(__PRETTY_FUNCTION__)) {
+ return;
+ }
+
+ // Do we have a buffer in progress already? If so, abort the buffer. In
+ // theory, this should never happen. If there were a discontinutity in the
+ // stream, the discon in the seq_nos at the RTP level should have already
+ // triggered a cleanup of the buffer in progress. To see a problem at this
+ // level is an indication either of a bug in the transmitter, or some form
+ // of terrible corruption/tampering on the wire.
+ if (NULL != buffer_in_progress_) {
+ ALOGE("processPayloadStart is aborting payload already in progress.");
+ cleanupBufferInProgress();
+ }
+
+ // Parse enough of the header to know where we stand. Since this is a
+ // payload start, it should begin with a TRTP header which has to be at
+ // least 6 bytes long.
+ if (amt < min_length) {
+ ALOGV("Discarding payload too short to contain TRTP header (len = %u)",
+ amt);
+ return;
+ }
+
+ // Check the TRTP version number.
+ if (0x01 != buf[0]) {
+ ALOGV("Unexpected TRTP version (%u) in header. Expected %u.",
+ buf[0], 1);
+ return;
+ }
+
+ // Extract the substream type field and make sure its one we understand (and
+ // one that does not conflict with any previously received substream type.
+ uint8_t header_type = (buf[1] >> 4) & 0xF;
+ switch (header_type) {
+ case 0x01:
+ // Audio, yay! Just break. We understand audio payloads.
+ break;
+ case 0x02:
+ ALOGV("RXed packet with unhandled TRTP header type (Video).");
+ return;
+ case 0x03:
+ ALOGV("RXed packet with unhandled TRTP header type (Subpicture).");
+ return;
+ case 0x04:
+ ALOGV("RXed packet with unhandled TRTP header type (Control).");
+ return;
+ default:
+ ALOGV("RXed packet with unhandled TRTP header type (%u).",
+ header_type);
+ return;
+ }
+
+ if (substream_details_known_ && (header_type != substream_type_)) {
+ ALOGV("RXed TRTP Payload for SSRC=0x%08x where header type (%u) does not"
+ " match previously received header type (%u)",
+ ssrc_, header_type, substream_type_);
+ return;
+ }
+
+ // Check the flags to see if there is another 32 bits of timestamp present.
+ uint32_t trtp_header_len = 6;
+ bool ts_valid = buf[1] & 0x1;
+ if (ts_valid) {
+ min_length += 4;
+ trtp_header_len += 4;
+ if (amt < min_length) {
+ ALOGV("Discarding payload too short to contain TRTP timestamp"
+ " (len = %u)", amt);
+ return;
+ }
+ }
+
+ // Extract the TRTP length field and sanity check it.
+ uint32_t trtp_len;
+ trtp_len = (static_cast<uint32_t>(buf[2]) << 24) |
+ (static_cast<uint32_t>(buf[3]) << 16) |
+ (static_cast<uint32_t>(buf[4]) << 8) |
+ static_cast<uint32_t>(buf[5]);
+ if (trtp_len < min_length) {
+ ALOGV("TRTP length (%u) is too short to be valid. Must be at least %u"
+ " bytes.", trtp_len, min_length);
+ return;
+ }
+
+ // Extract the rest of the timestamp field if valid.
+ int64_t ts = 0;
+ uint32_t parse_offset = 6;
+ if (ts_valid) {
+ ts = (static_cast<int64_t>(buf[parse_offset ]) << 56) |
+ (static_cast<int64_t>(buf[parse_offset + 1]) << 48) |
+ (static_cast<int64_t>(buf[parse_offset + 2]) << 40) |
+ (static_cast<int64_t>(buf[parse_offset + 3]) << 32);
+ ts |= ts_lower;
+ parse_offset += 4;
+ }
+
+ // Check the flags to see if there is another 24 bytes of timestamp
+ // transformation present.
+ if (buf[1] & 0x2) {
+ min_length += 24;
+ parse_offset += 24;
+ trtp_header_len += 24;
+ if (amt < min_length) {
+ ALOGV("Discarding payload too short to contain TRTP timestamp"
+ " transformation (len = %u)", amt);
+ return;
+ }
+ }
+
+ // TODO : break the parsing into individual parsers for the different
+ // payload types (audio, video, etc).
+ //
+ // At this point in time, we know that this is audio. Go ahead and parse
+ // the basic header, check the codec type, and find the payload portion of
+ // the packet.
+ min_length += 3;
+ if (trtp_len < min_length) {
+ ALOGV("TRTP length (%u) is too short to be a valid audio payload. Must"
+ " be at least %u bytes.", trtp_len, min_length);
+ return;
+ }
+
+ if (amt < min_length) {
+ ALOGV("TRTP porttion of RTP payload (%u bytes) too small to contain"
+ " entire TRTP header. TRTP does not currently support fragmenting"
+ " TRTP headers across RTP payloads", amt);
+ return;
+ }
+
+ uint8_t codec_type = buf[parse_offset ];
+ uint8_t flags = buf[parse_offset + 1];
+ uint8_t volume = buf[parse_offset + 2];
+ parse_offset += 3;
+ trtp_header_len += 3;
+
+ if (!setupSubstreamType(header_type, codec_type)) {
+ return;
+ }
+
+ if (decoder_ != NULL) {
+ decoder_->setRenderVolume(volume);
+ }
+
+ // TODO : move all of the constant flag and offset definitions for TRTP up
+ // into some sort of common header file.
+ if (waiting_for_rap_ && !(flags & 0x08)) {
+ ALOGV("Dropping non-RAP TRTP Audio Payload while waiting for RAP.");
+ return;
+ }
+
+ if (flags & 0x10) {
+ ALOGV("Dropping TRTP Audio Payload with aux codec data present (only"
+ " handle MP3 right now, and it has no aux data)");
+ return;
+ }
+
+ // OK - everything left is just payload. Compute the payload size, start
+ // the buffer in progress and pack as much payload as we can into it. If
+ // the payload is finished once we are done, go ahead and send the payload
+ // to the decoder.
+ expected_buffer_size_ = trtp_len - trtp_header_len;
+ if (!expected_buffer_size_) {
+ ALOGV("Dropping TRTP Audio Payload with 0 Access Unit length");
+ return;
+ }
+
+ CHECK(amt >= trtp_header_len);
+ uint32_t todo = amt - trtp_header_len;
+ if (expected_buffer_size_ < todo) {
+ ALOGV("Extra data (%u > %u) present in initial TRTP Audio Payload;"
+ " dropping payload.", todo, expected_buffer_size_);
+ return;
+ }
+
+ buffer_filled_ = 0;
+ buffer_in_progress_ = new MediaBuffer(expected_buffer_size_);
+ if ((NULL == buffer_in_progress_) ||
+ (NULL == buffer_in_progress_->data())) {
+ ALOGV("Failed to allocate MediaBuffer of length %u",
+ expected_buffer_size_);
+ cleanupBufferInProgress();
+ return;
+ }
+
+ sp<MetaData> meta = buffer_in_progress_->meta_data();
+ if (meta == NULL) {
+ ALOGV("Missing metadata structure in allocated MediaBuffer; dropping"
+ " payload");
+ cleanupBufferInProgress();
+ return;
+ }
+
+ // TODO : set this based on the codec type indicated in the TRTP stream.
+ // Right now, we only support MP3, so the choice is obvious.
+ meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
+ if (ts_valid) {
+ meta->setInt64(kKeyTime, ts);
+ }
+
+ if (amt > 0) {
+ uint8_t* tgt =
+ reinterpret_cast<uint8_t*>(buffer_in_progress_->data());
+ memcpy(tgt + buffer_filled_, buf + trtp_header_len, todo);
+ buffer_filled_ += amt;
+ }
+
+ if (buffer_filled_ >= expected_buffer_size_) {
+ processCompletedBuffer();
+ }
+}
+
+void AAH_RXPlayer::Substream::processPayloadCont(uint8_t* buf,
+ uint32_t amt) {
+ if (shouldAbort(__PRETTY_FUNCTION__)) {
+ return;
+ }
+
+ if (NULL == buffer_in_progress_) {
+ ALOGV("TRTP Receiver skipping payload continuation; no buffer currently"
+ " in progress.");
+ return;
+ }
+
+ CHECK(buffer_filled_ < expected_buffer_size_);
+ uint32_t buffer_left = expected_buffer_size_ - buffer_filled_;
+ if (amt > buffer_left) {
+ ALOGV("Extra data (%u > %u) present in continued TRTP Audio Payload;"
+ " dropping payload.", amt, buffer_left);
+ cleanupBufferInProgress();
+ return;
+ }
+
+ if (amt > 0) {
+ uint8_t* tgt =
+ reinterpret_cast<uint8_t*>(buffer_in_progress_->data());
+ memcpy(tgt + buffer_filled_, buf, amt);
+ buffer_filled_ += amt;
+ }
+
+ if (buffer_filled_ >= expected_buffer_size_) {
+ processCompletedBuffer();
+ }
+}
+
+void AAH_RXPlayer::Substream::processCompletedBuffer() {
+ const uint8_t* buffer_data = NULL;
+ int sample_rate;
+ int channel_count;
+ size_t frame_size;
+ status_t res;
+
+ CHECK(NULL != buffer_in_progress_);
+
+ if (decoder_ == NULL) {
+ ALOGV("Dropping complete buffer, no decoder pump allocated");
+ goto bailout;
+ }
+
+ buffer_data = reinterpret_cast<const uint8_t*>(buffer_in_progress_->data());
+ if (buffer_in_progress_->size() < 4) {
+ ALOGV("MP3 payload too short to contain header, dropping payload.");
+ goto bailout;
+ }
+
+ // Extract the channel count and the sample rate from the MP3 header. The
+ // stagefright MP3 requires that these be delivered before decoing can
+ // begin.
+ if (!GetMPEGAudioFrameSize(U32_AT(buffer_data),
+ &frame_size,
+ &sample_rate,
+ &channel_count,
+ NULL,
+ NULL)) {
+ ALOGV("Failed to parse MP3 header in payload, droping payload.");
+ goto bailout;
+ }
+
+
+ // Make sure that our substream metadata is set up properly. If there has
+ // been a format change, be sure to reset the underlying decoder. In
+ // stagefright, it seems like the only way to do this is to destroy and
+ // recreate the decoder.
+ if (substream_meta_ == NULL) {
+ substream_meta_ = new MetaData();
+
+ if (substream_meta_ == NULL) {
+ ALOGE("Failed to allocate MetaData structure for substream");
+ goto bailout;
+ }
+
+ substream_meta_->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
+ substream_meta_->setInt32 (kKeyChannelCount, channel_count);
+ substream_meta_->setInt32 (kKeySampleRate, sample_rate);
+ } else {
+ int32_t prev_sample_rate;
+ int32_t prev_channel_count;
+ substream_meta_->findInt32(kKeySampleRate, &prev_sample_rate);
+ substream_meta_->findInt32(kKeyChannelCount, &prev_channel_count);
+
+ if ((prev_channel_count != channel_count) ||
+ (prev_sample_rate != sample_rate)) {
+ ALOGW("Format change detected, forcing decoder reset.");
+ cleanupDecoder();
+
+ substream_meta_->setInt32(kKeyChannelCount, channel_count);
+ substream_meta_->setInt32(kKeySampleRate, sample_rate);
+ }
+ }
+
+ // If our decoder has not be set up, do so now.
+ res = decoder_->init(substream_meta_);
+ if (OK != res) {
+ ALOGE("Failed to init decoder (res = %d)", res);
+ cleanupDecoder();
+ substream_meta_ = NULL;
+ goto bailout;
+ }
+
+ // Queue the payload for decode.
+ res = decoder_->queueForDecode(buffer_in_progress_);
+
+ if (res != OK) {
+ ALOGD("Failed to queue payload for decode, resetting decoder pump!"
+ " (res = %d)", res);
+ status_ = res;
+ cleanupDecoder();
+ cleanupBufferInProgress();
+ }
+
+ // NULL out buffer_in_progress before calling the cleanup helper.
+ //
+ // MediaBuffers use something of a hybrid ref-counting pattern which prevent
+ // the AAH_DecoderPump's input queue from adding their own reference to the
+ // MediaBuffer. MediaBuffers start life with a reference count of 0, as
+ // well as an observer which starts as NULL. Before being given an
+ // observer, the ref count cannot be allowed to become non-zero as it will
+ // cause calls to release() to assert. Basically, before a MediaBuffer has
+ // an observer, they behave like non-ref counted obects where release()
+ // serves the roll of delete. After a MediaBuffer has an observer, they
+ // become more like ref counted objects where add ref and release can be
+ // used, and when the ref count hits zero, the MediaBuffer is handed off to
+ // the observer.
+ //
+ // Given all of this, when we give the buffer to the decoder pump to wait in
+ // the to-be-processed queue, the decoder cannot add a ref to the buffer as
+ // it would in a traditional ref counting system. Instead it needs to
+ // "steal" the non-existent ref. In the case of queue failure, we need to
+ // make certain to release this non-existent reference so that the buffer is
+ // cleaned up during the cleanupBufferInProgress helper. In the case of a
+ // successful queue operation, we need to make certain that the
+ // cleanupBufferInProgress helper does not release the buffer since it needs
+ // to remain alive in the queue. We acomplish this by NULLing out the
+ // buffer pointer before calling the cleanup helper.
+ buffer_in_progress_ = NULL;
+
+bailout:
+ cleanupBufferInProgress();
+}
+
+
+void AAH_RXPlayer::Substream::processTSTransform(const LinearTransform& trans) {
+ if (decoder_ != NULL) {
+ decoder_->setRenderTSTransform(trans);
+ }
+}
+
+bool AAH_RXPlayer::Substream::isAboutToUnderflow() {
+ if (decoder_ == NULL) {
+ return false;
+ }
+
+ return decoder_->isAboutToUnderflow(kAboutToUnderflowThreshold);
+}
+
+bool AAH_RXPlayer::Substream::setupSubstreamType(uint8_t substream_type,
+ uint8_t codec_type) {
+ // Sanity check the codec type. Right now we only support MP3. Also check
+ // for conflicts with previously delivered codec types.
+ if (substream_details_known_ && (codec_type != codec_type_)) {
+ ALOGV("RXed TRTP Payload for SSRC=0x%08x where codec type (%u) does not"
+ " match previously received codec type (%u)",
+ ssrc_, codec_type, codec_type_);
+ return false;
+ }
+
+ if (codec_type != 0x03) {
+ ALOGV("RXed TRTP Audio Payload for SSRC=0x%08x with unsupported codec"
+ " type (%u)", ssrc_, codec_type);
+ return false;
+ }
+
+ if (!substream_details_known_) {
+ substream_type_ = substream_type;
+ codec_type_ = codec_type;
+ substream_details_known_ = true;
+ }
+
+ return true;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_packet.cpp b/media/libaah_rtp/aah_tx_packet.cpp
new file mode 100644
index 0000000..3f6e0e9
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_packet.cpp
@@ -0,0 +1,331 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+#include <utils/Log.h>
+
+#include <arpa/inet.h>
+#include <string.h>
+
+#include <media/stagefright/foundation/ADebug.h>
+
+#include "aah_tx_packet.h"
+
+namespace android {
+
+const int TRTPPacket::kRTPHeaderLen;
+const uint32_t TRTPPacket::kTRTPEpochMask;
+
+TRTPPacket::~TRTPPacket() {
+ delete mPacket;
+}
+
+/*** TRTP packet properties ***/
+
+void TRTPPacket::setSeqNumber(uint16_t val) {
+ mSeqNumber = val;
+
+ if (mIsPacked) {
+ const int kTRTPSeqNumberOffset = 2;
+ uint16_t* buf = reinterpret_cast<uint16_t*>(
+ mPacket + kTRTPSeqNumberOffset);
+ *buf = htons(mSeqNumber);
+ }
+}
+
+uint16_t TRTPPacket::getSeqNumber() const {
+ return mSeqNumber;
+}
+
+void TRTPPacket::setPTS(int64_t val) {
+ CHECK(!mIsPacked);
+ mPTS = val;
+ mPTSValid = true;
+}
+
+int64_t TRTPPacket::getPTS() const {
+ return mPTS;
+}
+
+void TRTPPacket::setEpoch(uint32_t val) {
+ mEpoch = val;
+
+ if (mIsPacked) {
+ const int kTRTPEpochOffset = 8;
+ uint32_t* buf = reinterpret_cast<uint32_t*>(
+ mPacket + kTRTPEpochOffset);
+ uint32_t val = ntohl(*buf);
+ val &= ~(kTRTPEpochMask << kTRTPEpochShift);
+ val |= (mEpoch & kTRTPEpochMask) << kTRTPEpochShift;
+ *buf = htonl(val);
+ }
+}
+
+void TRTPPacket::setProgramID(uint16_t val) {
+ CHECK(!mIsPacked);
+ mProgramID = val;
+}
+
+void TRTPPacket::setSubstreamID(uint16_t val) {
+ CHECK(!mIsPacked);
+ mSubstreamID = val;
+}
+
+
+void TRTPPacket::setClockTransform(const LinearTransform& trans) {
+ CHECK(!mIsPacked);
+ mClockTranform = trans;
+ mClockTranformValid = true;
+}
+
+uint8_t* TRTPPacket::getPacket() const {
+ CHECK(mIsPacked);
+ return mPacket;
+}
+
+int TRTPPacket::getPacketLen() const {
+ CHECK(mIsPacked);
+ return mPacketLen;
+}
+
+void TRTPPacket::setExpireTime(nsecs_t val) {
+ CHECK(!mIsPacked);
+ mExpireTime = val;
+}
+
+nsecs_t TRTPPacket::getExpireTime() const {
+ return mExpireTime;
+}
+
+/*** TRTP audio packet properties ***/
+
+void TRTPAudioPacket::setCodecType(TRTPAudioCodecType val) {
+ CHECK(!mIsPacked);
+ mCodecType = val;
+}
+
+void TRTPAudioPacket::setRandomAccessPoint(bool val) {
+ CHECK(!mIsPacked);
+ mRandomAccessPoint = val;
+}
+
+void TRTPAudioPacket::setDropable(bool val) {
+ CHECK(!mIsPacked);
+ mDropable = val;
+}
+
+void TRTPAudioPacket::setDiscontinuity(bool val) {
+ CHECK(!mIsPacked);
+ mDiscontinuity = val;
+}
+
+void TRTPAudioPacket::setEndOfStream(bool val) {
+ CHECK(!mIsPacked);
+ mEndOfStream = val;
+}
+
+void TRTPAudioPacket::setVolume(uint8_t val) {
+ CHECK(!mIsPacked);
+ mVolume = val;
+}
+
+void TRTPAudioPacket::setAccessUnitData(void* data, int len) {
+ CHECK(!mIsPacked);
+ mAccessUnitData = data;
+ mAccessUnitLen = len;
+}
+
+/*** TRTP control packet properties ***/
+
+void TRTPControlPacket::setCommandID(TRTPCommandID val) {
+ CHECK(!mIsPacked);
+ mCommandID = val;
+}
+
+/*** TRTP packet serializers ***/
+
+void TRTPPacket::writeU8(uint8_t*& buf, uint8_t val) {
+ *buf = val;
+ buf++;
+}
+
+void TRTPPacket::writeU16(uint8_t*& buf, uint16_t val) {
+ *reinterpret_cast<uint16_t*>(buf) = htons(val);
+ buf += 2;
+}
+
+void TRTPPacket::writeU32(uint8_t*& buf, uint32_t val) {
+ *reinterpret_cast<uint32_t*>(buf) = htonl(val);
+ buf += 4;
+}
+
+void TRTPPacket::writeU64(uint8_t*& buf, uint64_t val) {
+ buf[0] = static_cast<uint8_t>(val >> 56);
+ buf[1] = static_cast<uint8_t>(val >> 48);
+ buf[2] = static_cast<uint8_t>(val >> 40);
+ buf[3] = static_cast<uint8_t>(val >> 32);
+ buf[4] = static_cast<uint8_t>(val >> 24);
+ buf[5] = static_cast<uint8_t>(val >> 16);
+ buf[6] = static_cast<uint8_t>(val >> 8);
+ buf[7] = static_cast<uint8_t>(val);
+ buf += 8;
+}
+
+void TRTPPacket::writeTRTPHeader(uint8_t*& buf,
+ bool isFirstFragment,
+ int totalPacketLen) {
+ // RTP header
+ writeU8(buf,
+ ((mVersion & 0x03) << 6) |
+ (static_cast<int>(mPadding) << 5) |
+ (static_cast<int>(mExtension) << 4) |
+ (mCsrcCount & 0x0F));
+ writeU8(buf,
+ (static_cast<int>(isFirstFragment) << 7) |
+ (mPayloadType & 0x7F));
+ writeU16(buf, mSeqNumber);
+ if (isFirstFragment && mPTSValid) {
+ writeU32(buf, mPTS & 0xFFFFFFFF);
+ } else {
+ writeU32(buf, 0);
+ }
+ writeU32(buf,
+ ((mEpoch & kTRTPEpochMask) << kTRTPEpochShift) |
+ ((mProgramID & 0x1F) << 5) |
+ (mSubstreamID & 0x1F));
+
+ // TRTP header
+ writeU8(buf, mTRTPVersion);
+ writeU8(buf,
+ ((mTRTPHeaderType & 0x0F) << 4) |
+ (mClockTranformValid ? 0x02 : 0x00) |
+ (mPTSValid ? 0x01 : 0x00));
+ writeU32(buf, totalPacketLen - kRTPHeaderLen);
+ if (mPTSValid) {
+ writeU32(buf, mPTS >> 32);
+ }
+
+ if (mClockTranformValid) {
+ writeU64(buf, mClockTranform.a_zero);
+ writeU32(buf, mClockTranform.a_to_b_numer);
+ writeU32(buf, mClockTranform.a_to_b_denom);
+ writeU64(buf, mClockTranform.b_zero);
+ }
+}
+
+bool TRTPAudioPacket::pack() {
+ if (mIsPacked) {
+ return false;
+ }
+
+ int packetLen = kRTPHeaderLen +
+ mAccessUnitLen +
+ TRTPHeaderLen();
+
+ // TODO : support multiple fragments
+ const int kMaxUDPPayloadLen = 65507;
+ if (packetLen > kMaxUDPPayloadLen) {
+ return false;
+ }
+
+ mPacket = new uint8_t[packetLen];
+ if (!mPacket) {
+ return false;
+ }
+
+ mPacketLen = packetLen;
+
+ uint8_t* cur = mPacket;
+
+ writeTRTPHeader(cur, true, packetLen);
+ writeU8(cur, mCodecType);
+ writeU8(cur,
+ (static_cast<int>(mRandomAccessPoint) << 3) |
+ (static_cast<int>(mDropable) << 2) |
+ (static_cast<int>(mDiscontinuity) << 1) |
+ (static_cast<int>(mEndOfStream)));
+ writeU8(cur, mVolume);
+
+ memcpy(cur, mAccessUnitData, mAccessUnitLen);
+
+ mIsPacked = true;
+ return true;
+}
+
+int TRTPPacket::TRTPHeaderLen() const {
+ // 6 bytes for version, payload type, flags and length. An additional 4 if
+ // there are upper timestamp bits present and another 24 if there is a clock
+ // transformation present.
+ return 6 +
+ (mClockTranformValid ? 24 : 0) +
+ (mPTSValid ? 4 : 0);
+}
+
+int TRTPAudioPacket::TRTPHeaderLen() const {
+ // TRTPPacket::TRTPHeaderLen() for the base TRTPHeader. 3 bytes for audio's
+ // codec type, flags and volume field. Another 5 bytes if the codec type is
+ // PCM and we are sending sample rate/channel count. as well as however long
+ // the aux data (if present) is.
+
+ int pcmParamLength;
+ switch(mCodecType) {
+ case kCodecPCMBigEndian:
+ case kCodecPCMLittleEndian:
+ pcmParamLength = 5;
+ break;
+
+ default:
+ pcmParamLength = 0;
+ break;
+ }
+
+
+ // TODO : properly compute aux data length. Currently, nothing
+ // uses aux data, so its length is always 0.
+ int auxDataLength = 0;
+ return TRTPPacket::TRTPHeaderLen() +
+ 3 +
+ auxDataLength +
+ pcmParamLength;
+}
+
+bool TRTPControlPacket::pack() {
+ if (mIsPacked) {
+ return false;
+ }
+
+ // command packets contain a 2-byte command ID
+ int packetLen = kRTPHeaderLen +
+ TRTPHeaderLen() +
+ 2;
+
+ mPacket = new uint8_t[packetLen];
+ if (!mPacket) {
+ return false;
+ }
+
+ mPacketLen = packetLen;
+
+ uint8_t* cur = mPacket;
+
+ writeTRTPHeader(cur, true, packetLen);
+ writeU16(cur, mCommandID);
+
+ mIsPacked = true;
+ return true;
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_packet.h b/media/libaah_rtp/aah_tx_packet.h
new file mode 100644
index 0000000..833803e
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_packet.h
@@ -0,0 +1,191 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AAH_TX_PACKET_H__
+#define __AAH_TX_PACKET_H__
+
+#include <media/stagefright/foundation/ABase.h>
+#include <utils/LinearTransform.h>
+#include <utils/RefBase.h>
+#include <utils/Timers.h>
+
+namespace android {
+
+class TRTPPacket : public RefBase {
+ protected:
+ enum TRTPHeaderType {
+ kHeaderTypeAudio = 1,
+ kHeaderTypeVideo = 2,
+ kHeaderTypeSubpicture = 3,
+ kHeaderTypeControl = 4,
+ };
+
+ TRTPPacket(TRTPHeaderType headerType)
+ : mIsPacked(false)
+ , mVersion(2)
+ , mPadding(false)
+ , mExtension(false)
+ , mCsrcCount(0)
+ , mPayloadType(100)
+ , mSeqNumber(0)
+ , mPTSValid(false)
+ , mPTS(0)
+ , mEpoch(0)
+ , mProgramID(0)
+ , mSubstreamID(0)
+ , mClockTranformValid(false)
+ , mTRTPVersion(1)
+ , mTRTPLength(0)
+ , mTRTPHeaderType(headerType)
+ , mPacket(NULL)
+ , mPacketLen(0) { }
+
+ public:
+ virtual ~TRTPPacket();
+
+ void setSeqNumber(uint16_t val);
+ uint16_t getSeqNumber() const;
+
+ void setPTS(int64_t val);
+ int64_t getPTS() const;
+
+ void setEpoch(uint32_t val);
+ void setProgramID(uint16_t val);
+ void setSubstreamID(uint16_t val);
+ void setClockTransform(const LinearTransform& trans);
+
+ uint8_t* getPacket() const;
+ int getPacketLen() const;
+
+ void setExpireTime(nsecs_t val);
+ nsecs_t getExpireTime() const;
+
+ virtual bool pack() = 0;
+
+ // mask for the number of bits in a TRTP epoch
+ static const uint32_t kTRTPEpochMask = (1 << 22) - 1;
+ static const int kTRTPEpochShift = 10;
+
+ protected:
+ static const int kRTPHeaderLen = 12;
+ virtual int TRTPHeaderLen() const;
+
+ void writeTRTPHeader(uint8_t*& buf,
+ bool isFirstFragment,
+ int totalPacketLen);
+
+ void writeU8(uint8_t*& buf, uint8_t val);
+ void writeU16(uint8_t*& buf, uint16_t val);
+ void writeU32(uint8_t*& buf, uint32_t val);
+ void writeU64(uint8_t*& buf, uint64_t val);
+
+ bool mIsPacked;
+
+ uint8_t mVersion;
+ bool mPadding;
+ bool mExtension;
+ uint8_t mCsrcCount;
+ uint8_t mPayloadType;
+ uint16_t mSeqNumber;
+ bool mPTSValid;
+ int64_t mPTS;
+ uint32_t mEpoch;
+ uint16_t mProgramID;
+ uint16_t mSubstreamID;
+ LinearTransform mClockTranform;
+ bool mClockTranformValid;
+ uint8_t mTRTPVersion;
+ uint32_t mTRTPLength;
+ TRTPHeaderType mTRTPHeaderType;
+
+ uint8_t* mPacket;
+ int mPacketLen;
+
+ nsecs_t mExpireTime;
+
+ DISALLOW_EVIL_CONSTRUCTORS(TRTPPacket);
+};
+
+class TRTPAudioPacket : public TRTPPacket {
+ public:
+ TRTPAudioPacket()
+ : TRTPPacket(kHeaderTypeAudio)
+ , mCodecType(kCodecInvalid)
+ , mRandomAccessPoint(false)
+ , mDropable(false)
+ , mDiscontinuity(false)
+ , mEndOfStream(false)
+ , mVolume(0)
+ , mAccessUnitData(NULL) { }
+
+ enum TRTPAudioCodecType {
+ kCodecInvalid = 0,
+ kCodecPCMBigEndian = 1,
+ kCodecPCMLittleEndian = 2,
+ kCodecMPEG1Audio = 3,
+ };
+
+ void setCodecType(TRTPAudioCodecType val);
+ void setRandomAccessPoint(bool val);
+ void setDropable(bool val);
+ void setDiscontinuity(bool val);
+ void setEndOfStream(bool val);
+ void setVolume(uint8_t val);
+ void setAccessUnitData(void* data, int len);
+
+ virtual bool pack();
+
+ protected:
+ virtual int TRTPHeaderLen() const;
+
+ private:
+ TRTPAudioCodecType mCodecType;
+ bool mRandomAccessPoint;
+ bool mDropable;
+ bool mDiscontinuity;
+ bool mEndOfStream;
+ uint8_t mVolume;
+ void* mAccessUnitData;
+ int mAccessUnitLen;
+
+ DISALLOW_EVIL_CONSTRUCTORS(TRTPAudioPacket);
+};
+
+class TRTPControlPacket : public TRTPPacket {
+ public:
+ TRTPControlPacket()
+ : TRTPPacket(kHeaderTypeControl)
+ , mCommandID(kCommandNop) {}
+
+ enum TRTPCommandID {
+ kCommandNop = 1,
+ kCommandFlush = 2,
+ kCommandEOS = 3,
+ };
+
+ void setCommandID(TRTPCommandID val);
+
+ virtual bool pack();
+
+ private:
+ TRTPCommandID mCommandID;
+
+ DISALLOW_EVIL_CONSTRUCTORS(TRTPControlPacket);
+};
+
+} // namespace android
+
+#endif // __AAH_TX_PLAYER_H__
diff --git a/media/libaah_rtp/aah_tx_player.cpp b/media/libaah_rtp/aah_tx_player.cpp
new file mode 100644
index 0000000..a79a989
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_player.cpp
@@ -0,0 +1,1139 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+#include <utils/Log.h>
+
+#define __STDC_FORMAT_MACROS
+#include <inttypes.h>
+#include <netdb.h>
+#include <netinet/ip.h>
+
+#include <common_time/cc_helper.h>
+#include <media/IMediaPlayer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/FileSource.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MetaData.h>
+#include <utils/Timers.h>
+
+#include "aah_tx_packet.h"
+#include "aah_tx_player.h"
+
+namespace android {
+
+static int64_t kLowWaterMarkUs = 2000000ll; // 2secs
+static int64_t kHighWaterMarkUs = 10000000ll; // 10secs
+static const size_t kLowWaterMarkBytes = 40000;
+static const size_t kHighWaterMarkBytes = 200000;
+
+// When we start up, how much lead time should we put on the first access unit?
+static const int64_t kAAHStartupLeadTimeUs = 300000LL;
+
+// How much time do we attempt to lead the clock by in steady state?
+static const int64_t kAAHBufferTimeUs = 1000000LL;
+
+// how long do we keep data in our retransmit buffer after sending it.
+const int64_t AAH_TXPlayer::kAAHRetryKeepAroundTimeNs =
+ kAAHBufferTimeUs * 1100;
+
+sp<MediaPlayerBase> createAAH_TXPlayer() {
+ sp<MediaPlayerBase> ret = new AAH_TXPlayer();
+ return ret;
+}
+
+template <typename T> static T clamp(T val, T min, T max) {
+ if (val < min) {
+ return min;
+ } else if (val > max) {
+ return max;
+ } else {
+ return val;
+ }
+}
+
+struct AAH_TXEvent : public TimedEventQueue::Event {
+ AAH_TXEvent(AAH_TXPlayer *player,
+ void (AAH_TXPlayer::*method)()) : mPlayer(player)
+ , mMethod(method) {}
+
+ protected:
+ virtual ~AAH_TXEvent() {}
+
+ virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) {
+ (mPlayer->*mMethod)();
+ }
+
+ private:
+ AAH_TXPlayer *mPlayer;
+ void (AAH_TXPlayer::*mMethod)();
+
+ AAH_TXEvent(const AAH_TXEvent &);
+ AAH_TXEvent& operator=(const AAH_TXEvent &);
+};
+
+AAH_TXPlayer::AAH_TXPlayer()
+ : mQueueStarted(false)
+ , mFlags(0)
+ , mExtractorFlags(0) {
+ DataSource::RegisterDefaultSniffers();
+
+ mBufferingEvent = new AAH_TXEvent(this, &AAH_TXPlayer::onBufferingUpdate);
+ mBufferingEventPending = false;
+
+ mPumpAudioEvent = new AAH_TXEvent(this, &AAH_TXPlayer::onPumpAudio);
+ mPumpAudioEventPending = false;
+
+ reset_l();
+}
+
+AAH_TXPlayer::~AAH_TXPlayer() {
+ if (mQueueStarted) {
+ mQueue.stop();
+ }
+
+ reset_l();
+}
+
+void AAH_TXPlayer::cancelPlayerEvents(bool keepBufferingGoing) {
+ if (!keepBufferingGoing) {
+ mQueue.cancelEvent(mBufferingEvent->eventID());
+ mBufferingEventPending = false;
+
+ mQueue.cancelEvent(mPumpAudioEvent->eventID());
+ mPumpAudioEventPending = false;
+ }
+}
+
+status_t AAH_TXPlayer::initCheck() {
+ // Check for the presense of the common time service by attempting to query
+ // for CommonTime's frequency. If we get an error back, we cannot talk to
+ // the service at all and should abort now.
+ status_t res;
+ uint64_t freq;
+ res = mCCHelper.getCommonFreq(&freq);
+ if (OK != res) {
+ ALOGE("Failed to connect to common time service! (res %d)", res);
+ return res;
+ }
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setDataSource(
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ Mutex::Autolock autoLock(mLock);
+ return setDataSource_l(url, headers);
+}
+
+status_t AAH_TXPlayer::setDataSource_l(
+ const char *url,
+ const KeyedVector<String8, String8> *headers) {
+ reset_l();
+
+ // the URL must consist of "aahTX://" followed by the real URL of
+ // the data source
+ const char *kAAHPrefix = "aahTX://";
+ if (strncasecmp(url, kAAHPrefix, strlen(kAAHPrefix))) {
+ return INVALID_OPERATION;
+ }
+
+ mUri.setTo(url + strlen(kAAHPrefix));
+
+ if (headers) {
+ mUriHeaders = *headers;
+
+ ssize_t index = mUriHeaders.indexOfKey(String8("x-hide-urls-from-log"));
+ if (index >= 0) {
+ // Browser is in "incognito" mode, suppress logging URLs.
+
+ // This isn't something that should be passed to the server.
+ mUriHeaders.removeItemsAt(index);
+
+ mFlags |= INCOGNITO;
+ }
+ }
+
+ // The URL may optionally contain a "#" character followed by a Skyjam
+ // cookie. Ideally the cookie header should just be passed in the headers
+ // argument, but the Java API for supplying headers is apparently not yet
+ // exposed in the SDK used by application developers.
+ const char kSkyjamCookieDelimiter = '#';
+ char* skyjamCookie = strrchr(mUri.string(), kSkyjamCookieDelimiter);
+ if (skyjamCookie) {
+ skyjamCookie++;
+ mUriHeaders.add(String8("Cookie"), String8(skyjamCookie));
+ mUri = String8(mUri.string(), skyjamCookie - mUri.string());
+ }
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setDataSource(int fd, int64_t offset, int64_t length) {
+ Mutex::Autolock autoLock(mLock);
+
+ reset_l();
+
+ sp<DataSource> dataSource = new FileSource(dup(fd), offset, length);
+
+ status_t err = dataSource->initCheck();
+
+ if (err != OK) {
+ return err;
+ }
+
+ mFileSource = dataSource;
+
+ sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);
+
+ if (extractor == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ return setDataSource_l(extractor);
+}
+
+status_t AAH_TXPlayer::setVideoSurface(const sp<Surface>& surface) {
+ return OK;
+}
+
+status_t AAH_TXPlayer::setVideoSurfaceTexture(
+ const sp<ISurfaceTexture>& surfaceTexture) {
+ return OK;
+}
+
+status_t AAH_TXPlayer::prepare() {
+ return INVALID_OPERATION;
+}
+
+status_t AAH_TXPlayer::prepareAsync() {
+ Mutex::Autolock autoLock(mLock);
+
+ return prepareAsync_l();
+}
+
+status_t AAH_TXPlayer::prepareAsync_l() {
+ if (mFlags & PREPARING) {
+ return UNKNOWN_ERROR; // async prepare already pending
+ }
+
+ mAAH_Sender = AAH_TXSender::GetInstance();
+ if (mAAH_Sender == NULL) {
+ return NO_MEMORY;
+ }
+
+ if (!mQueueStarted) {
+ mQueue.start();
+ mQueueStarted = true;
+ }
+
+ mFlags |= PREPARING;
+ mAsyncPrepareEvent = new AAH_TXEvent(
+ this, &AAH_TXPlayer::onPrepareAsyncEvent);
+
+ mQueue.postEvent(mAsyncPrepareEvent);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::finishSetDataSource_l() {
+ sp<DataSource> dataSource;
+
+ if (!strncasecmp("http://", mUri.string(), 7) ||
+ !strncasecmp("https://", mUri.string(), 8)) {
+
+ mConnectingDataSource = HTTPBase::Create(
+ (mFlags & INCOGNITO)
+ ? HTTPBase::kFlagIncognito
+ : 0);
+
+ mLock.unlock();
+ status_t err = mConnectingDataSource->connect(mUri, &mUriHeaders);
+ mLock.lock();
+
+ if (err != OK) {
+ mConnectingDataSource.clear();
+
+ ALOGI("mConnectingDataSource->connect() returned %d", err);
+ return err;
+ }
+
+ mCachedSource = new NuCachedSource2(mConnectingDataSource);
+ mConnectingDataSource.clear();
+
+ dataSource = mCachedSource;
+
+ // We're going to prefill the cache before trying to instantiate
+ // the extractor below, as the latter is an operation that otherwise
+ // could block on the datasource for a significant amount of time.
+ // During that time we'd be unable to abort the preparation phase
+ // without this prefill.
+
+ mLock.unlock();
+
+ for (;;) {
+ status_t finalStatus;
+ size_t cachedDataRemaining =
+ mCachedSource->approxDataRemaining(&finalStatus);
+
+ if (finalStatus != OK ||
+ cachedDataRemaining >= kHighWaterMarkBytes ||
+ (mFlags & PREPARE_CANCELLED)) {
+ break;
+ }
+
+ usleep(200000);
+ }
+
+ mLock.lock();
+
+ if (mFlags & PREPARE_CANCELLED) {
+ ALOGI("Prepare cancelled while waiting for initial cache fill.");
+ return UNKNOWN_ERROR;
+ }
+ } else {
+ dataSource = DataSource::CreateFromURI(mUri.string(), &mUriHeaders);
+ }
+
+ if (dataSource == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ sp<MediaExtractor> extractor = MediaExtractor::Create(dataSource);
+
+ if (extractor == NULL) {
+ return UNKNOWN_ERROR;
+ }
+
+ return setDataSource_l(extractor);
+}
+
+status_t AAH_TXPlayer::setDataSource_l(const sp<MediaExtractor> &extractor) {
+ // Attempt to approximate overall stream bitrate by summing all
+ // tracks' individual bitrates, if not all of them advertise bitrate,
+ // we have to fail.
+
+ int64_t totalBitRate = 0;
+
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<MetaData> meta = extractor->getTrackMetaData(i);
+
+ int32_t bitrate;
+ if (!meta->findInt32(kKeyBitRate, &bitrate)) {
+ totalBitRate = -1;
+ break;
+ }
+
+ totalBitRate += bitrate;
+ }
+
+ mBitrate = totalBitRate;
+
+ ALOGV("mBitrate = %lld bits/sec", mBitrate);
+
+ bool haveAudio = false;
+ for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ sp<MetaData> meta = extractor->getTrackMetaData(i);
+
+ const char *mime;
+ CHECK(meta->findCString(kKeyMIMEType, &mime));
+
+ if (!strncasecmp(mime, "audio/", 6)) {
+ mAudioSource = extractor->getTrack(i);
+ CHECK(mAudioSource != NULL);
+ haveAudio = true;
+ break;
+ }
+ }
+
+ if (!haveAudio) {
+ return UNKNOWN_ERROR;
+ }
+
+ mExtractorFlags = extractor->flags();
+
+ return OK;
+}
+
+void AAH_TXPlayer::abortPrepare(status_t err) {
+ CHECK(err != OK);
+
+ notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, err);
+
+ mPrepareResult = err;
+ mFlags &= ~(PREPARING|PREPARE_CANCELLED|PREPARING_CONNECTED);
+ mPreparedCondition.broadcast();
+}
+
+void AAH_TXPlayer::onPrepareAsyncEvent() {
+ Mutex::Autolock autoLock(mLock);
+
+ if (mFlags & PREPARE_CANCELLED) {
+ ALOGI("prepare was cancelled before doing anything");
+ abortPrepare(UNKNOWN_ERROR);
+ return;
+ }
+
+ if (mUri.size() > 0) {
+ status_t err = finishSetDataSource_l();
+
+ if (err != OK) {
+ abortPrepare(err);
+ return;
+ }
+ }
+
+ mAudioSource->getFormat()->findInt64(kKeyDuration, &mDurationUs);
+
+ status_t err = mAudioSource->start();
+ if (err != OK) {
+ ALOGI("failed to start audio source, err=%d", err);
+ abortPrepare(err);
+ return;
+ }
+
+ mFlags |= PREPARING_CONNECTED;
+
+ if (mCachedSource != NULL) {
+ postBufferingEvent_l();
+ } else {
+ finishAsyncPrepare_l();
+ }
+}
+
+void AAH_TXPlayer::finishAsyncPrepare_l() {
+ notifyListener_l(MEDIA_PREPARED);
+
+ mPrepareResult = OK;
+ mFlags &= ~(PREPARING|PREPARE_CANCELLED|PREPARING_CONNECTED);
+ mFlags |= PREPARED;
+ mPreparedCondition.broadcast();
+}
+
+status_t AAH_TXPlayer::start() {
+ Mutex::Autolock autoLock(mLock);
+
+ mFlags &= ~CACHE_UNDERRUN;
+
+ return play_l();
+}
+
+status_t AAH_TXPlayer::play_l() {
+ if (mFlags & PLAYING) {
+ return OK;
+ }
+
+ if (!(mFlags & PREPARED)) {
+ return INVALID_OPERATION;
+ }
+
+ {
+ Mutex::Autolock lock(mEndpointLock);
+ if (!mEndpointValid) {
+ return INVALID_OPERATION;
+ }
+ if (!mEndpointRegistered) {
+ mProgramID = mAAH_Sender->registerEndpoint(mEndpoint);
+ mEndpointRegistered = true;
+ }
+ }
+
+ mFlags |= PLAYING;
+
+ updateClockTransform_l(false);
+
+ postPumpAudioEvent_l(-1);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::stop() {
+ status_t ret = pause();
+ sendEOS_l();
+ return ret;
+}
+
+status_t AAH_TXPlayer::pause() {
+ Mutex::Autolock autoLock(mLock);
+
+ mFlags &= ~CACHE_UNDERRUN;
+
+ return pause_l();
+}
+
+status_t AAH_TXPlayer::pause_l(bool doClockUpdate) {
+ if (!(mFlags & PLAYING)) {
+ return OK;
+ }
+
+ cancelPlayerEvents(true /* keepBufferingGoing */);
+
+ mFlags &= ~PLAYING;
+
+ if (doClockUpdate) {
+ updateClockTransform_l(true);
+ }
+
+ return OK;
+}
+
+void AAH_TXPlayer::updateClockTransform_l(bool pause) {
+ // record the new pause status so that onPumpAudio knows what rate to apply
+ // when it initializes the transform
+ mPlayRateIsPaused = pause;
+
+ // if we haven't yet established a valid clock transform, then we can't
+ // do anything here
+ if (!mCurrentClockTransformValid) {
+ return;
+ }
+
+ // sample the current common time
+ int64_t commonTimeNow;
+ if (OK != mCCHelper.getCommonTime(&commonTimeNow)) {
+ ALOGE("updateClockTransform_l get common time failed");
+ mCurrentClockTransformValid = false;
+ return;
+ }
+
+ // convert the current common time to media time using the old
+ // transform
+ int64_t mediaTimeNow;
+ if (!mCurrentClockTransform.doReverseTransform(
+ commonTimeNow, &mediaTimeNow)) {
+ ALOGE("updateClockTransform_l reverse transform failed");
+ mCurrentClockTransformValid = false;
+ return;
+ }
+
+ // calculate a new transform that preserves the old transform's
+ // result for the current time
+ mCurrentClockTransform.a_zero = mediaTimeNow;
+ mCurrentClockTransform.b_zero = commonTimeNow;
+ mCurrentClockTransform.a_to_b_numer = 1;
+ mCurrentClockTransform.a_to_b_denom = pause ? 0 : 1;
+
+ // send a packet announcing the new transform
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setClockTransform(mCurrentClockTransform);
+ packet->setCommandID(TRTPControlPacket::kCommandNop);
+ queuePacketToSender_l(packet);
+}
+
+void AAH_TXPlayer::sendEOS_l() {
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setCommandID(TRTPControlPacket::kCommandEOS);
+ queuePacketToSender_l(packet);
+}
+
+bool AAH_TXPlayer::isPlaying() {
+ return (mFlags & PLAYING) || (mFlags & CACHE_UNDERRUN);
+}
+
+status_t AAH_TXPlayer::seekTo(int msec) {
+ if (mExtractorFlags & MediaExtractor::CAN_SEEK) {
+ Mutex::Autolock autoLock(mLock);
+ return seekTo_l(static_cast<int64_t>(msec) * 1000);
+ }
+
+ notifyListener_l(MEDIA_SEEK_COMPLETE);
+ return OK;
+}
+
+status_t AAH_TXPlayer::seekTo_l(int64_t timeUs) {
+ mIsSeeking = true;
+ mSeekTimeUs = timeUs;
+
+ mCurrentClockTransformValid = false;
+ mLastQueuedMediaTimePTSValid = false;
+
+ // send a flush command packet
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setCommandID(TRTPControlPacket::kCommandFlush);
+ queuePacketToSender_l(packet);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::getCurrentPosition(int *msec) {
+ if (!msec) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mLock);
+
+ int position;
+
+ if (mIsSeeking) {
+ position = mSeekTimeUs / 1000;
+ } else if (mCurrentClockTransformValid) {
+ // sample the current common time
+ int64_t commonTimeNow;
+ if (OK != mCCHelper.getCommonTime(&commonTimeNow)) {
+ ALOGE("getCurrentPosition get common time failed");
+ return INVALID_OPERATION;
+ }
+
+ int64_t mediaTimeNow;
+ if (!mCurrentClockTransform.doReverseTransform(commonTimeNow,
+ &mediaTimeNow)) {
+ ALOGE("getCurrentPosition reverse transform failed");
+ return INVALID_OPERATION;
+ }
+
+ position = static_cast<int>(mediaTimeNow / 1000);
+ } else {
+ position = 0;
+ }
+
+ int duration;
+ if (getDuration_l(&duration) == OK) {
+ *msec = clamp(position, 0, duration);
+ } else {
+ *msec = (position >= 0) ? position : 0;
+ }
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::getDuration(int* msec) {
+ if (!msec) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mLock);
+
+ return getDuration_l(msec);
+}
+
+status_t AAH_TXPlayer::getDuration_l(int* msec) {
+ if (mDurationUs < 0) {
+ return UNKNOWN_ERROR;
+ }
+
+ *msec = (mDurationUs + 500) / 1000;
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::reset() {
+ Mutex::Autolock autoLock(mLock);
+ reset_l();
+ return OK;
+}
+
+void AAH_TXPlayer::reset_l() {
+ if (mFlags & PREPARING) {
+ mFlags |= PREPARE_CANCELLED;
+ if (mConnectingDataSource != NULL) {
+ ALOGI("interrupting the connection process");
+ mConnectingDataSource->disconnect();
+ }
+
+ if (mFlags & PREPARING_CONNECTED) {
+ // We are basically done preparing, we're just buffering
+ // enough data to start playback, we can safely interrupt that.
+ finishAsyncPrepare_l();
+ }
+ }
+
+ while (mFlags & PREPARING) {
+ mPreparedCondition.wait(mLock);
+ }
+
+ cancelPlayerEvents();
+
+ sendEOS_l();
+
+ mCachedSource.clear();
+
+ if (mAudioSource != NULL) {
+ mAudioSource->stop();
+ }
+ mAudioSource.clear();
+
+ mFlags = 0;
+ mExtractorFlags = 0;
+
+ mDurationUs = -1;
+ mIsSeeking = false;
+ mSeekTimeUs = 0;
+
+ mUri.setTo("");
+ mUriHeaders.clear();
+
+ mFileSource.clear();
+
+ mBitrate = -1;
+
+ {
+ Mutex::Autolock lock(mEndpointLock);
+ if (mAAH_Sender != NULL && mEndpointRegistered) {
+ mAAH_Sender->unregisterEndpoint(mEndpoint);
+ }
+ mEndpointRegistered = false;
+ mEndpointValid = false;
+ }
+
+ mProgramID = 0;
+
+ mAAH_Sender.clear();
+ mLastQueuedMediaTimePTSValid = false;
+ mCurrentClockTransformValid = false;
+ mPlayRateIsPaused = false;
+
+ mTRTPVolume = 255;
+}
+
+status_t AAH_TXPlayer::setLooping(int loop) {
+ return OK;
+}
+
+player_type AAH_TXPlayer::playerType() {
+ return AAH_TX_PLAYER;
+}
+
+status_t AAH_TXPlayer::setParameter(int key, const Parcel &request) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_TXPlayer::getParameter(int key, Parcel *reply) {
+ return ERROR_UNSUPPORTED;
+}
+
+status_t AAH_TXPlayer::invoke(const Parcel& request, Parcel *reply) {
+ if (!reply) {
+ return BAD_VALUE;
+ }
+
+ int32_t methodID;
+ status_t err = request.readInt32(&methodID);
+ if (err != android::OK) {
+ return err;
+ }
+
+ switch (methodID) {
+ case kInvokeSetAAHDstIPPort:
+ case kInvokeSetAAHConfigBlob: {
+ if (mEndpointValid) {
+ return INVALID_OPERATION;
+ }
+
+ String8 addr;
+ uint16_t port;
+
+ if (methodID == kInvokeSetAAHDstIPPort) {
+ addr = String8(request.readString16());
+
+ int32_t port32;
+ err = request.readInt32(&port32);
+ if (err != android::OK) {
+ return err;
+ }
+ port = static_cast<uint16_t>(port32);
+ } else {
+ String8 blob(request.readString16());
+
+ char addr_buf[101];
+ if (sscanf(blob.string(), "V1:%100s %" SCNu16,
+ addr_buf, &port) != 2) {
+ return BAD_VALUE;
+ }
+ if (addr.setTo(addr_buf) != OK) {
+ return NO_MEMORY;
+ }
+ }
+
+ struct hostent* ent = gethostbyname(addr.string());
+ if (ent == NULL) {
+ return ERROR_UNKNOWN_HOST;
+ }
+ if (!(ent->h_addrtype == AF_INET && ent->h_length == 4)) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mEndpointLock);
+ mEndpoint = AAH_TXSender::Endpoint(
+ reinterpret_cast<struct in_addr*>(ent->h_addr)->s_addr,
+ port);
+ mEndpointValid = true;
+ return OK;
+ };
+
+ default:
+ return INVALID_OPERATION;
+ }
+}
+
+status_t AAH_TXPlayer::getMetadata(const media::Metadata::Filter& ids,
+ Parcel* records) {
+ using media::Metadata;
+
+ Metadata metadata(records);
+
+ metadata.appendBool(Metadata::kPauseAvailable, true);
+ metadata.appendBool(Metadata::kSeekBackwardAvailable, false);
+ metadata.appendBool(Metadata::kSeekForwardAvailable, false);
+ metadata.appendBool(Metadata::kSeekAvailable, false);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setVolume(float leftVolume, float rightVolume) {
+ if (leftVolume != rightVolume) {
+ ALOGE("%s does not support per channel volume: %f, %f",
+ __PRETTY_FUNCTION__, leftVolume, rightVolume);
+ }
+
+ float volume = clamp(leftVolume, 0.0f, 1.0f);
+
+ Mutex::Autolock lock(mLock);
+ mTRTPVolume = static_cast<uint8_t>((leftVolume * 255.0) + 0.5);
+
+ return OK;
+}
+
+status_t AAH_TXPlayer::setAudioStreamType(audio_stream_type_t streamType) {
+ return OK;
+}
+
+void AAH_TXPlayer::notifyListener_l(int msg, int ext1, int ext2) {
+ sendEvent(msg, ext1, ext2);
+}
+
+bool AAH_TXPlayer::getBitrate_l(int64_t *bitrate) {
+ off64_t size;
+ if (mDurationUs >= 0 &&
+ mCachedSource != NULL &&
+ mCachedSource->getSize(&size) == OK) {
+ *bitrate = size * 8000000ll / mDurationUs; // in bits/sec
+ return true;
+ }
+
+ if (mBitrate >= 0) {
+ *bitrate = mBitrate;
+ return true;
+ }
+
+ *bitrate = 0;
+
+ return false;
+}
+
+// Returns true iff cached duration is available/applicable.
+bool AAH_TXPlayer::getCachedDuration_l(int64_t *durationUs, bool *eos) {
+ int64_t bitrate;
+
+ if (mCachedSource != NULL && getBitrate_l(&bitrate)) {
+ status_t finalStatus;
+ size_t cachedDataRemaining = mCachedSource->approxDataRemaining(
+ &finalStatus);
+ *durationUs = cachedDataRemaining * 8000000ll / bitrate;
+ *eos = (finalStatus != OK);
+ return true;
+ }
+
+ return false;
+}
+
+void AAH_TXPlayer::ensureCacheIsFetching_l() {
+ if (mCachedSource != NULL) {
+ mCachedSource->resumeFetchingIfNecessary();
+ }
+}
+
+void AAH_TXPlayer::postBufferingEvent_l() {
+ if (mBufferingEventPending) {
+ return;
+ }
+ mBufferingEventPending = true;
+ mQueue.postEventWithDelay(mBufferingEvent, 1000000ll);
+}
+
+void AAH_TXPlayer::postPumpAudioEvent_l(int64_t delayUs) {
+ if (mPumpAudioEventPending) {
+ return;
+ }
+ mPumpAudioEventPending = true;
+ mQueue.postEventWithDelay(mPumpAudioEvent, delayUs < 0 ? 10000 : delayUs);
+}
+
+void AAH_TXPlayer::onBufferingUpdate() {
+ Mutex::Autolock autoLock(mLock);
+ if (!mBufferingEventPending) {
+ return;
+ }
+ mBufferingEventPending = false;
+
+ if (mCachedSource != NULL) {
+ status_t finalStatus;
+ size_t cachedDataRemaining = mCachedSource->approxDataRemaining(
+ &finalStatus);
+ bool eos = (finalStatus != OK);
+
+ if (eos) {
+ if (finalStatus == ERROR_END_OF_STREAM) {
+ notifyListener_l(MEDIA_BUFFERING_UPDATE, 100);
+ }
+ if (mFlags & PREPARING) {
+ ALOGV("cache has reached EOS, prepare is done.");
+ finishAsyncPrepare_l();
+ }
+ } else {
+ int64_t bitrate;
+ if (getBitrate_l(&bitrate)) {
+ size_t cachedSize = mCachedSource->cachedSize();
+ int64_t cachedDurationUs = cachedSize * 8000000ll / bitrate;
+
+ int percentage = (100.0 * (double) cachedDurationUs)
+ / mDurationUs;
+ if (percentage > 100) {
+ percentage = 100;
+ }
+
+ notifyListener_l(MEDIA_BUFFERING_UPDATE, percentage);
+ } else {
+ // We don't know the bitrate of the stream, use absolute size
+ // limits to maintain the cache.
+
+ if ((mFlags & PLAYING) &&
+ !eos &&
+ (cachedDataRemaining < kLowWaterMarkBytes)) {
+ ALOGI("cache is running low (< %d) , pausing.",
+ kLowWaterMarkBytes);
+ mFlags |= CACHE_UNDERRUN;
+ pause_l();
+ ensureCacheIsFetching_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);
+ } else if (eos || cachedDataRemaining > kHighWaterMarkBytes) {
+ if (mFlags & CACHE_UNDERRUN) {
+ ALOGI("cache has filled up (> %d), resuming.",
+ kHighWaterMarkBytes);
+ mFlags &= ~CACHE_UNDERRUN;
+ play_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_END);
+ } else if (mFlags & PREPARING) {
+ ALOGV("cache has filled up (> %d), prepare is done",
+ kHighWaterMarkBytes);
+ finishAsyncPrepare_l();
+ }
+ }
+ }
+ }
+ }
+
+ int64_t cachedDurationUs;
+ bool eos;
+ if (getCachedDuration_l(&cachedDurationUs, &eos)) {
+ ALOGV("cachedDurationUs = %.2f secs, eos=%d",
+ cachedDurationUs / 1E6, eos);
+
+ if ((mFlags & PLAYING) &&
+ !eos &&
+ (cachedDurationUs < kLowWaterMarkUs)) {
+ ALOGI("cache is running low (%.2f secs) , pausing.",
+ cachedDurationUs / 1E6);
+ mFlags |= CACHE_UNDERRUN;
+ pause_l();
+ ensureCacheIsFetching_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_START);
+ } else if (eos || cachedDurationUs > kHighWaterMarkUs) {
+ if (mFlags & CACHE_UNDERRUN) {
+ ALOGI("cache has filled up (%.2f secs), resuming.",
+ cachedDurationUs / 1E6);
+ mFlags &= ~CACHE_UNDERRUN;
+ play_l();
+ notifyListener_l(MEDIA_INFO, MEDIA_INFO_BUFFERING_END);
+ } else if (mFlags & PREPARING) {
+ ALOGV("cache has filled up (%.2f secs), prepare is done",
+ cachedDurationUs / 1E6);
+ finishAsyncPrepare_l();
+ }
+ }
+ }
+
+ postBufferingEvent_l();
+}
+
+void AAH_TXPlayer::onPumpAudio() {
+ while (true) {
+ Mutex::Autolock autoLock(mLock);
+ // If this flag is clear, its because someone has externally canceled
+ // this pump operation (probably because we a resetting/shutting down).
+ // Get out immediately, do not reschedule ourselves.
+ if (!mPumpAudioEventPending) {
+ return;
+ }
+
+ // Start by checking if there is still work to be doing. If we have
+ // never queued a payload (so we don't know what the last queued PTS is)
+ // or we have never established a MediaTime->CommonTime transformation,
+ // then we have work to do (one time through this loop should establish
+ // both). Otherwise, we want to keep a fixed amt of presentation time
+ // worth of data buffered. If we cannot get common time (service is
+ // unavailable, or common time is undefined)) then we don't have a lot
+ // of good options here. For now, signal an error up to the app level
+ // and shut down the transmission pump.
+ int64_t commonTimeNow;
+ if (OK != mCCHelper.getCommonTime(&commonTimeNow)) {
+ // Failed to get common time; either the service is down or common
+ // time is not synced. Raise an error and shutdown the player.
+ ALOGE("*** Cannot pump audio, unable to fetch common time."
+ " Shutting down.");
+ notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, UNKNOWN_ERROR);
+ mPumpAudioEventPending = false;
+ break;
+ }
+
+ if (mCurrentClockTransformValid && mLastQueuedMediaTimePTSValid) {
+ int64_t mediaTimeNow;
+ bool conversionResult = mCurrentClockTransform.doReverseTransform(
+ commonTimeNow,
+ &mediaTimeNow);
+ CHECK(conversionResult);
+
+ if ((mediaTimeNow +
+ kAAHBufferTimeUs -
+ mLastQueuedMediaTimePTS) <= 0) {
+ break;
+ }
+ }
+
+ MediaSource::ReadOptions options;
+ if (mIsSeeking) {
+ options.setSeekTo(mSeekTimeUs);
+ }
+
+ MediaBuffer* mediaBuffer;
+ status_t err = mAudioSource->read(&mediaBuffer, &options);
+ if (err != NO_ERROR) {
+ if (err == ERROR_END_OF_STREAM) {
+ ALOGI("*** %s reached end of stream", __PRETTY_FUNCTION__);
+ notifyListener_l(MEDIA_BUFFERING_UPDATE, 100);
+ notifyListener_l(MEDIA_PLAYBACK_COMPLETE);
+ pause_l(false);
+ sendEOS_l();
+ } else {
+ ALOGE("*** %s read failed err=%d", __PRETTY_FUNCTION__, err);
+ }
+ return;
+ }
+
+ if (mIsSeeking) {
+ mIsSeeking = false;
+ notifyListener_l(MEDIA_SEEK_COMPLETE);
+ }
+
+ uint8_t* data = (static_cast<uint8_t*>(mediaBuffer->data()) +
+ mediaBuffer->range_offset());
+ ALOGV("*** %s got media buffer data=[%02hhx %02hhx %02hhx %02hhx]"
+ " offset=%d length=%d", __PRETTY_FUNCTION__,
+ data[0], data[1], data[2], data[3],
+ mediaBuffer->range_offset(), mediaBuffer->range_length());
+
+ int64_t mediaTimeUs;
+ CHECK(mediaBuffer->meta_data()->findInt64(kKeyTime, &mediaTimeUs));
+ ALOGV("*** timeUs=%lld", mediaTimeUs);
+
+ if (!mCurrentClockTransformValid) {
+ if (OK == mCCHelper.getCommonTime(&commonTimeNow)) {
+ mCurrentClockTransform.a_zero = mediaTimeUs;
+ mCurrentClockTransform.b_zero = commonTimeNow +
+ kAAHStartupLeadTimeUs;
+ mCurrentClockTransform.a_to_b_numer = 1;
+ mCurrentClockTransform.a_to_b_denom = mPlayRateIsPaused ? 0 : 1;
+ mCurrentClockTransformValid = true;
+ } else {
+ // Failed to get common time; either the service is down or
+ // common time is not synced. Raise an error and shutdown the
+ // player.
+ ALOGE("*** Cannot begin transmission, unable to fetch common"
+ " time. Dropping sample with pts=%lld", mediaTimeUs);
+ notifyListener_l(MEDIA_ERROR,
+ MEDIA_ERROR_UNKNOWN,
+ UNKNOWN_ERROR);
+ mPumpAudioEventPending = false;
+ break;
+ }
+ }
+
+ ALOGV("*** transmitting packet with pts=%lld", mediaTimeUs);
+
+ sp<TRTPAudioPacket> packet = new TRTPAudioPacket();
+ packet->setPTS(mediaTimeUs);
+ packet->setSubstreamID(1);
+
+ packet->setCodecType(TRTPAudioPacket::kCodecMPEG1Audio);
+ packet->setVolume(mTRTPVolume);
+ // TODO : introduce a throttle for this so we can control the
+ // frequency with which transforms get sent.
+ packet->setClockTransform(mCurrentClockTransform);
+ packet->setAccessUnitData(data, mediaBuffer->range_length());
+ packet->setRandomAccessPoint(true);
+
+ queuePacketToSender_l(packet);
+ mediaBuffer->release();
+
+ mLastQueuedMediaTimePTSValid = true;
+ mLastQueuedMediaTimePTS = mediaTimeUs;
+ }
+
+ { // Explicit scope for the autolock pattern.
+ Mutex::Autolock autoLock(mLock);
+
+ // If someone externally has cleared this flag, its because we should be
+ // shutting down. Do not reschedule ourselves.
+ if (!mPumpAudioEventPending) {
+ return;
+ }
+
+ // Looks like no one canceled us explicitly. Clear our flag and post a
+ // new event to ourselves.
+ mPumpAudioEventPending = false;
+ postPumpAudioEvent_l(10000);
+ }
+}
+
+void AAH_TXPlayer::queuePacketToSender_l(const sp<TRTPPacket>& packet) {
+ if (mAAH_Sender == NULL) {
+ return;
+ }
+
+ sp<AMessage> message = new AMessage(AAH_TXSender::kWhatSendPacket,
+ mAAH_Sender->handlerID());
+
+ {
+ Mutex::Autolock lock(mEndpointLock);
+ if (!mEndpointValid) {
+ return;
+ }
+
+ message->setInt32(AAH_TXSender::kSendPacketIPAddr, mEndpoint.addr);
+ message->setInt32(AAH_TXSender::kSendPacketPort, mEndpoint.port);
+ }
+
+ packet->setProgramID(mProgramID);
+ packet->setExpireTime(systemTime() + kAAHRetryKeepAroundTimeNs);
+ packet->pack();
+
+ message->setObject(AAH_TXSender::kSendPacketTRTPPacket, packet);
+
+ message->post();
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_player.h b/media/libaah_rtp/aah_tx_player.h
new file mode 100644
index 0000000..64cf5dc
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_player.h
@@ -0,0 +1,179 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AAH_TX_PLAYER_H__
+#define __AAH_TX_PLAYER_H__
+
+#include <common_time/cc_helper.h>
+#include <libstagefright/include/HTTPBase.h>
+#include <libstagefright/include/NuCachedSource2.h>
+#include <libstagefright/include/TimedEventQueue.h>
+#include <media/MediaPlayerInterface.h>
+#include <media/stagefright/MediaExtractor.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/LinearTransform.h>
+#include <utils/String8.h>
+#include <utils/threads.h>
+
+#include "aah_tx_sender.h"
+
+namespace android {
+
+class AAH_TXPlayer : public MediaPlayerHWInterface {
+ public:
+ AAH_TXPlayer();
+
+ virtual status_t initCheck();
+ virtual status_t setDataSource(const char *url,
+ const KeyedVector<String8, String8>*
+ headers);
+ virtual status_t setDataSource(int fd, int64_t offset, int64_t length);
+ virtual status_t setVideoSurface(const sp<Surface>& surface);
+ virtual status_t setVideoSurfaceTexture(const sp<ISurfaceTexture>&
+ surfaceTexture);
+ virtual status_t prepare();
+ virtual status_t prepareAsync();
+ virtual status_t start();
+ virtual status_t stop();
+ virtual status_t pause();
+ virtual bool isPlaying();
+ virtual status_t seekTo(int msec);
+ virtual status_t getCurrentPosition(int *msec);
+ virtual status_t getDuration(int *msec);
+ virtual status_t reset();
+ virtual status_t setLooping(int loop);
+ virtual player_type playerType();
+ virtual status_t setParameter(int key, const Parcel &request);
+ virtual status_t getParameter(int key, Parcel *reply);
+ virtual status_t invoke(const Parcel& request, Parcel *reply);
+ virtual status_t getMetadata(const media::Metadata::Filter& ids,
+ Parcel* records);
+ virtual status_t setVolume(float leftVolume, float rightVolume);
+ virtual status_t setAudioStreamType(audio_stream_type_t streamType);
+
+ // invoke method IDs
+ enum {
+ // set the IP address and port of the A@H receiver
+ kInvokeSetAAHDstIPPort = 1,
+
+ // set the destination IP address and port (and perhaps any additional
+ // parameters added in the future) packaged in one string
+ kInvokeSetAAHConfigBlob,
+ };
+
+ static const int64_t kAAHRetryKeepAroundTimeNs;
+
+ protected:
+ virtual ~AAH_TXPlayer();
+
+ private:
+ friend struct AwesomeEvent;
+
+ enum {
+ PLAYING = 1,
+ PREPARING = 8,
+ PREPARED = 16,
+ PREPARE_CANCELLED = 64,
+ CACHE_UNDERRUN = 128,
+
+ // We are basically done preparing but are currently buffering
+ // sufficient data to begin playback and finish the preparation
+ // phase for good.
+ PREPARING_CONNECTED = 2048,
+
+ INCOGNITO = 32768,
+ };
+
+ status_t setDataSource_l(const char *url,
+ const KeyedVector<String8, String8> *headers);
+ status_t setDataSource_l(const sp<MediaExtractor>& extractor);
+ status_t finishSetDataSource_l();
+ status_t prepareAsync_l();
+ void onPrepareAsyncEvent();
+ void finishAsyncPrepare_l();
+ void abortPrepare(status_t err);
+ status_t play_l();
+ status_t pause_l(bool doClockUpdate = true);
+ status_t seekTo_l(int64_t timeUs);
+ void updateClockTransform_l(bool pause);
+ void sendEOS_l();
+ void cancelPlayerEvents(bool keepBufferingGoing = false);
+ void reset_l();
+ void notifyListener_l(int msg, int ext1 = 0, int ext2 = 0);
+ bool getBitrate_l(int64_t* bitrate);
+ status_t getDuration_l(int* msec);
+ bool getCachedDuration_l(int64_t* durationUs, bool* eos);
+ void ensureCacheIsFetching_l();
+ void postBufferingEvent_l();
+ void postPumpAudioEvent_l(int64_t delayUs);
+ void onBufferingUpdate();
+ void onPumpAudio();
+ void queuePacketToSender_l(const sp<TRTPPacket>& packet);
+
+ Mutex mLock;
+
+ TimedEventQueue mQueue;
+ bool mQueueStarted;
+
+ sp<TimedEventQueue::Event> mBufferingEvent;
+ bool mBufferingEventPending;
+
+ uint32_t mFlags;
+ uint32_t mExtractorFlags;
+
+ String8 mUri;
+ KeyedVector<String8, String8> mUriHeaders;
+
+ sp<DataSource> mFileSource;
+
+ sp<TimedEventQueue::Event> mAsyncPrepareEvent;
+ Condition mPreparedCondition;
+ status_t mPrepareResult;
+
+ bool mIsSeeking;
+ int64_t mSeekTimeUs;
+
+ sp<TimedEventQueue::Event> mPumpAudioEvent;
+ bool mPumpAudioEventPending;
+
+ sp<HTTPBase> mConnectingDataSource;
+ sp<NuCachedSource2> mCachedSource;
+
+ sp<MediaSource> mAudioSource;
+ int64_t mDurationUs;
+ int64_t mBitrate;
+
+ sp<AAH_TXSender> mAAH_Sender;
+ LinearTransform mCurrentClockTransform;
+ bool mCurrentClockTransformValid;
+ int64_t mLastQueuedMediaTimePTS;
+ bool mLastQueuedMediaTimePTSValid;
+ bool mPlayRateIsPaused;
+ CCHelper mCCHelper;
+
+ Mutex mEndpointLock;
+ AAH_TXSender::Endpoint mEndpoint;
+ bool mEndpointValid;
+ bool mEndpointRegistered;
+ uint16_t mProgramID;
+ uint8_t mTRTPVolume;
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_TXPlayer);
+};
+
+} // namespace android
+
+#endif // __AAH_TX_PLAYER_H__
diff --git a/media/libaah_rtp/aah_tx_sender.cpp b/media/libaah_rtp/aah_tx_sender.cpp
new file mode 100644
index 0000000..d991ea7
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_sender.cpp
@@ -0,0 +1,602 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+#include <media/stagefright/foundation/ADebug.h>
+
+#include <netinet/in.h>
+#include <poll.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <unistd.h>
+
+#include <media/stagefright/foundation/AMessage.h>
+#include <utils/misc.h>
+
+#include "aah_tx_player.h"
+#include "aah_tx_sender.h"
+
+namespace android {
+
+const char* AAH_TXSender::kSendPacketIPAddr = "ipaddr";
+const char* AAH_TXSender::kSendPacketPort = "port";
+const char* AAH_TXSender::kSendPacketTRTPPacket = "trtp";
+
+const int AAH_TXSender::kRetryTrimIntervalUs = 100000;
+const int AAH_TXSender::kHeartbeatIntervalUs = 1000000;
+const int AAH_TXSender::kRetryBufferCapacity = 100;
+const nsecs_t AAH_TXSender::kHeartbeatTimeout = 600ull * 1000000000ull;
+
+Mutex AAH_TXSender::sLock;
+wp<AAH_TXSender> AAH_TXSender::sInstance;
+uint32_t AAH_TXSender::sNextEpoch;
+bool AAH_TXSender::sNextEpochValid = false;
+
+AAH_TXSender::AAH_TXSender() : mSocket(-1) {
+ mLastSentPacketTime = systemTime();
+}
+
+sp<AAH_TXSender> AAH_TXSender::GetInstance() {
+ Mutex::Autolock autoLock(sLock);
+
+ sp<AAH_TXSender> sender = sInstance.promote();
+
+ if (sender == NULL) {
+ sender = new AAH_TXSender();
+ if (sender == NULL) {
+ return NULL;
+ }
+
+ sender->mLooper = new ALooper();
+ if (sender->mLooper == NULL) {
+ return NULL;
+ }
+
+ sender->mReflector = new AHandlerReflector<AAH_TXSender>(sender.get());
+ if (sender->mReflector == NULL) {
+ return NULL;
+ }
+
+ sender->mSocket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
+ if (sender->mSocket == -1) {
+ ALOGW("%s unable to create socket", __PRETTY_FUNCTION__);
+ return NULL;
+ }
+
+ struct sockaddr_in bind_addr;
+ memset(&bind_addr, 0, sizeof(bind_addr));
+ bind_addr.sin_family = AF_INET;
+ if (bind(sender->mSocket,
+ reinterpret_cast<const sockaddr*>(&bind_addr),
+ sizeof(bind_addr)) < 0) {
+ ALOGW("%s unable to bind socket (errno %d)",
+ __PRETTY_FUNCTION__, errno);
+ return NULL;
+ }
+
+ sender->mRetryReceiver = new RetryReceiver(sender.get());
+ if (sender->mRetryReceiver == NULL) {
+ return NULL;
+ }
+
+ sender->mLooper->setName("AAH_TXSender");
+ sender->mLooper->registerHandler(sender->mReflector);
+ sender->mLooper->start(false, false, PRIORITY_AUDIO);
+
+ if (sender->mRetryReceiver->run("AAH_TXSenderRetry", PRIORITY_AUDIO)
+ != OK) {
+ ALOGW("%s unable to start retry thread", __PRETTY_FUNCTION__);
+ return NULL;
+ }
+
+ sInstance = sender;
+ }
+
+ return sender;
+}
+
+AAH_TXSender::~AAH_TXSender() {
+ mLooper->stop();
+ mLooper->unregisterHandler(mReflector->id());
+
+ if (mRetryReceiver != NULL) {
+ mRetryReceiver->requestExit();
+ mRetryReceiver->mWakeupEvent.setEvent();
+ if (mRetryReceiver->requestExitAndWait() != OK) {
+ ALOGW("%s shutdown of retry receiver failed", __PRETTY_FUNCTION__);
+ }
+ mRetryReceiver->mSender = NULL;
+ mRetryReceiver.clear();
+ }
+
+ if (mSocket != -1) {
+ close(mSocket);
+ }
+}
+
+// Return the next epoch number usable for a newly instantiated endpoint.
+uint32_t AAH_TXSender::getNextEpoch() {
+ Mutex::Autolock autoLock(sLock);
+
+ if (sNextEpochValid) {
+ sNextEpoch = (sNextEpoch + 1) & TRTPPacket::kTRTPEpochMask;
+ } else {
+ sNextEpoch = ns2ms(systemTime()) & TRTPPacket::kTRTPEpochMask;
+ sNextEpochValid = true;
+ }
+
+ return sNextEpoch;
+}
+
+// Notify the sender that a player has started sending to this endpoint.
+// Returns a program ID for use by the calling player.
+uint16_t AAH_TXSender::registerEndpoint(const Endpoint& endpoint) {
+ Mutex::Autolock lock(mEndpointLock);
+
+ EndpointState* eps = mEndpointMap.valueFor(endpoint);
+ if (eps) {
+ eps->playerRefCount++;
+ } else {
+ eps = new EndpointState(getNextEpoch());
+ mEndpointMap.add(endpoint, eps);
+ }
+
+ // if this is the first registered endpoint, then send a message to start
+ // trimming retry buffers and a message to start sending heartbeats.
+ if (mEndpointMap.size() == 1) {
+ sp<AMessage> trimMessage = new AMessage(kWhatTrimRetryBuffers,
+ handlerID());
+ trimMessage->post(kRetryTrimIntervalUs);
+
+ sp<AMessage> heartbeatMessage = new AMessage(kWhatSendHeartbeats,
+ handlerID());
+ heartbeatMessage->post(kHeartbeatIntervalUs);
+ }
+
+ eps->nextProgramID++;
+ return eps->nextProgramID;
+}
+
+// Notify the sender that a player has ceased sending to this endpoint.
+// An endpoint's state can not be deleted until all of the endpoint's
+// registered players have called unregisterEndpoint.
+void AAH_TXSender::unregisterEndpoint(const Endpoint& endpoint) {
+ Mutex::Autolock lock(mEndpointLock);
+
+ EndpointState* eps = mEndpointMap.valueFor(endpoint);
+ if (eps) {
+ eps->playerRefCount--;
+ CHECK(eps->playerRefCount >= 0);
+ }
+}
+
+void AAH_TXSender::onMessageReceived(const sp<AMessage>& msg) {
+ switch (msg->what()) {
+ case kWhatSendPacket:
+ onSendPacket(msg);
+ break;
+
+ case kWhatTrimRetryBuffers:
+ trimRetryBuffers();
+ break;
+
+ case kWhatSendHeartbeats:
+ sendHeartbeats();
+ break;
+
+ default:
+ TRESPASS();
+ break;
+ }
+}
+
+void AAH_TXSender::onSendPacket(const sp<AMessage>& msg) {
+ sp<RefBase> obj;
+ CHECK(msg->findObject(kSendPacketTRTPPacket, &obj));
+ sp<TRTPPacket> packet = static_cast<TRTPPacket*>(obj.get());
+
+ uint32_t ipAddr;
+ CHECK(msg->findInt32(kSendPacketIPAddr,
+ reinterpret_cast<int32_t*>(&ipAddr)));
+
+ int32_t port32;
+ CHECK(msg->findInt32(kSendPacketPort, &port32));
+ uint16_t port = port32;
+
+ Mutex::Autolock lock(mEndpointLock);
+ doSendPacket_l(packet, Endpoint(ipAddr, port));
+ mLastSentPacketTime = systemTime();
+}
+
+void AAH_TXSender::doSendPacket_l(const sp<TRTPPacket>& packet,
+ const Endpoint& endpoint) {
+ EndpointState* eps = mEndpointMap.valueFor(endpoint);
+ if (!eps) {
+ // the endpoint state has disappeared, so the player that sent this
+ // packet must be dead.
+ return;
+ }
+
+ // assign the packet's sequence number
+ packet->setEpoch(eps->epoch);
+ packet->setSeqNumber(eps->trtpSeqNumber++);
+
+ // add the packet to the retry buffer
+ RetryBuffer& retry = eps->retry;
+ retry.push_back(packet);
+
+ // send the packet
+ struct sockaddr_in addr;
+ memset(&addr, 0, sizeof(addr));
+ addr.sin_family = AF_INET;
+ addr.sin_addr.s_addr = endpoint.addr;
+ addr.sin_port = htons(endpoint.port);
+
+ ssize_t result = sendto(mSocket,
+ packet->getPacket(),
+ packet->getPacketLen(),
+ 0,
+ (const struct sockaddr *) &addr,
+ sizeof(addr));
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+}
+
+void AAH_TXSender::trimRetryBuffers() {
+ Mutex::Autolock lock(mEndpointLock);
+
+ nsecs_t localTimeNow = systemTime();
+
+ Vector<Endpoint> endpointsToRemove;
+
+ for (size_t i = 0; i < mEndpointMap.size(); i++) {
+ EndpointState* eps = mEndpointMap.editValueAt(i);
+ RetryBuffer& retry = eps->retry;
+
+ while (!retry.isEmpty()) {
+ if (retry[0]->getExpireTime() < localTimeNow) {
+ retry.pop_front();
+ } else {
+ break;
+ }
+ }
+
+ if (retry.isEmpty() && eps->playerRefCount == 0) {
+ endpointsToRemove.add(mEndpointMap.keyAt(i));
+ }
+ }
+
+ // remove the state for any endpoints that are no longer in use
+ for (size_t i = 0; i < endpointsToRemove.size(); i++) {
+ Endpoint& e = endpointsToRemove.editItemAt(i);
+ ALOGD("*** %s removing endpoint addr=%08x", __PRETTY_FUNCTION__, e.addr);
+ size_t index = mEndpointMap.indexOfKey(e);
+ delete mEndpointMap.valueAt(index);
+ mEndpointMap.removeItemsAt(index);
+ }
+
+ // schedule the next trim
+ if (mEndpointMap.size()) {
+ sp<AMessage> trimMessage = new AMessage(kWhatTrimRetryBuffers,
+ handlerID());
+ trimMessage->post(kRetryTrimIntervalUs);
+ }
+}
+
+void AAH_TXSender::sendHeartbeats() {
+ Mutex::Autolock lock(mEndpointLock);
+
+ if (shouldSendHeartbeats_l()) {
+ for (size_t i = 0; i < mEndpointMap.size(); i++) {
+ EndpointState* eps = mEndpointMap.editValueAt(i);
+ const Endpoint& ep = mEndpointMap.keyAt(i);
+
+ sp<TRTPControlPacket> packet = new TRTPControlPacket();
+ packet->setCommandID(TRTPControlPacket::kCommandNop);
+
+ packet->setExpireTime(systemTime() +
+ AAH_TXPlayer::kAAHRetryKeepAroundTimeNs);
+ packet->pack();
+
+ doSendPacket_l(packet, ep);
+ }
+ }
+
+ // schedule the next heartbeat
+ if (mEndpointMap.size()) {
+ sp<AMessage> heartbeatMessage = new AMessage(kWhatSendHeartbeats,
+ handlerID());
+ heartbeatMessage->post(kHeartbeatIntervalUs);
+ }
+}
+
+bool AAH_TXSender::shouldSendHeartbeats_l() {
+ // assert(holding endpoint lock)
+ return (systemTime() < (mLastSentPacketTime + kHeartbeatTimeout));
+}
+
+// Receiver
+
+// initial 4-byte ID of a retry request packet
+const uint32_t AAH_TXSender::RetryReceiver::kRetryRequestID = 'Treq';
+
+// initial 4-byte ID of a retry NAK packet
+const uint32_t AAH_TXSender::RetryReceiver::kRetryNakID = 'Tnak';
+
+// initial 4-byte ID of a fast start request packet
+const uint32_t AAH_TXSender::RetryReceiver::kFastStartRequestID = 'Tfst';
+
+AAH_TXSender::RetryReceiver::RetryReceiver(AAH_TXSender* sender)
+ : Thread(false),
+ mSender(sender) {}
+
+ AAH_TXSender::RetryReceiver::~RetryReceiver() {
+ mWakeupEvent.clearPendingEvents();
+ }
+
+// Returns true if val is within the interval bounded inclusively by
+// start and end. Also handles the case where there is a rollover of the
+// range between start and end.
+template <typename T>
+static inline bool withinIntervalWithRollover(T val, T start, T end) {
+ return ((start <= end && val >= start && val <= end) ||
+ (start > end && (val >= start || val <= end)));
+}
+
+bool AAH_TXSender::RetryReceiver::threadLoop() {
+ struct pollfd pollFds[2];
+ pollFds[0].fd = mSender->mSocket;
+ pollFds[0].events = POLLIN;
+ pollFds[0].revents = 0;
+ pollFds[1].fd = mWakeupEvent.getWakeupHandle();
+ pollFds[1].events = POLLIN;
+ pollFds[1].revents = 0;
+
+ int pollResult = poll(pollFds, NELEM(pollFds), -1);
+ if (pollResult == -1) {
+ ALOGE("%s poll failed", __PRETTY_FUNCTION__);
+ return false;
+ }
+
+ if (exitPending()) {
+ ALOGI("*** %s exiting", __PRETTY_FUNCTION__);
+ return false;
+ }
+
+ if (pollFds[0].revents) {
+ handleRetryRequest();
+ }
+
+ return true;
+}
+
+void AAH_TXSender::RetryReceiver::handleRetryRequest() {
+ ALOGV("*** RX %s start", __PRETTY_FUNCTION__);
+
+ RetryPacket request;
+ struct sockaddr requestSrcAddr;
+ socklen_t requestSrcAddrLen = sizeof(requestSrcAddr);
+
+ ssize_t result = recvfrom(mSender->mSocket, &request, sizeof(request), 0,
+ &requestSrcAddr, &requestSrcAddrLen);
+ if (result == -1) {
+ ALOGE("%s recvfrom failed, errno=%d", __PRETTY_FUNCTION__, errno);
+ return;
+ }
+
+ if (static_cast<size_t>(result) < sizeof(RetryPacket)) {
+ ALOGW("%s short packet received", __PRETTY_FUNCTION__);
+ return;
+ }
+
+ uint32_t host_request_id = ntohl(request.id);
+ if ((host_request_id != kRetryRequestID) &&
+ (host_request_id != kFastStartRequestID)) {
+ ALOGW("%s received retry request with bogus ID (%08x)",
+ __PRETTY_FUNCTION__, host_request_id);
+ return;
+ }
+
+ Endpoint endpoint(request.endpointIP, ntohs(request.endpointPort));
+
+ Mutex::Autolock lock(mSender->mEndpointLock);
+
+ EndpointState* eps = mSender->mEndpointMap.valueFor(endpoint);
+
+ if (eps == NULL || eps->retry.isEmpty()) {
+ // we have no retry buffer or an empty retry buffer for this endpoint,
+ // so NAK the entire request
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+ return;
+ }
+
+ RetryBuffer& retry = eps->retry;
+
+ uint16_t startSeq = ntohs(request.seqStart);
+ uint16_t endSeq = ntohs(request.seqEnd);
+
+ uint16_t retryFirstSeq = retry[0]->getSeqNumber();
+ uint16_t retryLastSeq = retry[retry.size() - 1]->getSeqNumber();
+
+ // If this is a fast start, then force the start of the retry to match the
+ // start of the retransmit ring buffer (unless the end of the retransmit
+ // ring buffer is already past the point of fast start)
+ if ((host_request_id == kFastStartRequestID) &&
+ !((startSeq - retryFirstSeq) & 0x8000)) {
+ startSeq = retryFirstSeq;
+ }
+
+ int startIndex;
+ if (withinIntervalWithRollover(startSeq, retryFirstSeq, retryLastSeq)) {
+ startIndex = static_cast<uint16_t>(startSeq - retryFirstSeq);
+ } else {
+ startIndex = -1;
+ }
+
+ int endIndex;
+ if (withinIntervalWithRollover(endSeq, retryFirstSeq, retryLastSeq)) {
+ endIndex = static_cast<uint16_t>(endSeq - retryFirstSeq);
+ } else {
+ endIndex = -1;
+ }
+
+ if (startIndex == -1 && endIndex == -1) {
+ // no part of the request range is found in the retry buffer
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+ return;
+ }
+
+ if (startIndex == -1) {
+ // NAK a subrange at the front of the request range
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ nak.seqEnd = htons(retryFirstSeq - 1);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+
+ startIndex = 0;
+ } else if (endIndex == -1) {
+ // NAK a subrange at the back of the request range
+ RetryPacket nak = request;
+ nak.id = htonl(kRetryNakID);
+ nak.seqStart = htons(retryLastSeq + 1);
+ result = sendto(mSender->mSocket, &nak, sizeof(nak), 0,
+ &requestSrcAddr, requestSrcAddrLen);
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+
+ endIndex = retry.size() - 1;
+ }
+
+ // send the retry packets
+ for (int i = startIndex; i <= endIndex; i++) {
+ const sp<TRTPPacket>& replyPacket = retry[i];
+
+ result = sendto(mSender->mSocket,
+ replyPacket->getPacket(),
+ replyPacket->getPacketLen(),
+ 0,
+ &requestSrcAddr,
+ requestSrcAddrLen);
+
+ if (result == -1) {
+ ALOGW("%s sendto failed", __PRETTY_FUNCTION__);
+ }
+ }
+}
+
+// Endpoint
+
+AAH_TXSender::Endpoint::Endpoint()
+ : addr(0)
+ , port(0) { }
+
+AAH_TXSender::Endpoint::Endpoint(uint32_t a, uint16_t p)
+ : addr(a)
+ , port(p) {}
+
+bool AAH_TXSender::Endpoint::operator<(const Endpoint& other) const {
+ return ((addr < other.addr) ||
+ (addr == other.addr && port < other.port));
+}
+
+// EndpointState
+
+AAH_TXSender::EndpointState::EndpointState(uint32_t _epoch)
+ : retry(kRetryBufferCapacity)
+ , playerRefCount(1)
+ , trtpSeqNumber(0)
+ , nextProgramID(0)
+ , epoch(_epoch) { }
+
+// CircularBuffer
+
+template <typename T>
+CircularBuffer<T>::CircularBuffer(size_t capacity)
+ : mCapacity(capacity)
+ , mHead(0)
+ , mTail(0)
+ , mFillCount(0) {
+ mBuffer = new T[capacity];
+}
+
+template <typename T>
+CircularBuffer<T>::~CircularBuffer() {
+ delete [] mBuffer;
+}
+
+template <typename T>
+void CircularBuffer<T>::push_back(const T& item) {
+ if (this->isFull()) {
+ this->pop_front();
+ }
+ mBuffer[mHead] = item;
+ mHead = (mHead + 1) % mCapacity;
+ mFillCount++;
+}
+
+template <typename T>
+void CircularBuffer<T>::pop_front() {
+ CHECK(!isEmpty());
+ mBuffer[mTail] = T();
+ mTail = (mTail + 1) % mCapacity;
+ mFillCount--;
+}
+
+template <typename T>
+size_t CircularBuffer<T>::size() const {
+ return mFillCount;
+}
+
+template <typename T>
+bool CircularBuffer<T>::isFull() const {
+ return (mFillCount == mCapacity);
+}
+
+template <typename T>
+bool CircularBuffer<T>::isEmpty() const {
+ return (mFillCount == 0);
+}
+
+template <typename T>
+const T& CircularBuffer<T>::itemAt(size_t index) const {
+ CHECK(index < mFillCount);
+ return mBuffer[(mTail + index) % mCapacity];
+}
+
+template <typename T>
+const T& CircularBuffer<T>::operator[](size_t index) const {
+ return itemAt(index);
+}
+
+} // namespace android
diff --git a/media/libaah_rtp/aah_tx_sender.h b/media/libaah_rtp/aah_tx_sender.h
new file mode 100644
index 0000000..74206c4
--- /dev/null
+++ b/media/libaah_rtp/aah_tx_sender.h
@@ -0,0 +1,162 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __AAH_TX_SENDER_H__
+#define __AAH_TX_SENDER_H__
+
+#include <media/stagefright/foundation/ALooper.h>
+#include <media/stagefright/foundation/AHandlerReflector.h>
+#include <utils/RefBase.h>
+#include <utils/threads.h>
+
+#include "aah_tx_packet.h"
+#include "pipe_event.h"
+
+namespace android {
+
+template <typename T> class CircularBuffer {
+ public:
+ CircularBuffer(size_t capacity);
+ ~CircularBuffer();
+ void push_back(const T& item);;
+ void pop_front();
+ size_t size() const;
+ bool isFull() const;
+ bool isEmpty() const;
+ const T& itemAt(size_t index) const;
+ const T& operator[](size_t index) const;
+
+ private:
+ T* mBuffer;
+ size_t mCapacity;
+ size_t mHead;
+ size_t mTail;
+ size_t mFillCount;
+
+ DISALLOW_EVIL_CONSTRUCTORS(CircularBuffer);
+};
+
+class AAH_TXSender : public virtual RefBase {
+ public:
+ ~AAH_TXSender();
+
+ static sp<AAH_TXSender> GetInstance();
+
+ ALooper::handler_id handlerID() { return mReflector->id(); }
+
+ // an IP address and port
+ struct Endpoint {
+ Endpoint();
+ Endpoint(uint32_t a, uint16_t p);
+ bool operator<(const Endpoint& other) const;
+
+ uint32_t addr;
+ uint16_t port;
+ };
+
+ uint16_t registerEndpoint(const Endpoint& endpoint);
+ void unregisterEndpoint(const Endpoint& endpoint);
+
+ enum {
+ kWhatSendPacket,
+ kWhatTrimRetryBuffers,
+ kWhatSendHeartbeats,
+ };
+
+ // fields for SendPacket messages
+ static const char* kSendPacketIPAddr;
+ static const char* kSendPacketPort;
+ static const char* kSendPacketTRTPPacket;
+
+ private:
+ AAH_TXSender();
+
+ static Mutex sLock;
+ static wp<AAH_TXSender> sInstance;
+ static uint32_t sNextEpoch;
+ static bool sNextEpochValid;
+
+ static uint32_t getNextEpoch();
+
+ typedef CircularBuffer<sp<TRTPPacket> > RetryBuffer;
+
+ // state maintained on a per-endpoint basis
+ struct EndpointState {
+ EndpointState(uint32_t epoch);
+ RetryBuffer retry;
+ int playerRefCount;
+ uint16_t trtpSeqNumber;
+ uint16_t nextProgramID;
+ uint32_t epoch;
+ };
+
+ friend class AHandlerReflector<AAH_TXSender>;
+ void onMessageReceived(const sp<AMessage>& msg);
+ void onSendPacket(const sp<AMessage>& msg);
+ void doSendPacket_l(const sp<TRTPPacket>& packet,
+ const Endpoint& endpoint);
+ void trimRetryBuffers();
+ void sendHeartbeats();
+ bool shouldSendHeartbeats_l();
+
+ sp<ALooper> mLooper;
+ sp<AHandlerReflector<AAH_TXSender> > mReflector;
+
+ int mSocket;
+ nsecs_t mLastSentPacketTime;
+
+ DefaultKeyedVector<Endpoint, EndpointState*> mEndpointMap;
+ Mutex mEndpointLock;
+
+ static const int kRetryTrimIntervalUs;
+ static const int kHeartbeatIntervalUs;
+ static const int kRetryBufferCapacity;
+ static const nsecs_t kHeartbeatTimeout;
+
+ class RetryReceiver : public Thread {
+ private:
+ friend class AAH_TXSender;
+
+ RetryReceiver(AAH_TXSender* sender);
+ virtual ~RetryReceiver();
+ virtual bool threadLoop();
+ void handleRetryRequest();
+
+ static const int kMaxReceiverPacketLen;
+ static const uint32_t kRetryRequestID;
+ static const uint32_t kFastStartRequestID;
+ static const uint32_t kRetryNakID;
+
+ AAH_TXSender* mSender;
+ PipeEvent mWakeupEvent;
+ };
+
+ sp<RetryReceiver> mRetryReceiver;
+
+ DISALLOW_EVIL_CONSTRUCTORS(AAH_TXSender);
+};
+
+struct RetryPacket {
+ uint32_t id;
+ uint32_t endpointIP;
+ uint16_t endpointPort;
+ uint16_t seqStart;
+ uint16_t seqEnd;
+} __attribute__((packed));
+
+} // namespace android
+
+#endif // __AAH_TX_SENDER_H__
diff --git a/media/libaah_rtp/pipe_event.cpp b/media/libaah_rtp/pipe_event.cpp
new file mode 100644
index 0000000..b8e6960
--- /dev/null
+++ b/media/libaah_rtp/pipe_event.cpp
@@ -0,0 +1,86 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "LibAAH_RTP"
+#include <utils/Log.h>
+
+#include <errno.h>
+#include <fcntl.h>
+#include <poll.h>
+#include <unistd.h>
+
+#include "pipe_event.h"
+
+namespace android {
+
+PipeEvent::PipeEvent() {
+ pipe_[0] = -1;
+ pipe_[1] = -1;
+
+ // Create the pipe.
+ if (pipe(pipe_) >= 0) {
+ // Set non-blocking mode on the read side of the pipe so we can
+ // easily drain it whenever we wakeup.
+ fcntl(pipe_[0], F_SETFL, O_NONBLOCK);
+ } else {
+ ALOGE("Failed to create pipe event %d %d %d",
+ pipe_[0], pipe_[1], errno);
+ pipe_[0] = -1;
+ pipe_[1] = -1;
+ }
+}
+
+PipeEvent::~PipeEvent() {
+ if (pipe_[0] >= 0) {
+ close(pipe_[0]);
+ }
+
+ if (pipe_[1] >= 0) {
+ close(pipe_[1]);
+ }
+}
+
+void PipeEvent::clearPendingEvents() {
+ char drain_buffer[16];
+ while (read(pipe_[0], drain_buffer, sizeof(drain_buffer)) > 0) {
+ // No body.
+ }
+}
+
+bool PipeEvent::wait(int timeout) {
+ struct pollfd wait_fd;
+
+ wait_fd.fd = getWakeupHandle();
+ wait_fd.events = POLLIN;
+ wait_fd.revents = 0;
+
+ int res = poll(&wait_fd, 1, timeout);
+
+ if (res < 0) {
+ ALOGE("Wait error in PipeEvent; sleeping to prevent overload!");
+ usleep(1000);
+ }
+
+ return (res > 0);
+}
+
+void PipeEvent::setEvent() {
+ char foo = 'q';
+ write(pipe_[1], &foo, 1);
+}
+
+} // namespace android
+
diff --git a/media/libaah_rtp/pipe_event.h b/media/libaah_rtp/pipe_event.h
new file mode 100644
index 0000000..e53b0fd
--- /dev/null
+++ b/media/libaah_rtp/pipe_event.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef __PIPE_EVENT_H__
+#define __PIPE_EVENT_H__
+
+#include <media/stagefright/foundation/ABase.h>
+
+namespace android {
+
+class PipeEvent {
+ public:
+ PipeEvent();
+ ~PipeEvent();
+
+ bool initCheck() const {
+ return ((pipe_[0] >= 0) && (pipe_[1] >= 0));
+ }
+
+ int getWakeupHandle() const { return pipe_[0]; }
+
+ // block until the event fires; returns true if the event fired and false if
+ // the wait timed out. Timeout is expressed in milliseconds; negative
+ // values mean wait forever.
+ bool wait(int timeout = -1);
+
+ void clearPendingEvents();
+ void setEvent();
+
+ private:
+ int pipe_[2];
+
+ DISALLOW_EVIL_CONSTRUCTORS(PipeEvent);
+};
+
+} // namespace android
+
+#endif // __PIPE_EVENT_H__
diff --git a/media/libeffects/factory/EffectsFactory.c b/media/libeffects/factory/EffectsFactory.c
index 9f6599f..59cd9e3 100644
--- a/media/libeffects/factory/EffectsFactory.c
+++ b/media/libeffects/factory/EffectsFactory.c
@@ -53,8 +53,8 @@
static lib_entry_t *getLibrary(const char *path);
static void resetEffectEnumeration();
static uint32_t updateNumEffects();
-static int findEffect(effect_uuid_t *type,
- effect_uuid_t *uuid,
+static int findEffect(const effect_uuid_t *type,
+ const effect_uuid_t *uuid,
lib_entry_t **lib,
effect_descriptor_t **desc);
static void dumpEffectDescriptor(effect_descriptor_t *desc, char *str, size_t len);
@@ -236,7 +236,7 @@
return ret;
}
-int EffectGetDescriptor(effect_uuid_t *uuid, effect_descriptor_t *pDescriptor)
+int EffectGetDescriptor(const effect_uuid_t *uuid, effect_descriptor_t *pDescriptor)
{
lib_entry_t *l = NULL;
effect_descriptor_t *d = NULL;
@@ -257,7 +257,7 @@
return ret;
}
-int EffectCreate(effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle)
+int EffectCreate(const effect_uuid_t *uuid, int32_t sessionId, int32_t ioId, effect_handle_t *pHandle)
{
list_elem_t *e = gLibraryList;
lib_entry_t *l = NULL;
@@ -372,7 +372,7 @@
return ret;
}
-int EffectIsNullUuid(effect_uuid_t *uuid)
+int EffectIsNullUuid(const effect_uuid_t *uuid)
{
if (memcmp(uuid, EFFECT_UUID_NULL, sizeof(effect_uuid_t))) {
return 0;
@@ -628,8 +628,8 @@
return cnt;
}
-int findEffect(effect_uuid_t *type,
- effect_uuid_t *uuid,
+int findEffect(const effect_uuid_t *type,
+ const effect_uuid_t *uuid,
lib_entry_t **lib,
effect_descriptor_t **desc)
{
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 62be78c..3714283 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -133,7 +133,8 @@
int LvmEffect_enable (EffectContext *pContext);
int LvmEffect_disable (EffectContext *pContext);
void LvmEffect_free (EffectContext *pContext);
-int Effect_configure (EffectContext *pContext, effect_config_t *pConfig);
+int Effect_setConfig (EffectContext *pContext, effect_config_t *pConfig);
+void Effect_getConfig (EffectContext *pContext, effect_config_t *pConfig);
int BassBoost_setParameter (EffectContext *pContext, void *pParam, void *pValue);
int BassBoost_getParameter (EffectContext *pContext,
void *pParam,
@@ -194,7 +195,7 @@
return 0;
} /* end EffectQueryEffect */
-extern "C" int EffectCreate(effect_uuid_t *uuid,
+extern "C" int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle){
@@ -470,7 +471,7 @@
} /* end EffectRelease */
-extern "C" int EffectGetDescriptor(effect_uuid_t *uuid,
+extern "C" int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
const effect_descriptor_t *desc = NULL;
@@ -936,7 +937,7 @@
} /* end LvmEffect_free */
//----------------------------------------------------------------------------
-// Effect_configure()
+// Effect_setConfig()
//----------------------------------------------------------------------------
// Purpose: Set input and output audio configuration.
//
@@ -949,9 +950,9 @@
//
//----------------------------------------------------------------------------
-int Effect_configure(EffectContext *pContext, effect_config_t *pConfig){
+int Effect_setConfig(EffectContext *pContext, effect_config_t *pConfig){
LVM_Fs_en SampleRate;
- //ALOGV("\tEffect_configure start");
+ //ALOGV("\tEffect_setConfig start");
CHECK_ARG(pContext != NULL);
CHECK_ARG(pConfig != NULL);
@@ -992,7 +993,7 @@
pContext->pBundledContext->SamplesPerSecond = 48000*2; // 2 secs Stereo
break;
default:
- ALOGV("\tEffect_Configure invalid sampling rate %d", pConfig->inputCfg.samplingRate);
+ ALOGV("\tEffect_setConfig invalid sampling rate %d", pConfig->inputCfg.samplingRate);
return -EINVAL;
}
@@ -1001,28 +1002,47 @@
LVM_ControlParams_t ActiveParams;
LVM_ReturnStatus_en LvmStatus = LVM_SUCCESS;
- ALOGV("\tEffect_configure change sampling rate to %d", SampleRate);
+ ALOGV("\tEffect_setConfig change sampling rate to %d", SampleRate);
/* Get the current settings */
LvmStatus = LVM_GetControlParameters(pContext->pBundledContext->hInstance,
&ActiveParams);
- LVM_ERROR_CHECK(LvmStatus, "LVM_GetControlParameters", "Effect_configure")
+ LVM_ERROR_CHECK(LvmStatus, "LVM_GetControlParameters", "Effect_setConfig")
if(LvmStatus != LVM_SUCCESS) return -EINVAL;
LvmStatus = LVM_SetControlParameters(pContext->pBundledContext->hInstance, &ActiveParams);
- LVM_ERROR_CHECK(LvmStatus, "LVM_SetControlParameters", "Effect_configure")
- ALOGV("\tEffect_configure Succesfully called LVM_SetControlParameters\n");
+ LVM_ERROR_CHECK(LvmStatus, "LVM_SetControlParameters", "Effect_setConfig")
+ ALOGV("\tEffect_setConfig Succesfully called LVM_SetControlParameters\n");
pContext->pBundledContext->SampleRate = SampleRate;
}else{
- //ALOGV("\tEffect_configure keep sampling rate at %d", SampleRate);
+ //ALOGV("\tEffect_setConfig keep sampling rate at %d", SampleRate);
}
- //ALOGV("\tEffect_configure End....");
+ //ALOGV("\tEffect_setConfig End....");
return 0;
-} /* end Effect_configure */
+} /* end Effect_setConfig */
+
+//----------------------------------------------------------------------------
+// Effect_getConfig()
+//----------------------------------------------------------------------------
+// Purpose: Get input and output audio configuration.
+//
+// Inputs:
+// pContext: effect engine context
+// pConfig: pointer to effect_config_t structure holding input and output
+// configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+void Effect_getConfig(EffectContext *pContext, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pContext->config, sizeof(effect_config_t));
+} /* end Effect_getConfig */
//----------------------------------------------------------------------------
// BassGetStrength()
@@ -2778,23 +2798,34 @@
}
break;
- case EFFECT_CMD_CONFIGURE:
- //ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_CONFIGURE start");
+ case EFFECT_CMD_SET_CONFIG:
+ //ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_SET_CONFIG start");
if (pCmdData == NULL||
cmdSize != sizeof(effect_config_t)||
pReplyData == NULL||
*replySize != sizeof(int)){
ALOGV("\tLVM_ERROR : Effect_command cmdCode Case: "
- "EFFECT_CMD_CONFIGURE: ERROR");
+ "EFFECT_CMD_SET_CONFIG: ERROR");
return -EINVAL;
}
- *(int *) pReplyData = android::Effect_configure(pContext, (effect_config_t *) pCmdData);
- //ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_CONFIGURE end");
+ *(int *) pReplyData = android::Effect_setConfig(pContext, (effect_config_t *) pCmdData);
+ //ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_SET_CONFIG end");
+ break;
+
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)) {
+ ALOGV("\tLVM_ERROR : Effect_command cmdCode Case: "
+ "EFFECT_CMD_GET_CONFIG: ERROR");
+ return -EINVAL;
+ }
+
+ android::Effect_getConfig(pContext, (effect_config_t *)pReplyData);
break;
case EFFECT_CMD_RESET:
//ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_RESET start");
- android::Effect_configure(pContext, &pContext->config);
+ android::Effect_setConfig(pContext, &pContext->config);
//ALOGV("\tEffect_command cmdCode Case: EFFECT_CMD_RESET end");
break;
@@ -3078,20 +3109,20 @@
if (pContext->pBundledContext->bBassEnabled == LVM_TRUE) {
ALOGV("\tEFFECT_CMD_SET_DEVICE disable LVM_BASS_BOOST %d",
- *(int32_t *)pCmdData);
+ *(int32_t *)pCmdData);
android::LvmEffect_disable(pContext);
}
pContext->pBundledContext->bBassTempDisabled = LVM_TRUE;
} else {
ALOGV("\tEFFECT_CMD_SET_DEVICE device is valid for LVM_BASS_BOOST %d",
- *(int32_t *)pCmdData);
+ *(int32_t *)pCmdData);
// If a device supports bassboost and the effect has been temporarily disabled
// previously then re-enable it
if (pContext->pBundledContext->bBassEnabled == LVM_TRUE) {
ALOGV("\tEFFECT_CMD_SET_DEVICE re-enable LVM_BASS_BOOST %d",
- *(int32_t *)pCmdData);
+ *(int32_t *)pCmdData);
android::LvmEffect_enable(pContext);
}
pContext->pBundledContext->bBassTempDisabled = LVM_FALSE;
diff --git a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
index 1825aab..358357e 100755
--- a/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
+++ b/media/libeffects/lvm/wrapper/Reverb/EffectReverb.cpp
@@ -175,7 +175,8 @@
//--- local function prototypes
int Reverb_init (ReverbContext *pContext);
void Reverb_free (ReverbContext *pContext);
-int Reverb_configure (ReverbContext *pContext, effect_config_t *pConfig);
+int Reverb_setConfig (ReverbContext *pContext, effect_config_t *pConfig);
+void Reverb_getConfig (ReverbContext *pContext, effect_config_t *pConfig);
int Reverb_setParameter (ReverbContext *pContext, void *pParam, void *pValue);
int Reverb_getParameter (ReverbContext *pContext,
void *pParam,
@@ -209,7 +210,7 @@
return 0;
} /* end EffectQueryEffect */
-extern "C" int EffectCreate(effect_uuid_t *uuid,
+extern "C" int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle){
@@ -316,7 +317,7 @@
return 0;
} /* end EffectRelease */
-extern "C" int EffectGetDescriptor(effect_uuid_t *uuid,
+extern "C" int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
int i;
int length = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
@@ -609,7 +610,7 @@
} /* end Reverb_free */
//----------------------------------------------------------------------------
-// Reverb_configure()
+// Reverb_setConfig()
//----------------------------------------------------------------------------
// Purpose: Set input and output audio configuration.
//
@@ -622,9 +623,9 @@
//
//----------------------------------------------------------------------------
-int Reverb_configure(ReverbContext *pContext, effect_config_t *pConfig){
+int Reverb_setConfig(ReverbContext *pContext, effect_config_t *pConfig){
LVM_Fs_en SampleRate;
- //ALOGV("\tReverb_configure start");
+ //ALOGV("\tReverb_setConfig start");
CHECK_ARG(pContext != NULL);
CHECK_ARG(pConfig != NULL);
@@ -642,7 +643,7 @@
return -EINVAL;
}
- //ALOGV("\tReverb_configure calling memcpy");
+ //ALOGV("\tReverb_setConfig calling memcpy");
memcpy(&pContext->config, pConfig, sizeof(effect_config_t));
@@ -666,7 +667,7 @@
SampleRate = LVM_FS_48000;
break;
default:
- ALOGV("\rReverb_Configure invalid sampling rate %d", pConfig->inputCfg.samplingRate);
+ ALOGV("\rReverb_setConfig invalid sampling rate %d", pConfig->inputCfg.samplingRate);
return -EINVAL;
}
@@ -675,28 +676,46 @@
LVREV_ControlParams_st ActiveParams;
LVREV_ReturnStatus_en LvmStatus = LVREV_SUCCESS;
- //ALOGV("\tReverb_configure change sampling rate to %d", SampleRate);
+ //ALOGV("\tReverb_setConfig change sampling rate to %d", SampleRate);
/* Get the current settings */
LvmStatus = LVREV_GetControlParameters(pContext->hInstance,
&ActiveParams);
- LVM_ERROR_CHECK(LvmStatus, "LVREV_GetControlParameters", "Reverb_configure")
+ LVM_ERROR_CHECK(LvmStatus, "LVREV_GetControlParameters", "Reverb_setConfig")
if(LvmStatus != LVREV_SUCCESS) return -EINVAL;
LvmStatus = LVREV_SetControlParameters(pContext->hInstance, &ActiveParams);
- LVM_ERROR_CHECK(LvmStatus, "LVREV_SetControlParameters", "Reverb_configure")
- //ALOGV("\tReverb_configure Succesfully called LVREV_SetControlParameters\n");
+ LVM_ERROR_CHECK(LvmStatus, "LVREV_SetControlParameters", "Reverb_setConfig")
+ //ALOGV("\tReverb_setConfig Succesfully called LVREV_SetControlParameters\n");
}else{
- //ALOGV("\tReverb_configure keep sampling rate at %d", SampleRate);
+ //ALOGV("\tReverb_setConfig keep sampling rate at %d", SampleRate);
}
- //ALOGV("\tReverb_configure End");
+ //ALOGV("\tReverb_setConfig End");
return 0;
-} /* end Reverb_configure */
+} /* end Reverb_setConfig */
+//----------------------------------------------------------------------------
+// Reverb_getConfig()
+//----------------------------------------------------------------------------
+// Purpose: Get input and output audio configuration.
+//
+// Inputs:
+// pContext: effect engine context
+// pConfig: pointer to effect_config_t structure holding input and output
+// configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+void Reverb_getConfig(ReverbContext *pContext, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pContext->config, sizeof(effect_config_t));
+} /* end Reverb_getConfig */
//----------------------------------------------------------------------------
// Reverb_init()
@@ -1924,24 +1943,36 @@
*(int *) pReplyData = 0;
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
//ALOGV("\tReverb_command cmdCode Case: "
- // "EFFECT_CMD_CONFIGURE start");
- if (pCmdData == NULL||
- cmdSize != sizeof(effect_config_t)||
- pReplyData == NULL||
- *replySize != sizeof(int)){
+ // "EFFECT_CMD_SET_CONFIG start");
+ if (pCmdData == NULL ||
+ cmdSize != sizeof(effect_config_t) ||
+ pReplyData == NULL ||
+ *replySize != sizeof(int)) {
ALOGV("\tLVM_ERROR : Reverb_command cmdCode Case: "
- "EFFECT_CMD_CONFIGURE: ERROR");
+ "EFFECT_CMD_SET_CONFIG: ERROR");
return -EINVAL;
}
- *(int *) pReplyData = Reverb_configure(pContext, (effect_config_t *) pCmdData);
+ *(int *) pReplyData = android::Reverb_setConfig(pContext,
+ (effect_config_t *) pCmdData);
+ break;
+
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)) {
+ ALOGV("\tLVM_ERROR : Reverb_command cmdCode Case: "
+ "EFFECT_CMD_GET_CONFIG: ERROR");
+ return -EINVAL;
+ }
+
+ android::Reverb_getConfig(pContext, (effect_config_t *)pReplyData);
break;
case EFFECT_CMD_RESET:
//ALOGV("\tReverb_command cmdCode Case: "
// "EFFECT_CMD_RESET start");
- Reverb_configure(pContext, &pContext->config);
+ Reverb_setConfig(pContext, &pContext->config);
break;
case EFFECT_CMD_GET_PARAM:{
diff --git a/media/libeffects/preprocessing/Android.mk b/media/libeffects/preprocessing/Android.mk
index 77d40b6..7f7c7e1 100755
--- a/media/libeffects/preprocessing/Android.mk
+++ b/media/libeffects/preprocessing/Android.mk
@@ -13,7 +13,7 @@
LOCAL_C_INCLUDES += \
external/webrtc/src \
external/webrtc/src/modules/interface \
- external/webrtc/src/modules/audio_processing/main/interface \
+ external/webrtc/src/modules/audio_processing/interface \
system/media/audio_effects/include
LOCAL_C_INCLUDES += $(call include-path-for, speex)
diff --git a/media/libeffects/preprocessing/PreProcessing.cpp b/media/libeffects/preprocessing/PreProcessing.cpp
index 6267d1d..098a1a2 100755
--- a/media/libeffects/preprocessing/PreProcessing.cpp
+++ b/media/libeffects/preprocessing/PreProcessing.cpp
@@ -24,8 +24,8 @@
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_agc.h>
#include <audio_effects/effect_ns.h>
-#include "modules/interface/module_common_types.h"
-#include "modules/audio_processing/main/interface/audio_processing.h"
+#include <module_common_types.h>
+#include <audio_processing.h>
#include "speex/speex_resampler.h"
@@ -220,8 +220,8 @@
// Automatic Gain Control (AGC)
//------------------------------------------------------------------------------
-static const int kAgcDefaultTargetLevel = 0;
-static const int kAgcDefaultCompGain = 90;
+static const int kAgcDefaultTargetLevel = 3;
+static const int kAgcDefaultCompGain = 9;
static const bool kAgcDefaultLimiter = true;
int AgcInit (preproc_effect_t *effect)
@@ -940,6 +940,19 @@
return 0;
}
+void Session_GetConfig(preproc_session_t *session, effect_config_t *config)
+{
+ memset(config, 0, sizeof(effect_config_t));
+ config->inputCfg.samplingRate = config->outputCfg.samplingRate = session->samplingRate;
+ config->inputCfg.format = config->outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ config->inputCfg.channels = session->inChannelCount == 1 ?
+ AUDIO_CHANNEL_IN_MONO : AUDIO_CHANNEL_IN_STEREO;
+ config->outputCfg.channels = session->outChannelCount == 1 ?
+ AUDIO_CHANNEL_IN_MONO : AUDIO_CHANNEL_IN_STEREO;
+ config->inputCfg.mask = config->outputCfg.mask =
+ (EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT);
+}
+
int Session_SetReverseConfig(preproc_session_t *session, effect_config_t *config)
{
if (config->inputCfg.samplingRate != config->outputCfg.samplingRate ||
@@ -969,6 +982,17 @@
return 0;
}
+void Session_GetReverseConfig(preproc_session_t *session, effect_config_t *config)
+{
+ memset(config, 0, sizeof(effect_config_t));
+ config->inputCfg.samplingRate = config->outputCfg.samplingRate = session->samplingRate;
+ config->inputCfg.format = config->outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+ config->inputCfg.channels = config->outputCfg.channels =
+ session->revChannelCount == 1 ? AUDIO_CHANNEL_IN_MONO : AUDIO_CHANNEL_IN_STEREO;
+ config->inputCfg.mask = config->outputCfg.mask =
+ (EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT);
+}
+
void Session_SetProcEnabled(preproc_session_t *session, uint32_t procId, bool enabled)
{
if (enabled) {
@@ -1048,7 +1072,7 @@
return sInitStatus;
}
-const effect_descriptor_t *PreProc_GetDescriptor(effect_uuid_t *uuid)
+const effect_descriptor_t *PreProc_GetDescriptor(const effect_uuid_t *uuid)
{
size_t i;
for (i = 0; i < PREPROC_NUM_EFFECTS; i++) {
@@ -1250,13 +1274,13 @@
*(int *)pReplyData = 0;
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
if (pCmdData == NULL||
cmdSize != sizeof(effect_config_t)||
pReplyData == NULL||
*replySize != sizeof(int)){
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_CONFIGURE: ERROR");
+ "EFFECT_CMD_SET_CONFIG: ERROR");
return -EINVAL;
}
*(int *)pReplyData = Session_SetConfig(effect->session, (effect_config_t *)pCmdData);
@@ -1266,13 +1290,24 @@
*(int *)pReplyData = Effect_SetState(effect, PREPROC_EFFECT_STATE_CONFIG);
break;
- case EFFECT_CMD_CONFIGURE_REVERSE:
- if (pCmdData == NULL||
- cmdSize != sizeof(effect_config_t)||
- pReplyData == NULL||
- *replySize != sizeof(int)){
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)) {
+ ALOGV("\tLVM_ERROR : PreProcessingFx_Command cmdCode Case: "
+ "EFFECT_CMD_GET_CONFIG: ERROR");
+ return -EINVAL;
+ }
+
+ Session_GetConfig(effect->session, (effect_config_t *)pReplyData);
+ break;
+
+ case EFFECT_CMD_SET_CONFIG_REVERSE:
+ if (pCmdData == NULL ||
+ cmdSize != sizeof(effect_config_t) ||
+ pReplyData == NULL ||
+ *replySize != sizeof(int)) {
ALOGV("PreProcessingFx_Command cmdCode Case: "
- "EFFECT_CMD_CONFIGURE_REVERSE: ERROR");
+ "EFFECT_CMD_SET_CONFIG_REVERSE: ERROR");
return -EINVAL;
}
*(int *)pReplyData = Session_SetReverseConfig(effect->session,
@@ -1282,6 +1317,16 @@
}
break;
+ case EFFECT_CMD_GET_CONFIG_REVERSE:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)){
+ ALOGV("PreProcessingFx_Command cmdCode Case: "
+ "EFFECT_CMD_GET_CONFIG_REVERSE: ERROR");
+ return -EINVAL;
+ }
+ Session_GetReverseConfig(effect->session, (effect_config_t *)pCmdData);
+ break;
+
case EFFECT_CMD_RESET:
if (effect->ops->reset) {
effect->ops->reset(effect);
@@ -1523,7 +1568,7 @@
return 0;
}
-int PreProcessingLib_Create(effect_uuid_t *uuid,
+int PreProcessingLib_Create(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pInterface)
@@ -1575,7 +1620,7 @@
return Session_ReleaseEffect(fx->session, fx);
}
-int PreProcessingLib_GetDescriptor(effect_uuid_t *uuid,
+int PreProcessingLib_GetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
if (pDescriptor == NULL || uuid == NULL){
diff --git a/media/libeffects/testlibs/AudioBiquadFilter.cpp b/media/libeffects/testlibs/AudioBiquadFilter.cpp
index 72917a3..16dd1c5 100644
--- a/media/libeffects/testlibs/AudioBiquadFilter.cpp
+++ b/media/libeffects/testlibs/AudioBiquadFilter.cpp
@@ -17,12 +17,10 @@
#include <string.h>
#include <assert.h>
+#include <cutils/compiler.h>
#include "AudioBiquadFilter.h"
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
-
namespace android {
const audio_coef_t AudioBiquadFilter::IDENTITY_COEFS[AudioBiquadFilter::NUM_COEFS] = { AUDIO_COEF_ONE, 0, 0, 0, 0 };
@@ -55,7 +53,7 @@
void AudioBiquadFilter::setCoefs(const audio_coef_t coefs[NUM_COEFS], bool immediate) {
memcpy(mTargetCoefs, coefs, sizeof(mTargetCoefs));
if (mState & STATE_ENABLED_MASK) {
- if (UNLIKELY(immediate)) {
+ if (CC_UNLIKELY(immediate)) {
memcpy(mCoefs, coefs, sizeof(mCoefs));
setState(STATE_NORMAL);
} else {
@@ -70,7 +68,7 @@
}
void AudioBiquadFilter::enable(bool immediate) {
- if (UNLIKELY(immediate)) {
+ if (CC_UNLIKELY(immediate)) {
memcpy(mCoefs, mTargetCoefs, sizeof(mCoefs));
setState(STATE_NORMAL);
} else {
@@ -79,7 +77,7 @@
}
void AudioBiquadFilter::disable(bool immediate) {
- if (UNLIKELY(immediate)) {
+ if (CC_UNLIKELY(immediate)) {
memcpy(mCoefs, IDENTITY_COEFS, sizeof(mCoefs));
setState(STATE_BYPASS);
} else {
@@ -142,7 +140,7 @@
audio_sample_t * out,
int frameCount) {
// The common case is in-place processing, because this is what the EQ does.
- if (UNLIKELY(in != out)) {
+ if (CC_UNLIKELY(in != out)) {
memcpy(out, in, frameCount * mNumChannels * sizeof(audio_sample_t));
}
}
diff --git a/media/libeffects/testlibs/AudioCoefInterpolator.cpp b/media/libeffects/testlibs/AudioCoefInterpolator.cpp
index 039ab9f..6b56922 100644
--- a/media/libeffects/testlibs/AudioCoefInterpolator.cpp
+++ b/media/libeffects/testlibs/AudioCoefInterpolator.cpp
@@ -16,10 +16,10 @@
*/
#include <string.h>
-#include "AudioCoefInterpolator.h"
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
+
+#include "AudioCoefInterpolator.h"
namespace android {
@@ -44,9 +44,9 @@
size_t index = 0;
size_t dim = mNumInDims;
while (dim-- > 0) {
- if (UNLIKELY(intCoord[dim] < 0)) {
+ if (CC_UNLIKELY(intCoord[dim] < 0)) {
fracCoord[dim] = 0;
- } else if (UNLIKELY(intCoord[dim] >= (int)mInDims[dim] - 1)) {
+ } else if (CC_UNLIKELY(intCoord[dim] >= (int)mInDims[dim] - 1)) {
fracCoord[dim] = 0;
index += mInDimOffsets[dim] * (mInDims[dim] - 1);
} else {
@@ -63,7 +63,7 @@
memcpy(out, mTable + index, mNumOutDims * sizeof(audio_coef_t));
} else {
getCoefRecurse(index, fracCoord, out, dim + 1);
- if (LIKELY(fracCoord != 0)) {
+ if (CC_LIKELY(fracCoord != 0)) {
audio_coef_t tempCoef[MAX_OUT_DIMS];
getCoefRecurse(index + mInDimOffsets[dim], fracCoord, tempCoef,
dim + 1);
diff --git a/media/libeffects/testlibs/AudioCommon.h b/media/libeffects/testlibs/AudioCommon.h
index 444f93a..e8080dc 100644
--- a/media/libeffects/testlibs/AudioCommon.h
+++ b/media/libeffects/testlibs/AudioCommon.h
@@ -20,6 +20,7 @@
#include <stdint.h>
#include <stddef.h>
+#include <cutils/compiler.h>
namespace android {
@@ -76,9 +77,9 @@
// Convert a audio_sample_t sample to S15 (with clipping)
inline int16_t audio_sample_t_to_s15_clip(audio_sample_t sample) {
// TODO: optimize for targets supporting this as an atomic operation.
- if (__builtin_expect(sample >= (0x7FFF << 9), 0)) {
+ if (CC_UNLIKELY(sample >= (0x7FFF << 9))) {
return 0x7FFF;
- } else if (__builtin_expect(sample <= -(0x8000 << 9), 0)) {
+ } else if (CC_UNLIKELY(sample <= -(0x8000 << 9))) {
return 0x8000;
} else {
return audio_sample_t_to_s15(sample);
diff --git a/media/libeffects/testlibs/AudioPeakingFilter.cpp b/media/libeffects/testlibs/AudioPeakingFilter.cpp
index 60fefe6..99323ac 100644
--- a/media/libeffects/testlibs/AudioPeakingFilter.cpp
+++ b/media/libeffects/testlibs/AudioPeakingFilter.cpp
@@ -21,9 +21,7 @@
#include <new>
#include <assert.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// Format of the coefficient table:
@@ -66,12 +64,12 @@
void AudioPeakingFilter::setFrequency(uint32_t millihertz) {
mNominalFrequency = millihertz;
- if (UNLIKELY(millihertz > mNiquistFreq / 2)) {
+ if (CC_UNLIKELY(millihertz > mNiquistFreq / 2)) {
millihertz = mNiquistFreq / 2;
}
uint32_t normFreq = static_cast<uint32_t>(
(static_cast<uint64_t>(millihertz) * mFrequencyFactor) >> 10);
- if (LIKELY(normFreq > (1 << 23))) {
+ if (CC_LIKELY(normFreq > (1 << 23))) {
mFrequency = (Effects_log2(normFreq) - ((32-9) << 15)) << (FREQ_PRECISION_BITS - 15);
} else {
mFrequency = 0;
@@ -107,11 +105,11 @@
int32_t halfBW = (((mBandwidth + 1) / 2) << 15) / 1200;
low = static_cast<uint32_t>((static_cast<uint64_t>(mNominalFrequency) * Effects_exp2(-halfBW + (16 << 15))) >> 16);
- if (UNLIKELY(halfBW >= (16 << 15))) {
+ if (CC_UNLIKELY(halfBW >= (16 << 15))) {
high = mNiquistFreq;
} else {
high = static_cast<uint32_t>((static_cast<uint64_t>(mNominalFrequency) * Effects_exp2(halfBW + (16 << 15))) >> 16);
- if (UNLIKELY(high > mNiquistFreq)) {
+ if (CC_UNLIKELY(high > mNiquistFreq)) {
high = mNiquistFreq;
}
}
diff --git a/media/libeffects/testlibs/AudioShelvingFilter.cpp b/media/libeffects/testlibs/AudioShelvingFilter.cpp
index b8650ba..e031287 100644
--- a/media/libeffects/testlibs/AudioShelvingFilter.cpp
+++ b/media/libeffects/testlibs/AudioShelvingFilter.cpp
@@ -21,9 +21,7 @@
#include <new>
#include <assert.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// Format of the coefficient tables:
@@ -71,13 +69,13 @@
void AudioShelvingFilter::setFrequency(uint32_t millihertz) {
mNominalFrequency = millihertz;
- if (UNLIKELY(millihertz > mNiquistFreq / 2)) {
+ if (CC_UNLIKELY(millihertz > mNiquistFreq / 2)) {
millihertz = mNiquistFreq / 2;
}
uint32_t normFreq = static_cast<uint32_t>(
(static_cast<uint64_t>(millihertz) * mFrequencyFactor) >> 10);
uint32_t log2minFreq = (mType == kLowShelf ? (32-10) : (32-2));
- if (LIKELY(normFreq > (1U << log2minFreq))) {
+ if (CC_LIKELY(normFreq > (1U << log2minFreq))) {
mFrequency = (Effects_log2(normFreq) - (log2minFreq << 15)) << (FREQ_PRECISION_BITS - 15);
} else {
mFrequency = 0;
diff --git a/media/libeffects/testlibs/EffectEqualizer.cpp b/media/libeffects/testlibs/EffectEqualizer.cpp
index 43f34de..35a4a61 100644
--- a/media/libeffects/testlibs/EffectEqualizer.cpp
+++ b/media/libeffects/testlibs/EffectEqualizer.cpp
@@ -114,7 +114,7 @@
//--- local function prototypes
int Equalizer_init(EqualizerContext *pContext);
-int Equalizer_configure(EqualizerContext *pContext, effect_config_t *pConfig);
+int Equalizer_setConfig(EqualizerContext *pContext, effect_config_t *pConfig);
int Equalizer_getParameter(AudioEqualizer * pEqualizer, int32_t *pParam, size_t *pValueSize, void *pValue);
int Equalizer_setParameter(AudioEqualizer * pEqualizer, int32_t *pParam, void *pValue);
@@ -140,7 +140,7 @@
return 0;
} /* end EffectQueryNext */
-extern "C" int EffectCreate(effect_uuid_t *uuid,
+extern "C" int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle) {
@@ -195,7 +195,7 @@
return 0;
} /* end EffectRelease */
-extern "C" int EffectGetDescriptor(effect_uuid_t *uuid,
+extern "C" int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
if (pDescriptor == NULL || uuid == NULL){
@@ -224,7 +224,7 @@
}
//----------------------------------------------------------------------------
-// Equalizer_configure()
+// Equalizer_setConfig()
//----------------------------------------------------------------------------
// Purpose: Set input and output audio configuration.
//
@@ -237,9 +237,9 @@
//
//----------------------------------------------------------------------------
-int Equalizer_configure(EqualizerContext *pContext, effect_config_t *pConfig)
+int Equalizer_setConfig(EqualizerContext *pContext, effect_config_t *pConfig)
{
- ALOGV("Equalizer_configure start");
+ ALOGV("Equalizer_setConfig start");
CHECK_ARG(pContext != NULL);
CHECK_ARG(pConfig != NULL);
@@ -272,7 +272,26 @@
pConfig->outputCfg.accessMode);
return 0;
-} // end Equalizer_configure
+} // end Equalizer_setConfig
+
+//----------------------------------------------------------------------------
+// Equalizer_getConfig()
+//----------------------------------------------------------------------------
+// Purpose: Get input and output audio configuration.
+//
+// Inputs:
+// pContext: effect engine context
+// pConfig: pointer to effect_config_t structure holding input and output
+// configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+void Equalizer_getConfig(EqualizerContext *pContext, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pContext->config, sizeof(effect_config_t));
+} // end Equalizer_getConfig
//----------------------------------------------------------------------------
@@ -332,7 +351,7 @@
pContext->pEqualizer->enable(true);
- Equalizer_configure(pContext, &pContext->config);
+ Equalizer_setConfig(pContext, &pContext->config);
return 0;
} // end Equalizer_init
@@ -643,16 +662,22 @@
}
*(int *) pReplyData = Equalizer_init(pContext);
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
if (pCmdData == NULL || cmdSize != sizeof(effect_config_t)
|| pReplyData == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
- *(int *) pReplyData = Equalizer_configure(pContext,
+ *(int *) pReplyData = Equalizer_setConfig(pContext,
(effect_config_t *) pCmdData);
break;
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL || *replySize != sizeof(effect_config_t)) {
+ return -EINVAL;
+ }
+ Equalizer_getConfig(pContext, (effect_config_t *) pCmdData);
+ break;
case EFFECT_CMD_RESET:
- Equalizer_configure(pContext, &pContext->config);
+ Equalizer_setConfig(pContext, &pContext->config);
break;
case EFFECT_CMD_GET_PARAM: {
if (pCmdData == NULL || cmdSize < (int)(sizeof(effect_param_t) + sizeof(int32_t)) ||
diff --git a/media/libeffects/testlibs/EffectReverb.c b/media/libeffects/testlibs/EffectReverb.c
index d22868a..8351712 100644
--- a/media/libeffects/testlibs/EffectReverb.c
+++ b/media/libeffects/testlibs/EffectReverb.c
@@ -111,7 +111,7 @@
return 0;
}
-int EffectCreate(effect_uuid_t *uuid,
+int EffectCreate(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle) {
@@ -182,7 +182,7 @@
return 0;
}
-int EffectGetDescriptor(effect_uuid_t *uuid,
+int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
int i;
int length = sizeof(gDescriptors) / sizeof(const effect_descriptor_t *);
@@ -318,14 +318,20 @@
pRvbModule->context.mState = REVERB_STATE_INITIALIZED;
}
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
if (pCmdData == NULL || cmdSize != sizeof(effect_config_t)
|| pReplyData == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
- *(int *) pReplyData = Reverb_Configure(pRvbModule,
+ *(int *) pReplyData = Reverb_setConfig(pRvbModule,
(effect_config_t *)pCmdData, false);
break;
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL || *replySize != sizeof(effect_config_t)) {
+ return -EINVAL;
+ }
+ Reverb_getConfig(pRvbModule, (effect_config_t *) pCmdData);
+ break;
case EFFECT_CMD_RESET:
Reverb_Reset(pReverb, false);
break;
@@ -492,7 +498,7 @@
pRvbModule->config.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
pRvbModule->config.outputCfg.mask = EFFECT_CONFIG_ALL;
- ret = Reverb_Configure(pRvbModule, &pRvbModule->config, true);
+ ret = Reverb_setConfig(pRvbModule, &pRvbModule->config, true);
if (ret < 0) {
ALOGV("Reverb_Init error %d on module %p", ret, pRvbModule);
}
@@ -501,7 +507,7 @@
}
/*----------------------------------------------------------------------------
- * Reverb_Init()
+ * Reverb_setConfig()
*----------------------------------------------------------------------------
* Purpose:
* Set input and output audio configuration.
@@ -518,7 +524,7 @@
*----------------------------------------------------------------------------
*/
-int Reverb_Configure(reverb_module_t *pRvbModule, effect_config_t *pConfig,
+int Reverb_setConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig,
bool init) {
reverb_object_t *pReverb = &pRvbModule->context;
int bufferSizeInSamples;
@@ -531,12 +537,12 @@
|| pConfig->outputCfg.channels != OUTPUT_CHANNELS
|| pConfig->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT
|| pConfig->outputCfg.format != AUDIO_FORMAT_PCM_16_BIT) {
- ALOGV("Reverb_Configure invalid config");
+ ALOGV("Reverb_setConfig invalid config");
return -EINVAL;
}
if ((pReverb->m_Aux && (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_MONO)) ||
(!pReverb->m_Aux && (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO))) {
- ALOGV("Reverb_Configure invalid config");
+ ALOGV("Reverb_setConfig invalid config");
return -EINVAL;
}
@@ -576,7 +582,7 @@
pReverb->m_nCosWT_5KHz = 25997;
break;
default:
- ALOGV("Reverb_Configure invalid sampling rate %d", pReverb->m_nSamplingRate);
+ ALOGV("Reverb_setConfig invalid sampling rate %d", pReverb->m_nSamplingRate);
return -EINVAL;
}
@@ -620,6 +626,28 @@
}
/*----------------------------------------------------------------------------
+ * Reverb_getConfig()
+ *----------------------------------------------------------------------------
+ * Purpose:
+ * Get input and output audio configuration.
+ *
+ * Inputs:
+ * pRvbModule - pointer to reverb effect module
+ * pConfig - pointer to effect_config_t structure containing input
+ * and output audio parameters configuration
+ * Outputs:
+ *
+ * Side Effects:
+ *
+ *----------------------------------------------------------------------------
+ */
+
+void Reverb_getConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pRvbModule->config, sizeof(effect_config_t));
+}
+
+/*----------------------------------------------------------------------------
* Reverb_Reset()
*----------------------------------------------------------------------------
* Purpose:
@@ -844,7 +872,7 @@
if (param == REVERB_PARAM_ROOM_HF_LEVEL) {
break;
}
- pValue32 = &pProperties->decayTime;
+ pValue32 = (int32_t *)&pProperties->decayTime;
/* FALL THROUGH */
case REVERB_PARAM_DECAY_TIME:
@@ -916,7 +944,7 @@
if (param == REVERB_PARAM_REFLECTIONS_LEVEL) {
break;
}
- pValue32 = &pProperties->reflectionsDelay;
+ pValue32 = (int32_t *)&pProperties->reflectionsDelay;
/* FALL THROUGH */
case REVERB_PARAM_REFLECTIONS_DELAY:
@@ -940,7 +968,7 @@
if (param == REVERB_PARAM_REVERB_LEVEL) {
break;
}
- pValue32 = &pProperties->reverbDelay;
+ pValue32 = (int32_t *)&pProperties->reverbDelay;
/* FALL THROUGH */
case REVERB_PARAM_REVERB_DELAY:
diff --git a/media/libeffects/testlibs/EffectReverb.h b/media/libeffects/testlibs/EffectReverb.h
index 8e2cc31..1fb14a7 100644
--- a/media/libeffects/testlibs/EffectReverb.h
+++ b/media/libeffects/testlibs/EffectReverb.h
@@ -303,12 +303,12 @@
int EffectQueryNumberEffects(uint32_t *pNumEffects);
int EffectQueryEffect(uint32_t index,
effect_descriptor_t *pDescriptor);
-int EffectCreate(effect_uuid_t *effectUID,
+int EffectCreate(const effect_uuid_t *effectUID,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle);
int EffectRelease(effect_handle_t handle);
-int EffectGetDescriptor(effect_uuid_t *uuid,
+int EffectGetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor);
static int Reverb_Process(effect_handle_t self,
@@ -329,7 +329,8 @@
*/
int Reverb_Init(reverb_module_t *pRvbModule, int aux, int preset);
-int Reverb_Configure(reverb_module_t *pRvbModule, effect_config_t *pConfig, bool init);
+int Reverb_setConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig, bool init);
+void Reverb_getConfig(reverb_module_t *pRvbModule, effect_config_t *pConfig);
void Reverb_Reset(reverb_object_t *pReverb, bool init);
int Reverb_setParameter (reverb_object_t *pReverb, int32_t param, size_t size, void *pValue);
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index c441710..51c8b68 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -78,7 +78,7 @@
}
//----------------------------------------------------------------------------
-// Visualizer_configure()
+// Visualizer_setConfig()
//----------------------------------------------------------------------------
// Purpose: Set input and output audio configuration.
//
@@ -91,9 +91,9 @@
//
//----------------------------------------------------------------------------
-int Visualizer_configure(VisualizerContext *pContext, effect_config_t *pConfig)
+int Visualizer_setConfig(VisualizerContext *pContext, effect_config_t *pConfig)
{
- ALOGV("Visualizer_configure start");
+ ALOGV("Visualizer_setConfig start");
if (pConfig->inputCfg.samplingRate != pConfig->outputCfg.samplingRate) return -EINVAL;
if (pConfig->inputCfg.channels != pConfig->outputCfg.channels) return -EINVAL;
@@ -112,6 +112,26 @@
//----------------------------------------------------------------------------
+// Visualizer_getConfig()
+//----------------------------------------------------------------------------
+// Purpose: Get input and output audio configuration.
+//
+// Inputs:
+// pContext: effect engine context
+// pConfig: pointer to effect_config_t structure holding input and output
+// configuration parameters
+//
+// Outputs:
+//
+//----------------------------------------------------------------------------
+
+void Visualizer_getConfig(VisualizerContext *pContext, effect_config_t *pConfig)
+{
+ memcpy(pConfig, &pContext->mConfig, sizeof(effect_config_t));
+}
+
+
+//----------------------------------------------------------------------------
// Visualizer_init()
//----------------------------------------------------------------------------
// Purpose: Initialize engine with default configuration.
@@ -144,7 +164,7 @@
pContext->mCaptureSize = VISUALIZER_CAPTURE_SIZE_MAX;
- Visualizer_configure(pContext, &pContext->mConfig);
+ Visualizer_setConfig(pContext, &pContext->mConfig);
return 0;
}
@@ -170,7 +190,7 @@
return 0;
}
-int VisualizerLib_Create(effect_uuid_t *uuid,
+int VisualizerLib_Create(const effect_uuid_t *uuid,
int32_t sessionId,
int32_t ioId,
effect_handle_t *pHandle) {
@@ -220,7 +240,7 @@
return 0;
}
-int VisualizerLib_GetDescriptor(effect_uuid_t *uuid,
+int VisualizerLib_GetDescriptor(const effect_uuid_t *uuid,
effect_descriptor_t *pDescriptor) {
if (pDescriptor == NULL || uuid == NULL){
@@ -337,14 +357,21 @@
}
*(int *) pReplyData = Visualizer_init(pContext);
break;
- case EFFECT_CMD_CONFIGURE:
+ case EFFECT_CMD_SET_CONFIG:
if (pCmdData == NULL || cmdSize != sizeof(effect_config_t)
|| pReplyData == NULL || *replySize != sizeof(int)) {
return -EINVAL;
}
- *(int *) pReplyData = Visualizer_configure(pContext,
+ *(int *) pReplyData = Visualizer_setConfig(pContext,
(effect_config_t *) pCmdData);
break;
+ case EFFECT_CMD_GET_CONFIG:
+ if (pReplyData == NULL ||
+ *replySize != sizeof(effect_config_t)) {
+ return -EINVAL;
+ }
+ Visualizer_getConfig(pContext, (effect_config_t *)pReplyData);
+ break;
case EFFECT_CMD_RESET:
Visualizer_reset(pContext);
break;
diff --git a/media/libmedia/Android.mk b/media/libmedia/Android.mk
index 7af4a87..23670df 100644
--- a/media/libmedia/Android.mk
+++ b/media/libmedia/Android.mk
@@ -43,13 +43,12 @@
IEffectClient.cpp \
AudioEffect.cpp \
Visualizer.cpp \
- MemoryLeakTrackUtil.cpp \
- fixedfft.cpp.arm
+ MemoryLeakTrackUtil.cpp
LOCAL_SHARED_LIBRARIES := \
libui libcutils libutils libbinder libsonivox libicuuc libexpat \
libcamera_client libstagefright_foundation \
- libgui libdl
+ libgui libdl libaudioutils
LOCAL_WHOLE_STATIC_LIBRARY := libmedia_helper
@@ -61,6 +60,7 @@
$(TOP)/frameworks/base/include/media/stagefright/openmax \
external/icu4c/common \
external/expat/lib \
- system/media/audio_effects/include
+ system/media/audio_effects/include \
+ system/media/audio_utils/include
include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libmedia/AudioEffect.cpp b/media/libmedia/AudioEffect.cpp
index 6639d06..6808aa2 100644
--- a/media/libmedia/AudioEffect.cpp
+++ b/media/libmedia/AudioEffect.cpp
@@ -159,7 +159,7 @@
mCblk->buffer = (uint8_t *)mCblk + bufOffset;
iEffect->asBinder()->linkToDeath(mIEffectClient);
- ALOGV("set() %p OK effect: %s id: %d status %d enabled %d, ", this, mDescriptor.name, mId, mStatus, mEnabled);
+ ALOGV("set() %p OK effect: %s id: %d status %d enabled %d", this, mDescriptor.name, mId, mStatus, mEnabled);
return mStatus;
}
@@ -202,7 +202,7 @@
status_t AudioEffect::setEnabled(bool enabled)
{
if (mStatus != NO_ERROR) {
- return INVALID_OPERATION;
+ return (mStatus == ALREADY_EXISTS) ? (status_t) INVALID_OPERATION : mStatus;
}
status_t status = NO_ERROR;
@@ -231,7 +231,7 @@
{
if (mStatus != NO_ERROR && mStatus != ALREADY_EXISTS) {
ALOGV("command() bad status %d", mStatus);
- return INVALID_OPERATION;
+ return mStatus;
}
if (cmdCode == EFFECT_CMD_ENABLE || cmdCode == EFFECT_CMD_DISABLE) {
@@ -263,7 +263,7 @@
status_t AudioEffect::setParameter(effect_param_t *param)
{
if (mStatus != NO_ERROR) {
- return INVALID_OPERATION;
+ return (mStatus == ALREADY_EXISTS) ? (status_t) INVALID_OPERATION : mStatus;
}
if (param == NULL || param->psize == 0 || param->vsize == 0) {
@@ -281,7 +281,7 @@
status_t AudioEffect::setParameterDeferred(effect_param_t *param)
{
if (mStatus != NO_ERROR) {
- return INVALID_OPERATION;
+ return (mStatus == ALREADY_EXISTS) ? (status_t) INVALID_OPERATION : mStatus;
}
if (param == NULL || param->psize == 0 || param->vsize == 0) {
@@ -307,7 +307,7 @@
status_t AudioEffect::setParameterCommit()
{
if (mStatus != NO_ERROR) {
- return INVALID_OPERATION;
+ return (mStatus == ALREADY_EXISTS) ? (status_t) INVALID_OPERATION : mStatus;
}
Mutex::Autolock _l(mCblk->lock);
@@ -321,7 +321,7 @@
status_t AudioEffect::getParameter(effect_param_t *param)
{
if (mStatus != NO_ERROR && mStatus != ALREADY_EXISTS) {
- return INVALID_OPERATION;
+ return mStatus;
}
if (param == NULL || param->psize == 0 || param->vsize == 0) {
@@ -341,8 +341,8 @@
void AudioEffect::binderDied()
{
ALOGW("IEffect died");
- mStatus = NO_INIT;
- if (mCbf) {
+ mStatus = DEAD_OBJECT;
+ if (mCbf != NULL) {
status_t status = DEAD_OBJECT;
mCbf(EVENT_ERROR, mUserData, &status);
}
@@ -363,7 +363,7 @@
mStatus = ALREADY_EXISTS;
}
}
- if (mCbf) {
+ if (mCbf != NULL) {
mCbf(EVENT_CONTROL_STATUS_CHANGED, mUserData, &controlGranted);
}
}
@@ -373,7 +373,7 @@
ALOGV("enableStatusChanged %p enabled %d mCbf %p", this, enabled, mCbf);
if (mStatus == ALREADY_EXISTS) {
mEnabled = enabled;
- if (mCbf) {
+ if (mCbf != NULL) {
mCbf(EVENT_ENABLE_STATUS_CHANGED, mUserData, &enabled);
}
}
@@ -389,7 +389,7 @@
return;
}
- if (mCbf && cmdCode == EFFECT_CMD_SET_PARAM) {
+ if (mCbf != NULL && cmdCode == EFFECT_CMD_SET_PARAM) {
effect_param_t *cmd = (effect_param_t *)cmdData;
cmd->status = *(int32_t *)replyData;
mCbf(EVENT_PARAMETER_CHANGED, mUserData, cmd);
@@ -412,7 +412,8 @@
return af->queryEffect(index, descriptor);
}
-status_t AudioEffect::getEffectDescriptor(effect_uuid_t *uuid, effect_descriptor_t *descriptor)
+status_t AudioEffect::getEffectDescriptor(const effect_uuid_t *uuid,
+ effect_descriptor_t *descriptor) /*const*/
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
diff --git a/media/libmedia/AudioRecord.cpp b/media/libmedia/AudioRecord.cpp
index 34a5eb7..a4068ff 100644
--- a/media/libmedia/AudioRecord.cpp
+++ b/media/libmedia/AudioRecord.cpp
@@ -39,9 +39,7 @@
#include <system/audio.h>
#include <cutils/bitops.h>
-
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <cutils/compiler.h>
namespace android {
// ---------------------------------------------------------------------------
@@ -50,7 +48,7 @@
status_t AudioRecord::getMinFrameCount(
int* frameCount,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelCount)
{
size_t size = 0;
@@ -80,14 +78,15 @@
// ---------------------------------------------------------------------------
AudioRecord::AudioRecord()
- : mStatus(NO_INIT), mSessionId(0)
+ : mStatus(NO_INIT), mSessionId(0),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
}
AudioRecord::AudioRecord(
- int inputSource,
+ audio_source_t inputSource,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -95,7 +94,8 @@
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT), mSessionId(0)
+ : mStatus(NO_INIT), mSessionId(0),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
mStatus = set(inputSource, sampleRate, format, channelMask,
frameCount, flags, cbf, user, notificationFrames, sessionId);
@@ -119,9 +119,9 @@
}
status_t AudioRecord::set(
- int inputSource,
+ audio_source_t inputSource,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -148,7 +148,7 @@
sampleRate = DEFAULT_SAMPLE_RATE;
}
// these below should probably come from the audioFlinger too...
- if (format == 0) {
+ if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
// validate parameters
@@ -206,11 +206,8 @@
return status;
}
- if (cbf != 0) {
+ if (cbf != NULL) {
mClientRecordThread = new ClientRecordThread(*this, threadCanCallJava);
- if (mClientRecordThread == 0) {
- return NO_INIT;
- }
}
mStatus = NO_ERROR;
@@ -231,7 +228,7 @@
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
- mInputSource = (uint8_t)inputSource;
+ mInputSource = inputSource;
mFlags = flags;
mInput = input;
AudioSystem::acquireAudioSessionId(mSessionId);
@@ -251,7 +248,7 @@
return mLatency;
}
-int AudioRecord::format() const
+audio_format_t AudioRecord::format() const
{
return mFormat;
}
@@ -266,7 +263,7 @@
return mFrameCount;
}
-int AudioRecord::frameSize() const
+size_t AudioRecord::frameSize() const
{
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
@@ -275,9 +272,9 @@
}
}
-int AudioRecord::inputSource() const
+audio_source_t AudioRecord::inputSource() const
{
- return (int)mInputSource;
+ return mInputSource;
}
// -------------------------------------------------------------------------
@@ -296,7 +293,6 @@
return WOULD_BLOCK;
}
}
- t->mLock.lock();
}
AutoMutex lock(mLock);
@@ -308,10 +304,25 @@
if (mActive == 0) {
mActive = 1;
+ pid_t tid;
+ if (t != 0) {
+ mReadyToRun = WOULD_BLOCK;
+ t->run("ClientRecordThread", ANDROID_PRIORITY_AUDIO);
+ tid = t->getTid(); // pid_t is unknown until run()
+ ALOGV("getTid=%d", tid);
+ if (tid == -1) {
+ tid = 0;
+ }
+ // thread blocks in readyToRun()
+ } else {
+ tid = 0; // not gettid()
+ }
+
cblk->lock.lock();
if (!(cblk->flags & CBLK_INVALID_MSK)) {
cblk->lock.unlock();
- ret = mAudioRecord->start();
+ ALOGV("mAudioRecord->start(tid=%d)", tid);
+ ret = mAudioRecord->start(tid);
cblk->lock.lock();
if (ret == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
@@ -326,19 +337,22 @@
cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
cblk->waitTimeMs = 0;
if (t != 0) {
- t->run("ClientRecordThread", ANDROID_PRIORITY_AUDIO);
+ // thread unblocks in readyToRun() and returns NO_ERROR
+ mReadyToRun = NO_ERROR;
+ mCondition.signal();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+ mPreviousPriority = getpriority(PRIO_PROCESS, 0);
+ mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0);
+ androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
}
} else {
mActive = 0;
+ // thread unblocks in readyToRun() and returns NO_INIT
+ mReadyToRun = NO_INIT;
+ mCondition.signal();
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
-
return ret;
}
@@ -348,10 +362,6 @@
ALOGV("stop");
- if (t != 0) {
- t->mLock.lock();
- }
-
AutoMutex lock(mLock);
if (mActive == 1) {
mActive = 0;
@@ -363,14 +373,11 @@
if (t != 0) {
t->requestExit();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
-
return NO_ERROR;
}
@@ -387,7 +394,7 @@
status_t AudioRecord::setMarkerPosition(uint32_t marker)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
@@ -397,7 +404,7 @@
status_t AudioRecord::getMarkerPosition(uint32_t *marker)
{
- if (marker == 0) return BAD_VALUE;
+ if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
@@ -406,7 +413,7 @@
status_t AudioRecord::setPositionUpdatePeriod(uint32_t updatePeriod)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
@@ -418,7 +425,7 @@
status_t AudioRecord::getPositionUpdatePeriod(uint32_t *updatePeriod)
{
- if (updatePeriod == 0) return BAD_VALUE;
+ if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
@@ -427,7 +434,7 @@
status_t AudioRecord::getPosition(uint32_t *position)
{
- if (position == 0) return BAD_VALUE;
+ if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mCblk->user;
@@ -448,7 +455,7 @@
// must be called with mLock held
status_t AudioRecord::openRecord_l(
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -508,11 +515,11 @@
goto start_loop_here;
while (framesReady == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount)) {
+ if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
@@ -529,13 +536,13 @@
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_record;
}
- if (__builtin_expect(result!=NO_ERROR, false)) {
+ if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
ALOGW( "obtainBuffer timed out (is the CPU pegged?) "
"user=%08x, server=%08x", cblk->user, cblk->server);
cblk->lock.unlock();
- result = mAudioRecord->start();
+ result = mAudioRecord->start(0); // callback thread hasn't changed
cblk->lock.lock();
if (result == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
@@ -773,7 +780,7 @@
result = openRecord_l(cblk->sampleRate, mFormat, mChannelMask,
mFrameCount, mFlags, getInput_l());
if (result == NO_ERROR) {
- result = mAudioRecord->start();
+ result = mAudioRecord->start(0); // callback thread hasn't changed
}
if (result != NO_ERROR) {
mActive = false;
@@ -824,6 +831,15 @@
return mReceiver.processAudioBuffer(this);
}
+status_t AudioRecord::ClientRecordThread::readyToRun()
+{
+ AutoMutex(mReceiver.mLock);
+ while (mReceiver.mReadyToRun == WOULD_BLOCK) {
+ mReceiver.mCondition.wait(mReceiver.mLock);
+ }
+ return mReceiver.mReadyToRun;
+}
+
// -------------------------------------------------------------------------
}; // namespace android
diff --git a/media/libmedia/AudioSystem.cpp b/media/libmedia/AudioSystem.cpp
index f7f129c..ec4c044 100644
--- a/media/libmedia/AudioSystem.cpp
+++ b/media/libmedia/AudioSystem.cpp
@@ -35,12 +35,13 @@
sp<AudioSystem::AudioFlingerClient> AudioSystem::gAudioFlingerClient;
audio_error_callback AudioSystem::gAudioErrorCallback = NULL;
// Cached values
-DefaultKeyedVector<int, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
+
+DefaultKeyedVector<audio_stream_type_t, audio_io_handle_t> AudioSystem::gStreamOutputMap(0);
DefaultKeyedVector<audio_io_handle_t, AudioSystem::OutputDescriptor *> AudioSystem::gOutputs(0);
-// Cached values for recording queries
+// Cached values for recording queries, all protected by gLock
uint32_t AudioSystem::gPrevInSamplingRate = 16000;
-int AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
+audio_format_t AudioSystem::gPrevInFormat = AUDIO_FORMAT_PCM_16_BIT;
int AudioSystem::gPrevInChannelCount = 1;
size_t AudioSystem::gInBuffSize = 0;
@@ -49,7 +50,7 @@
const sp<IAudioFlinger>& AudioSystem::get_audio_flinger()
{
Mutex::Autolock _l(gLock);
- if (gAudioFlinger.get() == 0) {
+ if (gAudioFlinger == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
@@ -120,7 +121,8 @@
return NO_ERROR;
}
-status_t AudioSystem::setStreamVolume(int stream, float value, int output)
+status_t AudioSystem::setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -129,7 +131,7 @@
return NO_ERROR;
}
-status_t AudioSystem::setStreamMute(int stream, bool mute)
+status_t AudioSystem::setStreamMute(audio_stream_type_t stream, bool mute)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -138,7 +140,8 @@
return NO_ERROR;
}
-status_t AudioSystem::getStreamVolume(int stream, float* volume, int output)
+status_t AudioSystem::getStreamVolume(audio_stream_type_t stream, float* volume,
+ audio_io_handle_t output)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -147,7 +150,7 @@
return NO_ERROR;
}
-status_t AudioSystem::getStreamMute(int stream, bool* mute)
+status_t AudioSystem::getStreamMute(audio_stream_type_t stream, bool* mute)
{
if (uint32_t(stream) >= AUDIO_STREAM_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -156,9 +159,9 @@
return NO_ERROR;
}
-status_t AudioSystem::setMode(int mode)
+status_t AudioSystem::setMode(audio_mode_t mode)
{
- if (mode >= AUDIO_MODE_CNT) return BAD_VALUE;
+ if (uint32_t(mode) >= AUDIO_MODE_CNT) return BAD_VALUE;
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
return af->setMode(mode);
@@ -203,7 +206,12 @@
return volume ? 100 - int(dBConvertInverse * log(volume) + 0.5) : 0;
}
-status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType)
+// DEPRECATED
+status_t AudioSystem::getOutputSamplingRate(int* samplingRate, int streamType) {
+ return getOutputSamplingRate(samplingRate, (audio_stream_type_t)streamType);
+}
+
+status_t AudioSystem::getOutputSamplingRate(int* samplingRate, audio_stream_type_t streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
@@ -212,14 +220,14 @@
streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((audio_stream_type_t)streamType);
+ output = getOutput(streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
ALOGV("getOutputSamplingRate() no output descriptor for output %d in gOutputs", output);
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -236,7 +244,12 @@
return NO_ERROR;
}
-status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType)
+// DEPRECATED
+status_t AudioSystem::getOutputFrameCount(int* frameCount, int streamType) {
+ return getOutputFrameCount(frameCount, (audio_stream_type_t)streamType);
+}
+
+status_t AudioSystem::getOutputFrameCount(int* frameCount, audio_stream_type_t streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
@@ -245,14 +258,14 @@
streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((audio_stream_type_t)streamType);
+ output = getOutput(streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -267,7 +280,7 @@
return NO_ERROR;
}
-status_t AudioSystem::getOutputLatency(uint32_t* latency, int streamType)
+status_t AudioSystem::getOutputLatency(uint32_t* latency, audio_stream_type_t streamType)
{
OutputDescriptor *outputDesc;
audio_io_handle_t output;
@@ -276,14 +289,14 @@
streamType = AUDIO_STREAM_MUSIC;
}
- output = getOutput((audio_stream_type_t)streamType);
+ output = getOutput(streamType);
if (output == 0) {
return PERMISSION_DENIED;
}
gLock.lock();
outputDesc = AudioSystem::gOutputs.valueFor(output);
- if (outputDesc == 0) {
+ if (outputDesc == NULL) {
gLock.unlock();
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -298,25 +311,30 @@
return NO_ERROR;
}
-status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, int format, int channelCount,
+status_t AudioSystem::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount,
size_t* buffSize)
{
+ gLock.lock();
// Do we have a stale gInBufferSize or are we requesting the input buffer size for new values
- if ((gInBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
+ size_t inBuffSize = gInBuffSize;
+ if ((inBuffSize == 0) || (sampleRate != gPrevInSamplingRate) || (format != gPrevInFormat)
|| (channelCount != gPrevInChannelCount)) {
+ gLock.unlock();
+ const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+ if (af == 0) {
+ return PERMISSION_DENIED;
+ }
+ inBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
+ gLock.lock();
// save the request params
gPrevInSamplingRate = sampleRate;
gPrevInFormat = format;
gPrevInChannelCount = channelCount;
- gInBuffSize = 0;
- const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
- if (af == 0) {
- return PERMISSION_DENIED;
- }
- gInBuffSize = af->getInputBufferSize(sampleRate, format, channelCount);
+ gInBuffSize = inBuffSize;
}
- *buffSize = gInBuffSize;
+ gLock.unlock();
+ *buffSize = inBuffSize;
return NO_ERROR;
}
@@ -328,7 +346,7 @@
return af->setVoiceVolume(value);
}
-status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream)
+status_t AudioSystem::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, audio_stream_type_t stream)
{
const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
if (af == 0) return PERMISSION_DENIED;
@@ -337,7 +355,7 @@
stream = AUDIO_STREAM_MUSIC;
}
- return af->getRenderPosition(halFrames, dspFrames, getOutput((audio_stream_type_t)stream));
+ return af->getRenderPosition(halFrames, dspFrames, getOutput(stream));
}
unsigned int AudioSystem::getInputFramesLost(audio_io_handle_t ioHandle) {
@@ -386,10 +404,11 @@
ALOGW("AudioFlinger server died!");
}
-void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, int ioHandle, void *param2) {
+void AudioSystem::AudioFlingerClient::ioConfigChanged(int event, audio_io_handle_t ioHandle,
+ void *param2) {
ALOGV("ioConfigChanged() event %d", event);
OutputDescriptor *desc;
- uint32_t stream;
+ audio_stream_type_t stream;
if (ioHandle == 0) return;
@@ -397,8 +416,8 @@
switch (event) {
case STREAM_CONFIG_CHANGED:
- if (param2 == 0) break;
- stream = *(uint32_t *)param2;
+ if (param2 == NULL) break;
+ stream = *(audio_stream_type_t *)param2;
ALOGV("ioConfigChanged() STREAM_CONFIG_CHANGED stream %d, output %d", stream, ioHandle);
if (gStreamOutputMap.indexOfKey(stream) >= 0) {
gStreamOutputMap.replaceValueFor(stream, ioHandle);
@@ -409,7 +428,7 @@
ALOGV("ioConfigChanged() opening already existing output! %d", ioHandle);
break;
}
- if (param2 == 0) break;
+ if (param2 == NULL) break;
desc = (OutputDescriptor *)param2;
OutputDescriptor *outputDesc = new OutputDescriptor(*desc);
@@ -438,7 +457,7 @@
ALOGW("ioConfigChanged() modifying unknow output! %d", ioHandle);
break;
}
- if (param2 == 0) break;
+ if (param2 == NULL) break;
desc = (OutputDescriptor *)param2;
ALOGV("ioConfigChanged() new config for output %d samplingRate %d, format %d channels %d frameCount %d latency %d",
@@ -462,7 +481,7 @@
gAudioErrorCallback = cb;
}
-bool AudioSystem::routedToA2dpOutput(int streamType) {
+bool AudioSystem::routedToA2dpOutput(audio_stream_type_t streamType) {
switch(streamType) {
case AUDIO_STREAM_MUSIC:
case AUDIO_STREAM_VOICE_CALL:
@@ -484,7 +503,7 @@
const sp<IAudioPolicyService>& AudioSystem::get_audio_policy_service()
{
gLock.lock();
- if (gAudioPolicyService.get() == 0) {
+ if (gAudioPolicyService == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
@@ -531,21 +550,15 @@
return aps->getDeviceConnectionState(device, device_address);
}
-status_t AudioSystem::setPhoneState(int state)
+status_t AudioSystem::setPhoneState(audio_mode_t state)
{
+ if (uint32_t(state) >= AUDIO_MODE_CNT) return BAD_VALUE;
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
return aps->setPhoneState(state);
}
-status_t AudioSystem::setRingerMode(uint32_t mode, uint32_t mask)
-{
- const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
- if (aps == 0) return PERMISSION_DENIED;
- return aps->setRingerMode(mode, mask);
-}
-
status_t AudioSystem::setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
@@ -563,7 +576,7 @@
audio_io_handle_t AudioSystem::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_policy_output_flags_t flags)
{
@@ -621,9 +634,9 @@
aps->releaseOutput(output);
}
-audio_io_handle_t AudioSystem::getInput(int inputSource,
+audio_io_handle_t AudioSystem::getInput(audio_source_t inputSource,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_in_acoustics_t acoustics,
int sessionId)
@@ -663,18 +676,22 @@
return aps->initStreamVolume(stream, indexMin, indexMax);
}
-status_t AudioSystem::setStreamVolumeIndex(audio_stream_type_t stream, int index)
+status_t AudioSystem::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- return aps->setStreamVolumeIndex(stream, index);
+ return aps->setStreamVolumeIndex(stream, index, device);
}
-status_t AudioSystem::getStreamVolumeIndex(audio_stream_type_t stream, int *index)
+status_t AudioSystem::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
- return aps->getStreamVolumeIndex(stream, index);
+ return aps->getStreamVolumeIndex(stream, index, device);
}
uint32_t AudioSystem::getStrategyForStream(audio_stream_type_t stream)
@@ -723,7 +740,7 @@
return aps->setEffectEnabled(id, enabled);
}
-status_t AudioSystem::isStreamActive(int stream, bool* state, uint32_t inPastMs)
+status_t AudioSystem::isStreamActive(audio_stream_type_t stream, bool* state, uint32_t inPastMs)
{
const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
if (aps == 0) return PERMISSION_DENIED;
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index d51cd69..74c97ed 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -1,4 +1,4 @@
-/* //device/extlibs/pv/android/AudioTrack.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -38,12 +38,12 @@
#include <utils/Atomic.h>
#include <cutils/bitops.h>
+#include <cutils/compiler.h>
#include <system/audio.h>
#include <system/audio_policy.h>
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
+#include <audio_utils/primitives.h>
namespace android {
// ---------------------------------------------------------------------------
@@ -51,7 +51,7 @@
// static
status_t AudioTrack::getMinFrameCount(
int* frameCount,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate)
{
int afSampleRate;
@@ -79,11 +79,35 @@
// ---------------------------------------------------------------------------
AudioTrack::AudioTrack()
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mIsTimed(false),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+ mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
}
AudioTrack::AudioTrack(
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ int channelMask,
+ int frameCount,
+ uint32_t flags,
+ callback_t cbf,
+ void* user,
+ int notificationFrames,
+ int sessionId)
+ : mStatus(NO_INIT),
+ mIsTimed(false),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+ mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
+{
+ mStatus = set(streamType, sampleRate, format, channelMask,
+ frameCount, flags, cbf, user, notificationFrames,
+ 0, false, sessionId);
+}
+
+AudioTrack::AudioTrack(
int streamType,
uint32_t sampleRate,
int format,
@@ -94,17 +118,18 @@
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
- mStatus = set(streamType, sampleRate, format, channelMask,
+ mStatus = set((audio_stream_type_t)streamType, sampleRate, (audio_format_t)format, channelMask,
frameCount, flags, cbf, user, notificationFrames,
0, false, sessionId);
}
AudioTrack::AudioTrack(
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
const sp<IMemory>& sharedBuffer,
uint32_t flags,
@@ -112,7 +137,10 @@
void* user,
int notificationFrames,
int sessionId)
- : mStatus(NO_INIT)
+ : mStatus(NO_INIT),
+ mIsTimed(false),
+ mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+ mPreviousSchedulingGroup(ANDROID_TGROUP_DEFAULT)
{
mStatus = set(streamType, sampleRate, format, channelMask,
0, flags, cbf, user, notificationFrames,
@@ -139,9 +167,9 @@
}
status_t AudioTrack::set(
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
int frameCount,
uint32_t flags,
@@ -178,7 +206,7 @@
sampleRate = afSampleRate;
}
// these below should probably come from the audioFlinger too...
- if (format == 0) {
+ if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
}
if (channelMask == 0) {
@@ -203,8 +231,8 @@
uint32_t channelCount = popcount(channelMask);
audio_io_handle_t output = AudioSystem::getOutput(
- (audio_stream_type_t)streamType,
- sampleRate,format, channelMask,
+ streamType,
+ sampleRate, format, channelMask,
(audio_policy_output_flags_t)flags);
if (output == 0) {
@@ -214,7 +242,7 @@
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
- mSendLevel = 0;
+ mSendLevel = 0.0f;
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mSessionId = sessionId;
@@ -223,7 +251,7 @@
// create the IAudioTrack
status_t status = createTrack_l(streamType,
sampleRate,
- (uint32_t)format,
+ format,
(uint32_t)channelMask,
frameCount,
flags,
@@ -235,23 +263,19 @@
return status;
}
- if (cbf != 0) {
+ if (cbf != NULL) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
- if (mAudioTrackThread == 0) {
- ALOGE("Could not create callback thread");
- return NO_INIT;
- }
}
mStatus = NO_ERROR;
mStreamType = streamType;
- mFormat = (uint32_t)format;
+ mFormat = format;
mChannelMask = (uint32_t)channelMask;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
- mActive = 0;
+ mActive = false;
mCbf = cbf;
mUserData = user;
mLoopCount = 0;
@@ -278,12 +302,12 @@
return mLatency;
}
-int AudioTrack::streamType() const
+audio_stream_type_t AudioTrack::streamType() const
{
return mStreamType;
}
-int AudioTrack::format() const
+audio_format_t AudioTrack::format() const
{
return mFormat;
}
@@ -298,7 +322,7 @@
return mCblk->frameCount;
}
-int AudioTrack::frameSize() const
+size_t AudioTrack::frameSize() const
{
if (audio_is_linear_pcm(mFormat)) {
return channelCount()*audio_bytes_per_sample(mFormat);
@@ -327,7 +351,6 @@
return;
}
}
- t->mLock.lock();
}
AutoMutex lock(mLock);
@@ -337,24 +360,34 @@
sp <IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
- if (mActive == 0) {
+ if (!mActive) {
mFlushed = false;
- mActive = 1;
+ mActive = true;
mNewPosition = cblk->server + mUpdatePeriod;
cblk->lock.lock();
cblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
cblk->waitTimeMs = 0;
android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
+ pid_t tid;
if (t != 0) {
- t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO);
+ t->run("AudioTrackThread", ANDROID_PRIORITY_AUDIO);
+ tid = t->getTid(); // pid_t is unknown until run()
+ ALOGV("getTid=%d", tid);
+ if (tid == -1) {
+ tid = 0;
+ }
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+ mPreviousPriority = getpriority(PRIO_PROCESS, 0);
+ mPreviousSchedulingGroup = androidGetThreadSchedulingGroup(0);
+ androidSetThreadPriority(0, ANDROID_PRIORITY_AUDIO);
+ tid = 0; // not gettid()
}
ALOGV("start %p before lock cblk %p", this, mCblk);
if (!(cblk->flags & CBLK_INVALID_MSK)) {
cblk->lock.unlock();
- status = mAudioTrack->start();
+ ALOGV("mAudioTrack->start(tid=%d)", tid);
+ status = mAudioTrack->start(tid);
cblk->lock.lock();
if (status == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
@@ -366,18 +399,16 @@
cblk->lock.unlock();
if (status != NO_ERROR) {
ALOGV("start() failed");
- mActive = 0;
+ mActive = false;
if (t != 0) {
t->requestExit();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
}
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
}
void AudioTrack::stop()
@@ -385,13 +416,10 @@
sp<AudioTrackThread> t = mAudioTrackThread;
ALOGV("stop %p", this);
- if (t != 0) {
- t->mLock.lock();
- }
AutoMutex lock(mLock);
- if (mActive == 1) {
- mActive = 0;
+ if (mActive) {
+ mActive = false;
mCblk->cv.signal();
mAudioTrack->stop();
// Cancel loops (If we are in the middle of a loop, playback
@@ -408,18 +436,17 @@
if (t != 0) {
t->requestExit();
} else {
- setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
+ setpriority(PRIO_PROCESS, 0, mPreviousPriority);
+ androidSetThreadSchedulingGroup(0, mPreviousSchedulingGroup);
}
}
- if (t != 0) {
- t->mLock.unlock();
- }
}
bool AudioTrack::stopped() const
{
- return !mActive;
+ AutoMutex lock(mLock);
+ return stopped_l();
}
void AudioTrack::flush()
@@ -451,8 +478,8 @@
{
ALOGV("pause");
AutoMutex lock(mLock);
- if (mActive == 1) {
- mActive = 0;
+ if (mActive) {
+ mActive = false;
mAudioTrack->pause();
}
}
@@ -470,7 +497,7 @@
status_t AudioTrack::setVolume(float left, float right)
{
- if (left > 1.0f || right > 1.0f) {
+ if (left < 0.0f || left > 1.0f || right < 0.0f || right > 1.0f) {
return BAD_VALUE;
}
@@ -478,13 +505,12 @@
mVolume[LEFT] = left;
mVolume[RIGHT] = right;
- // write must be atomic
- mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
+ mCblk->setVolumeLR((uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000));
return NO_ERROR;
}
-void AudioTrack::getVolume(float* left, float* right)
+void AudioTrack::getVolume(float* left, float* right) const
{
if (left != NULL) {
*left = mVolume[LEFT];
@@ -497,19 +523,19 @@
status_t AudioTrack::setAuxEffectSendLevel(float level)
{
ALOGV("setAuxEffectSendLevel(%f)", level);
- if (level > 1.0f) {
+ if (level < 0.0f || level > 1.0f) {
return BAD_VALUE;
}
AutoMutex lock(mLock);
mSendLevel = level;
- mCblk->sendLevel = uint16_t(level * 0x1000);
+ mCblk->setSendLevel(level);
return NO_ERROR;
}
-void AudioTrack::getAuxEffectSendLevel(float* level)
+void AudioTrack::getAuxEffectSendLevel(float* level) const
{
if (level != NULL) {
*level = mSendLevel;
@@ -520,6 +546,10 @@
{
int afSamplingRate;
+ if (mIsTimed) {
+ return INVALID_OPERATION;
+ }
+
if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
return NO_INIT;
}
@@ -531,8 +561,12 @@
return NO_ERROR;
}
-uint32_t AudioTrack::getSampleRate()
+uint32_t AudioTrack::getSampleRate() const
{
+ if (mIsTimed) {
+ return INVALID_OPERATION;
+ }
+
AutoMutex lock(mLock);
return mCblk->sampleRate;
}
@@ -558,6 +592,10 @@
return NO_ERROR;
}
+ if (mIsTimed) {
+ return INVALID_OPERATION;
+ }
+
if (loopStart >= loopEnd ||
loopEnd - loopStart > cblk->frameCount ||
cblk->server > loopStart) {
@@ -579,29 +617,9 @@
return NO_ERROR;
}
-status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
-{
- AutoMutex lock(mLock);
- if (loopStart != 0) {
- *loopStart = mCblk->loopStart;
- }
- if (loopEnd != 0) {
- *loopEnd = mCblk->loopEnd;
- }
- if (loopCount != 0) {
- if (mCblk->loopCount < 0) {
- *loopCount = -1;
- } else {
- *loopCount = mCblk->loopCount;
- }
- }
-
- return NO_ERROR;
-}
-
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
@@ -609,9 +627,9 @@
return NO_ERROR;
}
-status_t AudioTrack::getMarkerPosition(uint32_t *marker)
+status_t AudioTrack::getMarkerPosition(uint32_t *marker) const
{
- if (marker == 0) return BAD_VALUE;
+ if (marker == NULL) return BAD_VALUE;
*marker = mMarkerPosition;
@@ -620,7 +638,7 @@
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
- if (mCbf == 0) return INVALID_OPERATION;
+ if (mCbf == NULL) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
@@ -630,9 +648,9 @@
return NO_ERROR;
}
-status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
+status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod) const
{
- if (updatePeriod == 0) return BAD_VALUE;
+ if (updatePeriod == NULL) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
@@ -641,10 +659,13 @@
status_t AudioTrack::setPosition(uint32_t position)
{
- AutoMutex lock(mLock);
- Mutex::Autolock _l(mCblk->lock);
+ if (mIsTimed) return INVALID_OPERATION;
- if (!stopped()) return INVALID_OPERATION;
+ AutoMutex lock(mLock);
+
+ if (!stopped_l()) return INVALID_OPERATION;
+
+ Mutex::Autolock _l(mCblk->lock);
if (position > mCblk->user) return BAD_VALUE;
@@ -656,7 +677,7 @@
status_t AudioTrack::getPosition(uint32_t *position)
{
- if (position == 0) return BAD_VALUE;
+ if (position == NULL) return BAD_VALUE;
AutoMutex lock(mLock);
*position = mFlushed ? 0 : mCblk->server;
@@ -667,7 +688,7 @@
{
AutoMutex lock(mLock);
- if (!stopped()) return INVALID_OPERATION;
+ if (!stopped_l()) return INVALID_OPERATION;
flush_l();
@@ -685,11 +706,11 @@
// must be called with mLock held
audio_io_handle_t AudioTrack::getOutput_l()
{
- return AudioSystem::getOutput((audio_stream_type_t)mStreamType,
+ return AudioSystem::getOutput(mStreamType,
mCblk->sampleRate, mFormat, mChannelMask, (audio_policy_output_flags_t)mFlags);
}
-int AudioTrack::getSessionId()
+int AudioTrack::getSessionId() const
{
return mSessionId;
}
@@ -708,9 +729,9 @@
// must be called with mLock held
status_t AudioTrack::createTrack_l(
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -771,7 +792,7 @@
}
}
} else {
- // Ensure that buffer alignment matches channelcount
+ // Ensure that buffer alignment matches channelCount
int channelCount = popcount(channelMask);
if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
ALOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
@@ -790,6 +811,7 @@
((uint16_t)flags) << 16,
sharedBuffer,
output,
+ mIsTimed,
&mSessionId,
&status);
@@ -802,9 +824,7 @@
ALOGE("Could not get control block");
return NO_INIT;
}
- mAudioTrack.clear();
mAudioTrack = track;
- mCblkMemory.clear();
mCblkMemory = cblk;
mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
android_atomic_or(CBLK_DIRECTION_OUT, &mCblk->flags);
@@ -816,8 +836,8 @@
mCblk->stepUser(mCblk->frameCount);
}
- mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
- mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);
+ mCblk->setVolumeLR((uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000));
+ mCblk->setSendLevel(mSendLevel);
mAudioTrack->attachAuxEffect(mAuxEffectId);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
@@ -829,7 +849,7 @@
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
AutoMutex lock(mLock);
- int active;
+ bool active;
status_t result = NO_ERROR;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
@@ -851,12 +871,12 @@
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount)) {
+ if (CC_UNLIKELY(!waitCount)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
@@ -865,7 +885,7 @@
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
cblk->lock.unlock();
mLock.lock();
- if (mActive == 0) {
+ if (!mActive) {
return status_t(STOPPED);
}
cblk->lock.lock();
@@ -874,7 +894,7 @@
if (cblk->flags & CBLK_INVALID_MSK) {
goto create_new_track;
}
- if (__builtin_expect(result!=NO_ERROR, false)) {
+ if (CC_UNLIKELY(result != NO_ERROR)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
// timing out when a loop has been set and we have already written upto loop end
@@ -884,7 +904,7 @@
"user=%08x, server=%08x", this, cblk->user, cblk->server);
//unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
cblk->lock.unlock();
- result = mAudioTrack->start();
+ result = mAudioTrack->start(0); // callback thread hasn't changed
cblk->lock.lock();
if (result == DEAD_OBJECT) {
android_atomic_or(CBLK_INVALID_ON, &cblk->flags);
@@ -916,7 +936,7 @@
if (mActive && (cblk->flags & CBLK_DISABLED_MSK)) {
android_atomic_and(~CBLK_DISABLED_ON, &cblk->flags);
ALOGW("obtainBuffer() track %p disabled, restarting", this);
- mAudioTrack->start();
+ mAudioTrack->start(0); // callback thread hasn't changed
}
cblk->waitTimeMs = 0;
@@ -958,9 +978,11 @@
{
if (mSharedBuffer != 0) return INVALID_OPERATION;
+ if (mIsTimed) return INVALID_OPERATION;
if (ssize_t(userSize) < 0) {
- // sanity-check. user is most-likely passing an error code.
+ // Sanity-check: user is most-likely passing an error code, and it would
+ // make the return value ambiguous (actualSize vs error).
ALOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
buffer, userSize, userSize);
return BAD_VALUE;
@@ -978,13 +1000,11 @@
ssize_t written = 0;
const int8_t *src = (const int8_t *)buffer;
Buffer audioBuffer;
- size_t frameSz = (size_t)frameSize();
+ size_t frameSz = frameSize();
do {
audioBuffer.frameCount = userSize/frameSz;
- // Calling obtainBuffer() with a negative wait count causes
- // an (almost) infinite wait time.
status_t err = obtainBuffer(&audioBuffer, -1);
if (err < 0) {
// out of buffers, return #bytes written
@@ -998,12 +1018,7 @@
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
toWrite = audioBuffer.size>>1;
- // 8 to 16 bit conversion
- int count = toWrite;
- int16_t *dst = (int16_t *)(audioBuffer.i8);
- while(count--) {
- *dst++ = (int16_t)(*src++^0x80) << 8;
- }
+ memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) src, toWrite);
} else {
toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, src, toWrite);
@@ -1020,6 +1035,59 @@
// -------------------------------------------------------------------------
+TimedAudioTrack::TimedAudioTrack() {
+ mIsTimed = true;
+}
+
+status_t TimedAudioTrack::allocateTimedBuffer(size_t size, sp<IMemory>* buffer)
+{
+ status_t result = UNKNOWN_ERROR;
+
+ // If the track is not invalid already, try to allocate a buffer. alloc
+ // fails indicating that the server is dead, flag the track as invalid so
+ // we can attempt to restore in in just a bit.
+ if (!(mCblk->flags & CBLK_INVALID_MSK)) {
+ result = mAudioTrack->allocateTimedBuffer(size, buffer);
+ if (result == DEAD_OBJECT) {
+ android_atomic_or(CBLK_INVALID_ON, &mCblk->flags);
+ }
+ }
+
+ // If the track is invalid at this point, attempt to restore it. and try the
+ // allocation one more time.
+ if (mCblk->flags & CBLK_INVALID_MSK) {
+ mCblk->lock.lock();
+ result = restoreTrack_l(mCblk, false);
+ mCblk->lock.unlock();
+
+ if (result == OK)
+ result = mAudioTrack->allocateTimedBuffer(size, buffer);
+ }
+
+ return result;
+}
+
+status_t TimedAudioTrack::queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts)
+{
+ // restart track if it was disabled by audioflinger due to previous underrun
+ if (mActive && (mCblk->flags & CBLK_DISABLED_MSK)) {
+ android_atomic_and(~CBLK_DISABLED_ON, &mCblk->flags);
+ ALOGW("queueTimedBuffer() track %p disabled, restarting", this);
+ mAudioTrack->start(0);
+ }
+
+ return mAudioTrack->queueTimedBuffer(buffer, pts);
+}
+
+status_t TimedAudioTrack::setMediaTimeTransform(const LinearTransform& xform,
+ TargetTimeline target)
+{
+ return mAudioTrack->setMediaTimeTransform(xform, target);
+}
+
+// -------------------------------------------------------------------------
+
bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
{
Buffer audioBuffer;
@@ -1032,10 +1100,11 @@
sp <IAudioTrack> audioTrack = mAudioTrack;
sp <IMemory> iMem = mCblkMemory;
audio_track_cblk_t* cblk = mCblk;
+ bool active = mActive;
mLock.unlock();
// Manage underrun callback
- if (mActive && (cblk->framesAvailable() == cblk->frameCount)) {
+ if (active && (cblk->framesAvailable() == cblk->frameCount)) {
ALOGV("Underrun user: %x, server: %x, flags %04x", cblk->user, cblk->server, cblk->flags);
if (!(android_atomic_or(CBLK_UNDERRUN_ON, &cblk->flags) & CBLK_UNDERRUN_MSK)) {
mCbf(EVENT_UNDERRUN, mUserData, 0);
@@ -1078,6 +1147,9 @@
frames = mRemainingFrames;
}
+ // See description of waitCount parameter at declaration of obtainBuffer().
+ // The logic below prevents us from being stuck below at obtainBuffer()
+ // not being able to handle timed events (position, markers, loops).
int32_t waitCount = -1;
if (mUpdatePeriod || (!mMarkerReached && mMarkerPosition) || mLoopCount) {
waitCount = 1;
@@ -1087,9 +1159,6 @@
audioBuffer.frameCount = frames;
- // Calling obtainBuffer() with a wait count of 1
- // limits wait time to WAIT_PERIOD_MS. This prevents from being
- // stuck here not being able to handle timed events (position, markers, loops).
status_t err = obtainBuffer(&audioBuffer, waitCount);
if (err < NO_ERROR) {
if (err != TIMED_OUT) {
@@ -1123,19 +1192,14 @@
if (writtenSize > reqSize) writtenSize = reqSize;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT)) {
- // 8 to 16 bit conversion
- const int8_t *src = audioBuffer.i8 + writtenSize-1;
- int count = writtenSize;
- int16_t *dst = audioBuffer.i16 + writtenSize-1;
- while(count--) {
- *dst-- = (int16_t)(*src--^0x80) << 8;
- }
+ // 8 to 16 bit conversion, note that source and destination are the same address
+ memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) audioBuffer.i8, writtenSize);
writtenSize <<= 1;
}
audioBuffer.size = writtenSize;
// NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
- // 8 bit PCM data: in this case, mCblk->frameSize is based on a sampel size of
+ // 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
// 16 bit.
audioBuffer.frameCount = writtenSize/mCblk->frameSize;
@@ -1216,7 +1280,7 @@
}
}
if (mActive) {
- result = mAudioTrack->start();
+ result = mAudioTrack->start(0); // callback thread hasn't changed
ALOGW_IF(result != NO_ERROR, "restoreTrack_l() start() failed status %d", result);
}
if (fromStart && result == NO_ERROR) {
@@ -1307,15 +1371,15 @@
audio_track_cblk_t::audio_track_cblk_t()
: lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
- userBase(0), serverBase(0), buffers(0), frameCount(0),
- loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
- sendLevel(0), flags(0)
+ userBase(0), serverBase(0), buffers(NULL), frameCount(0),
+ loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), mVolumeLR(0x10001000),
+ mSendLevel(0), flags(0)
{
}
uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
{
- uint32_t u = this->user;
+ uint32_t u = user;
u += frameCount;
// Ensure that user is never ahead of server for AudioRecord
@@ -1324,16 +1388,16 @@
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
}
- } else if (u > this->server) {
- ALOGW("stepServer occured after track reset");
- u = this->server;
+ } else if (u > server) {
+ ALOGW("stepServer occurred after track reset");
+ u = server;
}
if (u >= userBase + this->frameCount) {
userBase += this->frameCount;
}
- this->user = u;
+ user = u;
// Clear flow control error condition as new data has been written/read to/from buffer.
if (flags & CBLK_UNDERRUN_MSK) {
@@ -1350,7 +1414,7 @@
return false;
}
- uint32_t s = this->server;
+ uint32_t s = server;
s += frameCount;
if (flags & CBLK_DIRECTION_MSK) {
@@ -1363,9 +1427,9 @@
// while the mixer is processing a block: in this case,
// stepServer() is called After the flush() has reset u & s and
// we have s > u
- if (s > this->user) {
- ALOGW("stepServer occured after track reset");
- s = this->user;
+ if (s > user) {
+ ALOGW("stepServer occurred after track reset");
+ s = user;
}
}
@@ -1381,7 +1445,7 @@
serverBase += this->frameCount;
}
- this->server = s;
+ server = s;
if (!(flags & CBLK_INVALID_MSK)) {
cv.signal();
@@ -1392,7 +1456,7 @@
void* audio_track_cblk_t::buffer(uint32_t offset) const
{
- return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
+ return (int8_t *)buffers + (offset - userBase) * frameSize;
}
uint32_t audio_track_cblk_t::framesAvailable()
@@ -1403,8 +1467,8 @@
uint32_t audio_track_cblk_t::framesAvailable_l()
{
- uint32_t u = this->user;
- uint32_t s = this->server;
+ uint32_t u = user;
+ uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
uint32_t limit = (s < loopStart) ? s : loopStart;
@@ -1416,8 +1480,8 @@
uint32_t audio_track_cblk_t::framesReady()
{
- uint32_t u = this->user;
- uint32_t s = this->server;
+ uint32_t u = user;
+ uint32_t s = server;
if (flags & CBLK_DIRECTION_MSK) {
if (u < loopEnd) {
@@ -1462,4 +1526,3 @@
// -------------------------------------------------------------------------
}; // namespace android
-
diff --git a/media/libmedia/IAudioFlinger.cpp b/media/libmedia/IAudioFlinger.cpp
index abd491f..ebadbfa 100644
--- a/media/libmedia/IAudioFlinger.cpp
+++ b/media/libmedia/IAudioFlinger.cpp
@@ -82,14 +82,15 @@
virtual sp<IAudioTrack> createTrack(
pid_t pid,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
+ bool isTimed,
int *sessionId,
status_t *status)
{
@@ -97,14 +98,15 @@
sp<IAudioTrack> track;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(pid);
- data.writeInt32(streamType);
+ data.writeInt32((int32_t) streamType);
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
data.writeInt32(frameCount);
data.writeInt32(flags);
data.writeStrongBinder(sharedBuffer->asBinder());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
+ data.writeInt32(isTimed);
int lSessionId = 0;
if (sessionId != NULL) {
lSessionId = *sessionId;
@@ -129,9 +131,9 @@
virtual sp<IAudioRecord> openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
@@ -142,7 +144,7 @@
sp<IAudioRecord> record;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(pid);
- data.writeInt32(input);
+ data.writeInt32((int32_t) input);
data.writeInt32(sampleRate);
data.writeInt32(format);
data.writeInt32(channelMask);
@@ -170,47 +172,47 @@
return record;
}
- virtual uint32_t sampleRate(int output) const
+ virtual uint32_t sampleRate(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SAMPLE_RATE, data, &reply);
return reply.readInt32();
}
- virtual int channelCount(int output) const
+ virtual int channelCount(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(CHANNEL_COUNT, data, &reply);
return reply.readInt32();
}
- virtual uint32_t format(int output) const
+ virtual audio_format_t format(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(FORMAT, data, &reply);
- return reply.readInt32();
+ return (audio_format_t) reply.readInt32();
}
- virtual size_t frameCount(int output) const
+ virtual size_t frameCount(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(FRAME_COUNT, data, &reply);
return reply.readInt32();
}
- virtual uint32_t latency(int output) const
+ virtual uint32_t latency(audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(LATENCY, data, &reply);
return reply.readInt32();
}
@@ -249,47 +251,48 @@
return reply.readInt32();
}
- virtual status_t setStreamVolume(int stream, float value, int output)
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeFloat(value);
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SET_STREAM_VOLUME, data, &reply);
return reply.readInt32();
}
- virtual status_t setStreamMute(int stream, bool muted)
+ virtual status_t setStreamMute(audio_stream_type_t stream, bool muted)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(muted);
remote()->transact(SET_STREAM_MUTE, data, &reply);
return reply.readInt32();
}
- virtual float streamVolume(int stream, int output) const
+ virtual float streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
- data.writeInt32(output);
+ data.writeInt32((int32_t) stream);
+ data.writeInt32((int32_t) output);
remote()->transact(STREAM_VOLUME, data, &reply);
return reply.readFloat();
}
- virtual bool streamMute(int stream) const
+ virtual bool streamMute(audio_stream_type_t stream) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
remote()->transact(STREAM_MUTE, data, &reply);
return reply.readInt32();
}
- virtual status_t setMode(int mode)
+ virtual status_t setMode(audio_mode_t mode)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -315,21 +318,21 @@
return reply.readInt32();
}
- virtual status_t setParameters(int ioHandle, const String8& keyValuePairs)
+ virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
data.writeString8(keyValuePairs);
remote()->transact(SET_PARAMETERS, data, &reply);
return reply.readInt32();
}
- virtual String8 getParameters(int ioHandle, const String8& keys)
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
data.writeString8(keys);
remote()->transact(GET_PARAMETERS, data, &reply);
return reply.readString8();
@@ -343,7 +346,7 @@
remote()->transact(REGISTER_CLIENT, data, &reply);
}
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -354,9 +357,9 @@
return reply.readInt32();
}
- virtual int openOutput(uint32_t *pDevices,
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags)
@@ -364,7 +367,7 @@
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
@@ -376,13 +379,13 @@
data.writeInt32(latency);
data.writeInt32(flags);
remote()->transact(OPEN_OUTPUT, data, &reply);
- int output = reply.readInt32();
- ALOGV("openOutput() returned output, %p", output);
+ audio_io_handle_t output = (audio_io_handle_t) reply.readInt32();
+ ALOGV("openOutput() returned output, %d", output);
devices = reply.readInt32();
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
if (pSamplingRate) *pSamplingRate = samplingRate;
- format = reply.readInt32();
+ format = (audio_format_t) reply.readInt32();
if (pFormat) *pFormat = format;
channels = reply.readInt32();
if (pChannels) *pChannels = channels;
@@ -391,53 +394,54 @@
return output;
}
- virtual int openDuplicateOutput(int output1, int output2)
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output1);
- data.writeInt32(output2);
+ data.writeInt32((int32_t) output1);
+ data.writeInt32((int32_t) output2);
remote()->transact(OPEN_DUPLICATE_OUTPUT, data, &reply);
- return reply.readInt32();
+ return (audio_io_handle_t) reply.readInt32();
}
- virtual status_t closeOutput(int output)
+ virtual status_t closeOutput(audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(CLOSE_OUTPUT, data, &reply);
return reply.readInt32();
}
- virtual status_t suspendOutput(int output)
+ virtual status_t suspendOutput(audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(SUSPEND_OUTPUT, data, &reply);
return reply.readInt32();
}
- virtual status_t restoreOutput(int output)
+ virtual status_t restoreOutput(audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(RESTORE_OUTPUT, data, &reply);
return reply.readInt32();
}
- virtual int openInput(uint32_t *pDevices,
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
Parcel data, reply;
uint32_t devices = pDevices ? *pDevices : 0;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -445,14 +449,14 @@
data.writeInt32(samplingRate);
data.writeInt32(format);
data.writeInt32(channels);
- data.writeInt32(acoustics);
+ data.writeInt32((int32_t) acoustics);
remote()->transact(OPEN_INPUT, data, &reply);
- int input = reply.readInt32();
+ audio_io_handle_t input = (audio_io_handle_t) reply.readInt32();
devices = reply.readInt32();
if (pDevices) *pDevices = devices;
samplingRate = reply.readInt32();
if (pSamplingRate) *pSamplingRate = samplingRate;
- format = reply.readInt32();
+ format = (audio_format_t) reply.readInt32();
if (pFormat) *pFormat = format;
channels = reply.readInt32();
if (pChannels) *pChannels = channels;
@@ -468,12 +472,12 @@
return reply.readInt32();
}
- virtual status_t setStreamOutput(uint32_t stream, int output)
+ virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(stream);
- data.writeInt32(output);
+ data.writeInt32((int32_t) stream);
+ data.writeInt32((int32_t) output);
remote()->transact(SET_STREAM_OUTPUT, data, &reply);
return reply.readInt32();
}
@@ -487,11 +491,12 @@
return reply.readInt32();
}
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
remote()->transact(GET_RENDER_POSITION, data, &reply);
status_t status = reply.readInt32();
if (status == NO_ERROR) {
@@ -507,11 +512,11 @@
return status;
}
- virtual unsigned int getInputFramesLost(int ioHandle)
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
remote()->transact(GET_INPUT_FRAMES_LOST, data, &reply);
return reply.readInt32();
}
@@ -544,7 +549,7 @@
remote()->transact(RELEASE_AUDIO_SESSION_ID, data, &reply);
}
- virtual status_t queryNumberEffects(uint32_t *numEffects)
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -556,13 +561,13 @@
if (status != NO_ERROR) {
return status;
}
- if (numEffects) {
+ if (numEffects != NULL) {
*numEffects = (uint32_t)reply.readInt32();
}
return NO_ERROR;
}
- virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor)
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const
{
if (pDescriptor == NULL) {
return BAD_VALUE;
@@ -582,7 +587,8 @@
return NO_ERROR;
}
- virtual status_t getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *pDescriptor)
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
+ effect_descriptor_t *pDescriptor) const
{
if (pUuid == NULL || pDescriptor == NULL) {
return BAD_VALUE;
@@ -606,7 +612,7 @@
effect_descriptor_t *pDesc,
const sp<IEffectClient>& client,
int32_t priority,
- int output,
+ audio_io_handle_t output,
int sessionId,
status_t *status,
int *id,
@@ -627,7 +633,7 @@
data.write(pDesc, sizeof(effect_descriptor_t));
data.writeStrongBinder(client->asBinder());
data.writeInt32(priority);
- data.writeInt32(output);
+ data.writeInt32((int32_t) output);
data.writeInt32(sessionId);
status_t lStatus = remote()->transact(CREATE_EFFECT, data, &reply);
@@ -640,7 +646,7 @@
*id = tmp;
}
tmp = reply.readInt32();
- if (enabled) {
+ if (enabled != NULL) {
*enabled = tmp;
}
effect = interface_cast<IEffect>(reply.readStrongBinder());
@@ -653,13 +659,14 @@
return effect;
}
- virtual status_t moveEffects(int session, int srcOutput, int dstOutput)
+ virtual status_t moveEffects(int session, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
data.writeInt32(session);
- data.writeInt32(srcOutput);
- data.writeInt32(dstOutput);
+ data.writeInt32((int32_t) srcOutput);
+ data.writeInt32((int32_t) dstOutput);
remote()->transact(MOVE_EFFECTS, data, &reply);
return reply.readInt32();
}
@@ -678,17 +685,18 @@
pid_t pid = data.readInt32();
int streamType = data.readInt32();
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
size_t bufferCount = data.readInt32();
uint32_t flags = data.readInt32();
sp<IMemory> buffer = interface_cast<IMemory>(data.readStrongBinder());
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
+ bool isTimed = data.readInt32();
int sessionId = data.readInt32();
status_t status;
sp<IAudioTrack> track = createTrack(pid,
- streamType, sampleRate, format,
- channelCount, bufferCount, flags, buffer, output, &sessionId, &status);
+ (audio_stream_type_t) streamType, sampleRate, format,
+ channelCount, bufferCount, flags, buffer, output, isTimed, &sessionId, &status);
reply->writeInt32(sessionId);
reply->writeInt32(status);
reply->writeStrongBinder(track->asBinder());
@@ -697,9 +705,9 @@
case OPEN_RECORD: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
pid_t pid = data.readInt32();
- int input = data.readInt32();
+ audio_io_handle_t input = (audio_io_handle_t) data.readInt32();
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
size_t bufferCount = data.readInt32();
uint32_t flags = data.readInt32();
@@ -714,27 +722,27 @@
} break;
case SAMPLE_RATE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( sampleRate(data.readInt32()) );
+ reply->writeInt32( sampleRate((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case CHANNEL_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( channelCount(data.readInt32()) );
+ reply->writeInt32( channelCount((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case FORMAT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( format(data.readInt32()) );
+ reply->writeInt32( format((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case FRAME_COUNT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( frameCount(data.readInt32()) );
+ reply->writeInt32( frameCount((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case LATENCY: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32( latency(data.readInt32()) );
+ reply->writeInt32( latency((audio_io_handle_t) data.readInt32()) );
return NO_ERROR;
} break;
case SET_MASTER_VOLUME: {
@@ -761,32 +769,32 @@
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
float volume = data.readFloat();
- int output = data.readInt32();
- reply->writeInt32( setStreamVolume(stream, volume, output) );
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
+ reply->writeInt32( setStreamVolume((audio_stream_type_t) stream, volume, output) );
return NO_ERROR;
} break;
case SET_STREAM_MUTE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
- reply->writeInt32( setStreamMute(stream, data.readInt32()) );
+ reply->writeInt32( setStreamMute((audio_stream_type_t) stream, data.readInt32()) );
return NO_ERROR;
} break;
case STREAM_VOLUME: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
int output = data.readInt32();
- reply->writeFloat( streamVolume(stream, output) );
+ reply->writeFloat( streamVolume((audio_stream_type_t) stream, output) );
return NO_ERROR;
} break;
case STREAM_MUTE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int stream = data.readInt32();
- reply->writeInt32( streamMute(stream) );
+ reply->writeInt32( streamMute((audio_stream_type_t) stream) );
return NO_ERROR;
} break;
case SET_MODE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int mode = data.readInt32();
+ audio_mode_t mode = (audio_mode_t) data.readInt32();
reply->writeInt32( setMode(mode) );
return NO_ERROR;
} break;
@@ -803,14 +811,14 @@
} break;
case SET_PARAMETERS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
String8 keyValuePairs(data.readString8());
reply->writeInt32(setParameters(ioHandle, keyValuePairs));
return NO_ERROR;
} break;
case GET_PARAMETERS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
String8 keys(data.readString8());
reply->writeString8(getParameters(ioHandle, keys));
return NO_ERROR;
@@ -825,7 +833,7 @@
case GET_INPUTBUFFERSIZE: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t sampleRate = data.readInt32();
- int format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
int channelCount = data.readInt32();
reply->writeInt32( getInputBufferSize(sampleRate, format, channelCount) );
return NO_ERROR;
@@ -834,18 +842,18 @@
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t devices = data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
uint32_t latency = data.readInt32();
uint32_t flags = data.readInt32();
- int output = openOutput(&devices,
+ audio_io_handle_t output = openOutput(&devices,
&samplingRate,
&format,
&channels,
&latency,
flags);
ALOGV("OPEN_OUTPUT output, %p", output);
- reply->writeInt32(output);
+ reply->writeInt32((int32_t) output);
reply->writeInt32(devices);
reply->writeInt32(samplingRate);
reply->writeInt32(format);
@@ -855,40 +863,40 @@
} break;
case OPEN_DUPLICATE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int output1 = data.readInt32();
- int output2 = data.readInt32();
- reply->writeInt32(openDuplicateOutput(output1, output2));
+ audio_io_handle_t output1 = (audio_io_handle_t) data.readInt32();
+ audio_io_handle_t output2 = (audio_io_handle_t) data.readInt32();
+ reply->writeInt32((int32_t) openDuplicateOutput(output1, output2));
return NO_ERROR;
} break;
case CLOSE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(closeOutput(data.readInt32()));
+ reply->writeInt32(closeOutput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case SUSPEND_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(suspendOutput(data.readInt32()));
+ reply->writeInt32(suspendOutput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case RESTORE_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(restoreOutput(data.readInt32()));
+ reply->writeInt32(restoreOutput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case OPEN_INPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t devices = data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
- uint32_t acoutics = data.readInt32();
+ audio_in_acoustics_t acoustics = (audio_in_acoustics_t) data.readInt32();
- int input = openInput(&devices,
+ audio_io_handle_t input = openInput(&devices,
&samplingRate,
&format,
&channels,
- acoutics);
- reply->writeInt32(input);
+ acoustics);
+ reply->writeInt32((int32_t) input);
reply->writeInt32(devices);
reply->writeInt32(samplingRate);
reply->writeInt32(format);
@@ -897,14 +905,14 @@
} break;
case CLOSE_INPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- reply->writeInt32(closeInput(data.readInt32()));
+ reply->writeInt32(closeInput((audio_io_handle_t) data.readInt32()));
return NO_ERROR;
} break;
case SET_STREAM_OUTPUT: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
uint32_t stream = data.readInt32();
- int output = data.readInt32();
- reply->writeInt32(setStreamOutput(stream, output));
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
+ reply->writeInt32(setStreamOutput((audio_stream_type_t) stream, output));
return NO_ERROR;
} break;
case SET_VOICE_VOLUME: {
@@ -915,7 +923,7 @@
} break;
case GET_RENDER_POSITION: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
uint32_t halFrames;
uint32_t dspFrames;
status_t status = getRenderPosition(&halFrames, &dspFrames, output);
@@ -928,7 +936,7 @@
}
case GET_INPUT_FRAMES_LOST: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
- int ioHandle = data.readInt32();
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
reply->writeInt32(getInputFramesLost(ioHandle));
return NO_ERROR;
} break;
@@ -988,7 +996,7 @@
data.read(&desc, sizeof(effect_descriptor_t));
sp<IEffectClient> client = interface_cast<IEffectClient>(data.readStrongBinder());
int32_t priority = data.readInt32();
- int output = data.readInt32();
+ audio_io_handle_t output = (audio_io_handle_t) data.readInt32();
int sessionId = data.readInt32();
status_t status;
int id;
@@ -1005,8 +1013,8 @@
case MOVE_EFFECTS: {
CHECK_INTERFACE(IAudioFlinger, data, reply);
int session = data.readInt32();
- int srcOutput = data.readInt32();
- int dstOutput = data.readInt32();
+ audio_io_handle_t srcOutput = (audio_io_handle_t) data.readInt32();
+ audio_io_handle_t dstOutput = (audio_io_handle_t) data.readInt32();
reply->writeInt32(moveEffects(session, srcOutput, dstOutput));
return NO_ERROR;
} break;
diff --git a/media/libmedia/IAudioFlingerClient.cpp b/media/libmedia/IAudioFlingerClient.cpp
index 5a3f250..ce28b33 100644
--- a/media/libmedia/IAudioFlingerClient.cpp
+++ b/media/libmedia/IAudioFlingerClient.cpp
@@ -39,12 +39,12 @@
{
}
- void ioConfigChanged(int event, int ioHandle, void *param2)
+ void ioConfigChanged(int event, audio_io_handle_t ioHandle, void *param2)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioFlingerClient::getInterfaceDescriptor());
data.writeInt32(event);
- data.writeInt32(ioHandle);
+ data.writeInt32((int32_t) ioHandle);
if (event == AudioSystem::STREAM_CONFIG_CHANGED) {
uint32_t stream = *(uint32_t *)param2;
ALOGV("ioConfigChanged stream %d", stream);
@@ -72,8 +72,8 @@
case IO_CONFIG_CHANGED: {
CHECK_INTERFACE(IAudioFlingerClient, data, reply);
int event = data.readInt32();
- int ioHandle = data.readInt32();
- void *param2 = 0;
+ audio_io_handle_t ioHandle = (audio_io_handle_t) data.readInt32();
+ void *param2 = NULL;
AudioSystem::OutputDescriptor desc;
uint32_t stream;
if (event == AudioSystem::STREAM_CONFIG_CHANGED) {
diff --git a/media/libmedia/IAudioPolicyService.cpp b/media/libmedia/IAudioPolicyService.cpp
index 50b4855..99385aa 100644
--- a/media/libmedia/IAudioPolicyService.cpp
+++ b/media/libmedia/IAudioPolicyService.cpp
@@ -33,7 +33,7 @@
SET_DEVICE_CONNECTION_STATE = IBinder::FIRST_CALL_TRANSACTION,
GET_DEVICE_CONNECTION_STATE,
SET_PHONE_STATE,
- SET_RINGER_MODE,
+ SET_RINGER_MODE, // reserved, no longer used
SET_FORCE_USE,
GET_FORCE_USE,
GET_OUTPUT,
@@ -91,7 +91,7 @@
return static_cast <audio_policy_dev_state_t>(reply.readInt32());
}
- virtual status_t setPhoneState(int state)
+ virtual status_t setPhoneState(audio_mode_t state)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
@@ -100,16 +100,6 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t setRingerMode(uint32_t mode, uint32_t mask)
- {
- Parcel data, reply;
- data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(mode);
- data.writeInt32(mask);
- remote()->transact(SET_RINGER_MODE, data, &reply);
- return static_cast <status_t> (reply.readInt32());
- }
-
virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config)
{
Parcel data, reply;
@@ -132,7 +122,7 @@
virtual audio_io_handle_t getOutput(
audio_stream_type_t stream,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_policy_output_flags_t flags)
{
@@ -154,7 +144,7 @@
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(output);
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(session);
remote()->transact(START_OUTPUT, data, &reply);
return static_cast <status_t> (reply.readInt32());
@@ -167,7 +157,7 @@
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(output);
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(session);
remote()->transact(STOP_OUTPUT, data, &reply);
return static_cast <status_t> (reply.readInt32());
@@ -182,16 +172,16 @@
}
virtual audio_io_handle_t getInput(
- int inputSource,
+ audio_source_t inputSource,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_in_acoustics_t acoustics,
int audioSession)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(inputSource);
+ data.writeInt32((int32_t) inputSource);
data.writeInt32(samplingRate);
data.writeInt32(static_cast <uint32_t>(format));
data.writeInt32(channels);
@@ -240,21 +230,28 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index)
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(stream));
data.writeInt32(index);
+ data.writeInt32(static_cast <uint32_t>(device));
remote()->transact(SET_STREAM_VOLUME, data, &reply);
return static_cast <status_t> (reply.readInt32());
}
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index)
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
data.writeInt32(static_cast <uint32_t>(stream));
+ data.writeInt32(static_cast <uint32_t>(device));
+
remote()->transact(GET_STREAM_VOLUME, data, &reply);
int lIndex = reply.readInt32();
if (index) *index = lIndex;
@@ -324,11 +321,11 @@
return static_cast <status_t> (reply.readInt32());
}
- virtual bool isStreamActive(int stream, uint32_t inPastMs) const
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
{
Parcel data, reply;
data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
- data.writeInt32(stream);
+ data.writeInt32((int32_t) stream);
data.writeInt32(inPastMs);
remote()->transact(IS_STREAM_ACTIVE, data, &reply);
return reply.readInt32();
@@ -394,15 +391,7 @@
case SET_PHONE_STATE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- reply->writeInt32(static_cast <uint32_t>(setPhoneState(data.readInt32())));
- return NO_ERROR;
- } break;
-
- case SET_RINGER_MODE: {
- CHECK_INTERFACE(IAudioPolicyService, data, reply);
- uint32_t mode = data.readInt32();
- uint32_t mask = data.readInt32();
- reply->writeInt32(static_cast <uint32_t>(setRingerMode(mode, mask)));
+ reply->writeInt32(static_cast <uint32_t>(setPhoneState((audio_mode_t) data.readInt32())));
return NO_ERROR;
} break;
@@ -427,7 +416,7 @@
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
audio_policy_output_flags_t flags =
static_cast <audio_policy_output_flags_t>(data.readInt32());
@@ -472,9 +461,9 @@
case GET_INPUT: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- int inputSource = data.readInt32();
+ audio_source_t inputSource = (audio_source_t) data.readInt32();
uint32_t samplingRate = data.readInt32();
- uint32_t format = data.readInt32();
+ audio_format_t format = (audio_format_t) data.readInt32();
uint32_t channels = data.readInt32();
audio_in_acoustics_t acoustics =
static_cast <audio_in_acoustics_t>(data.readInt32());
@@ -525,7 +514,10 @@
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
int index = data.readInt32();
- reply->writeInt32(static_cast <uint32_t>(setStreamVolumeIndex(stream, index)));
+ audio_devices_t device = static_cast <audio_devices_t>(data.readInt32());
+ reply->writeInt32(static_cast <uint32_t>(setStreamVolumeIndex(stream,
+ index,
+ device)));
return NO_ERROR;
} break;
@@ -533,8 +525,9 @@
CHECK_INTERFACE(IAudioPolicyService, data, reply);
audio_stream_type_t stream =
static_cast <audio_stream_type_t>(data.readInt32());
+ audio_devices_t device = static_cast <audio_devices_t>(data.readInt32());
int index;
- status_t status = getStreamVolumeIndex(stream, &index);
+ status_t status = getStreamVolumeIndex(stream, &index, device);
reply->writeInt32(index);
reply->writeInt32(static_cast <uint32_t>(status));
return NO_ERROR;
@@ -598,9 +591,9 @@
case IS_STREAM_ACTIVE: {
CHECK_INTERFACE(IAudioPolicyService, data, reply);
- int stream = data.readInt32();
+ audio_stream_type_t stream = (audio_stream_type_t) data.readInt32();
uint32_t inPastMs = (uint32_t)data.readInt32();
- reply->writeInt32( isStreamActive(stream, inPastMs) );
+ reply->writeInt32( isStreamActive((audio_stream_type_t) stream, inPastMs) );
return NO_ERROR;
} break;
diff --git a/media/libmedia/IAudioRecord.cpp b/media/libmedia/IAudioRecord.cpp
index 8c7a960..6b473c9 100644
--- a/media/libmedia/IAudioRecord.cpp
+++ b/media/libmedia/IAudioRecord.cpp
@@ -42,10 +42,11 @@
{
}
- virtual status_t start()
+ virtual status_t start(pid_t tid)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioRecord::getInterfaceDescriptor());
+ data.writeInt32(tid);
status_t status = remote()->transact(START, data, &reply);
if (status == NO_ERROR) {
status = reply.readInt32();
@@ -90,7 +91,7 @@
} break;
case START: {
CHECK_INTERFACE(IAudioRecord, data, reply);
- reply->writeInt32(start());
+ reply->writeInt32(start(data.readInt32()));
return NO_ERROR;
} break;
case STOP: {
diff --git a/media/libmedia/IAudioTrack.cpp b/media/libmedia/IAudioTrack.cpp
index 0b372f3..28ebbbf 100644
--- a/media/libmedia/IAudioTrack.cpp
+++ b/media/libmedia/IAudioTrack.cpp
@@ -1,4 +1,4 @@
-/* //device/extlibs/pv/android/IAudioTrack.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -35,7 +35,10 @@
FLUSH,
MUTE,
PAUSE,
- ATTACH_AUX_EFFECT
+ ATTACH_AUX_EFFECT,
+ ALLOCATE_TIMED_BUFFER,
+ QUEUE_TIMED_BUFFER,
+ SET_MEDIA_TIME_TRANSFORM,
};
class BpAudioTrack : public BpInterface<IAudioTrack>
@@ -46,10 +49,23 @@
{
}
- virtual status_t start()
+ virtual sp<IMemory> getCblk() const
+ {
+ Parcel data, reply;
+ sp<IMemory> cblk;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ status_t status = remote()->transact(GET_CBLK, data, &reply);
+ if (status == NO_ERROR) {
+ cblk = interface_cast<IMemory>(reply.readStrongBinder());
+ }
+ return cblk;
+ }
+
+ virtual status_t start(pid_t tid)
{
Parcel data, reply;
data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeInt32(tid);
status_t status = remote()->transact(START, data, &reply);
if (status == NO_ERROR) {
status = reply.readInt32();
@@ -88,18 +104,6 @@
remote()->transact(PAUSE, data, &reply);
}
- virtual sp<IMemory> getCblk() const
- {
- Parcel data, reply;
- sp<IMemory> cblk;
- data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
- status_t status = remote()->transact(GET_CBLK, data, &reply);
- if (status == NO_ERROR) {
- cblk = interface_cast<IMemory>(reply.readStrongBinder());
- }
- return cblk;
- }
-
virtual status_t attachAuxEffect(int effectId)
{
Parcel data, reply;
@@ -113,6 +117,52 @@
}
return status;
}
+
+ virtual status_t allocateTimedBuffer(size_t size, sp<IMemory>* buffer) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeInt32(size);
+ status_t status = remote()->transact(ALLOCATE_TIMED_BUFFER,
+ data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ if (status == NO_ERROR) {
+ *buffer = interface_cast<IMemory>(reply.readStrongBinder());
+ }
+ }
+ return status;
+ }
+
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeStrongBinder(buffer->asBinder());
+ data.writeInt64(pts);
+ status_t status = remote()->transact(QUEUE_TIMED_BUFFER,
+ data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ }
+ return status;
+ }
+
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target) {
+ Parcel data, reply;
+ data.writeInterfaceToken(IAudioTrack::getInterfaceDescriptor());
+ data.writeInt64(xform.a_zero);
+ data.writeInt64(xform.b_zero);
+ data.writeInt32(xform.a_to_b_numer);
+ data.writeInt32(xform.a_to_b_denom);
+ data.writeInt32(target);
+ status_t status = remote()->transact(SET_MEDIA_TIME_TRANSFORM,
+ data, &reply);
+ if (status == NO_ERROR) {
+ status = reply.readInt32();
+ }
+ return status;
+ }
};
IMPLEMENT_META_INTERFACE(AudioTrack, "android.media.IAudioTrack");
@@ -130,7 +180,7 @@
} break;
case START: {
CHECK_INTERFACE(IAudioTrack, data, reply);
- reply->writeInt32(start());
+ reply->writeInt32(start(data.readInt32()));
return NO_ERROR;
} break;
case STOP: {
@@ -158,10 +208,38 @@
reply->writeInt32(attachAuxEffect(data.readInt32()));
return NO_ERROR;
} break;
+ case ALLOCATE_TIMED_BUFFER: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ sp<IMemory> buffer;
+ status_t status = allocateTimedBuffer(data.readInt32(), &buffer);
+ reply->writeInt32(status);
+ if (status == NO_ERROR) {
+ reply->writeStrongBinder(buffer->asBinder());
+ }
+ return NO_ERROR;
+ } break;
+ case QUEUE_TIMED_BUFFER: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ sp<IMemory> buffer = interface_cast<IMemory>(
+ data.readStrongBinder());
+ uint64_t pts = data.readInt64();
+ reply->writeInt32(queueTimedBuffer(buffer, pts));
+ return NO_ERROR;
+ } break;
+ case SET_MEDIA_TIME_TRANSFORM: {
+ CHECK_INTERFACE(IAudioTrack, data, reply);
+ LinearTransform xform;
+ xform.a_zero = data.readInt64();
+ xform.b_zero = data.readInt64();
+ xform.a_to_b_numer = data.readInt32();
+ xform.a_to_b_denom = data.readInt32();
+ int target = data.readInt32();
+ reply->writeInt32(setMediaTimeTransform(xform, target));
+ return NO_ERROR;
+ } break;
default:
return BBinder::onTransact(code, data, reply, flags);
}
}
}; // namespace android
-
diff --git a/media/libmedia/IEffect.cpp b/media/libmedia/IEffect.cpp
index d469e28..5d40cc8 100644
--- a/media/libmedia/IEffect.cpp
+++ b/media/libmedia/IEffect.cpp
@@ -83,8 +83,15 @@
size = *pReplySize;
}
data.writeInt32(size);
- remote()->transact(COMMAND, data, &reply);
- status_t status = reply.readInt32();
+
+ status_t status = remote()->transact(COMMAND, data, &reply);
+ if (status != NO_ERROR) {
+ if (pReplySize != NULL)
+ *pReplySize = 0;
+ return status;
+ }
+
+ status = reply.readInt32();
size = reply.readInt32();
if (size != 0 && pReplyData != NULL && pReplySize != NULL) {
reply.read(pReplyData, size);
diff --git a/media/libmedia/IMediaDeathNotifier.cpp b/media/libmedia/IMediaDeathNotifier.cpp
index 8525482..aeb35a5 100644
--- a/media/libmedia/IMediaDeathNotifier.cpp
+++ b/media/libmedia/IMediaDeathNotifier.cpp
@@ -36,7 +36,7 @@
{
ALOGV("getMediaPlayerService");
Mutex::Autolock _l(sServiceLock);
- if (sMediaPlayerService.get() == 0) {
+ if (sMediaPlayerService == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
diff --git a/media/libmedia/IMediaPlayer.cpp b/media/libmedia/IMediaPlayer.cpp
index 9c1e6b7..86d65db 100644
--- a/media/libmedia/IMediaPlayer.cpp
+++ b/media/libmedia/IMediaPlayer.cpp
@@ -23,8 +23,6 @@
#include <media/IMediaPlayer.h>
#include <media/IStreamSource.h>
-#include <surfaceflinger/ISurface.h>
-#include <surfaceflinger/Surface.h>
#include <gui/ISurfaceTexture.h>
#include <utils/String8.h>
@@ -198,11 +196,11 @@
return reply.readInt32();
}
- status_t setAudioStreamType(int type)
+ status_t setAudioStreamType(audio_stream_type_t stream)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayer::getInterfaceDescriptor());
- data.writeInt32(type);
+ data.writeInt32((int32_t) stream);
remote()->transact(SET_AUDIO_STREAM_TYPE, data, &reply);
return reply.readInt32();
}
@@ -397,7 +395,7 @@
} break;
case SET_AUDIO_STREAM_TYPE: {
CHECK_INTERFACE(IMediaPlayer, data, reply);
- reply->writeInt32(setAudioStreamType(data.readInt32()));
+ reply->writeInt32(setAudioStreamType((audio_stream_type_t) data.readInt32()));
return NO_ERROR;
} break;
case SET_LOOPING: {
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index 8e4dd04..f5b5cbd 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -78,7 +78,7 @@
return interface_cast<IMediaRecorder>(reply.readStrongBinder());
}
- virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+ virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
@@ -86,11 +86,11 @@
remote()->transact(DECODE_URL, data, &reply);
*pSampleRate = uint32_t(reply.readInt32());
*pNumChannels = reply.readInt32();
- *pFormat = reply.readInt32();
+ *pFormat = (audio_format_t) reply.readInt32();
return interface_cast<IMemory>(reply.readStrongBinder());
}
- virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+ virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
Parcel data, reply;
data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
@@ -100,7 +100,7 @@
remote()->transact(DECODE_FD, data, &reply);
*pSampleRate = uint32_t(reply.readInt32());
*pNumChannels = reply.readInt32();
- *pFormat = reply.readInt32();
+ *pFormat = (audio_format_t) reply.readInt32();
return interface_cast<IMemory>(reply.readStrongBinder());
}
@@ -148,11 +148,11 @@
const char* url = data.readCString();
uint32_t sampleRate;
int numChannels;
- int format;
+ audio_format_t format;
sp<IMemory> player = decode(url, &sampleRate, &numChannels, &format);
reply->writeInt32(sampleRate);
reply->writeInt32(numChannels);
- reply->writeInt32(format);
+ reply->writeInt32((int32_t) format);
reply->writeStrongBinder(player->asBinder());
return NO_ERROR;
} break;
@@ -163,11 +163,11 @@
int64_t length = data.readInt64();
uint32_t sampleRate;
int numChannels;
- int format;
+ audio_format_t format;
sp<IMemory> player = decode(fd, offset, length, &sampleRate, &numChannels, &format);
reply->writeInt32(sampleRate);
reply->writeInt32(numChannels);
- reply->writeInt32(format);
+ reply->writeInt32((int32_t) format);
reply->writeStrongBinder(player->asBinder());
return NO_ERROR;
} break;
diff --git a/media/libmedia/IMediaRecorder.cpp b/media/libmedia/IMediaRecorder.cpp
index 42f55c2..2f4e31a 100644
--- a/media/libmedia/IMediaRecorder.cpp
+++ b/media/libmedia/IMediaRecorder.cpp
@@ -19,10 +19,10 @@
#define LOG_TAG "IMediaRecorder"
#include <utils/Log.h>
#include <binder/Parcel.h>
-#include <surfaceflinger/Surface.h>
#include <camera/ICamera.h>
#include <media/IMediaRecorderClient.h>
#include <media/IMediaRecorder.h>
+#include <gui/Surface.h>
#include <gui/ISurfaceTexture.h>
#include <unistd.h>
diff --git a/media/libmedia/IOMX.cpp b/media/libmedia/IOMX.cpp
index d2f5f71..48e427a 100644
--- a/media/libmedia/IOMX.cpp
+++ b/media/libmedia/IOMX.cpp
@@ -22,8 +22,6 @@
#include <binder/Parcel.h>
#include <media/IOMX.h>
#include <media/stagefright/foundation/ADebug.h>
-#include <surfaceflinger/ISurface.h>
-#include <surfaceflinger/Surface.h>
namespace android {
@@ -59,9 +57,10 @@
: BpInterface<IOMX>(impl) {
}
- virtual bool livesLocally(pid_t pid) {
+ virtual bool livesLocally(node_id node, pid_t pid) {
Parcel data, reply;
data.writeInterfaceToken(IOMX::getInterfaceDescriptor());
+ data.writeIntPtr((intptr_t)node);
data.writeInt32(pid);
remote()->transact(LIVES_LOCALLY, data, &reply);
@@ -417,7 +416,9 @@
case LIVES_LOCALLY:
{
CHECK_INTERFACE(IOMX, data, reply);
- reply->writeInt32(livesLocally((pid_t)data.readInt32()));
+ node_id node = (void *)data.readIntPtr();
+ pid_t pid = (pid_t)data.readInt32();
+ reply->writeInt32(livesLocally(node, pid));
return OK;
}
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
index fa5b67a..6cb5b82 100644
--- a/media/libmedia/JetPlayer.cpp
+++ b/media/libmedia/JetPlayer.cpp
@@ -91,7 +91,7 @@
mAudioTrack = new AudioTrack();
mAudioTrack->set(AUDIO_STREAM_MUSIC, //TODO parametrize this
pLibConfig->sampleRate,
- 1, // format = PCM 16bits per sample,
+ AUDIO_FORMAT_PCM_16_BIT,
(pLibConfig->numChannels == 2) ? AUDIO_CHANNEL_OUT_STEREO : AUDIO_CHANNEL_OUT_MONO,
mTrackBufferSize,
0);
@@ -100,7 +100,8 @@
{
Mutex::Autolock l(mMutex);
ALOGV("JetPlayer::init(): trying to start render thread");
- createThreadEtc(renderThread, this, "jetRenderThread", ANDROID_PRIORITY_AUDIO);
+ mThread = new JetPlayerThread(this);
+ mThread->run("jetRenderThread", ANDROID_PRIORITY_AUDIO);
mCondition.wait(mMutex);
}
if (mTid > 0) {
@@ -156,12 +157,6 @@
//-------------------------------------------------------------------------------------------------
-int JetPlayer::renderThread(void* p) {
-
- return ((JetPlayer*)p)->render();
-}
-
-//-------------------------------------------------------------------------------------------------
int JetPlayer::render() {
EAS_RESULT result = EAS_FAILURE;
EAS_I32 count;
@@ -173,10 +168,6 @@
// allocate render buffer
mAudioBuffer =
new EAS_PCM[pLibConfig->mixBufferSize * pLibConfig->numChannels * MIX_NUM_BUFFERS];
- if (!mAudioBuffer) {
- ALOGE("JetPlayer::render(): mAudioBuffer allocate failed");
- goto threadExit;
- }
// signal main thread that we started
{
@@ -255,14 +246,12 @@
}//while (1)
threadExit:
- if (mAudioTrack) {
+ if (mAudioTrack != NULL) {
mAudioTrack->stop();
mAudioTrack->flush();
}
- if (mAudioBuffer) {
- delete [] mAudioBuffer;
- mAudioBuffer = NULL;
- }
+ delete [] mAudioBuffer;
+ mAudioBuffer = NULL;
mMutex.lock();
mTid = -1;
mCondition.signal();
@@ -343,8 +332,8 @@
Mutex::Autolock lock(mMutex);
mEasJetFileLoc = (EAS_FILE_LOCATOR) malloc(sizeof(EAS_FILE));
- memset(mJetFilePath, 0, 256);
- strncpy(mJetFilePath, path, strlen(path));
+ strncpy(mJetFilePath, path, sizeof(mJetFilePath));
+ mJetFilePath[sizeof(mJetFilePath) - 1] = '\0';
mEasJetFileLoc->path = mJetFilePath;
mEasJetFileLoc->fd = 0;
diff --git a/media/libmedia/MediaProfiles.cpp b/media/libmedia/MediaProfiles.cpp
index c905762..93ddca8 100644
--- a/media/libmedia/MediaProfiles.cpp
+++ b/media/libmedia/MediaProfiles.cpp
@@ -25,7 +25,7 @@
#include <cutils/properties.h>
#include <expat.h>
#include <media/MediaProfiles.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/openmax/OMX_Video.h>
namespace android {
@@ -349,7 +349,7 @@
{
CHECK(!strcmp("quality", atts[0]));
int quality = atoi(atts[1]);
- ALOGV("%s: cameraId=%d, quality=%d\n", __func__, cameraId, quality);
+ ALOGV("%s: cameraId=%d, quality=%d", __func__, cameraId, quality);
ImageEncodingQualityLevels *levels = findImageEncodingQualityLevels(cameraId);
if (levels == NULL) {
diff --git a/media/libmedia/MediaScanner.cpp b/media/libmedia/MediaScanner.cpp
index 79cab74..73d4519 100644
--- a/media/libmedia/MediaScanner.cpp
+++ b/media/libmedia/MediaScanner.cpp
@@ -143,7 +143,7 @@
if (pathRemaining >= 8 /* strlen(".nomedia") */ ) {
strcpy(fileSpot, ".nomedia");
if (access(path, F_OK) == 0) {
- ALOGV("found .nomedia, setting noMedia flag\n");
+ ALOGV("found .nomedia, setting noMedia flag");
noMedia = true;
}
diff --git a/media/libmedia/MediaScannerClient.cpp b/media/libmedia/MediaScannerClient.cpp
index 40b8188..cdfd477 100644
--- a/media/libmedia/MediaScannerClient.cpp
+++ b/media/libmedia/MediaScannerClient.cpp
@@ -142,12 +142,12 @@
UConverter *conv = ucnv_open(enc, &status);
if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for %s\n", enc);
+ ALOGE("could not create UConverter for %s", enc);
return;
}
UConverter *utf8Conv = ucnv_open("UTF-8", &status);
if (U_FAILURE(status)) {
- ALOGE("could not create UConverter for UTF-8\n");
+ ALOGE("could not create UConverter for UTF-8");
ucnv_close(conv);
return;
}
@@ -173,6 +173,7 @@
const char* source = mValues->getEntry(i);
int targetLength = len * 3 + 1;
char* buffer = new char[targetLength];
+ // don't normally check for NULL, but in this case targetLength may be large
if (!buffer)
break;
char* target = buffer;
@@ -180,7 +181,7 @@
ucnv_convertEx(utf8Conv, conv, &target, target + targetLength,
&source, (const char *)dest, NULL, NULL, NULL, NULL, TRUE, TRUE, &status);
if (U_FAILURE(status)) {
- ALOGE("ucnv_convertEx failed: %d\n", status);
+ ALOGE("ucnv_convertEx failed: %d", status);
mValues->setEntry(i, "???");
} else {
// zero terminate
diff --git a/media/libmedia/ToneGenerator.cpp b/media/libmedia/ToneGenerator.cpp
index 35dfbb8..54eb98a 100644
--- a/media/libmedia/ToneGenerator.cpp
+++ b/media/libmedia/ToneGenerator.cpp
@@ -751,7 +751,7 @@
// Used by ToneGenerator::getToneForRegion() to convert user specified supervisory tone type
// to actual tone for current region.
-const unsigned char ToneGenerator::sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONES] = {
+const unsigned char /*tone_type*/ ToneGenerator::sToneMappingTable[NUM_REGIONS-1][NUM_SUP_TONES] = {
{ // ANSI
TONE_ANSI_DIAL, // TONE_SUP_DIAL
TONE_ANSI_BUSY, // TONE_SUP_BUSY
@@ -791,16 +791,16 @@
// generators, instantiates output audio track.
//
// Input:
-// streamType: Type of stream used for tone playback (enum AudioTrack::stream_type)
+// streamType: Type of stream used for tone playback
// volume: volume applied to tone (0.0 to 1.0)
//
// Output:
// none
//
////////////////////////////////////////////////////////////////////////////////
-ToneGenerator::ToneGenerator(int streamType, float volume, bool threadCanCallJava) {
+ToneGenerator::ToneGenerator(audio_stream_type_t streamType, float volume, bool threadCanCallJava) {
- ALOGV("ToneGenerator constructor: streamType=%d, volume=%f\n", streamType, volume);
+ ALOGV("ToneGenerator constructor: streamType=%d, volume=%f", streamType, volume);
mState = TONE_IDLE;
@@ -811,9 +811,9 @@
mThreadCanCallJava = threadCanCallJava;
mStreamType = streamType;
mVolume = volume;
- mpAudioTrack = 0;
- mpToneDesc = 0;
- mpNewToneDesc = 0;
+ mpAudioTrack = NULL;
+ mpToneDesc = NULL;
+ mpNewToneDesc = NULL;
// Generate tone by chunks of 20 ms to keep cadencing precision
mProcessSize = (mSamplingRate * 20) / 1000;
@@ -829,9 +829,9 @@
}
if (initAudioTrack()) {
- ALOGV("ToneGenerator INIT OK, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("ToneGenerator INIT OK, time: %d", (unsigned int)(systemTime()/1000000));
} else {
- ALOGV("!!!ToneGenerator INIT FAILED!!!\n");
+ ALOGV("!!!ToneGenerator INIT FAILED!!!");
}
}
@@ -853,11 +853,11 @@
//
////////////////////////////////////////////////////////////////////////////////
ToneGenerator::~ToneGenerator() {
- ALOGV("ToneGenerator destructor\n");
+ ALOGV("ToneGenerator destructor");
- if (mpAudioTrack) {
+ if (mpAudioTrack != NULL) {
stopTone();
- ALOGV("Delete Track: %p\n", mpAudioTrack);
+ ALOGV("Delete Track: %p", mpAudioTrack);
delete mpAudioTrack;
}
}
@@ -878,7 +878,7 @@
// none
//
////////////////////////////////////////////////////////////////////////////////
-bool ToneGenerator::startTone(int toneType, int durationMs) {
+bool ToneGenerator::startTone(tone_type toneType, int durationMs) {
bool lResult = false;
status_t lStatus;
@@ -892,7 +892,7 @@
}
}
- ALOGV("startTone\n");
+ ALOGV("startTone");
mLock.lock();
@@ -915,7 +915,7 @@
if (mState == TONE_INIT) {
if (prepareWave()) {
- ALOGV("Immediate start, time %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("Immediate start, time %d", (unsigned int)(systemTime()/1000000));
lResult = true;
mState = TONE_STARTING;
mLock.unlock();
@@ -934,7 +934,7 @@
mState = TONE_IDLE;
}
} else {
- ALOGV("Delayed start\n");
+ ALOGV("Delayed start");
mState = TONE_RESTARTING;
lStatus = mWaitCbkCond.waitRelative(mLock, seconds(3));
if (lStatus == NO_ERROR) {
@@ -949,8 +949,8 @@
}
mLock.unlock();
- ALOGV_IF(lResult, "Tone started, time %d\n", (unsigned int)(systemTime()/1000000));
- ALOGW_IF(!lResult, "Tone start failed!!!, time %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV_IF(lResult, "Tone started, time %d", (unsigned int)(systemTime()/1000000));
+ ALOGW_IF(!lResult, "Tone start failed!!!, time %d", (unsigned int)(systemTime()/1000000));
return lResult;
}
@@ -1012,16 +1012,12 @@
if (mpAudioTrack) {
delete mpAudioTrack;
- mpAudioTrack = 0;
+ mpAudioTrack = NULL;
}
// Open audio track in mono, PCM 16bit, default sampling rate, default buffer size
mpAudioTrack = new AudioTrack();
- if (mpAudioTrack == 0) {
- ALOGE("AudioTrack allocation failed");
- goto initAudioTrack_exit;
- }
- ALOGV("Create Track: %p\n", mpAudioTrack);
+ ALOGV("Create Track: %p", mpAudioTrack);
mpAudioTrack->set(mStreamType,
0,
@@ -1049,10 +1045,10 @@
initAudioTrack_exit:
// Cleanup
- if (mpAudioTrack) {
- ALOGV("Delete Track I: %p\n", mpAudioTrack);
+ if (mpAudioTrack != NULL) {
+ ALOGV("Delete Track I: %p", mpAudioTrack);
delete mpAudioTrack;
- mpAudioTrack = 0;
+ mpAudioTrack = NULL;
}
return false;
@@ -1145,7 +1141,7 @@
if (lpToneGen->mTotalSmp > lpToneGen->mNextSegSmp) {
// Time to go to next sequence segment
- ALOGV("End Segment, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("End Segment, time: %d", (unsigned int)(systemTime()/1000000));
lGenSmp = lReqSmp;
@@ -1160,13 +1156,13 @@
lpWaveGen->getSamples(lpOut, lGenSmp, lWaveCmd);
lFrequency = lpToneDesc->segments[lpToneGen->mCurSegment].waveFreq[++lFreqIdx];
}
- ALOGV("ON->OFF, lGenSmp: %d, lReqSmp: %d\n", lGenSmp, lReqSmp);
+ ALOGV("ON->OFF, lGenSmp: %d, lReqSmp: %d", lGenSmp, lReqSmp);
}
// check if we need to loop and loop for the reqd times
if (lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt) {
if (lpToneGen->mLoopCounter < lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt) {
- ALOGV ("in if loop loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("in if loop loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
@@ -1176,14 +1172,14 @@
// completed loop. go to next segment
lpToneGen->mLoopCounter = 0;
lpToneGen->mCurSegment++;
- ALOGV ("in else loop loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("in else loop loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
}
} else {
lpToneGen->mCurSegment++;
- ALOGV ("Goto next seg loopCnt(%d) loopctr(%d), CurSeg(%d) \n",
+ ALOGV ("Goto next seg loopCnt(%d) loopctr(%d), CurSeg(%d)",
lpToneDesc->segments[lpToneGen->mCurSegment].loopCnt,
lpToneGen->mLoopCounter,
lpToneGen->mCurSegment);
@@ -1192,32 +1188,32 @@
// Handle loop if last segment reached
if (lpToneDesc->segments[lpToneGen->mCurSegment].duration == 0) {
- ALOGV("Last Seg: %d\n", lpToneGen->mCurSegment);
+ ALOGV("Last Seg: %d", lpToneGen->mCurSegment);
// Pre increment loop count and restart if total count not reached. Stop sequence otherwise
if (++lpToneGen->mCurCount <= lpToneDesc->repeatCnt) {
- ALOGV("Repeating Count: %d\n", lpToneGen->mCurCount);
+ ALOGV("Repeating Count: %d", lpToneGen->mCurCount);
lpToneGen->mCurSegment = lpToneDesc->repeatSegment;
if (lpToneDesc->segments[lpToneDesc->repeatSegment].waveFreq[0] != 0) {
lWaveCmd = WaveGenerator::WAVEGEN_START;
}
- ALOGV("New segment %d, Next Time: %d\n", lpToneGen->mCurSegment,
+ ALOGV("New segment %d, Next Time: %d", lpToneGen->mCurSegment,
(lpToneGen->mNextSegSmp*1000)/lpToneGen->mSamplingRate);
} else {
lGenSmp = 0;
- ALOGV("End repeat, time: %d\n", (unsigned int)(systemTime()/1000000));
+ ALOGV("End repeat, time: %d", (unsigned int)(systemTime()/1000000));
}
} else {
- ALOGV("New segment %d, Next Time: %d\n", lpToneGen->mCurSegment,
+ ALOGV("New segment %d, Next Time: %d", lpToneGen->mCurSegment,
(lpToneGen->mNextSegSmp*1000)/lpToneGen->mSamplingRate);
if (lpToneDesc->segments[lpToneGen->mCurSegment].waveFreq[0] != 0) {
// If next segment is not silent, OFF -> ON transition : reset wave generator
lWaveCmd = WaveGenerator::WAVEGEN_START;
- ALOGV("OFF->ON, lGenSmp: %d, lReqSmp: %d\n", lGenSmp, lReqSmp);
+ ALOGV("OFF->ON, lGenSmp: %d, lReqSmp: %d", lGenSmp, lReqSmp);
} else {
lGenSmp = 0;
}
@@ -1255,13 +1251,13 @@
switch (lpToneGen->mState) {
case TONE_RESTARTING:
- ALOGV("Cbk restarting track\n");
+ ALOGV("Cbk restarting track");
if (lpToneGen->prepareWave()) {
lpToneGen->mState = TONE_STARTING;
// must reload lpToneDesc as prepareWave() may change mpToneDesc
lpToneDesc = lpToneGen->mpToneDesc;
} else {
- ALOGW("Cbk restarting prepareWave() failed\n");
+ ALOGW("Cbk restarting prepareWave() failed");
lpToneGen->mState = TONE_IDLE;
lpToneGen->mpAudioTrack->stop();
// Force loop exit
@@ -1270,14 +1266,14 @@
lSignal = true;
break;
case TONE_STOPPING:
- ALOGV("Cbk Stopping\n");
+ ALOGV("Cbk Stopping");
lpToneGen->mState = TONE_STOPPED;
// Force loop exit
lNumSmp = 0;
break;
case TONE_STOPPED:
lpToneGen->mState = TONE_INIT;
- ALOGV("Cbk Stopped track\n");
+ ALOGV("Cbk Stopped track");
lpToneGen->mpAudioTrack->stop();
// Force loop exit
lNumSmp = 0;
@@ -1285,7 +1281,7 @@
lSignal = true;
break;
case TONE_STARTING:
- ALOGV("Cbk starting track\n");
+ ALOGV("Cbk starting track");
lpToneGen->mState = TONE_PLAYING;
lSignal = true;
break;
@@ -1321,7 +1317,7 @@
bool ToneGenerator::prepareWave() {
unsigned int segmentIdx = 0;
- if (!mpNewToneDesc) {
+ if (mpNewToneDesc == NULL) {
return false;
}
@@ -1353,9 +1349,6 @@
new ToneGenerator::WaveGenerator((unsigned short)mSamplingRate,
frequency,
TONEGEN_GAIN/lNumWaves);
- if (lpWaveGen == 0) {
- goto prepareWave_exit;
- }
mWaveGens.add(frequency, lpWaveGen);
}
frequency = mpNewToneDesc->segments[segmentIdx].waveFreq[++freqIdx];
@@ -1375,12 +1368,6 @@
}
return true;
-
-prepareWave_exit:
-
- clearWaveGens();
-
- return false;
}
@@ -1447,13 +1434,13 @@
// none
//
////////////////////////////////////////////////////////////////////////////////
-int ToneGenerator::getToneForRegion(int toneType) {
- int regionTone;
+ToneGenerator::tone_type ToneGenerator::getToneForRegion(tone_type toneType) {
+ tone_type regionTone;
if (mRegion == CEPT || toneType < FIRST_SUP_TONE || toneType > LAST_SUP_TONE) {
regionTone = toneType;
} else {
- regionTone = sToneMappingTable[mRegion][toneType - FIRST_SUP_TONE];
+ regionTone = (tone_type) sToneMappingTable[mRegion][toneType - FIRST_SUP_TONE];
}
ALOGV("getToneForRegion, tone %d, region %d, regionTone %d", toneType, mRegion, regionTone);
@@ -1504,7 +1491,7 @@
d0 = 32767;
mA1_Q14 = (short) d0;
- ALOGV("WaveGenerator init, mA1_Q14: %d, mS2_0: %d, mAmplitude_Q15: %d\n",
+ ALOGV("WaveGenerator init, mA1_Q14: %d, mS2_0: %d, mAmplitude_Q15: %d",
mA1_Q14, mS2_0, mAmplitude_Q15);
}
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
index d08ffa5..70f8c0c 100644
--- a/media/libmedia/Visualizer.cpp
+++ b/media/libmedia/Visualizer.cpp
@@ -27,8 +27,7 @@
#include <cutils/bitops.h>
#include <media/Visualizer.h>
-
-extern void fixed_fft_real(int n, int32_t *v);
+#include <audio_utils/fixedfft.h>
namespace android {
@@ -54,7 +53,7 @@
status_t Visualizer::setEnabled(bool enabled)
{
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
sp<CaptureThread> t = mCaptureThread;
if (t != 0) {
@@ -74,7 +73,7 @@
if (status == NO_ERROR) {
if (t != 0) {
if (enabled) {
- t->run("AudioTrackThread");
+ t->run("Visualizer");
} else {
t->requestExit();
}
@@ -93,7 +92,7 @@
if (rate > CAPTURE_RATE_MAX) {
return BAD_VALUE;
}
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
if (mEnabled) {
return INVALID_OPERATION;
@@ -115,10 +114,6 @@
if (cbk != NULL) {
mCaptureThread = new CaptureThread(*this, rate, ((flags & CAPTURE_CALL_JAVA) != 0));
- if (mCaptureThread == 0) {
- ALOGE("Could not create callback thread");
- return NO_INIT;
- }
}
ALOGV("setCaptureCallBack() rate: %d thread %p flags 0x%08x",
rate, mCaptureThread.get(), mCaptureFlags);
@@ -133,7 +128,7 @@
return BAD_VALUE;
}
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
if (mEnabled) {
return INVALID_OPERATION;
}
@@ -173,7 +168,7 @@
uint32_t replySize = mCaptureSize;
status = command(VISUALIZER_CMD_CAPTURE, 0, NULL, &replySize, waveform);
ALOGV("getWaveForm() command returned %d", status);
- if (replySize == 0) {
+ if ((status == NO_ERROR) && (replySize == 0)) {
status = NOT_ENOUGH_DATA;
}
} else {
@@ -235,7 +230,7 @@
void Visualizer::periodicCapture()
{
- Mutex::Autolock _l(mLock);
+ Mutex::Autolock _l(mCaptureLock);
ALOGV("periodicCapture() %p mCaptureCallBack %p mCaptureFlags 0x%08x",
this, mCaptureCallBack, mCaptureFlags);
if (mCaptureCallBack != NULL &&
diff --git a/media/libmedia/autodetect.cpp b/media/libmedia/autodetect.cpp
index dfcc6a0..be5c3b2 100644
--- a/media/libmedia/autodetect.cpp
+++ b/media/libmedia/autodetect.cpp
@@ -16,7 +16,7 @@
#include "autodetect.h"
-typedef struct CharRange {
+struct CharRange {
uint16_t first;
uint16_t last;
};
diff --git a/media/libmedia/fixedfft.cpp b/media/libmedia/fixedfft.cpp
deleted file mode 100644
index 2b495e6..0000000
--- a/media/libmedia/fixedfft.cpp
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-/* A Fixed point implementation of Fast Fourier Transform (FFT). Complex numbers
- * are represented by 32-bit integers, where higher 16 bits are real part and
- * lower ones are imaginary part. Few compromises are made between efficiency,
- * accuracy, and maintainability. To make it fast, arithmetic shifts are used
- * instead of divisions, and bitwise inverses are used instead of negates. To
- * keep it small, only radix-2 Cooley-Tukey algorithm is implemented, and only
- * half of the twiddle factors are stored. Although there are still ways to make
- * it even faster or smaller, it costs too much on one of the aspects.
- */
-
-#include <stdio.h>
-#include <stdint.h>
-#ifdef __arm__
-#include <machine/cpu-features.h>
-#endif
-
-#define LOG_FFT_SIZE 10
-#define MAX_FFT_SIZE (1 << LOG_FFT_SIZE)
-
-static const int32_t twiddle[MAX_FFT_SIZE / 4] = {
- 0x00008000, 0xff378001, 0xfe6e8002, 0xfda58006, 0xfcdc800a, 0xfc13800f,
- 0xfb4a8016, 0xfa81801e, 0xf9b88027, 0xf8ef8032, 0xf827803e, 0xf75e804b,
- 0xf6958059, 0xf5cd8068, 0xf5058079, 0xf43c808b, 0xf374809e, 0xf2ac80b2,
- 0xf1e480c8, 0xf11c80de, 0xf05580f6, 0xef8d8110, 0xeec6812a, 0xedff8146,
- 0xed388163, 0xec718181, 0xebab81a0, 0xeae481c1, 0xea1e81e2, 0xe9588205,
- 0xe892822a, 0xe7cd824f, 0xe7078276, 0xe642829d, 0xe57d82c6, 0xe4b982f1,
- 0xe3f4831c, 0xe3308349, 0xe26d8377, 0xe1a983a6, 0xe0e683d6, 0xe0238407,
- 0xdf61843a, 0xde9e846e, 0xdddc84a3, 0xdd1b84d9, 0xdc598511, 0xdb998549,
- 0xdad88583, 0xda1885be, 0xd95885fa, 0xd8988637, 0xd7d98676, 0xd71b86b6,
- 0xd65c86f6, 0xd59e8738, 0xd4e1877b, 0xd42487c0, 0xd3678805, 0xd2ab884c,
- 0xd1ef8894, 0xd13488dd, 0xd0798927, 0xcfbe8972, 0xcf0489be, 0xce4b8a0c,
- 0xcd928a5a, 0xccd98aaa, 0xcc218afb, 0xcb698b4d, 0xcab28ba0, 0xc9fc8bf5,
- 0xc9468c4a, 0xc8908ca1, 0xc7db8cf8, 0xc7278d51, 0xc6738dab, 0xc5c08e06,
- 0xc50d8e62, 0xc45b8ebf, 0xc3a98f1d, 0xc2f88f7d, 0xc2488fdd, 0xc198903e,
- 0xc0e990a1, 0xc03a9105, 0xbf8c9169, 0xbedf91cf, 0xbe329236, 0xbd86929e,
- 0xbcda9307, 0xbc2f9371, 0xbb8593dc, 0xbadc9448, 0xba3394b5, 0xb98b9523,
- 0xb8e39592, 0xb83c9603, 0xb7969674, 0xb6f196e6, 0xb64c9759, 0xb5a897ce,
- 0xb5059843, 0xb46298b9, 0xb3c09930, 0xb31f99a9, 0xb27f9a22, 0xb1df9a9c,
- 0xb1409b17, 0xb0a29b94, 0xb0059c11, 0xaf689c8f, 0xaecc9d0e, 0xae319d8e,
- 0xad979e0f, 0xacfd9e91, 0xac659f14, 0xabcd9f98, 0xab36a01c, 0xaaa0a0a2,
- 0xaa0aa129, 0xa976a1b0, 0xa8e2a238, 0xa84fa2c2, 0xa7bda34c, 0xa72ca3d7,
- 0xa69ca463, 0xa60ca4f0, 0xa57ea57e, 0xa4f0a60c, 0xa463a69c, 0xa3d7a72c,
- 0xa34ca7bd, 0xa2c2a84f, 0xa238a8e2, 0xa1b0a976, 0xa129aa0a, 0xa0a2aaa0,
- 0xa01cab36, 0x9f98abcd, 0x9f14ac65, 0x9e91acfd, 0x9e0fad97, 0x9d8eae31,
- 0x9d0eaecc, 0x9c8faf68, 0x9c11b005, 0x9b94b0a2, 0x9b17b140, 0x9a9cb1df,
- 0x9a22b27f, 0x99a9b31f, 0x9930b3c0, 0x98b9b462, 0x9843b505, 0x97ceb5a8,
- 0x9759b64c, 0x96e6b6f1, 0x9674b796, 0x9603b83c, 0x9592b8e3, 0x9523b98b,
- 0x94b5ba33, 0x9448badc, 0x93dcbb85, 0x9371bc2f, 0x9307bcda, 0x929ebd86,
- 0x9236be32, 0x91cfbedf, 0x9169bf8c, 0x9105c03a, 0x90a1c0e9, 0x903ec198,
- 0x8fddc248, 0x8f7dc2f8, 0x8f1dc3a9, 0x8ebfc45b, 0x8e62c50d, 0x8e06c5c0,
- 0x8dabc673, 0x8d51c727, 0x8cf8c7db, 0x8ca1c890, 0x8c4ac946, 0x8bf5c9fc,
- 0x8ba0cab2, 0x8b4dcb69, 0x8afbcc21, 0x8aaaccd9, 0x8a5acd92, 0x8a0cce4b,
- 0x89becf04, 0x8972cfbe, 0x8927d079, 0x88ddd134, 0x8894d1ef, 0x884cd2ab,
- 0x8805d367, 0x87c0d424, 0x877bd4e1, 0x8738d59e, 0x86f6d65c, 0x86b6d71b,
- 0x8676d7d9, 0x8637d898, 0x85fad958, 0x85beda18, 0x8583dad8, 0x8549db99,
- 0x8511dc59, 0x84d9dd1b, 0x84a3dddc, 0x846ede9e, 0x843adf61, 0x8407e023,
- 0x83d6e0e6, 0x83a6e1a9, 0x8377e26d, 0x8349e330, 0x831ce3f4, 0x82f1e4b9,
- 0x82c6e57d, 0x829de642, 0x8276e707, 0x824fe7cd, 0x822ae892, 0x8205e958,
- 0x81e2ea1e, 0x81c1eae4, 0x81a0ebab, 0x8181ec71, 0x8163ed38, 0x8146edff,
- 0x812aeec6, 0x8110ef8d, 0x80f6f055, 0x80def11c, 0x80c8f1e4, 0x80b2f2ac,
- 0x809ef374, 0x808bf43c, 0x8079f505, 0x8068f5cd, 0x8059f695, 0x804bf75e,
- 0x803ef827, 0x8032f8ef, 0x8027f9b8, 0x801efa81, 0x8016fb4a, 0x800ffc13,
- 0x800afcdc, 0x8006fda5, 0x8002fe6e, 0x8001ff37,
-};
-
-/* Returns the multiplication of \conj{a} and {b}. */
-static inline int32_t mult(int32_t a, int32_t b)
-{
-#if __ARM_ARCH__ >= 6
- int32_t t = b;
- __asm__("smuad %0, %0, %1" : "+r" (t) : "r" (a));
- __asm__("smusdx %0, %0, %1" : "+r" (b) : "r" (a));
- __asm__("pkhtb %0, %0, %1, ASR #16" : "+r" (t) : "r" (b));
- return t;
-#else
- return (((a >> 16) * (b >> 16) + (int16_t)a * (int16_t)b) & ~0xFFFF) |
- ((((a >> 16) * (int16_t)b - (int16_t)a * (b >> 16)) >> 16) & 0xFFFF);
-#endif
-}
-
-static inline int32_t half(int32_t a)
-{
-#if __ARM_ARCH__ >= 6
- __asm__("shadd16 %0, %0, %1" : "+r" (a) : "r" (0));
- return a;
-#else
- return ((a >> 1) & ~0x8000) | (a & 0x8000);
-#endif
-}
-
-void fixed_fft(int n, int32_t *v)
-{
- int scale = LOG_FFT_SIZE, i, p, r;
-
- for (r = 0, i = 1; i < n; ++i) {
- for (p = n; !(p & r); p >>= 1, r ^= p);
- if (i < r) {
- int32_t t = v[i];
- v[i] = v[r];
- v[r] = t;
- }
- }
-
- for (p = 1; p < n; p <<= 1) {
- --scale;
-
- for (i = 0; i < n; i += p << 1) {
- int32_t x = half(v[i]);
- int32_t y = half(v[i + p]);
- v[i] = x + y;
- v[i + p] = x - y;
- }
-
- for (r = 1; r < p; ++r) {
- int32_t w = MAX_FFT_SIZE / 4 - (r << scale);
- i = w >> 31;
- w = twiddle[(w ^ i) - i] ^ (i << 16);
- for (i = r; i < n; i += p << 1) {
- int32_t x = half(v[i]);
- int32_t y = mult(w, v[i + p]);
- v[i] = x - y;
- v[i + p] = x + y;
- }
- }
- }
-}
-
-void fixed_fft_real(int n, int32_t *v)
-{
- int scale = LOG_FFT_SIZE, m = n >> 1, i;
-
- fixed_fft(n, v);
- for (i = 1; i <= n; i <<= 1, --scale);
- v[0] = mult(~v[0], 0x80008000);
- v[m] = half(v[m]);
-
- for (i = 1; i < n >> 1; ++i) {
- int32_t x = half(v[i]);
- int32_t z = half(v[n - i]);
- int32_t y = z - (x ^ 0xFFFF);
- x = half(x + (z ^ 0xFFFF));
- y = mult(y, twiddle[i << scale]);
- v[i] = x - y;
- v[n - i] = (x + y) ^ 0xFFFF;
- }
-}
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index 88e269f..8d53357 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -35,7 +35,7 @@
const sp<IMediaPlayerService>& MediaMetadataRetriever::getService()
{
Mutex::Autolock lock(sServiceLock);
- if (sService.get() == 0) {
+ if (sService == 0) {
sp<IServiceManager> sm = defaultServiceManager();
sp<IBinder> binder;
do {
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 2284927..9d45907 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -1,4 +1,4 @@
-/* mediaplayer.cpp
+/*
**
** Copyright 2006, The Android Open Source Project
**
@@ -30,9 +30,7 @@
#include <gui/SurfaceTextureClient.h>
#include <media/mediaplayer.h>
-#include <media/AudioTrack.h>
-
-#include <surfaceflinger/Surface.h>
+#include <media/AudioSystem.h>
#include <binder/MemoryBase.h>
@@ -478,7 +476,7 @@
return reset_l();
}
-status_t MediaPlayer::setAudioStreamType(int type)
+status_t MediaPlayer::setAudioStreamType(audio_stream_type_t type)
{
ALOGV("MediaPlayer::setAudioStreamType");
Mutex::Autolock _l(mLock);
@@ -709,7 +707,7 @@
}
}
-/*static*/ sp<IMemory> MediaPlayer::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+/*static*/ sp<IMemory> MediaPlayer::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%s)", url);
sp<IMemory> p;
@@ -729,7 +727,7 @@
notify(MEDIA_ERROR, MEDIA_ERROR_SERVER_DIED, 0);
}
-/*static*/ sp<IMemory> MediaPlayer::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+/*static*/ sp<IMemory> MediaPlayer::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
sp<IMemory> p;
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index 8d947d8..cc73014 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -18,7 +18,6 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MediaRecorder"
#include <utils/Log.h>
-#include <surfaceflinger/Surface.h>
#include <media/mediarecorder.h>
#include <binder/IServiceManager.h>
#include <utils/String8.h>
diff --git a/media/libmediaplayerservice/Android.mk b/media/libmediaplayerservice/Android.mk
index a3e2517..e521648 100644
--- a/media/libmediaplayerservice/Android.mk
+++ b/media/libmediaplayerservice/Android.mk
@@ -29,7 +29,8 @@
libstagefright_omx \
libstagefright_foundation \
libgui \
- libdl
+ libdl \
+ libaah_rtp
LOCAL_STATIC_LIBRARIES := \
libstagefright_nuplayer \
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index f5cb019..764eddc 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -70,6 +70,11 @@
#include <OMX.h>
+namespace android {
+sp<MediaPlayerBase> createAAH_TXPlayer();
+sp<MediaPlayerBase> createAAH_RXPlayer();
+}
+
namespace {
using android::media::Metadata;
using android::status_t;
@@ -593,6 +598,14 @@
return NU_PLAYER;
}
+ if (!strncasecmp("aahRX://", url, 8)) {
+ return AAH_RX_PLAYER;
+ }
+
+ if (!strncasecmp("aahTX://", url, 8)) {
+ return AAH_TX_PLAYER;
+ }
+
// use MidiFile for MIDI extensions
int lenURL = strlen(url);
for (int i = 0; i < NELEM(FILE_EXTS); ++i) {
@@ -629,6 +642,14 @@
ALOGV("Create Test Player stub");
p = new TestPlayerStub();
break;
+ case AAH_RX_PLAYER:
+ ALOGV(" create A@H RX Player");
+ p = createAAH_RXPlayer();
+ break;
+ case AAH_TX_PLAYER:
+ ALOGV(" create A@H TX Player");
+ p = createAAH_TXPlayer();
+ break;
default:
ALOGE("Unknown player type: %d", playerType);
return NULL;
@@ -1010,7 +1031,7 @@
return p->reset();
}
-status_t MediaPlayerService::Client::setAudioStreamType(int type)
+status_t MediaPlayerService::Client::setAudioStreamType(audio_stream_type_t type)
{
ALOGV("[%d] setAudioStreamType(%d)", mConnId, type);
// TODO: for hardware output, call player instead
@@ -1031,9 +1052,21 @@
status_t MediaPlayerService::Client::setVolume(float leftVolume, float rightVolume)
{
ALOGV("[%d] setVolume(%f, %f)", mConnId, leftVolume, rightVolume);
- // TODO: for hardware output, call player instead
- Mutex::Autolock l(mLock);
- if (mAudioOutput != 0) mAudioOutput->setVolume(leftVolume, rightVolume);
+
+ // for hardware output, call player instead
+ sp<MediaPlayerBase> p = getPlayer();
+ {
+ Mutex::Autolock l(mLock);
+ if (p != 0 && p->hardwareOutput()) {
+ MediaPlayerHWInterface* hwp =
+ reinterpret_cast<MediaPlayerHWInterface*>(p.get());
+ return hwp->setVolume(leftVolume, rightVolume);
+ } else {
+ if (mAudioOutput != 0) mAudioOutput->setVolume(leftVolume, rightVolume);
+ return NO_ERROR;
+ }
+ }
+
return NO_ERROR;
}
@@ -1149,7 +1182,7 @@
static size_t kDefaultHeapSize = 1024 * 1024; // 1MB
-sp<IMemory> MediaPlayerService::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+sp<IMemory> MediaPlayerService::decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%s)", url);
sp<MemoryBase> mem;
@@ -1197,7 +1230,7 @@
mem = new MemoryBase(cache->getHeap(), 0, cache->size());
*pSampleRate = cache->sampleRate();
*pNumChannels = cache->channelCount();
- *pFormat = (int)cache->format();
+ *pFormat = cache->format();
ALOGV("return memory @ %p, sampleRate=%u, channelCount = %d, format = %d", mem->pointer(), *pSampleRate, *pNumChannels, *pFormat);
Exit:
@@ -1205,7 +1238,7 @@
return mem;
}
-sp<IMemory> MediaPlayerService::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat)
+sp<IMemory> MediaPlayerService::decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat)
{
ALOGV("decode(%d, %lld, %lld)", fd, offset, length);
sp<MemoryBase> mem;
@@ -1265,6 +1298,8 @@
mStreamType = AUDIO_STREAM_MUSIC;
mLeftVolume = 1.0;
mRightVolume = 1.0;
+ mPlaybackRatePermille = 1000;
+ mSampleRateHz = 0;
mLatency = 0;
mMsecsPerFrame = 0;
mAuxEffectId = 0;
@@ -1339,7 +1374,7 @@
}
status_t MediaPlayerService::AudioOutput::open(
- uint32_t sampleRate, int channelCount, int format, int bufferCount,
+ uint32_t sampleRate, int channelCount, audio_format_t format, int bufferCount,
AudioCallback cb, void *cookie)
{
mCallback = cb;
@@ -1402,10 +1437,15 @@
ALOGV("setVolume");
t->setVolume(mLeftVolume, mRightVolume);
- mMsecsPerFrame = 1.e3 / (float) sampleRate;
+ mSampleRateHz = sampleRate;
+ mMsecsPerFrame = mPlaybackRatePermille / (float) sampleRate;
mLatency = t->latency();
mTrack = t;
+ status_t res = t->setSampleRate(mPlaybackRatePermille * mSampleRateHz / 1000);
+ if (res != NO_ERROR) {
+ return res;
+ }
t->setAuxEffectSendLevel(mSendLevel);
return t->attachAuxEffect(mAuxEffectId);;
}
@@ -1469,6 +1509,22 @@
}
}
+status_t MediaPlayerService::AudioOutput::setPlaybackRatePermille(int32_t ratePermille)
+{
+ ALOGV("setPlaybackRatePermille(%d)", ratePermille);
+ status_t res = NO_ERROR;
+ if (mTrack) {
+ res = mTrack->setSampleRate(ratePermille * mSampleRateHz / 1000);
+ } else {
+ res = NO_INIT;
+ }
+ mPlaybackRatePermille = ratePermille;
+ if (mSampleRateHz != 0) {
+ mMsecsPerFrame = mPlaybackRatePermille / (float) mSampleRateHz;
+ }
+ return res;
+}
+
status_t MediaPlayerService::AudioOutput::setAuxEffectSendLevel(float level)
{
ALOGV("setAuxEffectSendLevel(%f)", level);
@@ -1611,7 +1667,7 @@
////////////////////////////////////////////////////////////////////////////////
status_t MediaPlayerService::AudioCache::open(
- uint32_t sampleRate, int channelCount, int format, int bufferCount,
+ uint32_t sampleRate, int channelCount, audio_format_t format, int bufferCount,
AudioCallback cb, void *cookie)
{
ALOGV("open(%u, %d, %d, %d)", sampleRate, channelCount, format, bufferCount);
@@ -1621,7 +1677,7 @@
mSampleRate = sampleRate;
mChannelCount = (uint16_t)channelCount;
- mFormat = (uint16_t)format;
+ mFormat = format;
mMsecsPerFrame = 1.e3 / (float) sampleRate;
if (cb != NULL) {
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 04d9e28..52af64d 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -34,6 +34,7 @@
namespace android {
+class AudioTrack;
class IMediaRecorder;
class IMediaMetadataRetriever;
class IOMX;
@@ -83,7 +84,7 @@
virtual status_t open(
uint32_t sampleRate, int channelCount,
- int format, int bufferCount,
+ audio_format_t format, int bufferCount,
AudioCallback cb, void *cookie);
virtual void start();
@@ -92,8 +93,9 @@
virtual void flush();
virtual void pause();
virtual void close();
- void setAudioStreamType(int streamType) { mStreamType = streamType; }
+ void setAudioStreamType(audio_stream_type_t streamType) { mStreamType = streamType; }
void setVolume(float left, float right);
+ virtual status_t setPlaybackRatePermille(int32_t ratePermille);
status_t setAuxEffectSendLevel(float level);
status_t attachAuxEffect(int effectId);
virtual status_t dump(int fd, const Vector<String16>& args) const;
@@ -108,9 +110,11 @@
AudioTrack* mTrack;
AudioCallback mCallback;
void * mCallbackCookie;
- int mStreamType;
+ audio_stream_type_t mStreamType;
float mLeftVolume;
float mRightVolume;
+ int32_t mPlaybackRatePermille;
+ uint32_t mSampleRateHz; // sample rate of the content, as set in open()
float mMsecsPerFrame;
uint32_t mLatency;
int mSessionId;
@@ -139,7 +143,7 @@
virtual int getSessionId();
virtual status_t open(
- uint32_t sampleRate, int channelCount, int format,
+ uint32_t sampleRate, int channelCount, audio_format_t format,
int bufferCount = 1,
AudioCallback cb = NULL, void *cookie = NULL);
@@ -149,10 +153,11 @@
virtual void flush() {}
virtual void pause() {}
virtual void close() {}
- void setAudioStreamType(int streamType) {}
+ void setAudioStreamType(audio_stream_type_t streamType) {}
void setVolume(float left, float right) {}
+ virtual status_t setPlaybackRatePermille(int32_t ratePermille) { return INVALID_OPERATION; }
uint32_t sampleRate() const { return mSampleRate; }
- uint32_t format() const { return (uint32_t)mFormat; }
+ audio_format_t format() const { return mFormat; }
size_t size() const { return mSize; }
status_t wait();
@@ -170,7 +175,7 @@
sp<MemoryHeapBase> mHeap;
float mMsecsPerFrame;
uint16_t mChannelCount;
- uint16_t mFormat;
+ audio_format_t mFormat;
ssize_t mFrameCount;
uint32_t mSampleRate;
uint32_t mSize;
@@ -190,8 +195,8 @@
virtual sp<IMediaPlayer> create(pid_t pid, const sp<IMediaPlayerClient>& client, int audioSessionId);
- virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, int* pFormat);
- virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, int* pFormat);
+ virtual sp<IMemory> decode(const char* url, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
+ virtual sp<IMemory> decode(int fd, int64_t offset, int64_t length, uint32_t *pSampleRate, int* pNumChannels, audio_format_t* pFormat);
virtual sp<IOMX> getOMX();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -259,7 +264,7 @@
virtual status_t getCurrentPosition(int* msec);
virtual status_t getDuration(int* msec);
virtual status_t reset();
- virtual status_t setAudioStreamType(int type);
+ virtual status_t setAudioStreamType(audio_stream_type_t type);
virtual status_t setLooping(int loop);
virtual status_t setVolume(float leftVolume, float rightVolume);
virtual status_t invoke(const Parcel& request, Parcel *reply);
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index d219fc2..beda945 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -33,8 +33,6 @@
#include <utils/String16.h>
-#include <media/AudioTrack.h>
-
#include <system/audio.h>
#include "MediaRecorderClient.h"
diff --git a/media/libmediaplayerservice/MidiFile.cpp b/media/libmediaplayerservice/MidiFile.cpp
index d89b5f4..7cb8c29 100644
--- a/media/libmediaplayerservice/MidiFile.cpp
+++ b/media/libmediaplayerservice/MidiFile.cpp
@@ -86,7 +86,8 @@
// create playback thread
{
Mutex::Autolock l(mMutex);
- createThreadEtc(renderThread, this, "midithread", ANDROID_PRIORITY_AUDIO);
+ mThread = new MidiFileThread(this);
+ mThread->run("midithread", ANDROID_PRIORITY_AUDIO);
mCondition.wait(mMutex);
ALOGV("thread started");
}
@@ -427,11 +428,6 @@
return NO_ERROR;
}
-int MidiFile::renderThread(void* p) {
-
- return ((MidiFile*)p)->render();
-}
-
int MidiFile::render() {
EAS_RESULT result = EAS_FAILURE;
EAS_I32 count;
diff --git a/media/libmediaplayerservice/MidiFile.h b/media/libmediaplayerservice/MidiFile.h
index 3469389..f6f8f7b 100644
--- a/media/libmediaplayerservice/MidiFile.h
+++ b/media/libmediaplayerservice/MidiFile.h
@@ -19,11 +19,11 @@
#define ANDROID_MIDIFILE_H
#include <media/MediaPlayerInterface.h>
-#include <media/AudioTrack.h>
#include <libsonivox/eas.h>
namespace android {
+// Note that the name MidiFile is misleading; this actually represents a MIDI file player
class MidiFile : public MediaPlayerInterface {
public:
MidiFile();
@@ -65,7 +65,6 @@
private:
status_t createOutputTrack();
status_t reset_nosync();
- static int renderThread(void*);
int render();
void updateState(){ EAS_State(mEasData, mEasHandle, &mState); }
@@ -78,12 +77,35 @@
EAS_I32 mDuration;
EAS_STATE mState;
EAS_FILE mFileLocator;
- int mStreamType;
+ audio_stream_type_t mStreamType;
bool mLoop;
volatile bool mExit;
bool mPaused;
volatile bool mRender;
pid_t mTid;
+
+ class MidiFileThread : public Thread {
+ public:
+ MidiFileThread(MidiFile *midiPlayer) : mMidiFile(midiPlayer) {
+ }
+
+ protected:
+ virtual ~MidiFileThread() {}
+
+ private:
+ MidiFile *mMidiFile;
+
+ bool threadLoop() {
+ int result;
+ result = mMidiFile->render();
+ return false;
+ }
+
+ MidiFileThread(const MidiFileThread &);
+ MidiFileThread &operator=(const MidiFileThread &);
+ };
+
+ sp<MidiFileThread> mThread;
};
}; // namespace android
diff --git a/media/libmediaplayerservice/StagefrightPlayer.cpp b/media/libmediaplayerservice/StagefrightPlayer.cpp
index 6d7771a..052ebf0 100644
--- a/media/libmediaplayerservice/StagefrightPlayer.cpp
+++ b/media/libmediaplayerservice/StagefrightPlayer.cpp
@@ -176,7 +176,7 @@
}
status_t StagefrightPlayer::setParameter(int key, const Parcel &request) {
- ALOGV("setParameter");
+ ALOGV("setParameter(key=%d)", key);
return mPlayer->setParameter(key, request);
}
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index 4632016..ca79657 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -24,6 +24,7 @@
#include <binder/IServiceManager.h>
#include <media/IMediaPlayerService.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AudioSource.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/AACWriter.h>
@@ -31,7 +32,6 @@
#include <media/stagefright/CameraSourceTimeLapse.h>
#include <media/stagefright/MPEG2TSWriter.h>
#include <media/stagefright/MPEG4Writer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXClient.h>
@@ -40,7 +40,7 @@
#include <media/MediaProfiles.h>
#include <camera/ICamera.h>
#include <camera/CameraParameters.h>
-#include <surfaceflinger/Surface.h>
+#include <gui/Surface.h>
#include <utils/Errors.h>
#include <sys/types.h>
@@ -241,8 +241,8 @@
status_t StagefrightRecorder::setOutputFile(int fd, int64_t offset, int64_t length) {
ALOGV("setOutputFile: %d, %lld, %lld", fd, offset, length);
// These don't make any sense, do they?
- CHECK_EQ(offset, 0);
- CHECK_EQ(length, 0);
+ CHECK_EQ(offset, 0ll);
+ CHECK_EQ(length, 0ll);
if (fd < 0) {
ALOGE("Invalid file descriptor: %d", fd);
@@ -734,7 +734,7 @@
}
status_t StagefrightRecorder::start() {
- CHECK(mOutputFd >= 0);
+ CHECK_GE(mOutputFd, 0);
if (mWriter != NULL) {
ALOGE("File writer is not avaialble");
@@ -837,7 +837,7 @@
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
sp<MediaSource> audioEncoder =
OMXCodec::Create(client.interface(), encMeta,
@@ -850,9 +850,9 @@
status_t StagefrightRecorder::startAACRecording() {
// FIXME:
// Add support for OUTPUT_FORMAT_AAC_ADIF
- CHECK(mOutputFormat == OUTPUT_FORMAT_AAC_ADTS);
+ CHECK_EQ(mOutputFormat, OUTPUT_FORMAT_AAC_ADTS);
- CHECK(mAudioEncoder == AUDIO_ENCODER_AAC);
+ CHECK_EQ(mAudioEncoder, AUDIO_ENCODER_AAC);
CHECK(mAudioSource != AUDIO_SOURCE_CNT);
mWriter = new AACWriter(mOutputFd);
@@ -1291,6 +1291,12 @@
videoSize.width = mVideoWidth;
videoSize.height = mVideoHeight;
if (mCaptureTimeLapse) {
+ if (mTimeBetweenTimeLapseFrameCaptureUs < 0) {
+ ALOGE("Invalid mTimeBetweenTimeLapseFrameCaptureUs value: %lld",
+ mTimeBetweenTimeLapseFrameCaptureUs);
+ return BAD_VALUE;
+ }
+
mCameraSourceTimeLapse = CameraSourceTimeLapse::CreateFromCamera(
mCamera, mCameraProxy, mCameraId,
videoSize, mFrameRate, mPreviewSurface,
@@ -1386,7 +1392,7 @@
}
OMXClient client;
- CHECK_EQ(client.connect(), OK);
+ CHECK_EQ(client.connect(), (status_t)OK);
uint32_t encoder_flags = 0;
if (mIsMetaDataStoredInVideoBuffers) {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index a00aaa5..e618f67 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -38,7 +38,6 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
-#include <surfaceflinger/Surface.h>
#include <gui/ISurfaceTexture.h>
#include "avc_utils.h"
@@ -387,10 +386,10 @@
audio ? "audio" : "video");
mRenderer->queueEOS(audio, UNKNOWN_ERROR);
- } else {
- CHECK_EQ((int)what, (int)ACodec::kWhatDrainThisBuffer);
-
+ } else if (what == ACodec::kWhatDrainThisBuffer) {
renderBuffer(audio, codecRequest);
+ } else {
+ ALOGV("Unhandled codec notification %d.", what);
}
break;
@@ -480,7 +479,7 @@
// completed.
ALOGV("postponing reset mFlushingAudio=%d, mFlushingVideo=%d",
- mFlushingAudio, mFlushingVideo);
+ mFlushingAudio, mFlushingVideo);
mResetPostponed = true;
break;
@@ -690,7 +689,7 @@
bool timeChange = (type & ATSParser::DISCONTINUITY_TIME) != 0;
ALOGI("%s discontinuity (formatChange=%d, time=%d)",
- audio ? "audio" : "video", formatChange, timeChange);
+ audio ? "audio" : "video", formatChange, timeChange);
if (audio) {
mSkipRenderingAudioUntilMediaTimeUs = -1;
@@ -768,7 +767,7 @@
mediaTimeUs / 1E6);
#endif
- reply->setObject("buffer", accessUnit);
+ reply->setBuffer("buffer", accessUnit);
reply->post();
return OK;
@@ -793,10 +792,8 @@
return;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
-
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
int64_t &skipUntilMediaTimeUs =
audio
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index ffc710e..6be14be 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -21,8 +21,6 @@
#include <media/MediaPlayerInterface.h>
#include <media/stagefright/foundation/AHandler.h>
#include <media/stagefright/NativeWindowWrapper.h>
-#include <gui/SurfaceTextureClient.h>
-#include <surfaceflinger/Surface.h>
namespace android {
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 56c2773..460fc98 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -29,8 +29,6 @@
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
-#include <surfaceflinger/Surface.h>
-#include <gui/ISurfaceTexture.h>
namespace android {
@@ -214,8 +212,6 @@
buffer->meta()->setInt32("csd", true);
mCSD.push(buffer);
-
- msg->setObject("csd", buffer);
} else if (meta->findData(kKeyESDS, &type, &data, &size)) {
ESDS esds((const char *)data, size);
CHECK_EQ(esds.InitCheck(), (status_t)OK);
@@ -242,9 +238,8 @@
CHECK(msg->findMessage("reply", &reply));
#if 0
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<ABuffer> outBuffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> outBuffer;
+ CHECK(msg->findBuffer("buffer", &outBuffer));
#else
sp<ABuffer> outBuffer;
#endif
@@ -253,7 +248,7 @@
outBuffer = mCSD.editItemAt(mCSDIndex++);
outBuffer->meta()->setInt64("timeUs", 0);
- reply->setObject("buffer", outBuffer);
+ reply->setBuffer("buffer", outBuffer);
reply->post();
return;
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
index 074cb4f..5738ecb 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp
@@ -60,7 +60,7 @@
const sp<AMessage> ¬ifyConsumed) {
sp<AMessage> msg = new AMessage(kWhatQueueBuffer, id());
msg->setInt32("audio", static_cast<int32_t>(audio));
- msg->setObject("buffer", buffer);
+ msg->setBuffer("buffer", buffer);
msg->setMessage("notifyConsumed", notifyConsumed);
msg->post();
}
@@ -376,7 +376,7 @@
bool tooLate = (mVideoLateByUs > 40000);
if (tooLate) {
- ALOGV("video late by %lld us (%.2f secs)", lateByUs, lateByUs / 1E6);
+ ALOGV("video late by %lld us (%.2f secs)", mVideoLateByUs, mVideoLateByUs / 1E6);
} else {
ALOGV("rendering video at media time %.2f secs", mediaTimeUs / 1E6);
}
@@ -411,9 +411,8 @@
return;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
sp<AMessage> notifyConsumed;
CHECK(msg->findMessage("notifyConsumed", ¬ifyConsumed));
@@ -629,7 +628,7 @@
}
ALOGV("now paused audio queue has %d entries, video has %d entries",
- mAudioQueue.size(), mVideoQueue.size());
+ mAudioQueue.size(), mVideoQueue.size());
mPaused = true;
}
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index 6eb0d07..4c65b65 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -218,10 +218,8 @@
CHECK(msg->findSize("trackIndex", &trackIndex));
CHECK_LT(trackIndex, mTracks.size());
- sp<RefBase> obj;
- CHECK(msg->findObject("accessUnit", &obj));
-
- sp<ABuffer> accessUnit = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("accessUnit", &accessUnit));
int32_t damaged;
if (accessUnit->meta()->findInt32("damaged", &damaged)
diff --git a/media/libstagefright/AACExtractor.cpp b/media/libstagefright/AACExtractor.cpp
index 52b1200..4d1072f 100644
--- a/media/libstagefright/AACExtractor.cpp
+++ b/media/libstagefright/AACExtractor.cpp
@@ -22,9 +22,10 @@
#include "include/avc_utils.h"
#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
@@ -131,18 +132,28 @@
return frameSize;
}
-AACExtractor::AACExtractor(const sp<DataSource> &source)
+AACExtractor::AACExtractor(
+ const sp<DataSource> &source, const sp<AMessage> &_meta)
: mDataSource(source),
mInitCheck(NO_INIT),
mFrameDurationUs(0) {
- String8 mimeType;
- float confidence;
- if (!SniffAAC(mDataSource, &mimeType, &confidence, NULL)) {
- return;
+ sp<AMessage> meta = _meta;
+
+ if (meta == NULL) {
+ String8 mimeType;
+ float confidence;
+ sp<AMessage> _meta;
+
+ if (!SniffAAC(mDataSource, &mimeType, &confidence, &meta)) {
+ return;
+ }
}
+ int64_t offset;
+ CHECK(meta->findInt64("offset", &offset));
+
uint8_t profile, sf_index, channel, header[2];
- if (mDataSource->readAt(2, &header, 2) < 2) {
+ if (mDataSource->readAt(offset + 2, &header, 2) < 2) {
return;
}
@@ -156,7 +167,6 @@
mMeta = MakeAACCodecSpecificData(profile, sf_index, channel);
- off64_t offset = 0;
off64_t streamSize, numFrames = 0;
size_t frameSize = 0;
int64_t duration = 0;
@@ -245,7 +255,12 @@
status_t AACSource::start(MetaData *params) {
CHECK(!mStarted);
- mOffset = 0;
+ if (mOffsetVector.empty()) {
+ mOffset = 0;
+ } else {
+ mOffset = mOffsetVector.itemAt(0);
+ }
+
mCurrentTimeUs = 0;
mGroup = new MediaBufferGroup;
mGroup->add_buffer(new MediaBuffer(kMaxFrameSize));
@@ -318,10 +333,39 @@
bool SniffAAC(
const sp<DataSource> &source, String8 *mimeType, float *confidence,
- sp<AMessage> *) {
+ sp<AMessage> *meta) {
+ off64_t pos = 0;
+
+ for (;;) {
+ uint8_t id3header[10];
+ if (source->readAt(pos, id3header, sizeof(id3header))
+ < (ssize_t)sizeof(id3header)) {
+ return false;
+ }
+
+ if (memcmp("ID3", id3header, 3)) {
+ break;
+ }
+
+ // Skip the ID3v2 header.
+
+ size_t len =
+ ((id3header[6] & 0x7f) << 21)
+ | ((id3header[7] & 0x7f) << 14)
+ | ((id3header[8] & 0x7f) << 7)
+ | (id3header[9] & 0x7f);
+
+ len += 10;
+
+ pos += len;
+
+ ALOGV("skipped ID3 tag, new starting offset is %lld (0x%016llx)",
+ pos, pos);
+ }
+
uint8_t header[2];
- if (source->readAt(0, &header, 2) != 2) {
+ if (source->readAt(pos, &header, 2) != 2) {
return false;
}
@@ -329,6 +373,10 @@
if ((header[0] == 0xff) && ((header[1] & 0xf6) == 0xf0)) {
*mimeType = MEDIA_MIMETYPE_AUDIO_AAC_ADTS;
*confidence = 0.2;
+
+ *meta = new AMessage;
+ (*meta)->setInt64("offset", pos);
+
return true;
}
diff --git a/media/libstagefright/AACWriter.cpp b/media/libstagefright/AACWriter.cpp
index 1673ccd..9cdb463 100644
--- a/media/libstagefright/AACWriter.cpp
+++ b/media/libstagefright/AACWriter.cpp
@@ -60,7 +60,7 @@
AACWriter::~AACWriter() {
if (mStarted) {
- stop();
+ reset();
}
if (mFd != -1) {
@@ -152,7 +152,7 @@
return OK;
}
-status_t AACWriter::stop() {
+status_t AACWriter::reset() {
if (!mStarted) {
return OK;
}
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index ca44ea3..85bd7ba 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -31,9 +31,6 @@
#include <media/stagefright/OMXClient.h>
#include <media/stagefright/OMXCodec.h>
-#include <surfaceflinger/Surface.h>
-#include <gui/SurfaceTextureClient.h>
-
#include <OMX_Component.h>
namespace android {
@@ -171,6 +168,9 @@
private:
void onSetup(const sp<AMessage> &msg);
+ void onAllocateComponent(const sp<AMessage> &msg);
+ void onConfigureComponent(const sp<AMessage> &msg);
+ void onStart();
DISALLOW_EVIL_CONSTRUCTORS(UninitializedState);
};
@@ -265,6 +265,8 @@
private:
void changeStateIfWeOwnAllBuffers();
+ bool mComponentNowIdle;
+
DISALLOW_EVIL_CONSTRUCTORS(ExecutingToIdleState);
};
@@ -309,7 +311,8 @@
ACodec::ACodec()
: mNode(NULL),
- mSentFormat(false) {
+ mSentFormat(false),
+ mIsEncoder(false) {
mUninitializedState = new UninitializedState(this);
mLoadedToIdleState = new LoadedToIdleState(this);
mIdleToExecutingState = new IdleToExecutingState(this);
@@ -341,6 +344,22 @@
msg->post();
}
+void ACodec::initiateAllocateComponent(const sp<AMessage> &msg) {
+ msg->setWhat(kWhatAllocateComponent);
+ msg->setTarget(id());
+ msg->post();
+}
+
+void ACodec::initiateConfigureComponent(const sp<AMessage> &msg) {
+ msg->setWhat(kWhatConfigureComponent);
+ msg->setTarget(id());
+ msg->post();
+}
+
+void ACodec::initiateStart() {
+ (new AMessage(kWhatStart, id()))->post();
+}
+
void ACodec::signalFlush() {
ALOGV("[%s] signalFlush", mComponentName.c_str());
(new AMessage(kWhatFlush, id()))->post();
@@ -360,62 +379,75 @@
CHECK(mDealer[portIndex] == NULL);
CHECK(mBuffers[portIndex].isEmpty());
+ status_t err;
if (mNativeWindow != NULL && portIndex == kPortIndexOutput) {
- return allocateOutputBuffersFromNativeWindow();
+ err = allocateOutputBuffersFromNativeWindow();
+ } else {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+ def.nPortIndex = portIndex;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err == OK) {
+ ALOGV("[%s] Allocating %lu buffers of size %lu on %s port",
+ mComponentName.c_str(),
+ def.nBufferCountActual, def.nBufferSize,
+ portIndex == kPortIndexInput ? "input" : "output");
+
+ size_t totalSize = def.nBufferCountActual * def.nBufferSize;
+ mDealer[portIndex] = new MemoryDealer(totalSize, "ACodec");
+
+ for (OMX_U32 i = 0; i < def.nBufferCountActual; ++i) {
+ sp<IMemory> mem = mDealer[portIndex]->allocate(def.nBufferSize);
+ CHECK(mem.get() != NULL);
+
+ IOMX::buffer_id buffer;
+
+ if (!strncasecmp(
+ mComponentName.c_str(), "OMX.TI.DUCATI1.VIDEO.", 21)) {
+ if (portIndex == kPortIndexInput && i == 0) {
+ // Only log this warning once per allocation round.
+
+ ALOGW("OMX.TI.DUCATI1.VIDEO.* require the use of "
+ "OMX_AllocateBuffer instead of the preferred "
+ "OMX_UseBuffer. Vendor must fix this.");
+ }
+
+ err = mOMX->allocateBufferWithBackup(
+ mNode, portIndex, mem, &buffer);
+ } else {
+ err = mOMX->useBuffer(mNode, portIndex, mem, &buffer);
+ }
+
+ BufferInfo info;
+ info.mBufferID = buffer;
+ info.mStatus = BufferInfo::OWNED_BY_US;
+ info.mData = new ABuffer(mem->pointer(), def.nBufferSize);
+ mBuffers[portIndex].push(info);
+ }
+ }
}
- OMX_PARAM_PORTDEFINITIONTYPE def;
- InitOMXParams(&def);
- def.nPortIndex = portIndex;
-
- status_t err = mOMX->getParameter(
- mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
-
if (err != OK) {
return err;
}
- ALOGV("[%s] Allocating %lu buffers of size %lu on %s port",
- mComponentName.c_str(),
- def.nBufferCountActual, def.nBufferSize,
- portIndex == kPortIndexInput ? "input" : "output");
+ sp<AMessage> notify = mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatBuffersAllocated);
- size_t totalSize = def.nBufferCountActual * def.nBufferSize;
- mDealer[portIndex] = new MemoryDealer(totalSize, "OMXCodec");
+ notify->setInt32("portIndex", portIndex);
+ for (size_t i = 0; i < mBuffers[portIndex].size(); ++i) {
+ AString name = StringPrintf("buffer-id_%d", i);
+ notify->setPointer(name.c_str(), mBuffers[portIndex][i].mBufferID);
- for (OMX_U32 i = 0; i < def.nBufferCountActual; ++i) {
- sp<IMemory> mem = mDealer[portIndex]->allocate(def.nBufferSize);
- CHECK(mem.get() != NULL);
-
- IOMX::buffer_id buffer;
-
- if (!strcasecmp(
- mComponentName.c_str(), "OMX.TI.DUCATI1.VIDEO.DECODER")) {
- if (portIndex == kPortIndexInput && i == 0) {
- // Only log this warning once per allocation round.
-
- ALOGW("OMX.TI.DUCATI1.VIDEO.DECODER requires the use of "
- "OMX_AllocateBuffer instead of the preferred "
- "OMX_UseBuffer. Vendor must fix this.");
- }
-
- err = mOMX->allocateBufferWithBackup(
- mNode, portIndex, mem, &buffer);
- } else {
- err = mOMX->useBuffer(mNode, portIndex, mem, &buffer);
- }
-
- if (err != OK) {
- return err;
- }
-
- BufferInfo info;
- info.mBufferID = buffer;
- info.mStatus = BufferInfo::OWNED_BY_US;
- info.mData = new ABuffer(mem->pointer(), def.nBufferSize);
- mBuffers[portIndex].push(info);
+ name = StringPrintf("data_%d", i);
+ notify->setBuffer(name.c_str(), mBuffers[portIndex][i].mData);
}
+ notify->post();
+
return OK;
}
@@ -671,7 +703,7 @@
return NULL;
}
-void ACodec::setComponentRole(
+status_t ACodec::setComponentRole(
bool isEncoder, const char *mime) {
struct MimeToRole {
const char *mime;
@@ -700,6 +732,8 @@
"video_decoder.mpeg4", "video_encoder.mpeg4" },
{ MEDIA_MIMETYPE_VIDEO_H263,
"video_decoder.h263", "video_encoder.h263" },
+ { MEDIA_MIMETYPE_VIDEO_VPX,
+ "video_decoder.vpx", "video_encoder.vpx" },
};
static const size_t kNumMimeToRole =
@@ -713,7 +747,7 @@
}
if (i == kNumMimeToRole) {
- return;
+ return ERROR_UNSUPPORTED;
}
const char *role =
@@ -736,50 +770,83 @@
if (err != OK) {
ALOGW("[%s] Failed to set standard component role '%s'.",
mComponentName.c_str(), role);
+
+ return err;
}
}
+
+ return OK;
}
-void ACodec::configureCodec(
+status_t ACodec::configureCodec(
const char *mime, const sp<AMessage> &msg) {
- setComponentRole(false /* isEncoder */, mime);
+ int32_t encoder;
+ if (!msg->findInt32("encoder", &encoder)) {
+ encoder = false;
+ }
+
+ mIsEncoder = encoder;
+
+ status_t err = setComponentRole(encoder /* isEncoder */, mime);
+
+ if (err != OK) {
+ return err;
+ }
+
+ int32_t bitRate = 0;
+ if (encoder && !msg->findInt32("bitrate", &bitRate)) {
+ return INVALID_OPERATION;
+ }
if (!strncasecmp(mime, "video/", 6)) {
- int32_t width, height;
- CHECK(msg->findInt32("width", &width));
- CHECK(msg->findInt32("height", &height));
-
- CHECK_EQ(setupVideoDecoder(mime, width, height),
- (status_t)OK);
+ if (encoder) {
+ err = setupVideoEncoder(mime, msg);
+ } else {
+ int32_t width, height;
+ if (!msg->findInt32("width", &width)
+ || !msg->findInt32("height", &height)) {
+ err = INVALID_OPERATION;
+ } else {
+ err = setupVideoDecoder(mime, width, height);
+ }
+ }
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC)) {
int32_t numChannels, sampleRate;
- CHECK(msg->findInt32("channel-count", &numChannels));
- CHECK(msg->findInt32("sample-rate", &sampleRate));
-
- CHECK_EQ(setupAACDecoder(numChannels, sampleRate), (status_t)OK);
+ if (!msg->findInt32("channel-count", &numChannels)
+ || !msg->findInt32("sample-rate", &sampleRate)) {
+ err = INVALID_OPERATION;
+ } else {
+ err = setupAACCodec(encoder, numChannels, sampleRate, bitRate);
+ }
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AMR_NB)) {
- CHECK_EQ(setupAMRDecoder(false /* isWAMR */), (status_t)OK);
+ err = setupAMRCodec(encoder, false /* isWAMR */, bitRate);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AMR_WB)) {
- CHECK_EQ(setupAMRDecoder(true /* isWAMR */), (status_t)OK);
+ err = setupAMRCodec(encoder, true /* isWAMR */, bitRate);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_G711_ALAW)
|| !strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_G711_MLAW)) {
// These are PCM-like formats with a fixed sample rate but
// a variable number of channels.
int32_t numChannels;
- CHECK(msg->findInt32("channel-count", &numChannels));
+ if (!msg->findInt32("channel-count", &numChannels)) {
+ err = INVALID_OPERATION;
+ } else {
+ err = setupG711Codec(encoder, numChannels);
+ }
+ }
- CHECK_EQ(setupG711Decoder(numChannels), (status_t)OK);
+ if (err != OK) {
+ return err;
}
int32_t maxInputSize;
if (msg->findInt32("max-input-size", &maxInputSize)) {
- CHECK_EQ(setMinBufferSize(kPortIndexInput, (size_t)maxInputSize),
- (status_t)OK);
+ err = setMinBufferSize(kPortIndexInput, (size_t)maxInputSize);
} else if (!strcmp("OMX.Nvidia.aac.decoder", mComponentName.c_str())) {
- CHECK_EQ(setMinBufferSize(kPortIndexInput, 8192), // XXX
- (status_t)OK);
+ err = setMinBufferSize(kPortIndexInput, 8192); // XXX
}
+
+ return err;
}
status_t ACodec::setMinBufferSize(OMX_U32 portIndex, size_t size) {
@@ -819,12 +886,113 @@
return OK;
}
-status_t ACodec::setupAACDecoder(int32_t numChannels, int32_t sampleRate) {
+status_t ACodec::selectAudioPortFormat(
+ OMX_U32 portIndex, OMX_AUDIO_CODINGTYPE desiredFormat) {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE format;
+ InitOMXParams(&format);
+
+ format.nPortIndex = portIndex;
+ for (OMX_U32 index = 0;; ++index) {
+ format.nIndex = index;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamAudioPortFormat,
+ &format, sizeof(format));
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (format.eEncoding == desiredFormat) {
+ break;
+ }
+ }
+
+ return mOMX->setParameter(
+ mNode, OMX_IndexParamAudioPortFormat, &format, sizeof(format));
+}
+
+status_t ACodec::setupAACCodec(
+ bool encoder,
+ int32_t numChannels, int32_t sampleRate, int32_t bitRate) {
+ status_t err = setupRawAudioFormat(
+ encoder ? kPortIndexInput : kPortIndexOutput,
+ sampleRate,
+ numChannels);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (encoder) {
+ err = selectAudioPortFormat(kPortIndexOutput, OMX_AUDIO_CodingAAC);
+
+ if (err != OK) {
+ return err;
+ }
+
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+ def.nPortIndex = kPortIndexOutput;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ def.format.audio.bFlagErrorConcealment = OMX_TRUE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ OMX_AUDIO_PARAM_AACPROFILETYPE profile;
+ InitOMXParams(&profile);
+ profile.nPortIndex = kPortIndexOutput;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
+
+ if (err != OK) {
+ return err;
+ }
+
+ profile.nChannels = numChannels;
+
+ profile.eChannelMode =
+ (numChannels == 1)
+ ? OMX_AUDIO_ChannelModeMono: OMX_AUDIO_ChannelModeStereo;
+
+ profile.nSampleRate = sampleRate;
+ profile.nBitRate = bitRate;
+ profile.nAudioBandWidth = 0;
+ profile.nFrameLength = 0;
+ profile.nAACtools = OMX_AUDIO_AACToolAll;
+ profile.nAACERtools = OMX_AUDIO_AACERNone;
+ profile.eAACProfile = OMX_AUDIO_AACObjectLC;
+ profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4FF;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
+
+ if (err != OK) {
+ return err;
+ }
+
+ return err;
+ }
+
OMX_AUDIO_PARAM_AACPROFILETYPE profile;
InitOMXParams(&profile);
profile.nPortIndex = kPortIndexInput;
- status_t err = mOMX->getParameter(
+ err = mOMX->getParameter(
mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
if (err != OK) {
@@ -835,16 +1003,59 @@
profile.nSampleRate = sampleRate;
profile.eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4ADTS;
- err = mOMX->setParameter(
+ return mOMX->setParameter(
mNode, OMX_IndexParamAudioAac, &profile, sizeof(profile));
-
- return err;
}
-status_t ACodec::setupAMRDecoder(bool isWAMR) {
+static OMX_AUDIO_AMRBANDMODETYPE pickModeFromBitRate(
+ bool isAMRWB, int32_t bps) {
+ if (isAMRWB) {
+ if (bps <= 6600) {
+ return OMX_AUDIO_AMRBandModeWB0;
+ } else if (bps <= 8850) {
+ return OMX_AUDIO_AMRBandModeWB1;
+ } else if (bps <= 12650) {
+ return OMX_AUDIO_AMRBandModeWB2;
+ } else if (bps <= 14250) {
+ return OMX_AUDIO_AMRBandModeWB3;
+ } else if (bps <= 15850) {
+ return OMX_AUDIO_AMRBandModeWB4;
+ } else if (bps <= 18250) {
+ return OMX_AUDIO_AMRBandModeWB5;
+ } else if (bps <= 19850) {
+ return OMX_AUDIO_AMRBandModeWB6;
+ } else if (bps <= 23050) {
+ return OMX_AUDIO_AMRBandModeWB7;
+ }
+
+ // 23850 bps
+ return OMX_AUDIO_AMRBandModeWB8;
+ } else { // AMRNB
+ if (bps <= 4750) {
+ return OMX_AUDIO_AMRBandModeNB0;
+ } else if (bps <= 5150) {
+ return OMX_AUDIO_AMRBandModeNB1;
+ } else if (bps <= 5900) {
+ return OMX_AUDIO_AMRBandModeNB2;
+ } else if (bps <= 6700) {
+ return OMX_AUDIO_AMRBandModeNB3;
+ } else if (bps <= 7400) {
+ return OMX_AUDIO_AMRBandModeNB4;
+ } else if (bps <= 7950) {
+ return OMX_AUDIO_AMRBandModeNB5;
+ } else if (bps <= 10200) {
+ return OMX_AUDIO_AMRBandModeNB6;
+ }
+
+ // 12200 bps
+ return OMX_AUDIO_AMRBandModeNB7;
+ }
+}
+
+status_t ACodec::setupAMRCodec(bool encoder, bool isWAMR, int32_t bitrate) {
OMX_AUDIO_PARAM_AMRTYPE def;
InitOMXParams(&def);
- def.nPortIndex = kPortIndexInput;
+ def.nPortIndex = encoder ? kPortIndexOutput : kPortIndexInput;
status_t err =
mOMX->getParameter(mNode, OMX_IndexParamAudioAmr, &def, sizeof(def));
@@ -854,14 +1065,24 @@
}
def.eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+ def.eAMRBandMode = pickModeFromBitRate(isWAMR, bitrate);
- def.eAMRBandMode =
- isWAMR ? OMX_AUDIO_AMRBandModeWB0 : OMX_AUDIO_AMRBandModeNB0;
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamAudioAmr, &def, sizeof(def));
- return mOMX->setParameter(mNode, OMX_IndexParamAudioAmr, &def, sizeof(def));
+ if (err != OK) {
+ return err;
+ }
+
+ return setupRawAudioFormat(
+ encoder ? kPortIndexInput : kPortIndexOutput,
+ isWAMR ? 16000 : 8000 /* sampleRate */,
+ 1 /* numChannels */);
}
-status_t ACodec::setupG711Decoder(int32_t numChannels) {
+status_t ACodec::setupG711Codec(bool encoder, int32_t numChannels) {
+ CHECK(!encoder); // XXX TODO
+
return setupRawAudioFormat(
kPortIndexInput, 8000 /* sampleRate */, numChannels);
}
@@ -1001,22 +1222,36 @@
&format, sizeof(format));
}
-status_t ACodec::setupVideoDecoder(
- const char *mime, int32_t width, int32_t height) {
- OMX_VIDEO_CODINGTYPE compressionFormat = OMX_VIDEO_CodingUnused;
+static status_t GetVideoCodingTypeFromMime(
+ const char *mime, OMX_VIDEO_CODINGTYPE *codingType) {
if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_AVC, mime)) {
- compressionFormat = OMX_VIDEO_CodingAVC;
+ *codingType = OMX_VIDEO_CodingAVC;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_MPEG4, mime)) {
- compressionFormat = OMX_VIDEO_CodingMPEG4;
+ *codingType = OMX_VIDEO_CodingMPEG4;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_H263, mime)) {
- compressionFormat = OMX_VIDEO_CodingH263;
+ *codingType = OMX_VIDEO_CodingH263;
} else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_MPEG2, mime)) {
- compressionFormat = OMX_VIDEO_CodingMPEG2;
+ *codingType = OMX_VIDEO_CodingMPEG2;
+ } else if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_VPX, mime)) {
+ *codingType = OMX_VIDEO_CodingVPX;
} else {
- TRESPASS();
+ *codingType = OMX_VIDEO_CodingUnused;
+ return ERROR_UNSUPPORTED;
}
- status_t err = setVideoPortFormatType(
+ return OK;
+}
+
+status_t ACodec::setupVideoDecoder(
+ const char *mime, int32_t width, int32_t height) {
+ OMX_VIDEO_CODINGTYPE compressionFormat;
+ status_t err = GetVideoCodingTypeFromMime(mime, &compressionFormat);
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = setVideoPortFormatType(
kPortIndexInput, compressionFormat, OMX_COLOR_FormatUnused);
if (err != OK) {
@@ -1046,6 +1281,489 @@
return OK;
}
+status_t ACodec::setupVideoEncoder(const char *mime, const sp<AMessage> &msg) {
+ int32_t tmp;
+ if (!msg->findInt32("color-format", &tmp)) {
+ return INVALID_OPERATION;
+ }
+
+ OMX_COLOR_FORMATTYPE colorFormat =
+ static_cast<OMX_COLOR_FORMATTYPE>(tmp);
+
+ status_t err = setVideoPortFormatType(
+ kPortIndexInput, OMX_VIDEO_CodingUnused, colorFormat);
+
+ if (err != OK) {
+ ALOGE("[%s] does not support color format %d",
+ mComponentName.c_str(), colorFormat);
+
+ return err;
+ }
+
+ /* Input port configuration */
+
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ OMX_VIDEO_PORTDEFINITIONTYPE *video_def = &def.format.video;
+
+ def.nPortIndex = kPortIndexInput;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ int32_t width, height, bitrate;
+ if (!msg->findInt32("width", &width)
+ || !msg->findInt32("height", &height)
+ || !msg->findInt32("bitrate", &bitrate)) {
+ return INVALID_OPERATION;
+ }
+
+ video_def->nFrameWidth = width;
+ video_def->nFrameHeight = height;
+
+ int32_t stride;
+ if (!msg->findInt32("stride", &stride)) {
+ stride = width;
+ }
+
+ video_def->nStride = stride;
+
+ int32_t sliceHeight;
+ if (!msg->findInt32("slice-height", &sliceHeight)) {
+ sliceHeight = height;
+ }
+
+ video_def->nSliceHeight = sliceHeight;
+
+ def.nBufferSize = (video_def->nStride * video_def->nSliceHeight * 3) / 2;
+
+ float frameRate;
+ if (!msg->findFloat("frame-rate", &frameRate)) {
+ int32_t tmp;
+ if (!msg->findInt32("frame-rate", &tmp)) {
+ return INVALID_OPERATION;
+ }
+ frameRate = (float)tmp;
+ }
+
+ video_def->xFramerate = (OMX_U32)(frameRate * 65536.0f);
+ video_def->eCompressionFormat = OMX_VIDEO_CodingUnused;
+ video_def->eColorFormat = colorFormat;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ ALOGE("[%s] failed to set input port definition parameters.",
+ mComponentName.c_str());
+
+ return err;
+ }
+
+ /* Output port configuration */
+
+ OMX_VIDEO_CODINGTYPE compressionFormat;
+ err = GetVideoCodingTypeFromMime(mime, &compressionFormat);
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = setVideoPortFormatType(
+ kPortIndexOutput, compressionFormat, OMX_COLOR_FormatUnused);
+
+ if (err != OK) {
+ ALOGE("[%s] does not support compression format %d",
+ mComponentName.c_str(), compressionFormat);
+
+ return err;
+ }
+
+ def.nPortIndex = kPortIndexOutput;
+
+ err = mOMX->getParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ return err;
+ }
+
+ video_def->nFrameWidth = width;
+ video_def->nFrameHeight = height;
+ video_def->xFramerate = 0;
+ video_def->nBitrate = bitrate;
+ video_def->eCompressionFormat = compressionFormat;
+ video_def->eColorFormat = OMX_COLOR_FormatUnused;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamPortDefinition, &def, sizeof(def));
+
+ if (err != OK) {
+ ALOGE("[%s] failed to set output port definition parameters.",
+ mComponentName.c_str());
+
+ return err;
+ }
+
+ switch (compressionFormat) {
+ case OMX_VIDEO_CodingMPEG4:
+ err = setupMPEG4EncoderParameters(msg);
+ break;
+
+ case OMX_VIDEO_CodingH263:
+ err = setupH263EncoderParameters(msg);
+ break;
+
+ case OMX_VIDEO_CodingAVC:
+ err = setupAVCEncoderParameters(msg);
+ break;
+
+ default:
+ break;
+ }
+
+ ALOGI("setupVideoEncoder succeeded");
+
+ return err;
+}
+
+static OMX_U32 setPFramesSpacing(int32_t iFramesInterval, int32_t frameRate) {
+ if (iFramesInterval < 0) {
+ return 0xFFFFFFFF;
+ } else if (iFramesInterval == 0) {
+ return 0;
+ }
+ OMX_U32 ret = frameRate * iFramesInterval;
+ CHECK(ret > 1);
+ return ret;
+}
+
+status_t ACodec::setupMPEG4EncoderParameters(const sp<AMessage> &msg) {
+ int32_t bitrate, iFrameInterval;
+ if (!msg->findInt32("bitrate", &bitrate)
+ || !msg->findInt32("i-frame-interval", &iFrameInterval)) {
+ return INVALID_OPERATION;
+ }
+
+ float frameRate;
+ if (!msg->findFloat("frame-rate", &frameRate)) {
+ int32_t tmp;
+ if (!msg->findInt32("frame-rate", &tmp)) {
+ return INVALID_OPERATION;
+ }
+ frameRate = (float)tmp;
+ }
+
+ OMX_VIDEO_PARAM_MPEG4TYPE mpeg4type;
+ InitOMXParams(&mpeg4type);
+ mpeg4type.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoMpeg4, &mpeg4type, sizeof(mpeg4type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ mpeg4type.nSliceHeaderSpacing = 0;
+ mpeg4type.bSVH = OMX_FALSE;
+ mpeg4type.bGov = OMX_FALSE;
+
+ mpeg4type.nAllowedPictureTypes =
+ OMX_VIDEO_PictureTypeI | OMX_VIDEO_PictureTypeP;
+
+ mpeg4type.nPFrames = setPFramesSpacing(iFrameInterval, frameRate);
+ if (mpeg4type.nPFrames == 0) {
+ mpeg4type.nAllowedPictureTypes = OMX_VIDEO_PictureTypeI;
+ }
+ mpeg4type.nBFrames = 0;
+ mpeg4type.nIDCVLCThreshold = 0;
+ mpeg4type.bACPred = OMX_TRUE;
+ mpeg4type.nMaxPacketSize = 256;
+ mpeg4type.nTimeIncRes = 1000;
+ mpeg4type.nHeaderExtension = 0;
+ mpeg4type.bReversibleVLC = OMX_FALSE;
+
+ int32_t profile;
+ if (msg->findInt32("profile", &profile)) {
+ int32_t level;
+ if (!msg->findInt32("level", &level)) {
+ return INVALID_OPERATION;
+ }
+
+ err = verifySupportForProfileAndLevel(profile, level);
+
+ if (err != OK) {
+ return err;
+ }
+
+ mpeg4type.eProfile = static_cast<OMX_VIDEO_MPEG4PROFILETYPE>(profile);
+ mpeg4type.eLevel = static_cast<OMX_VIDEO_MPEG4LEVELTYPE>(level);
+ }
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamVideoMpeg4, &mpeg4type, sizeof(mpeg4type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = configureBitrate(bitrate);
+
+ if (err != OK) {
+ return err;
+ }
+
+ return setupErrorCorrectionParameters();
+}
+
+status_t ACodec::setupH263EncoderParameters(const sp<AMessage> &msg) {
+ int32_t bitrate, iFrameInterval;
+ if (!msg->findInt32("bitrate", &bitrate)
+ || !msg->findInt32("i-frame-interval", &iFrameInterval)) {
+ return INVALID_OPERATION;
+ }
+
+ float frameRate;
+ if (!msg->findFloat("frame-rate", &frameRate)) {
+ int32_t tmp;
+ if (!msg->findInt32("frame-rate", &tmp)) {
+ return INVALID_OPERATION;
+ }
+ frameRate = (float)tmp;
+ }
+
+ OMX_VIDEO_PARAM_H263TYPE h263type;
+ InitOMXParams(&h263type);
+ h263type.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoH263, &h263type, sizeof(h263type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ h263type.nAllowedPictureTypes =
+ OMX_VIDEO_PictureTypeI | OMX_VIDEO_PictureTypeP;
+
+ h263type.nPFrames = setPFramesSpacing(iFrameInterval, frameRate);
+ if (h263type.nPFrames == 0) {
+ h263type.nAllowedPictureTypes = OMX_VIDEO_PictureTypeI;
+ }
+ h263type.nBFrames = 0;
+
+ int32_t profile;
+ if (msg->findInt32("profile", &profile)) {
+ int32_t level;
+ if (!msg->findInt32("level", &level)) {
+ return INVALID_OPERATION;
+ }
+
+ err = verifySupportForProfileAndLevel(profile, level);
+
+ if (err != OK) {
+ return err;
+ }
+
+ h263type.eProfile = static_cast<OMX_VIDEO_H263PROFILETYPE>(profile);
+ h263type.eLevel = static_cast<OMX_VIDEO_H263LEVELTYPE>(level);
+ }
+
+ h263type.bPLUSPTYPEAllowed = OMX_FALSE;
+ h263type.bForceRoundingTypeToZero = OMX_FALSE;
+ h263type.nPictureHeaderRepetition = 0;
+ h263type.nGOBHeaderInterval = 0;
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamVideoH263, &h263type, sizeof(h263type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ err = configureBitrate(bitrate);
+
+ if (err != OK) {
+ return err;
+ }
+
+ return setupErrorCorrectionParameters();
+}
+
+status_t ACodec::setupAVCEncoderParameters(const sp<AMessage> &msg) {
+ int32_t bitrate, iFrameInterval;
+ if (!msg->findInt32("bitrate", &bitrate)
+ || !msg->findInt32("i-frame-interval", &iFrameInterval)) {
+ return INVALID_OPERATION;
+ }
+
+ float frameRate;
+ if (!msg->findFloat("frame-rate", &frameRate)) {
+ int32_t tmp;
+ if (!msg->findInt32("frame-rate", &tmp)) {
+ return INVALID_OPERATION;
+ }
+ frameRate = (float)tmp;
+ }
+
+ OMX_VIDEO_PARAM_AVCTYPE h264type;
+ InitOMXParams(&h264type);
+ h264type.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoAvc, &h264type, sizeof(h264type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ h264type.nAllowedPictureTypes =
+ OMX_VIDEO_PictureTypeI | OMX_VIDEO_PictureTypeP;
+
+ int32_t profile;
+ if (msg->findInt32("profile", &profile)) {
+ int32_t level;
+ if (!msg->findInt32("level", &level)) {
+ return INVALID_OPERATION;
+ }
+
+ err = verifySupportForProfileAndLevel(profile, level);
+
+ if (err != OK) {
+ return err;
+ }
+
+ h264type.eProfile = static_cast<OMX_VIDEO_AVCPROFILETYPE>(profile);
+ h264type.eLevel = static_cast<OMX_VIDEO_AVCLEVELTYPE>(level);
+ }
+
+ // XXX
+ if (!strncmp(mComponentName.c_str(), "OMX.TI.DUCATI1", 14)) {
+ h264type.eProfile = OMX_VIDEO_AVCProfileBaseline;
+ }
+
+ if (h264type.eProfile == OMX_VIDEO_AVCProfileBaseline) {
+ h264type.nSliceHeaderSpacing = 0;
+ h264type.bUseHadamard = OMX_TRUE;
+ h264type.nRefFrames = 1;
+ h264type.nBFrames = 0;
+ h264type.nPFrames = setPFramesSpacing(iFrameInterval, frameRate);
+ if (h264type.nPFrames == 0) {
+ h264type.nAllowedPictureTypes = OMX_VIDEO_PictureTypeI;
+ }
+ h264type.nRefIdx10ActiveMinus1 = 0;
+ h264type.nRefIdx11ActiveMinus1 = 0;
+ h264type.bEntropyCodingCABAC = OMX_FALSE;
+ h264type.bWeightedPPrediction = OMX_FALSE;
+ h264type.bconstIpred = OMX_FALSE;
+ h264type.bDirect8x8Inference = OMX_FALSE;
+ h264type.bDirectSpatialTemporal = OMX_FALSE;
+ h264type.nCabacInitIdc = 0;
+ }
+
+ if (h264type.nBFrames != 0) {
+ h264type.nAllowedPictureTypes |= OMX_VIDEO_PictureTypeB;
+ }
+
+ h264type.bEnableUEP = OMX_FALSE;
+ h264type.bEnableFMO = OMX_FALSE;
+ h264type.bEnableASO = OMX_FALSE;
+ h264type.bEnableRS = OMX_FALSE;
+ h264type.bFrameMBsOnly = OMX_TRUE;
+ h264type.bMBAFF = OMX_FALSE;
+ h264type.eLoopFilterMode = OMX_VIDEO_AVCLoopFilterEnable;
+
+ if (!strcasecmp("OMX.Nvidia.h264.encoder", mComponentName.c_str())) {
+ h264type.eLevel = OMX_VIDEO_AVCLevelMax;
+ }
+
+ err = mOMX->setParameter(
+ mNode, OMX_IndexParamVideoAvc, &h264type, sizeof(h264type));
+
+ if (err != OK) {
+ return err;
+ }
+
+ return configureBitrate(bitrate);
+}
+
+status_t ACodec::verifySupportForProfileAndLevel(
+ int32_t profile, int32_t level) {
+ OMX_VIDEO_PARAM_PROFILELEVELTYPE params;
+ InitOMXParams(¶ms);
+ params.nPortIndex = kPortIndexOutput;
+
+ for (params.nProfileIndex = 0;; ++params.nProfileIndex) {
+ status_t err = mOMX->getParameter(
+ mNode,
+ OMX_IndexParamVideoProfileLevelQuerySupported,
+ ¶ms,
+ sizeof(params));
+
+ if (err != OK) {
+ return err;
+ }
+
+ int32_t supportedProfile = static_cast<int32_t>(params.eProfile);
+ int32_t supportedLevel = static_cast<int32_t>(params.eLevel);
+
+ if (profile == supportedProfile && level <= supportedLevel) {
+ return OK;
+ }
+ }
+}
+
+status_t ACodec::configureBitrate(int32_t bitrate) {
+ OMX_VIDEO_PARAM_BITRATETYPE bitrateType;
+ InitOMXParams(&bitrateType);
+ bitrateType.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoBitrate,
+ &bitrateType, sizeof(bitrateType));
+
+ if (err != OK) {
+ return err;
+ }
+
+ bitrateType.eControlRate = OMX_Video_ControlRateVariable;
+ bitrateType.nTargetBitrate = bitrate;
+
+ return mOMX->setParameter(
+ mNode, OMX_IndexParamVideoBitrate,
+ &bitrateType, sizeof(bitrateType));
+}
+
+status_t ACodec::setupErrorCorrectionParameters() {
+ OMX_VIDEO_PARAM_ERRORCORRECTIONTYPE errorCorrectionType;
+ InitOMXParams(&errorCorrectionType);
+ errorCorrectionType.nPortIndex = kPortIndexOutput;
+
+ status_t err = mOMX->getParameter(
+ mNode, OMX_IndexParamVideoErrorCorrection,
+ &errorCorrectionType, sizeof(errorCorrectionType));
+
+ if (err != OK) {
+ return OK; // Optional feature. Ignore this failure
+ }
+
+ errorCorrectionType.bEnableHEC = OMX_FALSE;
+ errorCorrectionType.bEnableResync = OMX_TRUE;
+ errorCorrectionType.nResynchMarkerSpacing = 256;
+ errorCorrectionType.bEnableDataPartitioning = OMX_FALSE;
+ errorCorrectionType.bEnableRVLC = OMX_FALSE;
+
+ return mOMX->setParameter(
+ mNode, OMX_IndexParamVideoErrorCorrection,
+ &errorCorrectionType, sizeof(errorCorrectionType));
+}
+
status_t ACodec::setVideoFormatOnPort(
OMX_U32 portIndex,
int32_t width, int32_t height, OMX_VIDEO_CODINGTYPE compressionFormat) {
@@ -1166,6 +1884,9 @@
notify->setString("mime", MEDIA_MIMETYPE_VIDEO_RAW);
notify->setInt32("width", videoDef->nFrameWidth);
notify->setInt32("height", videoDef->nFrameHeight);
+ notify->setInt32("stride", videoDef->nStride);
+ notify->setInt32("slice-height", videoDef->nSliceHeight);
+ notify->setInt32("color-format", videoDef->eColorFormat);
OMX_CONFIG_RECTTYPE rect;
InitOMXParams(&rect);
@@ -1241,10 +1962,11 @@
mSentFormat = true;
}
-void ACodec::signalError(OMX_ERRORTYPE error) {
+void ACodec::signalError(OMX_ERRORTYPE error, status_t internalError) {
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", ACodec::kWhatError);
notify->setInt32("omx-error", error);
+ notify->setInt32("err", internalError);
notify->post();
}
@@ -1417,7 +2139,7 @@
notify->setPointer("buffer-id", info->mBufferID);
info->mData->meta()->clear();
- notify->setObject("buffer", info->mData);
+ notify->setBuffer("buffer", info->mData);
sp<AMessage> reply = new AMessage(kWhatInputBufferFilled, mCodec->id());
reply->setPointer("buffer-id", info->mBufferID);
@@ -1433,18 +2155,26 @@
IOMX::buffer_id bufferID;
CHECK(msg->findPointer("buffer-id", &bufferID));
- sp<RefBase> obj;
+ sp<ABuffer> buffer;
int32_t err = OK;
- if (!msg->findObject("buffer", &obj)) {
+ bool eos = false;
+
+ if (!msg->findBuffer("buffer", &buffer)) {
CHECK(msg->findInt32("err", &err));
ALOGV("[%s] saw error %d instead of an input buffer",
mCodec->mComponentName.c_str(), err);
- obj.clear();
+ buffer.clear();
+
+ eos = true;
}
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ int32_t tmp;
+ if (buffer != NULL && buffer->meta()->findInt32("eos", &tmp) && tmp) {
+ eos = true;
+ err = ERROR_END_OF_STREAM;
+ }
BufferInfo *info = mCodec->findBufferByID(kPortIndexInput, bufferID);
CHECK_EQ((int)info->mStatus, (int)BufferInfo::OWNED_BY_UPSTREAM);
@@ -1456,7 +2186,7 @@
switch (mode) {
case KEEP_BUFFERS:
{
- if (buffer == NULL) {
+ if (eos) {
if (!mCodec->mPortEOS[kPortIndexInput]) {
mCodec->mPortEOS[kPortIndexInput] = true;
mCodec->mInputEOSResult = err;
@@ -1467,9 +2197,7 @@
case RESUBMIT_BUFFERS:
{
- if (buffer != NULL) {
- CHECK(!mCodec->mPortEOS[kPortIndexInput]);
-
+ if (buffer != NULL && !mCodec->mPortEOS[kPortIndexInput]) {
int64_t timeUs;
CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
@@ -1480,6 +2208,10 @@
flags |= OMX_BUFFERFLAG_CODECCONFIG;
}
+ if (eos) {
+ flags |= OMX_BUFFERFLAG_EOS;
+ }
+
if (buffer != info->mData) {
if (0 && !(flags & OMX_BUFFERFLAG_CODECCONFIG)) {
ALOGV("[%s] Needs to copy input data.",
@@ -1493,6 +2225,9 @@
if (flags & OMX_BUFFERFLAG_CODECCONFIG) {
ALOGV("[%s] calling emptyBuffer %p w/ codec specific data",
mCodec->mComponentName.c_str(), bufferID);
+ } else if (flags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("[%s] calling emptyBuffer %p w/ EOS",
+ mCodec->mComponentName.c_str(), bufferID);
} else {
ALOGV("[%s] calling emptyBuffer %p w/ time %lld us",
mCodec->mComponentName.c_str(), bufferID, timeUs);
@@ -1509,7 +2244,15 @@
info->mStatus = BufferInfo::OWNED_BY_COMPONENT;
- getMoreInputDataIfPossible();
+ if (!eos) {
+ getMoreInputDataIfPossible();
+ } else {
+ ALOGV("[%s] Signalled EOS on the input port",
+ mCodec->mComponentName.c_str());
+
+ mCodec->mPortEOS[kPortIndexInput] = true;
+ mCodec->mInputEOSResult = err;
+ }
} else if (!mCodec->mPortEOS[kPortIndexInput]) {
if (err != ERROR_END_OF_STREAM) {
ALOGV("[%s] Signalling EOS on the input port "
@@ -1582,8 +2325,8 @@
int64_t timeUs,
void *platformPrivate,
void *dataPtr) {
- ALOGV("[%s] onOMXFillBufferDone %p time %lld us",
- mCodec->mComponentName.c_str(), bufferID, timeUs);
+ ALOGV("[%s] onOMXFillBufferDone %p time %lld us, flags = 0x%08lx",
+ mCodec->mComponentName.c_str(), bufferID, timeUs, flags);
ssize_t index;
BufferInfo *info =
@@ -1601,46 +2344,48 @@
case RESUBMIT_BUFFERS:
{
- if (rangeLength == 0) {
- if (!(flags & OMX_BUFFERFLAG_EOS)) {
- ALOGV("[%s] calling fillBuffer %p",
- mCodec->mComponentName.c_str(), info->mBufferID);
+ if (rangeLength == 0 && !(flags & OMX_BUFFERFLAG_EOS)) {
+ ALOGV("[%s] calling fillBuffer %p",
+ mCodec->mComponentName.c_str(), info->mBufferID);
- CHECK_EQ(mCodec->mOMX->fillBuffer(
- mCodec->mNode, info->mBufferID),
- (status_t)OK);
+ CHECK_EQ(mCodec->mOMX->fillBuffer(
+ mCodec->mNode, info->mBufferID),
+ (status_t)OK);
- info->mStatus = BufferInfo::OWNED_BY_COMPONENT;
- }
- } else {
- if (!mCodec->mSentFormat) {
- mCodec->sendFormatChange();
- }
-
- if (mCodec->mNativeWindow == NULL) {
- info->mData->setRange(rangeOffset, rangeLength);
- }
-
- info->mData->meta()->setInt64("timeUs", timeUs);
-
- sp<AMessage> notify = mCodec->mNotify->dup();
- notify->setInt32("what", ACodec::kWhatDrainThisBuffer);
- notify->setPointer("buffer-id", info->mBufferID);
- notify->setObject("buffer", info->mData);
-
- sp<AMessage> reply =
- new AMessage(kWhatOutputBufferDrained, mCodec->id());
-
- reply->setPointer("buffer-id", info->mBufferID);
-
- notify->setMessage("reply", reply);
-
- notify->post();
-
- info->mStatus = BufferInfo::OWNED_BY_DOWNSTREAM;
+ info->mStatus = BufferInfo::OWNED_BY_COMPONENT;
+ break;
}
+ if (!mCodec->mIsEncoder && !mCodec->mSentFormat) {
+ mCodec->sendFormatChange();
+ }
+
+ if (mCodec->mNativeWindow == NULL) {
+ info->mData->setRange(rangeOffset, rangeLength);
+ }
+
+ info->mData->meta()->setInt64("timeUs", timeUs);
+
+ sp<AMessage> notify = mCodec->mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatDrainThisBuffer);
+ notify->setPointer("buffer-id", info->mBufferID);
+ notify->setBuffer("buffer", info->mData);
+ notify->setInt32("flags", flags);
+
+ sp<AMessage> reply =
+ new AMessage(kWhatOutputBufferDrained, mCodec->id());
+
+ reply->setPointer("buffer-id", info->mBufferID);
+
+ notify->setMessage("reply", reply);
+
+ notify->post();
+
+ info->mStatus = BufferInfo::OWNED_BY_DOWNSTREAM;
+
if (flags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("[%s] saw output EOS", mCodec->mComponentName.c_str());
+
sp<AMessage> notify = mCodec->mNotify->dup();
notify->setInt32("what", ACodec::kWhatEOS);
notify->setInt32("err", mCodec->mInputEOSResult);
@@ -1678,12 +2423,13 @@
&& msg->findInt32("render", &render) && render != 0) {
// The client wants this buffer to be rendered.
- if (mCodec->mNativeWindow->queueBuffer(
+ status_t err;
+ if ((err = mCodec->mNativeWindow->queueBuffer(
mCodec->mNativeWindow.get(),
- info->mGraphicBuffer.get()) == OK) {
+ info->mGraphicBuffer.get())) == OK) {
info->mStatus = BufferInfo::OWNED_BY_NATIVE_WINDOW;
} else {
- mCodec->signalError();
+ mCodec->signalError(OMX_ErrorUndefined, err);
info->mStatus = BufferInfo::OWNED_BY_US;
}
} else {
@@ -1758,6 +2504,27 @@
break;
}
+ case ACodec::kWhatAllocateComponent:
+ {
+ onAllocateComponent(msg);
+ handled = true;
+ break;
+ }
+
+ case ACodec::kWhatConfigureComponent:
+ {
+ onConfigureComponent(msg);
+ handled = true;
+ break;
+ }
+
+ case ACodec::kWhatStart:
+ {
+ onStart();
+ handled = true;
+ break;
+ }
+
case ACodec::kWhatShutdown:
{
sp<AMessage> notify = mCodec->mNotify->dup();
@@ -1787,27 +2554,54 @@
void ACodec::UninitializedState::onSetup(
const sp<AMessage> &msg) {
+ onAllocateComponent(msg);
+ onConfigureComponent(msg);
+ onStart();
+}
+
+void ACodec::UninitializedState::onAllocateComponent(const sp<AMessage> &msg) {
+ ALOGV("onAllocateComponent");
+
+ if (mCodec->mNode != NULL) {
+ CHECK_EQ(mCodec->mOMX->freeNode(mCodec->mNode), (status_t)OK);
+
+ mCodec->mNativeWindow.clear();
+ mCodec->mNode = NULL;
+ mCodec->mOMX.clear();
+ mCodec->mComponentName.clear();
+ }
+
OMXClient client;
CHECK_EQ(client.connect(), (status_t)OK);
sp<IOMX> omx = client.interface();
- AString mime;
- CHECK(msg->findString("mime", &mime));
-
Vector<String8> matchingCodecs;
- OMXCodec::findMatchingCodecs(
- mime.c_str(),
- false, // createEncoder
- NULL, // matchComponentName
- 0, // flags
- &matchingCodecs);
+
+ AString mime;
+
+ AString componentName;
+ if (msg->findString("componentName", &componentName)) {
+ matchingCodecs.push_back(String8(componentName.c_str()));
+ } else {
+ CHECK(msg->findString("mime", &mime));
+
+ int32_t encoder;
+ if (!msg->findInt32("encoder", &encoder)) {
+ encoder = false;
+ }
+
+ OMXCodec::findMatchingCodecs(
+ mime.c_str(),
+ encoder, // createEncoder
+ NULL, // matchComponentName
+ 0, // flags
+ &matchingCodecs);
+ }
sp<CodecObserver> observer = new CodecObserver;
IOMX::node_id node = NULL;
- AString componentName;
-
for (size_t matchIndex = 0; matchIndex < matchingCodecs.size();
++matchIndex) {
componentName = matchingCodecs.itemAt(matchIndex).string();
@@ -1826,7 +2620,12 @@
}
if (node == NULL) {
- ALOGE("Unable to instantiate a decoder for type '%s'.", mime.c_str());
+ if (!mime.empty()) {
+ ALOGE("Unable to instantiate a decoder for type '%s'.",
+ mime.c_str());
+ } else {
+ ALOGE("Unable to instantiate decoder '%s'.", componentName.c_str());
+ }
mCodec->signalError(OMX_ErrorComponentNotFound);
return;
@@ -1844,20 +2643,52 @@
mCodec->mInputEOSResult = OK;
- mCodec->configureCodec(mime.c_str(), msg);
+ {
+ sp<AMessage> notify = mCodec->mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatComponentAllocated);
+ notify->setString("componentName", mCodec->mComponentName.c_str());
+ notify->post();
+ }
+}
+
+void ACodec::UninitializedState::onConfigureComponent(
+ const sp<AMessage> &msg) {
+ ALOGV("onConfigureComponent");
+
+ CHECK(mCodec->mNode != NULL);
+
+ AString mime;
+ CHECK(msg->findString("mime", &mime));
+
+ status_t err = mCodec->configureCodec(mime.c_str(), msg);
+
+ if (err != OK) {
+ mCodec->signalError(OMX_ErrorUndefined, err);
+ return;
+ }
sp<RefBase> obj;
if (msg->findObject("native-window", &obj)
- && strncmp("OMX.google.", componentName.c_str(), 11)) {
+ && strncmp("OMX.google.", mCodec->mComponentName.c_str(), 11)) {
sp<NativeWindowWrapper> nativeWindow(
static_cast<NativeWindowWrapper *>(obj.get()));
CHECK(nativeWindow != NULL);
mCodec->mNativeWindow = nativeWindow->getNativeWindow();
}
-
CHECK_EQ((status_t)OK, mCodec->initNativeWindow());
- CHECK_EQ(omx->sendCommand(node, OMX_CommandStateSet, OMX_StateIdle),
+ {
+ sp<AMessage> notify = mCodec->mNotify->dup();
+ notify->setInt32("what", ACodec::kWhatComponentConfigured);
+ notify->post();
+ }
+}
+
+void ACodec::UninitializedState::onStart() {
+ ALOGV("onStart");
+
+ CHECK_EQ(mCodec->mOMX->sendCommand(
+ mCodec->mNode, OMX_CommandStateSet, OMX_StateIdle),
(status_t)OK);
mCodec->changeState(mCodec->mLoadedToIdleState);
@@ -1878,7 +2709,7 @@
"(error 0x%08x)",
err);
- mCodec->signalError();
+ mCodec->signalError(OMX_ErrorUndefined, err);
}
}
@@ -2202,7 +3033,7 @@
"port reconfiguration (error 0x%08x)",
err);
- mCodec->signalError();
+ mCodec->signalError(OMX_ErrorUndefined, err);
// This is technically not correct, since we were unable
// to allocate output buffers and therefore the output port
@@ -2240,7 +3071,8 @@
////////////////////////////////////////////////////////////////////////////////
ACodec::ExecutingToIdleState::ExecutingToIdleState(ACodec *codec)
- : BaseState(codec) {
+ : BaseState(codec),
+ mComponentNowIdle(false) {
}
bool ACodec::ExecutingToIdleState::onMessageReceived(const sp<AMessage> &msg) {
@@ -2274,6 +3106,7 @@
void ACodec::ExecutingToIdleState::stateEntered() {
ALOGV("[%s] Now Executing->Idle", mCodec->mComponentName.c_str());
+ mComponentNowIdle = false;
mCodec->mSentFormat = false;
}
@@ -2285,6 +3118,8 @@
CHECK_EQ(data1, (OMX_U32)OMX_CommandStateSet);
CHECK_EQ(data2, (OMX_U32)OMX_StateIdle);
+ mComponentNowIdle = true;
+
changeStateIfWeOwnAllBuffers();
return true;
@@ -2303,7 +3138,7 @@
}
void ACodec::ExecutingToIdleState::changeStateIfWeOwnAllBuffers() {
- if (mCodec->allYourBuffersAreBelongToUs()) {
+ if (mComponentNowIdle && mCodec->allYourBuffersAreBelongToUs()) {
CHECK_EQ(mCodec->mOMX->sendCommand(
mCodec->mNode, OMX_CommandStateSet, OMX_StateLoaded),
(status_t)OK);
diff --git a/media/libstagefright/AMRExtractor.cpp b/media/libstagefright/AMRExtractor.cpp
index 5a28347..03dcbf9 100644
--- a/media/libstagefright/AMRExtractor.cpp
+++ b/media/libstagefright/AMRExtractor.cpp
@@ -20,9 +20,9 @@
#include "include/AMRExtractor.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/AMRWriter.cpp b/media/libstagefright/AMRWriter.cpp
index 6c4e307..ca85640 100644
--- a/media/libstagefright/AMRWriter.cpp
+++ b/media/libstagefright/AMRWriter.cpp
@@ -14,9 +14,9 @@
* limitations under the License.
*/
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/AMRWriter.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
@@ -52,7 +52,7 @@
AMRWriter::~AMRWriter() {
if (mStarted) {
- stop();
+ reset();
}
if (mFd != -1) {
@@ -152,7 +152,7 @@
return OK;
}
-status_t AMRWriter::stop() {
+status_t AMRWriter::reset() {
if (!mStarted) {
return OK;
}
diff --git a/media/libstagefright/AVIExtractor.cpp b/media/libstagefright/AVIExtractor.cpp
index a3187b7..5a6211e 100644
--- a/media/libstagefright/AVIExtractor.cpp
+++ b/media/libstagefright/AVIExtractor.cpp
@@ -577,6 +577,7 @@
case FOURCC('a', 'v', 'c', '1'):
case FOURCC('d', 'a', 'v', 'c'):
case FOURCC('x', '2', '6', '4'):
+ case FOURCC('H', '2', '6', '4'):
case FOURCC('v', 's', 's', 'h'):
return MEDIA_MIMETYPE_VIDEO_AVC;
diff --git a/media/libstagefright/Android.mk b/media/libstagefright/Android.mk
index 690deac..cfb1e29 100644
--- a/media/libstagefright/Android.mk
+++ b/media/libstagefright/Android.mk
@@ -9,6 +9,7 @@
AACWriter.cpp \
AMRExtractor.cpp \
AMRWriter.cpp \
+ AVIExtractor.cpp \
AudioPlayer.cpp \
AudioSource.cpp \
AwesomePlayer.cpp \
@@ -28,12 +29,14 @@
MPEG4Writer.cpp \
MediaBuffer.cpp \
MediaBufferGroup.cpp \
+ MediaCodec.cpp \
MediaDefs.cpp \
MediaExtractor.cpp \
MediaSource.cpp \
MediaSourceSplitter.cpp \
MetaData.cpp \
NuCachedSource2.cpp \
+ NuMediaExtractor.cpp \
OMXClient.cpp \
OMXCodec.cpp \
OggExtractor.cpp \
@@ -60,25 +63,30 @@
$(TOP)/external/openssl/include \
LOCAL_SHARED_LIBRARIES := \
- libbinder \
- libmedia \
- libutils \
- libcutils \
- libui \
- libsonivox \
- libvorbisidec \
+ libbinder \
+ libmedia \
+ libutils \
+ libcutils \
+ libui \
+ libsonivox \
+ libvorbisidec \
libstagefright_yuv \
libcamera_client \
- libdrmframework \
- libcrypto \
- libssl \
- libgui \
+ libdrmframework \
+ libcrypto \
+ libssl \
+ libgui \
+ libstagefright_omx \
+ liblog \
+ libicuuc \
+ libicui18n \
+ libz \
+ libdl \
+ libchromium_net \
LOCAL_STATIC_LIBRARIES := \
libstagefright_color_conversion \
libstagefright_aacenc \
- libstagefright_amrnbenc \
- libstagefright_amrwbenc \
libstagefright_avcenc \
libstagefright_m4vh263enc \
libstagefright_matroska \
@@ -88,59 +96,15 @@
libstagefright_httplive \
libstagefright_id3 \
libFLAC \
+ libstagefright_chromium_http \
-################################################################################
-
-# The following was shamelessly copied from external/webkit/Android.mk and
-# currently must follow the same logic to determine how webkit was built and
-# if it's safe to link against libchromium.net
-
-# V8 also requires an ARMv7 CPU, and since we must use jsc, we cannot
-# use the Chrome http stack either.
-ifneq ($(strip $(ARCH_ARM_HAVE_ARMV7A)),true)
- USE_ALT_HTTP := true
-endif
-
-# See if the user has specified a stack they want to use
-HTTP_STACK = $(HTTP)
-# We default to the Chrome HTTP stack.
-DEFAULT_HTTP = chrome
-ALT_HTTP = android
-
-ifneq ($(HTTP_STACK),chrome)
- ifneq ($(HTTP_STACK),android)
- # No HTTP stack is specified, pickup the one we want as default.
- ifeq ($(USE_ALT_HTTP),true)
- HTTP_STACK = $(ALT_HTTP)
- else
- HTTP_STACK = $(DEFAULT_HTTP)
- endif
- endif
-endif
-
-ifeq ($(HTTP_STACK),chrome)
-
-LOCAL_SHARED_LIBRARIES += \
- liblog \
- libicuuc \
- libicui18n \
- libz \
- libdl \
-
-LOCAL_STATIC_LIBRARIES += \
- libstagefright_chromium_http
-
-LOCAL_SHARED_LIBRARIES += libstlport libchromium_net
+LOCAL_SHARED_LIBRARIES += libstlport
include external/stlport/libstlport.mk
+# TODO: Chromium is always available, so this flag can be removed.
LOCAL_CPPFLAGS += -DCHROMIUM_AVAILABLE=1
-endif # ifeq ($(HTTP_STACK),chrome)
-
-################################################################################
-
LOCAL_SHARED_LIBRARIES += \
- libstagefright_amrnb_common \
libstagefright_enc_common \
libstagefright_avc_common \
libstagefright_foundation \
diff --git a/media/libstagefright/AudioPlayer.cpp b/media/libstagefright/AudioPlayer.cpp
index 9a9c3ef..df27566 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/media/libstagefright/AudioPlayer.cpp
@@ -268,6 +268,16 @@
return mReachedEOS;
}
+status_t AudioPlayer::setPlaybackRatePermille(int32_t ratePermille) {
+ if (mAudioSink.get() != NULL) {
+ return mAudioSink->setPlaybackRatePermille(ratePermille);
+ } else if (mAudioTrack != NULL){
+ return mAudioTrack->setSampleRate(ratePermille * mSampleRate / 1000);
+ } else {
+ return NO_INIT;
+ }
+}
+
// static
size_t AudioPlayer::AudioSinkCallback(
MediaPlayerBase::AudioSink *audioSink,
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 2172cc0..5b2ea1f 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -47,7 +47,7 @@
}
AudioSource::AudioSource(
- int inputSource, uint32_t sampleRate, uint32_t channels)
+ audio_source_t inputSource, uint32_t sampleRate, uint32_t channels)
: mStarted(false),
mSampleRate(sampleRate),
mPrevSampleTimeUs(0),
@@ -72,7 +72,7 @@
AudioSource::~AudioSource() {
if (mStarted) {
- stop();
+ reset();
}
delete mRecord;
@@ -130,7 +130,7 @@
}
}
-status_t AudioSource::stop() {
+status_t AudioSource::reset() {
Mutex::Autolock autoLock(mLock);
if (!mStarted) {
return UNKNOWN_ERROR;
@@ -282,8 +282,6 @@
mPrevSampleTimeUs = mStartTimeUs;
}
- int64_t timestampUs = mPrevSampleTimeUs;
-
size_t numLostBytes = 0;
if (mNumFramesReceived > 0) { // Ignore earlier frame lost
// getInputFramesLost() returns the number of lost frames.
@@ -293,37 +291,58 @@
CHECK_EQ(numLostBytes & 1, 0u);
CHECK_EQ(audioBuffer.size & 1, 0u);
- size_t bufferSize = numLostBytes + audioBuffer.size;
- MediaBuffer *buffer = new MediaBuffer(bufferSize);
if (numLostBytes > 0) {
- memset(buffer->data(), 0, numLostBytes);
- memcpy((uint8_t *) buffer->data() + numLostBytes,
- audioBuffer.i16, audioBuffer.size);
- } else {
- if (audioBuffer.size == 0) {
- ALOGW("Nothing is available from AudioRecord callback buffer");
- buffer->release();
- return OK;
- }
- memcpy((uint8_t *) buffer->data(),
- audioBuffer.i16, audioBuffer.size);
+ // Loss of audio frames should happen rarely; thus the LOGW should
+ // not cause a logging spam
+ ALOGW("Lost audio record data: %d bytes", numLostBytes);
}
+ while (numLostBytes > 0) {
+ size_t bufferSize = numLostBytes;
+ if (numLostBytes > kMaxBufferSize) {
+ numLostBytes -= kMaxBufferSize;
+ bufferSize = kMaxBufferSize;
+ } else {
+ numLostBytes = 0;
+ }
+ MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize);
+ memset(lostAudioBuffer->data(), 0, bufferSize);
+ lostAudioBuffer->set_range(0, bufferSize);
+ queueInputBuffer_l(lostAudioBuffer, timeUs);
+ }
+
+ if (audioBuffer.size == 0) {
+ ALOGW("Nothing is available from AudioRecord callback buffer");
+ return OK;
+ }
+
+ const size_t bufferSize = audioBuffer.size;
+ MediaBuffer *buffer = new MediaBuffer(bufferSize);
+ memcpy((uint8_t *) buffer->data(),
+ audioBuffer.i16, audioBuffer.size);
buffer->set_range(0, bufferSize);
- timestampUs += ((1000000LL * (bufferSize >> 1)) +
- (mSampleRate >> 1)) / mSampleRate;
+ queueInputBuffer_l(buffer, timeUs);
+ return OK;
+}
+
+void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
+ const size_t bufferSize = buffer->range_length();
+ const size_t frameSize = mRecord->frameSize();
+ const int64_t timestampUs =
+ mPrevSampleTimeUs +
+ ((1000000LL * (bufferSize / frameSize)) +
+ (mSampleRate >> 1)) / mSampleRate;
if (mNumFramesReceived == 0) {
buffer->meta_data()->setInt64(kKeyAnchorTime, mStartTimeUs);
}
+
buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
buffer->meta_data()->setInt64(kKeyDriftTime, timeUs - mInitialReadTimeUs);
mPrevSampleTimeUs = timestampUs;
- mNumFramesReceived += buffer->range_length() / sizeof(int16_t);
+ mNumFramesReceived += bufferSize / frameSize;
mBuffersReceived.push_back(buffer);
mFrameAvailableCondition.signal();
-
- return OK;
}
void AudioSource::trackMaxAmplitude(int16_t *data, int nSamples) {
diff --git a/media/libstagefright/AwesomePlayer.cpp b/media/libstagefright/AwesomePlayer.cpp
index d0cb7ff..8cfb8d3 100644
--- a/media/libstagefright/AwesomePlayer.cpp
+++ b/media/libstagefright/AwesomePlayer.cpp
@@ -30,7 +30,7 @@
#include "include/MPEG2TSExtractor.h"
#include "include/WVMExtractor.h"
-#include "timedtext/TimedTextPlayer.h"
+#include "timedtext/TimedTextDriver.h"
#include <binder/IPCThreadState.h>
#include <binder/IServiceManager.h>
@@ -47,10 +47,8 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXCodec.h>
-#include <surfaceflinger/Surface.h>
#include <gui/ISurfaceTexture.h>
#include <gui/SurfaceTextureClient.h>
-#include <surfaceflinger/ISurfaceComposer.h>
#include <media/stagefright/foundation/AMessage.h>
@@ -192,7 +190,7 @@
mVideoBuffer(NULL),
mDecryptHandle(NULL),
mLastVideoTimeUs(-1),
- mTextPlayer(NULL) {
+ mTextDriver(NULL) {
CHECK_EQ(mClient.connect(), (status_t)OK);
DataSource::RegisterDefaultSniffers();
@@ -335,6 +333,14 @@
return UNKNOWN_ERROR;
}
+ if (extractor->getDrmFlag()) {
+ checkDrmStatus(dataSource);
+ }
+
+ return setDataSource_l(extractor);
+}
+
+void AwesomePlayer::checkDrmStatus(const sp<DataSource>& dataSource) {
dataSource->getDrmInfo(mDecryptHandle, &mDrmManagerClient);
if (mDecryptHandle != NULL) {
CHECK(mDrmManagerClient);
@@ -342,8 +348,6 @@
notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, ERROR_DRM_NO_LICENSE);
}
}
-
- return setDataSource_l(extractor);
}
status_t AwesomePlayer::setDataSource_l(const sp<MediaExtractor> &extractor) {
@@ -524,9 +528,9 @@
delete mAudioPlayer;
mAudioPlayer = NULL;
- if (mTextPlayer != NULL) {
- delete mTextPlayer;
- mTextPlayer = NULL;
+ if (mTextDriver != NULL) {
+ delete mTextDriver;
+ mTextDriver = NULL;
}
mVideoRenderer.clear();
@@ -1112,7 +1116,7 @@
}
if (mFlags & TEXTPLAYER_STARTED) {
- mTextPlayer->pause();
+ mTextDriver->pause();
modifyFlags(TEXT_RUNNING, CLEAR);
}
@@ -1266,9 +1270,9 @@
}
status_t AwesomePlayer::setTimedTextTrackIndex(int32_t index) {
- if (mTextPlayer != NULL) {
+ if (mTextDriver != NULL) {
if (index >= 0) { // to turn on a text track
- status_t err = mTextPlayer->setTimedTextTrackIndex(index);
+ status_t err = mTextDriver->setTimedTextTrackIndex(index);
if (err != OK) {
return err;
}
@@ -1284,7 +1288,7 @@
modifyFlags(TEXTPLAYER_STARTED, CLEAR);
}
- return mTextPlayer->setTimedTextTrackIndex(index);
+ return mTextDriver->setTimedTextTrackIndex(index);
}
} else {
return INVALID_OPERATION;
@@ -1313,7 +1317,7 @@
seekAudioIfNecessary_l();
if (mFlags & TEXTPLAYER_STARTED) {
- mTextPlayer->seekTo(mSeekTimeUs);
+ mTextDriver->seekToAsync(mSeekTimeUs);
}
if (!(mFlags & PLAYING)) {
@@ -1354,15 +1358,15 @@
mAudioTrack = source;
}
-void AwesomePlayer::addTextSource(sp<MediaSource> source) {
+void AwesomePlayer::addTextSource(const sp<MediaSource>& source) {
Mutex::Autolock autoLock(mTimedTextLock);
CHECK(source != NULL);
- if (mTextPlayer == NULL) {
- mTextPlayer = new TimedTextPlayer(this, mListener, &mQueue);
+ if (mTextDriver == NULL) {
+ mTextDriver = new TimedTextDriver(mListener);
}
- mTextPlayer->addTextSource(source);
+ mTextDriver->addInBandTextSource(source);
}
status_t AwesomePlayer::initAudioDecoder() {
@@ -1603,7 +1607,7 @@
mSeekTimeUs,
mSeeking == SEEK_VIDEO_ONLY
? MediaSource::ReadOptions::SEEK_NEXT_SYNC
- : MediaSource::ReadOptions::SEEK_CLOSEST_SYNC);
+ : MediaSource::ReadOptions::SEEK_CLOSEST);
}
for (;;) {
status_t err = mVideoSource->read(&mVideoBuffer, &options);
@@ -1689,7 +1693,7 @@
}
if ((mFlags & TEXTPLAYER_STARTED) && !(mFlags & (TEXT_RUNNING | SEEK_PREVIEW))) {
- mTextPlayer->resume();
+ mTextDriver->resume();
modifyFlags(TEXT_RUNNING, SET);
}
@@ -2095,7 +2099,7 @@
String8 mimeType;
float confidence;
sp<AMessage> dummy;
- bool success = SniffDRM(dataSource, &mimeType, &confidence, &dummy);
+ bool success = SniffWVM(dataSource, &mimeType, &confidence, &dummy);
if (!success
|| strcasecmp(
@@ -2115,13 +2119,8 @@
}
}
- dataSource->getDrmInfo(mDecryptHandle, &mDrmManagerClient);
-
- if (mDecryptHandle != NULL) {
- CHECK(mDrmManagerClient);
- if (RightsStatus::RIGHTS_VALID != mDecryptHandle->status) {
- notifyListener_l(MEDIA_ERROR, MEDIA_ERROR_UNKNOWN, ERROR_DRM_NO_LICENSE);
- }
+ if (extractor->getDrmFlag()) {
+ checkDrmStatus(dataSource);
}
status_t err = setDataSource_l(extractor);
@@ -2240,16 +2239,24 @@
case KEY_PARAMETER_TIMED_TEXT_ADD_OUT_OF_BAND_SOURCE:
{
Mutex::Autolock autoLock(mTimedTextLock);
- if (mTextPlayer == NULL) {
- mTextPlayer = new TimedTextPlayer(this, mListener, &mQueue);
+ if (mTextDriver == NULL) {
+ mTextDriver = new TimedTextDriver(mListener);
}
- return mTextPlayer->setParameter(key, request);
+ return mTextDriver->addOutOfBandTextSource(request);
}
case KEY_PARAMETER_CACHE_STAT_COLLECT_FREQ_MS:
{
return setCacheStatCollectFreq(request);
}
+ case KEY_PARAMETER_PLAYBACK_RATE_PERMILLE:
+ {
+ if (mAudioPlayer != NULL) {
+ return mAudioPlayer->setPlaybackRatePermille(request.readInt32());
+ } else {
+ return NO_INIT;
+ }
+ }
default:
{
return ERROR_UNSUPPORTED;
diff --git a/media/libstagefright/CameraSource.cpp b/media/libstagefright/CameraSource.cpp
index 1850c9c..2df5528 100755
--- a/media/libstagefright/CameraSource.cpp
+++ b/media/libstagefright/CameraSource.cpp
@@ -20,14 +20,14 @@
#include <OMX_Component.h>
#include <binder/IPCThreadState.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/CameraSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
#include <camera/Camera.h>
#include <camera/CameraParameters.h>
-#include <surfaceflinger/Surface.h>
+#include <gui/Surface.h>
#include <utils/String8.h>
#include <cutils/properties.h>
@@ -114,7 +114,7 @@
ALOGE("Uknown color format (%s), please add it to "
"CameraSource::getColorFormat", colorFormat);
- CHECK_EQ(0, "Unknown color format");
+ CHECK(!"Unknown color format");
}
CameraSource *CameraSource::Create() {
@@ -517,7 +517,7 @@
// This CHECK is good, since we just passed the lock/unlock
// check earlier by calling mCamera->setParameters().
- CHECK_EQ(OK, mCamera->setPreviewDisplay(mSurface));
+ CHECK_EQ((status_t)OK, mCamera->setPreviewDisplay(mSurface));
// By default, do not store metadata in video buffers
mIsMetaDataStoredInVideoBuffers = false;
@@ -548,7 +548,7 @@
CameraSource::~CameraSource() {
if (mStarted) {
- stop();
+ reset();
} else if (mInitCheck == OK) {
// Camera is initialized but because start() is never called,
// the lock on Camera is never released(). This makes sure
@@ -566,7 +566,8 @@
if (mCameraFlags & FLAGS_HOT_CAMERA) {
mCamera->unlock();
mCamera.clear();
- CHECK_EQ(OK, mCameraRecordingProxy->startRecording(new ProxyListener(this)));
+ CHECK_EQ((status_t)OK,
+ mCameraRecordingProxy->startRecording(new ProxyListener(this)));
} else {
mCamera->setListener(new CameraSourceListener(this));
mCamera->startRecording();
@@ -632,8 +633,8 @@
mCameraFlags = 0;
}
-status_t CameraSource::stop() {
- ALOGD("stop: E");
+status_t CameraSource::reset() {
+ ALOGD("reset: E");
Mutex::Autolock autoLock(mLock);
mStarted = false;
mFrameAvailableCondition.signal();
@@ -670,7 +671,7 @@
}
CHECK_EQ(mNumFramesReceived, mNumFramesEncoded + mNumFramesDropped);
- ALOGD("stop: X");
+ ALOGD("reset: X");
return OK;
}
@@ -718,7 +719,7 @@
return;
}
}
- CHECK_EQ(0, "signalBufferReturned: bogus buffer");
+ CHECK(!"signalBufferReturned: bogus buffer");
}
status_t CameraSource::read(
diff --git a/media/libstagefright/CameraSourceTimeLapse.cpp b/media/libstagefright/CameraSourceTimeLapse.cpp
index 263ab50..26ce7ae 100644
--- a/media/libstagefright/CameraSourceTimeLapse.cpp
+++ b/media/libstagefright/CameraSourceTimeLapse.cpp
@@ -20,9 +20,9 @@
#include <binder/IPCThreadState.h>
#include <binder/MemoryBase.h>
#include <binder/MemoryHeapBase.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/CameraSource.h>
#include <media/stagefright/CameraSourceTimeLapse.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MetaData.h>
#include <camera/Camera.h>
#include <camera/CameraParameters.h>
@@ -87,6 +87,10 @@
}
CameraSourceTimeLapse::~CameraSourceTimeLapse() {
+ if (mLastReadBufferCopy) {
+ mLastReadBufferCopy->release();
+ mLastReadBufferCopy = NULL;
+ }
}
void CameraSourceTimeLapse::startQuickReadReturns() {
@@ -204,15 +208,6 @@
}
}
-void CameraSourceTimeLapse::stopCameraRecording() {
- ALOGV("stopCameraRecording");
- CameraSource::stopCameraRecording();
- if (mLastReadBufferCopy) {
- mLastReadBufferCopy->release();
- mLastReadBufferCopy = NULL;
- }
-}
-
sp<IMemory> CameraSourceTimeLapse::createIMemoryCopy(
const sp<IMemory> &source_data) {
diff --git a/media/libstagefright/DRMExtractor.cpp b/media/libstagefright/DRMExtractor.cpp
index 9452ab1..524c3aa 100644
--- a/media/libstagefright/DRMExtractor.cpp
+++ b/media/libstagefright/DRMExtractor.cpp
@@ -23,6 +23,7 @@
#include <arpa/inet.h>
#include <utils/String8.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/Utils.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaSource.h>
@@ -30,7 +31,6 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <drm/drm_framework_common.h>
#include <utils/Errors.h>
@@ -282,13 +282,13 @@
if (decryptHandle != NULL) {
if (decryptHandle->decryptApiType == DecryptApiType::CONTAINER_BASED) {
*mimeType = String8("drm+container_based+") + decryptHandle->mimeType;
+ *confidence = 10.0f;
} else if (decryptHandle->decryptApiType == DecryptApiType::ELEMENTARY_STREAM_BASED) {
*mimeType = String8("drm+es_based+") + decryptHandle->mimeType;
- } else if (decryptHandle->decryptApiType == DecryptApiType::WV_BASED) {
- *mimeType = MEDIA_MIMETYPE_CONTAINER_WVM;
- ALOGW("SniffWVM: found match\n");
+ *confidence = 10.0f;
+ } else {
+ return false;
}
- *confidence = 10.0f;
return true;
}
diff --git a/media/libstagefright/DataSource.cpp b/media/libstagefright/DataSource.cpp
index 43539bb..d0a7880 100644
--- a/media/libstagefright/DataSource.cpp
+++ b/media/libstagefright/DataSource.cpp
@@ -15,6 +15,12 @@
*/
#include "include/AMRExtractor.h"
+#include "include/AVIExtractor.h"
+
+#if CHROMIUM_AVAILABLE
+#include "include/DataUriSource.h"
+#endif
+
#include "include/MP3Extractor.h"
#include "include/MPEG4Extractor.h"
#include "include/WAVExtractor.h"
@@ -26,6 +32,7 @@
#include "include/DRMExtractor.h"
#include "include/FLACExtractor.h"
#include "include/AACExtractor.h"
+#include "include/WVMExtractor.h"
#include "matroska/MatroskaExtractor.h"
@@ -112,7 +119,9 @@
RegisterSniffer(SniffMPEG2TS);
RegisterSniffer(SniffMP3);
RegisterSniffer(SniffAAC);
+ RegisterSniffer(SniffAVI);
RegisterSniffer(SniffMPEG2PS);
+ RegisterSniffer(SniffWVM);
char value[PROPERTY_VALUE_MAX];
if (property_get("drm.service.enabled", value, NULL)
@@ -134,6 +143,10 @@
return NULL;
}
source = new NuCachedSource2(httpSource);
+# if CHROMIUM_AVAILABLE
+ } else if (!strncasecmp("data:", uri, 5)) {
+ source = new DataUriSource(uri);
+#endif
} else {
// Assume it's a filename.
source = new FileSource(uri);
diff --git a/media/libstagefright/FileSource.cpp b/media/libstagefright/FileSource.cpp
index 73cb48c..73c8d03 100644
--- a/media/libstagefright/FileSource.cpp
+++ b/media/libstagefright/FileSource.cpp
@@ -14,8 +14,8 @@
* limitations under the License.
*/
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <sys/types.h>
#include <unistd.h>
#include <sys/types.h>
@@ -127,7 +127,7 @@
return OK;
}
-sp<DecryptHandle> FileSource::DrmInitialization() {
+sp<DecryptHandle> FileSource::DrmInitialization(const char *mime) {
if (mDrmManagerClient == NULL) {
mDrmManagerClient = new DrmManagerClient();
}
@@ -138,7 +138,7 @@
if (mDecryptHandle == NULL) {
mDecryptHandle = mDrmManagerClient->openDecryptSession(
- mFd, mOffset, mLength);
+ mFd, mOffset, mLength, mime);
}
if (mDecryptHandle == NULL) {
diff --git a/media/libstagefright/JPEGSource.cpp b/media/libstagefright/JPEGSource.cpp
index e818115..bafa4b2 100644
--- a/media/libstagefright/JPEGSource.cpp
+++ b/media/libstagefright/JPEGSource.cpp
@@ -18,10 +18,10 @@
#define LOG_TAG "JPEGSource"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/JPEGSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -59,7 +59,7 @@
mWidth(0),
mHeight(0),
mOffset(0) {
- CHECK_EQ(parseJPEG(), OK);
+ CHECK_EQ(parseJPEG(), (status_t)OK);
CHECK(mSource->getSize(&mSize) == OK);
}
diff --git a/media/libstagefright/MPEG2TSWriter.cpp b/media/libstagefright/MPEG2TSWriter.cpp
index 36009ab..f702376 100644
--- a/media/libstagefright/MPEG2TSWriter.cpp
+++ b/media/libstagefright/MPEG2TSWriter.cpp
@@ -244,7 +244,7 @@
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kNotifyBuffer);
- notify->setObject("buffer", out);
+ notify->setBuffer("buffer", out);
notify->setInt32("oob", true);
notify->post();
}
@@ -270,7 +270,7 @@
copy->meta()->setInt32("isSync", true);
}
- notify->setObject("buffer", copy);
+ notify->setBuffer("buffer", copy);
notify->post();
}
@@ -351,7 +351,7 @@
sp<AMessage> notify = mNotify->dup();
notify->setInt32("what", kNotifyBuffer);
- notify->setObject("buffer", mAACBuffer);
+ notify->setBuffer("buffer", mAACBuffer);
notify->post();
mAACBuffer.clear();
@@ -513,7 +513,7 @@
MPEG2TSWriter::~MPEG2TSWriter() {
if (mStarted) {
- stop();
+ reset();
}
mLooper->unregisterHandler(mReflector->id());
@@ -564,7 +564,7 @@
return OK;
}
-status_t MPEG2TSWriter::stop() {
+status_t MPEG2TSWriter::reset() {
CHECK(mStarted);
for (size_t i = 0; i < mSources.size(); ++i) {
@@ -614,10 +614,8 @@
++mNumSourcesDone;
} else if (what == SourceInfo::kNotifyBuffer) {
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
-
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
int32_t oob;
if (msg->findInt32("oob", &oob) && oob) {
diff --git a/media/libstagefright/MPEG4Extractor.cpp b/media/libstagefright/MPEG4Extractor.cpp
index 22bdd95..6c95d4e 100644
--- a/media/libstagefright/MPEG4Extractor.cpp
+++ b/media/libstagefright/MPEG4Extractor.cpp
@@ -20,7 +20,6 @@
#include "include/MPEG4Extractor.h"
#include "include/SampleTable.h"
#include "include/ESDS.h"
-#include "timedtext/TimedTextPlayer.h"
#include <arpa/inet.h>
@@ -1372,8 +1371,9 @@
uint32_t type = ntohl(buffer);
// For the 3GPP file format, the handler-type within the 'hdlr' box
- // shall be 'text'
- if (type == FOURCC('t', 'e', 'x', 't')) {
+ // shall be 'text'. We also want to support 'sbtl' handler type
+ // for a practical reason as various MPEG4 containers use it.
+ if (type == FOURCC('t', 'e', 'x', 't') || type == FOURCC('s', 'b', 't', 'l')) {
mLastTrack->meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_TEXT_3GPP);
}
@@ -2429,4 +2429,3 @@
}
} // namespace android
-
diff --git a/media/libstagefright/MPEG4Writer.cpp b/media/libstagefright/MPEG4Writer.cpp
index 06dd875..7ebbe1d 100755
--- a/media/libstagefright/MPEG4Writer.cpp
+++ b/media/libstagefright/MPEG4Writer.cpp
@@ -23,10 +23,10 @@
#include <pthread.h>
#include <sys/prctl.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MPEG4Writer.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MetaData.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
@@ -70,6 +70,10 @@
status_t dump(int fd, const Vector<String16>& args) const;
private:
+ enum {
+ kMaxCttsOffsetTimeUs = 1000000LL, // 1 second
+ };
+
MPEG4Writer *mOwner;
sp<MetaData> mMeta;
sp<MediaSource> mSource;
@@ -137,11 +141,12 @@
: sampleCount(count), sampleDuration(timescaledDur) {}
uint32_t sampleCount;
- int32_t sampleDuration; // time scale based
+ uint32_t sampleDuration; // time scale based
};
- bool mHasNegativeCttsDeltaDuration;
size_t mNumCttsTableEntries;
List<CttsTableEntry> mCttsTableEntries;
+ int64_t mMinCttsOffsetTimeUs;
+ int64_t mMaxCttsOffsetTimeUs;
// Sequence parameter set or picture parameter set
struct AVCParamSet {
@@ -172,6 +177,8 @@
// Update the audio track's drift information.
void updateDriftTime(const sp<MetaData>& meta);
+ int32_t getStartTimeOffsetScaledTime() const;
+
static void *ThreadWrapper(void *me);
status_t threadEntry();
@@ -282,7 +289,7 @@
}
MPEG4Writer::~MPEG4Writer() {
- stop();
+ reset();
while (!mTracks.empty()) {
List<Track *>::iterator it = mTracks.begin();
@@ -471,7 +478,7 @@
!param->findInt32(kKeyTimeScale, &mTimeScale)) {
mTimeScale = 1000;
}
- CHECK(mTimeScale > 0);
+ CHECK_GT(mTimeScale, 0);
ALOGV("movie time scale: %d", mTimeScale);
mStreamableFile = true;
@@ -490,7 +497,7 @@
}
mEstimatedMoovBoxSize = estimateMoovBoxSize(bitRate);
}
- CHECK(mEstimatedMoovBoxSize >= 8);
+ CHECK_GE(mEstimatedMoovBoxSize, 8);
lseek64(mFd, mFreeBoxOffset, SEEK_SET);
writeInt32(mEstimatedMoovBoxSize);
write("free", 4);
@@ -616,7 +623,7 @@
mStarted = false;
}
-status_t MPEG4Writer::stop() {
+status_t MPEG4Writer::reset() {
if (mInitCheck != OK) {
return OK;
} else {
@@ -684,7 +691,7 @@
mWriteMoovBoxToMemory = false;
if (mStreamableFile) {
- CHECK(mMoovBoxBufferOffset + 8 <= mEstimatedMoovBoxSize);
+ CHECK_LE(mMoovBoxBufferOffset + 8, mEstimatedMoovBoxSize);
// Moov box
lseek64(mFd, mFreeBoxOffset, SEEK_SET);
@@ -856,7 +863,7 @@
mOffset += length + 4;
} else {
- CHECK(length < 65536);
+ CHECK_LT(length, 65536);
uint8_t x = length >> 8;
::write(mFd, &x, 1);
@@ -1085,7 +1092,7 @@
void MPEG4Writer::setStartTimestampUs(int64_t timeUs) {
ALOGI("setStartTimestampUs: %lld", timeUs);
- CHECK(timeUs >= 0);
+ CHECK_GE(timeUs, 0ll);
Mutex::Autolock autoLock(mLock);
if (mStartTimestampUs < 0 || mStartTimestampUs > timeUs) {
mStartTimestampUs = timeUs;
@@ -1186,9 +1193,6 @@
if (mIsAudio) {
return;
}
- if (duration < 0 && !mHasNegativeCttsDeltaDuration) {
- mHasNegativeCttsDeltaDuration = true;
- }
CttsTableEntry cttsEntry(sampleCount, duration);
mCttsTableEntries.push_back(cttsEntry);
++mNumCttsTableEntries;
@@ -1218,7 +1222,7 @@
mTimeScale = timeScale;
}
- CHECK(mTimeScale > 0);
+ CHECK_GT(mTimeScale, 0);
}
void MPEG4Writer::Track::getCodecSpecificDataFromInputFormatIfPossible() {
@@ -1299,7 +1303,7 @@
}
}
- CHECK("Received a chunk for a unknown track" == 0);
+ CHECK(!"Received a chunk for a unknown track");
}
void MPEG4Writer::writeChunkToFile(Chunk* chunk) {
@@ -1509,7 +1513,6 @@
mMdatSizeBytes = 0;
mMaxChunkDurationUs = 0;
- mHasNegativeCttsDeltaDuration = false;
pthread_create(&mThread, &attr, ThreadWrapper, this);
pthread_attr_destroy(&attr);
@@ -1833,29 +1836,18 @@
int32_t nChunks = 0;
int32_t nZeroLengthFrames = 0;
int64_t lastTimestampUs = 0; // Previous sample time stamp
- int64_t lastCttsTimeUs = 0; // Previous sample time stamp
int64_t lastDurationUs = 0; // Between the previous two samples
int64_t currDurationTicks = 0; // Timescale based ticks
int64_t lastDurationTicks = 0; // Timescale based ticks
int32_t sampleCount = 1; // Sample count in the current stts table entry
- int64_t currCttsDurTicks = 0; // Timescale based ticks
- int64_t lastCttsDurTicks = 0; // Timescale based ticks
- int32_t cttsSampleCount = 1; // Sample count in the current ctts table entry
- uint32_t previousSampleSize = 0; // Size of the previous sample
+ uint32_t previousSampleSize = 0; // Size of the previous sample
int64_t previousPausedDurationUs = 0;
int64_t timestampUs = 0;
- int64_t cttsDeltaTimeUs = 0;
- bool hasBFrames = false;
+ int64_t cttsOffsetTimeUs = 0;
+ int64_t currCttsOffsetTimeTicks = 0; // Timescale based ticks
+ int64_t lastCttsOffsetTimeTicks = -1; // Timescale based ticks
+ int32_t cttsSampleCount = 0; // Sample count in the current ctts table entry
-#if 1
- // XXX: Samsung's video encoder's output buffer timestamp
- // is not correct. see bug 4724339
- char value[PROPERTY_VALUE_MAX];
- if (property_get("rw.media.record.hasb", value, NULL) &&
- (!strcasecmp(value, "true") || !strcasecmp(value, "1"))) {
- hasBFrames = true;
- }
-#endif
if (mIsAudio) {
prctl(PR_SET_NAME, (unsigned long)"AudioTrackEncoding", 0, 0, 0);
} else {
@@ -1897,7 +1889,7 @@
(const uint8_t *)buffer->data()
+ buffer->range_offset(),
buffer->range_length());
- CHECK_EQ(OK, err);
+ CHECK_EQ((status_t)OK, err);
} else if (mIsMPEG4) {
mCodecSpecificDataSize = buffer->range_length();
mCodecSpecificData = malloc(mCodecSpecificDataSize);
@@ -1963,32 +1955,64 @@
if (mResumed) {
int64_t durExcludingEarlierPausesUs = timestampUs - previousPausedDurationUs;
- CHECK(durExcludingEarlierPausesUs >= 0);
+ CHECK_GE(durExcludingEarlierPausesUs, 0ll);
int64_t pausedDurationUs = durExcludingEarlierPausesUs - mTrackDurationUs;
- CHECK(pausedDurationUs >= lastDurationUs);
+ CHECK_GE(pausedDurationUs, lastDurationUs);
previousPausedDurationUs += pausedDurationUs - lastDurationUs;
mResumed = false;
}
timestampUs -= previousPausedDurationUs;
- CHECK(timestampUs >= 0);
- if (!mIsAudio && hasBFrames) {
+ CHECK_GE(timestampUs, 0ll);
+ if (!mIsAudio) {
/*
* Composition time: timestampUs
* Decoding time: decodingTimeUs
- * Composition time delta = composition time - decoding time
- *
- * We save picture decoding time stamp delta in stts table entries,
- * and composition time delta duration in ctts table entries.
+ * Composition time offset = composition time - decoding time
*/
int64_t decodingTimeUs;
CHECK(meta_data->findInt64(kKeyDecodingTime, &decodingTimeUs));
decodingTimeUs -= previousPausedDurationUs;
- int64_t timeUs = decodingTimeUs;
- cttsDeltaTimeUs = timestampUs - decodingTimeUs;
+ cttsOffsetTimeUs =
+ timestampUs + kMaxCttsOffsetTimeUs - decodingTimeUs;
+ CHECK_GE(cttsOffsetTimeUs, 0ll);
timestampUs = decodingTimeUs;
- ALOGV("decoding time: %lld and ctts delta time: %lld",
- timestampUs, cttsDeltaTimeUs);
+ ALOGV("decoding time: %lld and ctts offset time: %lld",
+ timestampUs, cttsOffsetTimeUs);
+
+ // Update ctts box table if necessary
+ currCttsOffsetTimeTicks =
+ (cttsOffsetTimeUs * mTimeScale + 500000LL) / 1000000LL;
+ CHECK_LE(currCttsOffsetTimeTicks, 0x0FFFFFFFFLL);
+ if (mNumSamples == 0) {
+ // Force the first ctts table entry to have one single entry
+ // so that we can do adjustment for the initial track start
+ // time offset easily in writeCttsBox().
+ lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
+ addOneCttsTableEntry(1, currCttsOffsetTimeTicks);
+ cttsSampleCount = 0; // No sample in ctts box is pending
+ } else {
+ if (currCttsOffsetTimeTicks != lastCttsOffsetTimeTicks) {
+ addOneCttsTableEntry(cttsSampleCount, lastCttsOffsetTimeTicks);
+ lastCttsOffsetTimeTicks = currCttsOffsetTimeTicks;
+ cttsSampleCount = 1; // One sample in ctts box is pending
+ } else {
+ ++cttsSampleCount;
+ }
+ }
+
+ // Update ctts time offset range
+ if (mNumSamples == 0) {
+ mMinCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ mMaxCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ } else {
+ if (currCttsOffsetTimeTicks > mMaxCttsOffsetTimeUs) {
+ mMaxCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ } else if (currCttsOffsetTimeTicks < mMinCttsOffsetTimeUs) {
+ mMinCttsOffsetTimeUs = currCttsOffsetTimeTicks;
+ }
+ }
+
}
if (mIsRealTimeRecording) {
@@ -1997,7 +2021,7 @@
}
}
- CHECK(timestampUs >= 0);
+ CHECK_GE(timestampUs, 0ll);
ALOGV("%s media time stamp: %lld and previous paused duration %lld",
mIsAudio? "Audio": "Video", timestampUs, previousPausedDurationUs);
if (timestampUs > mTrackDurationUs) {
@@ -2012,6 +2036,7 @@
currDurationTicks =
((timestampUs * mTimeScale + 500000LL) / 1000000LL -
(lastTimestampUs * mTimeScale + 500000LL) / 1000000LL);
+ CHECK_GE(currDurationTicks, 0ll);
mSampleSizes.push_back(sampleSize);
++mNumSamples;
@@ -2020,25 +2045,12 @@
// Force the first sample to have its own stts entry so that
// we can adjust its value later to maintain the A/V sync.
if (mNumSamples == 3 || currDurationTicks != lastDurationTicks) {
- ALOGV("%s lastDurationUs: %lld us, currDurationTicks: %lld us",
- mIsAudio? "Audio": "Video", lastDurationUs, currDurationTicks);
addOneSttsTableEntry(sampleCount, lastDurationTicks);
sampleCount = 1;
} else {
++sampleCount;
}
- if (!mIsAudio) {
- currCttsDurTicks =
- ((cttsDeltaTimeUs * mTimeScale + 500000LL) / 1000000LL -
- (lastCttsTimeUs * mTimeScale + 500000LL) / 1000000LL);
- if (currCttsDurTicks != lastCttsDurTicks) {
- addOneCttsTableEntry(cttsSampleCount, lastCttsDurTicks);
- cttsSampleCount = 1;
- } else {
- ++cttsSampleCount;
- }
- }
}
if (mSamplesHaveSameSize) {
if (mNumSamples >= 2 && previousSampleSize != sampleSize) {
@@ -2052,11 +2064,6 @@
lastDurationTicks = currDurationTicks;
lastTimestampUs = timestampUs;
- if (!mIsAudio) {
- lastCttsDurTicks = currCttsDurTicks;
- lastCttsTimeUs = cttsDeltaTimeUs;
- }
-
if (isSync != 0) {
addOneStssTableEntry(mNumSamples);
}
@@ -2125,10 +2132,8 @@
if (mNumSamples == 1) {
lastDurationUs = 0; // A single sample's duration
lastDurationTicks = 0;
- lastCttsDurTicks = 0;
} else {
++sampleCount; // Count for the last sample
- ++cttsSampleCount;
}
if (mNumSamples <= 2) {
@@ -2140,7 +2145,14 @@
addOneSttsTableEntry(sampleCount, lastDurationTicks);
}
- addOneCttsTableEntry(cttsSampleCount, lastCttsDurTicks);
+ // The last ctts box may not have been written yet, and this
+ // is to make sure that we write out the last ctts box.
+ if (currCttsOffsetTimeTicks == lastCttsOffsetTimeTicks) {
+ if (cttsSampleCount > 0) {
+ addOneCttsTableEntry(cttsSampleCount, lastCttsOffsetTimeTicks);
+ }
+ }
+
mTrackDurationUs += lastDurationUs;
mReachedEOS = true;
@@ -2406,7 +2418,7 @@
mOwner->writeInt16(0x18); // depth
mOwner->writeInt16(-1); // predefined
- CHECK(23 + mCodecSpecificDataSize < 128);
+ CHECK_LT(23 + mCodecSpecificDataSize, 128);
if (!strcasecmp(MEDIA_MIMETYPE_VIDEO_MPEG4, mime)) {
writeMp4vEsdsBox();
@@ -2465,10 +2477,10 @@
void MPEG4Writer::Track::writeMp4aEsdsBox() {
mOwner->beginBox("esds");
CHECK(mCodecSpecificData);
- CHECK(mCodecSpecificDataSize > 0);
+ CHECK_GT(mCodecSpecificDataSize, 0);
// Make sure all sizes encode to a single byte.
- CHECK(mCodecSpecificDataSize + 23 < 128);
+ CHECK_LT(mCodecSpecificDataSize + 23, 128);
mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt8(0x03); // ES_DescrTag
@@ -2502,7 +2514,7 @@
void MPEG4Writer::Track::writeMp4vEsdsBox() {
CHECK(mCodecSpecificData);
- CHECK(mCodecSpecificDataSize > 0);
+ CHECK_GT(mCodecSpecificDataSize, 0);
mOwner->beginBox("esds");
mOwner->writeInt32(0); // version=0, flags=0
@@ -2662,7 +2674,7 @@
void MPEG4Writer::Track::writeAvccBox() {
CHECK(mCodecSpecificData);
- CHECK(mCodecSpecificDataSize >= 5);
+ CHECK_GE(mCodecSpecificDataSize, 5);
// Patch avcc's lengthSize field to match the number
// of bytes we use to indicate the size of a nal unit.
@@ -2690,23 +2702,26 @@
mOwner->endBox(); // pasp
}
+int32_t MPEG4Writer::Track::getStartTimeOffsetScaledTime() const {
+ int64_t trackStartTimeOffsetUs = 0;
+ int64_t moovStartTimeUs = mOwner->getStartTimestampUs();
+ if (mStartTimestampUs != moovStartTimeUs) {
+ CHECK_GT(mStartTimestampUs, moovStartTimeUs);
+ trackStartTimeOffsetUs = mStartTimestampUs - moovStartTimeUs;
+ }
+ return (trackStartTimeOffsetUs * mTimeScale + 500000LL) / 1000000LL;
+}
+
void MPEG4Writer::Track::writeSttsBox() {
mOwner->beginBox("stts");
mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt32(mNumSttsTableEntries);
// Compensate for small start time difference from different media tracks
- int64_t trackStartTimeOffsetUs = 0;
- int64_t moovStartTimeUs = mOwner->getStartTimestampUs();
- if (mStartTimestampUs != moovStartTimeUs) {
- CHECK(mStartTimestampUs > moovStartTimeUs);
- trackStartTimeOffsetUs = mStartTimestampUs - moovStartTimeUs;
- }
List<SttsTableEntry>::iterator it = mSttsTableEntries.begin();
CHECK(it != mSttsTableEntries.end() && it->sampleCount == 1);
mOwner->writeInt32(it->sampleCount);
- int32_t dur = (trackStartTimeOffsetUs * mTimeScale + 500000LL) / 1000000LL;
- mOwner->writeInt32(dur + it->sampleDuration);
+ mOwner->writeInt32(getStartTimeOffsetScaledTime() + it->sampleDuration);
int64_t totalCount = 1;
while (++it != mSttsTableEntries.end()) {
@@ -2714,7 +2729,7 @@
mOwner->writeInt32(it->sampleDuration);
totalCount += it->sampleCount;
}
- CHECK(totalCount == mNumSamples);
+ CHECK_EQ(totalCount, mNumSamples);
mOwner->endBox(); // stts
}
@@ -2723,6 +2738,11 @@
return;
}
+ // There is no B frame at all
+ if (mMinCttsOffsetTimeUs == mMaxCttsOffsetTimeUs) {
+ return;
+ }
+
// Do not write ctts box when there is no need to have it.
if ((mNumCttsTableEntries == 1 &&
mCttsTableEntries.begin()->sampleDuration == 0) ||
@@ -2730,24 +2750,29 @@
return;
}
- ALOGV("ctts box has %d entries", mNumCttsTableEntries);
+ ALOGD("ctts box has %d entries with range [%lld, %lld]",
+ mNumCttsTableEntries, mMinCttsOffsetTimeUs, mMaxCttsOffsetTimeUs);
mOwner->beginBox("ctts");
- if (mHasNegativeCttsDeltaDuration) {
- mOwner->writeInt32(0x00010000); // version=1, flags=0
- } else {
- mOwner->writeInt32(0); // version=0, flags=0
- }
+ // Version 1 allows to use negative offset time value, but
+ // we are sticking to version 0 for now.
+ mOwner->writeInt32(0); // version=0, flags=0
mOwner->writeInt32(mNumCttsTableEntries);
- int64_t totalCount = 0;
- for (List<CttsTableEntry>::iterator it = mCttsTableEntries.begin();
- it != mCttsTableEntries.end(); ++it) {
+ // Compensate for small start time difference from different media tracks
+ List<CttsTableEntry>::iterator it = mCttsTableEntries.begin();
+ CHECK(it != mCttsTableEntries.end() && it->sampleCount == 1);
+ mOwner->writeInt32(it->sampleCount);
+ mOwner->writeInt32(getStartTimeOffsetScaledTime() +
+ it->sampleDuration - mMinCttsOffsetTimeUs);
+
+ int64_t totalCount = 1;
+ while (++it != mCttsTableEntries.end()) {
mOwner->writeInt32(it->sampleCount);
- mOwner->writeInt32(it->sampleDuration);
+ mOwner->writeInt32(it->sampleDuration - mMinCttsOffsetTimeUs);
totalCount += it->sampleCount;
}
- CHECK(totalCount == mNumSamples);
+ CHECK_EQ(totalCount, mNumSamples);
mOwner->endBox(); // ctts
}
diff --git a/media/libstagefright/MediaBuffer.cpp b/media/libstagefright/MediaBuffer.cpp
index 96271e4..11b80bf 100644
--- a/media/libstagefright/MediaBuffer.cpp
+++ b/media/libstagefright/MediaBuffer.cpp
@@ -22,8 +22,8 @@
#include <stdlib.h>
#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MetaData.h>
#include <ui/GraphicBuffer.h>
@@ -157,7 +157,7 @@
}
MediaBuffer::~MediaBuffer() {
- CHECK_EQ(mObserver, NULL);
+ CHECK(mObserver == NULL);
if (mOwnsData && mData != NULL) {
free(mData);
@@ -188,7 +188,7 @@
}
MediaBuffer *MediaBuffer::clone() {
- CHECK_EQ(mGraphicBuffer, NULL);
+ CHECK(mGraphicBuffer == NULL);
MediaBuffer *buffer = new MediaBuffer(mData, mSize);
buffer->set_range(mRangeOffset, mRangeLength);
diff --git a/media/libstagefright/MediaBufferGroup.cpp b/media/libstagefright/MediaBufferGroup.cpp
index c8d05f4..80aae51 100644
--- a/media/libstagefright/MediaBufferGroup.cpp
+++ b/media/libstagefright/MediaBufferGroup.cpp
@@ -17,9 +17,9 @@
#define LOG_TAG "MediaBufferGroup"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
namespace android {
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
new file mode 100644
index 0000000..e14b1c4
--- /dev/null
+++ b/media/libstagefright/MediaCodec.cpp
@@ -0,0 +1,1179 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "MediaCodec"
+#include <utils/Log.h>
+
+#include <media/stagefright/MediaCodec.h>
+
+#include "include/SoftwareRenderer.h"
+
+#include <gui/SurfaceTextureClient.h>
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/ACodec.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/NativeWindowWrapper.h>
+
+namespace android {
+
+// static
+sp<MediaCodec> MediaCodec::CreateByType(
+ const sp<ALooper> &looper, const char *mime, bool encoder) {
+ sp<MediaCodec> codec = new MediaCodec(looper);
+ if (codec->init(mime, true /* nameIsType */, encoder) != OK) {
+ return NULL;
+ }
+
+ return codec;
+}
+
+// static
+sp<MediaCodec> MediaCodec::CreateByComponentName(
+ const sp<ALooper> &looper, const char *name) {
+ sp<MediaCodec> codec = new MediaCodec(looper);
+ if (codec->init(name, false /* nameIsType */, false /* encoder */) != OK) {
+ return NULL;
+ }
+
+ return codec;
+}
+
+MediaCodec::MediaCodec(const sp<ALooper> &looper)
+ : mState(UNINITIALIZED),
+ mLooper(looper),
+ mCodec(new ACodec),
+ mFlags(0),
+ mSoftRenderer(NULL),
+ mDequeueInputTimeoutGeneration(0),
+ mDequeueInputReplyID(0),
+ mDequeueOutputTimeoutGeneration(0),
+ mDequeueOutputReplyID(0) {
+}
+
+MediaCodec::~MediaCodec() {
+ CHECK_EQ(mState, UNINITIALIZED);
+}
+
+// static
+status_t MediaCodec::PostAndAwaitResponse(
+ const sp<AMessage> &msg, sp<AMessage> *response) {
+ status_t err = msg->postAndAwaitResponse(response);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (!(*response)->findInt32("err", &err)) {
+ err = OK;
+ }
+
+ return err;
+}
+
+status_t MediaCodec::init(const char *name, bool nameIsType, bool encoder) {
+ // Current video decoders do not return from OMX_FillThisBuffer
+ // quickly, violating the OpenMAX specs, until that is remedied
+ // we need to invest in an extra looper to free the main event
+ // queue.
+ bool needDedicatedLooper = false;
+ if (nameIsType && !strncasecmp(name, "video/", 6)) {
+ needDedicatedLooper = true;
+ } else if (!nameIsType && !strncmp(name, "OMX.TI.DUCATI1.VIDEO.", 21)) {
+ needDedicatedLooper = true;
+ }
+
+ if (needDedicatedLooper) {
+ if (mCodecLooper == NULL) {
+ mCodecLooper = new ALooper;
+ mCodecLooper->setName("CodecLooper");
+ mCodecLooper->start(false, false, ANDROID_PRIORITY_AUDIO);
+ }
+
+ mCodecLooper->registerHandler(mCodec);
+ } else {
+ mLooper->registerHandler(mCodec);
+ }
+
+ mLooper->registerHandler(this);
+
+ mCodec->setNotificationMessage(new AMessage(kWhatCodecNotify, id()));
+
+ sp<AMessage> msg = new AMessage(kWhatInit, id());
+ msg->setString("name", name);
+ msg->setInt32("nameIsType", nameIsType);
+
+ if (nameIsType) {
+ msg->setInt32("encoder", encoder);
+ }
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::configure(
+ const sp<AMessage> &format,
+ const sp<SurfaceTextureClient> &nativeWindow,
+ uint32_t flags) {
+ sp<AMessage> msg = new AMessage(kWhatConfigure, id());
+
+ msg->setMessage("format", format);
+ msg->setInt32("flags", flags);
+
+ if (nativeWindow != NULL) {
+ if (!(mFlags & kFlagIsSoftwareCodec)) {
+ msg->setObject(
+ "native-window",
+ new NativeWindowWrapper(nativeWindow));
+ } else {
+ mNativeWindow = nativeWindow;
+ }
+ }
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::start() {
+ sp<AMessage> msg = new AMessage(kWhatStart, id());
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::stop() {
+ sp<AMessage> msg = new AMessage(kWhatStop, id());
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::queueInputBuffer(
+ size_t index,
+ size_t offset,
+ size_t size,
+ int64_t presentationTimeUs,
+ uint32_t flags) {
+ sp<AMessage> msg = new AMessage(kWhatQueueInputBuffer, id());
+ msg->setSize("index", index);
+ msg->setSize("offset", offset);
+ msg->setSize("size", size);
+ msg->setInt64("timeUs", presentationTimeUs);
+ msg->setInt32("flags", flags);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::dequeueInputBuffer(size_t *index, int64_t timeoutUs) {
+ sp<AMessage> msg = new AMessage(kWhatDequeueInputBuffer, id());
+ msg->setInt64("timeoutUs", timeoutUs);
+
+ sp<AMessage> response;
+ status_t err;
+ if ((err = PostAndAwaitResponse(msg, &response)) != OK) {
+ return err;
+ }
+
+ CHECK(response->findSize("index", index));
+
+ return OK;
+}
+
+status_t MediaCodec::dequeueOutputBuffer(
+ size_t *index,
+ size_t *offset,
+ size_t *size,
+ int64_t *presentationTimeUs,
+ uint32_t *flags,
+ int64_t timeoutUs) {
+ sp<AMessage> msg = new AMessage(kWhatDequeueOutputBuffer, id());
+ msg->setInt64("timeoutUs", timeoutUs);
+
+ sp<AMessage> response;
+ status_t err;
+ if ((err = PostAndAwaitResponse(msg, &response)) != OK) {
+ return err;
+ }
+
+ CHECK(response->findSize("index", index));
+ CHECK(response->findSize("offset", offset));
+ CHECK(response->findSize("size", size));
+ CHECK(response->findInt64("timeUs", presentationTimeUs));
+ CHECK(response->findInt32("flags", (int32_t *)flags));
+
+ return OK;
+}
+
+status_t MediaCodec::renderOutputBufferAndRelease(size_t index) {
+ sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, id());
+ msg->setSize("index", index);
+ msg->setInt32("render", true);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::releaseOutputBuffer(size_t index) {
+ sp<AMessage> msg = new AMessage(kWhatReleaseOutputBuffer, id());
+ msg->setSize("index", index);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::getOutputFormat(sp<AMessage> *format) const {
+ sp<AMessage> msg = new AMessage(kWhatGetOutputFormat, id());
+
+ sp<AMessage> response;
+ status_t err;
+ if ((err = PostAndAwaitResponse(msg, &response)) != OK) {
+ return err;
+ }
+
+ CHECK(response->findMessage("format", format));
+
+ return OK;
+}
+
+status_t MediaCodec::getInputBuffers(Vector<sp<ABuffer> > *buffers) const {
+ sp<AMessage> msg = new AMessage(kWhatGetBuffers, id());
+ msg->setInt32("portIndex", kPortIndexInput);
+ msg->setPointer("buffers", buffers);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::getOutputBuffers(Vector<sp<ABuffer> > *buffers) const {
+ sp<AMessage> msg = new AMessage(kWhatGetBuffers, id());
+ msg->setInt32("portIndex", kPortIndexOutput);
+ msg->setPointer("buffers", buffers);
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+status_t MediaCodec::flush() {
+ sp<AMessage> msg = new AMessage(kWhatFlush, id());
+
+ sp<AMessage> response;
+ return PostAndAwaitResponse(msg, &response);
+}
+
+////////////////////////////////////////////////////////////////////////////////
+
+void MediaCodec::cancelPendingDequeueOperations() {
+ if (mFlags & kFlagDequeueInputPending) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+ response->postReply(mDequeueInputReplyID);
+
+ ++mDequeueInputTimeoutGeneration;
+ mDequeueInputReplyID = 0;
+ mFlags &= ~kFlagDequeueInputPending;
+ }
+
+ if (mFlags & kFlagDequeueOutputPending) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+ response->postReply(mDequeueOutputReplyID);
+
+ ++mDequeueOutputTimeoutGeneration;
+ mDequeueOutputReplyID = 0;
+ mFlags &= ~kFlagDequeueOutputPending;
+ }
+}
+
+bool MediaCodec::handleDequeueInputBuffer(uint32_t replyID, bool newRequest) {
+ if (mState != STARTED
+ || (mFlags & kFlagStickyError)
+ || (newRequest && (mFlags & kFlagDequeueInputPending))) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+
+ return true;
+ }
+
+ ssize_t index = dequeuePortBuffer(kPortIndexInput);
+
+ if (index < 0) {
+ CHECK_EQ(index, -EAGAIN);
+ return false;
+ }
+
+ sp<AMessage> response = new AMessage;
+ response->setSize("index", index);
+ response->postReply(replyID);
+
+ return true;
+}
+
+bool MediaCodec::handleDequeueOutputBuffer(uint32_t replyID, bool newRequest) {
+ sp<AMessage> response = new AMessage;
+
+ if (mState != STARTED
+ || (mFlags & kFlagStickyError)
+ || (newRequest && (mFlags & kFlagDequeueOutputPending))) {
+ response->setInt32("err", INVALID_OPERATION);
+ } else if (mFlags & kFlagOutputBuffersChanged) {
+ response->setInt32("err", INFO_OUTPUT_BUFFERS_CHANGED);
+ mFlags &= ~kFlagOutputBuffersChanged;
+ } else if (mFlags & kFlagOutputFormatChanged) {
+ response->setInt32("err", INFO_FORMAT_CHANGED);
+ mFlags &= ~kFlagOutputFormatChanged;
+ } else {
+ ssize_t index = dequeuePortBuffer(kPortIndexOutput);
+
+ if (index < 0) {
+ CHECK_EQ(index, -EAGAIN);
+ return false;
+ }
+
+ const sp<ABuffer> &buffer =
+ mPortBuffers[kPortIndexOutput].itemAt(index).mData;
+
+ response->setSize("index", index);
+ response->setSize("offset", buffer->offset());
+ response->setSize("size", buffer->size());
+
+ int64_t timeUs;
+ CHECK(buffer->meta()->findInt64("timeUs", &timeUs));
+
+ response->setInt64("timeUs", timeUs);
+
+ int32_t omxFlags;
+ CHECK(buffer->meta()->findInt32("omxFlags", &omxFlags));
+
+ uint32_t flags = 0;
+ if (omxFlags & OMX_BUFFERFLAG_SYNCFRAME) {
+ flags |= BUFFER_FLAG_SYNCFRAME;
+ }
+ if (omxFlags & OMX_BUFFERFLAG_CODECCONFIG) {
+ flags |= BUFFER_FLAG_CODECCONFIG;
+ }
+ if (omxFlags & OMX_BUFFERFLAG_EOS) {
+ flags |= BUFFER_FLAG_EOS;
+ }
+
+ response->setInt32("flags", flags);
+ }
+
+ response->postReply(replyID);
+
+ return true;
+}
+
+void MediaCodec::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatCodecNotify:
+ {
+ int32_t what;
+ CHECK(msg->findInt32("what", &what));
+
+ switch (what) {
+ case ACodec::kWhatError:
+ {
+ int32_t omxError, internalError;
+ CHECK(msg->findInt32("omx-error", &omxError));
+ CHECK(msg->findInt32("err", &internalError));
+
+ ALOGE("Codec reported an error. "
+ "(omx error 0x%08x, internalError %d)",
+ omxError, internalError);
+
+ bool sendErrorReponse = true;
+
+ switch (mState) {
+ case INITIALIZING:
+ {
+ setState(UNINITIALIZED);
+ break;
+ }
+
+ case CONFIGURING:
+ {
+ setState(INITIALIZED);
+ break;
+ }
+
+ case STARTING:
+ {
+ setState(CONFIGURED);
+ break;
+ }
+
+ case STOPPING:
+ {
+ // Ignore the error, assuming we'll still get
+ // the shutdown complete notification.
+
+ sendErrorReponse = false;
+ break;
+ }
+
+ case FLUSHING:
+ {
+ setState(STARTED);
+ break;
+ }
+
+ case STARTED:
+ {
+ sendErrorReponse = false;
+
+ mFlags |= kFlagStickyError;
+
+ cancelPendingDequeueOperations();
+ break;
+ }
+
+ default:
+ {
+ sendErrorReponse = false;
+
+ mFlags |= kFlagStickyError;
+ break;
+ }
+ }
+
+ if (sendErrorReponse) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", UNKNOWN_ERROR);
+
+ response->postReply(mReplyID);
+ }
+ break;
+ }
+
+ case ACodec::kWhatComponentAllocated:
+ {
+ CHECK_EQ(mState, INITIALIZING);
+ setState(INITIALIZED);
+
+ AString componentName;
+ CHECK(msg->findString("componentName", &componentName));
+
+ if (componentName.startsWith("OMX.google.")) {
+ mFlags |= kFlagIsSoftwareCodec;
+ } else {
+ mFlags &= ~kFlagIsSoftwareCodec;
+ }
+
+ (new AMessage)->postReply(mReplyID);
+ break;
+ }
+
+ case ACodec::kWhatComponentConfigured:
+ {
+ CHECK_EQ(mState, CONFIGURING);
+ setState(CONFIGURED);
+
+ (new AMessage)->postReply(mReplyID);
+ break;
+ }
+
+ case ACodec::kWhatBuffersAllocated:
+ {
+ int32_t portIndex;
+ CHECK(msg->findInt32("portIndex", &portIndex));
+
+ ALOGV("%s buffers allocated",
+ portIndex == kPortIndexInput ? "input" : "output");
+
+ CHECK(portIndex == kPortIndexInput
+ || portIndex == kPortIndexOutput);
+
+ mPortBuffers[portIndex].clear();
+
+ Vector<BufferInfo> *buffers = &mPortBuffers[portIndex];
+ for (size_t i = 0;; ++i) {
+ AString name = StringPrintf("buffer-id_%d", i);
+
+ void *bufferID;
+ if (!msg->findPointer(name.c_str(), &bufferID)) {
+ break;
+ }
+
+ name = StringPrintf("data_%d", i);
+
+ BufferInfo info;
+ info.mBufferID = bufferID;
+ info.mOwnedByClient = false;
+ CHECK(msg->findBuffer(name.c_str(), &info.mData));
+
+ buffers->push_back(info);
+ }
+
+ if (portIndex == kPortIndexOutput) {
+ if (mState == STARTING) {
+ // We're always allocating output buffers after
+ // allocating input buffers, so this is a good
+ // indication that now all buffers are allocated.
+ setState(STARTED);
+ (new AMessage)->postReply(mReplyID);
+ } else {
+ mFlags |= kFlagOutputBuffersChanged;
+ }
+ }
+ break;
+ }
+
+ case ACodec::kWhatOutputFormatChanged:
+ {
+ ALOGV("codec output format changed");
+
+ if ((mFlags & kFlagIsSoftwareCodec)
+ && mNativeWindow != NULL) {
+ AString mime;
+ CHECK(msg->findString("mime", &mime));
+
+ if (!strncasecmp("video/", mime.c_str(), 6)) {
+ delete mSoftRenderer;
+ mSoftRenderer = NULL;
+
+ int32_t width, height;
+ CHECK(msg->findInt32("width", &width));
+ CHECK(msg->findInt32("height", &height));
+
+ int32_t colorFormat;
+ CHECK(msg->findInt32(
+ "color-format", &colorFormat));
+
+ sp<MetaData> meta = new MetaData;
+ meta->setInt32(kKeyWidth, width);
+ meta->setInt32(kKeyHeight, height);
+ meta->setInt32(kKeyColorFormat, colorFormat);
+
+ mSoftRenderer =
+ new SoftwareRenderer(mNativeWindow, meta);
+ }
+ }
+
+ mOutputFormat = msg;
+ mFlags |= kFlagOutputFormatChanged;
+ break;
+ }
+
+ case ACodec::kWhatFillThisBuffer:
+ {
+ /* size_t index = */updateBuffers(kPortIndexInput, msg);
+
+ if (mState == FLUSHING || mState == STOPPING) {
+ returnBuffersToCodecOnPort(kPortIndexInput);
+ break;
+ }
+
+ if (mFlags & kFlagDequeueInputPending) {
+ CHECK(handleDequeueInputBuffer(mDequeueInputReplyID));
+
+ ++mDequeueInputTimeoutGeneration;
+ mFlags &= ~kFlagDequeueInputPending;
+ mDequeueInputReplyID = 0;
+ }
+ break;
+ }
+
+ case ACodec::kWhatDrainThisBuffer:
+ {
+ /* size_t index = */updateBuffers(kPortIndexOutput, msg);
+
+ if (mState == FLUSHING || mState == STOPPING) {
+ returnBuffersToCodecOnPort(kPortIndexOutput);
+ break;
+ }
+
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
+
+ int32_t omxFlags;
+ CHECK(msg->findInt32("flags", &omxFlags));
+
+ buffer->meta()->setInt32("omxFlags", omxFlags);
+
+ if (mFlags & kFlagDequeueOutputPending) {
+ CHECK(handleDequeueOutputBuffer(mDequeueOutputReplyID));
+
+ ++mDequeueOutputTimeoutGeneration;
+ mFlags &= ~kFlagDequeueOutputPending;
+ mDequeueOutputReplyID = 0;
+ }
+ break;
+ }
+
+ case ACodec::kWhatEOS:
+ {
+ // We already notify the client of this by using the
+ // corresponding flag in "onOutputBufferReady".
+ break;
+ }
+
+ case ACodec::kWhatShutdownCompleted:
+ {
+ CHECK_EQ(mState, STOPPING);
+ setState(UNINITIALIZED);
+
+ (new AMessage)->postReply(mReplyID);
+ break;
+ }
+
+ case ACodec::kWhatFlushCompleted:
+ {
+ CHECK_EQ(mState, FLUSHING);
+ setState(STARTED);
+
+ mCodec->signalResume();
+
+ (new AMessage)->postReply(mReplyID);
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+ break;
+ }
+
+ case kWhatInit:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != UNINITIALIZED) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(INITIALIZING);
+
+ AString name;
+ CHECK(msg->findString("name", &name));
+
+ int32_t nameIsType;
+ int32_t encoder = false;
+ CHECK(msg->findInt32("nameIsType", &nameIsType));
+ if (nameIsType) {
+ CHECK(msg->findInt32("encoder", &encoder));
+ }
+
+ sp<AMessage> format = new AMessage;
+
+ if (nameIsType) {
+ format->setString("mime", name.c_str());
+ format->setInt32("encoder", encoder);
+ } else {
+ format->setString("componentName", name.c_str());
+ }
+
+ mCodec->initiateAllocateComponent(format);
+ break;
+ }
+
+ case kWhatConfigure:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != INITIALIZED) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(CONFIGURING);
+
+ sp<RefBase> obj;
+ if (!msg->findObject("native-window", &obj)) {
+ obj.clear();
+ }
+
+ sp<AMessage> format;
+ CHECK(msg->findMessage("format", &format));
+
+ if (obj != NULL) {
+ format->setObject("native-window", obj);
+ }
+
+ uint32_t flags;
+ CHECK(msg->findInt32("flags", (int32_t *)&flags));
+
+ if (flags & CONFIGURE_FLAG_ENCODE) {
+ format->setInt32("encoder", true);
+ }
+
+ mCodec->initiateConfigureComponent(format);
+ break;
+ }
+
+ case kWhatStart:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != CONFIGURED) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(STARTING);
+
+ mCodec->initiateStart();
+ break;
+ }
+
+ case kWhatStop:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != INITIALIZED
+ && mState != CONFIGURED && mState != STARTED) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(STOPPING);
+
+ mCodec->initiateShutdown();
+ returnBuffersToCodec();
+ break;
+ }
+
+ case kWhatDequeueInputBuffer:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (handleDequeueInputBuffer(replyID, true /* new request */)) {
+ break;
+ }
+
+ int64_t timeoutUs;
+ CHECK(msg->findInt64("timeoutUs", &timeoutUs));
+
+ if (timeoutUs == 0ll) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", -EAGAIN);
+ response->postReply(replyID);
+ break;
+ }
+
+ mFlags |= kFlagDequeueInputPending;
+ mDequeueInputReplyID = replyID;
+
+ if (timeoutUs > 0ll) {
+ sp<AMessage> timeoutMsg =
+ new AMessage(kWhatDequeueInputTimedOut, id());
+ timeoutMsg->setInt32(
+ "generation", ++mDequeueInputTimeoutGeneration);
+ timeoutMsg->post(timeoutUs);
+ }
+ break;
+ }
+
+ case kWhatDequeueInputTimedOut:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mDequeueInputTimeoutGeneration) {
+ // Obsolete
+ break;
+ }
+
+ CHECK(mFlags & kFlagDequeueInputPending);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", -EAGAIN);
+ response->postReply(mDequeueInputReplyID);
+
+ mFlags &= ~kFlagDequeueInputPending;
+ mDequeueInputReplyID = 0;
+ break;
+ }
+
+ case kWhatQueueInputBuffer:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != STARTED || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ status_t err = onQueueInputBuffer(msg);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatDequeueOutputBuffer:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (handleDequeueOutputBuffer(replyID, true /* new request */)) {
+ break;
+ }
+
+ int64_t timeoutUs;
+ CHECK(msg->findInt64("timeoutUs", &timeoutUs));
+
+ if (timeoutUs == 0ll) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", -EAGAIN);
+ response->postReply(replyID);
+ break;
+ }
+
+ mFlags |= kFlagDequeueOutputPending;
+ mDequeueOutputReplyID = replyID;
+
+ if (timeoutUs > 0ll) {
+ sp<AMessage> timeoutMsg =
+ new AMessage(kWhatDequeueOutputTimedOut, id());
+ timeoutMsg->setInt32(
+ "generation", ++mDequeueOutputTimeoutGeneration);
+ timeoutMsg->post(timeoutUs);
+ }
+ break;
+ }
+
+ case kWhatDequeueOutputTimedOut:
+ {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+
+ if (generation != mDequeueOutputTimeoutGeneration) {
+ // Obsolete
+ break;
+ }
+
+ CHECK(mFlags & kFlagDequeueOutputPending);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", -EAGAIN);
+ response->postReply(mDequeueOutputReplyID);
+
+ mFlags &= ~kFlagDequeueOutputPending;
+ mDequeueOutputReplyID = 0;
+ break;
+ }
+
+ case kWhatReleaseOutputBuffer:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != STARTED || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ status_t err = onReleaseOutputBuffer(msg);
+
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", err);
+ response->postReply(replyID);
+ break;
+ }
+
+ case kWhatGetBuffers:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != STARTED || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ int32_t portIndex;
+ CHECK(msg->findInt32("portIndex", &portIndex));
+
+ Vector<sp<ABuffer> > *dstBuffers;
+ CHECK(msg->findPointer("buffers", (void **)&dstBuffers));
+
+ dstBuffers->clear();
+ const Vector<BufferInfo> &srcBuffers = mPortBuffers[portIndex];
+
+ for (size_t i = 0; i < srcBuffers.size(); ++i) {
+ const BufferInfo &info = srcBuffers.itemAt(i);
+
+ dstBuffers->push_back(info.mData);
+ }
+
+ (new AMessage)->postReply(replyID);
+ break;
+ }
+
+ case kWhatFlush:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if (mState != STARTED || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ mReplyID = replyID;
+ setState(FLUSHING);
+
+ mCodec->signalFlush();
+ returnBuffersToCodec();
+ break;
+ }
+
+ case kWhatGetOutputFormat:
+ {
+ uint32_t replyID;
+ CHECK(msg->senderAwaitsResponse(&replyID));
+
+ if ((mState != STARTED && mState != FLUSHING)
+ || (mFlags & kFlagStickyError)) {
+ sp<AMessage> response = new AMessage;
+ response->setInt32("err", INVALID_OPERATION);
+
+ response->postReply(replyID);
+ break;
+ }
+
+ sp<AMessage> response = new AMessage;
+ response->setMessage("format", mOutputFormat);
+ response->postReply(replyID);
+ break;
+ }
+
+ default:
+ TRESPASS();
+ }
+}
+
+void MediaCodec::setState(State newState) {
+ if (newState == UNINITIALIZED) {
+ delete mSoftRenderer;
+ mSoftRenderer = NULL;
+
+ mNativeWindow.clear();
+
+ mOutputFormat.clear();
+ mFlags &= ~kFlagOutputFormatChanged;
+ mFlags &= ~kFlagOutputBuffersChanged;
+ mFlags &= ~kFlagStickyError;
+ }
+
+ mState = newState;
+
+ cancelPendingDequeueOperations();
+}
+
+void MediaCodec::returnBuffersToCodec() {
+ returnBuffersToCodecOnPort(kPortIndexInput);
+ returnBuffersToCodecOnPort(kPortIndexOutput);
+}
+
+void MediaCodec::returnBuffersToCodecOnPort(int32_t portIndex) {
+ CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+
+ Vector<BufferInfo> *buffers = &mPortBuffers[portIndex];
+
+ for (size_t i = 0; i < buffers->size(); ++i) {
+ BufferInfo *info = &buffers->editItemAt(i);
+
+ if (info->mNotify != NULL) {
+ sp<AMessage> msg = info->mNotify;
+ info->mNotify = NULL;
+ info->mOwnedByClient = false;
+
+ if (portIndex == kPortIndexInput) {
+ msg->setInt32("err", ERROR_END_OF_STREAM);
+ }
+ msg->post();
+ }
+ }
+
+ mAvailPortBuffers[portIndex].clear();
+}
+
+size_t MediaCodec::updateBuffers(
+ int32_t portIndex, const sp<AMessage> &msg) {
+ CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+
+ void *bufferID;
+ CHECK(msg->findPointer("buffer-id", &bufferID));
+
+ Vector<BufferInfo> *buffers = &mPortBuffers[portIndex];
+
+ for (size_t i = 0; i < buffers->size(); ++i) {
+ BufferInfo *info = &buffers->editItemAt(i);
+
+ if (info->mBufferID == bufferID) {
+ CHECK(info->mNotify == NULL);
+ CHECK(msg->findMessage("reply", &info->mNotify));
+
+ mAvailPortBuffers[portIndex].push_back(i);
+
+ return i;
+ }
+ }
+
+ TRESPASS();
+
+ return 0;
+}
+
+status_t MediaCodec::onQueueInputBuffer(const sp<AMessage> &msg) {
+ size_t index;
+ size_t offset;
+ size_t size;
+ int64_t timeUs;
+ uint32_t flags;
+ CHECK(msg->findSize("index", &index));
+ CHECK(msg->findSize("offset", &offset));
+ CHECK(msg->findSize("size", &size));
+ CHECK(msg->findInt64("timeUs", &timeUs));
+ CHECK(msg->findInt32("flags", (int32_t *)&flags));
+
+ if (index >= mPortBuffers[kPortIndexInput].size()) {
+ return -ERANGE;
+ }
+
+ BufferInfo *info = &mPortBuffers[kPortIndexInput].editItemAt(index);
+
+ if (info->mNotify == NULL || !info->mOwnedByClient) {
+ return -EACCES;
+ }
+
+ if (offset + size > info->mData->capacity()) {
+ return -EINVAL;
+ }
+
+ sp<AMessage> reply = info->mNotify;
+ info->mNotify = NULL;
+ info->mOwnedByClient = false;
+
+ info->mData->setRange(offset, size);
+ info->mData->meta()->setInt64("timeUs", timeUs);
+
+ if (flags & BUFFER_FLAG_EOS) {
+ info->mData->meta()->setInt32("eos", true);
+ }
+
+ if (flags & BUFFER_FLAG_CODECCONFIG) {
+ info->mData->meta()->setInt32("csd", true);
+ }
+
+ reply->setBuffer("buffer", info->mData);
+ reply->post();
+
+ return OK;
+}
+
+status_t MediaCodec::onReleaseOutputBuffer(const sp<AMessage> &msg) {
+ size_t index;
+ CHECK(msg->findSize("index", &index));
+
+ int32_t render;
+ if (!msg->findInt32("render", &render)) {
+ render = 0;
+ }
+
+ if (mState != STARTED) {
+ return -EINVAL;
+ }
+
+ if (index >= mPortBuffers[kPortIndexOutput].size()) {
+ return -ERANGE;
+ }
+
+ BufferInfo *info = &mPortBuffers[kPortIndexOutput].editItemAt(index);
+
+ if (info->mNotify == NULL || !info->mOwnedByClient) {
+ return -EACCES;
+ }
+
+ if (render) {
+ info->mNotify->setInt32("render", true);
+
+ if (mSoftRenderer != NULL) {
+ mSoftRenderer->render(
+ info->mData->data(), info->mData->size(), NULL);
+ }
+ }
+
+ info->mNotify->post();
+ info->mNotify = NULL;
+ info->mOwnedByClient = false;
+
+ return OK;
+}
+
+ssize_t MediaCodec::dequeuePortBuffer(int32_t portIndex) {
+ CHECK(portIndex == kPortIndexInput || portIndex == kPortIndexOutput);
+
+ List<size_t> *availBuffers = &mAvailPortBuffers[portIndex];
+
+ if (availBuffers->empty()) {
+ return -EAGAIN;
+ }
+
+ size_t index = *availBuffers->begin();
+ availBuffers->erase(availBuffers->begin());
+
+ BufferInfo *info = &mPortBuffers[portIndex].editItemAt(index);
+ CHECK(!info->mOwnedByClient);
+ info->mOwnedByClient = true;
+
+ return index;
+}
+
+} // namespace android
diff --git a/media/libstagefright/MediaExtractor.cpp b/media/libstagefright/MediaExtractor.cpp
index 7b17d65..2171492 100644
--- a/media/libstagefright/MediaExtractor.cpp
+++ b/media/libstagefright/MediaExtractor.cpp
@@ -19,6 +19,7 @@
#include <utils/Log.h>
#include "include/AMRExtractor.h"
+#include "include/AVIExtractor.h"
#include "include/MP3Extractor.h"
#include "include/MPEG4Extractor.h"
#include "include/WAVExtractor.h"
@@ -109,10 +110,12 @@
ret = new MatroskaExtractor(source);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2TS)) {
ret = new MPEG2TSExtractor(source);
+ } else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_AVI)) {
+ ret = new AVIExtractor(source);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_WVM)) {
ret = new WVMExtractor(source);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AAC_ADTS)) {
- ret = new AACExtractor(source);
+ ret = new AACExtractor(source, meta);
} else if (!strcasecmp(mime, MEDIA_MIMETYPE_CONTAINER_MPEG2PS)) {
ret = new MPEG2PSExtractor(source);
}
diff --git a/media/libstagefright/MediaSourceSplitter.cpp b/media/libstagefright/MediaSourceSplitter.cpp
index 8af0694..3b64ded 100644
--- a/media/libstagefright/MediaSourceSplitter.cpp
+++ b/media/libstagefright/MediaSourceSplitter.cpp
@@ -18,8 +18,8 @@
#define LOG_TAG "MediaSourceSplitter"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaSourceSplitter.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MetaData.h>
diff --git a/media/libstagefright/MetaData.cpp b/media/libstagefright/MetaData.cpp
index 884f3b4..66dec90 100644
--- a/media/libstagefright/MetaData.cpp
+++ b/media/libstagefright/MetaData.cpp
@@ -17,7 +17,7 @@
#include <stdlib.h>
#include <string.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MetaData.h>
namespace android {
diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libstagefright/NuCachedSource2.cpp
index 249c298..0957426 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libstagefright/NuCachedSource2.cpp
@@ -370,6 +370,7 @@
&& (mSource->flags() & DataSource::kIsHTTPBasedSource)) {
ALOGV("Disconnecting at high watermark");
static_cast<HTTPBase *>(mSource.get())->disconnect();
+ mFinalStatus = -EAGAIN;
}
}
} else {
@@ -549,7 +550,7 @@
size_t delta = offset - mCacheOffset;
- if (mFinalStatus != OK) {
+ if (mFinalStatus != OK && mNumRetriesLeft == 0) {
if (delta >= mCache->totalSize()) {
return mFinalStatus;
}
@@ -591,7 +592,7 @@
size_t totalSize = mCache->totalSize();
CHECK_EQ(mCache->releaseFromStart(totalSize), totalSize);
- mFinalStatus = OK;
+ mNumRetriesLeft = kMaxNumRetries;
mFetching = true;
return OK;
@@ -603,8 +604,8 @@
restartPrefetcherIfNecessary_l(true /* ignore low water threshold */);
}
-sp<DecryptHandle> NuCachedSource2::DrmInitialization() {
- return mSource->DrmInitialization();
+sp<DecryptHandle> NuCachedSource2::DrmInitialization(const char* mime) {
+ return mSource->DrmInitialization(mime);
}
void NuCachedSource2::getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client) {
diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
new file mode 100644
index 0000000..afd4763
--- /dev/null
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -0,0 +1,433 @@
+/*
+ * Copyright 2012, The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "NuMediaExtractor"
+#include <utils/Log.h>
+
+#include <media/stagefright/NuMediaExtractor.h>
+
+#include "include/ESDS.h"
+
+#include <media/stagefright/foundation/ABuffer.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaExtractor.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+#include <media/stagefright/Utils.h>
+
+namespace android {
+
+NuMediaExtractor::NuMediaExtractor() {
+}
+
+NuMediaExtractor::~NuMediaExtractor() {
+ releaseTrackSamples();
+
+ for (size_t i = 0; i < mSelectedTracks.size(); ++i) {
+ TrackInfo *info = &mSelectedTracks.editItemAt(i);
+
+ CHECK_EQ((status_t)OK, info->mSource->stop());
+ }
+
+ mSelectedTracks.clear();
+}
+
+status_t NuMediaExtractor::setDataSource(const char *path) {
+ sp<DataSource> dataSource = DataSource::CreateFromURI(path);
+
+ if (dataSource == NULL) {
+ return -ENOENT;
+ }
+
+ mImpl = MediaExtractor::Create(dataSource);
+
+ if (mImpl == NULL) {
+ return ERROR_UNSUPPORTED;
+ }
+
+ return OK;
+}
+
+size_t NuMediaExtractor::countTracks() const {
+ return mImpl == NULL ? 0 : mImpl->countTracks();
+}
+
+status_t NuMediaExtractor::getTrackFormat(
+ size_t index, sp<AMessage> *format) const {
+ *format = NULL;
+
+ if (mImpl == NULL) {
+ return -EINVAL;
+ }
+
+ if (index >= mImpl->countTracks()) {
+ return -ERANGE;
+ }
+
+ sp<MetaData> meta = mImpl->getTrackMetaData(index);
+
+ const char *mime;
+ CHECK(meta->findCString(kKeyMIMEType, &mime));
+
+ sp<AMessage> msg = new AMessage;
+ msg->setString("mime", mime);
+
+ if (!strncasecmp("video/", mime, 6)) {
+ int32_t width, height;
+ CHECK(meta->findInt32(kKeyWidth, &width));
+ CHECK(meta->findInt32(kKeyHeight, &height));
+
+ msg->setInt32("width", width);
+ msg->setInt32("height", height);
+ } else {
+ CHECK(!strncasecmp("audio/", mime, 6));
+
+ int32_t numChannels, sampleRate;
+ CHECK(meta->findInt32(kKeyChannelCount, &numChannels));
+ CHECK(meta->findInt32(kKeySampleRate, &sampleRate));
+
+ msg->setInt32("channel-count", numChannels);
+ msg->setInt32("sample-rate", sampleRate);
+ }
+
+ int32_t maxInputSize;
+ if (meta->findInt32(kKeyMaxInputSize, &maxInputSize)) {
+ msg->setInt32("max-input-size", maxInputSize);
+ }
+
+ uint32_t type;
+ const void *data;
+ size_t size;
+ if (meta->findData(kKeyAVCC, &type, &data, &size)) {
+ // Parse the AVCDecoderConfigurationRecord
+
+ const uint8_t *ptr = (const uint8_t *)data;
+
+ CHECK(size >= 7);
+ CHECK_EQ((unsigned)ptr[0], 1u); // configurationVersion == 1
+ uint8_t profile = ptr[1];
+ uint8_t level = ptr[3];
+
+ // There is decodable content out there that fails the following
+ // assertion, let's be lenient for now...
+ // CHECK((ptr[4] >> 2) == 0x3f); // reserved
+
+ size_t lengthSize = 1 + (ptr[4] & 3);
+
+ // commented out check below as H264_QVGA_500_NO_AUDIO.3gp
+ // violates it...
+ // CHECK((ptr[5] >> 5) == 7); // reserved
+
+ size_t numSeqParameterSets = ptr[5] & 31;
+
+ ptr += 6;
+ size -= 6;
+
+ sp<ABuffer> buffer = new ABuffer(1024);
+ buffer->setRange(0, 0);
+
+ for (size_t i = 0; i < numSeqParameterSets; ++i) {
+ CHECK(size >= 2);
+ size_t length = U16_AT(ptr);
+
+ ptr += 2;
+ size -= 2;
+
+ CHECK(size >= length);
+
+ memcpy(buffer->data() + buffer->size(), "\x00\x00\x00\x01", 4);
+ memcpy(buffer->data() + buffer->size() + 4, ptr, length);
+ buffer->setRange(0, buffer->size() + 4 + length);
+
+ ptr += length;
+ size -= length;
+ }
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+
+ msg->setBuffer("csd-0", buffer);
+
+ buffer = new ABuffer(1024);
+ buffer->setRange(0, 0);
+
+ CHECK(size >= 1);
+ size_t numPictureParameterSets = *ptr;
+ ++ptr;
+ --size;
+
+ for (size_t i = 0; i < numPictureParameterSets; ++i) {
+ CHECK(size >= 2);
+ size_t length = U16_AT(ptr);
+
+ ptr += 2;
+ size -= 2;
+
+ CHECK(size >= length);
+
+ memcpy(buffer->data() + buffer->size(), "\x00\x00\x00\x01", 4);
+ memcpy(buffer->data() + buffer->size() + 4, ptr, length);
+ buffer->setRange(0, buffer->size() + 4 + length);
+
+ ptr += length;
+ size -= length;
+ }
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-1", buffer);
+ } else if (meta->findData(kKeyESDS, &type, &data, &size)) {
+ ESDS esds((const char *)data, size);
+ CHECK_EQ(esds.InitCheck(), (status_t)OK);
+
+ const void *codec_specific_data;
+ size_t codec_specific_data_size;
+ esds.getCodecSpecificInfo(
+ &codec_specific_data, &codec_specific_data_size);
+
+ sp<ABuffer> buffer = new ABuffer(codec_specific_data_size);
+
+ memcpy(buffer->data(), codec_specific_data,
+ codec_specific_data_size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-0", buffer);
+ } else if (meta->findData(kKeyVorbisInfo, &type, &data, &size)) {
+ sp<ABuffer> buffer = new ABuffer(size);
+ memcpy(buffer->data(), data, size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-0", buffer);
+
+ if (!meta->findData(kKeyVorbisBooks, &type, &data, &size)) {
+ return -EINVAL;
+ }
+
+ buffer = new ABuffer(size);
+ memcpy(buffer->data(), data, size);
+
+ buffer->meta()->setInt32("csd", true);
+ buffer->meta()->setInt64("timeUs", 0);
+ msg->setBuffer("csd-1", buffer);
+ }
+
+ *format = msg;
+
+ return OK;
+}
+
+status_t NuMediaExtractor::selectTrack(size_t index) {
+ if (mImpl == NULL) {
+ return -EINVAL;
+ }
+
+ if (index >= mImpl->countTracks()) {
+ return -ERANGE;
+ }
+
+ for (size_t i = 0; i < mSelectedTracks.size(); ++i) {
+ TrackInfo *info = &mSelectedTracks.editItemAt(i);
+
+ if (info->mTrackIndex == index) {
+ // This track has already been selected.
+ return OK;
+ }
+ }
+
+ sp<MediaSource> source = mImpl->getTrack(index);
+
+ CHECK_EQ((status_t)OK, source->start());
+
+ mSelectedTracks.push();
+ TrackInfo *info = &mSelectedTracks.editItemAt(mSelectedTracks.size() - 1);
+
+ info->mSource = source;
+ info->mTrackIndex = index;
+ info->mFinalResult = OK;
+ info->mSample = NULL;
+ info->mSampleTimeUs = -1ll;
+ info->mFlags = 0;
+
+ const char *mime;
+ CHECK(source->getFormat()->findCString(kKeyMIMEType, &mime));
+
+ if (!strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_VORBIS)) {
+ info->mFlags |= kIsVorbis;
+ }
+
+ return OK;
+}
+
+void NuMediaExtractor::releaseTrackSamples() {
+ for (size_t i = 0; i < mSelectedTracks.size(); ++i) {
+ TrackInfo *info = &mSelectedTracks.editItemAt(i);
+
+ if (info->mSample != NULL) {
+ info->mSample->release();
+ info->mSample = NULL;
+
+ info->mSampleTimeUs = -1ll;
+ }
+ }
+}
+
+ssize_t NuMediaExtractor::fetchTrackSamples(int64_t seekTimeUs) {
+ TrackInfo *minInfo = NULL;
+ ssize_t minIndex = -1;
+
+ for (size_t i = 0; i < mSelectedTracks.size(); ++i) {
+ TrackInfo *info = &mSelectedTracks.editItemAt(i);
+
+ if (seekTimeUs >= 0ll) {
+ info->mFinalResult = OK;
+
+ if (info->mSample != NULL) {
+ info->mSample->release();
+ info->mSample = NULL;
+ info->mSampleTimeUs = -1ll;
+ }
+ } else if (info->mFinalResult != OK) {
+ continue;
+ }
+
+ if (info->mSample == NULL) {
+ MediaSource::ReadOptions options;
+ if (seekTimeUs >= 0ll) {
+ options.setSeekTo(seekTimeUs);
+ }
+ status_t err = info->mSource->read(&info->mSample, &options);
+
+ if (err != OK) {
+ CHECK(info->mSample == NULL);
+
+ info->mFinalResult = err;
+ info->mSampleTimeUs = -1ll;
+ continue;
+ } else {
+ CHECK(info->mSample != NULL);
+ CHECK(info->mSample->meta_data()->findInt64(
+ kKeyTime, &info->mSampleTimeUs));
+ }
+ }
+
+ if (minInfo == NULL || info->mSampleTimeUs < minInfo->mSampleTimeUs) {
+ minInfo = info;
+ minIndex = i;
+ }
+ }
+
+ return minIndex;
+}
+
+status_t NuMediaExtractor::seekTo(int64_t timeUs) {
+ return fetchTrackSamples(timeUs);
+}
+
+status_t NuMediaExtractor::advance() {
+ ssize_t minIndex = fetchTrackSamples();
+
+ if (minIndex < 0) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ TrackInfo *info = &mSelectedTracks.editItemAt(minIndex);
+
+ info->mSample->release();
+ info->mSample = NULL;
+ info->mSampleTimeUs = -1ll;
+
+ return OK;
+}
+
+status_t NuMediaExtractor::readSampleData(const sp<ABuffer> &buffer) {
+ ssize_t minIndex = fetchTrackSamples();
+
+ if (minIndex < 0) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ TrackInfo *info = &mSelectedTracks.editItemAt(minIndex);
+
+ size_t sampleSize = info->mSample->range_length();
+
+ if (info->mFlags & kIsVorbis) {
+ // Each sample's data is suffixed by the number of page samples
+ // or -1 if not available.
+ sampleSize += sizeof(int32_t);
+ }
+
+ if (buffer->capacity() < sampleSize) {
+ return -ENOMEM;
+ }
+
+ const uint8_t *src =
+ (const uint8_t *)info->mSample->data()
+ + info->mSample->range_offset();
+
+ memcpy((uint8_t *)buffer->data(), src, info->mSample->range_length());
+
+ if (info->mFlags & kIsVorbis) {
+ int32_t numPageSamples;
+ if (!info->mSample->meta_data()->findInt32(
+ kKeyValidSamples, &numPageSamples)) {
+ numPageSamples = -1;
+ }
+
+ memcpy((uint8_t *)buffer->data() + info->mSample->range_length(),
+ &numPageSamples,
+ sizeof(numPageSamples));
+ }
+
+ buffer->setRange(0, sampleSize);
+
+ return OK;
+}
+
+status_t NuMediaExtractor::getSampleTrackIndex(size_t *trackIndex) {
+ ssize_t minIndex = fetchTrackSamples();
+
+ if (minIndex < 0) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ TrackInfo *info = &mSelectedTracks.editItemAt(minIndex);
+ *trackIndex = info->mTrackIndex;
+
+ return OK;
+}
+
+status_t NuMediaExtractor::getSampleTime(int64_t *sampleTimeUs) {
+ ssize_t minIndex = fetchTrackSamples();
+
+ if (minIndex < 0) {
+ return ERROR_END_OF_STREAM;
+ }
+
+ TrackInfo *info = &mSelectedTracks.editItemAt(minIndex);
+ *sampleTimeUs = info->mSampleTimeUs;
+
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/OMXClient.cpp b/media/libstagefright/OMXClient.cpp
index 9de873e..7cdb793 100644
--- a/media/libstagefright/OMXClient.cpp
+++ b/media/libstagefright/OMXClient.cpp
@@ -20,11 +20,299 @@
#include <binder/IServiceManager.h>
#include <media/IMediaPlayerService.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/OMXClient.h>
+#include <utils/KeyedVector.h>
+
+#include "include/OMX.h"
namespace android {
+struct MuxOMX : public IOMX {
+ MuxOMX(const sp<IOMX> &remoteOMX);
+ virtual ~MuxOMX();
+
+ virtual IBinder *onAsBinder() { return NULL; }
+
+ virtual bool livesLocally(node_id node, pid_t pid);
+
+ virtual status_t listNodes(List<ComponentInfo> *list);
+
+ virtual status_t allocateNode(
+ const char *name, const sp<IOMXObserver> &observer,
+ node_id *node);
+
+ virtual status_t freeNode(node_id node);
+
+ virtual status_t sendCommand(
+ node_id node, OMX_COMMANDTYPE cmd, OMX_S32 param);
+
+ virtual status_t getParameter(
+ node_id node, OMX_INDEXTYPE index,
+ void *params, size_t size);
+
+ virtual status_t setParameter(
+ node_id node, OMX_INDEXTYPE index,
+ const void *params, size_t size);
+
+ virtual status_t getConfig(
+ node_id node, OMX_INDEXTYPE index,
+ void *params, size_t size);
+
+ virtual status_t setConfig(
+ node_id node, OMX_INDEXTYPE index,
+ const void *params, size_t size);
+
+ virtual status_t getState(
+ node_id node, OMX_STATETYPE* state);
+
+ virtual status_t storeMetaDataInBuffers(
+ node_id node, OMX_U32 port_index, OMX_BOOL enable);
+
+ virtual status_t enableGraphicBuffers(
+ node_id node, OMX_U32 port_index, OMX_BOOL enable);
+
+ virtual status_t getGraphicBufferUsage(
+ node_id node, OMX_U32 port_index, OMX_U32* usage);
+
+ virtual status_t useBuffer(
+ node_id node, OMX_U32 port_index, const sp<IMemory> ¶ms,
+ buffer_id *buffer);
+
+ virtual status_t useGraphicBuffer(
+ node_id node, OMX_U32 port_index,
+ const sp<GraphicBuffer> &graphicBuffer, buffer_id *buffer);
+
+ virtual status_t allocateBuffer(
+ node_id node, OMX_U32 port_index, size_t size,
+ buffer_id *buffer, void **buffer_data);
+
+ virtual status_t allocateBufferWithBackup(
+ node_id node, OMX_U32 port_index, const sp<IMemory> ¶ms,
+ buffer_id *buffer);
+
+ virtual status_t freeBuffer(
+ node_id node, OMX_U32 port_index, buffer_id buffer);
+
+ virtual status_t fillBuffer(node_id node, buffer_id buffer);
+
+ virtual status_t emptyBuffer(
+ node_id node,
+ buffer_id buffer,
+ OMX_U32 range_offset, OMX_U32 range_length,
+ OMX_U32 flags, OMX_TICKS timestamp);
+
+ virtual status_t getExtensionIndex(
+ node_id node,
+ const char *parameter_name,
+ OMX_INDEXTYPE *index);
+
+private:
+ mutable Mutex mLock;
+
+ sp<IOMX> mRemoteOMX;
+ sp<IOMX> mLocalOMX;
+
+ KeyedVector<node_id, bool> mIsLocalNode;
+
+ bool isLocalNode(node_id node) const;
+ bool isLocalNode_l(node_id node) const;
+ const sp<IOMX> &getOMX(node_id node) const;
+ const sp<IOMX> &getOMX_l(node_id node) const;
+
+ static bool IsSoftwareComponent(const char *name);
+
+ DISALLOW_EVIL_CONSTRUCTORS(MuxOMX);
+};
+
+MuxOMX::MuxOMX(const sp<IOMX> &remoteOMX)
+ : mRemoteOMX(remoteOMX) {
+}
+
+MuxOMX::~MuxOMX() {
+}
+
+bool MuxOMX::isLocalNode(node_id node) const {
+ Mutex::Autolock autoLock(mLock);
+
+ return isLocalNode_l(node);
+}
+
+bool MuxOMX::isLocalNode_l(node_id node) const {
+ return mIsLocalNode.indexOfKey(node) >= 0;
+}
+
+// static
+bool MuxOMX::IsSoftwareComponent(const char *name) {
+ return !strncasecmp(name, "OMX.google.", 11);
+}
+
+const sp<IOMX> &MuxOMX::getOMX(node_id node) const {
+ return isLocalNode(node) ? mLocalOMX : mRemoteOMX;
+}
+
+const sp<IOMX> &MuxOMX::getOMX_l(node_id node) const {
+ return isLocalNode_l(node) ? mLocalOMX : mRemoteOMX;
+}
+
+bool MuxOMX::livesLocally(node_id node, pid_t pid) {
+ return getOMX(node)->livesLocally(node, pid);
+}
+
+status_t MuxOMX::listNodes(List<ComponentInfo> *list) {
+ Mutex::Autolock autoLock(mLock);
+
+ if (mLocalOMX == NULL) {
+ mLocalOMX = new OMX;
+ }
+
+ return mLocalOMX->listNodes(list);
+}
+
+status_t MuxOMX::allocateNode(
+ const char *name, const sp<IOMXObserver> &observer,
+ node_id *node) {
+ Mutex::Autolock autoLock(mLock);
+
+ sp<IOMX> omx;
+
+ if (IsSoftwareComponent(name)) {
+ if (mLocalOMX == NULL) {
+ mLocalOMX = new OMX;
+ }
+ omx = mLocalOMX;
+ } else {
+ omx = mRemoteOMX;
+ }
+
+ status_t err = omx->allocateNode(name, observer, node);
+
+ if (err != OK) {
+ return err;
+ }
+
+ if (omx == mLocalOMX) {
+ mIsLocalNode.add(*node, true);
+ }
+
+ return OK;
+}
+
+status_t MuxOMX::freeNode(node_id node) {
+ Mutex::Autolock autoLock(mLock);
+
+ status_t err = getOMX_l(node)->freeNode(node);
+
+ if (err != OK) {
+ return err;
+ }
+
+ mIsLocalNode.removeItem(node);
+
+ return OK;
+}
+
+status_t MuxOMX::sendCommand(
+ node_id node, OMX_COMMANDTYPE cmd, OMX_S32 param) {
+ return getOMX(node)->sendCommand(node, cmd, param);
+}
+
+status_t MuxOMX::getParameter(
+ node_id node, OMX_INDEXTYPE index,
+ void *params, size_t size) {
+ return getOMX(node)->getParameter(node, index, params, size);
+}
+
+status_t MuxOMX::setParameter(
+ node_id node, OMX_INDEXTYPE index,
+ const void *params, size_t size) {
+ return getOMX(node)->setParameter(node, index, params, size);
+}
+
+status_t MuxOMX::getConfig(
+ node_id node, OMX_INDEXTYPE index,
+ void *params, size_t size) {
+ return getOMX(node)->getConfig(node, index, params, size);
+}
+
+status_t MuxOMX::setConfig(
+ node_id node, OMX_INDEXTYPE index,
+ const void *params, size_t size) {
+ return getOMX(node)->setConfig(node, index, params, size);
+}
+
+status_t MuxOMX::getState(
+ node_id node, OMX_STATETYPE* state) {
+ return getOMX(node)->getState(node, state);
+}
+
+status_t MuxOMX::storeMetaDataInBuffers(
+ node_id node, OMX_U32 port_index, OMX_BOOL enable) {
+ return getOMX(node)->storeMetaDataInBuffers(node, port_index, enable);
+}
+
+status_t MuxOMX::enableGraphicBuffers(
+ node_id node, OMX_U32 port_index, OMX_BOOL enable) {
+ return getOMX(node)->enableGraphicBuffers(node, port_index, enable);
+}
+
+status_t MuxOMX::getGraphicBufferUsage(
+ node_id node, OMX_U32 port_index, OMX_U32* usage) {
+ return getOMX(node)->getGraphicBufferUsage(node, port_index, usage);
+}
+
+status_t MuxOMX::useBuffer(
+ node_id node, OMX_U32 port_index, const sp<IMemory> ¶ms,
+ buffer_id *buffer) {
+ return getOMX(node)->useBuffer(node, port_index, params, buffer);
+}
+
+status_t MuxOMX::useGraphicBuffer(
+ node_id node, OMX_U32 port_index,
+ const sp<GraphicBuffer> &graphicBuffer, buffer_id *buffer) {
+ return getOMX(node)->useGraphicBuffer(
+ node, port_index, graphicBuffer, buffer);
+}
+
+status_t MuxOMX::allocateBuffer(
+ node_id node, OMX_U32 port_index, size_t size,
+ buffer_id *buffer, void **buffer_data) {
+ return getOMX(node)->allocateBuffer(
+ node, port_index, size, buffer, buffer_data);
+}
+
+status_t MuxOMX::allocateBufferWithBackup(
+ node_id node, OMX_U32 port_index, const sp<IMemory> ¶ms,
+ buffer_id *buffer) {
+ return getOMX(node)->allocateBufferWithBackup(
+ node, port_index, params, buffer);
+}
+
+status_t MuxOMX::freeBuffer(
+ node_id node, OMX_U32 port_index, buffer_id buffer) {
+ return getOMX(node)->freeBuffer(node, port_index, buffer);
+}
+
+status_t MuxOMX::fillBuffer(node_id node, buffer_id buffer) {
+ return getOMX(node)->fillBuffer(node, buffer);
+}
+
+status_t MuxOMX::emptyBuffer(
+ node_id node,
+ buffer_id buffer,
+ OMX_U32 range_offset, OMX_U32 range_length,
+ OMX_U32 flags, OMX_TICKS timestamp) {
+ return getOMX(node)->emptyBuffer(
+ node, buffer, range_offset, range_length, flags, timestamp);
+}
+
+status_t MuxOMX::getExtensionIndex(
+ node_id node,
+ const char *parameter_name,
+ OMX_INDEXTYPE *index) {
+ return getOMX(node)->getExtensionIndex(node, parameter_name, index);
+}
+
OMXClient::OMXClient() {
}
@@ -38,10 +326,19 @@
mOMX = service->getOMX();
CHECK(mOMX.get() != NULL);
+ if (!mOMX->livesLocally(NULL /* node */, getpid())) {
+ ALOGI("Using client-side OMX mux.");
+ mOMX = new MuxOMX(mOMX);
+ }
+
return OK;
}
void OMXClient::disconnect() {
+ if (mOMX.get() != NULL) {
+ mOMX.clear();
+ mOMX = NULL;
+ }
}
} // namespace android
diff --git a/media/libstagefright/OMXCodec.cpp b/media/libstagefright/OMXCodec.cpp
index 60d9bb7..1325462 100755
--- a/media/libstagefright/OMXCodec.cpp
+++ b/media/libstagefright/OMXCodec.cpp
@@ -19,8 +19,6 @@
#include <utils/Log.h>
#include "include/AACEncoder.h"
-#include "include/AMRNBEncoder.h"
-#include "include/AMRWBEncoder.h"
#include "include/AVCEncoder.h"
#include "include/M4vH263Encoder.h"
@@ -71,8 +69,6 @@
#define FACTORY_REF(name) { #name, Make##name },
-FACTORY_CREATE_ENCODER(AMRNBEncoder)
-FACTORY_CREATE_ENCODER(AMRWBEncoder)
FACTORY_CREATE_ENCODER(AACEncoder)
FACTORY_CREATE_ENCODER(AVCEncoder)
FACTORY_CREATE_ENCODER(M4vH263Encoder)
@@ -86,8 +82,6 @@
};
static const FactoryInfo kFactoryInfo[] = {
- FACTORY_REF(AMRNBEncoder)
- FACTORY_REF(AMRWBEncoder)
FACTORY_REF(AACEncoder)
FACTORY_REF(AVCEncoder)
FACTORY_REF(M4vH263Encoder)
@@ -149,10 +143,11 @@
static const CodecInfo kEncoderInfo[] = {
{ MEDIA_MIMETYPE_AUDIO_AMR_NB, "OMX.TI.AMR.encode" },
- { MEDIA_MIMETYPE_AUDIO_AMR_NB, "AMRNBEncoder" },
+ { MEDIA_MIMETYPE_AUDIO_AMR_NB, "OMX.google.amrnb.encoder" },
{ MEDIA_MIMETYPE_AUDIO_AMR_WB, "OMX.TI.WBAMR.encode" },
- { MEDIA_MIMETYPE_AUDIO_AMR_WB, "AMRWBEncoder" },
+ { MEDIA_MIMETYPE_AUDIO_AMR_WB, "OMX.google.amrwb.encoder" },
{ MEDIA_MIMETYPE_AUDIO_AAC, "OMX.TI.AAC.encode" },
+ { MEDIA_MIMETYPE_AUDIO_AAC, "OMX.google.aac.encoder" },
{ MEDIA_MIMETYPE_AUDIO_AAC, "AACEncoder" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.TI.DUCATI1.VIDEO.MPEG4E" },
{ MEDIA_MIMETYPE_VIDEO_MPEG4, "OMX.qcom.7x30.video.encoder.mpeg4" },
@@ -1482,11 +1477,12 @@
const sp<MediaSource> &source,
const sp<ANativeWindow> &nativeWindow)
: mOMX(omx),
- mOMXLivesLocally(omx->livesLocally(getpid())),
+ mOMXLivesLocally(omx->livesLocally(node, getpid())),
mNode(node),
mQuirks(quirks),
mFlags(flags),
mIsEncoder(isEncoder),
+ mIsVideo(!strncasecmp("video/", mime, 6)),
mMIME(strdup(mime)),
mComponentName(strdup(componentName)),
mSource(source),
@@ -1545,6 +1541,8 @@
"video_decoder.mpeg4", "video_encoder.mpeg4" },
{ MEDIA_MIMETYPE_VIDEO_H263,
"video_decoder.h263", "video_encoder.h263" },
+ { MEDIA_MIMETYPE_VIDEO_VPX,
+ "video_decoder.vpx", "video_encoder.vpx" },
};
static const size_t kNumMimeToRole =
@@ -2191,8 +2189,8 @@
}
}
-int64_t OMXCodec::retrieveDecodingTimeUs(bool isCodecSpecific) {
- CHECK(mIsEncoder);
+int64_t OMXCodec::getDecodingTimeUs() {
+ CHECK(mIsEncoder && mIsVideo);
if (mDecodingTimeList.empty()) {
CHECK(mSignalledEOS || mNoMoreOutputData);
@@ -2203,12 +2201,7 @@
List<int64_t>::iterator it = mDecodingTimeList.begin();
int64_t timeUs = *it;
-
- // If the output buffer is codec specific configuration,
- // do not remove the decoding time from the list.
- if (!isCodecSpecific) {
- mDecodingTimeList.erase(it);
- }
+ mDecodingTimeList.erase(it);
return timeUs;
}
@@ -2387,8 +2380,8 @@
mNoMoreOutputData = true;
}
- if (mIsEncoder) {
- int64_t decodingTimeUs = retrieveDecodingTimeUs(isCodecSpecific);
+ if (mIsEncoder && mIsVideo) {
+ int64_t decodingTimeUs = isCodecSpecific? 0: getDecodingTimeUs();
buffer->meta_data()->setInt64(kKeyDecodingTime, decodingTimeUs);
}
@@ -3249,7 +3242,7 @@
int64_t lastBufferTimeUs;
CHECK(srcBuffer->meta_data()->findInt64(kKeyTime, &lastBufferTimeUs));
CHECK(lastBufferTimeUs >= 0);
- if (mIsEncoder) {
+ if (mIsEncoder && mIsVideo) {
mDecodingTimeList.push_back(lastBufferTimeUs);
}
@@ -3565,6 +3558,7 @@
//////////////// output port ////////////////////
// format
OMX_AUDIO_PARAM_PORTFORMATTYPE format;
+ InitOMXParams(&format);
format.nPortIndex = kPortIndexOutput;
format.nIndex = 0;
status_t err = OMX_ErrorNone;
diff --git a/media/libstagefright/OggExtractor.cpp b/media/libstagefright/OggExtractor.cpp
index 73efc27..5e79e78 100644
--- a/media/libstagefright/OggExtractor.cpp
+++ b/media/libstagefright/OggExtractor.cpp
@@ -21,10 +21,10 @@
#include "include/OggExtractor.h"
#include <cutils/properties.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/SampleIterator.cpp b/media/libstagefright/SampleIterator.cpp
index 81ec5c1..eae721b 100644
--- a/media/libstagefright/SampleIterator.cpp
+++ b/media/libstagefright/SampleIterator.cpp
@@ -22,8 +22,8 @@
#include <arpa/inet.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/Utils.h>
#include "include/SampleTable.h"
diff --git a/media/libstagefright/SampleTable.cpp b/media/libstagefright/SampleTable.cpp
index 8d80d63..d9858d7 100644
--- a/media/libstagefright/SampleTable.cpp
+++ b/media/libstagefright/SampleTable.cpp
@@ -618,26 +618,31 @@
}
uint32_t left = 0;
- while (left < mNumSyncSamples) {
- uint32_t x = mSyncSamples[left];
+ uint32_t right = mNumSyncSamples;
+ while (left < right) {
+ uint32_t center = left + (right - left) / 2;
+ uint32_t x = mSyncSamples[center];
- if (x >= start_sample_index) {
+ if (start_sample_index < x) {
+ right = center;
+ } else if (start_sample_index > x) {
+ left = center + 1;
+ } else {
+ left = center;
break;
}
-
- ++left;
}
-
if (left == mNumSyncSamples) {
if (flags == kFlagAfter) {
ALOGE("tried to find a sync frame after the last one: %d", left);
return ERROR_OUT_OF_RANGE;
}
+ left = left - 1;
}
- if (left > 0) {
- --left;
- }
+ // Now ssi[left] is the sync sample index just before (or at)
+ // start_sample_index.
+ // Also start_sample_index < ssi[left + 1], if left + 1 < mNumSyncSamples.
uint32_t x = mSyncSamples[left];
@@ -682,7 +687,11 @@
x = mSyncSamples[left - 1];
- CHECK(x <= start_sample_index);
+ if (x > start_sample_index) {
+ // The table of sync sample indices was not sorted
+ // properly.
+ return ERROR_MALFORMED;
+ }
}
break;
}
@@ -696,7 +705,11 @@
x = mSyncSamples[left + 1];
- CHECK(x >= start_sample_index);
+ if (x < start_sample_index) {
+ // The table of sync sample indices was not sorted
+ // properly.
+ return ERROR_MALFORMED;
+ }
}
break;
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index 43bfd9e..35f9c1f 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -20,10 +20,10 @@
#include "include/StagefrightMetadataRetriever.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/ColorConverter.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaExtractor.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/OMXCodec.h>
@@ -37,7 +37,7 @@
ALOGV("StagefrightMetadataRetriever()");
DataSource::RegisterDefaultSniffers();
- CHECK_EQ(mClient.connect(), OK);
+ CHECK_EQ(mClient.connect(), (status_t)OK);
}
StagefrightMetadataRetriever::~StagefrightMetadataRetriever() {
@@ -169,7 +169,7 @@
|| (buffer != NULL && buffer->range_length() == 0));
if (err != OK) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
ALOGV("decoding frame failed.");
decoder->stop();
diff --git a/media/libstagefright/SurfaceMediaSource.cpp b/media/libstagefright/SurfaceMediaSource.cpp
index 48df058..ab2cff0 100644
--- a/media/libstagefright/SurfaceMediaSource.cpp
+++ b/media/libstagefright/SurfaceMediaSource.cpp
@@ -16,22 +16,23 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "SurfaceMediaSource"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/SurfaceMediaSource.h>
-#include <ui/GraphicBuffer.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaDefs.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/openmax/OMX_IVCommon.h>
#include <media/stagefright/MetadataBufferType.h>
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
-#include <surfaceflinger/IGraphicBufferAlloc.h>
+#include <ui/GraphicBuffer.h>
+#include <gui/ISurfaceComposer.h>
+#include <gui/IGraphicBufferAlloc.h>
#include <OMX_Component.h>
#include <utils/Log.h>
#include <utils/String8.h>
+#include <private/gui/ComposerService.h>
+
namespace android {
SurfaceMediaSource::SurfaceMediaSource(uint32_t bufW, uint32_t bufH) :
@@ -58,7 +59,7 @@
SurfaceMediaSource::~SurfaceMediaSource() {
ALOGV("SurfaceMediaSource::~SurfaceMediaSource");
if (!mStopped) {
- stop();
+ reset();
}
}
@@ -714,9 +715,9 @@
}
-status_t SurfaceMediaSource::stop()
+status_t SurfaceMediaSource::reset()
{
- ALOGV("Stop");
+ ALOGV("Reset");
Mutex::Autolock lock(mMutex);
// TODO: Add waiting on mFrameCompletedCondition here?
@@ -853,7 +854,7 @@
}
if (!foundBuffer) {
- CHECK_EQ(0, "signalBufferReturned: bogus buffer");
+ CHECK(!"signalBufferReturned: bogus buffer");
}
}
diff --git a/media/libstagefright/ThrottledSource.cpp b/media/libstagefright/ThrottledSource.cpp
index 88e07b0..b1fcafd 100644
--- a/media/libstagefright/ThrottledSource.cpp
+++ b/media/libstagefright/ThrottledSource.cpp
@@ -16,7 +16,7 @@
#include "include/ThrottledSource.h"
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
namespace android {
diff --git a/media/libstagefright/TimedEventQueue.cpp b/media/libstagefright/TimedEventQueue.cpp
index 12c9c36..f4b5d4f 100644
--- a/media/libstagefright/TimedEventQueue.cpp
+++ b/media/libstagefright/TimedEventQueue.cpp
@@ -31,7 +31,7 @@
#include <sys/prctl.h>
#include <sys/time.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#ifdef ANDROID_SIMULATOR
#include <jni.h>
diff --git a/media/libstagefright/VideoSourceDownSampler.cpp b/media/libstagefright/VideoSourceDownSampler.cpp
index 1b66990..90a42c9 100644
--- a/media/libstagefright/VideoSourceDownSampler.cpp
+++ b/media/libstagefright/VideoSourceDownSampler.cpp
@@ -17,9 +17,9 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "VideoSourceDownSampler"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/VideoSourceDownSampler.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/YUVImage.h>
#include <media/stagefright/YUVCanvas.h>
diff --git a/media/libstagefright/WAVExtractor.cpp b/media/libstagefright/WAVExtractor.cpp
index 0bcaf08..501f480 100644
--- a/media/libstagefright/WAVExtractor.cpp
+++ b/media/libstagefright/WAVExtractor.cpp
@@ -20,9 +20,9 @@
#include "include/WAVExtractor.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
@@ -217,7 +217,7 @@
kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_ALAW);
break;
default:
- CHECK_EQ(mWaveFormat, WAVE_FORMAT_MULAW);
+ CHECK_EQ(mWaveFormat, (uint16_t)WAVE_FORMAT_MULAW);
mTrackMeta->setCString(
kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_G711_MLAW);
break;
@@ -362,7 +362,7 @@
// Convert 8-bit unsigned samples to 16-bit signed.
MediaBuffer *tmp;
- CHECK_EQ(mGroup->acquire_buffer(&tmp), OK);
+ CHECK_EQ(mGroup->acquire_buffer(&tmp), (status_t)OK);
// The new buffer holds the sample number of samples, but each
// one is 2 bytes wide.
diff --git a/media/libstagefright/WVMExtractor.cpp b/media/libstagefright/WVMExtractor.cpp
index 2092cb6..c7ad513 100644
--- a/media/libstagefright/WVMExtractor.cpp
+++ b/media/libstagefright/WVMExtractor.cpp
@@ -21,6 +21,7 @@
#include <arpa/inet.h>
#include <utils/String8.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/Utils.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaSource.h>
@@ -28,7 +29,6 @@
#include <media/stagefright/MetaData.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <dlfcn.h>
#include <utils/Errors.h>
@@ -45,17 +45,12 @@
static Mutex gWVMutex;
WVMExtractor::WVMExtractor(const sp<DataSource> &source)
- : mDataSource(source) {
- {
- Mutex::Autolock autoLock(gWVMutex);
- if (gVendorLibHandle == NULL) {
- gVendorLibHandle = dlopen("libwvm.so", RTLD_NOW);
- }
+ : mDataSource(source)
+{
+ Mutex::Autolock autoLock(gWVMutex);
- if (gVendorLibHandle == NULL) {
- ALOGE("Failed to open libwvm.so");
- return;
- }
+ if (!getVendorLibHandle()) {
+ return;
}
typedef WVMLoadableExtractor *(*GetInstanceFunc)(sp<DataSource>);
@@ -64,13 +59,28 @@
"_ZN7android11GetInstanceENS_2spINS_10DataSourceEEE");
if (getInstanceFunc) {
+ CHECK(source->DrmInitialization(MEDIA_MIMETYPE_CONTAINER_WVM) != NULL);
mImpl = (*getInstanceFunc)(source);
CHECK(mImpl != NULL);
+ setDrmFlag(true);
} else {
ALOGE("Failed to locate GetInstance in libwvm.so");
}
}
+bool WVMExtractor::getVendorLibHandle()
+{
+ if (gVendorLibHandle == NULL) {
+ gVendorLibHandle = dlopen("libwvm.so", RTLD_NOW);
+ }
+
+ if (gVendorLibHandle == NULL) {
+ ALOGE("Failed to open libwvm.so");
+ }
+
+ return gVendorLibHandle != NULL;
+}
+
WVMExtractor::~WVMExtractor() {
}
@@ -113,5 +123,33 @@
}
}
+bool SniffWVM(
+ const sp<DataSource> &source, String8 *mimeType, float *confidence,
+ sp<AMessage> *) {
+
+ Mutex::Autolock autoLock(gWVMutex);
+
+ if (!WVMExtractor::getVendorLibHandle()) {
+ return false;
+ }
+
+ typedef WVMLoadableExtractor *(*SnifferFunc)(const sp<DataSource>&);
+ SnifferFunc snifferFunc =
+ (SnifferFunc) dlsym(gVendorLibHandle,
+ "_ZN7android15IsWidevineMediaERKNS_2spINS_10DataSourceEEE");
+
+ if (snifferFunc) {
+ if ((*snifferFunc)(source)) {
+ *mimeType = MEDIA_MIMETYPE_CONTAINER_WVM;
+ *confidence = 10.0f;
+ return true;
+ }
+ } else {
+ ALOGE("IsWidevineMedia not found in libwvm.so");
+ }
+
+ return false;
+}
+
} //namespace android
diff --git a/media/libstagefright/chromium_http/Android.mk b/media/libstagefright/chromium_http/Android.mk
index 6573e3c..63775f1 100644
--- a/media/libstagefright/chromium_http/Android.mk
+++ b/media/libstagefright/chromium_http/Android.mk
@@ -3,8 +3,9 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- ChromiumHTTPDataSource.cpp \
- support.cpp \
+ DataUriSource.cpp \
+ ChromiumHTTPDataSource.cpp \
+ support.cpp
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
diff --git a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp b/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
index 180460b..76f7946 100644
--- a/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
+++ b/media/libstagefright/chromium_http/ChromiumHTTPDataSource.cpp
@@ -259,7 +259,7 @@
mCondition.broadcast();
}
-sp<DecryptHandle> ChromiumHTTPDataSource::DrmInitialization() {
+sp<DecryptHandle> ChromiumHTTPDataSource::DrmInitialization(const char* mime) {
Mutex::Autolock autoLock(mLock);
if (mDrmManagerClient == NULL) {
@@ -275,7 +275,7 @@
* original one
*/
mDecryptHandle = mDrmManagerClient->openDecryptSession(
- String8(mURI.c_str()));
+ String8(mURI.c_str()), mime);
}
if (mDecryptHandle == NULL) {
diff --git a/media/libstagefright/chromium_http/DataUriSource.cpp b/media/libstagefright/chromium_http/DataUriSource.cpp
new file mode 100644
index 0000000..ecf3fa1
--- /dev/null
+++ b/media/libstagefright/chromium_http/DataUriSource.cpp
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <include/DataUriSource.h>
+
+#include <net/base/data_url.h>
+#include <googleurl/src/gurl.h>
+
+
+namespace android {
+
+DataUriSource::DataUriSource(const char *uri) :
+ mDataUri(uri),
+ mInited(NO_INIT) {
+
+ // Copy1: const char *uri -> String8 mDataUri.
+ std::string mimeTypeStr, unusedCharsetStr, dataStr;
+ // Copy2: String8 mDataUri -> std::string
+ const bool ret = net::DataURL::Parse(
+ GURL(std::string(mDataUri.string())),
+ &mimeTypeStr, &unusedCharsetStr, &dataStr);
+ // Copy3: std::string dataStr -> AString mData
+ mData.setTo(dataStr.data(), dataStr.length());
+ mInited = ret ? OK : UNKNOWN_ERROR;
+
+ // The chromium data url implementation defaults to using "text/plain"
+ // if no mime type is specified. We prefer to leave this unspecified
+ // instead, since the mime type is sniffed in most cases.
+ if (mimeTypeStr != "text/plain") {
+ mMimeType = mimeTypeStr.c_str();
+ }
+}
+
+ssize_t DataUriSource::readAt(off64_t offset, void *out, size_t size) {
+ if (mInited != OK) {
+ return mInited;
+ }
+
+ const off64_t length = mData.size();
+ if (offset >= length) {
+ return UNKNOWN_ERROR;
+ }
+
+ const char *dataBuf = mData.c_str();
+ const size_t bytesToCopy =
+ offset + size >= length ? (length - offset) : size;
+
+ if (bytesToCopy > 0) {
+ memcpy(out, dataBuf + offset, bytesToCopy);
+ }
+
+ return bytesToCopy;
+}
+
+} // namespace android
diff --git a/media/libstagefright/codecs/aacdec/SoftAAC.cpp b/media/libstagefright/codecs/aacdec/SoftAAC.cpp
index da9d280..ea6c360 100644
--- a/media/libstagefright/codecs/aacdec/SoftAAC.cpp
+++ b/media/libstagefright/codecs/aacdec/SoftAAC.cpp
@@ -218,6 +218,18 @@
return OMX_ErrorNone;
}
+ case OMX_IndexParamAudioPcm:
+ {
+ const OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
default:
return SimpleSoftOMXComponent::internalSetParameter(index, params);
}
diff --git a/media/libstagefright/codecs/aacenc/AACEncoder.cpp b/media/libstagefright/codecs/aacenc/AACEncoder.cpp
index 2b8633d..8b5007e 100644
--- a/media/libstagefright/codecs/aacenc/AACEncoder.cpp
+++ b/media/libstagefright/codecs/aacenc/AACEncoder.cpp
@@ -22,8 +22,8 @@
#include "voAAC.h"
#include "cmnMemory.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -114,8 +114,8 @@
ALOGV("setAudioSpecificConfigData: %d hz, %d bps, and %d channels",
mSampleRate, mBitRate, mChannels);
- int32_t index;
- CHECK_EQ(OK, getSampleRateTableIndex(mSampleRate, index));
+ int32_t index = 0;
+ CHECK_EQ((status_t)OK, getSampleRateTableIndex(mSampleRate, index));
if (mChannels > 2 || mChannels <= 0) {
ALOGE("Unsupported number of channels(%d)", mChannels);
return UNKNOWN_ERROR;
@@ -142,7 +142,7 @@
mBufferGroup = new MediaBufferGroup;
mBufferGroup->add_buffer(new MediaBuffer(2048));
- CHECK_EQ(OK, initCheck());
+ CHECK_EQ((status_t)OK, initCheck());
mNumInputSamples = 0;
mAnchorTimeUs = 0;
@@ -183,7 +183,7 @@
mSource->stop();
if (mEncoderHandle) {
- CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ CHECK_EQ((VO_U32)VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
mEncoderHandle = NULL;
}
delete mApiHandle;
@@ -223,7 +223,7 @@
CHECK(options == NULL || !options->getSeekTo(&seekTimeUs, &mode));
MediaBuffer *buffer;
- CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), OK);
+ CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), (status_t)OK);
uint8_t *outPtr = (uint8_t *)buffer->data();
bool readFromSource = false;
int64_t wallClockTimeUs = -1;
@@ -255,7 +255,7 @@
}
size_t align = mInputBuffer->range_length() % sizeof(int16_t);
- CHECK_EQ(align, 0);
+ CHECK_EQ(align, (size_t)0);
int64_t timeUs;
if (mInputBuffer->meta_data()->findInt64(kKeyDriftTime, &timeUs)) {
diff --git a/media/libstagefright/codecs/aacenc/Android.mk b/media/libstagefright/codecs/aacenc/Android.mk
index 8318ba4..34a2796 100644
--- a/media/libstagefright/codecs/aacenc/Android.mk
+++ b/media/libstagefright/codecs/aacenc/Android.mk
@@ -85,3 +85,29 @@
endif
include $(BUILD_STATIC_LIBRARY)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ SoftAACEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/media/libstagefright/include \
+ frameworks/base/include/media/stagefright/openmax \
+ frameworks/base/media/libstagefright/codecs/common/include \
+
+LOCAL_CFLAGS := -DOSCL_IMPORT_REF=
+
+LOCAL_STATIC_LIBRARIES := \
+ libstagefright_aacenc
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx libstagefright_foundation libutils \
+ libstagefright_enc_common
+
+LOCAL_MODULE := libstagefright_soft_aacenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp b/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp
new file mode 100644
index 0000000..c6724c2
--- /dev/null
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder.cpp
@@ -0,0 +1,560 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAACEncoder"
+#include <utils/Log.h>
+
+#include "SoftAACEncoder.h"
+
+#include "voAAC.h"
+#include "cmnMemory.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftAACEncoder::SoftAACEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mEncoderHandle(NULL),
+ mApiHandle(NULL),
+ mMemOperator(NULL),
+ mNumChannels(1),
+ mSampleRate(44100),
+ mBitRate(0),
+ mSentCodecSpecificData(false),
+ mInputSize(0),
+ mInputFrame(NULL),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+
+ setAudioParams();
+}
+
+SoftAACEncoder::~SoftAACEncoder() {
+ delete[] mInputFrame;
+ mInputFrame = NULL;
+
+ if (mEncoderHandle) {
+ CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ mEncoderHandle = NULL;
+ }
+
+ delete mApiHandle;
+ mApiHandle = NULL;
+
+ delete mMemOperator;
+ mMemOperator = NULL;
+}
+
+void SoftAACEncoder::initPorts() {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = 0;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t) * 2;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+ addPort(def);
+
+ def.nPortIndex = 1;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = 8192;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/aac");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAAC;
+
+ addPort(def);
+}
+
+status_t SoftAACEncoder::initEncoder() {
+ mApiHandle = new VO_AUDIO_CODECAPI;
+
+ if (VO_ERR_NONE != voGetAACEncAPI(mApiHandle)) {
+ ALOGE("Failed to get api handle");
+ return UNKNOWN_ERROR;
+ }
+
+ mMemOperator = new VO_MEM_OPERATOR;
+ mMemOperator->Alloc = cmnMemAlloc;
+ mMemOperator->Copy = cmnMemCopy;
+ mMemOperator->Free = cmnMemFree;
+ mMemOperator->Set = cmnMemSet;
+ mMemOperator->Check = cmnMemCheck;
+
+ VO_CODEC_INIT_USERDATA userData;
+ memset(&userData, 0, sizeof(userData));
+ userData.memflag = VO_IMF_USERMEMOPERATOR;
+ userData.memData = (VO_PTR) mMemOperator;
+ if (VO_ERR_NONE !=
+ mApiHandle->Init(&mEncoderHandle, VO_AUDIO_CodingAAC, &userData)) {
+ ALOGE("Failed to init AAC encoder");
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAACEncoder::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamAudioPortFormat:
+ {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ formatParams->eEncoding =
+ (formatParams->nPortIndex == 0)
+ ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAAC;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAac:
+ {
+ OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+ (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+ if (aacParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ aacParams->nBitRate = mBitRate;
+ aacParams->nAudioBandWidth = 0;
+ aacParams->nAACtools = 0;
+ aacParams->nAACERtools = 0;
+ aacParams->eAACProfile = OMX_AUDIO_AACObjectMain;
+ aacParams->eAACStreamFormat = OMX_AUDIO_AACStreamFormatMP4FF;
+ aacParams->eChannelMode = OMX_AUDIO_ChannelModeStereo;
+
+ aacParams->nChannels = mNumChannels;
+ aacParams->nSampleRate = mSampleRate;
+ aacParams->nFrameLength = 0;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ pcmParams->eNumData = OMX_NumericalDataSigned;
+ pcmParams->eEndian = OMX_EndianBig;
+ pcmParams->bInterleaved = OMX_TRUE;
+ pcmParams->nBitPerSample = 16;
+ pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelLF;
+ pcmParams->eChannelMapping[1] = OMX_AUDIO_ChannelRF;
+
+ pcmParams->nChannels = mNumChannels;
+ pcmParams->nSamplingRate = mSampleRate;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftAACEncoder::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ "audio_encoder.aac",
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPortFormat:
+ {
+ const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if ((formatParams->nPortIndex == 0
+ && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+ || (formatParams->nPortIndex == 1
+ && formatParams->eEncoding != OMX_AUDIO_CodingAAC)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAac:
+ {
+ OMX_AUDIO_PARAM_AACPROFILETYPE *aacParams =
+ (OMX_AUDIO_PARAM_AACPROFILETYPE *)params;
+
+ if (aacParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ mBitRate = aacParams->nBitRate;
+ mNumChannels = aacParams->nChannels;
+ mSampleRate = aacParams->nSampleRate;
+
+ if (setAudioParams() != OK) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ mNumChannels = pcmParams->nChannels;
+ mSampleRate = pcmParams->nSamplingRate;
+
+ if (setAudioParams() != OK) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+status_t SoftAACEncoder::setAudioParams() {
+ // We call this whenever sample rate, number of channels or bitrate change
+ // in reponse to setParameter calls.
+
+ ALOGV("setAudioParams: %lu Hz, %lu channels, %lu bps",
+ mSampleRate, mNumChannels, mBitRate);
+
+ status_t err = setAudioSpecificConfigData();
+
+ if (err != OK) {
+ return err;
+ }
+
+ AACENC_PARAM params;
+ memset(¶ms, 0, sizeof(params));
+ params.sampleRate = mSampleRate;
+ params.bitRate = mBitRate;
+ params.nChannels = mNumChannels;
+ params.adtsUsed = 0; // We add adts header in the file writer if needed.
+ if (VO_ERR_NONE != mApiHandle->SetParam(
+ mEncoderHandle, VO_PID_AAC_ENCPARAM, ¶ms)) {
+ ALOGE("Failed to set AAC encoder parameters");
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+static status_t getSampleRateTableIndex(int32_t sampleRate, int32_t &index) {
+ static const int32_t kSampleRateTable[] = {
+ 96000, 88200, 64000, 48000, 44100, 32000,
+ 24000, 22050, 16000, 12000, 11025, 8000
+ };
+ const int32_t tableSize =
+ sizeof(kSampleRateTable) / sizeof(kSampleRateTable[0]);
+
+ for (int32_t i = 0; i < tableSize; ++i) {
+ if (sampleRate == kSampleRateTable[i]) {
+ index = i;
+ return OK;
+ }
+ }
+
+ return UNKNOWN_ERROR;
+}
+
+status_t SoftAACEncoder::setAudioSpecificConfigData() {
+ // The AAC encoder's audio specific config really only encodes
+ // number of channels and the sample rate (mapped to an index into
+ // a fixed sample rate table).
+
+ int32_t index;
+ status_t err = getSampleRateTableIndex(mSampleRate, index);
+ if (err != OK) {
+ ALOGE("Unsupported sample rate (%lu Hz)", mSampleRate);
+ return err;
+ }
+
+ if (mNumChannels > 2 || mNumChannels <= 0) {
+ ALOGE("Unsupported number of channels(%lu)", mNumChannels);
+ return UNKNOWN_ERROR;
+ }
+
+ // OMX_AUDIO_AACObjectLC
+ mAudioSpecificConfigData[0] = ((0x02 << 3) | (index >> 1));
+ mAudioSpecificConfigData[1] = ((index & 0x01) << 7) | (mNumChannels << 3);
+
+ return OK;
+}
+
+void SoftAACEncoder::onQueueFilled(OMX_U32 portIndex) {
+ if (mSignalledError) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+
+ if (!mSentCodecSpecificData) {
+ // The very first thing we want to output is the codec specific
+ // data. It does not require any input data but we will need an
+ // output buffer to store it in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+ outHeader->nFilledLen = sizeof(mAudioSpecificConfigData);
+ outHeader->nFlags = OMX_BUFFERFLAG_CODECCONFIG;
+
+ uint8_t *out = outHeader->pBuffer + outHeader->nOffset;
+ memcpy(out, mAudioSpecificConfigData, sizeof(mAudioSpecificConfigData));
+
+#if 0
+ ALOGI("sending codec specific data.");
+ hexdump(out, sizeof(mAudioSpecificConfigData));
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ mSentCodecSpecificData = true;
+ }
+
+ size_t numBytesPerInputFrame =
+ mNumChannels * kNumSamplesPerFrame * sizeof(int16_t);
+
+ for (;;) {
+ // We do the following until we run out of buffers.
+
+ while (mInputSize < numBytesPerInputFrame) {
+ // As long as there's still input data to be read we
+ // will drain "kNumSamplesPerFrame * mNumChannels" samples
+ // into the "mInputFrame" buffer and then encode those
+ // as a unit into an output buffer.
+
+ if (mSawInputEOS || inQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+ const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+ size_t copy = numBytesPerInputFrame - mInputSize;
+ if (copy > inHeader->nFilledLen) {
+ copy = inHeader->nFilledLen;
+ }
+
+ if (mInputFrame == NULL) {
+ mInputFrame = new int16_t[kNumSamplesPerFrame * mNumChannels];
+ }
+
+ if (mInputSize == 0) {
+ mInputTimeUs = inHeader->nTimeStamp;
+ }
+
+ memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+ mInputSize += copy;
+
+ inHeader->nOffset += copy;
+ inHeader->nFilledLen -= copy;
+
+ // "Time" on the input buffer has in effect advanced by the
+ // number of audio frames we just advanced nOffset by.
+ inHeader->nTimeStamp +=
+ (copy * 1000000ll / mSampleRate)
+ / (mNumChannels * sizeof(int16_t));
+
+ if (inHeader->nFilledLen == 0) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("saw input EOS");
+ mSawInputEOS = true;
+
+ // Pad any remaining data with zeroes.
+ memset((uint8_t *)mInputFrame + mInputSize,
+ 0,
+ numBytesPerInputFrame - mInputSize);
+
+ mInputSize = numBytesPerInputFrame;
+ }
+
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+
+ inData = NULL;
+ inHeader = NULL;
+ inInfo = NULL;
+ }
+ }
+
+ // At this point we have all the input data necessary to encode
+ // a single frame, all we need is an output buffer to store the result
+ // in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+ VO_CODECBUFFER inputData;
+ memset(&inputData, 0, sizeof(inputData));
+ inputData.Buffer = (unsigned char *)mInputFrame;
+ inputData.Length = numBytesPerInputFrame;
+ CHECK(VO_ERR_NONE ==
+ mApiHandle->SetInputData(mEncoderHandle, &inputData));
+
+ VO_CODECBUFFER outputData;
+ memset(&outputData, 0, sizeof(outputData));
+ VO_AUDIO_OUTPUTINFO outputInfo;
+ memset(&outputInfo, 0, sizeof(outputInfo));
+
+ uint8_t *outPtr = (uint8_t *)outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ VO_U32 ret = VO_ERR_NONE;
+ size_t nOutputBytes = 0;
+ do {
+ outputData.Buffer = outPtr;
+ outputData.Length = outAvailable - nOutputBytes;
+ ret = mApiHandle->GetOutputData(
+ mEncoderHandle, &outputData, &outputInfo);
+ if (ret == VO_ERR_NONE) {
+ outPtr += outputData.Length;
+ nOutputBytes += outputData.Length;
+ }
+ } while (ret != VO_ERR_INPUT_BUFFER_SMALL);
+
+ outHeader->nFilledLen = nOutputBytes;
+
+ outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+ if (mSawInputEOS) {
+ // We also tag this output buffer with EOS if it corresponds
+ // to the final input buffer.
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ }
+
+ outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+ ALOGI("sending %d bytes of data (time = %lld us, flags = 0x%08lx)",
+ nOutputBytes, mInputTimeUs, outHeader->nFlags);
+
+ hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ outHeader = NULL;
+ outInfo = NULL;
+
+ mInputSize = 0;
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+ const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+ return new android::SoftAACEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/aacenc/SoftAACEncoder.h b/media/libstagefright/codecs/aacenc/SoftAACEncoder.h
new file mode 100644
index 0000000..d148eb7
--- /dev/null
+++ b/media/libstagefright/codecs/aacenc/SoftAACEncoder.h
@@ -0,0 +1,82 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SOFT_AAC_ENCODER_H_
+
+#define SOFT_AAC_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+struct VO_AUDIO_CODECAPI;
+struct VO_MEM_OPERATOR;
+
+namespace android {
+
+struct SoftAACEncoder : public SimpleSoftOMXComponent {
+ SoftAACEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftAACEncoder();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+ enum {
+ kNumBuffers = 4,
+ kNumSamplesPerFrame = 1024,
+ };
+
+ void *mEncoderHandle;
+ VO_AUDIO_CODECAPI *mApiHandle;
+ VO_MEM_OPERATOR *mMemOperator;
+
+ OMX_U32 mNumChannels;
+ OMX_U32 mSampleRate;
+ OMX_U32 mBitRate;
+
+ bool mSentCodecSpecificData;
+ size_t mInputSize;
+ int16_t *mInputFrame;
+ int64_t mInputTimeUs;
+
+ bool mSawInputEOS;
+
+ uint8_t mAudioSpecificConfigData[2];
+
+ bool mSignalledError;
+
+ void initPorts();
+ status_t initEncoder();
+
+ status_t setAudioSpecificConfigData();
+ status_t setAudioParams();
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftAACEncoder);
+};
+
+} // namespace android
+
+#endif // SOFT_AAC_ENCODER_H_
diff --git a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/PrePostMDCT_v7.s b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/PrePostMDCT_v7.s
index b2bc9d9..7f6b881 100644
--- a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/PrePostMDCT_v7.s
+++ b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/PrePostMDCT_v7.s
@@ -23,9 +23,13 @@
.section .text
.global PreMDCT
+ .fnstart
PreMDCT:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
add r9, r0, r1, lsl #2
sub r3, r9, #32
@@ -74,14 +78,20 @@
bne PreMDCT_LOOP
PreMDCT_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
@ENDP @ |PreMDCT|
+ .fnend
.section .text
.global PostMDCT
+ .fnstart
PostMDCT:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
add r9, r0, r1, lsl #2
sub r3, r9, #32
@@ -129,7 +139,8 @@
bne PostMDCT_LOOP
PostMDCT_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
@ENDP @ |PostMDCT|
- .end
+ .fnend
diff --git a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/R4R8First_v7.s b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/R4R8First_v7.s
index 3033156..03fa6a9 100644
--- a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/R4R8First_v7.s
+++ b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/R4R8First_v7.s
@@ -23,9 +23,13 @@
.section .text
.global Radix8First
+ .fnstart
Radix8First:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
ldr r3, SQRT1_2
cmp r1, #0
@@ -103,17 +107,23 @@
bne Radix8First_LOOP
Radix8First_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
SQRT1_2:
.word 0x2d413ccd
@ENDP @ |Radix8First|
+ .fnend
.section .text
.global Radix4First
+ .fnstart
Radix4First:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
cmp r1, #0
beq Radix4First_END
@@ -140,7 +150,8 @@
bne Radix4First_LOOP
Radix4First_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
@ENDP @ |Radix4First|
- .end
+ .fnend
diff --git a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/Radix4FFT_v7.s b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/Radix4FFT_v7.s
index f874825..431bc30 100644
--- a/media/libstagefright/codecs/aacenc/src/asm/ARMV7/Radix4FFT_v7.s
+++ b/media/libstagefright/codecs/aacenc/src/asm/ARMV7/Radix4FFT_v7.s
@@ -23,9 +23,13 @@
.section .text
.global Radix4FFT
+ .fnstart
Radix4FFT:
stmdb sp!, {r4 - r11, lr}
+ .save {r4 - r11, lr}
+ fstmfdd sp!, {d8 - d15}
+ .vsave {d8 - d15}
mov r1, r1, asr #2
cmp r1, #0
@@ -137,7 +141,8 @@
bne Radix4FFT_LOOP1
Radix4FFT_END:
+ fldmfdd sp!, {d8 - d15}
ldmia sp!, {r4 - r11, pc}
@ENDP @ |Radix4FFT|
- .end
+ .fnend
diff --git a/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp b/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp
index 3afbc4f..27d7e4d 100644
--- a/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp
+++ b/media/libstagefright/codecs/amrnb/enc/AMRNBEncoder.cpp
@@ -18,8 +18,8 @@
#include "gsmamr_enc.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -210,7 +210,7 @@
}
MediaBuffer *buffer;
- CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), OK);
+ CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), (status_t)OK);
uint8_t *outPtr = (uint8_t *)buffer->data();
diff --git a/media/libstagefright/codecs/amrnb/enc/Android.mk b/media/libstagefright/codecs/amrnb/enc/Android.mk
index b6aed81..94e8726 100644
--- a/media/libstagefright/codecs/amrnb/enc/Android.mk
+++ b/media/libstagefright/codecs/amrnb/enc/Android.mk
@@ -74,3 +74,30 @@
LOCAL_MODULE := libstagefright_amrnbenc
include $(BUILD_STATIC_LIBRARY)
+
+################################################################################
+
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ SoftAMRNBEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/media/libstagefright/include \
+ frameworks/base/include/media/stagefright/openmax \
+ $(LOCAL_PATH)/src \
+ $(LOCAL_PATH)/include \
+ $(LOCAL_PATH)/../common/include \
+ $(LOCAL_PATH)/../common
+
+LOCAL_STATIC_LIBRARIES := \
+ libstagefright_amrnbenc
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx libstagefright_foundation libutils \
+ libstagefright_amrnb_common
+
+LOCAL_MODULE := libstagefright_soft_amrnbenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
new file mode 100644
index 0000000..07f8b4f
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.cpp
@@ -0,0 +1,404 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAMRNBEncoder"
+#include <utils/Log.h>
+
+#include "SoftAMRNBEncoder.h"
+
+#include "gsmamr_enc.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+static const int32_t kSampleRate = 8000;
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftAMRNBEncoder::SoftAMRNBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mEncState(NULL),
+ mSidState(NULL),
+ mBitRate(0),
+ mMode(MR475),
+ mInputSize(0),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+}
+
+SoftAMRNBEncoder::~SoftAMRNBEncoder() {
+ if (mEncState != NULL) {
+ AMREncodeExit(&mEncState, &mSidState);
+ mEncState = mSidState = NULL;
+ }
+}
+
+void SoftAMRNBEncoder::initPorts() {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = 0;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t);
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+ addPort(def);
+
+ def.nPortIndex = 1;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = 8192;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/3gpp");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAMR;
+
+ addPort(def);
+}
+
+status_t SoftAMRNBEncoder::initEncoder() {
+ if (AMREncodeInit(&mEncState, &mSidState, false /* dtx_enable */) != 0) {
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAMRNBEncoder::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamAudioPortFormat:
+ {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ formatParams->eEncoding =
+ (formatParams->nPortIndex == 0)
+ ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAMR;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ amrParams->nChannels = 1;
+ amrParams->nBitRate = mBitRate;
+ amrParams->eAMRBandMode = (OMX_AUDIO_AMRBANDMODETYPE)(mMode + 1);
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ pcmParams->eNumData = OMX_NumericalDataSigned;
+ pcmParams->eEndian = OMX_EndianBig;
+ pcmParams->bInterleaved = OMX_TRUE;
+ pcmParams->nBitPerSample = 16;
+ pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelCF;
+
+ pcmParams->nChannels = 1;
+ pcmParams->nSamplingRate = kSampleRate;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftAMRNBEncoder::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ "audio_encoder.amrnb",
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPortFormat:
+ {
+ const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if ((formatParams->nPortIndex == 0
+ && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+ || (formatParams->nPortIndex == 1
+ && formatParams->eEncoding != OMX_AUDIO_CodingAMR)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (amrParams->nChannels != 1
+ || amrParams->eAMRDTXMode != OMX_AUDIO_AMRDTXModeOff
+ || amrParams->eAMRFrameFormat
+ != OMX_AUDIO_AMRFrameFormatFSF
+ || amrParams->eAMRBandMode < OMX_AUDIO_AMRBandModeNB0
+ || amrParams->eAMRBandMode > OMX_AUDIO_AMRBandModeNB7) {
+ return OMX_ErrorUndefined;
+ }
+
+ mBitRate = amrParams->nBitRate;
+ mMode = amrParams->eAMRBandMode - 1;
+
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (pcmParams->nChannels != 1
+ || pcmParams->nSamplingRate != kSampleRate) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+void SoftAMRNBEncoder::onQueueFilled(OMX_U32 portIndex) {
+ if (mSignalledError) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+
+ size_t numBytesPerInputFrame = kNumSamplesPerFrame * sizeof(int16_t);
+
+ for (;;) {
+ // We do the following until we run out of buffers.
+
+ while (mInputSize < numBytesPerInputFrame) {
+ // As long as there's still input data to be read we
+ // will drain "kNumSamplesPerFrame" samples
+ // into the "mInputFrame" buffer and then encode those
+ // as a unit into an output buffer.
+
+ if (mSawInputEOS || inQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+ const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+ size_t copy = numBytesPerInputFrame - mInputSize;
+ if (copy > inHeader->nFilledLen) {
+ copy = inHeader->nFilledLen;
+ }
+
+ if (mInputSize == 0) {
+ mInputTimeUs = inHeader->nTimeStamp;
+ }
+
+ memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+ mInputSize += copy;
+
+ inHeader->nOffset += copy;
+ inHeader->nFilledLen -= copy;
+
+ // "Time" on the input buffer has in effect advanced by the
+ // number of audio frames we just advanced nOffset by.
+ inHeader->nTimeStamp +=
+ (copy * 1000000ll / kSampleRate) / sizeof(int16_t);
+
+ if (inHeader->nFilledLen == 0) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("saw input EOS");
+ mSawInputEOS = true;
+
+ // Pad any remaining data with zeroes.
+ memset((uint8_t *)mInputFrame + mInputSize,
+ 0,
+ numBytesPerInputFrame - mInputSize);
+
+ mInputSize = numBytesPerInputFrame;
+ }
+
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+
+ inData = NULL;
+ inHeader = NULL;
+ inInfo = NULL;
+ }
+ }
+
+ // At this point we have all the input data necessary to encode
+ // a single frame, all we need is an output buffer to store the result
+ // in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+ uint8_t *outPtr = outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ Frame_Type_3GPP frameType;
+ int res = AMREncode(
+ mEncState, mSidState, (Mode)mMode,
+ mInputFrame, outPtr, &frameType, AMR_TX_WMF);
+
+ CHECK_GE(res, 0);
+ CHECK_LE((size_t)res, outAvailable);
+
+ // Convert header byte from WMF to IETF format.
+ outPtr[0] = ((outPtr[0] << 3) | 4) & 0x7c;
+
+ outHeader->nFilledLen = res;
+ outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+ if (mSawInputEOS) {
+ // We also tag this output buffer with EOS if it corresponds
+ // to the final input buffer.
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ }
+
+ outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+ ALOGI("sending %d bytes of data (time = %lld us, flags = 0x%08lx)",
+ nOutputBytes, mInputTimeUs, outHeader->nFlags);
+
+ hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ outHeader = NULL;
+ outInfo = NULL;
+
+ mInputSize = 0;
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+ const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+ return new android::SoftAMRNBEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.h b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.h
new file mode 100644
index 0000000..50178c4
--- /dev/null
+++ b/media/libstagefright/codecs/amrnb/enc/SoftAMRNBEncoder.h
@@ -0,0 +1,72 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SOFT_AMRNB_ENCODER_H_
+
+#define SOFT_AMRNB_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+namespace android {
+
+struct SoftAMRNBEncoder : public SimpleSoftOMXComponent {
+ SoftAMRNBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftAMRNBEncoder();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+ enum {
+ kNumBuffers = 4,
+ kNumSamplesPerFrame = 160,
+ };
+
+ void *mEncState;
+ void *mSidState;
+
+ OMX_U32 mBitRate;
+ int mMode;
+
+ size_t mInputSize;
+ int16_t mInputFrame[kNumSamplesPerFrame];
+ int64_t mInputTimeUs;
+
+ bool mSawInputEOS;
+ bool mSignalledError;
+
+ void initPorts();
+ status_t initEncoder();
+
+ status_t setAudioParams();
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftAMRNBEncoder);
+};
+
+} // namespace android
+
+#endif // SOFT_AMRNB_ENCODER_H_
diff --git a/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp b/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp
index 60b1163..7fd3a95 100644
--- a/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp
+++ b/media/libstagefright/codecs/amrwbenc/AMRWBEncoder.cpp
@@ -22,8 +22,8 @@
#include "voAMRWB.h"
#include "cmnMemory.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -134,7 +134,7 @@
// The largest buffer size is header + 477 bits
mBufferGroup->add_buffer(new MediaBuffer(1024));
- CHECK_EQ(OK, initCheck());
+ CHECK_EQ((status_t)OK, initCheck());
mNumFramesOutput = 0;
@@ -163,7 +163,7 @@
mBufferGroup = NULL;
- CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ CHECK_EQ((VO_U32)VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
mEncoderHandle = NULL;
delete mApiHandle;
@@ -222,7 +222,7 @@
}
size_t align = mInputBuffer->range_length() % sizeof(int16_t);
- CHECK_EQ(align, 0);
+ CHECK_EQ(align, (size_t)0);
int64_t timeUs;
if (mInputBuffer->meta_data()->findInt64(kKeyDriftTime, &timeUs)) {
@@ -271,7 +271,7 @@
CHECK(VO_ERR_NONE == mApiHandle->SetInputData(mEncoderHandle,&inputData));
MediaBuffer *buffer;
- CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), OK);
+ CHECK_EQ(mBufferGroup->acquire_buffer(&buffer), (status_t)OK);
uint8_t *outPtr = (uint8_t *)buffer->data();
VO_CODECBUFFER outputData;
diff --git a/media/libstagefright/codecs/amrwbenc/Android.mk b/media/libstagefright/codecs/amrwbenc/Android.mk
index ae43870..6ce6171 100644
--- a/media/libstagefright/codecs/amrwbenc/Android.mk
+++ b/media/libstagefright/codecs/amrwbenc/Android.mk
@@ -117,4 +117,26 @@
include $(BUILD_STATIC_LIBRARY)
+################################################################################
+include $(CLEAR_VARS)
+
+LOCAL_SRC_FILES := \
+ SoftAMRWBEncoder.cpp
+
+LOCAL_C_INCLUDES := \
+ frameworks/base/media/libstagefright/include \
+ frameworks/base/include/media/stagefright/openmax \
+ frameworks/base/media/libstagefright/codecs/common/include \
+
+LOCAL_STATIC_LIBRARIES := \
+ libstagefright_amrwbenc
+
+LOCAL_SHARED_LIBRARIES := \
+ libstagefright_omx libstagefright_foundation libutils \
+ libstagefright_enc_common
+
+LOCAL_MODULE := libstagefright_soft_amrwbenc
+LOCAL_MODULE_TAGS := optional
+
+include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
new file mode 100644
index 0000000..9ccb49c
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.cpp
@@ -0,0 +1,459 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "SoftAMRWBEncoder"
+#include <utils/Log.h>
+
+#include "SoftAMRWBEncoder.h"
+
+#include "cmnMemory.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/hexdump.h>
+
+namespace android {
+
+static const int32_t kSampleRate = 16000;
+
+template<class T>
+static void InitOMXParams(T *params) {
+ params->nSize = sizeof(T);
+ params->nVersion.s.nVersionMajor = 1;
+ params->nVersion.s.nVersionMinor = 0;
+ params->nVersion.s.nRevision = 0;
+ params->nVersion.s.nStep = 0;
+}
+
+SoftAMRWBEncoder::SoftAMRWBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component)
+ : SimpleSoftOMXComponent(name, callbacks, appData, component),
+ mEncoderHandle(NULL),
+ mApiHandle(NULL),
+ mMemOperator(NULL),
+ mBitRate(0),
+ mMode(VOAMRWB_MD66),
+ mInputSize(0),
+ mInputTimeUs(-1ll),
+ mSawInputEOS(false),
+ mSignalledError(false) {
+ initPorts();
+ CHECK_EQ(initEncoder(), (status_t)OK);
+}
+
+SoftAMRWBEncoder::~SoftAMRWBEncoder() {
+ if (mEncoderHandle != NULL) {
+ CHECK_EQ(VO_ERR_NONE, mApiHandle->Uninit(mEncoderHandle));
+ mEncoderHandle = NULL;
+ }
+
+ delete mApiHandle;
+ mApiHandle = NULL;
+
+ delete mMemOperator;
+ mMemOperator = NULL;
+}
+
+void SoftAMRWBEncoder::initPorts() {
+ OMX_PARAM_PORTDEFINITIONTYPE def;
+ InitOMXParams(&def);
+
+ def.nPortIndex = 0;
+ def.eDir = OMX_DirInput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = kNumSamplesPerFrame * sizeof(int16_t);
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 1;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/raw");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingPCM;
+
+ addPort(def);
+
+ def.nPortIndex = 1;
+ def.eDir = OMX_DirOutput;
+ def.nBufferCountMin = kNumBuffers;
+ def.nBufferCountActual = def.nBufferCountMin;
+ def.nBufferSize = 8192;
+ def.bEnabled = OMX_TRUE;
+ def.bPopulated = OMX_FALSE;
+ def.eDomain = OMX_PortDomainAudio;
+ def.bBuffersContiguous = OMX_FALSE;
+ def.nBufferAlignment = 2;
+
+ def.format.audio.cMIMEType = const_cast<char *>("audio/amr-wb");
+ def.format.audio.pNativeRender = NULL;
+ def.format.audio.bFlagErrorConcealment = OMX_FALSE;
+ def.format.audio.eEncoding = OMX_AUDIO_CodingAMR;
+
+ addPort(def);
+}
+
+status_t SoftAMRWBEncoder::initEncoder() {
+ mApiHandle = new VO_AUDIO_CODECAPI;
+
+ if (VO_ERR_NONE != voGetAMRWBEncAPI(mApiHandle)) {
+ ALOGE("Failed to get api handle");
+ return UNKNOWN_ERROR;
+ }
+
+ mMemOperator = new VO_MEM_OPERATOR;
+ mMemOperator->Alloc = cmnMemAlloc;
+ mMemOperator->Copy = cmnMemCopy;
+ mMemOperator->Free = cmnMemFree;
+ mMemOperator->Set = cmnMemSet;
+ mMemOperator->Check = cmnMemCheck;
+
+ VO_CODEC_INIT_USERDATA userData;
+ memset(&userData, 0, sizeof(userData));
+ userData.memflag = VO_IMF_USERMEMOPERATOR;
+ userData.memData = (VO_PTR) mMemOperator;
+
+ if (VO_ERR_NONE != mApiHandle->Init(
+ &mEncoderHandle, VO_AUDIO_CodingAMRWB, &userData)) {
+ ALOGE("Failed to init AMRWB encoder");
+ return UNKNOWN_ERROR;
+ }
+
+ VOAMRWBFRAMETYPE type = VOAMRWB_RFC3267;
+ if (VO_ERR_NONE != mApiHandle->SetParam(
+ mEncoderHandle, VO_PID_AMRWB_FRAMETYPE, &type)) {
+ ALOGE("Failed to set AMRWB encoder frame type to %d", type);
+ return UNKNOWN_ERROR;
+ }
+
+ return OK;
+}
+
+OMX_ERRORTYPE SoftAMRWBEncoder::internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamAudioPortFormat:
+ {
+ OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ formatParams->eEncoding =
+ (formatParams->nPortIndex == 0)
+ ? OMX_AUDIO_CodingPCM : OMX_AUDIO_CodingAMR;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ amrParams->nChannels = 1;
+ amrParams->nBitRate = mBitRate;
+
+ amrParams->eAMRBandMode =
+ (OMX_AUDIO_AMRBANDMODETYPE)(mMode + OMX_AUDIO_AMRBandModeWB0);
+
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ pcmParams->eNumData = OMX_NumericalDataSigned;
+ pcmParams->eEndian = OMX_EndianBig;
+ pcmParams->bInterleaved = OMX_TRUE;
+ pcmParams->nBitPerSample = 16;
+ pcmParams->ePCMMode = OMX_AUDIO_PCMModeLinear;
+ pcmParams->eChannelMapping[0] = OMX_AUDIO_ChannelCF;
+
+ pcmParams->nChannels = 1;
+ pcmParams->nSamplingRate = kSampleRate;
+
+ return OMX_ErrorNone;
+ }
+
+ default:
+ return SimpleSoftOMXComponent::internalGetParameter(index, params);
+ }
+}
+
+OMX_ERRORTYPE SoftAMRWBEncoder::internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params) {
+ switch (index) {
+ case OMX_IndexParamStandardComponentRole:
+ {
+ const OMX_PARAM_COMPONENTROLETYPE *roleParams =
+ (const OMX_PARAM_COMPONENTROLETYPE *)params;
+
+ if (strncmp((const char *)roleParams->cRole,
+ "audio_encoder.amrwb",
+ OMX_MAX_STRINGNAME_SIZE - 1)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPortFormat:
+ {
+ const OMX_AUDIO_PARAM_PORTFORMATTYPE *formatParams =
+ (const OMX_AUDIO_PARAM_PORTFORMATTYPE *)params;
+
+ if (formatParams->nPortIndex > 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (formatParams->nIndex > 0) {
+ return OMX_ErrorNoMore;
+ }
+
+ if ((formatParams->nPortIndex == 0
+ && formatParams->eEncoding != OMX_AUDIO_CodingPCM)
+ || (formatParams->nPortIndex == 1
+ && formatParams->eEncoding != OMX_AUDIO_CodingAMR)) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioAmr:
+ {
+ OMX_AUDIO_PARAM_AMRTYPE *amrParams =
+ (OMX_AUDIO_PARAM_AMRTYPE *)params;
+
+ if (amrParams->nPortIndex != 1) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (amrParams->nChannels != 1
+ || amrParams->eAMRDTXMode != OMX_AUDIO_AMRDTXModeOff
+ || amrParams->eAMRFrameFormat
+ != OMX_AUDIO_AMRFrameFormatFSF
+ || amrParams->eAMRBandMode < OMX_AUDIO_AMRBandModeWB0
+ || amrParams->eAMRBandMode > OMX_AUDIO_AMRBandModeWB8) {
+ return OMX_ErrorUndefined;
+ }
+
+ mBitRate = amrParams->nBitRate;
+
+ mMode = (VOAMRWBMODE)(
+ amrParams->eAMRBandMode - OMX_AUDIO_AMRBandModeWB0);
+
+ amrParams->eAMRDTXMode = OMX_AUDIO_AMRDTXModeOff;
+ amrParams->eAMRFrameFormat = OMX_AUDIO_AMRFrameFormatFSF;
+
+ if (VO_ERR_NONE !=
+ mApiHandle->SetParam(
+ mEncoderHandle, VO_PID_AMRWB_MODE, &mMode)) {
+ ALOGE("Failed to set AMRWB encoder mode to %d", mMode);
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+ case OMX_IndexParamAudioPcm:
+ {
+ OMX_AUDIO_PARAM_PCMMODETYPE *pcmParams =
+ (OMX_AUDIO_PARAM_PCMMODETYPE *)params;
+
+ if (pcmParams->nPortIndex != 0) {
+ return OMX_ErrorUndefined;
+ }
+
+ if (pcmParams->nChannels != 1
+ || pcmParams->nSamplingRate != (OMX_U32)kSampleRate) {
+ return OMX_ErrorUndefined;
+ }
+
+ return OMX_ErrorNone;
+ }
+
+
+ default:
+ return SimpleSoftOMXComponent::internalSetParameter(index, params);
+ }
+}
+
+void SoftAMRWBEncoder::onQueueFilled(OMX_U32 portIndex) {
+ if (mSignalledError) {
+ return;
+ }
+
+ List<BufferInfo *> &inQueue = getPortQueue(0);
+ List<BufferInfo *> &outQueue = getPortQueue(1);
+
+ size_t numBytesPerInputFrame = kNumSamplesPerFrame * sizeof(int16_t);
+
+ for (;;) {
+ // We do the following until we run out of buffers.
+
+ while (mInputSize < numBytesPerInputFrame) {
+ // As long as there's still input data to be read we
+ // will drain "kNumSamplesPerFrame" samples
+ // into the "mInputFrame" buffer and then encode those
+ // as a unit into an output buffer.
+
+ if (mSawInputEOS || inQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *inInfo = *inQueue.begin();
+ OMX_BUFFERHEADERTYPE *inHeader = inInfo->mHeader;
+
+ const void *inData = inHeader->pBuffer + inHeader->nOffset;
+
+ size_t copy = numBytesPerInputFrame - mInputSize;
+ if (copy > inHeader->nFilledLen) {
+ copy = inHeader->nFilledLen;
+ }
+
+ if (mInputSize == 0) {
+ mInputTimeUs = inHeader->nTimeStamp;
+ }
+
+ memcpy((uint8_t *)mInputFrame + mInputSize, inData, copy);
+ mInputSize += copy;
+
+ inHeader->nOffset += copy;
+ inHeader->nFilledLen -= copy;
+
+ // "Time" on the input buffer has in effect advanced by the
+ // number of audio frames we just advanced nOffset by.
+ inHeader->nTimeStamp +=
+ (copy * 1000000ll / kSampleRate) / sizeof(int16_t);
+
+ if (inHeader->nFilledLen == 0) {
+ if (inHeader->nFlags & OMX_BUFFERFLAG_EOS) {
+ ALOGV("saw input EOS");
+ mSawInputEOS = true;
+
+ // Pad any remaining data with zeroes.
+ memset((uint8_t *)mInputFrame + mInputSize,
+ 0,
+ numBytesPerInputFrame - mInputSize);
+
+ mInputSize = numBytesPerInputFrame;
+ }
+
+ inQueue.erase(inQueue.begin());
+ inInfo->mOwnedByUs = false;
+ notifyEmptyBufferDone(inHeader);
+
+ inData = NULL;
+ inHeader = NULL;
+ inInfo = NULL;
+ }
+ }
+
+ // At this point we have all the input data necessary to encode
+ // a single frame, all we need is an output buffer to store the result
+ // in.
+
+ if (outQueue.empty()) {
+ return;
+ }
+
+ BufferInfo *outInfo = *outQueue.begin();
+ OMX_BUFFERHEADERTYPE *outHeader = outInfo->mHeader;
+
+ uint8_t *outPtr = outHeader->pBuffer + outHeader->nOffset;
+ size_t outAvailable = outHeader->nAllocLen - outHeader->nOffset;
+
+ VO_CODECBUFFER inputData;
+ memset(&inputData, 0, sizeof(inputData));
+ inputData.Buffer = (unsigned char *) mInputFrame;
+ inputData.Length = mInputSize;
+
+ CHECK_EQ(VO_ERR_NONE,
+ mApiHandle->SetInputData(mEncoderHandle, &inputData));
+
+ VO_CODECBUFFER outputData;
+ memset(&outputData, 0, sizeof(outputData));
+ VO_AUDIO_OUTPUTINFO outputInfo;
+ memset(&outputInfo, 0, sizeof(outputInfo));
+
+ outputData.Buffer = outPtr;
+ outputData.Length = outAvailable;
+ VO_U32 ret = mApiHandle->GetOutputData(
+ mEncoderHandle, &outputData, &outputInfo);
+ CHECK(ret == VO_ERR_NONE || ret == VO_ERR_INPUT_BUFFER_SMALL);
+
+ outHeader->nFilledLen = outputData.Length;
+ outHeader->nFlags = OMX_BUFFERFLAG_ENDOFFRAME;
+
+ if (mSawInputEOS) {
+ // We also tag this output buffer with EOS if it corresponds
+ // to the final input buffer.
+ outHeader->nFlags = OMX_BUFFERFLAG_EOS;
+ }
+
+ outHeader->nTimeStamp = mInputTimeUs;
+
+#if 0
+ ALOGI("sending %ld bytes of data (time = %lld us, flags = 0x%08lx)",
+ outHeader->nFilledLen, mInputTimeUs, outHeader->nFlags);
+
+ hexdump(outHeader->pBuffer + outHeader->nOffset, outHeader->nFilledLen);
+#endif
+
+ outQueue.erase(outQueue.begin());
+ outInfo->mOwnedByUs = false;
+ notifyFillBufferDone(outHeader);
+
+ outHeader = NULL;
+ outInfo = NULL;
+
+ mInputSize = 0;
+ }
+}
+
+} // namespace android
+
+android::SoftOMXComponent *createSoftOMXComponent(
+ const char *name, const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData, OMX_COMPONENTTYPE **component) {
+ return new android::SoftAMRWBEncoder(name, callbacks, appData, component);
+}
diff --git a/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.h b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.h
new file mode 100644
index 0000000..d0c1dab
--- /dev/null
+++ b/media/libstagefright/codecs/amrwbenc/SoftAMRWBEncoder.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef SOFT_AMRWB_ENCODER_H_
+
+#define SOFT_AMRWB_ENCODER_H_
+
+#include "SimpleSoftOMXComponent.h"
+
+#include "voAMRWB.h"
+
+struct VO_AUDIO_CODECAPI;
+struct VO_MEM_OPERATOR;
+
+namespace android {
+
+struct SoftAMRWBEncoder : public SimpleSoftOMXComponent {
+ SoftAMRWBEncoder(
+ const char *name,
+ const OMX_CALLBACKTYPE *callbacks,
+ OMX_PTR appData,
+ OMX_COMPONENTTYPE **component);
+
+protected:
+ virtual ~SoftAMRWBEncoder();
+
+ virtual OMX_ERRORTYPE internalGetParameter(
+ OMX_INDEXTYPE index, OMX_PTR params);
+
+ virtual OMX_ERRORTYPE internalSetParameter(
+ OMX_INDEXTYPE index, const OMX_PTR params);
+
+ virtual void onQueueFilled(OMX_U32 portIndex);
+
+private:
+ enum {
+ kNumBuffers = 4,
+ kNumSamplesPerFrame = 320,
+ };
+
+ void *mEncoderHandle;
+ VO_AUDIO_CODECAPI *mApiHandle;
+ VO_MEM_OPERATOR *mMemOperator;
+
+ OMX_U32 mBitRate;
+ VOAMRWBMODE mMode;
+
+ size_t mInputSize;
+ int16_t mInputFrame[kNumSamplesPerFrame];
+ int64_t mInputTimeUs;
+
+ bool mSawInputEOS;
+ bool mSignalledError;
+
+ void initPorts();
+ status_t initEncoder();
+
+ DISALLOW_EVIL_CONSTRUCTORS(SoftAMRWBEncoder);
+};
+
+} // namespace android
+
+#endif // SOFT_AMRWB_ENCODER_H_
diff --git a/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp b/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp
index e202a2b..7533f07 100644
--- a/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp
+++ b/media/libstagefright/codecs/avc/enc/AVCEncoder.cpp
@@ -24,8 +24,8 @@
#include "avcenc_int.h"
#include "OMX_Video.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -417,7 +417,7 @@
*out = NULL;
MediaBuffer *outputBuffer;
- CHECK_EQ(OK, mGroup->acquire_buffer(&outputBuffer));
+ CHECK_EQ((status_t)OK, mGroup->acquire_buffer(&outputBuffer));
uint8_t *outPtr = (uint8_t *) outputBuffer->data();
uint32_t dataLength = outputBuffer->size();
@@ -557,9 +557,9 @@
encoderStatus = PVAVCEncodeNAL(mHandle, outPtr, &dataLength, &type);
if (encoderStatus == AVCENC_SUCCESS) {
outputBuffer->meta_data()->setInt32(kKeyIsSyncFrame, mIsIDRFrame);
- CHECK_EQ(NULL, PVAVCEncGetOverrunBuffer(mHandle));
+ CHECK(NULL == PVAVCEncGetOverrunBuffer(mHandle));
} else if (encoderStatus == AVCENC_PICTURE_READY) {
- CHECK_EQ(NULL, PVAVCEncGetOverrunBuffer(mHandle));
+ CHECK(NULL == PVAVCEncGetOverrunBuffer(mHandle));
if (mIsIDRFrame) {
outputBuffer->meta_data()->setInt32(kKeyIsSyncFrame, mIsIDRFrame);
mIsIDRFrame = 0;
diff --git a/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp b/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp
index d538603..20b0f8d 100644
--- a/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp
+++ b/media/libstagefright/codecs/m4v_h263/enc/M4vH263Encoder.cpp
@@ -23,8 +23,8 @@
#include "mp4enc_api.h"
#include "OMX_Video.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
@@ -379,7 +379,7 @@
*out = NULL;
MediaBuffer *outputBuffer;
- CHECK_EQ(OK, mGroup->acquire_buffer(&outputBuffer));
+ CHECK_EQ((status_t)OK, mGroup->acquire_buffer(&outputBuffer));
uint8_t *outPtr = (uint8_t *) outputBuffer->data();
int32_t dataLength = outputBuffer->size();
@@ -467,7 +467,7 @@
mInputBuffer = NULL;
return UNKNOWN_ERROR;
}
- CHECK_EQ(NULL, PVGetOverrunBuffer(mHandle));
+ CHECK(NULL == PVGetOverrunBuffer(mHandle));
if (hintTrack.CodeType == 0) { // I-frame serves as sync frame
outputBuffer->meta_data()->setInt32(kKeyIsSyncFrame, 1);
}
diff --git a/media/libstagefright/colorconversion/ColorConverter.cpp b/media/libstagefright/colorconversion/ColorConverter.cpp
index 5cc3f78..597167f 100644
--- a/media/libstagefright/colorconversion/ColorConverter.cpp
+++ b/media/libstagefright/colorconversion/ColorConverter.cpp
@@ -18,8 +18,8 @@
#define LOG_TAG "ColorConverter"
#include <utils/Log.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/ColorConverter.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaErrors.h>
namespace android {
@@ -144,8 +144,8 @@
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_ptr = (const uint8_t *)src.mBits
+ (src.mCropTop * dst.mWidth + src.mCropLeft) * 2;
@@ -182,11 +182,15 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[b2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_ptr += src.mWidth * 2;
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
@@ -290,15 +294,14 @@
const BitmapParams &src, const BitmapParams &dst) {
uint8_t *kAdjustedClip = initClip();
- if (!((dst.mWidth & 3) == 0
- && (src.mCropLeft & 1) == 0
+ if (!((src.mCropLeft & 1) == 0
&& src.cropWidth() == dst.cropWidth()
&& src.cropHeight() == dst.cropHeight())) {
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_y =
(const uint8_t *)src.mBits + src.mCropTop * src.mWidth + src.mCropLeft;
@@ -340,7 +343,11 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[r2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_y += src.mWidth;
@@ -349,7 +356,7 @@
src_u += src.mWidth;
}
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
@@ -361,15 +368,14 @@
uint8_t *kAdjustedClip = initClip();
- if (!((dst.mWidth & 3) == 0
- && (src.mCropLeft & 1) == 0
+ if (!((src.mCropLeft & 1) == 0
&& src.cropWidth() == dst.cropWidth()
&& src.cropHeight() == dst.cropHeight())) {
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_y =
(const uint8_t *)src.mBits + src.mCropTop * src.mWidth + src.mCropLeft;
@@ -411,7 +417,11 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[r2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_y += src.mWidth;
@@ -420,7 +430,7 @@
src_u += src.mWidth;
}
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
@@ -430,15 +440,14 @@
const BitmapParams &src, const BitmapParams &dst) {
uint8_t *kAdjustedClip = initClip();
- if (!((dst.mWidth & 3) == 0
- && (src.mCropLeft & 1) == 0
+ if (!((src.mCropLeft & 1) == 0
&& src.cropWidth() == dst.cropWidth()
&& src.cropHeight() == dst.cropHeight())) {
return ERROR_UNSUPPORTED;
}
- uint32_t *dst_ptr = (uint32_t *)dst.mBits
- + (dst.mCropTop * dst.mWidth + dst.mCropLeft) / 2;
+ uint16_t *dst_ptr = (uint16_t *)dst.mBits
+ + dst.mCropTop * dst.mWidth + dst.mCropLeft;
const uint8_t *src_y = (const uint8_t *)src.mBits;
@@ -478,7 +487,11 @@
| ((kAdjustedClip[g2] >> 2) << 5)
| (kAdjustedClip[b2] >> 3);
- dst_ptr[x / 2] = (rgb2 << 16) | rgb1;
+ if (x + 1 < src.cropWidth()) {
+ *(uint32_t *)(&dst_ptr[x]) = (rgb2 << 16) | rgb1;
+ } else {
+ dst_ptr[x] = rgb1;
+ }
}
src_y += src.mWidth;
@@ -487,7 +500,7 @@
src_u += src.mWidth;
}
- dst_ptr += dst.mWidth / 2;
+ dst_ptr += dst.mWidth;
}
return OK;
diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
index e892f92..b8516af 100644
--- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp
+++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
@@ -23,8 +23,7 @@
#include <binder/MemoryHeapPmem.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MetaData.h>
-#include <surfaceflinger/Surface.h>
-#include <ui/android_native_buffer.h>
+#include <system/window.h>
#include <ui/GraphicBufferMapper.h>
#include <gui/ISurfaceTexture.h>
diff --git a/media/libstagefright/foundation/AMessage.cpp b/media/libstagefright/foundation/AMessage.cpp
index 0a6776e..9a00186 100644
--- a/media/libstagefright/foundation/AMessage.cpp
+++ b/media/libstagefright/foundation/AMessage.cpp
@@ -19,6 +19,7 @@
#include <ctype.h>
#include "AAtomizer.h"
+#include "ABuffer.h"
#include "ADebug.h"
#include "ALooperRoster.h"
#include "AString.h"
@@ -157,14 +158,23 @@
item->u.stringValue = new AString(s, len < 0 ? strlen(s) : len);
}
-void AMessage::setObject(const char *name, const sp<RefBase> &obj) {
+void AMessage::setObjectInternal(
+ const char *name, const sp<RefBase> &obj, Type type) {
Item *item = allocateItem(name);
- item->mType = kTypeObject;
+ item->mType = type;
if (obj != NULL) { obj->incStrong(this); }
item->u.refValue = obj.get();
}
+void AMessage::setObject(const char *name, const sp<RefBase> &obj) {
+ setObjectInternal(name, obj, kTypeObject);
+}
+
+void AMessage::setBuffer(const char *name, const sp<ABuffer> &buffer) {
+ setObjectInternal(name, sp<RefBase>(buffer), kTypeBuffer);
+}
+
void AMessage::setMessage(const char *name, const sp<AMessage> &obj) {
Item *item = allocateItem(name);
item->mType = kTypeMessage;
@@ -203,6 +213,15 @@
return false;
}
+bool AMessage::findBuffer(const char *name, sp<ABuffer> *buf) const {
+ const Item *item = findItem(name, kTypeBuffer);
+ if (item) {
+ *buf = (ABuffer *)(item->u.refValue);
+ return true;
+ }
+ return false;
+}
+
bool AMessage::findMessage(const char *name, sp<AMessage> *obj) const {
const Item *item = findItem(name, kTypeMessage);
if (item) {
@@ -542,4 +561,20 @@
}
}
+size_t AMessage::countEntries() const {
+ return mNumItems;
+}
+
+const char *AMessage::getEntryNameAt(size_t index, Type *type) const {
+ if (index >= mNumItems) {
+ *type = kTypeInt32;
+
+ return NULL;
+ }
+
+ *type = mItems[index].mType;
+
+ return mItems[index].mName;
+}
+
} // namespace android
diff --git a/media/libstagefright/httplive/LiveSession.cpp b/media/libstagefright/httplive/LiveSession.cpp
index 0df66f1..0cddd2e 100644
--- a/media/libstagefright/httplive/LiveSession.cpp
+++ b/media/libstagefright/httplive/LiveSession.cpp
@@ -215,7 +215,9 @@
mDisconnectPending = false;
}
-status_t LiveSession::fetchFile(const char *url, sp<ABuffer> *out) {
+status_t LiveSession::fetchFile(
+ const char *url, sp<ABuffer> *out,
+ int64_t range_offset, int64_t range_length) {
*out = NULL;
sp<DataSource> source;
@@ -234,8 +236,18 @@
}
}
- status_t err = mHTTPDataSource->connect(
- url, mExtraHeaders.isEmpty() ? NULL : &mExtraHeaders);
+ KeyedVector<String8, String8> headers = mExtraHeaders;
+ if (range_offset > 0 || range_length >= 0) {
+ headers.add(
+ String8("Range"),
+ String8(
+ StringPrintf(
+ "bytes=%lld-%s",
+ range_offset,
+ range_length < 0
+ ? "" : StringPrintf("%lld", range_offset + range_length - 1).c_str()).c_str()));
+ }
+ status_t err = mHTTPDataSource->connect(url, &headers);
if (err != OK) {
return err;
@@ -270,9 +282,21 @@
buffer = copy;
}
+ size_t maxBytesToRead = bufferRemaining;
+ if (range_length >= 0) {
+ int64_t bytesLeftInRange = range_length - buffer->size();
+ if (bytesLeftInRange < maxBytesToRead) {
+ maxBytesToRead = bytesLeftInRange;
+
+ if (bytesLeftInRange == 0) {
+ break;
+ }
+ }
+ }
+
ssize_t n = source->readAt(
buffer->size(), buffer->data() + buffer->size(),
- bufferRemaining);
+ maxBytesToRead);
if (n < 0) {
return n;
@@ -659,8 +683,15 @@
explicitDiscontinuity = true;
}
+ int64_t range_offset, range_length;
+ if (!itemMeta->findInt64("range-offset", &range_offset)
+ || !itemMeta->findInt64("range-length", &range_length)) {
+ range_offset = 0;
+ range_length = -1;
+ }
+
sp<ABuffer> buffer;
- status_t err = fetchFile(uri.c_str(), &buffer);
+ status_t err = fetchFile(uri.c_str(), &buffer, range_offset, range_length);
if (err != OK) {
ALOGE("failed to fetch .ts segment at url '%s'", uri.c_str());
mDataSource->queueEOS(err);
diff --git a/media/libstagefright/httplive/M3UParser.cpp b/media/libstagefright/httplive/M3UParser.cpp
index 5e30488..7d3cf05 100644
--- a/media/libstagefright/httplive/M3UParser.cpp
+++ b/media/libstagefright/httplive/M3UParser.cpp
@@ -152,6 +152,7 @@
const char *data = (const char *)_data;
size_t offset = 0;
+ uint64_t segmentRangeOffset = 0;
while (offset < size) {
size_t offsetLF = offset;
while (offsetLF < size && data[offsetLF] != '\n') {
@@ -218,6 +219,24 @@
}
mIsVariantPlaylist = true;
err = parseStreamInf(line, &itemMeta);
+ } else if (line.startsWith("#EXT-X-BYTERANGE")) {
+ if (mIsVariantPlaylist) {
+ return ERROR_MALFORMED;
+ }
+
+ uint64_t length, offset;
+ err = parseByteRange(line, segmentRangeOffset, &length, &offset);
+
+ if (err == OK) {
+ if (itemMeta == NULL) {
+ itemMeta = new AMessage;
+ }
+
+ itemMeta->setInt64("range-offset", offset);
+ itemMeta->setInt64("range-length", length);
+
+ segmentRangeOffset = offset + length;
+ }
}
if (err != OK) {
@@ -447,6 +466,52 @@
}
// static
+status_t M3UParser::parseByteRange(
+ const AString &line, uint64_t curOffset,
+ uint64_t *length, uint64_t *offset) {
+ ssize_t colonPos = line.find(":");
+
+ if (colonPos < 0) {
+ return ERROR_MALFORMED;
+ }
+
+ ssize_t atPos = line.find("@", colonPos + 1);
+
+ AString lenStr;
+ if (atPos < 0) {
+ lenStr = AString(line, colonPos + 1, line.size() - colonPos - 1);
+ } else {
+ lenStr = AString(line, colonPos + 1, atPos - colonPos - 1);
+ }
+
+ lenStr.trim();
+
+ const char *s = lenStr.c_str();
+ char *end;
+ *length = strtoull(s, &end, 10);
+
+ if (s == end || *end != '\0') {
+ return ERROR_MALFORMED;
+ }
+
+ if (atPos >= 0) {
+ AString offStr = AString(line, atPos + 1, line.size() - atPos - 1);
+ offStr.trim();
+
+ const char *s = offStr.c_str();
+ *offset = strtoull(s, &end, 10);
+
+ if (s == end || *end != '\0') {
+ return ERROR_MALFORMED;
+ }
+ } else {
+ *offset = curOffset;
+ }
+
+ return OK;
+}
+
+// static
status_t M3UParser::ParseInt32(const char *s, int32_t *x) {
char *end;
long lval = strtol(s, &end, 10);
diff --git a/media/libstagefright/id3/Android.mk b/media/libstagefright/id3/Android.mk
index 23c8e44..ff35d4a 100644
--- a/media/libstagefright/id3/Android.mk
+++ b/media/libstagefright/id3/Android.mk
@@ -16,7 +16,7 @@
testid3.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright libutils libbinder
+ libstagefright libutils libbinder libstagefright_foundation
LOCAL_STATIC_LIBRARIES := \
libstagefright_id3
diff --git a/media/libstagefright/id3/ID3.cpp b/media/libstagefright/id3/ID3.cpp
index 6dde9d8..2e92926 100644
--- a/media/libstagefright/id3/ID3.cpp
+++ b/media/libstagefright/id3/ID3.cpp
@@ -20,8 +20,8 @@
#include "../include/ID3.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/Utils.h>
#include <utils/String8.h>
#include <byteswap.h>
diff --git a/media/libstagefright/id3/testid3.cpp b/media/libstagefright/id3/testid3.cpp
index 0741045..bc4572c 100644
--- a/media/libstagefright/id3/testid3.cpp
+++ b/media/libstagefright/id3/testid3.cpp
@@ -23,7 +23,7 @@
#include <binder/ProcessState.h>
#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#define MAXPATHLEN 256
@@ -70,7 +70,7 @@
void scanFile(const char *path) {
sp<FileSource> file = new FileSource(path);
- CHECK_EQ(file->initCheck(), OK);
+ CHECK_EQ(file->initCheck(), (status_t)OK);
ID3 tag(file);
if (!tag.isValid()) {
diff --git a/media/libstagefright/include/AACExtractor.h b/media/libstagefright/include/AACExtractor.h
index 8e5657b..e98ca82 100644
--- a/media/libstagefright/include/AACExtractor.h
+++ b/media/libstagefright/include/AACExtractor.h
@@ -29,7 +29,7 @@
class AACExtractor : public MediaExtractor {
public:
- AACExtractor(const sp<DataSource> &source);
+ AACExtractor(const sp<DataSource> &source, const sp<AMessage> &meta);
virtual size_t countTracks();
virtual sp<MediaSource> getTrack(size_t index);
diff --git a/media/libstagefright/include/AwesomePlayer.h b/media/libstagefright/include/AwesomePlayer.h
index 0985f47..4c7bfa6 100644
--- a/media/libstagefright/include/AwesomePlayer.h
+++ b/media/libstagefright/include/AwesomePlayer.h
@@ -41,7 +41,7 @@
class DrmManagerClinet;
class DecryptHandle;
-class TimedTextPlayer;
+class TimedTextDriver;
struct WVMExtractor;
struct AwesomeRenderer : public RefBase {
@@ -232,7 +232,7 @@
sp<DecryptHandle> mDecryptHandle;
int64_t mLastVideoTimeUs;
- TimedTextPlayer *mTextPlayer;
+ TimedTextDriver *mTextDriver;
mutable Mutex mTimedTextLock;
sp<WVMExtractor> mWVMExtractor;
@@ -258,7 +258,7 @@
void setVideoSource(sp<MediaSource> source);
status_t initVideoDecoder(uint32_t flags = 0);
- void addTextSource(sp<MediaSource> source);
+ void addTextSource(const sp<MediaSource>& source);
void onStreamDone();
@@ -290,6 +290,7 @@
bool isStreamingHTTP() const;
void sendCacheStats();
+ void checkDrmStatus(const sp<DataSource>& dataSource);
enum FlagMode {
SET,
@@ -325,4 +326,3 @@
} // namespace android
#endif // AWESOME_PLAYER_H_
-
diff --git a/media/libstagefright/include/ChromiumHTTPDataSource.h b/media/libstagefright/include/ChromiumHTTPDataSource.h
index 18f8913..82e08fd 100644
--- a/media/libstagefright/include/ChromiumHTTPDataSource.h
+++ b/media/libstagefright/include/ChromiumHTTPDataSource.h
@@ -43,7 +43,7 @@
virtual status_t getSize(off64_t *size);
virtual uint32_t flags();
- virtual sp<DecryptHandle> DrmInitialization();
+ virtual sp<DecryptHandle> DrmInitialization(const char *mime);
virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
diff --git a/media/libstagefright/include/DataUriSource.h b/media/libstagefright/include/DataUriSource.h
new file mode 100644
index 0000000..d223c06
--- /dev/null
+++ b/media/libstagefright/include/DataUriSource.h
@@ -0,0 +1,76 @@
+/*
+ * Copyright (C) 2011 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef DATA_URI_SOURCE_H_
+
+#define DATA_URI_SOURCE_H_
+
+#include <stdio.h>
+
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/foundation/AString.h>
+
+namespace android {
+
+class DataUriSource : public DataSource {
+public:
+ DataUriSource(const char *uri);
+
+ virtual status_t initCheck() const {
+ return mInited;
+ }
+
+ virtual ssize_t readAt(off64_t offset, void *data, size_t size);
+
+ virtual status_t getSize(off64_t *size) {
+ if (mInited != OK) {
+ return mInited;
+ }
+
+ *size = mData.size();
+ return OK;
+ }
+
+ virtual String8 getUri() {
+ return mDataUri;
+ }
+
+ virtual String8 getMIMEType() const {
+ return mMimeType;
+ }
+
+protected:
+ virtual ~DataUriSource() {
+ // Nothing to delete.
+ }
+
+private:
+ const String8 mDataUri;
+
+ String8 mMimeType;
+ // Use AString because individual bytes may not be valid UTF8 chars.
+ AString mData;
+ status_t mInited;
+
+ // Disallow copy and assign.
+ DataUriSource(const DataUriSource &);
+ DataUriSource &operator=(const DataUriSource &);
+};
+
+} // namespace android
+
+#endif // DATA_URI_SOURCE_H_
diff --git a/media/libstagefright/include/LiveSession.h b/media/libstagefright/include/LiveSession.h
index 116ed0e..3a11612 100644
--- a/media/libstagefright/include/LiveSession.h
+++ b/media/libstagefright/include/LiveSession.h
@@ -120,7 +120,10 @@
void onMonitorQueue();
void onSeek(const sp<AMessage> &msg);
- status_t fetchFile(const char *url, sp<ABuffer> *out);
+ status_t fetchFile(
+ const char *url, sp<ABuffer> *out,
+ int64_t range_offset = 0, int64_t range_length = -1);
+
sp<M3UParser> fetchPlaylist(const char *url, bool *unchanged);
size_t getBandwidthIndex();
diff --git a/media/libstagefright/include/M3UParser.h b/media/libstagefright/include/M3UParser.h
index 478582d..e30d6fd 100644
--- a/media/libstagefright/include/M3UParser.h
+++ b/media/libstagefright/include/M3UParser.h
@@ -72,6 +72,10 @@
static status_t parseCipherInfo(
const AString &line, sp<AMessage> *meta, const AString &baseURI);
+ static status_t parseByteRange(
+ const AString &line, uint64_t curOffset,
+ uint64_t *length, uint64_t *offset);
+
static status_t ParseInt32(const char *s, int32_t *x);
static status_t ParseDouble(const char *s, double *x);
diff --git a/media/libstagefright/include/NuCachedSource2.h b/media/libstagefright/include/NuCachedSource2.h
index 7a03e7e..c27a29b 100644
--- a/media/libstagefright/include/NuCachedSource2.h
+++ b/media/libstagefright/include/NuCachedSource2.h
@@ -40,7 +40,7 @@
virtual status_t getSize(off64_t *size);
virtual uint32_t flags();
- virtual sp<DecryptHandle> DrmInitialization();
+ virtual sp<DecryptHandle> DrmInitialization(const char* mime);
virtual void getDrmInfo(sp<DecryptHandle> &handle, DrmManagerClient **client);
virtual String8 getUri();
diff --git a/media/libstagefright/include/OMX.h b/media/libstagefright/include/OMX.h
index 53e764f..2c87b34 100644
--- a/media/libstagefright/include/OMX.h
+++ b/media/libstagefright/include/OMX.h
@@ -31,7 +31,7 @@
public:
OMX();
- virtual bool livesLocally(pid_t pid);
+ virtual bool livesLocally(node_id node, pid_t pid);
virtual status_t listNodes(List<ComponentInfo> *list);
diff --git a/media/libstagefright/include/SoftwareRenderer.h b/media/libstagefright/include/SoftwareRenderer.h
index 8f2ea95..7ab0042 100644
--- a/media/libstagefright/include/SoftwareRenderer.h
+++ b/media/libstagefright/include/SoftwareRenderer.h
@@ -20,7 +20,7 @@
#include <media/stagefright/ColorConverter.h>
#include <utils/RefBase.h>
-#include <ui/android_native_buffer.h>
+#include <system/window.h>
namespace android {
diff --git a/media/libstagefright/include/ThrottledSource.h b/media/libstagefright/include/ThrottledSource.h
index 8928a4a..7fe7c06 100644
--- a/media/libstagefright/include/ThrottledSource.h
+++ b/media/libstagefright/include/ThrottledSource.h
@@ -35,6 +35,11 @@
virtual status_t getSize(off64_t *size);
virtual uint32_t flags();
+ virtual String8 getMIMEType() const {
+ return mSource->getMIMEType();
+ }
+
+
private:
Mutex mLock;
diff --git a/media/libstagefright/include/WVMExtractor.h b/media/libstagefright/include/WVMExtractor.h
index deecd25..9f763f9 100644
--- a/media/libstagefright/include/WVMExtractor.h
+++ b/media/libstagefright/include/WVMExtractor.h
@@ -23,6 +23,8 @@
namespace android {
+struct AMessage;
+class String8;
class DataSource;
class WVMLoadableExtractor : public MediaExtractor {
@@ -58,6 +60,8 @@
// is used.
void setAdaptiveStreamingMode(bool adaptive);
+ static bool getVendorLibHandle();
+
protected:
virtual ~WVMExtractor();
@@ -69,6 +73,10 @@
WVMExtractor &operator=(const WVMExtractor &);
};
+bool SniffWVM(
+ const sp<DataSource> &source, String8 *mimeType, float *confidence,
+ sp<AMessage> *);
+
} // namespace android
#endif // DRM_EXTRACTOR_H_
diff --git a/media/libstagefright/matroska/MatroskaExtractor.cpp b/media/libstagefright/matroska/MatroskaExtractor.cpp
index 4fbf47e..a0db719 100644
--- a/media/libstagefright/matroska/MatroskaExtractor.cpp
+++ b/media/libstagefright/matroska/MatroskaExtractor.cpp
@@ -93,7 +93,10 @@
void advance();
void reset();
- void seek(int64_t seekTimeUs, bool seekToKeyFrame);
+
+ void seek(
+ int64_t seekTimeUs, bool seekToKeyFrame,
+ int64_t *actualFrameTimeUs);
const mkvparser::Block *block() const;
int64_t blockTimeUs() const;
@@ -303,22 +306,52 @@
} while (!eos() && block()->GetTrackNumber() != mTrackNum);
}
-void BlockIterator::seek(int64_t seekTimeUs, bool seekToKeyFrame) {
+void BlockIterator::seek(
+ int64_t seekTimeUs, bool seekToKeyFrame,
+ int64_t *actualFrameTimeUs) {
Mutex::Autolock autoLock(mExtractor->mLock);
- mCluster = mExtractor->mSegment->FindCluster(seekTimeUs * 1000ll);
+ *actualFrameTimeUs = -1ll;
+
+ int64_t seekTimeNs = seekTimeUs * 1000ll;
+
+ mCluster = mExtractor->mSegment->FindCluster(seekTimeNs);
mBlockEntry = NULL;
mBlockEntryIndex = 0;
- do {
- advance_l();
- }
- while (!eos() && block()->GetTrackNumber() != mTrackNum);
+ long prevKeyFrameBlockEntryIndex = -1;
- if (seekToKeyFrame) {
- while (!eos() && !mBlockEntry->GetBlock()->IsKey()) {
- advance_l();
+ for (;;) {
+ advance_l();
+
+ if (eos()) {
+ break;
}
+
+ if (block()->GetTrackNumber() != mTrackNum) {
+ continue;
+ }
+
+ if (block()->IsKey()) {
+ prevKeyFrameBlockEntryIndex = mBlockEntryIndex - 1;
+ }
+
+ int64_t timeNs = block()->GetTime(mCluster);
+
+ if (timeNs >= seekTimeNs) {
+ *actualFrameTimeUs = (timeNs + 500ll) / 1000ll;
+ break;
+ }
+ }
+
+ if (eos()) {
+ return;
+ }
+
+ if (seekToKeyFrame && !block()->IsKey()) {
+ CHECK_GE(prevKeyFrameBlockEntryIndex, 0);
+ mBlockEntryIndex = prevKeyFrameBlockEntryIndex;
+ advance_l();
}
}
@@ -397,6 +430,8 @@
MediaBuffer **out, const ReadOptions *options) {
*out = NULL;
+ int64_t targetSampleTimeUs = -1ll;
+
int64_t seekTimeUs;
ReadOptions::SeekMode mode;
if (options && options->getSeekTo(&seekTimeUs, &mode)
@@ -406,10 +441,14 @@
// Apparently keyframe indication in audio tracks is unreliable,
// fortunately in all our currently supported audio encodings every
// frame is effectively a keyframe.
- mBlockIter.seek(seekTimeUs, !mIsAudio);
+ int64_t actualFrameTimeUs;
+ mBlockIter.seek(seekTimeUs, !mIsAudio, &actualFrameTimeUs);
+
+ if (mode == ReadOptions::SEEK_CLOSEST) {
+ targetSampleTimeUs = actualFrameTimeUs;
+ }
}
-again:
while (mPendingFrames.empty()) {
status_t err = readBlock();
@@ -424,6 +463,11 @@
mPendingFrames.erase(mPendingFrames.begin());
if (mType != AVC) {
+ if (targetSampleTimeUs >= 0ll) {
+ frame->meta_data()->setInt64(
+ kKeyTargetTime, targetSampleTimeUs);
+ }
+
*out = frame;
return OK;
@@ -506,6 +550,11 @@
frame->release();
frame = NULL;
+ if (targetSampleTimeUs >= 0ll) {
+ buffer->meta_data()->setInt64(
+ kKeyTargetTime, targetSampleTimeUs);
+ }
+
*out = buffer;
return OK;
@@ -610,36 +659,41 @@
return mIsLiveStreaming;
}
-static void addESDSFromAudioSpecificInfo(
- const sp<MetaData> &meta, const void *asi, size_t asiSize) {
+static void addESDSFromCodecPrivate(
+ const sp<MetaData> &meta,
+ bool isAudio, const void *priv, size_t privSize) {
static const uint8_t kStaticESDS[] = {
0x03, 22,
0x00, 0x00, // ES_ID
0x00, // streamDependenceFlag, URL_Flag, OCRstreamFlag
0x04, 17,
- 0x40, // Audio ISO/IEC 14496-3
+ 0x40, // ObjectTypeIndication
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x00, 0x00, 0x00, 0x00,
0x05,
- // AudioSpecificInfo (with size prefix) follows
+ // CodecSpecificInfo (with size prefix) follows
};
// Make sure all sizes can be coded in a single byte.
- CHECK(asiSize + 22 - 2 < 128);
- size_t esdsSize = sizeof(kStaticESDS) + asiSize + 1;
+ CHECK(privSize + 22 - 2 < 128);
+ size_t esdsSize = sizeof(kStaticESDS) + privSize + 1;
uint8_t *esds = new uint8_t[esdsSize];
memcpy(esds, kStaticESDS, sizeof(kStaticESDS));
uint8_t *ptr = esds + sizeof(kStaticESDS);
- *ptr++ = asiSize;
- memcpy(ptr, asi, asiSize);
+ *ptr++ = privSize;
+ memcpy(ptr, priv, privSize);
// Increment by codecPrivateSize less 2 bytes that are accounted for
// already in lengths of 22/17
- esds[1] += asiSize - 2;
- esds[6] += asiSize - 2;
+ esds[1] += privSize - 2;
+ esds[6] += privSize - 2;
+
+ // Set ObjectTypeIndication.
+ esds[7] = isAudio ? 0x40 // Audio ISO/IEC 14496-3
+ : 0x20; // Visual ISO/IEC 14496-2
meta->setData(kKeyESDS, 0, esds, esdsSize);
@@ -707,9 +761,21 @@
if (!strcmp("V_MPEG4/ISO/AVC", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
meta->setData(kKeyAVCC, 0, codecPrivate, codecPrivateSize);
+ } else if (!strcmp("V_MPEG4/ISO/ASP", codecID)) {
+ if (codecPrivateSize > 0) {
+ meta->setCString(
+ kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG4);
+ addESDSFromCodecPrivate(
+ meta, false, codecPrivate, codecPrivateSize);
+ } else {
+ ALOGW("%s is detected, but does not have configuration.",
+ codecID);
+ continue;
+ }
} else if (!strcmp("V_VP8", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_VPX);
} else {
+ ALOGW("%s is not supported.", codecID);
continue;
}
@@ -727,13 +793,16 @@
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AAC);
CHECK(codecPrivateSize >= 2);
- addESDSFromAudioSpecificInfo(
- meta, codecPrivate, codecPrivateSize);
+ addESDSFromCodecPrivate(
+ meta, true, codecPrivate, codecPrivateSize);
} else if (!strcmp("A_VORBIS", codecID)) {
meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_VORBIS);
addVorbisCodecInfo(meta, codecPrivate, codecPrivateSize);
+ } else if (!strcmp("A_MPEG/L3", codecID)) {
+ meta->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_MPEG);
} else {
+ ALOGW("%s is not supported.", codecID);
continue;
}
diff --git a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
index 03033f5..e1589b4 100644
--- a/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
+++ b/media/libstagefright/mpeg2ts/MPEG2TSExtractor.cpp
@@ -22,8 +22,8 @@
#include "include/LiveSession.h"
#include "include/NuCachedSource2.h"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaSource.h>
diff --git a/media/libstagefright/omx/OMX.cpp b/media/libstagefright/omx/OMX.cpp
index 694b12d..f11fcd2 100644
--- a/media/libstagefright/omx/OMX.cpp
+++ b/media/libstagefright/omx/OMX.cpp
@@ -25,7 +25,7 @@
#include "../include/OMXNodeInstance.h"
#include <binder/IMemory.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <utils/threads.h>
#include "OMXMaster.h"
@@ -102,7 +102,7 @@
if (status != WOULD_BLOCK) {
// Other than join to self, the only other error return codes are
// whatever readyToRun() returns, and we don't override that
- CHECK_EQ(status, NO_ERROR);
+ CHECK_EQ(status, (status_t)NO_ERROR);
}
}
@@ -185,7 +185,7 @@
instance->onObserverDied(mMaster);
}
-bool OMX::livesLocally(pid_t pid) {
+bool OMX::livesLocally(node_id node, pid_t pid) {
return pid == getpid();
}
diff --git a/media/libstagefright/omx/OMXComponentBase.cpp b/media/libstagefright/omx/OMXComponentBase.cpp
index 35227a0..7d11dce 100644
--- a/media/libstagefright/omx/OMXComponentBase.cpp
+++ b/media/libstagefright/omx/OMXComponentBase.cpp
@@ -18,7 +18,7 @@
#include <stdlib.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
namespace android {
@@ -33,7 +33,7 @@
OMXComponentBase::~OMXComponentBase() {}
void OMXComponentBase::setComponentHandle(OMX_COMPONENTTYPE *handle) {
- CHECK_EQ(mComponentHandle, NULL);
+ CHECK(mComponentHandle == NULL);
mComponentHandle = handle;
}
diff --git a/media/libstagefright/omx/OMXMaster.cpp b/media/libstagefright/omx/OMXMaster.cpp
index d698939..6b6d0ab 100644
--- a/media/libstagefright/omx/OMXMaster.cpp
+++ b/media/libstagefright/omx/OMXMaster.cpp
@@ -24,7 +24,7 @@
#include <dlfcn.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
namespace android {
diff --git a/media/libstagefright/omx/OMXNodeInstance.cpp b/media/libstagefright/omx/OMXNodeInstance.cpp
index 8938e33..099c4f5 100644
--- a/media/libstagefright/omx/OMXNodeInstance.cpp
+++ b/media/libstagefright/omx/OMXNodeInstance.cpp
@@ -24,8 +24,8 @@
#include <OMX_Component.h>
#include <binder/IMemory.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/HardwareAPI.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaErrors.h>
namespace android {
@@ -91,11 +91,11 @@
}
OMXNodeInstance::~OMXNodeInstance() {
- CHECK_EQ(mHandle, NULL);
+ CHECK(mHandle == NULL);
}
void OMXNodeInstance::setHandle(OMX::node_id node_id, OMX_HANDLETYPE handle) {
- CHECK_EQ(mHandle, NULL);
+ CHECK(mHandle == NULL);
mNodeID = node_id;
mHandle = handle;
}
diff --git a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
index 0914f32..c79e01f 100644
--- a/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
+++ b/media/libstagefright/omx/SimpleSoftOMXComponent.cpp
@@ -333,8 +333,9 @@
void SimpleSoftOMXComponent::onMessageReceived(const sp<AMessage> &msg) {
Mutex::Autolock autoLock(mLock);
-
- switch (msg->what()) {
+ uint32_t msgType = msg->what();
+ ALOGV("msgType = %d", msgType);
+ switch (msgType) {
case kWhatSendCommand:
{
int32_t cmd, param;
@@ -354,27 +355,27 @@
CHECK(mState == OMX_StateExecuting && mTargetState == mState);
bool found = false;
- for (size_t i = 0; i < mPorts.size(); ++i) {
- PortInfo *port = &mPorts.editItemAt(i);
+ size_t portIndex = (kWhatEmptyThisBuffer == msgType)?
+ header->nInputPortIndex: header->nOutputPortIndex;
+ PortInfo *port = &mPorts.editItemAt(portIndex);
- for (size_t j = 0; j < port->mBuffers.size(); ++j) {
- BufferInfo *buffer = &port->mBuffers.editItemAt(j);
+ for (size_t j = 0; j < port->mBuffers.size(); ++j) {
+ BufferInfo *buffer = &port->mBuffers.editItemAt(j);
- if (buffer->mHeader == header) {
- CHECK(!buffer->mOwnedByUs);
+ if (buffer->mHeader == header) {
+ CHECK(!buffer->mOwnedByUs);
- buffer->mOwnedByUs = true;
+ buffer->mOwnedByUs = true;
- CHECK((msg->what() == kWhatEmptyThisBuffer
- && port->mDef.eDir == OMX_DirInput)
- || (port->mDef.eDir == OMX_DirOutput));
+ CHECK((msgType == kWhatEmptyThisBuffer
+ && port->mDef.eDir == OMX_DirInput)
+ || (port->mDef.eDir == OMX_DirOutput));
- port->mQueue.push_back(buffer);
- onQueueFilled(i);
+ port->mQueue.push_back(buffer);
+ onQueueFilled(portIndex);
- found = true;
- break;
- }
+ found = true;
+ break;
}
}
diff --git a/media/libstagefright/omx/SoftOMXPlugin.cpp b/media/libstagefright/omx/SoftOMXPlugin.cpp
index da3ae42..99ffe7d 100644
--- a/media/libstagefright/omx/SoftOMXPlugin.cpp
+++ b/media/libstagefright/omx/SoftOMXPlugin.cpp
@@ -35,8 +35,11 @@
} kComponents[] = {
{ "OMX.google.aac.decoder", "aacdec", "audio_decoder.aac" },
+ { "OMX.google.aac.encoder", "aacenc", "audio_encoder.aac" },
{ "OMX.google.amrnb.decoder", "amrdec", "audio_decoder.amrnb" },
+ { "OMX.google.amrnb.encoder", "amrnbenc", "audio_encoder.amrnb" },
{ "OMX.google.amrwb.decoder", "amrdec", "audio_decoder.amrwb" },
+ { "OMX.google.amrwb.encoder", "amrwbenc", "audio_encoder.amrwb" },
{ "OMX.google.h264.decoder", "h264dec", "video_decoder.avc" },
{ "OMX.google.g711.alaw.decoder", "g711dec", "audio_decoder.g711alaw" },
{ "OMX.google.g711.mlaw.decoder", "g711dec", "audio_decoder.g711mlaw" },
diff --git a/media/libstagefright/omx/tests/Android.mk b/media/libstagefright/omx/tests/Android.mk
index bf69428..0c0a70c 100644
--- a/media/libstagefright/omx/tests/Android.mk
+++ b/media/libstagefright/omx/tests/Android.mk
@@ -5,13 +5,15 @@
OMXHarness.cpp \
LOCAL_SHARED_LIBRARIES := \
- libstagefright libbinder libmedia libutils
+ libstagefright libbinder libmedia libutils libstagefright_foundation
-LOCAL_C_INCLUDES:= \
+LOCAL_C_INCLUDES := \
$(JNI_H_INCLUDE) \
frameworks/base/media/libstagefright \
$(TOP)/frameworks/base/include/media/stagefright/openmax
-LOCAL_MODULE:= omx_tests
+LOCAL_MODULE := omx_tests
+
+LOCAL_MODULE_TAGS := tests
include $(BUILD_EXECUTABLE)
diff --git a/media/libstagefright/omx/tests/OMXHarness.cpp b/media/libstagefright/omx/tests/OMXHarness.cpp
index 8faf544..fab1771 100644
--- a/media/libstagefright/omx/tests/OMXHarness.cpp
+++ b/media/libstagefright/omx/tests/OMXHarness.cpp
@@ -26,9 +26,9 @@
#include <binder/IServiceManager.h>
#include <binder/MemoryDealer.h>
#include <media/IMediaPlayerService.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/DataSource.h>
#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/MediaDebug.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaExtractor.h>
@@ -155,7 +155,7 @@
if (err == TIMED_OUT) {
return err;
}
- CHECK_EQ(err, OK);
+ CHECK_EQ(err, (status_t)OK);
}
}
@@ -317,7 +317,7 @@
EXPECT_SUCCESS(err, "allocatePortBuffers(input)");
err = dequeueMessageForNode(node, &msg, DEFAULT_TIMEOUT);
- CHECK_EQ(err, TIMED_OUT);
+ CHECK_EQ(err, (status_t)TIMED_OUT);
Vector<Buffer> outputBuffers;
err = allocatePortBuffers(dealer, node, 1, &outputBuffers);
@@ -412,7 +412,7 @@
// Make sure node doesn't just transition to loaded before we are done
// freeing all input and output buffers.
err = dequeueMessageForNode(node, &msg, DEFAULT_TIMEOUT);
- CHECK_EQ(err, TIMED_OUT);
+ CHECK_EQ(err, (status_t)TIMED_OUT);
for (size_t i = 0; i < inputBuffers.size(); ++i) {
err = mOMX->freeBuffer(node, 0, inputBuffers[i].mID);
@@ -420,7 +420,7 @@
}
err = dequeueMessageForNode(node, &msg, DEFAULT_TIMEOUT);
- CHECK_EQ(err, TIMED_OUT);
+ CHECK_EQ(err, (status_t)TIMED_OUT);
for (size_t i = 0; i < outputBuffers.size(); ++i) {
err = mOMX->freeBuffer(node, 1, outputBuffers[i].mID);
@@ -584,7 +584,7 @@
return UNKNOWN_ERROR;
}
- CHECK_EQ(seekSource->start(), OK);
+ CHECK_EQ(seekSource->start(), (status_t)OK);
sp<MediaSource> codec = OMXCodec::Create(
mOMX, source->getFormat(), false /* createEncoder */,
@@ -592,7 +592,7 @@
CHECK(codec != NULL);
- CHECK_EQ(codec->start(), OK);
+ CHECK_EQ(codec->start(), (status_t)OK);
int64_t durationUs;
CHECK(source->getFormat()->findInt64(kKeyDuration, &durationUs));
@@ -638,7 +638,7 @@
requestedSeekTimeUs, MediaSource::ReadOptions::SEEK_NEXT_SYNC);
if (seekSource->read(&buffer, &options) != OK) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
actualSeekTimeUs = -1;
} else {
CHECK(buffer != NULL);
@@ -659,7 +659,7 @@
err = codec->read(&buffer, &options);
options.clearSeekTo();
if (err == INFO_FORMAT_CHANGED) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
continue;
}
if (err == OK) {
@@ -670,7 +670,7 @@
continue;
}
} else {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
}
break;
@@ -679,7 +679,7 @@
if (requestedSeekTimeUs < 0) {
// Linear read.
if (err != OK) {
- CHECK_EQ(buffer, NULL);
+ CHECK(buffer == NULL);
} else {
CHECK(buffer != NULL);
buffer->release();
@@ -694,8 +694,8 @@
"We attempted to seek beyond EOS and expected "
"ERROR_END_OF_STREAM to be returned, but instead "
"we found some other error.");
- CHECK_EQ(err, ERROR_END_OF_STREAM);
- CHECK_EQ(buffer, NULL);
+ CHECK_EQ(err, (status_t)ERROR_END_OF_STREAM);
+ CHECK(buffer == NULL);
} else {
EXPECT(err == OK,
"Expected a valid buffer to be returned from "
@@ -715,7 +715,7 @@
buffer->release();
buffer = NULL;
- CHECK_EQ(codec->stop(), OK);
+ CHECK_EQ(codec->stop(), (status_t)OK);
return UNKNOWN_ERROR;
}
@@ -725,7 +725,7 @@
}
}
- CHECK_EQ(codec->stop(), OK);
+ CHECK_EQ(codec->stop(), (status_t)OK);
return OK;
}
@@ -841,7 +841,7 @@
srand(seed);
sp<Harness> h = new Harness;
- CHECK_EQ(h->initCheck(), OK);
+ CHECK_EQ(h->initCheck(), (status_t)OK);
if (argc == 0) {
h->testAll();
diff --git a/media/libstagefright/rtsp/AAMRAssembler.cpp b/media/libstagefright/rtsp/AAMRAssembler.cpp
index 9d72b1f..fb8abc5 100644
--- a/media/libstagefright/rtsp/AAMRAssembler.cpp
+++ b/media/libstagefright/rtsp/AAMRAssembler.cpp
@@ -211,7 +211,7 @@
}
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
queue->erase(queue->begin());
diff --git a/media/libstagefright/rtsp/AAVCAssembler.cpp b/media/libstagefright/rtsp/AAVCAssembler.cpp
index ed8b1df..7ea132e 100644
--- a/media/libstagefright/rtsp/AAVCAssembler.cpp
+++ b/media/libstagefright/rtsp/AAVCAssembler.cpp
@@ -345,7 +345,7 @@
mAccessUnitDamaged = false;
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/rtsp/AH263Assembler.cpp b/media/libstagefright/rtsp/AH263Assembler.cpp
index 498295c..ded70fa 100644
--- a/media/libstagefright/rtsp/AH263Assembler.cpp
+++ b/media/libstagefright/rtsp/AH263Assembler.cpp
@@ -166,7 +166,7 @@
mAccessUnitDamaged = false;
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp b/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
index b0c7007..24c2f30 100644
--- a/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
+++ b/media/libstagefright/rtsp/AMPEG4AudioAssembler.cpp
@@ -571,7 +571,7 @@
mAccessUnitDamaged = false;
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/rtsp/AMPEG4ElementaryAssembler.cpp b/media/libstagefright/rtsp/AMPEG4ElementaryAssembler.cpp
index 2f2e2c2..687d72b 100644
--- a/media/libstagefright/rtsp/AMPEG4ElementaryAssembler.cpp
+++ b/media/libstagefright/rtsp/AMPEG4ElementaryAssembler.cpp
@@ -368,7 +368,7 @@
mAccessUnitDamaged = false;
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", accessUnit);
+ msg->setBuffer("access-unit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/rtsp/ARTPConnection.cpp b/media/libstagefright/rtsp/ARTPConnection.cpp
index 8c9dd8d..44988a3 100644
--- a/media/libstagefright/rtsp/ARTPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTPConnection.cpp
@@ -639,7 +639,7 @@
void ARTPConnection::injectPacket(int index, const sp<ABuffer> &buffer) {
sp<AMessage> msg = new AMessage(kWhatInjectPacket, id());
msg->setInt32("index", index);
- msg->setObject("buffer", buffer);
+ msg->setBuffer("buffer", buffer);
msg->post();
}
@@ -647,10 +647,8 @@
int32_t index;
CHECK(msg->findInt32("index", &index));
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
-
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
List<StreamInfo>::iterator it = mStreams.begin();
while (it != mStreams.end()
diff --git a/media/libstagefright/rtsp/ARTPSession.cpp b/media/libstagefright/rtsp/ARTPSession.cpp
index 7a05b88..ba4e33c 100644
--- a/media/libstagefright/rtsp/ARTPSession.cpp
+++ b/media/libstagefright/rtsp/ARTPSession.cpp
@@ -145,10 +145,8 @@
break;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("access-unit", &obj));
-
- sp<ABuffer> accessUnit = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("access-unit", &accessUnit));
uint64_t ntpTime;
CHECK(accessUnit->meta()->findInt64(
diff --git a/media/libstagefright/rtsp/ARTSPConnection.cpp b/media/libstagefright/rtsp/ARTSPConnection.cpp
index 80a010e..539a888 100644
--- a/media/libstagefright/rtsp/ARTSPConnection.cpp
+++ b/media/libstagefright/rtsp/ARTSPConnection.cpp
@@ -612,7 +612,7 @@
if (mObserveBinaryMessage != NULL) {
sp<AMessage> notify = mObserveBinaryMessage->dup();
- notify->setObject("buffer", buffer);
+ notify->setBuffer("buffer", buffer);
notify->post();
} else {
ALOGW("received binary data, but no one cares.");
diff --git a/media/libstagefright/rtsp/ARawAudioAssembler.cpp b/media/libstagefright/rtsp/ARawAudioAssembler.cpp
index 98bee82..0da5dd2 100644
--- a/media/libstagefright/rtsp/ARawAudioAssembler.cpp
+++ b/media/libstagefright/rtsp/ARawAudioAssembler.cpp
@@ -94,7 +94,7 @@
}
sp<AMessage> msg = mNotifyMsg->dup();
- msg->setObject("access-unit", buffer);
+ msg->setBuffer("access-unit", buffer);
msg->post();
queue->erase(queue->begin());
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index 2391c5c..deee30f 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -122,6 +122,7 @@
mSetupTracksSuccessful(false),
mSeekPending(false),
mFirstAccessUnit(true),
+ mAllTracksHaveTime(false),
mNTPAnchorUs(-1),
mMediaAnchorUs(-1),
mLastMediaTimeUs(0),
@@ -723,6 +724,7 @@
mSetupTracksSuccessful = false;
mSeekPending = false;
mFirstAccessUnit = true;
+ mAllTracksHaveTime = false;
mNTPAnchorUs = -1;
mMediaAnchorUs = -1;
mNumAccessUnitsReceived = 0;
@@ -855,10 +857,8 @@
return;
}
- sp<RefBase> obj;
- CHECK(msg->findObject("access-unit", &obj));
-
- sp<ABuffer> accessUnit = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> accessUnit;
+ CHECK(msg->findBuffer("access-unit", &accessUnit));
uint32_t seqNum = (uint32_t)accessUnit->int32Data();
@@ -930,6 +930,7 @@
info->mNTPAnchorUs = -1;
}
+ mAllTracksHaveTime = false;
mNTPAnchorUs = -1;
int64_t timeUs;
@@ -1002,9 +1003,8 @@
case 'biny':
{
- sp<RefBase> obj;
- CHECK(msg->findObject("buffer", &obj));
- sp<ABuffer> buffer = static_cast<ABuffer *>(obj.get());
+ sp<ABuffer> buffer;
+ CHECK(msg->findBuffer("buffer", &buffer));
int32_t index;
CHECK(buffer->meta()->findInt32("index", &index));
@@ -1037,6 +1037,14 @@
ALOGW("Never received any data, disconnecting.");
(new AMessage('abor', id()))->post();
}
+ } else {
+ if (!mAllTracksHaveTime) {
+ ALOGW("We received some RTCP packets, but time "
+ "could not be established on all tracks, now "
+ "using fake timestamps");
+
+ fakeTimestamps();
+ }
}
break;
}
@@ -1211,6 +1219,7 @@
bool mSeekPending;
bool mFirstAccessUnit;
+ bool mAllTracksHaveTime;
int64_t mNTPAnchorUs;
int64_t mMediaAnchorUs;
int64_t mLastMediaTimeUs;
@@ -1357,6 +1366,7 @@
}
void fakeTimestamps() {
+ mNTPAnchorUs = -1ll;
for (size_t i = 0; i < mTracks.size(); ++i) {
onTimeUpdate(i, 0, 0ll);
}
@@ -1377,6 +1387,21 @@
mNTPAnchorUs = ntpTimeUs;
mMediaAnchorUs = mLastMediaTimeUs;
}
+
+ if (!mAllTracksHaveTime) {
+ bool allTracksHaveTime = true;
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ TrackInfo *track = &mTracks.editItemAt(i);
+ if (track->mNTPAnchorUs < 0) {
+ allTracksHaveTime = false;
+ break;
+ }
+ }
+ if (allTracksHaveTime) {
+ mAllTracksHaveTime = true;
+ ALOGI("Time now established for all tracks.");
+ }
+ }
}
void onAccessUnitComplete(
@@ -1403,7 +1428,7 @@
TrackInfo *track = &mTracks.editItemAt(trackIndex);
- if (mNTPAnchorUs < 0 || mMediaAnchorUs < 0 || track->mNTPAnchorUs < 0) {
+ if (!mAllTracksHaveTime) {
ALOGV("storing accessUnit, no time established yet");
track->mPackets.push_back(accessUnit);
return;
@@ -1460,7 +1485,7 @@
sp<AMessage> msg = mNotify->dup();
msg->setInt32("what", kWhatAccessUnit);
msg->setSize("trackIndex", trackIndex);
- msg->setObject("accessUnit", accessUnit);
+ msg->setBuffer("accessUnit", accessUnit);
msg->post();
}
diff --git a/media/libstagefright/tests/SurfaceMediaSource_test.cpp b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
index 76b507f..3dcd9fc 100644
--- a/media/libstagefright/tests/SurfaceMediaSource_test.cpp
+++ b/media/libstagefright/tests/SurfaceMediaSource_test.cpp
@@ -26,16 +26,16 @@
#include <media/stagefright/SurfaceMediaSource.h>
#include <media/mediarecorder.h>
-#include <gui/SurfaceTextureClient.h>
#include <ui/GraphicBuffer.h>
-#include <surfaceflinger/ISurfaceComposer.h>
-#include <surfaceflinger/Surface.h>
-#include <surfaceflinger/SurfaceComposerClient.h>
+#include <gui/SurfaceTextureClient.h>
+#include <gui/ISurfaceComposer.h>
+#include <gui/Surface.h>
+#include <gui/SurfaceComposerClient.h>
#include <binder/ProcessState.h>
#include <ui/FramebufferNativeWindow.h>
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/MediaBufferGroup.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
@@ -475,7 +475,7 @@
mr->setVideoFrameRate(fps);
mr->prepare();
ALOGV("Starting MediaRecorder...");
- CHECK_EQ(OK, mr->start());
+ CHECK_EQ((status_t)OK, mr->start());
return mr;
}
@@ -757,7 +757,7 @@
ASSERT_EQ(NO_ERROR, native_window_api_disconnect(mANW.get(), NATIVE_WINDOW_API_CPU));
ALOGV("Stopping MediaRecorder...");
- CHECK_EQ(OK, mr->stop());
+ CHECK_EQ((status_t)OK, mr->stop());
mr.clear();
close(fd);
}
@@ -886,7 +886,7 @@
mEglSurface = EGL_NO_SURFACE;
ALOGV("Stopping MediaRecorder...");
- CHECK_EQ(OK, mr->stop());
+ CHECK_EQ((status_t)OK, mr->stop());
mr.clear();
close(fd);
}
@@ -929,7 +929,7 @@
mEglSurface = EGL_NO_SURFACE;
ALOGV("Stopping MediaRecorder...");
- CHECK_EQ(OK, mr->stop());
+ CHECK_EQ((status_t)OK, mr->stop());
mr.clear();
close(fd);
}
diff --git a/media/libstagefright/timedtext/Android.mk b/media/libstagefright/timedtext/Android.mk
index 59d0e15..dde2066 100644
--- a/media/libstagefright/timedtext/Android.mk
+++ b/media/libstagefright/timedtext/Android.mk
@@ -3,14 +3,16 @@
LOCAL_SRC_FILES:= \
TextDescriptions.cpp \
- TimedTextParser.cpp \
+ TimedTextDriver.cpp \
+ TimedText3GPPSource.cpp \
+ TimedTextSource.cpp \
+ TimedTextSRTSource.cpp \
TimedTextPlayer.cpp
LOCAL_CFLAGS += -Wno-multichar
LOCAL_C_INCLUDES:= \
$(JNI_H_INCLUDE) \
- $(TOP)/frameworks/base/media/libstagefright \
- $(TOP)/frameworks/base/include/media/stagefright/openmax
+ $(TOP)/frameworks/base/media/libstagefright
LOCAL_MODULE:= libstagefright_timedtext
diff --git a/media/libstagefright/timedtext/TimedText3GPPSource.cpp b/media/libstagefright/timedtext/TimedText3GPPSource.cpp
new file mode 100644
index 0000000..4a3bfd3
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedText3GPPSource.cpp
@@ -0,0 +1,113 @@
+ /*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "TimedText3GPPSource"
+#include <utils/Log.h>
+
+#include <binder/Parcel.h>
+#include <media/stagefright/foundation/ADebug.h> // CHECK_XX macro
+#include <media/stagefright/MediaBuffer.h>
+#include <media/stagefright/MediaDefs.h> // for MEDIA_MIMETYPE_xxx
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+
+#include "TimedText3GPPSource.h"
+#include "TextDescriptions.h"
+
+namespace android {
+
+TimedText3GPPSource::TimedText3GPPSource(const sp<MediaSource>& mediaSource)
+ : mSource(mediaSource) {
+}
+
+TimedText3GPPSource::~TimedText3GPPSource() {
+}
+
+status_t TimedText3GPPSource::read(
+ int64_t *timeUs, Parcel *parcel, const MediaSource::ReadOptions *options) {
+ MediaBuffer *textBuffer = NULL;
+ status_t err = mSource->read(&textBuffer, options);
+ if (err != OK) {
+ return err;
+ }
+ CHECK(textBuffer != NULL);
+ textBuffer->meta_data()->findInt64(kKeyTime, timeUs);
+ // TODO: this is legacy code. when 'timeUs' can be <= 0?
+ if (*timeUs > 0) {
+ extractAndAppendLocalDescriptions(*timeUs, textBuffer, parcel);
+ }
+ textBuffer->release();
+ return OK;
+}
+
+// Each text sample consists of a string of text, optionally with sample
+// modifier description. The modifier description could specify a new
+// text style for the string of text. These descriptions are present only
+// if they are needed. This method is used to extract the modifier
+// description and append it at the end of the text.
+status_t TimedText3GPPSource::extractAndAppendLocalDescriptions(
+ int64_t timeUs, const MediaBuffer *textBuffer, Parcel *parcel) {
+ const void *data;
+ size_t size = 0;
+ int32_t flag = TextDescriptions::LOCAL_DESCRIPTIONS;
+
+ const char *mime;
+ CHECK(mSource->getFormat()->findCString(kKeyMIMEType, &mime));
+ CHECK(strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) == 0);
+
+ data = textBuffer->data();
+ size = textBuffer->size();
+
+ if (size > 0) {
+ parcel->freeData();
+ flag |= TextDescriptions::IN_BAND_TEXT_3GPP;
+ return TextDescriptions::getParcelOfDescriptions(
+ (const uint8_t *)data, size, flag, timeUs / 1000, parcel);
+ }
+ return OK;
+}
+
+// To extract and send the global text descriptions for all the text samples
+// in the text track or text file.
+// TODO: send error message to application via notifyListener()...?
+status_t TimedText3GPPSource::extractGlobalDescriptions(Parcel *parcel) {
+ const void *data;
+ size_t size = 0;
+ int32_t flag = TextDescriptions::GLOBAL_DESCRIPTIONS;
+
+ const char *mime;
+ CHECK(mSource->getFormat()->findCString(kKeyMIMEType, &mime));
+ CHECK(strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) == 0);
+
+ uint32_t type;
+ // get the 'tx3g' box content. This box contains the text descriptions
+ // used to render the text track
+ if (!mSource->getFormat()->findData(
+ kKeyTextFormatData, &type, &data, &size)) {
+ return ERROR_MALFORMED;
+ }
+
+ if (size > 0) {
+ flag |= TextDescriptions::IN_BAND_TEXT_3GPP;
+ return TextDescriptions::getParcelOfDescriptions(
+ (const uint8_t *)data, size, flag, 0, parcel);
+ }
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedText3GPPSource.h b/media/libstagefright/timedtext/TimedText3GPPSource.h
new file mode 100644
index 0000000..cb7e47c9
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedText3GPPSource.h
@@ -0,0 +1,55 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TIMED_TEXT_3GPP_SOURCE_H_
+#define TIMED_TEXT_3GPP_SOURCE_H_
+
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+
+#include "TimedTextSource.h"
+
+namespace android {
+
+class MediaBuffer;
+class Parcel;
+
+class TimedText3GPPSource : public TimedTextSource {
+ public:
+ TimedText3GPPSource(const sp<MediaSource>& mediaSource);
+ virtual status_t start() { return mSource->start(); }
+ virtual status_t stop() { return mSource->stop(); }
+ virtual status_t read(
+ int64_t *timeUs,
+ Parcel *parcel,
+ const MediaSource::ReadOptions *options = NULL);
+ virtual status_t extractGlobalDescriptions(Parcel *parcel);
+
+ protected:
+ virtual ~TimedText3GPPSource();
+
+ private:
+ sp<MediaSource> mSource;
+
+ status_t extractAndAppendLocalDescriptions(
+ int64_t timeUs, const MediaBuffer *textBuffer, Parcel *parcel);
+
+ DISALLOW_EVIL_CONSTRUCTORS(TimedText3GPPSource);
+};
+
+} // namespace android
+
+#endif // TIMED_TEXT_3GPP_SOURCE_H_
diff --git a/media/libstagefright/timedtext/TimedTextDriver.cpp b/media/libstagefright/timedtext/TimedTextDriver.cpp
new file mode 100644
index 0000000..9ec9415
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextDriver.cpp
@@ -0,0 +1,223 @@
+ /*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "TimedTextDriver"
+#include <utils/Log.h>
+
+#include <binder/IPCThreadState.h>
+
+#include <media/MediaPlayerInterface.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/Utils.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/ALooper.h>
+
+#include "TimedTextDriver.h"
+
+#include "TextDescriptions.h"
+#include "TimedTextPlayer.h"
+#include "TimedTextSource.h"
+
+namespace android {
+
+TimedTextDriver::TimedTextDriver(
+ const wp<MediaPlayerBase> &listener)
+ : mLooper(new ALooper),
+ mListener(listener),
+ mState(UNINITIALIZED) {
+ mLooper->setName("TimedTextDriver");
+ mLooper->start();
+ mPlayer = new TimedTextPlayer(listener);
+ mLooper->registerHandler(mPlayer);
+}
+
+TimedTextDriver::~TimedTextDriver() {
+ mTextInBandVector.clear();
+ mTextOutOfBandVector.clear();
+ mLooper->stop();
+}
+
+status_t TimedTextDriver::setTimedTextTrackIndex_l(int32_t index) {
+ if (index >=
+ (int)(mTextInBandVector.size() + mTextOutOfBandVector.size())) {
+ return BAD_VALUE;
+ }
+
+ sp<TimedTextSource> source;
+ if (index < mTextInBandVector.size()) {
+ source = mTextInBandVector.itemAt(index);
+ } else {
+ source = mTextOutOfBandVector.itemAt(index - mTextInBandVector.size());
+ }
+ mPlayer->setDataSource(source);
+ return OK;
+}
+
+status_t TimedTextDriver::start() {
+ Mutex::Autolock autoLock(mLock);
+ switch (mState) {
+ case UNINITIALIZED:
+ return INVALID_OPERATION;
+ case STOPPED:
+ mPlayer->start();
+ break;
+ case PLAYING:
+ return OK;
+ case PAUSED:
+ mPlayer->resume();
+ break;
+ default:
+ TRESPASS();
+ }
+ mState = PLAYING;
+ return OK;
+}
+
+status_t TimedTextDriver::stop() {
+ return pause();
+}
+
+// TODO: Test if pause() works properly.
+// Scenario 1: start - pause - resume
+// Scenario 2: start - seek
+// Scenario 3: start - pause - seek - resume
+status_t TimedTextDriver::pause() {
+ Mutex::Autolock autoLock(mLock);
+ switch (mState) {
+ case UNINITIALIZED:
+ return INVALID_OPERATION;
+ case STOPPED:
+ return OK;
+ case PLAYING:
+ mPlayer->pause();
+ break;
+ case PAUSED:
+ return OK;
+ default:
+ TRESPASS();
+ }
+ mState = PAUSED;
+ return OK;
+}
+
+status_t TimedTextDriver::resume() {
+ return start();
+}
+
+status_t TimedTextDriver::seekToAsync(int64_t timeUs) {
+ mPlayer->seekToAsync(timeUs);
+ return OK;
+}
+
+status_t TimedTextDriver::setTimedTextTrackIndex(int32_t index) {
+ // TODO: This is current implementation for MediaPlayer::disableTimedText().
+ // Find better way for readability.
+ if (index < 0) {
+ mPlayer->pause();
+ return OK;
+ }
+
+ status_t ret = OK;
+ Mutex::Autolock autoLock(mLock);
+ switch (mState) {
+ case UNINITIALIZED:
+ ret = INVALID_OPERATION;
+ break;
+ case PAUSED:
+ ret = setTimedTextTrackIndex_l(index);
+ break;
+ case PLAYING:
+ mPlayer->pause();
+ ret = setTimedTextTrackIndex_l(index);
+ if (ret != OK) {
+ break;
+ }
+ mPlayer->start();
+ break;
+ case STOPPED:
+ // TODO: The only difference between STOPPED and PAUSED is this
+ // part. Revise the flow from "MediaPlayer::enableTimedText()" and
+ // remove one of the status, PAUSED and STOPPED, if possible.
+ ret = setTimedTextTrackIndex_l(index);
+ if (ret != OK) {
+ break;
+ }
+ mPlayer->start();
+ break;
+ defaut:
+ TRESPASS();
+ }
+ return ret;
+}
+
+status_t TimedTextDriver::addInBandTextSource(
+ const sp<MediaSource>& mediaSource) {
+ sp<TimedTextSource> source =
+ TimedTextSource::CreateTimedTextSource(mediaSource);
+ if (source == NULL) {
+ return ERROR_UNSUPPORTED;
+ }
+ Mutex::Autolock autoLock(mLock);
+ mTextInBandVector.add(source);
+ if (mState == UNINITIALIZED) {
+ mState = STOPPED;
+ }
+ return OK;
+}
+
+status_t TimedTextDriver::addOutOfBandTextSource(
+ const Parcel &request) {
+ // TODO: Define "TimedTextSource::CreateFromURI(uri)"
+ // and move below lines there..?
+
+ // String values written in Parcel are UTF-16 values.
+ const String16 uri16 = request.readString16();
+ String8 uri = String8(request.readString16());
+
+ uri.toLower();
+ // To support local subtitle file only for now
+ if (strncasecmp("file://", uri.string(), 7)) {
+ return ERROR_UNSUPPORTED;
+ }
+ sp<DataSource> dataSource =
+ DataSource::CreateFromURI(uri);
+ if (dataSource == NULL) {
+ return ERROR_UNSUPPORTED;
+ }
+
+ sp<TimedTextSource> source;
+ if (uri.getPathExtension() == String8(".srt")) {
+ source = TimedTextSource::CreateTimedTextSource(
+ dataSource, TimedTextSource::OUT_OF_BAND_FILE_SRT);
+ }
+
+ if (source == NULL) {
+ return ERROR_UNSUPPORTED;
+ }
+
+ Mutex::Autolock autoLock(mLock);
+
+ mTextOutOfBandVector.add(source);
+ if (mState == UNINITIALIZED) {
+ mState = STOPPED;
+ }
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextDriver.h b/media/libstagefright/timedtext/TimedTextDriver.h
new file mode 100644
index 0000000..efedb6e
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextDriver.h
@@ -0,0 +1,81 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TIMED_TEXT_DRIVER_H_
+#define TIMED_TEXT_DRIVER_H_
+
+#include <media/stagefright/foundation/ABase.h> // for DISALLOW_* macro
+#include <utils/Errors.h> // for status_t
+#include <utils/RefBase.h>
+#include <utils/threads.h>
+
+namespace android {
+
+class ALooper;
+class MediaPlayerBase;
+class MediaSource;
+class Parcel;
+class TimedTextPlayer;
+class TimedTextSource;
+
+class TimedTextDriver {
+public:
+ TimedTextDriver(const wp<MediaPlayerBase> &listener);
+
+ ~TimedTextDriver();
+
+ // TODO: pause-resume pair seems equivalent to stop-start pair.
+ // Check if it is replaceable with stop-start.
+ status_t start();
+ status_t stop();
+ status_t pause();
+ status_t resume();
+
+ status_t seekToAsync(int64_t timeUs);
+
+ status_t addInBandTextSource(const sp<MediaSource>& source);
+ status_t addOutOfBandTextSource(const Parcel &request);
+
+ status_t setTimedTextTrackIndex(int32_t index);
+
+private:
+ Mutex mLock;
+
+ enum State {
+ UNINITIALIZED,
+ STOPPED,
+ PLAYING,
+ PAUSED,
+ };
+
+ sp<ALooper> mLooper;
+ sp<TimedTextPlayer> mPlayer;
+ wp<MediaPlayerBase> mListener;
+
+ // Variables to be guarded by mLock.
+ State mState;
+ Vector<sp<TimedTextSource> > mTextInBandVector;
+ Vector<sp<TimedTextSource> > mTextOutOfBandVector;
+ // -- End of variables to be guarded by mLock
+
+ status_t setTimedTextTrackIndex_l(int32_t index);
+
+ DISALLOW_EVIL_CONSTRUCTORS(TimedTextDriver);
+};
+
+} // namespace android
+
+#endif // TIMED_TEXT_DRIVER_H_
diff --git a/media/libstagefright/timedtext/TimedTextParser.cpp b/media/libstagefright/timedtext/TimedTextParser.cpp
deleted file mode 100644
index 0bada16..0000000
--- a/media/libstagefright/timedtext/TimedTextParser.cpp
+++ /dev/null
@@ -1,257 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include "TimedTextParser.h"
-#include <media/stagefright/DataSource.h>
-
-namespace android {
-
-TimedTextParser::TimedTextParser()
- : mDataSource(NULL),
- mOffset(0),
- mIndex(0) {
-}
-
-TimedTextParser::~TimedTextParser() {
- reset();
-}
-
-status_t TimedTextParser::init(
- const sp<DataSource> &dataSource, FileType fileType) {
- mDataSource = dataSource;
- mFileType = fileType;
-
- status_t err;
- if ((err = scanFile()) != OK) {
- reset();
- return err;
- }
-
- return OK;
-}
-
-void TimedTextParser::reset() {
- mDataSource.clear();
- mTextVector.clear();
- mOffset = 0;
- mIndex = 0;
-}
-
-// scan the text file to get start/stop time and the
-// offset of each piece of text content
-status_t TimedTextParser::scanFile() {
- if (mFileType != OUT_OF_BAND_FILE_SRT) {
- return ERROR_UNSUPPORTED;
- }
-
- off64_t offset = 0;
- int64_t startTimeUs;
- bool endOfFile = false;
-
- while (!endOfFile) {
- TextInfo info;
- status_t err = getNextInSrtFileFormat(&offset, &startTimeUs, &info);
-
- if (err != OK) {
- if (err == ERROR_END_OF_STREAM) {
- endOfFile = true;
- } else {
- return err;
- }
- } else {
- mTextVector.add(startTimeUs, info);
- }
- }
-
- if (mTextVector.isEmpty()) {
- return ERROR_MALFORMED;
- }
- return OK;
-}
-
-// read one line started from *offset and store it into data.
-status_t TimedTextParser::readNextLine(off64_t *offset, AString *data) {
- char character;
-
- data->clear();
-
- while (true) {
- ssize_t err;
- if ((err = mDataSource->readAt(*offset, &character, 1)) < 1) {
- if (err == 0) {
- return ERROR_END_OF_STREAM;
- }
- return ERROR_IO;
- }
-
- (*offset) ++;
-
- // a line could end with CR, LF or CR + LF
- if (character == 10) {
- break;
- } else if (character == 13) {
- if ((err = mDataSource->readAt(*offset, &character, 1)) < 1) {
- if (err == 0) { // end of the stream
- return OK;
- }
- return ERROR_IO;
- }
-
- (*offset) ++;
-
- if (character != 10) {
- (*offset) --;
- }
- break;
- }
-
- data->append(character);
- }
-
- return OK;
-}
-
-/* SRT format:
- * Subtitle number
- * Start time --> End time
- * Text of subtitle (one or more lines)
- * Blank line
- *
- * .srt file example:
- * 1
- * 00:00:20,000 --> 00:00:24,400
- * Altocumulus clouds occur between six thousand
- *
- * 2
- * 00:00:24,600 --> 00:00:27,800
- * and twenty thousand feet above ground level.
- */
-status_t TimedTextParser::getNextInSrtFileFormat(
- off64_t *offset, int64_t *startTimeUs, TextInfo *info) {
- AString data;
- status_t err;
- if ((err = readNextLine(offset, &data)) != OK) {
- return err;
- }
-
- // to skip the first line
- if ((err = readNextLine(offset, &data)) != OK) {
- return err;
- }
-
- int hour1, hour2, min1, min2, sec1, sec2, msec1, msec2;
- // the start time format is: hours:minutes:seconds,milliseconds
- // 00:00:24,600 --> 00:00:27,800
- if (sscanf(data.c_str(), "%02d:%02d:%02d,%03d --> %02d:%02d:%02d,%03d",
- &hour1, &min1, &sec1, &msec1, &hour2, &min2, &sec2, &msec2) != 8) {
- return ERROR_MALFORMED;
- }
-
- *startTimeUs = ((hour1 * 3600 + min1 * 60 + sec1) * 1000 + msec1) * 1000ll;
- info->endTimeUs = ((hour2 * 3600 + min2 * 60 + sec2) * 1000 + msec2) * 1000ll;
- if (info->endTimeUs <= *startTimeUs) {
- return ERROR_MALFORMED;
- }
-
- info->offset = *offset;
-
- bool needMoreData = true;
- while (needMoreData) {
- if ((err = readNextLine(offset, &data)) != OK) {
- if (err == ERROR_END_OF_STREAM) {
- needMoreData = false;
- } else {
- return err;
- }
- }
-
- if (needMoreData) {
- data.trim();
- if (data.empty()) {
- // it's an empty line used to separate two subtitles
- needMoreData = false;
- }
- }
- }
-
- info->textLen = *offset - info->offset;
-
- return OK;
-}
-
-status_t TimedTextParser::getText(
- AString *text, int64_t *startTimeUs, int64_t *endTimeUs,
- const MediaSource::ReadOptions *options) {
- Mutex::Autolock autoLock(mLock);
-
- text->clear();
-
- int64_t seekTimeUs;
- MediaSource::ReadOptions::SeekMode mode;
- if (options && options->getSeekTo(&seekTimeUs, &mode)) {
- int64_t lastEndTimeUs = mTextVector.valueAt(mTextVector.size() - 1).endTimeUs;
- int64_t firstStartTimeUs = mTextVector.keyAt(0);
-
- if (seekTimeUs < 0 || seekTimeUs > lastEndTimeUs) {
- return ERROR_OUT_OF_RANGE;
- } else if (seekTimeUs < firstStartTimeUs) {
- mIndex = 0;
- } else {
- // binary search
- ssize_t low = 0;
- ssize_t high = mTextVector.size() - 1;
- ssize_t mid = 0;
- int64_t currTimeUs;
-
- while (low <= high) {
- mid = low + (high - low)/2;
- currTimeUs = mTextVector.keyAt(mid);
- const int diff = currTimeUs - seekTimeUs;
-
- if (diff == 0) {
- break;
- } else if (diff < 0) {
- low = mid + 1;
- } else {
- if ((high == mid + 1)
- && (seekTimeUs < mTextVector.keyAt(high))) {
- break;
- }
- high = mid - 1;
- }
- }
-
- mIndex = mid;
- }
- }
-
- TextInfo info = mTextVector.valueAt(mIndex);
- *startTimeUs = mTextVector.keyAt(mIndex);
- *endTimeUs = info.endTimeUs;
- mIndex ++;
-
- char *str = new char[info.textLen];
- if (mDataSource->readAt(info.offset, str, info.textLen) < info.textLen) {
- delete[] str;
- return ERROR_IO;
- }
-
- text->append(str, info.textLen);
- delete[] str;
- return OK;
-}
-
-} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextParser.h b/media/libstagefright/timedtext/TimedTextParser.h
deleted file mode 100644
index 44774c2..0000000
--- a/media/libstagefright/timedtext/TimedTextParser.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (C) 2011 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef TIMED_TEXT_PARSER_H_
-
-#define TIMED_TEXT_PARSER_H_
-
-#include <media/MediaPlayerInterface.h>
-#include <media/stagefright/foundation/ABase.h>
-#include <media/stagefright/foundation/AString.h>
-#include <media/stagefright/MediaSource.h>
-
-namespace android {
-
-class DataSource;
-
-class TimedTextParser : public RefBase {
-public:
- TimedTextParser();
- virtual ~TimedTextParser();
-
- enum FileType {
- OUT_OF_BAND_FILE_SRT = 1,
- };
-
- status_t getText(AString *text, int64_t *startTimeUs, int64_t *endTimeUs,
- const MediaSource::ReadOptions *options = NULL);
- status_t init(const sp<DataSource> &dataSource, FileType fileType);
- void reset();
-
-private:
- Mutex mLock;
-
- sp<DataSource> mDataSource;
- off64_t mOffset;
-
- struct TextInfo {
- int64_t endTimeUs;
- // the offset of the text in the original file
- off64_t offset;
- int textLen;
- };
-
- int mIndex;
- FileType mFileType;
-
- // the key indicated the start time of the text
- KeyedVector<int64_t, TextInfo> mTextVector;
-
- status_t getNextInSrtFileFormat(
- off64_t *offset, int64_t *startTimeUs, TextInfo *info);
- status_t readNextLine(off64_t *offset, AString *data);
-
- status_t scanFile();
-
- DISALLOW_EVIL_CONSTRUCTORS(TimedTextParser);
-};
-
-} // namespace android
-
-#endif // TIMED_TEXT_PARSER_H_
-
diff --git a/media/libstagefright/timedtext/TimedTextPlayer.cpp b/media/libstagefright/timedtext/TimedTextPlayer.cpp
index 3014b0b..bf7cbf6 100644
--- a/media/libstagefright/timedtext/TimedTextPlayer.cpp
+++ b/media/libstagefright/timedtext/TimedTextPlayer.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2011 The Android Open Source Project
+ * Copyright (C) 2012 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -18,399 +18,164 @@
#define LOG_TAG "TimedTextPlayer"
#include <utils/Log.h>
-#include <binder/IPCThreadState.h>
-
-#include <media/stagefright/MediaDebug.h>
-#include <media/stagefright/MediaDefs.h>
+#include <media/stagefright/foundation/ADebug.h>
+#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/MediaErrors.h>
-#include <media/stagefright/MediaSource.h>
-#include <media/stagefright/MetaData.h>
-#include <media/stagefright/MediaBuffer.h>
-#include <media/stagefright/FileSource.h>
-#include <media/stagefright/Utils.h>
+#include <media/MediaPlayerInterface.h>
-#include "include/AwesomePlayer.h"
#include "TimedTextPlayer.h"
-#include "TimedTextParser.h"
-#include "TextDescriptions.h"
+
+#include "TimedTextDriver.h"
+#include "TimedTextSource.h"
namespace android {
-struct TimedTextEvent : public TimedEventQueue::Event {
- TimedTextEvent(
- TimedTextPlayer *player,
- void (TimedTextPlayer::*method)())
- : mPlayer(player),
- mMethod(method) {
- }
+static const int64_t kAdjustmentProcessingTimeUs = 100000ll;
-protected:
- virtual ~TimedTextEvent() {}
-
- virtual void fire(TimedEventQueue *queue, int64_t /* now_us */) {
- (mPlayer->*mMethod)();
- }
-
-private:
- TimedTextPlayer *mPlayer;
- void (TimedTextPlayer::*mMethod)();
-
- TimedTextEvent(const TimedTextEvent &);
- TimedTextEvent &operator=(const TimedTextEvent &);
-};
-
-TimedTextPlayer::TimedTextPlayer(
- AwesomePlayer *observer,
- const wp<MediaPlayerBase> &listener,
- TimedEventQueue *queue)
- : mSource(NULL),
- mOutOfBandSource(NULL),
- mSeekTimeUs(0),
- mStarted(false),
- mTextEventPending(false),
- mQueue(queue),
- mListener(listener),
- mObserver(observer),
- mTextBuffer(NULL),
- mTextParser(NULL),
- mTextType(kNoText) {
- mTextEvent = new TimedTextEvent(this, &TimedTextPlayer::onTextEvent);
+TimedTextPlayer::TimedTextPlayer(const wp<MediaPlayerBase> &listener)
+ : mListener(listener),
+ mSource(NULL),
+ mSendSubtitleGeneration(0) {
}
TimedTextPlayer::~TimedTextPlayer() {
- if (mStarted) {
- reset();
+ if (mSource != NULL) {
+ mSource->stop();
+ mSource.clear();
+ mSource = NULL;
}
-
- mTextTrackVector.clear();
- mTextOutOfBandVector.clear();
}
-status_t TimedTextPlayer::start(uint8_t index) {
- CHECK(!mStarted);
-
- if (index >=
- mTextTrackVector.size() + mTextOutOfBandVector.size()) {
- ALOGE("Incorrect text track index: %d", index);
- return BAD_VALUE;
- }
-
- status_t err;
- if (index < mTextTrackVector.size()) { // start an in-band text
- mSource = mTextTrackVector.itemAt(index);
-
- err = mSource->start();
-
- if (err != OK) {
- return err;
- }
- mTextType = kInBandText;
- } else { // start an out-of-band text
- OutOfBandText text =
- mTextOutOfBandVector.itemAt(index - mTextTrackVector.size());
-
- mOutOfBandSource = text.source;
- TimedTextParser::FileType fileType = text.type;
-
- if (mTextParser == NULL) {
- mTextParser = new TimedTextParser();
- }
-
- if ((err = mTextParser->init(mOutOfBandSource, fileType)) != OK) {
- return err;
- }
- mTextType = kOutOfBandText;
- }
-
- // send sample description format
- if ((err = extractAndSendGlobalDescriptions()) != OK) {
- return err;
- }
-
- int64_t positionUs;
- mObserver->getPosition(&positionUs);
- seekTo(positionUs);
-
- postTextEvent();
-
- mStarted = true;
-
- return OK;
+void TimedTextPlayer::start() {
+ sp<AMessage> msg = new AMessage(kWhatSeek, id());
+ msg->setInt64("seekTimeUs", -1);
+ msg->post();
}
void TimedTextPlayer::pause() {
- CHECK(mStarted);
-
- cancelTextEvent();
+ (new AMessage(kWhatPause, id()))->post();
}
void TimedTextPlayer::resume() {
- CHECK(mStarted);
-
- postTextEvent();
+ start();
}
-void TimedTextPlayer::reset() {
- CHECK(mStarted);
+void TimedTextPlayer::seekToAsync(int64_t timeUs) {
+ sp<AMessage> msg = new AMessage(kWhatSeek, id());
+ msg->setInt64("seekTimeUs", timeUs);
+ msg->post();
+}
- // send an empty text to clear the screen
- notifyListener(MEDIA_TIMED_TEXT);
+void TimedTextPlayer::setDataSource(sp<TimedTextSource> source) {
+ sp<AMessage> msg = new AMessage(kWhatSetSource, id());
+ msg->setObject("source", source);
+ msg->post();
+}
- cancelTextEvent();
-
- mSeeking = false;
- mStarted = false;
-
- if (mTextType == kInBandText) {
- if (mTextBuffer != NULL) {
- mTextBuffer->release();
- mTextBuffer = NULL;
+void TimedTextPlayer::onMessageReceived(const sp<AMessage> &msg) {
+ switch (msg->what()) {
+ case kWhatPause: {
+ mSendSubtitleGeneration++;
+ break;
}
-
- if (mSource != NULL) {
- mSource->stop();
- mSource.clear();
- mSource = NULL;
+ case kWhatSeek: {
+ int64_t seekTimeUs = 0;
+ msg->findInt64("seekTimeUs", &seekTimeUs);
+ if (seekTimeUs < 0) {
+ sp<MediaPlayerBase> listener = mListener.promote();
+ if (listener != NULL) {
+ int32_t positionMs = 0;
+ listener->getCurrentPosition(&positionMs);
+ seekTimeUs = positionMs * 1000ll;
+ }
+ }
+ doSeekAndRead(seekTimeUs);
+ break;
}
- } else {
- if (mTextParser != NULL) {
- mTextParser.clear();
- mTextParser = NULL;
+ case kWhatSendSubtitle: {
+ int32_t generation;
+ CHECK(msg->findInt32("generation", &generation));
+ if (generation != mSendSubtitleGeneration) {
+ // Drop obsolete msg.
+ break;
+ }
+ sp<RefBase> obj;
+ msg->findObject("subtitle", &obj);
+ if (obj != NULL) {
+ sp<ParcelEvent> parcelEvent;
+ parcelEvent = static_cast<ParcelEvent*>(obj.get());
+ notifyListener(MEDIA_TIMED_TEXT, &(parcelEvent->parcel));
+ } else {
+ notifyListener(MEDIA_TIMED_TEXT);
+ }
+ doRead();
+ break;
}
- if (mOutOfBandSource != NULL) {
- mOutOfBandSource.clear();
- mOutOfBandSource = NULL;
+ case kWhatSetSource: {
+ sp<RefBase> obj;
+ msg->findObject("source", &obj);
+ if (obj == NULL) break;
+ if (mSource != NULL) {
+ mSource->stop();
+ }
+ mSource = static_cast<TimedTextSource*>(obj.get());
+ mSource->start();
+ Parcel parcel;
+ if (mSource->extractGlobalDescriptions(&parcel) == OK &&
+ parcel.dataSize() > 0) {
+ notifyListener(MEDIA_TIMED_TEXT, &parcel);
+ } else {
+ notifyListener(MEDIA_TIMED_TEXT);
+ }
+ break;
}
}
}
-status_t TimedTextPlayer::seekTo(int64_t time_us) {
- Mutex::Autolock autoLock(mLock);
-
- mSeeking = true;
- mSeekTimeUs = time_us;
-
- postTextEvent();
-
- return OK;
-}
-
-status_t TimedTextPlayer::setTimedTextTrackIndex(int32_t index) {
- if (index >=
- (int)(mTextTrackVector.size() + mTextOutOfBandVector.size())) {
- return BAD_VALUE;
- }
-
- if (mStarted) {
- reset();
- }
-
- if (index >= 0) {
- return start(index);
- }
- return OK;
-}
-
-void TimedTextPlayer::onTextEvent() {
- Mutex::Autolock autoLock(mLock);
-
- if (!mTextEventPending) {
- return;
- }
- mTextEventPending = false;
-
- if (mData.dataSize() > 0) {
- notifyListener(MEDIA_TIMED_TEXT, &mData);
- mData.freeData();
- }
-
+void TimedTextPlayer::doSeekAndRead(int64_t seekTimeUs) {
MediaSource::ReadOptions options;
- if (mSeeking) {
- options.setSeekTo(mSeekTimeUs,
- MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC);
- mSeeking = false;
-
- notifyListener(MEDIA_TIMED_TEXT); //empty text to clear the screen
- }
-
- int64_t positionUs, timeUs;
- mObserver->getPosition(&positionUs);
-
- if (mTextType == kInBandText) {
- if (mSource->read(&mTextBuffer, &options) != OK) {
- return;
- }
-
- mTextBuffer->meta_data()->findInt64(kKeyTime, &timeUs);
- } else {
- int64_t endTimeUs;
- if (mTextParser->getText(
- &mText, &timeUs, &endTimeUs, &options) != OK) {
- return;
- }
- }
-
- if (timeUs > 0) {
- extractAndAppendLocalDescriptions(timeUs);
- }
-
- if (mTextType == kInBandText) {
- if (mTextBuffer != NULL) {
- mTextBuffer->release();
- mTextBuffer = NULL;
- }
- } else {
- mText.clear();
- }
-
- //send the text now
- if (timeUs <= positionUs + 100000ll) {
- postTextEvent();
- } else {
- postTextEvent(timeUs - positionUs - 100000ll);
- }
+ options.setSeekTo(seekTimeUs, MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC);
+ doRead(&options);
}
-void TimedTextPlayer::postTextEvent(int64_t delayUs) {
- if (mTextEventPending) {
- return;
- }
-
- mTextEventPending = true;
- mQueue->postEventWithDelay(mTextEvent, delayUs < 0 ? 10000 : delayUs);
+void TimedTextPlayer::doRead(MediaSource::ReadOptions* options) {
+ int64_t timeUs = 0;
+ sp<ParcelEvent> parcelEvent = new ParcelEvent();
+ mSource->read(&timeUs, &(parcelEvent->parcel), options);
+ postTextEvent(parcelEvent, timeUs);
}
-void TimedTextPlayer::cancelTextEvent() {
- mQueue->cancelEvent(mTextEvent->eventID());
- mTextEventPending = false;
-}
+void TimedTextPlayer::postTextEvent(const sp<ParcelEvent>& parcel, int64_t timeUs) {
+ sp<MediaPlayerBase> listener = mListener.promote();
+ if (listener != NULL) {
+ int64_t positionUs, delayUs;
+ int32_t positionMs = 0;
+ listener->getCurrentPosition(&positionMs);
+ positionUs = positionMs * 1000;
-void TimedTextPlayer::addTextSource(sp<MediaSource> source) {
- Mutex::Autolock autoLock(mLock);
- mTextTrackVector.add(source);
-}
-
-status_t TimedTextPlayer::setParameter(int key, const Parcel &request) {
- Mutex::Autolock autoLock(mLock);
-
- if (key == KEY_PARAMETER_TIMED_TEXT_ADD_OUT_OF_BAND_SOURCE) {
- const String16 uri16 = request.readString16();
- String8 uri = String8(uri16);
- KeyedVector<String8, String8> headers;
-
- // To support local subtitle file only for now
- if (strncasecmp("file://", uri.string(), 7)) {
- return INVALID_OPERATION;
- }
- sp<DataSource> dataSource =
- DataSource::CreateFromURI(uri, &headers);
- status_t err = dataSource->initCheck();
-
- if (err != OK) {
- return err;
- }
-
- OutOfBandText text;
- text.source = dataSource;
- if (uri.getPathExtension() == String8(".srt")) {
- text.type = TimedTextParser::OUT_OF_BAND_FILE_SRT;
+ if (timeUs <= positionUs + kAdjustmentProcessingTimeUs) {
+ delayUs = 0;
} else {
- return ERROR_UNSUPPORTED;
+ delayUs = timeUs - positionUs - kAdjustmentProcessingTimeUs;
}
-
- mTextOutOfBandVector.add(text);
-
- return OK;
+ sp<AMessage> msg = new AMessage(kWhatSendSubtitle, id());
+ msg->setInt32("generation", mSendSubtitleGeneration);
+ if (parcel != NULL) {
+ msg->setObject("subtitle", parcel);
+ }
+ msg->post(delayUs);
}
- return INVALID_OPERATION;
}
void TimedTextPlayer::notifyListener(int msg, const Parcel *parcel) {
- if (mListener != NULL) {
- sp<MediaPlayerBase> listener = mListener.promote();
-
- if (listener != NULL) {
- if (parcel && (parcel->dataSize() > 0)) {
- listener->sendEvent(msg, 0, 0, parcel);
- } else { // send an empty timed text to clear the screen
- listener->sendEvent(msg);
- }
+ sp<MediaPlayerBase> listener = mListener.promote();
+ if (listener != NULL) {
+ if (parcel != NULL && (parcel->dataSize() > 0)) {
+ listener->sendEvent(msg, 0, 0, parcel);
+ } else { // send an empty timed text to clear the screen
+ listener->sendEvent(msg);
}
}
}
-// Each text sample consists of a string of text, optionally with sample
-// modifier description. The modifier description could specify a new
-// text style for the string of text. These descriptions are present only
-// if they are needed. This method is used to extract the modifier
-// description and append it at the end of the text.
-status_t TimedTextPlayer::extractAndAppendLocalDescriptions(int64_t timeUs) {
- const void *data;
- size_t size = 0;
- int32_t flag = TextDescriptions::LOCAL_DESCRIPTIONS;
-
- if (mTextType == kInBandText) {
- const char *mime;
- CHECK(mSource->getFormat()->findCString(kKeyMIMEType, &mime));
-
- if (!strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP)) {
- flag |= TextDescriptions::IN_BAND_TEXT_3GPP;
- data = mTextBuffer->data();
- size = mTextBuffer->size();
- } else {
- // support 3GPP only for now
- return ERROR_UNSUPPORTED;
- }
- } else {
- data = mText.c_str();
- size = mText.size();
- flag |= TextDescriptions::OUT_OF_BAND_TEXT_SRT;
- }
-
- if ((size > 0) && (flag != TextDescriptions::LOCAL_DESCRIPTIONS)) {
- mData.freeData();
- return TextDescriptions::getParcelOfDescriptions(
- (const uint8_t *)data, size, flag, timeUs / 1000, &mData);
- }
-
- return OK;
-}
-
-// To extract and send the global text descriptions for all the text samples
-// in the text track or text file.
-status_t TimedTextPlayer::extractAndSendGlobalDescriptions() {
- const void *data;
- size_t size = 0;
- int32_t flag = TextDescriptions::GLOBAL_DESCRIPTIONS;
-
- if (mTextType == kInBandText) {
- const char *mime;
- CHECK(mSource->getFormat()->findCString(kKeyMIMEType, &mime));
-
- // support 3GPP only for now
- if (!strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP)) {
- uint32_t type;
- // get the 'tx3g' box content. This box contains the text descriptions
- // used to render the text track
- if (!mSource->getFormat()->findData(
- kKeyTextFormatData, &type, &data, &size)) {
- return ERROR_MALFORMED;
- }
-
- flag |= TextDescriptions::IN_BAND_TEXT_3GPP;
- }
- }
-
- if ((size > 0) && (flag != TextDescriptions::GLOBAL_DESCRIPTIONS)) {
- Parcel parcel;
- if (TextDescriptions::getParcelOfDescriptions(
- (const uint8_t *)data, size, flag, 0, &parcel) == OK) {
- if (parcel.dataSize() > 0) {
- notifyListener(MEDIA_TIMED_TEXT, &parcel);
- }
- }
- }
-
- return OK;
-}
-}
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextPlayer.h b/media/libstagefright/timedtext/TimedTextPlayer.h
index a744db5..837beeb 100644
--- a/media/libstagefright/timedtext/TimedTextPlayer.h
+++ b/media/libstagefright/timedtext/TimedTextPlayer.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2011 The Android Open Source Project
+ * Copyright (C) 2012 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -15,99 +15,61 @@
*/
#ifndef TIMEDTEXT_PLAYER_H_
-
#define TIMEDTEXT_PLAYER_H_
-#include <media/MediaPlayerInterface.h>
+#include <binder/Parcel.h>
#include <media/stagefright/foundation/ABase.h>
-#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/foundation/AHandler.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/RefBase.h>
-#include "include/TimedEventQueue.h"
-#include "TimedTextParser.h"
+#include "TimedTextSource.h"
namespace android {
-class MediaSource;
-class AwesomePlayer;
-class MediaBuffer;
+class AMessage;
+class MediaPlayerBase;
+class TimedTextDriver;
+class TimedTextSource;
-class TimedTextPlayer {
+class TimedTextPlayer : public AHandler {
public:
- TimedTextPlayer(AwesomePlayer *observer,
- const wp<MediaPlayerBase> &listener,
- TimedEventQueue *queue);
+ TimedTextPlayer(const wp<MediaPlayerBase> &listener);
virtual ~TimedTextPlayer();
- // index: the index of the text track which will
- // be turned on
- status_t start(uint8_t index);
-
+ void start();
void pause();
-
void resume();
+ void seekToAsync(int64_t timeUs);
+ void setDataSource(sp<TimedTextSource> source);
- status_t seekTo(int64_t time_us);
-
- void addTextSource(sp<MediaSource> source);
-
- status_t setTimedTextTrackIndex(int32_t index);
- status_t setParameter(int key, const Parcel &request);
+protected:
+ virtual void onMessageReceived(const sp<AMessage> &msg);
private:
- enum TextType {
- kNoText = 0,
- kInBandText = 1,
- kOutOfBandText = 2,
+ enum {
+ kWhatPause = 'paus',
+ kWhatSeek = 'seek',
+ kWhatSendSubtitle = 'send',
+ kWhatSetSource = 'ssrc',
};
- Mutex mLock;
-
- sp<MediaSource> mSource;
- sp<DataSource> mOutOfBandSource;
-
- bool mSeeking;
- int64_t mSeekTimeUs;
-
- bool mStarted;
-
- sp<TimedEventQueue::Event> mTextEvent;
- bool mTextEventPending;
-
- TimedEventQueue *mQueue;
+ // To add Parcel into an AMessage as an object, it should be 'RefBase'.
+ struct ParcelEvent : public RefBase {
+ Parcel parcel;
+ };
wp<MediaPlayerBase> mListener;
- AwesomePlayer *mObserver;
+ sp<TimedTextSource> mSource;
+ int32_t mSendSubtitleGeneration;
- MediaBuffer *mTextBuffer;
- Parcel mData;
-
- // for in-band timed text
- Vector<sp<MediaSource> > mTextTrackVector;
-
- // for out-of-band timed text
- struct OutOfBandText {
- TimedTextParser::FileType type;
- sp<DataSource> source;
- };
- Vector<OutOfBandText > mTextOutOfBandVector;
-
- sp<TimedTextParser> mTextParser;
- AString mText;
-
- TextType mTextType;
-
- void reset();
-
+ void doSeekAndRead(int64_t seekTimeUs);
+ void doRead(MediaSource::ReadOptions* options = NULL);
void onTextEvent();
- void postTextEvent(int64_t delayUs = -1);
- void cancelTextEvent();
-
+ void postTextEvent(const sp<ParcelEvent>& parcel = NULL, int64_t timeUs = -1);
void notifyListener(int msg, const Parcel *parcel = NULL);
- status_t extractAndAppendLocalDescriptions(int64_t timeUs);
- status_t extractAndSendGlobalDescriptions();
-
DISALLOW_EVIL_CONSTRUCTORS(TimedTextPlayer);
};
diff --git a/media/libstagefright/timedtext/TimedTextSRTSource.cpp b/media/libstagefright/timedtext/TimedTextSRTSource.cpp
new file mode 100644
index 0000000..3752d34
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextSRTSource.cpp
@@ -0,0 +1,275 @@
+ /*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "TimedTextSRTSource"
+#include <utils/Log.h>
+
+#include <binder/Parcel.h>
+#include <media/stagefright/foundation/AString.h>
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+
+#include "TimedTextSRTSource.h"
+#include "TextDescriptions.h"
+
+namespace android {
+
+TimedTextSRTSource::TimedTextSRTSource(const sp<DataSource>& dataSource)
+ : mSource(dataSource),
+ mIndex(0) {
+}
+
+TimedTextSRTSource::~TimedTextSRTSource() {
+}
+
+status_t TimedTextSRTSource::start() {
+ status_t err = scanFile();
+ if (err != OK) {
+ reset();
+ }
+ return err;
+}
+
+void TimedTextSRTSource::reset() {
+ mTextVector.clear();
+ mIndex = 0;
+}
+
+status_t TimedTextSRTSource::stop() {
+ reset();
+ return OK;
+}
+
+status_t TimedTextSRTSource::read(
+ int64_t *timeUs,
+ Parcel *parcel,
+ const MediaSource::ReadOptions *options) {
+ int64_t endTimeUs;
+ AString text;
+ status_t err = getText(options, &text, timeUs, &endTimeUs);
+ if (err != OK) {
+ return err;
+ }
+
+ if (*timeUs > 0) {
+ extractAndAppendLocalDescriptions(*timeUs, text, parcel);
+ }
+ return OK;
+}
+
+status_t TimedTextSRTSource::scanFile() {
+ off64_t offset = 0;
+ int64_t startTimeUs;
+ bool endOfFile = false;
+
+ while (!endOfFile) {
+ TextInfo info;
+ status_t err = getNextSubtitleInfo(&offset, &startTimeUs, &info);
+ switch (err) {
+ case OK:
+ mTextVector.add(startTimeUs, info);
+ break;
+ case ERROR_END_OF_STREAM:
+ endOfFile = true;
+ break;
+ default:
+ return err;
+ }
+ }
+ if (mTextVector.isEmpty()) {
+ return ERROR_MALFORMED;
+ }
+ return OK;
+}
+
+/* SRT format:
+ * Subtitle number
+ * Start time --> End time
+ * Text of subtitle (one or more lines)
+ * Blank lines
+ *
+ * .srt file example:
+ * 1
+ * 00:00:20,000 --> 00:00:24,400
+ * Altocumulus clouds occr between six thousand
+ *
+ * 2
+ * 00:00:24,600 --> 00:00:27,800
+ * and twenty thousand feet above ground level.
+ */
+status_t TimedTextSRTSource::getNextSubtitleInfo(
+ off64_t *offset, int64_t *startTimeUs, TextInfo *info) {
+ AString data;
+ status_t err;
+
+ // To skip blank lines.
+ do {
+ if ((err = readNextLine(offset, &data)) != OK) {
+ return err;
+ }
+ data.trim();
+ } while (data.empty());
+
+ // Just ignore the first non-blank line which is subtitle sequence number.
+ if ((err = readNextLine(offset, &data)) != OK) {
+ return err;
+ }
+ int hour1, hour2, min1, min2, sec1, sec2, msec1, msec2;
+ // the start time format is: hours:minutes:seconds,milliseconds
+ // 00:00:24,600 --> 00:00:27,800
+ if (sscanf(data.c_str(), "%02d:%02d:%02d,%03d --> %02d:%02d:%02d,%03d",
+ &hour1, &min1, &sec1, &msec1, &hour2, &min2, &sec2, &msec2) != 8) {
+ return ERROR_MALFORMED;
+ }
+
+ *startTimeUs = ((hour1 * 3600 + min1 * 60 + sec1) * 1000 + msec1) * 1000ll;
+ info->endTimeUs = ((hour2 * 3600 + min2 * 60 + sec2) * 1000 + msec2) * 1000ll;
+ if (info->endTimeUs <= *startTimeUs) {
+ return ERROR_MALFORMED;
+ }
+
+ info->offset = *offset;
+ bool needMoreData = true;
+ while (needMoreData) {
+ if ((err = readNextLine(offset, &data)) != OK) {
+ if (err == ERROR_END_OF_STREAM) {
+ needMoreData = false;
+ } else {
+ return err;
+ }
+ }
+
+ if (needMoreData) {
+ data.trim();
+ if (data.empty()) {
+ // it's an empty line used to separate two subtitles
+ needMoreData = false;
+ }
+ }
+ }
+ info->textLen = *offset - info->offset;
+ return OK;
+}
+
+status_t TimedTextSRTSource::readNextLine(off64_t *offset, AString *data) {
+ data->clear();
+ while (true) {
+ ssize_t readSize;
+ char character;
+ if ((readSize = mSource->readAt(*offset, &character, 1)) < 1) {
+ if (readSize == 0) {
+ return ERROR_END_OF_STREAM;
+ }
+ return ERROR_IO;
+ }
+
+ (*offset)++;
+
+ // a line could end with CR, LF or CR + LF
+ if (character == 10) {
+ break;
+ } else if (character == 13) {
+ if ((readSize = mSource->readAt(*offset, &character, 1)) < 1) {
+ if (readSize == 0) { // end of the stream
+ return OK;
+ }
+ return ERROR_IO;
+ }
+
+ (*offset)++;
+ if (character != 10) {
+ (*offset)--;
+ }
+ break;
+ }
+ data->append(character);
+ }
+ return OK;
+}
+
+status_t TimedTextSRTSource::getText(
+ const MediaSource::ReadOptions *options,
+ AString *text, int64_t *startTimeUs, int64_t *endTimeUs) {
+ text->clear();
+ int64_t seekTimeUs;
+ MediaSource::ReadOptions::SeekMode mode;
+ if (options != NULL && options->getSeekTo(&seekTimeUs, &mode)) {
+ int64_t lastEndTimeUs =
+ mTextVector.valueAt(mTextVector.size() - 1).endTimeUs;
+ int64_t firstStartTimeUs = mTextVector.keyAt(0);
+ if (seekTimeUs < 0 || seekTimeUs > lastEndTimeUs) {
+ return ERROR_OUT_OF_RANGE;
+ } else if (seekTimeUs < firstStartTimeUs) {
+ mIndex = 0;
+ } else {
+ // binary search
+ ssize_t low = 0;
+ ssize_t high = mTextVector.size() - 1;
+ ssize_t mid = 0;
+ int64_t currTimeUs;
+
+ while (low <= high) {
+ mid = low + (high - low)/2;
+ currTimeUs = mTextVector.keyAt(mid);
+ const int diff = currTimeUs - seekTimeUs;
+
+ if (diff == 0) {
+ break;
+ } else if (diff < 0) {
+ low = mid + 1;
+ } else {
+ if ((high == mid + 1)
+ && (seekTimeUs < mTextVector.keyAt(high))) {
+ break;
+ }
+ high = mid - 1;
+ }
+ }
+ mIndex = mid;
+ }
+ }
+ const TextInfo &info = mTextVector.valueAt(mIndex);
+ *startTimeUs = mTextVector.keyAt(mIndex);
+ *endTimeUs = info.endTimeUs;
+ mIndex++;
+
+ char *str = new char[info.textLen];
+ if (mSource->readAt(info.offset, str, info.textLen) < info.textLen) {
+ delete[] str;
+ return ERROR_IO;
+ }
+ text->append(str, info.textLen);
+ delete[] str;
+ return OK;
+}
+
+status_t TimedTextSRTSource::extractAndAppendLocalDescriptions(
+ int64_t timeUs, const AString &text, Parcel *parcel) {
+ const void *data = text.c_str();
+ size_t size = text.size();
+ int32_t flag = TextDescriptions::LOCAL_DESCRIPTIONS |
+ TextDescriptions::OUT_OF_BAND_TEXT_SRT;
+
+ if (size > 0) {
+ return TextDescriptions::getParcelOfDescriptions(
+ (const uint8_t *)data, size, flag, timeUs / 1000, parcel);
+ }
+ return OK;
+}
+
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextSRTSource.h b/media/libstagefright/timedtext/TimedTextSRTSource.h
new file mode 100644
index 0000000..a0734d9
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextSRTSource.h
@@ -0,0 +1,75 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TIMED_TEXT_SRT_SOURCE_H_
+#define TIMED_TEXT_SRT_SOURCE_H_
+
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h>
+#include <utils/Compat.h> // off64_t
+
+#include "TimedTextSource.h"
+
+namespace android {
+
+class AString;
+class DataSource;
+class MediaBuffer;
+class Parcel;
+
+class TimedTextSRTSource : public TimedTextSource {
+ public:
+ TimedTextSRTSource(const sp<DataSource>& dataSource);
+ virtual status_t start();
+ virtual status_t stop();
+ virtual status_t read(
+ int64_t *timeUs,
+ Parcel *parcel,
+ const MediaSource::ReadOptions *options = NULL);
+
+ protected:
+ virtual ~TimedTextSRTSource();
+
+ private:
+ sp<DataSource> mSource;
+
+ struct TextInfo {
+ int64_t endTimeUs;
+ // The offset of the text in the original file.
+ off64_t offset;
+ int textLen;
+ };
+
+ int mIndex;
+ KeyedVector<int64_t, TextInfo> mTextVector;
+
+ void reset();
+ status_t scanFile();
+ status_t getNextSubtitleInfo(
+ off64_t *offset, int64_t *startTimeUs, TextInfo *info);
+ status_t readNextLine(off64_t *offset, AString *data);
+ status_t getText(
+ const MediaSource::ReadOptions *options,
+ AString *text, int64_t *startTimeUs, int64_t *endTimeUs);
+ status_t extractAndAppendLocalDescriptions(
+ int64_t timeUs, const AString &text, Parcel *parcel);
+
+ DISALLOW_EVIL_CONSTRUCTORS(TimedTextSRTSource);
+};
+
+} // namespace android
+
+#endif // TIMED_TEXT_SRT_SOURCE_H_
diff --git a/media/libstagefright/timedtext/TimedTextSource.cpp b/media/libstagefright/timedtext/TimedTextSource.cpp
new file mode 100644
index 0000000..ffbe1c3
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextSource.cpp
@@ -0,0 +1,62 @@
+ /*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "TimedTextSource"
+#include <utils/Log.h>
+
+#include <media/stagefright/foundation/ADebug.h> // CHECK_XX macro
+#include <media/stagefright/DataSource.h>
+#include <media/stagefright/MediaDefs.h> // for MEDIA_MIMETYPE_xxx
+#include <media/stagefright/MediaSource.h>
+#include <media/stagefright/MetaData.h>
+
+#include "TimedTextSource.h"
+
+#include "TimedText3GPPSource.h"
+#include "TimedTextSRTSource.h"
+
+namespace android {
+
+// static
+sp<TimedTextSource> TimedTextSource::CreateTimedTextSource(
+ const sp<MediaSource>& mediaSource) {
+ const char *mime;
+ CHECK(mediaSource->getFormat()->findCString(kKeyMIMEType, &mime));
+ if (strcasecmp(mime, MEDIA_MIMETYPE_TEXT_3GPP) == 0) {
+ return new TimedText3GPPSource(mediaSource);
+ }
+ ALOGE("Unsupported mime type for subtitle. : %s", mime);
+ return NULL;
+}
+
+// static
+sp<TimedTextSource> TimedTextSource::CreateTimedTextSource(
+ const sp<DataSource>& dataSource, FileType filetype) {
+ switch(filetype) {
+ case OUT_OF_BAND_FILE_SRT:
+ return new TimedTextSRTSource(dataSource);
+ case OUT_OF_BAND_FILE_SMI:
+ // TODO: Implement for SMI.
+ ALOGE("Supporting SMI is not implemented yet");
+ break;
+ default:
+ ALOGE("Undefined subtitle format. : %d", filetype);
+ }
+ return NULL;
+}
+
+} // namespace android
diff --git a/media/libstagefright/timedtext/TimedTextSource.h b/media/libstagefright/timedtext/TimedTextSource.h
new file mode 100644
index 0000000..06bae71
--- /dev/null
+++ b/media/libstagefright/timedtext/TimedTextSource.h
@@ -0,0 +1,61 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef TIMED_TEXT_SOURCE_H_
+#define TIMED_TEXT_SOURCE_H_
+
+#include <media/stagefright/foundation/ABase.h> // for DISALLOW_XXX macro.
+#include <media/stagefright/MediaErrors.h>
+#include <media/stagefright/MediaSource.h> // for MediaSource::ReadOptions
+#include <utils/RefBase.h>
+
+namespace android {
+
+class DataSource;
+class Parcel;
+
+class TimedTextSource : public RefBase {
+ public:
+ enum FileType {
+ OUT_OF_BAND_FILE_SRT = 1,
+ OUT_OF_BAND_FILE_SMI = 2,
+ };
+ static sp<TimedTextSource> CreateTimedTextSource(
+ const sp<MediaSource>& source);
+ static sp<TimedTextSource> CreateTimedTextSource(
+ const sp<DataSource>& source, FileType filetype);
+ TimedTextSource() {}
+ virtual status_t start() = 0;
+ virtual status_t stop() = 0;
+ // Returns subtitle parcel and its start time.
+ virtual status_t read(
+ int64_t *timeUs,
+ Parcel *parcel,
+ const MediaSource::ReadOptions *options = NULL) = 0;
+ virtual status_t extractGlobalDescriptions(Parcel *parcel) {
+ return INVALID_OPERATION;
+ }
+
+ protected:
+ virtual ~TimedTextSource() { }
+
+ private:
+ DISALLOW_EVIL_CONSTRUCTORS(TimedTextSource);
+};
+
+} // namespace android
+
+#endif // TIMED_TEXT_SOURCE_H_
diff --git a/media/libstagefright/yuv/YUVCanvas.cpp b/media/libstagefright/yuv/YUVCanvas.cpp
index 38aa779..4c9fee8 100644
--- a/media/libstagefright/yuv/YUVCanvas.cpp
+++ b/media/libstagefright/yuv/YUVCanvas.cpp
@@ -17,7 +17,7 @@
#define LOG_NDEBUG 0
#define LOG_TAG "YUVCanvas"
-#include <media/stagefright/MediaDebug.h>
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/YUVCanvas.h>
#include <media/stagefright/YUVImage.h>
#include <ui/Rect.h>
diff --git a/media/libstagefright/yuv/YUVImage.cpp b/media/libstagefright/yuv/YUVImage.cpp
index 0d67c96..7b9000b 100644
--- a/media/libstagefright/yuv/YUVImage.cpp
+++ b/media/libstagefright/yuv/YUVImage.cpp
@@ -17,9 +17,9 @@
#define LOG_NDEBUG 0
#define LOG_TAG "YUVImage"
+#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/YUVImage.h>
#include <ui/Rect.h>
-#include <media/stagefright/MediaDebug.h>
namespace android {
diff --git a/media/mediaserver/main_mediaserver.cpp b/media/mediaserver/main_mediaserver.cpp
index f078192..1520c01 100644
--- a/media/mediaserver/main_mediaserver.cpp
+++ b/media/mediaserver/main_mediaserver.cpp
@@ -15,10 +15,7 @@
** limitations under the License.
*/
-// System headers required for setgroups, etc.
-#include <sys/types.h>
-#include <unistd.h>
-#include <grp.h>
+#define LOG_TAG "mediaserver"
#include <binder/IPCThreadState.h>
#include <binder/ProcessState.h>
@@ -29,7 +26,6 @@
#include <CameraService.h>
#include <MediaPlayerService.h>
#include <AudioPolicyService.h>
-#include <private/android_filesystem_config.h>
using namespace android;
diff --git a/media/mtp/Android.mk b/media/mtp/Android.mk
index e590bab..fc7fc4f 100644
--- a/media/mtp/Android.mk
+++ b/media/mtp/Android.mk
@@ -39,6 +39,9 @@
LOCAL_CFLAGS := -DMTP_DEVICE -DMTP_HOST
+# Needed for <bionic_time.h>
+LOCAL_C_INCLUDES := bionic/libc/private
+
LOCAL_SHARED_LIBRARIES := libutils libcutils libusbhost libbinder
include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index fa49592..22fa752 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -6,14 +6,19 @@
AudioFlinger.cpp \
AudioMixer.cpp.arm \
AudioResampler.cpp.arm \
- AudioResamplerSinc.cpp.arm \
- AudioResamplerCubic.cpp.arm \
- AudioPolicyService.cpp
+ AudioPolicyService.cpp \
+ AudioBufferProvider.cpp \
+ ServiceUtilities.cpp
+# AudioResamplerSinc.cpp.arm
+# AudioResamplerCubic.cpp.arm
LOCAL_C_INCLUDES := \
- system/media/audio_effects/include
+ system/media/audio_effects/include \
+ system/media/audio_utils/include
LOCAL_SHARED_LIBRARIES := \
+ libaudioutils \
+ libcommon_time_client \
libcutils \
libutils \
libbinder \
diff --git a/include/private/ui/android_natives_priv.h b/services/audioflinger/AudioBufferProvider.cpp
similarity index 65%
rename from include/private/ui/android_natives_priv.h
rename to services/audioflinger/AudioBufferProvider.cpp
index 6b9f524..678fd58 100644
--- a/include/private/ui/android_natives_priv.h
+++ b/services/audioflinger/AudioBufferProvider.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2009 The Android Open Source Project
+ * Copyright (C) 2011 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,4 +14,15 @@
* limitations under the License.
*/
-#include <ui/android_native_buffer.h>
+#undef __STRICT_ANSI__
+#define __STDINT_LIMITS
+#define __STDC_LIMIT_MACROS
+#include <stdint.h>
+
+#include "AudioBufferProvider.h"
+
+namespace android {
+
+const int64_t AudioBufferProvider::kInvalidPTS = INT64_MAX;
+
+}; // namespace android
diff --git a/services/audioflinger/AudioBufferProvider.h b/services/audioflinger/AudioBufferProvider.h
index 81c5c39..62ad6bd 100644
--- a/services/audioflinger/AudioBufferProvider.h
+++ b/services/audioflinger/AudioBufferProvider.h
@@ -38,8 +38,15 @@
};
virtual ~AudioBufferProvider() {}
-
- virtual status_t getNextBuffer(Buffer* buffer) = 0;
+
+ // value representing an invalid presentation timestamp
+ static const int64_t kInvalidPTS;
+
+ // pts is the local time when the next sample yielded by getNextBuffer
+ // will be rendered.
+ // Pass kInvalidPTS if the PTS is unknown or not applicable.
+ virtual status_t getNextBuffer(Buffer* buffer, int64_t pts) = 0;
+
virtual void releaseBuffer(Buffer* buffer) = 0;
};
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 5a46e44..2687cd5 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioFlinger.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -35,10 +35,10 @@
#include <cutils/bitops.h>
#include <cutils/properties.h>
+#include <cutils/compiler.h>
-#include <media/AudioTrack.h>
-#include <media/AudioRecord.h>
#include <media/IMediaPlayerService.h>
+#include <media/IMediaDeathNotifier.h>
#include <private/media/AudioTrackShared.h>
#include <private/media/AudioEffectShared.h>
@@ -48,27 +48,32 @@
#include "AudioMixer.h"
#include "AudioFlinger.h"
+#include "ServiceUtilities.h"
#include <media/EffectsFactoryApi.h>
#include <audio_effects/effect_visualizer.h>
#include <audio_effects/effect_ns.h>
#include <audio_effects/effect_aec.h>
+#include <audio_utils/primitives.h>
+
#include <cpustats/ThreadCpuUsage.h>
#include <powermanager/PowerManager.h>
// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds
+#include <common_time/cc_helper.h>
+#include <common_time/local_clock.h>
+
// ----------------------------------------------------------------------------
namespace android {
-static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n";
-static const char* kHardwareLockedString = "Hardware lock is taken\n";
+static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
+static const char kHardwareLockedString[] = "Hardware lock is taken\n";
-//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
static const float MAX_GAIN = 4096.0f;
-static const float MAX_GAIN_INT = 0x1000;
+static const uint32_t MAX_GAIN_INT = 0x1000;
// retry counts for buffer fill timeout
// 50 * ~20msecs = 1 second
@@ -80,44 +85,31 @@
static const int8_t kMaxTrackRetriesDirect = 2;
static const int kDumpLockRetries = 50;
-static const int kDumpLockSleep = 20000;
+static const int kDumpLockSleepUs = 20000;
-static const nsecs_t kWarningThrottle = seconds(5);
+// don't warn about blocked writes or record buffer overflows more often than this
+static const nsecs_t kWarningThrottleNs = seconds(5);
// RecordThread loop sleep time upon application overrun or audio HAL read error
static const int kRecordThreadSleepUs = 5000;
-static const nsecs_t kSetParametersTimeout = seconds(2);
+// maximum time to wait for setParameters to complete
+static const nsecs_t kSetParametersTimeoutNs = seconds(2);
// minimum sleep time for the mixer thread loop when tracks are active but in underrun
static const uint32_t kMinThreadSleepTimeUs = 5000;
// maximum divider applied to the active sleep time in the mixer thread loop
static const uint32_t kMaxThreadSleepTimeShift = 2;
+nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
// ----------------------------------------------------------------------------
-static bool recordingAllowed() {
- if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
- bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
- if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
- return ok;
-}
-
-static bool settingsAllowed() {
- if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
- bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
- if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
- return ok;
-}
-
// To collect the amplifier usage
static void addBatteryData(uint32_t params) {
- sp<IBinder> binder =
- defaultServiceManager()->getService(String16("media.player"));
- sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder);
- if (service.get() == NULL) {
- ALOGW("Cannot connect to the MediaPlayerService for battery tracking");
+ sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService();
+ if (service == NULL) {
+ // it already logged
return;
}
@@ -147,7 +139,7 @@
return rc;
}
-static const char *audio_interfaces[] = {
+static const char * const audio_interfaces[] = {
"primary",
"a2dp",
"usb",
@@ -158,8 +150,14 @@
AudioFlinger::AudioFlinger()
: BnAudioFlinger(),
- mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1),
- mBtNrecIsOff(false)
+ mPrimaryHardwareDev(NULL),
+ mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef()
+ mMasterVolume(1.0f),
+ mMasterVolumeSupportLvl(MVS_NONE),
+ mMasterMute(false),
+ mNextUniqueId(1),
+ mMode(AUDIO_MODE_INVALID),
+ mBtNrecIsOff(false)
{
}
@@ -170,7 +168,18 @@
Mutex::Autolock _l(mLock);
/* TODO: move all this work into an Init() function */
- mHardwareStatus = AUDIO_HW_IDLE;
+ char val_str[PROPERTY_VALUE_MAX] = { 0 };
+ if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) {
+ uint32_t int_val;
+ if (1 == sscanf(val_str, "%u", &int_val)) {
+ mStandbyTimeInNsecs = milliseconds(int_val);
+ ALOGI("Using %u mSec as standby time.", int_val);
+ } else {
+ mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs;
+ ALOGI("Using default %u mSec as standby time.",
+ (uint32_t)(mStandbyTimeInNsecs / 1000000));
+ }
+ }
for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) {
const hw_module_t *mod;
@@ -184,49 +193,80 @@
mod->name, mod->id);
mAudioHwDevs.push(dev);
- if (!mPrimaryHardwareDev) {
+ if (mPrimaryHardwareDev == NULL) {
mPrimaryHardwareDev = dev;
ALOGI("Using '%s' (%s.%s) as the primary audio interface",
mod->name, mod->id, audio_interfaces[i]);
}
}
- mHardwareStatus = AUDIO_HW_INIT;
-
- if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) {
+ if (mPrimaryHardwareDev == NULL) {
ALOGE("Primary audio interface not found");
- return;
+ // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck()
}
+ // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the
+ // primary HW dev is selected can change so these conditions might not always be equivalent.
+ // When that happens, re-visit all the code that assumes this.
+
+ AutoMutex lock(mHardwareLock);
+
+ // Determine the level of master volume support the primary audio HAL has,
+ // and set the initial master volume at the same time.
+ float initialVolume = 1.0;
+ mMasterVolumeSupportLvl = MVS_NONE;
+ if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) {
+ audio_hw_device_t *dev = mPrimaryHardwareDev;
+
+ mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
+ if ((NULL != dev->get_master_volume) &&
+ (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) {
+ mMasterVolumeSupportLvl = MVS_FULL;
+ } else {
+ mMasterVolumeSupportLvl = MVS_SETONLY;
+ initialVolume = 1.0;
+ }
+
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ if ((NULL == dev->set_master_volume) ||
+ (NO_ERROR != dev->set_master_volume(dev, initialVolume))) {
+ mMasterVolumeSupportLvl = MVS_NONE;
+ }
+ mHardwareStatus = AUDIO_HW_INIT;
+ }
+
+ // Set the mode for each audio HAL, and try to set the initial volume (if
+ // supported) for all of the non-primary audio HALs.
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
mHardwareStatus = AUDIO_HW_INIT;
rc = dev->init_check(dev);
+ mHardwareStatus = AUDIO_HW_IDLE;
if (rc == 0) {
- AutoMutex lock(mHardwareLock);
-
- mMode = AUDIO_MODE_NORMAL;
+ mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value
mHardwareStatus = AUDIO_HW_SET_MODE;
dev->set_mode(dev, mMode);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- dev->set_master_volume(dev, 1.0f);
- mHardwareStatus = AUDIO_HW_IDLE;
+
+ if ((dev != mPrimaryHardwareDev) &&
+ (NULL != dev->set_master_volume)) {
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ dev->set_master_volume(dev, initialVolume);
+ }
+
+ mHardwareStatus = AUDIO_HW_INIT;
}
}
-}
-status_t AudioFlinger::initCheck() const
-{
- Mutex::Autolock _l(mLock);
- if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0)
- return NO_INIT;
- return NO_ERROR;
+ mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl)
+ ? initialVolume
+ : 1.0;
+ mMasterVolume = initialVolume;
+ mHardwareStatus = AUDIO_HW_IDLE;
}
AudioFlinger::~AudioFlinger()
{
- int num_devs = mAudioHwDevs.size();
while (!mRecordThreads.isEmpty()) {
// closeInput() will remove first entry from mRecordThreads
@@ -237,11 +277,10 @@
closeOutput(mPlaybackThreads.keyAt(0));
}
- for (int i = 0; i < num_devs; i++) {
- audio_hw_device_t *dev = mAudioHwDevs[i];
- audio_hw_device_close(dev);
+ for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+ // no mHardwareLock needed, as there are no other references to this
+ audio_hw_device_close(mAudioHwDevs[i]);
}
- mAudioHwDevs.clear();
}
audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices)
@@ -263,13 +302,10 @@
result.append("Clients:\n");
for (size_t i = 0; i < mClients.size(); ++i) {
- wp<Client> wClient = mClients.valueAt(i);
- if (wClient != 0) {
- sp<Client> client = wClient.promote();
- if (client != 0) {
- snprintf(buffer, SIZE, " pid: %d\n", client->pid());
- result.append(buffer);
- }
+ sp<Client> client = mClients.valueAt(i).promote();
+ if (client != 0) {
+ snprintf(buffer, SIZE, " pid: %d\n", client->pid());
+ result.append(buffer);
}
}
@@ -290,9 +326,12 @@
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- int hardwareStatus = mHardwareStatus;
+ hardware_call_state hardwareStatus = mHardwareStatus;
- snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus);
+ snprintf(buffer, SIZE, "Hardware status: %d\n"
+ "Standby Time mSec: %u\n",
+ hardwareStatus,
+ (uint32_t)(mStandbyTimeInNsecs / 1000000));
result.append(buffer);
write(fd, result.string(), result.size());
return NO_ERROR;
@@ -320,14 +359,14 @@
locked = true;
break;
}
- usleep(kDumpLockSleep);
+ usleep(kDumpLockSleepUs);
}
return locked;
}
status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
{
- if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+ if (!dumpAllowed()) {
dumpPermissionDenial(fd, args);
} else {
// get state of hardware lock
@@ -370,32 +409,47 @@
return NO_ERROR;
}
+sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid)
+{
+ // If pid is already in the mClients wp<> map, then use that entry
+ // (for which promote() is always != 0), otherwise create a new entry and Client.
+ sp<Client> client = mClients.valueFor(pid).promote();
+ if (client == 0) {
+ client = new Client(this, pid);
+ mClients.add(pid, client);
+ }
+
+ return client;
+}
// IAudioFlinger interface
sp<IAudioTrack> AudioFlinger::createTrack(
pid_t pid,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
+ // FIXME dead, remove from IAudioFlinger
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
+ bool isTimed,
int *sessionId,
status_t *status)
{
sp<PlaybackThread::Track> track;
sp<TrackHandle> trackHandle;
sp<Client> client;
- wp<Client> wclient;
status_t lStatus;
int lSessionId;
- if (streamType >= AUDIO_STREAM_CNT) {
- ALOGE("invalid stream type");
+ // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC,
+ // but if someone uses binder directly they could bypass that and cause us to crash
+ if (uint32_t(streamType) >= AUDIO_STREAM_CNT) {
+ ALOGE("createTrack() invalid stream type %d", streamType);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -410,14 +464,7 @@
goto Exit;
}
- wclient = mClients.valueFor(pid);
-
- if (wclient != NULL) {
- client = wclient.promote();
- } else {
- client = new Client(this, pid);
- mClients.add(pid, client);
- }
+ client = registerPid_l(pid);
ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId);
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
@@ -427,6 +474,7 @@
// prevent same audio session on different output threads
uint32_t sessions = t->hasAudioSession(*sessionId);
if (sessions & PlaybackThread::TRACK_SESSION) {
+ ALOGE("createTrack() session ID %d already in use", *sessionId);
lStatus = BAD_VALUE;
goto Exit;
}
@@ -447,7 +495,7 @@
ALOGV("createTrack() lSessionId: %d", lSessionId);
track = thread->createTrack_l(client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, lSessionId, &lStatus);
+ channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus);
// move effect chain to this output thread if an effect on same session was waiting
// for a track to be created
@@ -473,7 +521,7 @@
return trackHandle;
}
-uint32_t AudioFlinger::sampleRate(int output) const
+uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -484,7 +532,7 @@
return thread->sampleRate();
}
-int AudioFlinger::channelCount(int output) const
+int AudioFlinger::channelCount(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -495,18 +543,18 @@
return thread->channelCount();
}
-uint32_t AudioFlinger::format(int output) const
+audio_format_t AudioFlinger::format(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
if (thread == NULL) {
ALOGW("format() unknown thread %d", output);
- return 0;
+ return AUDIO_FORMAT_INVALID;
}
return thread->format();
}
-size_t AudioFlinger::frameCount(int output) const
+size_t AudioFlinger::frameCount(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -517,7 +565,7 @@
return thread->frameCount();
}
-uint32_t AudioFlinger::latency(int output) const
+uint32_t AudioFlinger::latency(audio_io_handle_t output) const
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -540,25 +588,34 @@
return PERMISSION_DENIED;
}
+ float swmv = value;
+
// when hw supports master volume, don't scale in sw mixer
- { // scope for the lock
- AutoMutex lock(mHardwareLock);
- mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
- if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) {
- value = 1.0f;
+ if (MVS_NONE != mMasterVolumeSupportLvl) {
+ for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
+ AutoMutex lock(mHardwareLock);
+ audio_hw_device_t *dev = mAudioHwDevs[i];
+
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ if (NULL != dev->set_master_volume) {
+ dev->set_master_volume(dev, value);
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
}
- mHardwareStatus = AUDIO_HW_IDLE;
+
+ swmv = 1.0;
}
Mutex::Autolock _l(mLock);
- mMasterVolume = value;
- for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setMasterVolume(value);
+ mMasterVolume = value;
+ mMasterVolumeSW = swmv;
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++)
+ mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
return NO_ERROR;
}
-status_t AudioFlinger::setMode(int mode)
+status_t AudioFlinger::setMode(audio_mode_t mode)
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
@@ -569,7 +626,7 @@
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) {
+ if (uint32_t(mode) >= AUDIO_MODE_CNT) {
ALOGW("Illegal value: setMode(%d)", mode);
return BAD_VALUE;
}
@@ -584,7 +641,7 @@
if (NO_ERROR == ret) {
Mutex::Autolock _l(mLock);
mMode = mode;
- for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMode(mode);
}
@@ -618,6 +675,7 @@
}
bool state = AUDIO_MODE_INVALID;
+ AutoMutex lock(mHardwareLock);
mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state);
mHardwareStatus = AUDIO_HW_IDLE;
@@ -632,8 +690,9 @@
}
Mutex::Autolock _l(mLock);
+ // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger
mMasterMute = muted;
- for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++)
mPlaybackThreads.valueAt(i)->setMasterMute(muted);
return NO_ERROR;
@@ -641,22 +700,50 @@
float AudioFlinger::masterVolume() const
{
- return mMasterVolume;
+ Mutex::Autolock _l(mLock);
+ return masterVolume_l();
+}
+
+float AudioFlinger::masterVolumeSW() const
+{
+ Mutex::Autolock _l(mLock);
+ return masterVolumeSW_l();
}
bool AudioFlinger::masterMute() const
{
- return mMasterMute;
+ Mutex::Autolock _l(mLock);
+ return masterMute_l();
}
-status_t AudioFlinger::setStreamVolume(int stream, float value, int output)
+float AudioFlinger::masterVolume_l() const
+{
+ if (MVS_FULL == mMasterVolumeSupportLvl) {
+ float ret_val;
+ AutoMutex lock(mHardwareLock);
+
+ mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME;
+ assert(NULL != mPrimaryHardwareDev);
+ assert(NULL != mPrimaryHardwareDev->get_master_volume);
+
+ mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return ret_val;
+ }
+
+ return mMasterVolume;
+}
+
+status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ ALOGE("setStreamVolume() invalid stream %d", stream);
return BAD_VALUE;
}
@@ -672,7 +759,7 @@
mStreamTypes[stream].volume = value;
if (thread == NULL) {
- for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) {
+ for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
}
} else {
@@ -682,15 +769,16 @@
return NO_ERROR;
}
-status_t AudioFlinger::setStreamMute(int stream, bool muted)
+status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
{
// check calling permissions
if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT ||
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT ||
uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) {
+ ALOGE("setStreamMute() invalid stream %d", stream);
return BAD_VALUE;
}
@@ -702,9 +790,9 @@
return NO_ERROR;
}
-float AudioFlinger::streamVolume(int stream, int output) const
+float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const
{
- if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return 0.0f;
}
@@ -717,26 +805,27 @@
}
volume = thread->streamVolume(stream);
} else {
- volume = mStreamTypes[stream].volume;
+ volume = streamVolume_l(stream);
}
return volume;
}
-bool AudioFlinger::streamMute(int stream) const
+bool AudioFlinger::streamMute(audio_stream_type_t stream) const
{
- if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return true;
}
- return mStreamTypes[stream].mute;
+ AutoMutex lock(mLock);
+ return streamMute_l(stream);
}
-status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs)
+status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs)
{
status_t result;
- ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d",
+ ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d",
ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid());
// check calling permissions
if (!settingsAllowed()) {
@@ -790,7 +879,7 @@
thread = checkPlaybackThread_l(ioHandle);
if (thread == NULL) {
thread = checkRecordThread_l(ioHandle);
- } else if (thread.get() == primaryPlaybackThread_l()) {
+ } else if (thread == primaryPlaybackThread_l()) {
// indicate output device change to all input threads for pre processing
AudioParameter param = AudioParameter(keyValuePairs);
int value;
@@ -801,16 +890,15 @@
}
}
}
- if (thread != NULL) {
- result = thread->setParameters(keyValuePairs);
- return result;
+ if (thread != 0) {
+ return thread->setParameters(keyValuePairs);
}
return BAD_VALUE;
}
-String8 AudioFlinger::getParameters(int ioHandle, const String8& keys)
+String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const
{
-// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d",
+// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d",
// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid());
if (ioHandle == 0) {
@@ -819,7 +907,7 @@
for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
audio_hw_device_t *dev = mAudioHwDevs[i];
char *s = dev->get_parameters(dev, keys.string());
- out_s8 += String8(s);
+ out_s8 += String8(s ? s : "");
free(s);
}
return out_s8;
@@ -838,17 +926,21 @@
return String8("");
}
-size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const
{
status_t ret = initCheck();
if (ret != NO_ERROR) {
return 0;
}
- return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE;
+ size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return size;
}
-unsigned int AudioFlinger::getInputFramesLost(int ioHandle)
+unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const
{
if (ioHandle == 0) {
return 0;
@@ -883,7 +975,8 @@
return ret;
}
-status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output)
+status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const
{
status_t status;
@@ -902,7 +995,7 @@
Mutex::Autolock _l(mLock);
- int pid = IPCThreadState::self()->getCallingPid();
+ pid_t pid = IPCThreadState::self()->getCallingPid();
if (mNotificationClients.indexOfKey(pid) < 0) {
sp<NotificationClient> notificationClient = new NotificationClient(this,
client,
@@ -930,7 +1023,7 @@
{
Mutex::Autolock _l(mLock);
- int index = mNotificationClients.indexOfKey(pid);
+ ssize_t index = mNotificationClients.indexOfKey(pid);
if (index >= 0) {
sp <NotificationClient> client = mNotificationClients.valueFor(pid);
ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid);
@@ -938,9 +1031,9 @@
}
ALOGV("%d died, releasing its sessions", pid);
- int num = mAudioSessionRefs.size();
+ size_t num = mAudioSessionRefs.size();
bool removed = false;
- for (int i = 0; i< num; i++) {
+ for (size_t i = 0; i< num; ) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
ALOGV(" pid %d @ %d", ref->pid, i);
if (ref->pid == pid) {
@@ -948,8 +1041,9 @@
mAudioSessionRefs.removeAt(i);
delete ref;
removed = true;
- i--;
num--;
+ } else {
+ i++;
}
}
if (removed) {
@@ -958,11 +1052,12 @@
}
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
-void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2)
+void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2)
{
size_t size = mNotificationClients.size();
for (size_t i = 0; i < size; i++) {
- mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2);
+ mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle,
+ param2);
}
}
@@ -976,19 +1071,24 @@
// ----------------------------------------------------------------------------
-AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device)
+AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id,
+ uint32_t device, type_t type)
: Thread(false),
- mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
- mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false),
- mDevice(device)
+ mType(type),
+ mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0),
+ // mChannelMask
+ mChannelCount(0),
+ mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID),
+ mParamStatus(NO_ERROR),
+ mStandby(false), mId(id),
+ mDevice(device),
+ mDeathRecipient(new PMDeathRecipient(this))
{
- mDeathRecipient = new PMDeathRecipient(this);
}
AudioFlinger::ThreadBase::~ThreadBase()
{
mParamCond.broadcast();
- mNewParameters.clear();
// do not lock the mutex in destructor
releaseWakeLock_l();
if (mPowerManager != 0) {
@@ -999,40 +1099,26 @@
void AudioFlinger::ThreadBase::exit()
{
- // keep a strong ref on ourself so that we wont get
- // destroyed in the middle of requestExitAndWait()
- sp <ThreadBase> strongMe = this;
-
ALOGV("ThreadBase::exit");
{
- AutoMutex lock(&mLock);
- mExiting = true;
+ // This lock prevents the following race in thread (uniprocessor for illustration):
+ // if (!exitPending()) {
+ // // context switch from here to exit()
+ // // exit() calls requestExit(), what exitPending() observes
+ // // exit() calls signal(), which is dropped since no waiters
+ // // context switch back from exit() to here
+ // mWaitWorkCV.wait(...);
+ // // now thread is hung
+ // }
+ AutoMutex lock(mLock);
requestExit();
mWaitWorkCV.signal();
}
+ // When Thread::requestExitAndWait is made virtual and this method is renamed to
+ // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();"
requestExitAndWait();
}
-uint32_t AudioFlinger::ThreadBase::sampleRate() const
-{
- return mSampleRate;
-}
-
-int AudioFlinger::ThreadBase::channelCount() const
-{
- return (int)mChannelCount;
-}
-
-uint32_t AudioFlinger::ThreadBase::format() const
-{
- return mFormat;
-}
-
-size_t AudioFlinger::ThreadBase::frameCount() const
-{
- return mFrameCount;
-}
-
status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs)
{
status_t status;
@@ -1044,7 +1130,7 @@
mWaitWorkCV.signal();
// wait condition with timeout in case the thread loop has exited
// before the request could be processed
- if (mParamCond.waitRelative(mLock, kSetParametersTimeout) == NO_ERROR) {
+ if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) {
status = mParamStatus;
mWaitWorkCV.signal();
} else {
@@ -1062,9 +1148,9 @@
// sendConfigEvent_l() must be called with ThreadBase::mLock held
void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
{
- ConfigEvent *configEvent = new ConfigEvent();
- configEvent->mEvent = event;
- configEvent->mParam = param;
+ ConfigEvent configEvent;
+ configEvent.mEvent = event;
+ configEvent.mParam = param;
mConfigEvents.add(configEvent);
ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
mWaitWorkCV.signal();
@@ -1075,15 +1161,14 @@
mLock.lock();
while(!mConfigEvents.isEmpty()) {
ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
- ConfigEvent *configEvent = mConfigEvents[0];
+ ConfigEvent configEvent = mConfigEvents[0];
mConfigEvents.removeAt(0);
// release mLock before locking AudioFlinger mLock: lock order is always
// AudioFlinger then ThreadBase to avoid cross deadlock
mLock.unlock();
mAudioFlinger->mLock.lock();
- audioConfigChanged_l(configEvent->mEvent, configEvent->mParam);
+ audioConfigChanged_l(configEvent.mEvent, configEvent.mParam);
mAudioFlinger->mLock.unlock();
- delete configEvent;
mLock.lock();
}
mLock.unlock();
@@ -1113,7 +1198,7 @@
result.append(buffer);
snprintf(buffer, SIZE, "Format: %d\n", mFormat);
result.append(buffer);
- snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize);
+ snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize);
result.append(buffer);
snprintf(buffer, SIZE, "\nPending setParameters commands: \n");
@@ -1130,7 +1215,7 @@
snprintf(buffer, SIZE, " Index event param\n");
result.append(buffer);
for (size_t i = 0; i < mConfigEvents.size(); i++) {
- snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam);
+ snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam);
result.append(buffer);
}
result.append("\n");
@@ -1235,8 +1320,7 @@
void AudioFlinger::ThreadBase::setEffectSuspended_l(
const effect_uuid_t *type, bool suspend, int sessionId)
{
- sp<EffectChain> chain;
- chain = getEffectChain_l(sessionId);
+ sp<EffectChain> chain = getEffectChain_l(sessionId);
if (chain != 0) {
if (type != NULL) {
chain->setEffectSuspended_l(type, suspend);
@@ -1250,7 +1334,7 @@
void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain)
{
- int index = mSuspendedSessions.indexOfKey(chain->sessionId());
+ ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId());
if (index < 0) {
return;
}
@@ -1276,7 +1360,7 @@
bool suspend,
int sessionId)
{
- int index = mSuspendedSessions.indexOfKey(sessionId);
+ ssize_t index = mSuspendedSessions.indexOfKey(sessionId);
KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects;
@@ -1364,24 +1448,36 @@
AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
- int id,
- uint32_t device)
- : ThreadBase(audioFlinger, id, device),
- mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output),
+ audio_io_handle_t id,
+ uint32_t device,
+ type_t type)
+ : ThreadBase(audioFlinger, id, device, type),
+ mMixBuffer(NULL), mSuspended(0), mBytesWritten(0),
+ // Assumes constructor is called by AudioFlinger with it's mLock held,
+ // but it would be safer to explicitly pass initial masterMute as parameter
+ mMasterMute(audioFlinger->masterMute_l()),
+ // mStreamTypes[] initialized in constructor body
+ mOutput(output),
+ // Assumes constructor is called by AudioFlinger with it's mLock held,
+ // but it would be safer to explicitly pass initial masterVolume as parameter
+ mMasterVolume(audioFlinger->masterVolumeSW_l()),
mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false)
{
snprintf(mName, kNameLength, "AudioOut_%d", id);
readOutputParameters();
- mMasterVolume = mAudioFlinger->masterVolume();
- mMasterMute = mAudioFlinger->masterMute();
-
- for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) {
- mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream);
- mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream);
- mStreamTypes[stream].valid = true;
+ // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor
+ // There is no AUDIO_STREAM_MIN, and ++ operator does not compile
+ for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT;
+ stream = (audio_stream_type_t) (stream + 1)) {
+ mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream);
+ mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream);
+ // initialized by stream_type_t default constructor
+ // mStreamTypes[stream].valid = true;
}
+ // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here,
+ // because mAudioFlinger doesn't have one to copy from
}
AudioFlinger::PlaybackThread::~PlaybackThread()
@@ -1418,13 +1514,10 @@
result.append(buffer);
result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n");
for (size_t i = 0; i < mActiveTracks.size(); ++i) {
- wp<Track> wTrack = mActiveTracks[i];
- if (wTrack != 0) {
- sp<Track> track = wTrack.promote();
- if (track != 0) {
- track->dump(buffer, SIZE);
- result.append(buffer);
- }
+ sp<Track> track = mActiveTracks[i].promote();
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
}
}
write(fd, result.string(), result.size());
@@ -1478,13 +1571,14 @@
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
+ bool isTimed,
status_t *status)
{
sp<Track> track;
@@ -1521,22 +1615,28 @@
// all tracks in same audio session must share the same routing strategy otherwise
// conflicts will happen when tracks are moved from one output to another by audio policy
// manager
- uint32_t strategy =
- AudioSystem::getStrategyForStream((audio_stream_type_t)streamType);
+ uint32_t strategy = AudioSystem::getStrategyForStream(streamType);
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> t = mTracks[i];
if (t != 0) {
- if (sessionId == t->sessionId() &&
- strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) {
+ uint32_t actual = AudioSystem::getStrategyForStream(t->streamType());
+ if (sessionId == t->sessionId() && strategy != actual) {
+ ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
+ strategy, actual);
lStatus = BAD_VALUE;
goto Exit;
}
}
}
- track = new Track(this, client, streamType, sampleRate, format,
- channelMask, frameCount, sharedBuffer, sessionId);
- if (track->getCblk() == NULL || track->name() < 0) {
+ if (!isTimed) {
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelMask, frameCount, sharedBuffer, sessionId);
+ } else {
+ track = TimedTrack::create(this, client, streamType, sampleRate, format,
+ channelMask, frameCount, sharedBuffer, sessionId);
+ }
+ if (track == NULL || track->getCblk() == NULL || track->name() < 0) {
lStatus = NO_MEMORY;
goto Exit;
}
@@ -1546,7 +1646,7 @@
if (chain != 0) {
ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
track->setMainBuffer(chain->inBuffer());
- chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type()));
+ chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
chain->incTrackCnt();
}
@@ -1577,50 +1677,36 @@
}
}
-status_t AudioFlinger::PlaybackThread::setMasterVolume(float value)
+void AudioFlinger::PlaybackThread::setMasterVolume(float value)
{
+ Mutex::Autolock _l(mLock);
mMasterVolume = value;
- return NO_ERROR;
}
-status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted)
+void AudioFlinger::PlaybackThread::setMasterMute(bool muted)
{
- mMasterMute = muted;
- return NO_ERROR;
+ Mutex::Autolock _l(mLock);
+ setMasterMute_l(muted);
}
-float AudioFlinger::PlaybackThread::masterVolume() const
+void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value)
{
- return mMasterVolume;
-}
-
-bool AudioFlinger::PlaybackThread::masterMute() const
-{
- return mMasterMute;
-}
-
-status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value)
-{
+ Mutex::Autolock _l(mLock);
mStreamTypes[stream].volume = value;
- return NO_ERROR;
}
-status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted)
+void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted)
{
+ Mutex::Autolock _l(mLock);
mStreamTypes[stream].mute = muted;
- return NO_ERROR;
}
-float AudioFlinger::PlaybackThread::streamVolume(int stream) const
+float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const
{
+ Mutex::Autolock _l(mLock);
return mStreamTypes[stream].volume;
}
-bool AudioFlinger::PlaybackThread::streamMute(int stream) const
-{
- return mStreamTypes[stream].mute;
-}
-
// addTrack_l() must be called with ThreadBase::mLock held
status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track)
{
@@ -1690,7 +1776,7 @@
// audioConfigChanged_l() must be called with AudioFlinger::mLock held
void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
- void *param2 = 0;
+ void *param2 = NULL;
ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
@@ -1720,12 +1806,12 @@
mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common);
mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mOutput->stream->common.get_format(&mOutput->stream->common);
- mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common);
+ mFrameSize = audio_stream_frame_size(&mOutput->stream->common);
mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize;
// FIXME - Current mixer implementation only supports stereo output: Always
// Allocate a stereo buffer even if HW output is mono.
- if (mMixBuffer != NULL) delete[] mMixBuffer;
+ delete[] mMixBuffer;
mMixBuffer = new int16_t[mFrameCount * 2];
memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
@@ -1743,7 +1829,7 @@
status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames)
{
- if (halFrames == 0 || dspFrames == 0) {
+ if (halFrames == NULL || dspFrames == NULL) {
return BAD_VALUE;
}
Mutex::Autolock _l(mLock);
@@ -1786,14 +1872,14 @@
sp<Track> track = mTracks[i];
if (sessionId == track->sessionId() &&
!(track->mCblk->flags & CBLK_INVALID_MSK)) {
- return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type());
+ return AudioSystem::getStrategyForStream(track->streamType());
}
}
return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
}
-AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput()
+AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const
{
Mutex::Autolock _l(mLock);
return mOutput;
@@ -1830,13 +1916,12 @@
// ----------------------------------------------------------------------------
-AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
- : PlaybackThread(audioFlinger, output, id, device),
- mAudioMixer(0), mPrevMixerStatus(MIXER_IDLE)
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, uint32_t device, type_t type)
+ : PlaybackThread(audioFlinger, output, id, device, type),
+ mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)),
+ mPrevMixerStatus(MIXER_IDLE)
{
- mType = ThreadBase::MIXER;
- mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
-
// FIXME - Current mixer implementation only supports stereo output
if (mChannelCount == 1) {
ALOGE("Invalid audio hardware channel count");
@@ -1848,10 +1933,66 @@
delete mAudioMixer;
}
+class CpuStats {
+public:
+ void sample();
+#ifdef DEBUG_CPU_USAGE
+private:
+ ThreadCpuUsage mCpu;
+#endif
+};
+
+void CpuStats::sample() {
+#ifdef DEBUG_CPU_USAGE
+ const CentralTendencyStatistics& stats = mCpu.statistics();
+ mCpu.sampleAndEnable();
+ unsigned n = stats.n();
+ // mCpu.elapsed() is expensive, so don't call it every loop
+ if ((n & 127) == 1) {
+ long long elapsed = mCpu.elapsed();
+ if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
+ double perLoop = elapsed / (double) n;
+ double perLoop100 = perLoop * 0.01;
+ double mean = stats.mean();
+ double stddev = stats.stddev();
+ double minimum = stats.minimum();
+ double maximum = stats.maximum();
+ mCpu.resetStatistics();
+ ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
+ elapsed * .000000001, n, perLoop * .000001,
+ mean * .001,
+ stddev * .001,
+ minimum * .001,
+ maximum * .001,
+ mean / perLoop100,
+ stddev / perLoop100,
+ minimum / perLoop100,
+ maximum / perLoop100);
+ }
+ }
+#endif
+};
+
+void AudioFlinger::PlaybackThread::checkSilentMode_l()
+{
+ if (!mMasterMute) {
+ char value[PROPERTY_VALUE_MAX];
+ if (property_get("ro.audio.silent", value, "0") > 0) {
+ char *endptr;
+ unsigned long ul = strtoul(value, &endptr, 0);
+ if (*endptr == '\0' && ul != 0) {
+ ALOGD("Silence is golden");
+ // The setprop command will not allow a property to be changed after
+ // the first time it is set, so we don't have to worry about un-muting.
+ setMasterMute_l(true);
+ }
+ }
+ }
+}
+
bool AudioFlinger::MixerThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
- uint32_t mixerStatus = MIXER_IDLE;
nsecs_t standbyTime = systemTime();
size_t mixBufferSize = mFrameCount * mFrameSize;
// FIXME: Relaxed timing because of a certain device that can't meet latency
@@ -1866,45 +2007,16 @@
uint32_t sleepTime = idleSleepTime;
uint32_t sleepTimeShift = 0;
Vector< sp<EffectChain> > effectChains;
-#ifdef DEBUG_CPU_USAGE
- ThreadCpuUsage cpu;
- const CentralTendencyStatistics& stats = cpu.statistics();
-#endif
+ CpuStats cpuStats;
acquireWakeLock();
while (!exitPending())
{
-#ifdef DEBUG_CPU_USAGE
- cpu.sampleAndEnable();
- unsigned n = stats.n();
- // cpu.elapsed() is expensive, so don't call it every loop
- if ((n & 127) == 1) {
- long long elapsed = cpu.elapsed();
- if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
- double perLoop = elapsed / (double) n;
- double perLoop100 = perLoop * 0.01;
- double mean = stats.mean();
- double stddev = stats.stddev();
- double minimum = stats.minimum();
- double maximum = stats.maximum();
- cpu.resetStatistics();
- ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f",
- elapsed * .000000001, n, perLoop * .000001,
- mean * .001,
- stddev * .001,
- minimum * .001,
- maximum * .001,
- mean / perLoop100,
- stddev / perLoop100,
- minimum / perLoop100,
- maximum / perLoop100);
- }
- }
-#endif
+ cpuStats.sample();
processConfigEvents();
- mixerStatus = MIXER_IDLE;
+ mixer_state mixerStatus = MIXER_IDLE;
{ // scope for mLock
Mutex::Autolock _l(mLock);
@@ -1923,10 +2035,10 @@
const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
// put audio hardware into standby after short delay
- if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
- mSuspended) {
+ if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+ mSuspended)) {
if (!mStandby) {
- ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended);
+ ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
@@ -1940,22 +2052,15 @@
releaseWakeLock_l();
// wait until we have something to do...
- ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid());
+ ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid());
mWaitWorkCV.wait(mLock);
- ALOGV("MixerThread %p TID %d waking up\n", this, gettid());
+ ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid());
acquireWakeLock_l();
mPrevMixerStatus = MIXER_IDLE;
- if (mMasterMute == false) {
- char value[PROPERTY_VALUE_MAX];
- property_get("ro.audio.silent", value, "0");
- if (atoi(value)) {
- ALOGD("Silence is golden");
- setMasterMute(true);
- }
- }
+ checkSilentMode_l();
- standbyTime = systemTime() + kStandbyTimeInNsecs;
+ standbyTime = systemTime() + mStandbyTimeInNsecs;
sleepTime = idleSleepTime;
sleepTimeShift = 0;
continue;
@@ -1968,11 +2073,24 @@
// during mixing and effect process as the audio buffers could be deleted
// or modified if an effect is created or deleted
lockEffectChains_l(effectChains);
- }
+ }
- if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+ if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+ // obtain the presentation timestamp of the next output buffer
+ int64_t pts;
+ status_t status = INVALID_OPERATION;
+
+ if (NULL != mOutput->stream->get_next_write_timestamp) {
+ status = mOutput->stream->get_next_write_timestamp(
+ mOutput->stream, &pts);
+ }
+
+ if (status != NO_ERROR) {
+ pts = AudioBufferProvider::kInvalidPTS;
+ }
+
// mix buffers...
- mAudioMixer->process();
+ mAudioMixer->process(pts);
// increase sleep time progressively when application underrun condition clears.
// Only increase sleep time if the mixer is ready for two consecutive times to avoid
// that a steady state of alternating ready/not ready conditions keeps the sleep time
@@ -1981,7 +2099,7 @@
sleepTimeShift--;
}
sleepTime = 0;
- standbyTime = systemTime() + kStandbyTimeInNsecs;
+ standbyTime = systemTime() + mStandbyTimeInNsecs;
//TODO: delay standby when effects have a tail
} else {
// If no tracks are ready, sleep once for the duration of an output
@@ -2016,11 +2134,11 @@
}
// sleepTime == 0 means we must write to audio hardware
if (sleepTime == 0) {
- for (size_t i = 0; i < effectChains.size(); i ++) {
- effectChains[i]->process_l();
- }
- // enable changes in effect chain
- unlockEffectChains(effectChains);
+ for (size_t i = 0; i < effectChains.size(); i ++) {
+ effectChains[i]->process_l();
+ }
+ // enable changes in effect chain
+ unlockEffectChains(effectChains);
mLastWriteTime = systemTime();
mInWrite = true;
mBytesWritten += mixBufferSize;
@@ -2033,7 +2151,7 @@
nsecs_t delta = now - mLastWriteTime;
if (!mStandby && delta > maxPeriod) {
mNumDelayedWrites++;
- if ((now - lastWarning) > kWarningThrottle) {
+ if ((now - lastWarning) > kWarningThrottleNs) {
ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p",
ns2ms(delta), mNumDelayedWrites, this);
lastWarning = now;
@@ -2065,15 +2183,16 @@
releaseWakeLock();
- ALOGV("MixerThread %p exiting", this);
+ ALOGV("Thread %p type %d exiting", this, mType);
return false;
}
// prepareTracks_l() must be called with ThreadBase::mLock held
-uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
+AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l(
+ const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove)
{
- uint32_t mixerStatus = MIXER_IDLE;
+ mixer_state mixerStatus = MIXER_IDLE;
// find out which tracks need to be processed
size_t count = activeTracks.size();
size_t mixedTracks = 0;
@@ -2098,12 +2217,13 @@
sp<Track> t = activeTracks[i].promote();
if (t == 0) continue;
+ // this const just means the local variable doesn't change
Track* const track = t.get();
audio_track_cblk_t* cblk = track->cblk();
// The first time a track is added we wait
// for all its buffers to be filled before processing it
- mAudioMixer->setActiveTrack(track->name());
+ int name = track->name();
// make sure that we have enough frames to mix one full buffer.
// enforce this condition only once to enable draining the buffer in case the client
// app does not call stop() and relies on underrun to stop:
@@ -2123,13 +2243,13 @@
// the minimum track buffer size is normally twice the number of frames necessary
// to fill one buffer and the resampler should not leave more than one buffer worth
// of unreleased frames after each pass, but just in case...
- LOG_ASSERT(minFrames <= cblk->frameCount);
+ ALOG_ASSERT(minFrames <= cblk->frameCount);
}
}
- if ((cblk->framesReady() >= minFrames) && track->isReady() &&
+ if ((track->framesReady() >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
- //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this);
+ //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
mixedTracks++;
@@ -2142,8 +2262,8 @@
if (chain != 0) {
tracksWithEffect++;
} else {
- ALOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d",
- track->name(), track->sessionId());
+ ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
+ name, track->sessionId());
}
}
@@ -2156,7 +2276,7 @@
track->mState = TrackBase::ACTIVE;
param = AudioMixer::RAMP_VOLUME;
}
- mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
+ mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL);
} else if (cblk->server != 0) {
// If the track is stopped before the first frame was mixed,
// do not apply ramp
@@ -2166,7 +2286,7 @@
// compute volume for this track
uint32_t vl, vr, va;
if (track->isMuted() || track->isPausing() ||
- mStreamTypes[track->type()].mute) {
+ mStreamTypes[track->streamType()].mute) {
vl = vr = va = 0;
if (track->isPausing()) {
track->setPaused();
@@ -2174,12 +2294,33 @@
} else {
// read original volumes with volume control
- float typeVolume = mStreamTypes[track->type()].volume;
+ float typeVolume = mStreamTypes[track->streamType()].volume;
float v = masterVolume * typeVolume;
- vl = (uint32_t)(v * cblk->volume[0]) << 12;
- vr = (uint32_t)(v * cblk->volume[1]) << 12;
+ uint32_t vlr = cblk->getVolumeLR();
+ vl = vlr & 0xFFFF;
+ vr = vlr >> 16;
+ // track volumes come from shared memory, so can't be trusted and must be clamped
+ if (vl > MAX_GAIN_INT) {
+ ALOGV("Track left volume out of range: %04X", vl);
+ vl = MAX_GAIN_INT;
+ }
+ if (vr > MAX_GAIN_INT) {
+ ALOGV("Track right volume out of range: %04X", vr);
+ vr = MAX_GAIN_INT;
+ }
+ // now apply the master volume and stream type volume
+ vl = (uint32_t)(v * vl) << 12;
+ vr = (uint32_t)(v * vr) << 12;
+ // assuming master volume and stream type volume each go up to 1.0,
+ // vl and vr are now in 8.24 format
- va = (uint32_t)(v * cblk->sendLevel);
+ uint16_t sendLevel = cblk->getSendLevel_U4_12();
+ // send level comes from shared memory and so may be corrupt
+ if (sendLevel > MAX_GAIN_INT) {
+ ALOGV("Track send level out of range: %04X", sendLevel);
+ sendLevel = MAX_GAIN_INT;
+ }
+ va = (uint32_t)(v * sendLevel);
}
// Delegate volume control to effect in track effect chain if needed
if (chain != 0 && chain->setVolume_l(&vl, &vr)) {
@@ -2196,38 +2337,40 @@
}
// Convert volumes from 8.24 to 4.12 format
- int16_t left, right, aux;
- uint32_t v_clamped = (vl + (1 << 11)) >> 12;
- if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
- left = int16_t(v_clamped);
- v_clamped = (vr + (1 << 11)) >> 12;
- if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT;
- right = int16_t(v_clamped);
+ // This additional clamping is needed in case chain->setVolume_l() overshot
+ vl = (vl + (1 << 11)) >> 12;
+ if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT;
+ vr = (vr + (1 << 11)) >> 12;
+ if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT;
- if (va > MAX_GAIN_INT) va = MAX_GAIN_INT;
- aux = int16_t(va);
+ if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for -
// XXX: these things DON'T need to be done each time
- mAudioMixer->setBufferProvider(track);
- mAudioMixer->enable(AudioMixer::MIXING);
+ mAudioMixer->setBufferProvider(name, track);
+ mAudioMixer->enable(name);
- mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left);
- mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right);
- mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl);
+ mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr);
+ mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va);
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::FORMAT, (void *)track->format());
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::CHANNEL_MASK, (void *)track->channelMask());
mAudioMixer->setParameter(
+ name,
AudioMixer::RESAMPLE,
AudioMixer::SAMPLE_RATE,
(void *)(cblk->sampleRate));
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer());
mAudioMixer->setParameter(
+ name,
AudioMixer::TRACK,
AudioMixer::AUX_BUFFER, (void *)track->auxBuffer());
@@ -2241,7 +2384,7 @@
mixerStatus = MIXER_TRACKS_READY;
}
} else {
- //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this);
+ //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
if (track->isStopped()) {
track->reset();
}
@@ -2253,7 +2396,7 @@
// No buffers for this track. Give it a few chances to
// fill a buffer, then remove it from active list.
if (--(track->mRetryCount) <= 0) {
- ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this);
+ ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this);
tracksToRemove->add(track);
// indicate to client process that the track was disabled because of underrun
android_atomic_or(CBLK_DISABLED_ON, &cblk->flags);
@@ -2265,13 +2408,13 @@
mixerStatus = MIXER_TRACKS_ENABLED;
}
}
- mAudioMixer->disable(AudioMixer::MIXING);
+ mAudioMixer->disable(name);
}
}
// remove all the tracks that need to be...
count = tracksToRemove->size();
- if (UNLIKELY(count)) {
+ if (CC_UNLIKELY(count)) {
for (size_t i=0 ; i<count ; i++) {
const sp<Track>& track = tracksToRemove->itemAt(i);
mActiveTracks.remove(track);
@@ -2299,7 +2442,7 @@
return mixerStatus;
}
-void AudioFlinger::MixerThread::invalidateTracks(int streamType)
+void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType)
{
ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d",
this, streamType, mTracks.size());
@@ -2308,14 +2451,14 @@
size_t size = mTracks.size();
for (size_t i = 0; i < size; i++) {
sp<Track> t = mTracks[i];
- if (t->type() == streamType) {
+ if (t->streamType() == streamType) {
android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags);
t->mCblk->cv.signal();
}
}
}
-void AudioFlinger::PlaybackThread::setStreamValid(int streamType, bool valid)
+void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid)
{
ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d",
this, streamType, valid);
@@ -2352,7 +2495,7 @@
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- if (value != AUDIO_FORMAT_PCM_16_BIT) {
+ if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) {
status = BAD_VALUE;
} else {
reconfig = true;
@@ -2367,7 +2510,7 @@
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
- // size depends on frame count and correct behavior would not be garantied
+ // size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (!mTracks.isEmpty()) {
status = INVALID_OPERATION;
@@ -2417,6 +2560,8 @@
}
if (status == NO_ERROR && reconfig) {
delete mAudioMixer;
+ // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't)
+ mAudioMixer = NULL;
readOutputParameters();
mAudioMixer = new AudioMixer(mFrameCount, mSampleRate);
for (size_t i = 0; i < mTracks.size() ; i++) {
@@ -2438,7 +2583,7 @@
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
- mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
@@ -2468,39 +2613,18 @@
}
// ----------------------------------------------------------------------------
-AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device)
- : PlaybackThread(audioFlinger, output, id, device)
+AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger,
+ AudioStreamOut* output, audio_io_handle_t id, uint32_t device)
+ : PlaybackThread(audioFlinger, output, id, device, DIRECT)
+ // mLeftVolFloat, mRightVolFloat
+ // mLeftVolShort, mRightVolShort
{
- mType = ThreadBase::DIRECT;
}
AudioFlinger::DirectOutputThread::~DirectOutputThread()
{
}
-
-static inline int16_t clamp16(int32_t sample)
-{
- if ((sample>>15) ^ (sample>>31))
- sample = 0x7FFF ^ (sample>>31);
- return sample;
-}
-
-static inline
-int32_t mul(int16_t in, int16_t v)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smulbb %[out], %[in], %[v] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v)
- : );
- return out;
-#else
- return in * int32_t(v);
-#endif
-}
-
void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp)
{
// Do not apply volume on compressed audio
@@ -2577,11 +2701,9 @@
bool AudioFlinger::DirectOutputThread::threadLoop()
{
- uint32_t mixerStatus = MIXER_IDLE;
sp<Track> trackToRemove;
sp<Track> activeTrack;
nsecs_t standbyTime = systemTime();
- int8_t *curBuf;
size_t mixBufferSize = mFrameCount*mFrameSize;
uint32_t activeSleepTime = activeSleepTimeUs();
uint32_t idleSleepTime = idleSleepTimeUs();
@@ -2601,8 +2723,7 @@
processConfigEvents();
- mixerStatus = MIXER_IDLE;
-
+ mixer_state mixerStatus = MIXER_IDLE;
{ // scope for the mLock
Mutex::Autolock _l(mLock);
@@ -2615,11 +2736,11 @@
}
// put audio hardware into standby after short delay
- if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
- mSuspended) {
+ if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) ||
+ mSuspended)) {
// wait until we have something to do...
if (!mStandby) {
- ALOGV("Audio hardware entering standby, mixer %p\n", this);
+ ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended);
mOutput->stream->common.standby(&mOutput->stream->common);
mStandby = true;
mBytesWritten = 0;
@@ -2632,19 +2753,12 @@
if (exitPending()) break;
releaseWakeLock_l();
- ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid());
+ ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid());
mWaitWorkCV.wait(mLock);
- ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid());
+ ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid());
acquireWakeLock_l();
- if (mMasterMute == false) {
- char value[PROPERTY_VALUE_MAX];
- property_get("ro.audio.silent", value, "0");
- if (atoi(value)) {
- ALOGD("Silence is golden");
- setMasterMute(true);
- }
- }
+ checkSilentMode_l();
standbyTime = systemTime() + standbyDelay;
sleepTime = idleSleepTime;
@@ -2685,18 +2799,19 @@
// compute volume for this track
float left, right;
if (track->isMuted() || mMasterMute || track->isPausing() ||
- mStreamTypes[track->type()].mute) {
+ mStreamTypes[track->streamType()].mute) {
left = right = 0;
if (track->isPausing()) {
track->setPaused();
}
} else {
- float typeVolume = mStreamTypes[track->type()].volume;
+ float typeVolume = mStreamTypes[track->streamType()].volume;
float v = mMasterVolume * typeVolume;
- float v_clamped = v * cblk->volume[0];
+ uint32_t vlr = cblk->getVolumeLR();
+ float v_clamped = v * (vlr & 0xFFFF);
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
left = v_clamped/MAX_GAIN;
- v_clamped = v * cblk->volume[1];
+ v_clamped = v * (vlr >> 16);
if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
right = v_clamped/MAX_GAIN;
}
@@ -2766,7 +2881,7 @@
}
// remove all the tracks that need to be...
- if (UNLIKELY(trackToRemove != 0)) {
+ if (CC_UNLIKELY(trackToRemove != 0)) {
mActiveTracks.remove(trackToRemove);
if (!effectChains.isEmpty()) {
ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(),
@@ -2781,15 +2896,16 @@
lockEffectChains_l(effectChains);
}
- if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+ if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
AudioBufferProvider::Buffer buffer;
size_t frameCount = mFrameCount;
- curBuf = (int8_t *)mMixBuffer;
+ int8_t *curBuf = (int8_t *)mMixBuffer;
// output audio to hardware
while (frameCount) {
buffer.frameCount = frameCount;
- activeTrack->getNextBuffer(&buffer);
- if (UNLIKELY(buffer.raw == 0)) {
+ activeTrack->getNextBuffer(&buffer,
+ AudioBufferProvider::kInvalidPTS);
+ if (CC_UNLIKELY(buffer.raw == NULL)) {
memset(curBuf, 0, frameCount * mFrameSize);
break;
}
@@ -2856,7 +2972,7 @@
releaseWakeLock();
- ALOGV("DirectOutputThread %p exiting", this);
+ ALOGV("Thread %p type %d exiting", this, mType);
return false;
}
@@ -2914,7 +3030,7 @@
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
- mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
@@ -2955,10 +3071,11 @@
// ----------------------------------------------------------------------------
-AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id)
- : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX)
+AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
+ AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
+ : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING),
+ mWaitTimeMs(UINT_MAX)
{
- mType = ThreadBase::DUPLICATING;
addOutputTrack(mainThread);
}
@@ -2967,13 +3084,11 @@
for (size_t i = 0; i < mOutputTracks.size(); i++) {
mOutputTracks[i]->destroy();
}
- mOutputTracks.clear();
}
bool AudioFlinger::DuplicatingThread::threadLoop()
{
Vector< sp<Track> > tracksToRemove;
- uint32_t mixerStatus = MIXER_IDLE;
nsecs_t standbyTime = systemTime();
size_t mixBufferSize = mFrameCount*mFrameSize;
SortedVector< sp<OutputTrack> > outputTracks;
@@ -2989,7 +3104,7 @@
{
processConfigEvents();
- mixerStatus = MIXER_IDLE;
+ mixer_state mixerStatus = MIXER_IDLE;
{ // scope for the mLock
Mutex::Autolock _l(mLock);
@@ -3008,8 +3123,8 @@
}
// put audio hardware into standby after short delay
- if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
- mSuspended) {
+ if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) ||
+ mSuspended)) {
if (!mStandby) {
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->stop();
@@ -3026,22 +3141,15 @@
if (exitPending()) break;
releaseWakeLock_l();
- ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid());
+ // wait until we have something to do...
+ ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid());
mWaitWorkCV.wait(mLock);
- ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid());
+ ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid());
acquireWakeLock_l();
- mPrevMixerStatus = MIXER_IDLE;
- if (mMasterMute == false) {
- char value[PROPERTY_VALUE_MAX];
- property_get("ro.audio.silent", value, "0");
- if (atoi(value)) {
- ALOGD("Silence is golden");
- setMasterMute(true);
- }
- }
+ checkSilentMode_l();
- standbyTime = systemTime() + kStandbyTimeInNsecs;
+ standbyTime = systemTime() + mStandbyTimeInNsecs;
sleepTime = idleSleepTime;
continue;
}
@@ -3055,10 +3163,10 @@
lockEffectChains_l(effectChains);
}
- if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
+ if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) {
// mix buffers...
if (outputsReady(outputTracks)) {
- mAudioMixer->process();
+ mAudioMixer->process(AudioBufferProvider::kInvalidPTS);
} else {
memset(mMixBuffer, 0, mixBufferSize);
}
@@ -3095,7 +3203,7 @@
// enable changes in effect chain
unlockEffectChains(effectChains);
- standbyTime = systemTime() + kStandbyTimeInNsecs;
+ standbyTime = systemTime() + mStandbyTimeInNsecs;
for (size_t i = 0; i < outputTracks.size(); i++) {
outputTracks[i]->write(mMixBuffer, writeFrames);
}
@@ -3120,13 +3228,15 @@
releaseWakeLock();
+ ALOGV("Thread %p type %d exiting", this, mType);
return false;
}
void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread)
{
+ // FIXME explain this formula
int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate();
- OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread,
+ OutputTrack *outputTrack = new OutputTrack(thread,
this,
mSampleRate,
mFormat,
@@ -3144,7 +3254,7 @@
{
Mutex::Autolock _l(mLock);
for (size_t i = 0; i < mOutputTracks.size(); i++) {
- if (mOutputTracks[i]->thread() == (ThreadBase *)thread) {
+ if (mOutputTracks[i]->thread() == thread) {
mOutputTracks[i]->destroy();
mOutputTracks.removeAt(i);
updateWaitTime();
@@ -3159,7 +3269,7 @@
mWaitTimeMs = UINT_MAX;
for (size_t i = 0; i < mOutputTracks.size(); i++) {
sp<ThreadBase> strong = mOutputTracks[i]->thread().promote();
- if (strong != NULL) {
+ if (strong != 0) {
uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate();
if (waitTimeMs < mWaitTimeMs) {
mWaitTimeMs = waitTimeMs;
@@ -3195,25 +3305,27 @@
// TrackBase constructor must be called with AudioFlinger::mLock held
AudioFlinger::ThreadBase::TrackBase::TrackBase(
- const wp<ThreadBase>& thread,
+ ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId)
: RefBase(),
mThread(thread),
mClient(client),
- mCblk(0),
+ mCblk(NULL),
+ // mBuffer
+ // mBufferEnd
mFrameCount(0),
mState(IDLE),
- mClientTid(-1),
mFormat(format),
- mFlags(flags & ~SYSTEM_FLAGS_MASK),
+ mStepServerFailed(false),
mSessionId(sessionId)
+ // mChannelCount
+ // mChannelMask
{
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
@@ -3229,7 +3341,7 @@
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
- if (mCblk) { // construct the shared structure in-place.
+ if (mCblk != NULL) { // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
@@ -3254,7 +3366,7 @@
}
} else {
mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
- if (mCblk) { // construct the shared structure in-place.
+ // construct the shared structure in-place.
new(mCblk) audio_track_cblk_t();
// clear all buffers
mCblk->frameCount = frameCount;
@@ -3267,28 +3379,32 @@
// written to buffer (other flags are cleared)
mCblk->flags = CBLK_UNDERRUN_ON;
mBufferEnd = (uint8_t *)mBuffer + bufferSize;
- }
}
}
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
{
- if (mCblk) {
- mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
- if (mClient == NULL) {
+ if (mCblk != NULL) {
+ if (mClient == 0) {
delete mCblk;
+ } else {
+ mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
}
}
- mCblkMemory.clear(); // and free the shared memory
- if (mClient != NULL) {
+ mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
+ if (mClient != 0) {
+ // Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
+ // If the client's reference count drops to zero, the associated destructor
+ // must run with AudioFlinger lock held. Thus the explicit clear() rather than
+ // relying on the automatic clear() at end of scope.
mClient.clear();
}
}
void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
{
- buffer->raw = 0;
+ buffer->raw = NULL;
mFrameCount = buffer->frameCount;
step();
buffer->frameCount = 0;
@@ -3301,7 +3417,7 @@
result = cblk->stepServer(mFrameCount);
if (!result) {
ALOGV("stepServer failed acquiring cblk mutex");
- mFlags |= STEPSERVER_FAILED;
+ mStepServerFailed = true;
}
return result;
}
@@ -3313,40 +3429,28 @@
cblk->server = 0;
cblk->userBase = 0;
cblk->serverBase = 0;
- mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
+ mStepServerFailed = false;
ALOGV("TrackBase::reset");
}
-sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const
-{
- return mCblkMemory;
-}
-
int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
return (int)mCblk->sampleRate;
}
-int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
- return (const int)mChannelCount;
-}
-
-uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const {
- return mChannelMask;
-}
-
void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
audio_track_cblk_t* cblk = this->cblk();
- int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize;
- int8_t *bufferEnd = bufferStart + frames * cblk->frameSize;
+ size_t frameSize = cblk->frameSize;
+ int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize;
+ int8_t *bufferEnd = bufferStart + frames * frameSize;
// Check validity of returned pointer in case the track control block would have been corrupted.
if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd ||
- ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
+ ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) {
ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
server %d, serverBase %d, user %d, userBase %d",
bufferStart, bufferEnd, mBuffer, mBufferEnd,
cblk->server, cblk->serverBase, cblk->user, cblk->userBase);
- return 0;
+ return NULL;
}
return bufferStart;
@@ -3356,32 +3460,28 @@
// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held
AudioFlinger::PlaybackThread::Track::Track(
- const wp<ThreadBase>& thread,
+ PlaybackThread *thread,
const sp<Client>& client,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId)
- : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId),
+ : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId),
mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL),
mAuxEffectId(0), mHasVolumeController(false)
{
if (mCblk != NULL) {
- sp<ThreadBase> baseThread = thread.promote();
- if (baseThread != 0) {
- PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get();
- mName = playbackThread->getTrackName_l();
- mMainBuffer = playbackThread->mixBuffer();
+ if (thread != NULL) {
+ mName = thread->getTrackName_l();
+ mMainBuffer = thread->mixBuffer();
}
- ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
if (mName < 0) {
ALOGE("no more track names available");
}
- mVolume[0] = 1.0f;
- mVolume[1] = 1.0f;
mStreamType = streamType;
// NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
@@ -3403,7 +3503,7 @@
{
// NOTE: destroyTrack_l() can remove a strong reference to this Track
// by removing it from mTracks vector, so there is a risk that this Tracks's
- // desctructor is called. As the destructor needs to lock mLock,
+ // destructor is called. As the destructor needs to lock mLock,
// we must acquire a strong reference on this Track before locking mLock
// here so that the destructor is called only when exiting this function.
// On the other hand, as long as Track::destroy() is only called by
@@ -3415,9 +3515,7 @@
if (thread != 0) {
if (!isOutputTrack()) {
if (mState == ACTIVE || mState == RESUMING) {
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
// to track the speaker usage
addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
@@ -3433,9 +3531,10 @@
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
{
+ uint32_t vlr = mCblk->getVolumeLR();
snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n",
mName - AudioMixer::TRACK0,
- (mClient == NULL) ? getpid() : mClient->pid(),
+ (mClient == 0) ? getpid_cached : mClient->pid(),
mStreamType,
mFormat,
mChannelMask,
@@ -3445,30 +3544,31 @@
mMute,
mFillingUpStatus,
mCblk->sampleRate,
- mCblk->volume[0],
- mCblk->volume[1],
+ vlr & 0xFFFF,
+ vlr >> 16,
mCblk->server,
mCblk->user,
(int)mMainBuffer,
(int)mAuxBuffer);
}
-status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
+ AudioBufferProvider::Buffer* buffer, int64_t pts)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesReady;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
- if (mFlags & TrackBase::STEPSERVER_FAILED) {
+ if (mStepServerFailed) {
if (!step()) goto getNextBuffer_exit;
ALOGV("stepServer recovered");
- mFlags &= ~TrackBase::STEPSERVER_FAILED;
+ mStepServerFailed = false;
}
framesReady = cblk->framesReady();
- if (LIKELY(framesReady)) {
+ if (CC_LIKELY(framesReady)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
@@ -3481,23 +3581,27 @@
}
buffer->raw = getBuffer(s, framesReq);
- if (buffer->raw == 0) goto getNextBuffer_exit;
+ if (buffer->raw == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
return NOT_ENOUGH_DATA;
}
+uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{
+ return mCblk->framesReady();
+}
+
bool AudioFlinger::PlaybackThread::Track::isReady() const {
if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true;
- if (mCblk->framesReady() >= mCblk->frameCount ||
+ if (framesReady() >= mCblk->frameCount ||
(mCblk->flags & CBLK_FORCEREADY_MSK)) {
mFillingUpStatus = FS_FILLED;
android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags);
@@ -3506,15 +3610,15 @@
return false;
}
-status_t AudioFlinger::PlaybackThread::Track::start()
+status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid)
{
status_t status = NO_ERROR;
- ALOGV("start(%d), calling thread %d session %d",
- mName, IPCThreadState::self()->getCallingPid(), mSessionId);
+ ALOGV("start(%d), calling pid %d session %d tid %d",
+ mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
- int state = mState;
+ track_state state = mState;
// here the track could be either new, or restarted
// in both cases "unstop" the track
if (mState == PAUSED) {
@@ -3527,9 +3631,7 @@
if (!isOutputTrack() && state != ACTIVE && state != RESUMING) {
thread->mLock.unlock();
- status = AudioSystem::startOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
@@ -3551,11 +3653,11 @@
void AudioFlinger::PlaybackThread::Track::stop()
{
- ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
- int state = mState;
+ track_state state = mState;
if (mState > STOPPED) {
mState = STOPPED;
// If the track is not active (PAUSED and buffers full), flush buffers
@@ -3567,9 +3669,7 @@
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
@@ -3580,7 +3680,7 @@
void AudioFlinger::PlaybackThread::Track::pause()
{
- ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid());
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
Mutex::Autolock _l(thread->mLock);
@@ -3589,9 +3689,7 @@
ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get());
if (!isOutputTrack()) {
thread->mLock.unlock();
- AudioSystem::stopOutput(thread->id(),
- (audio_stream_type_t)mStreamType,
- mSessionId);
+ AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId);
thread->mLock.lock();
// to track the speaker usage
@@ -3643,12 +3741,6 @@
mMute = muted;
}
-void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right)
-{
- mVolume[0] = left;
- mVolume[1] = right;
-}
-
status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
{
status_t status = DEAD_OBJECT;
@@ -3666,20 +3758,410 @@
mAuxBuffer = buffer;
}
+// timed audio tracks
+
+sp<AudioFlinger::PlaybackThread::TimedTrack>
+AudioFlinger::PlaybackThread::TimedTrack::create(
+ PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId) {
+ if (!client->reserveTimedTrack())
+ return NULL;
+
+ sp<TimedTrack> track = new TimedTrack(
+ thread, client, streamType, sampleRate, format, channelMask, frameCount,
+ sharedBuffer, sessionId);
+
+ if (track == NULL) {
+ client->releaseTimedTrack();
+ return NULL;
+ }
+
+ return track;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedTrack(
+ PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId)
+ : Track(thread, client, streamType, sampleRate, format, channelMask,
+ frameCount, sharedBuffer, sessionId),
+ mTimedSilenceBuffer(NULL),
+ mTimedSilenceBufferSize(0),
+ mTimedAudioOutputOnTime(false),
+ mMediaTimeTransformValid(false)
+{
+ LocalClock lc;
+ mLocalTimeFreq = lc.getLocalFreq();
+
+ mLocalTimeToSampleTransform.a_zero = 0;
+ mLocalTimeToSampleTransform.b_zero = 0;
+ mLocalTimeToSampleTransform.a_to_b_numer = sampleRate;
+ mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq;
+ LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer,
+ &mLocalTimeToSampleTransform.a_to_b_denom);
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() {
+ mClient->releaseTimedTrack();
+ delete [] mTimedSilenceBuffer;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer(
+ size_t size, sp<IMemory>* buffer) {
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ trimTimedBufferQueue_l();
+
+ // lazily initialize the shared memory heap for timed buffers
+ if (mTimedMemoryDealer == NULL) {
+ const int kTimedBufferHeapSize = 512 << 10;
+
+ mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize,
+ "AudioFlingerTimed");
+ if (mTimedMemoryDealer == NULL)
+ return NO_MEMORY;
+ }
+
+ sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size);
+ if (newBuffer == NULL) {
+ newBuffer = mTimedMemoryDealer->allocate(size);
+ if (newBuffer == NULL)
+ return NO_MEMORY;
+ }
+
+ *buffer = newBuffer;
+ return NO_ERROR;
+}
+
+// caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() {
+ int64_t mediaTimeNow;
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+ if (!mMediaTimeTransformValid)
+ return;
+
+ int64_t targetTimeNow;
+ status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME)
+ ? mCCHelper.getCommonTime(&targetTimeNow)
+ : mCCHelper.getLocalTime(&targetTimeNow);
+
+ if (OK != res)
+ return;
+
+ if (!mMediaTimeTransform.doReverseTransform(targetTimeNow,
+ &mediaTimeNow)) {
+ return;
+ }
+ }
+
+ size_t trimIndex;
+ for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) {
+ if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow)
+ break;
+ }
+
+ if (trimIndex) {
+ mTimedBufferQueue.removeItemsAt(0, trimIndex);
+ }
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer(
+ const sp<IMemory>& buffer, int64_t pts) {
+
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+ if (!mMediaTimeTransformValid)
+ return INVALID_OPERATION;
+ }
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ mTimedBufferQueue.add(TimedBuffer(buffer, pts));
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform(
+ const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) {
+
+ ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__,
+ xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom,
+ target);
+
+ if (!(target == TimedAudioTrack::LOCAL_TIME ||
+ target == TimedAudioTrack::COMMON_TIME)) {
+ return BAD_VALUE;
+ }
+
+ Mutex::Autolock lock(mMediaTimeTransformLock);
+ mMediaTimeTransform = xform;
+ mMediaTimeTransformTarget = target;
+ mMediaTimeTransformValid = true;
+
+ return NO_ERROR;
+}
+
+#define min(a, b) ((a) < (b) ? (a) : (b))
+
+// implementation of getNextBuffer for tracks whose buffers have timestamps
+status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer(
+ AudioBufferProvider::Buffer* buffer, int64_t pts)
+{
+ if (pts == AudioBufferProvider::kInvalidPTS) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return INVALID_OPERATION;
+ }
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ while (true) {
+
+ // if we have no timed buffers, then fail
+ if (mTimedBufferQueue.isEmpty()) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+ }
+
+ TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+ // calculate the PTS of the head of the timed buffer queue expressed in
+ // local time
+ int64_t headLocalPTS;
+ {
+ Mutex::Autolock mttLock(mMediaTimeTransformLock);
+
+ assert(mMediaTimeTransformValid);
+
+ if (mMediaTimeTransform.a_to_b_denom == 0) {
+ // the transform represents a pause, so yield silence
+ timedYieldSilence(buffer->frameCount, buffer);
+ return NO_ERROR;
+ }
+
+ int64_t transformedPTS;
+ if (!mMediaTimeTransform.doForwardTransform(head.pts(),
+ &transformedPTS)) {
+ // the transform failed. this shouldn't happen, but if it does
+ // then just drop this buffer
+ ALOGW("timedGetNextBuffer transform failed");
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ mTimedBufferQueue.removeAt(0);
+ return NO_ERROR;
+ }
+
+ if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) {
+ if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS,
+ &headLocalPTS)) {
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return INVALID_OPERATION;
+ }
+ } else {
+ headLocalPTS = transformedPTS;
+ }
+ }
+
+ // adjust the head buffer's PTS to reflect the portion of the head buffer
+ // that has already been consumed
+ int64_t effectivePTS = headLocalPTS +
+ ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate());
+
+ // Calculate the delta in samples between the head of the input buffer
+ // queue and the start of the next output buffer that will be written.
+ // If the transformation fails because of over or underflow, it means
+ // that the sample's position in the output stream is so far out of
+ // whack that it should just be dropped.
+ int64_t sampleDelta;
+ if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) {
+ ALOGV("*** head buffer is too far from PTS: dropped buffer");
+ mTimedBufferQueue.removeAt(0);
+ continue;
+ }
+ if (!mLocalTimeToSampleTransform.doForwardTransform(
+ (effectivePTS - pts) << 32, &sampleDelta)) {
+ ALOGV("*** too late during sample rate transform: dropped buffer");
+ mTimedBufferQueue.removeAt(0);
+ continue;
+ }
+
+ ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]",
+ __PRETTY_FUNCTION__, head.pts(), head.position(), pts,
+ static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)),
+ static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF));
+
+ // if the delta between the ideal placement for the next input sample and
+ // the current output position is within this threshold, then we will
+ // concatenate the next input samples to the previous output
+ const int64_t kSampleContinuityThreshold =
+ (static_cast<int64_t>(sampleRate()) << 32) / 10;
+
+ // if this is the first buffer of audio that we're emitting from this track
+ // then it should be almost exactly on time.
+ const int64_t kSampleStartupThreshold = 1LL << 32;
+
+ if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) ||
+ (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) {
+ // the next input is close enough to being on time, so concatenate it
+ // with the last output
+ timedYieldSamples(buffer);
+
+ ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+ return NO_ERROR;
+ } else if (sampleDelta > 0) {
+ // the gap between the current output position and the proper start of
+ // the next input sample is too big, so fill it with silence
+ uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32;
+
+ timedYieldSilence(framesUntilNextInput, buffer);
+ ALOGV("*** silence: frameCount=%u", buffer->frameCount);
+ return NO_ERROR;
+ } else {
+ // the next input sample is late
+ uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32));
+ size_t onTimeSamplePosition =
+ head.position() + lateFrames * mCblk->frameSize;
+
+ if (onTimeSamplePosition > head.buffer()->size()) {
+ // all the remaining samples in the head are too late, so
+ // drop it and move on
+ ALOGV("*** too late: dropped buffer");
+ mTimedBufferQueue.removeAt(0);
+ continue;
+ } else {
+ // skip over the late samples
+ head.setPosition(onTimeSamplePosition);
+
+ // yield the available samples
+ timedYieldSamples(buffer);
+
+ ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount);
+ return NO_ERROR;
+ }
+ }
+ }
+}
+
+// Yield samples from the timed buffer queue head up to the given output
+// buffer's capacity.
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples(
+ AudioBufferProvider::Buffer* buffer) {
+
+ const TimedBuffer& head = mTimedBufferQueue[0];
+
+ buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) +
+ head.position());
+
+ uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) /
+ mCblk->frameSize);
+ size_t framesRequested = buffer->frameCount;
+ buffer->frameCount = min(framesLeftInHead, framesRequested);
+
+ mTimedAudioOutputOnTime = true;
+}
+
+// Yield samples of silence up to the given output buffer's capacity
+//
+// Caller must hold mTimedBufferQueueLock
+void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence(
+ uint32_t numFrames, AudioBufferProvider::Buffer* buffer) {
+
+ // lazily allocate a buffer filled with silence
+ if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) {
+ delete [] mTimedSilenceBuffer;
+ mTimedSilenceBufferSize = numFrames * mCblk->frameSize;
+ mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize];
+ memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize);
+ }
+
+ buffer->raw = mTimedSilenceBuffer;
+ size_t framesRequested = buffer->frameCount;
+ buffer->frameCount = min(numFrames, framesRequested);
+
+ mTimedAudioOutputOnTime = false;
+}
+
+void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer(
+ AudioBufferProvider::Buffer* buffer) {
+
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ // If the buffer which was just released is part of the buffer at the head
+ // of the queue, be sure to update the amt of the buffer which has been
+ // consumed. If the buffer being returned is not part of the head of the
+ // queue, its either because the buffer is part of the silence buffer, or
+ // because the head of the timed queue was trimmed after the mixer called
+ // getNextBuffer but before the mixer called releaseBuffer.
+ if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) {
+ TimedBuffer& head = mTimedBufferQueue.editItemAt(0);
+
+ void* start = head.buffer()->pointer();
+ void* end = (char *) head.buffer()->pointer() + head.buffer()->size();
+
+ if ((buffer->raw >= start) && (buffer->raw <= end)) {
+ head.setPosition(head.position() +
+ (buffer->frameCount * mCblk->frameSize));
+ if (static_cast<size_t>(head.position()) >= head.buffer()->size()) {
+ mTimedBufferQueue.removeAt(0);
+ }
+ }
+ }
+
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+}
+
+uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const {
+ Mutex::Autolock _l(mTimedBufferQueueLock);
+
+ uint32_t frames = 0;
+ for (size_t i = 0; i < mTimedBufferQueue.size(); i++) {
+ const TimedBuffer& tb = mTimedBufferQueue[i];
+ frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize;
+ }
+
+ return frames;
+}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer()
+ : mPTS(0), mPosition(0) {}
+
+AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer(
+ const sp<IMemory>& buffer, int64_t pts)
+ : mBuffer(buffer), mPTS(pts), mPosition(0) {}
+
// ----------------------------------------------------------------------------
// RecordTrack constructor must be called with AudioFlinger::mLock held
AudioFlinger::RecordThread::RecordTrack::RecordTrack(
- const wp<ThreadBase>& thread,
+ RecordThread *thread,
const sp<Client>& client,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
int sessionId)
: TrackBase(thread, client, sampleRate, format,
- channelMask, frameCount, flags, 0, sessionId),
+ channelMask, frameCount, 0 /*sharedBuffer*/, sessionId),
mOverflow(false)
{
if (mCblk != NULL) {
@@ -3702,22 +4184,22 @@
}
}
-status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
uint32_t framesReq = buffer->frameCount;
// Check if last stepServer failed, try to step now
- if (mFlags & TrackBase::STEPSERVER_FAILED) {
+ if (mStepServerFailed) {
if (!step()) goto getNextBuffer_exit;
ALOGV("stepServer recovered");
- mFlags &= ~TrackBase::STEPSERVER_FAILED;
+ mStepServerFailed = false;
}
framesAvail = cblk->framesAvailable_l();
- if (LIKELY(framesAvail)) {
+ if (CC_LIKELY(framesAvail)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
@@ -3729,24 +4211,24 @@
}
buffer->raw = getBuffer(s, framesReq);
- if (buffer->raw == 0) goto getNextBuffer_exit;
+ if (buffer->raw == NULL) goto getNextBuffer_exit;
buffer->frameCount = framesReq;
return NO_ERROR;
}
getNextBuffer_exit:
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
-status_t AudioFlinger::RecordThread::RecordTrack::start()
+status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid)
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
RecordThread *recordThread = (RecordThread *)thread.get();
- return recordThread->start(this);
+ return recordThread->start(this, tid);
} else {
return BAD_VALUE;
}
@@ -3768,7 +4250,7 @@
void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
{
snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n",
- (mClient == NULL) ? getpid() : mClient->pid(),
+ (mClient == 0) ? getpid_cached : mClient->pid(),
mFormat,
mChannelMask,
mSessionId,
@@ -3783,21 +4265,19 @@
// ----------------------------------------------------------------------------
AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
- const wp<ThreadBase>& thread,
+ PlaybackThread *playbackThread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount)
- : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
+ : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0),
mActive(false), mSourceThread(sourceThread)
{
- PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get();
if (mCblk != NULL) {
mCblk->flags |= CBLK_DIRECTION_OUT;
mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
- mCblk->volume[0] = mCblk->volume[1] = 0x1000;
mOutBuffer.frameCount = 0;
playbackThread->mTracks.add(this);
ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \
@@ -3814,9 +4294,9 @@
clearBufferQueue();
}
-status_t AudioFlinger::PlaybackThread::OutputTrack::start()
+status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid)
{
- status_t status = Track::start();
+ status_t status = Track::start(tid);
if (status != NO_ERROR) {
return status;
}
@@ -3846,7 +4326,7 @@
uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs();
if (!mActive && frames != 0) {
- start();
+ start(0);
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
MixerThread *mixerThread = (MixerThread *)thread.get();
@@ -3881,7 +4361,7 @@
if (mOutBuffer.frameCount == 0) {
mOutBuffer.frameCount = pInBuffer->frameCount;
nsecs_t startTime = systemTime();
- if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+ if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
outputBufferFull = true;
break;
@@ -3970,13 +4450,13 @@
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
- if (UNLIKELY(!active)) {
+ if (CC_UNLIKELY(!active)) {
ALOGV("Not active and NO_MORE_BUFFERS");
- return AudioTrack::NO_MORE_BUFFERS;
+ return NO_MORE_BUFFERS;
}
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
if (result != NO_ERROR) {
- return AudioTrack::NO_MORE_BUFFERS;
+ return NO_MORE_BUFFERS;
}
// read the server count again
start_loop_here:
@@ -3985,7 +4465,7 @@
}
// if (framesAvail < framesReq) {
-// return AudioTrack::NO_MORE_BUFFERS;
+// return NO_MORE_BUFFERS;
// }
if (framesReq > framesAvail) {
@@ -4008,10 +4488,9 @@
void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue()
{
size_t size = mBufferQueue.size();
- Buffer *pBuffer;
for (size_t i = 0; i < size; i++) {
- pBuffer = mBufferQueue.itemAt(i);
+ Buffer *pBuffer = mBufferQueue.itemAt(i);
delete [] pBuffer->mBuffer;
delete pBuffer;
}
@@ -4023,8 +4502,10 @@
AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
: RefBase(),
mAudioFlinger(audioFlinger),
+ // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below
mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")),
- mPid(pid)
+ mPid(pid),
+ mTimedTrackCount(0)
{
// 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
}
@@ -4035,31 +4516,53 @@
mAudioFlinger->removeClient_l(mPid);
}
-const sp<MemoryDealer>& AudioFlinger::Client::heap() const
+sp<MemoryDealer> AudioFlinger::Client::heap() const
{
return mMemoryDealer;
}
+// Reserve one of the limited slots for a timed audio track associated
+// with this client
+bool AudioFlinger::Client::reserveTimedTrack()
+{
+ const int kMaxTimedTracksPerClient = 4;
+
+ Mutex::Autolock _l(mTimedTrackLock);
+
+ if (mTimedTrackCount >= kMaxTimedTracksPerClient) {
+ ALOGW("can not create timed track - pid %d has exceeded the limit",
+ mPid);
+ return false;
+ }
+
+ mTimedTrackCount++;
+ return true;
+}
+
+// Release a slot for a timed audio track
+void AudioFlinger::Client::releaseTimedTrack()
+{
+ Mutex::Autolock _l(mTimedTrackLock);
+ mTimedTrackCount--;
+}
+
// ----------------------------------------------------------------------------
AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger,
const sp<IAudioFlingerClient>& client,
pid_t pid)
- : mAudioFlinger(audioFlinger), mPid(pid), mClient(client)
+ : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client)
{
}
AudioFlinger::NotificationClient::~NotificationClient()
{
- mClient.clear();
}
void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who)
{
sp<NotificationClient> keep(this);
- {
- mAudioFlinger->removeNotificationClient(mPid);
- }
+ mAudioFlinger->removeNotificationClient(mPid);
}
// ----------------------------------------------------------------------------
@@ -4078,8 +4581,12 @@
mTrack->destroy();
}
-status_t AudioFlinger::TrackHandle::start() {
- return mTrack->start();
+sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
+ return mTrack->getCblk();
+}
+
+status_t AudioFlinger::TrackHandle::start(pid_t tid) {
+ return mTrack->start(tid);
}
void AudioFlinger::TrackHandle::stop() {
@@ -4098,19 +4605,43 @@
mTrack->pause();
}
-void AudioFlinger::TrackHandle::setVolume(float left, float right) {
- mTrack->setVolume(left, right);
-}
-
-sp<IMemory> AudioFlinger::TrackHandle::getCblk() const {
- return mTrack->getCblk();
-}
-
status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId)
{
return mTrack->attachAuxEffect(EffectId);
}
+status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer) {
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->allocateTimedBuffer(size, buffer);
+}
+
+status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts) {
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->queueTimedBuffer(buffer, pts);
+}
+
+status_t AudioFlinger::TrackHandle::setMediaTimeTransform(
+ const LinearTransform& xform, int target) {
+
+ if (!mTrack->isTimedTrack())
+ return INVALID_OPERATION;
+
+ PlaybackThread::TimedTrack* tt =
+ reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get());
+ return tt->setMediaTimeTransform(
+ xform, static_cast<TimedAudioTrack::TargetTimeline>(target));
+}
+
status_t AudioFlinger::TrackHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
@@ -4121,11 +4652,12 @@
sp<IAudioRecord> AudioFlinger::openRecord(
pid_t pid,
- int input,
+ audio_io_handle_t input,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
+ // FIXME dead, remove from IAudioFlinger
uint32_t flags,
int *sessionId,
status_t *status)
@@ -4133,7 +4665,6 @@
sp<RecordThread::RecordTrack> recordTrack;
sp<RecordHandle> recordHandle;
sp<Client> client;
- wp<Client> wclient;
status_t lStatus;
RecordThread *thread;
size_t inFrameCount;
@@ -4154,13 +4685,7 @@
goto Exit;
}
- wclient = mClients.valueFor(pid);
- if (wclient != NULL) {
- client = wclient.promote();
- } else {
- client = new Client(this, pid);
- mClients.add(pid, client);
- }
+ client = registerPid_l(pid);
// If no audio session id is provided, create one here
if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) {
@@ -4177,7 +4702,6 @@
format,
channelMask,
frameCount,
- flags,
lSessionId,
&lStatus);
}
@@ -4212,9 +4736,13 @@
stop();
}
-status_t AudioFlinger::RecordHandle::start() {
+sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
+ return mRecordTrack->getCblk();
+}
+
+status_t AudioFlinger::RecordHandle::start(pid_t tid) {
ALOGV("RecordHandle::start()");
- return mRecordTrack->start();
+ return mRecordTrack->start(tid);
}
void AudioFlinger::RecordHandle::stop() {
@@ -4222,10 +4750,6 @@
mRecordTrack->stop();
}
-sp<IMemory> AudioFlinger::RecordHandle::getCblk() const {
- return mRecordTrack->getCblk();
-}
-
status_t AudioFlinger::RecordHandle::onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
{
@@ -4238,17 +4762,18 @@
AudioStreamIn *input,
uint32_t sampleRate,
uint32_t channels,
- int id,
+ audio_io_handle_t id,
uint32_t device) :
- ThreadBase(audioFlinger, id, device),
- mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0)
+ ThreadBase(audioFlinger, id, device, RECORD),
+ mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL),
+ // mRsmpInIndex and mInputBytes set by readInputParameters()
+ mReqChannelCount(popcount(channels)),
+ mReqSampleRate(sampleRate)
+ // mBytesRead is only meaningful while active, and so is cleared in start()
+ // (but might be better to also clear here for dump?)
{
- mType = ThreadBase::RECORD;
-
snprintf(mName, kNameLength, "AudioIn_%d", id);
- mReqChannelCount = popcount(channels);
- mReqSampleRate = sampleRate;
readInputParameters();
}
@@ -4256,10 +4781,8 @@
AudioFlinger::RecordThread::~RecordThread()
{
delete[] mRsmpInBuffer;
- if (mResampler != 0) {
- delete mResampler;
- delete[] mRsmpOutBuffer;
- }
+ delete mResampler;
+ delete[] mRsmpOutBuffer;
}
void AudioFlinger::RecordThread::onFirstRef()
@@ -4348,9 +4871,10 @@
}
buffer.frameCount = mFrameCount;
- if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ if (CC_LIKELY(mActiveTrack->getNextBuffer(
+ &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) {
size_t framesOut = buffer.frameCount;
- if (mResampler == 0) {
+ if (mResampler == NULL) {
// no resampling
while (framesOut) {
size_t framesIn = mFrameCount - mRsmpInIndex;
@@ -4415,7 +4939,7 @@
// ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
// are 32 bit aligned which should be always true.
if (mChannelCount == 2 && mReqChannelCount == 1) {
- AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
+ ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
// the resampler always outputs stereo samples: do post stereo to mono conversion
int16_t *src = (int16_t *)mRsmpOutBuffer;
int16_t *dst = buffer.i16;
@@ -4424,7 +4948,7 @@
src += 2;
}
} else {
- AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
+ ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut);
}
}
@@ -4435,7 +4959,7 @@
else {
if (!mActiveTrack->setOverflow()) {
nsecs_t now = systemTime();
- if ((now - lastWarning) > kWarningThrottle) {
+ if ((now - lastWarning) > kWarningThrottleNs) {
ALOGW("RecordThread: buffer overflow");
lastWarning = now;
}
@@ -4468,10 +4992,9 @@
sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
int frameCount,
- uint32_t flags,
int sessionId,
status_t *status)
{
@@ -4488,9 +5011,9 @@
Mutex::Autolock _l(mLock);
track = new RecordTrack(this, client, sampleRate,
- format, channelMask, frameCount, flags, sessionId);
+ format, channelMask, frameCount, sessionId);
- if (track->getCblk() == NULL) {
+ if (track->getCblk() == 0) {
lStatus = NO_MEMORY;
goto Exit;
}
@@ -4511,13 +5034,13 @@
return track;
}
-status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack)
+status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid)
{
- ALOGV("RecordThread::start");
+ ALOGV("RecordThread::start tid=%d", tid);
sp <ThreadBase> strongMe = this;
status_t status = NO_ERROR;
{
- AutoMutex lock(&mLock);
+ AutoMutex lock(mLock);
if (mActiveTrack != 0) {
if (recordTrack != mActiveTrack.get()) {
status = -EBUSY;
@@ -4546,7 +5069,7 @@
ALOGV("Signal record thread");
mWaitWorkCV.signal();
// do not wait for mStartStopCond if exiting
- if (mExiting) {
+ if (exitPending()) {
mActiveTrack.clear();
status = INVALID_OPERATION;
goto startError;
@@ -4569,11 +5092,11 @@
ALOGV("RecordThread::stop");
sp <ThreadBase> strongMe = this;
{
- AutoMutex lock(&mLock);
+ AutoMutex lock(mLock);
if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
mActiveTrack->mState = TrackBase::PAUSING;
// do not wait for mStartStopCond if exiting
- if (mExiting) {
+ if (exitPending()) {
return;
}
mStartStopCond.wait(mLock);
@@ -4593,7 +5116,6 @@
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
- pid_t pid = 0;
snprintf(buffer, SIZE, "\nInput thread %p internals\n", this);
result.append(buffer);
@@ -4608,7 +5130,7 @@
result.append(buffer);
snprintf(buffer, SIZE, "In size: %d\n", mInputBytes);
result.append(buffer);
- snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0));
+ snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL));
result.append(buffer);
snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount);
result.append(buffer);
@@ -4627,7 +5149,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
{
size_t framesReq = buffer->frameCount;
size_t framesReady = mFrameCount - mRsmpInIndex;
@@ -4643,7 +5165,7 @@
mInput->stream->common.standby(&mInput->stream->common);
usleep(kRecordThreadSleepUs);
}
- buffer->raw = 0;
+ buffer->raw = NULL;
buffer->frameCount = 0;
return NOT_ENOUGH_DATA;
}
@@ -4680,7 +5202,7 @@
String8 keyValuePair = mNewParameters[0];
AudioParameter param = AudioParameter(keyValuePair);
int value;
- int reqFormat = mFormat;
+ audio_format_t reqFormat = mFormat;
int reqSamplingRate = mReqSampleRate;
int reqChannelCount = mReqChannelCount;
@@ -4689,7 +5211,7 @@
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) {
- reqFormat = value;
+ reqFormat = (audio_format_t) value;
reconfig = true;
}
if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
@@ -4698,7 +5220,7 @@
}
if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) {
// do not accept frame count changes if tracks are open as the track buffer
- // size depends on frame count and correct behavior would not be garantied
+ // size depends on frame count and correct behavior would not be guaranteed
// if frame count is changed after track creation
if (mActiveTrack != 0) {
status = INVALID_OPERATION;
@@ -4758,7 +5280,7 @@
mParamCond.signal();
// wait for condition with time out in case the thread calling ThreadBase::setParameters()
// already timed out waiting for the status and will never signal the condition.
- mWaitWorkCV.waitRelative(mLock, kSetParametersTimeout);
+ mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs);
}
return reconfig;
}
@@ -4781,7 +5303,7 @@
void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) {
AudioSystem::OutputDescriptor desc;
- void *param2 = 0;
+ void *param2 = NULL;
switch (event) {
case AudioSystem::INPUT_OPENED:
@@ -4803,16 +5325,18 @@
void AudioFlinger::RecordThread::readInputParameters()
{
- if (mRsmpInBuffer) delete mRsmpInBuffer;
- if (mRsmpOutBuffer) delete mRsmpOutBuffer;
- if (mResampler) delete mResampler;
- mResampler = 0;
+ delete mRsmpInBuffer;
+ // mRsmpInBuffer is always assigned a new[] below
+ delete mRsmpOutBuffer;
+ mRsmpOutBuffer = NULL;
+ delete mResampler;
+ mResampler = NULL;
mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common);
mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common);
mChannelCount = (uint16_t)popcount(mChannelMask);
mFormat = mInput->stream->common.get_format(&mInput->stream->common);
- mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common);
+ mFrameSize = audio_stream_frame_size(&mInput->stream->common);
mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common);
mFrameCount = mInputBytes / mFrameSize;
mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];
@@ -4872,7 +5396,7 @@
return mTrack;
}
-AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput()
+AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const
{
Mutex::Autolock _l(mLock);
return mInput;
@@ -4898,9 +5422,9 @@
// ----------------------------------------------------------------------------
-int AudioFlinger::openOutput(uint32_t *pDevices,
+audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags)
@@ -4909,7 +5433,7 @@
PlaybackThread *thread = NULL;
mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t latency = pLatencyMs ? *pLatencyMs : 0;
audio_stream_out_t *outStream;
@@ -4932,7 +5456,7 @@
if (outHwDev == NULL)
return 0;
- status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format,
+ status = outHwDev->open_output_stream(outHwDev, *pDevices, &format,
&channels, &samplingRate, &outStream);
ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
outStream,
@@ -4944,7 +5468,7 @@
mHardwareStatus = AUDIO_HW_IDLE;
if (outStream != NULL) {
AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream);
- int id = nextUniqueId();
+ audio_io_handle_t id = nextUniqueId();
if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) ||
(format != AUDIO_FORMAT_PCM_16_BIT) ||
@@ -4957,10 +5481,10 @@
}
mPlaybackThreads.add(id, thread);
- if (pSamplingRate) *pSamplingRate = samplingRate;
- if (pFormat) *pFormat = format;
- if (pChannels) *pChannels = channels;
- if (pLatencyMs) *pLatencyMs = thread->latency();
+ if (pSamplingRate != NULL) *pSamplingRate = samplingRate;
+ if (pFormat != NULL) *pFormat = format;
+ if (pChannels != NULL) *pChannels = channels;
+ if (pLatencyMs != NULL) *pLatencyMs = thread->latency();
// notify client processes of the new output creation
thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED);
@@ -4970,7 +5494,8 @@
return 0;
}
-int AudioFlinger::openDuplicateOutput(int output1, int output2)
+audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2)
{
Mutex::Autolock _l(mLock);
MixerThread *thread1 = checkMixerThread_l(output1);
@@ -4981,7 +5506,7 @@
return 0;
}
- int id = nextUniqueId();
+ audio_io_handle_t id = nextUniqueId();
DuplicatingThread *thread = new DuplicatingThread(this, thread1, id);
thread->addOutputTrack(thread2);
mPlaybackThreads.add(id, thread);
@@ -4990,7 +5515,7 @@
return id;
}
-status_t AudioFlinger::closeOutput(int output)
+status_t AudioFlinger::closeOutput(audio_io_handle_t output)
{
// keep strong reference on the playback thread so that
// it is not destroyed while exit() is executed
@@ -5012,14 +5537,16 @@
}
}
}
- void *param2 = 0;
- audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2);
+ audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL);
mPlaybackThreads.removeItem(output);
}
thread->exit();
+ // The thread entity (active unit of execution) is no longer running here,
+ // but the ThreadBase container still exists.
if (thread->type() != ThreadBase::DUPLICATING) {
AudioStreamOut *out = thread->clearOutput();
+ assert(out != NULL);
// from now on thread->mOutput is NULL
out->hwDev->close_output_stream(out->hwDev, out->stream);
delete out;
@@ -5027,7 +5554,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::suspendOutput(int output)
+status_t AudioFlinger::suspendOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -5042,7 +5569,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::restoreOutput(int output)
+status_t AudioFlinger::restoreOutput(audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
PlaybackThread *thread = checkPlaybackThread_l(output);
@@ -5058,19 +5585,19 @@
return NO_ERROR;
}
-int AudioFlinger::openInput(uint32_t *pDevices,
+audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
status_t status;
RecordThread *thread = NULL;
uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0;
- uint32_t format = pFormat ? *pFormat : 0;
+ audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT;
uint32_t channels = pChannels ? *pChannels : 0;
uint32_t reqSamplingRate = samplingRate;
- uint32_t reqFormat = format;
+ audio_format_t reqFormat = format;
uint32_t reqChannels = channels;
audio_stream_in_t *inStream;
audio_hw_device_t *inHwDev;
@@ -5085,9 +5612,9 @@
if (inHwDev == NULL)
return 0;
- status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
+ status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
&channels, &samplingRate,
- (audio_in_acoustics_t)acoustics,
+ acoustics,
&inStream);
ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d",
inStream,
@@ -5105,16 +5632,16 @@
(samplingRate <= 2 * reqSamplingRate) &&
(popcount(channels) < 3) && (popcount(reqChannels) < 3)) {
ALOGV("openInput() reopening with proposed sampling rate and channels");
- status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format,
+ status = inHwDev->open_input_stream(inHwDev, *pDevices, &format,
&channels, &samplingRate,
- (audio_in_acoustics_t)acoustics,
+ acoustics,
&inStream);
}
if (inStream != NULL) {
AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream);
- int id = nextUniqueId();
+ audio_io_handle_t id = nextUniqueId();
// Start record thread
// RecorThread require both input and output device indication to forward to audio
// pre processing modules
@@ -5127,9 +5654,9 @@
device);
mRecordThreads.add(id, thread);
ALOGV("openInput() created record thread: ID %d thread %p", id, thread);
- if (pSamplingRate) *pSamplingRate = reqSamplingRate;
- if (pFormat) *pFormat = format;
- if (pChannels) *pChannels = reqChannels;
+ if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate;
+ if (pFormat != NULL) *pFormat = format;
+ if (pChannels != NULL) *pChannels = reqChannels;
input->stream->common.standby(&input->stream->common);
@@ -5141,7 +5668,7 @@
return 0;
}
-status_t AudioFlinger::closeInput(int input)
+status_t AudioFlinger::closeInput(audio_io_handle_t input)
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
@@ -5154,13 +5681,15 @@
}
ALOGV("closeInput() %d", input);
- void *param2 = 0;
- audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2);
+ audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL);
mRecordThreads.removeItem(input);
}
thread->exit();
+ // The thread entity (active unit of execution) is no longer running here,
+ // but the ThreadBase container still exists.
AudioStreamIn *in = thread->clearInput();
+ assert(in != NULL);
// from now on thread->mInput is NULL
in->hwDev->close_input_stream(in->hwDev, in->stream);
delete in;
@@ -5168,7 +5697,7 @@
return NO_ERROR;
}
-status_t AudioFlinger::setStreamOutput(uint32_t stream, int output)
+status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output)
{
Mutex::Autolock _l(mLock);
MixerThread *dstThread = checkMixerThread_l(output);
@@ -5184,8 +5713,7 @@
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
- if (thread != dstThread &&
- thread->type() != ThreadBase::DIRECT) {
+ if (thread != dstThread && thread->type() != ThreadBase::DIRECT) {
MixerThread *srcThread = (MixerThread *)thread;
srcThread->setStreamValid(stream, false);
srcThread->invalidateTracks(stream);
@@ -5204,10 +5732,10 @@
void AudioFlinger::acquireAudioSessionId(int audioSession)
{
Mutex::Autolock _l(mLock);
- int caller = IPCThreadState::self()->getCallingPid();
+ pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("acquiring %d from %d", audioSession, caller);
- int num = mAudioSessionRefs.size();
- for (int i = 0; i< num; i++) {
+ size_t num = mAudioSessionRefs.size();
+ for (size_t i = 0; i< num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i);
if (ref->sessionid == audioSession && ref->pid == caller) {
ref->cnt++;
@@ -5215,21 +5743,17 @@
return;
}
}
- AudioSessionRef *ref = new AudioSessionRef();
- ref->sessionid = audioSession;
- ref->pid = caller;
- ref->cnt = 1;
- mAudioSessionRefs.push(ref);
- ALOGV(" added new entry for %d", ref->sessionid);
+ mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller));
+ ALOGV(" added new entry for %d", audioSession);
}
void AudioFlinger::releaseAudioSessionId(int audioSession)
{
Mutex::Autolock _l(mLock);
- int caller = IPCThreadState::self()->getCallingPid();
+ pid_t caller = IPCThreadState::self()->getCallingPid();
ALOGV("releasing %d from %d", audioSession, caller);
- int num = mAudioSessionRefs.size();
- for (int i = 0; i< num; i++) {
+ size_t num = mAudioSessionRefs.size();
+ for (size_t i = 0; i< num; i++) {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(i);
if (ref->sessionid == audioSession && ref->pid == caller) {
ref->cnt--;
@@ -5310,35 +5834,22 @@
}
// checkPlaybackThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const
+AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const
{
- PlaybackThread *thread = NULL;
- if (mPlaybackThreads.indexOfKey(output) >= 0) {
- thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get();
- }
- return thread;
+ return mPlaybackThreads.valueFor(output).get();
}
// checkMixerThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const
+AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const
{
PlaybackThread *thread = checkPlaybackThread_l(output);
- if (thread != NULL) {
- if (thread->type() == ThreadBase::DIRECT) {
- thread = NULL;
- }
- }
- return (MixerThread *)thread;
+ return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL;
}
// checkRecordThread_l() must be called with AudioFlinger::mLock held
-AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const
+AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const
{
- RecordThread *thread = NULL;
- if (mRecordThreads.indexOfKey(input) >= 0) {
- thread = (RecordThread *)mRecordThreads.valueFor(input).get();
- }
- return thread;
+ return mRecordThreads.valueFor(input).get();
}
uint32_t AudioFlinger::nextUniqueId()
@@ -5375,19 +5886,20 @@
// ----------------------------------------------------------------------------
-status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects)
+status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const
{
Mutex::Autolock _l(mLock);
return EffectQueryNumberEffects(numEffects);
}
-status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor)
+status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectQueryEffect(index, descriptor);
}
-status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor)
+status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
+ effect_descriptor_t *descriptor) const
{
Mutex::Autolock _l(mLock);
return EffectGetDescriptor(pUuid, descriptor);
@@ -5398,7 +5910,7 @@
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
- int io,
+ audio_io_handle_t io,
int sessionId,
status_t *status,
int *id,
@@ -5407,10 +5919,8 @@
status_t lStatus = NO_ERROR;
sp<EffectHandle> handle;
effect_descriptor_t desc;
- sp<Client> client;
- wp<Client> wclient;
- ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d",
+ ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d",
pid, effectClient.get(), priority, sessionId, io);
if (pDesc == NULL) {
@@ -5426,7 +5936,7 @@
// Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
// that can only be created by audio policy manager (running in same process)
- if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) {
+ if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
lStatus = PERMISSION_DENIED;
goto Exit;
}
@@ -5561,14 +6071,7 @@
}
}
- wclient = mClients.valueFor(pid);
-
- if (wclient != NULL) {
- client = wclient.promote();
- } else {
- client = new Client(this, pid);
- mClients.add(pid, client);
- }
+ sp<Client> client = registerPid_l(pid);
// create effect on selected output thread
handle = thread->createEffect_l(client, effectClient, priority, sessionId,
@@ -5585,7 +6088,8 @@
return handle;
}
-status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput)
+status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput)
{
ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d",
sessionId, srcOutput, dstOutput);
@@ -5636,7 +6140,7 @@
// transfer all effects one by one so that new effect chain is created on new thread with
// correct buffer sizes and audio parameters and effect engines reconfigured accordingly
- int dstOutput = dstThread->id();
+ audio_io_handle_t dstOutput = dstThread->id();
sp<EffectChain> dstChain;
uint32_t strategy = 0; // prevent compiler warning
sp<EffectModule> effect = chain->getEffectFromId_l(0);
@@ -5708,10 +6212,7 @@
goto Exit;
}
// Only Pre processor effects are allowed on input threads and only on input threads
- if ((mType == RECORD &&
- (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) ||
- (mType != RECORD &&
- (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
+ if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) {
ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d",
desc->name, desc->flags, mType);
lStatus = BAD_VALUE;
@@ -5736,7 +6237,7 @@
effect = chain->getEffectFromDesc_l(desc);
}
- ALOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get());
+ ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get());
if (effect == 0) {
int id = mAudioFlinger->nextUniqueId();
@@ -5764,7 +6265,7 @@
// create effect handle and connect it to effect module
handle = new EffectHandle(effect, client, effectClient, priority);
lStatus = effect->addHandle(handle);
- if (enabled) {
+ if (enabled != NULL) {
*enabled = (int)effect->isEnabled();
}
}
@@ -5792,13 +6293,8 @@
sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId)
{
- sp<EffectModule> effect;
-
sp<EffectChain> chain = getEffectChain_l(sessionId);
- if (chain != 0) {
- effect = chain->getEffectFromId_l(effectId);
- }
- return effect;
+ return chain != 0 ? chain->getEffectFromId_l(effectId) : 0;
}
// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and
@@ -5883,19 +6379,16 @@
sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId)
{
- sp<EffectChain> chain;
-
size_t size = mEffectChains.size();
for (size_t i = 0; i < size; i++) {
if (mEffectChains[i]->sessionId() == sessionId) {
- chain = mEffectChains[i];
- break;
+ return mEffectChains[i];
}
}
- return chain;
+ return 0;
}
-void AudioFlinger::ThreadBase::setMode(uint32_t mode)
+void AudioFlinger::ThreadBase::setMode(audio_mode_t mode)
{
Mutex::Autolock _l(mLock);
size_t size = mEffectChains.size();
@@ -5906,13 +6399,13 @@
void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect,
const wp<EffectHandle>& handle,
- bool unpiniflast) {
+ bool unpinIfLast) {
Mutex::Autolock _l(mLock);
ALOGV("disconnectEffect() %p effect %p", this, effect.get());
// delete the effect module if removing last handle on it
if (effect->removeHandle(handle) == 0) {
- if (!effect->isPinned() || unpiniflast) {
+ if (!effect->isPinned() || unpinIfLast) {
removeEffect_l(effect);
AudioSystem::unregisterEffect(effect->id());
}
@@ -6092,18 +6585,17 @@
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectModule"
-AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread,
+AudioFlinger::EffectModule::EffectModule(ThreadBase *thread,
const wp<AudioFlinger::EffectChain>& chain,
effect_descriptor_t *desc,
int id,
int sessionId)
- : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
+ : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL),
mStatus(NO_INIT), mState(IDLE), mSuspended(false)
{
ALOGV("Constructor %p", this);
int lStatus;
- sp<ThreadBase> thread = mThread.promote();
- if (thread == 0) {
+ if (thread == NULL) {
return;
}
@@ -6151,12 +6643,11 @@
}
}
-status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle)
+status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle)
{
status_t status;
Mutex::Autolock _l(mLock);
- // First handle in mHandles has highest priority and controls the effect module
int priority = handle->priority();
size_t size = mHandles.size();
sp<EffectHandle> h;
@@ -6198,7 +6689,7 @@
bool enabled = false;
EffectHandle *hdl = handle.unsafe_get();
- if (hdl) {
+ if (hdl != NULL) {
ALOGV("removeHandle() unsafe_get OK");
enabled = hdl->enabled();
}
@@ -6225,23 +6716,19 @@
sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle()
{
Mutex::Autolock _l(mLock);
- sp<EffectHandle> handle;
- if (mHandles.size() != 0) {
- handle = mHandles[0].promote();
- }
- return handle;
+ return mHandles.size() != 0 ? mHandles[0].promote() : 0;
}
-void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast)
+void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast)
{
- ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get());
+ ALOGV("disconnect() %p handle %p", this, handle.unsafe_get());
// keep a strong reference on this EffectModule to avoid calling the
// destructor before we exit
sp<EffectModule> keep(this);
{
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
- thread->disconnectEffect(keep, handle, unpiniflast);
+ thread->disconnectEffect(keep, handle, unpinIfLast);
}
}
}
@@ -6295,7 +6782,7 @@
if (isProcessEnabled()) {
// do 32 bit to 16 bit conversion for auxiliary effect input buffer
if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
- AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32,
+ ditherAndClamp(mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.s32,
mConfig.inputCfg.buffer.frameCount/2);
}
@@ -6400,7 +6887,7 @@
status_t cmdStatus;
uint32_t size = sizeof(int);
status_t status = (*mEffectInterface)->command(mEffectInterface,
- EFFECT_CMD_CONFIGURE,
+ EFFECT_CMD_SET_CONFIG,
sizeof(effect_config_t),
&mConfig,
&size,
@@ -6584,7 +7071,7 @@
return NO_ERROR;
}
-bool AudioFlinger::EffectModule::isEnabled()
+bool AudioFlinger::EffectModule::isEnabled() const
{
switch (mState) {
case RESTART:
@@ -6600,7 +7087,7 @@
}
}
-bool AudioFlinger::EffectModule::isProcessEnabled()
+bool AudioFlinger::EffectModule::isProcessEnabled() const
{
switch (mState) {
case RESTART:
@@ -6693,7 +7180,7 @@
return status;
}
-status_t AudioFlinger::EffectModule::setMode(uint32_t mode)
+status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode)
{
Mutex::Autolock _l(mLock);
status_t status = NO_ERROR;
@@ -6702,7 +7189,7 @@
uint32_t size = sizeof(status_t);
status = (*mEffectInterface)->command(mEffectInterface,
EFFECT_CMD_SET_AUDIO_MODE,
- sizeof(int),
+ sizeof(audio_mode_t),
&mode,
&size,
&cmdStatus);
@@ -6718,7 +7205,8 @@
Mutex::Autolock _l(mLock);
mSuspended = suspended;
}
-bool AudioFlinger::EffectModule::suspended()
+
+bool AudioFlinger::EffectModule::suspended() const
{
Mutex::Autolock _l(mLock);
return mSuspended;
@@ -6833,7 +7321,7 @@
if (mCblkMemory != 0) {
mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer());
- if (mCblk) {
+ if (mCblk != NULL) {
new(mCblk) effect_param_cblk_t();
mBuffer = (uint8_t *)mCblk + bufOffset;
}
@@ -6912,13 +7400,13 @@
disconnect(true);
}
-void AudioFlinger::EffectHandle::disconnect(bool unpiniflast)
+void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast)
{
- ALOGV("disconnect(%s)", unpiniflast ? "true" : "false");
+ ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false");
if (mEffect == 0) {
return;
}
- mEffect->disconnect(this, unpiniflast);
+ mEffect->disconnect(this, unpinIfLast);
if (mHasControl && mEnabled) {
sp<ThreadBase> thread = mEffect->thread().promote();
@@ -6930,10 +7418,12 @@
// release sp on module => module destructor can be called now
mEffect.clear();
if (mClient != 0) {
- if (mCblk) {
+ if (mCblk != NULL) {
+ // unlike ~TrackBase(), mCblk is never a local new, so don't delete
mCblk->~effect_param_cblk_t(); // destroy our shared-structure.
}
- mCblkMemory.clear(); // and free the shared memory
+ mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to
+ // Client destructor must run with AudioFlinger mutex locked
Mutex::Autolock _l(mClient->audioFlinger()->mLock);
mClient.clear();
}
@@ -7015,10 +7505,6 @@
return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
}
-sp<IMemory> AudioFlinger::EffectHandle::getCblk() const {
- return mCblkMemory;
-}
-
void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled)
{
ALOGV("setControl %p control %d", this, hasControl);
@@ -7060,10 +7546,10 @@
void AudioFlinger::EffectHandle::dump(char* buffer, size_t size)
{
- bool locked = mCblk ? tryLock(mCblk->lock) : false;
+ bool locked = mCblk != NULL && tryLock(mCblk->lock);
snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n",
- (mClient == NULL) ? getpid() : mClient->pid(),
+ (mClient == 0) ? getpid_cached : mClient->pid(),
mPriority,
mHasControl,
!locked,
@@ -7079,15 +7565,14 @@
#undef LOG_TAG
#define LOG_TAG "AudioFlinger::EffectChain"
-AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread,
+AudioFlinger::EffectChain::EffectChain(ThreadBase *thread,
int sessionId)
- : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
+ : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0),
mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX),
mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX)
{
mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
- sp<ThreadBase> thread = mThread.promote();
- if (thread == 0) {
+ if (thread == NULL) {
return;
}
mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) /
@@ -7105,48 +7590,42 @@
// getEffectFromDesc_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
{
- sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) {
- effect = mEffects[i];
- break;
+ return mEffects[i];
}
}
- return effect;
+ return 0;
}
// getEffectFromId_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id)
{
- sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
// by convention, return first effect if id provided is 0 (0 is never a valid id)
if (id == 0 || mEffects[i]->id() == id) {
- effect = mEffects[i];
- break;
+ return mEffects[i];
}
}
- return effect;
+ return 0;
}
// getEffectFromType_l() must be called with ThreadBase::mLock held
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l(
const effect_uuid_t *type)
{
- sp<EffectModule> effect;
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) {
- effect = mEffects[i];
- break;
+ return mEffects[i];
}
}
- return effect;
+ return 0;
}
// Must be called with EffectChain::mLock locked
@@ -7237,12 +7716,12 @@
// Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is
// already present
- int size = (int)mEffects.size();
- int idx_insert = size;
- int idx_insert_first = -1;
- int idx_insert_last = -1;
+ size_t size = mEffects.size();
+ size_t idx_insert = size;
+ ssize_t idx_insert_first = -1;
+ ssize_t idx_insert_last = -1;
- for (int i = 0; i < size; i++) {
+ for (size_t i = 0; i < size; i++) {
effect_descriptor_t d = mEffects[i]->desc();
uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK;
uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK;
@@ -7311,11 +7790,10 @@
size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect)
{
Mutex::Autolock _l(mLock);
- int size = (int)mEffects.size();
- int i;
+ size_t size = mEffects.size();
uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK;
- for (i = 0; i < size; i++) {
+ for (size_t i = 0; i < size; i++) {
if (effect == mEffects[i]) {
// calling stop here will remove pre-processing effect from the audio HAL.
// This is safe as we hold the EffectChain mutex which guarantees that we are not in
@@ -7351,7 +7829,7 @@
}
// setMode_l() must be called with PlaybackThread::mLock held
-void AudioFlinger::EffectChain::setMode_l(uint32_t mode)
+void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode)
{
size_t size = mEffects.size();
for (size_t i = 0; i < size; i++) {
@@ -7462,7 +7940,7 @@
sp<SuspendedEffectDesc> desc;
// use effect type UUID timelow as key as there is no real risk of identical
// timeLow fields among effect type UUIDs.
- int index = mSuspendedEffects.indexOfKey(type->timeLow);
+ ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow);
if (suspend) {
if (index >= 0) {
desc = mSuspendedEffects.valueAt(index);
@@ -7512,7 +7990,7 @@
{
sp<SuspendedEffectDesc> desc;
- int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
+ ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll);
if (suspend) {
if (index >= 0) {
desc = mSuspendedEffects.valueAt(index);
@@ -7522,7 +8000,8 @@
ALOGV("setEffectSuspendedAll_l() add entry for 0");
}
if (desc->mRefCount++ == 0) {
- Vector< sp<EffectModule> > effects = getSuspendEligibleEffects();
+ Vector< sp<EffectModule> > effects;
+ getSuspendEligibleEffects(effects);
for (size_t i = 0; i < effects.size(); i++) {
setEffectSuspended_l(&effects[i]->desc().type, true);
}
@@ -7573,33 +8052,27 @@
return true;
}
-Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects()
+void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects)
{
- Vector< sp<EffectModule> > effects;
+ effects.clear();
for (size_t i = 0; i < mEffects.size(); i++) {
- if (!isEffectEligibleForSuspend(mEffects[i]->desc())) {
- continue;
+ if (isEffectEligibleForSuspend(mEffects[i]->desc())) {
+ effects.add(mEffects[i]);
}
- effects.add(mEffects[i]);
}
- return effects;
}
sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled(
const effect_uuid_t *type)
{
- sp<EffectModule> effect;
- effect = getEffectFromType_l(type);
- if (effect != 0 && !effect->isEnabled()) {
- effect.clear();
- }
- return effect;
+ sp<EffectModule> effect = getEffectFromType_l(type);
+ return effect != 0 && effect->isEnabled() ? effect : 0;
}
void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect,
bool enabled)
{
- int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
+ ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow);
if (enabled) {
if (index < 0) {
// if the effect is not suspend check if all effects are suspended
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 9bd2c7f..2a5d805 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioFlinger.h
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -22,10 +22,13 @@
#include <sys/types.h>
#include <limits.h>
+#include <common_time/cc_helper.h>
+
#include <media/IAudioFlinger.h>
#include <media/IAudioFlingerClient.h>
#include <media/IAudioTrack.h>
#include <media/IAudioRecord.h>
+#include <media/AudioSystem.h>
#include <media/AudioTrack.h>
#include <utils/Atomic.h>
@@ -55,13 +58,7 @@
// ----------------------------------------------------------------------------
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
-
-
-// ----------------------------------------------------------------------------
-
-static const nsecs_t kStandbyTimeInNsecs = seconds(3);
+static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3);
class AudioFlinger :
public BinderService<AudioFlinger>,
@@ -69,83 +66,101 @@
{
friend class BinderService<AudioFlinger>;
public:
- static char const* getServiceName() { return "media.audio_flinger"; }
+ static const char* getServiceName() { return "media.audio_flinger"; }
virtual status_t dump(int fd, const Vector<String16>& args);
- // IAudioFlinger interface
+ // IAudioFlinger interface, in binder opcode order
virtual sp<IAudioTrack> createTrack(
pid_t pid,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
uint32_t flags,
const sp<IMemory>& sharedBuffer,
- int output,
+ audio_io_handle_t output,
+ bool isTimed,
int *sessionId,
status_t *status);
- virtual uint32_t sampleRate(int output) const;
- virtual int channelCount(int output) const;
- virtual uint32_t format(int output) const;
- virtual size_t frameCount(int output) const;
- virtual uint32_t latency(int output) const;
+ virtual sp<IAudioRecord> openRecord(
+ pid_t pid,
+ audio_io_handle_t input,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ uint32_t flags,
+ int *sessionId,
+ status_t *status);
+
+ virtual uint32_t sampleRate(audio_io_handle_t output) const;
+ virtual int channelCount(audio_io_handle_t output) const;
+ virtual audio_format_t format(audio_io_handle_t output) const;
+ virtual size_t frameCount(audio_io_handle_t output) const;
+ virtual uint32_t latency(audio_io_handle_t output) const;
virtual status_t setMasterVolume(float value);
virtual status_t setMasterMute(bool muted);
virtual float masterVolume() const;
+ virtual float masterVolumeSW() const;
virtual bool masterMute() const;
- virtual status_t setStreamVolume(int stream, float value, int output);
- virtual status_t setStreamMute(int stream, bool muted);
+ virtual status_t setStreamVolume(audio_stream_type_t stream, float value,
+ audio_io_handle_t output);
+ virtual status_t setStreamMute(audio_stream_type_t stream, bool muted);
- virtual float streamVolume(int stream, int output) const;
- virtual bool streamMute(int stream) const;
+ virtual float streamVolume(audio_stream_type_t stream,
+ audio_io_handle_t output) const;
+ virtual bool streamMute(audio_stream_type_t stream) const;
- virtual status_t setMode(int mode);
+ virtual status_t setMode(audio_mode_t mode);
virtual status_t setMicMute(bool state);
virtual bool getMicMute() const;
- virtual status_t setParameters(int ioHandle, const String8& keyValuePairs);
- virtual String8 getParameters(int ioHandle, const String8& keys);
+ virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
+ virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
virtual void registerClient(const sp<IAudioFlingerClient>& client);
- virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
- virtual unsigned int getInputFramesLost(int ioHandle);
+ virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const;
- virtual int openOutput(uint32_t *pDevices,
+ virtual audio_io_handle_t openOutput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
uint32_t flags);
- virtual int openDuplicateOutput(int output1, int output2);
+ virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1,
+ audio_io_handle_t output2);
- virtual status_t closeOutput(int output);
+ virtual status_t closeOutput(audio_io_handle_t output);
- virtual status_t suspendOutput(int output);
+ virtual status_t suspendOutput(audio_io_handle_t output);
- virtual status_t restoreOutput(int output);
+ virtual status_t restoreOutput(audio_io_handle_t output);
- virtual int openInput(uint32_t *pDevices,
+ virtual audio_io_handle_t openInput(uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics);
+ audio_in_acoustics_t acoustics);
- virtual status_t closeInput(int input);
+ virtual status_t closeInput(audio_io_handle_t input);
- virtual status_t setStreamOutput(uint32_t stream, int output);
+ virtual status_t setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output);
virtual status_t setVoiceVolume(float volume);
- virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output);
+ virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames,
+ audio_io_handle_t output) const;
+
+ virtual unsigned int getInputFramesLost(audio_io_handle_t ioHandle) const;
virtual int newAudioSessionId();
@@ -153,54 +168,25 @@
virtual void releaseAudioSessionId(int audioSession);
- virtual status_t queryNumberEffects(uint32_t *numEffects);
+ virtual status_t queryNumberEffects(uint32_t *numEffects) const;
- virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor);
+ virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
- virtual status_t getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor);
+ virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
+ effect_descriptor_t *descriptor) const;
virtual sp<IEffect> createEffect(pid_t pid,
effect_descriptor_t *pDesc,
const sp<IEffectClient>& effectClient,
int32_t priority,
- int io,
+ audio_io_handle_t io,
int sessionId,
status_t *status,
int *id,
int *enabled);
- virtual status_t moveEffects(int sessionId, int srcOutput, int dstOutput);
-
- enum hardware_call_state {
- AUDIO_HW_IDLE = 0,
- AUDIO_HW_INIT,
- AUDIO_HW_OUTPUT_OPEN,
- AUDIO_HW_OUTPUT_CLOSE,
- AUDIO_HW_INPUT_OPEN,
- AUDIO_HW_INPUT_CLOSE,
- AUDIO_HW_STANDBY,
- AUDIO_HW_SET_MASTER_VOLUME,
- AUDIO_HW_GET_ROUTING,
- AUDIO_HW_SET_ROUTING,
- AUDIO_HW_GET_MODE,
- AUDIO_HW_SET_MODE,
- AUDIO_HW_GET_MIC_MUTE,
- AUDIO_HW_SET_MIC_MUTE,
- AUDIO_SET_VOICE_VOLUME,
- AUDIO_SET_PARAMETER,
- };
-
- // record interface
- virtual sp<IAudioRecord> openRecord(
- pid_t pid,
- int input,
- uint32_t sampleRate,
- uint32_t format,
- uint32_t channelMask,
- int frameCount,
- uint32_t flags,
- int *sessionId,
- status_t *status);
+ virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput,
+ audio_io_handle_t dstOutput);
virtual status_t onTransact(
uint32_t code,
@@ -208,19 +194,25 @@
Parcel* reply,
uint32_t flags);
- uint32_t getMode() { return mMode; }
-
- bool btNrecIsOff() { return mBtNrecIsOff; }
+ // end of IAudioFlinger interface
private:
+ audio_mode_t getMode() const { return mMode; }
+
+ bool btNrecIsOff() const { return mBtNrecIsOff; }
+
AudioFlinger();
virtual ~AudioFlinger();
- status_t initCheck() const;
+ // call in any IAudioFlinger method that accesses mPrimaryHardwareDev
+ status_t initCheck() const { return mPrimaryHardwareDev == NULL ? NO_INIT : NO_ERROR; }
+
virtual void onFirstRef();
audio_hw_device_t* findSuitableHwDev_l(uint32_t devices);
void purgeStaleEffects_l();
+ static nsecs_t mStandbyTimeInNsecs;
+
// Internal dump utilites.
status_t dumpPermissionDenial(int fd, const Vector<String16>& args);
status_t dumpClients(int fd, const Vector<String16>& args);
@@ -231,16 +223,22 @@
public:
Client(const sp<AudioFlinger>& audioFlinger, pid_t pid);
virtual ~Client();
- const sp<MemoryDealer>& heap() const;
+ sp<MemoryDealer> heap() const;
pid_t pid() const { return mPid; }
- sp<AudioFlinger> audioFlinger() { return mAudioFlinger; }
+ sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; }
+
+ bool reserveTimedTrack();
+ void releaseTimedTrack();
private:
Client(const Client&);
Client& operator = (const Client&);
- sp<AudioFlinger> mAudioFlinger;
- sp<MemoryDealer> mMemoryDealer;
- pid_t mPid;
+ const sp<AudioFlinger> mAudioFlinger;
+ const sp<MemoryDealer> mMemoryDealer;
+ const pid_t mPid;
+
+ Mutex mTimedTrackLock;
+ int mTimedTrackCount;
};
// --- Notification Client ---
@@ -251,7 +249,7 @@
pid_t pid);
virtual ~NotificationClient();
- sp<IAudioFlingerClient> client() { return mClient; }
+ sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; }
// IBinder::DeathRecipient
virtual void binderDied(const wp<IBinder>& who);
@@ -260,9 +258,9 @@
NotificationClient(const NotificationClient&);
NotificationClient& operator = (const NotificationClient&);
- sp<AudioFlinger> mAudioFlinger;
- pid_t mPid;
- sp<IAudioFlingerClient> mClient;
+ const sp<AudioFlinger> mAudioFlinger;
+ const pid_t mPid;
+ const sp<IAudioFlingerClient> mAudioFlingerClient;
};
class TrackHandle;
@@ -282,17 +280,17 @@
class ThreadBase : public Thread {
public:
- ThreadBase (const sp<AudioFlinger>& audioFlinger, int id, uint32_t device);
- virtual ~ThreadBase();
-
- enum type {
+ enum type_t {
MIXER, // Thread class is MixerThread
DIRECT, // Thread class is DirectOutputThread
DUPLICATING, // Thread class is DuplicatingThread
RECORD // Thread class is RecordThread
};
+ ThreadBase (const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, uint32_t device, type_t type);
+ virtual ~ThreadBase();
+
status_t dumpBase(int fd, const Vector<String16>& args);
status_t dumpEffectChains(int fd, const Vector<String16>& args);
@@ -305,6 +303,8 @@
enum track_state {
IDLE,
TERMINATED,
+ // These are order-sensitive; do not change order without reviewing the impact.
+ // In particular there are assumptions about > STOPPED.
STOPPED,
RESUMING,
ACTIVE,
@@ -312,28 +312,21 @@
PAUSED
};
- enum track_flags {
- STEPSERVER_FAILED = 0x01, // StepServer could not acquire cblk->lock mutex
- SYSTEM_FLAGS_MASK = 0x0000ffffUL,
- // The upper 16 bits are used for track-specific flags.
- };
-
- TrackBase(const wp<ThreadBase>& thread,
+ TrackBase(ThreadBase *thread,
const sp<Client>& client,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
const sp<IMemory>& sharedBuffer,
int sessionId);
- ~TrackBase();
+ virtual ~TrackBase();
- virtual status_t start() = 0;
+ virtual status_t start(pid_t tid) = 0;
virtual void stop() = 0;
- sp<IMemory> getCblk() const;
+ sp<IMemory> getCblk() const { return mCblkMemory; }
audio_track_cblk_t* cblk() const { return mCblk; }
- int sessionId() { return mSessionId; }
+ int sessionId() const { return mSessionId; }
protected:
friend class ThreadBase;
@@ -346,18 +339,20 @@
TrackBase(const TrackBase&);
TrackBase& operator = (const TrackBase&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
+ virtual status_t getNextBuffer(
+ AudioBufferProvider::Buffer* buffer,
+ int64_t pts) = 0;
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
- uint32_t format() const {
+ audio_format_t format() const {
return mFormat;
}
- int channelCount() const ;
+ int channelCount() const { return mChannelCount; }
- uint32_t channelMask() const;
+ uint32_t channelMask() const { return mChannelMask; }
- int sampleRate() const;
+ int sampleRate() const; // FIXME inline after cblk sr moved
void* getBuffer(uint32_t offset, uint32_t frames) const;
@@ -372,19 +367,18 @@
bool step();
void reset();
- wp<ThreadBase> mThread;
- sp<Client> mClient;
+ const wp<ThreadBase> mThread;
+ /*const*/ sp<Client> mClient; // see explanation at ~TrackBase() why not const
sp<IMemory> mCblkMemory;
audio_track_cblk_t* mCblk;
void* mBuffer;
void* mBufferEnd;
uint32_t mFrameCount;
// we don't really need a lock for these
- int mState;
- int mClientTid;
- uint32_t mFormat;
- uint32_t mFlags;
- int mSessionId;
+ track_state mState;
+ const audio_format_t mFormat;
+ bool mStepServerFailed;
+ const int mSessionId;
uint8_t mChannelCount;
uint32_t mChannelMask;
};
@@ -413,12 +407,14 @@
};
virtual status_t initCheck() const = 0;
- int type() const { return mType; }
- uint32_t sampleRate() const;
- int channelCount() const;
- uint32_t format() const;
- size_t frameCount() const;
+ type_t type() const { return mType; }
+ uint32_t sampleRate() const { return mSampleRate; }
+ int channelCount() const { return mChannelCount; }
+ audio_format_t format() const { return mFormat; }
+ size_t frameCount() const { return mFrameCount; }
void wakeUp() { mWaitWorkCV.broadcast(); }
+ // Should be "virtual status_t requestExitAndWait()" and override same
+ // method in Thread, but Thread::requestExitAndWait() is not yet virtual.
void exit();
virtual bool checkForNewParameters_l() = 0;
virtual status_t setParameters(const String8& keyValuePairs);
@@ -427,7 +423,7 @@
void sendConfigEvent(int event, int param = 0);
void sendConfigEvent_l(int event, int param = 0);
void processConfigEvents();
- int id() const { return mId;}
+ audio_io_handle_t id() const { return mId;}
bool standby() { return mStandby; }
uint32_t device() { return mDevice; }
virtual audio_stream_t* stream() = 0;
@@ -442,7 +438,7 @@
status_t *status);
void disconnectEffect(const sp< EffectModule>& effect,
const wp<EffectHandle>& handle,
- bool unpiniflast);
+ bool unpinIfLast);
// return values for hasAudioSession (bit field)
enum effect_state {
@@ -467,7 +463,7 @@
// unlock effect chains after process
void unlockEffectChains(Vector<sp <EffectChain> >& effectChains);
// set audio mode to all effect chains
- void setMode(uint32_t mode);
+ void setMode(audio_mode_t mode);
// get effect module with corresponding ID on specified audio session
sp<AudioFlinger::EffectModule> getEffect_l(int sessionId, int effectId);
// add and effect module. Also creates the effect chain is none exists for
@@ -535,22 +531,21 @@
friend class RecordThread;
friend class RecordTrack;
- int mType;
+ const type_t mType;
Condition mWaitWorkCV;
- sp<AudioFlinger> mAudioFlinger;
+ const sp<AudioFlinger> mAudioFlinger;
uint32_t mSampleRate;
size_t mFrameCount;
uint32_t mChannelMask;
uint16_t mChannelCount;
- uint16_t mFrameSize;
- uint32_t mFormat;
+ size_t mFrameSize;
+ audio_format_t mFormat;
Condition mParamCond;
Vector<String8> mNewParameters;
status_t mParamStatus;
- Vector<ConfigEvent *> mConfigEvents;
+ Vector<ConfigEvent> mConfigEvents;
bool mStandby;
- int mId;
- bool mExiting;
+ const audio_io_handle_t mId;
Vector< sp<EffectChain> > mEffectChains;
uint32_t mDevice; // output device for PlaybackThread
// input + output devices for RecordThread
@@ -558,12 +553,24 @@
char mName[kNameLength];
sp<IPowerManager> mPowerManager;
sp<IBinder> mWakeLockToken;
- sp<PMDeathRecipient> mDeathRecipient;
+ const sp<PMDeathRecipient> mDeathRecipient;
// list of suspended effects per session and per type. The first vector is
// keyed by session ID, the second by type UUID timeLow field
KeyedVector< int, KeyedVector< int, sp<SuspendedSessionDesc> > > mSuspendedSessions;
};
+ struct stream_type_t {
+ stream_type_t()
+ : volume(1.0f),
+ mute(false),
+ valid(true)
+ {
+ }
+ float volume;
+ bool mute;
+ bool valid;
+ };
+
// --- PlaybackThread ---
class PlaybackThread : public ThreadBase {
public:
@@ -577,40 +584,38 @@
// playback track
class Track : public TrackBase {
public:
- Track( const wp<ThreadBase>& thread,
+ Track( PlaybackThread *thread,
const sp<Client>& client,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId);
- ~Track();
+ virtual ~Track();
void dump(char* buffer, size_t size);
- virtual status_t start();
+ virtual status_t start(pid_t tid);
virtual void stop();
void pause();
void flush();
void destroy();
void mute(bool);
- void setVolume(float left, float right);
int name() const {
return mName;
}
- int type() const {
+ audio_stream_type_t streamType() const {
return mStreamType;
}
status_t attachAuxEffect(int EffectId);
void setAuxBuffer(int EffectId, int32_t *buffer);
- int32_t *auxBuffer() { return mAuxBuffer; }
+ int32_t *auxBuffer() const { return mAuxBuffer; }
void setMainBuffer(int16_t *buffer) { mMainBuffer = buffer; }
- int16_t *mainBuffer() { return mMainBuffer; }
- int auxEffectId() { return mAuxEffectId; }
-
+ int16_t *mainBuffer() const { return mMainBuffer; }
+ int auxEffectId() const { return mAuxEffectId; }
protected:
friend class ThreadBase;
@@ -622,8 +627,12 @@
Track(const Track&);
Track& operator = (const Track&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
- bool isMuted() { return mMute; }
+ virtual status_t getNextBuffer(
+ AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
+ virtual uint32_t framesReady() const;
+
+ bool isMuted() const { return mMute; }
bool isPausing() const {
return mState == PAUSING;
}
@@ -638,8 +647,9 @@
return (mStreamType == AUDIO_STREAM_CNT);
}
+ virtual bool isTimedTrack() const { return false; }
+
// we don't really need a lock for these
- float mVolume[2];
volatile bool mMute;
// FILLED state is used for suppressing volume ramp at begin of playing
enum {FS_FILLING, FS_FILLED, FS_ACTIVE};
@@ -647,7 +657,7 @@
int8_t mRetryCount;
sp<IMemory> mSharedBuffer;
bool mResetDone;
- int mStreamType;
+ audio_stream_type_t mStreamType;
int mName;
int16_t *mMainBuffer;
int32_t *mAuxBuffer;
@@ -655,6 +665,79 @@
bool mHasVolumeController;
}; // end of Track
+ class TimedTrack : public Track {
+ public:
+ static sp<TimedTrack> create(PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId);
+ ~TimedTrack();
+
+ class TimedBuffer {
+ public:
+ TimedBuffer();
+ TimedBuffer(const sp<IMemory>& buffer, int64_t pts);
+ const sp<IMemory>& buffer() const { return mBuffer; }
+ int64_t pts() const { return mPTS; }
+ int position() const { return mPosition; }
+ void setPosition(int pos) { mPosition = pos; }
+ private:
+ sp<IMemory> mBuffer;
+ int64_t mPTS;
+ int mPosition;
+ };
+
+ virtual bool isTimedTrack() const { return true; }
+
+ virtual uint32_t framesReady() const;
+
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
+ virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
+ void timedYieldSamples(AudioBufferProvider::Buffer* buffer);
+ void timedYieldSilence(uint32_t numFrames,
+ AudioBufferProvider::Buffer* buffer);
+
+ status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer);
+ status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts);
+ status_t setMediaTimeTransform(const LinearTransform& xform,
+ TimedAudioTrack::TargetTimeline target);
+ void trimTimedBufferQueue_l();
+
+ private:
+ TimedTrack(PlaybackThread *thread,
+ const sp<Client>& client,
+ audio_stream_type_t streamType,
+ uint32_t sampleRate,
+ audio_format_t format,
+ uint32_t channelMask,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ int sessionId);
+
+ uint64_t mLocalTimeFreq;
+ LinearTransform mLocalTimeToSampleTransform;
+ sp<MemoryDealer> mTimedMemoryDealer;
+ Vector<TimedBuffer> mTimedBufferQueue;
+ uint8_t* mTimedSilenceBuffer;
+ uint32_t mTimedSilenceBufferSize;
+ mutable Mutex mTimedBufferQueueLock;
+ bool mTimedAudioOutputOnTime;
+ CCHelper mCCHelper;
+
+ Mutex mMediaTimeTransformLock;
+ LinearTransform mMediaTimeTransform;
+ bool mMediaTimeTransformValid;
+ TimedAudioTrack::TargetTimeline mMediaTimeTransformTarget;
+ };
+
// playback track
class OutputTrack : public Track {
@@ -665,23 +748,27 @@
int16_t *mBuffer;
};
- OutputTrack( const wp<ThreadBase>& thread,
+ OutputTrack(PlaybackThread *thread,
DuplicatingThread *sourceThread,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount);
- ~OutputTrack();
+ virtual ~OutputTrack();
- virtual status_t start();
+ virtual status_t start(pid_t tid);
virtual void stop();
bool write(int16_t* data, uint32_t frames);
- bool bufferQueueEmpty() { return (mBufferQueue.size() == 0) ? true : false; }
- bool isActive() { return mActive; }
- wp<ThreadBase>& thread() { return mThread; }
+ bool bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
+ bool isActive() const { return mActive; }
+ const wp<ThreadBase>& thread() const { return mThread; }
private:
+ enum {
+ NO_MORE_BUFFERS = 0x80000001, // same in AudioTrack.h, ok to be different value
+ };
+
status_t obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs);
void clearBufferQueue();
@@ -691,10 +778,11 @@
Vector < Buffer* > mBufferQueue;
AudioBufferProvider::Buffer mOutBuffer;
bool mActive;
- DuplicatingThread* mSourceThread;
+ DuplicatingThread* const mSourceThread; // for waitTimeMs() in write()
}; // end of OutputTrack
- PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
+ PlaybackThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, uint32_t device, type_t type);
virtual ~PlaybackThread();
virtual status_t dump(int fd, const Vector<String16>& args);
@@ -703,44 +791,41 @@
virtual status_t readyToRun();
virtual void onFirstRef();
- virtual status_t initCheck() const { return (mOutput == 0) ? NO_INIT : NO_ERROR; }
+ virtual status_t initCheck() const { return (mOutput == NULL) ? NO_INIT : NO_ERROR; }
virtual uint32_t latency() const;
- virtual status_t setMasterVolume(float value);
- virtual status_t setMasterMute(bool muted);
+ void setMasterVolume(float value);
+ void setMasterMute(bool muted);
- virtual float masterVolume() const;
- virtual bool masterMute() const;
+ void setStreamVolume(audio_stream_type_t stream, float value);
+ void setStreamMute(audio_stream_type_t stream, bool muted);
- virtual status_t setStreamVolume(int stream, float value);
- virtual status_t setStreamMute(int stream, bool muted);
-
- virtual float streamVolume(int stream) const;
- virtual bool streamMute(int stream) const;
+ float streamVolume(audio_stream_type_t stream) const;
sp<Track> createTrack_l(
const sp<AudioFlinger::Client>& client,
- int streamType,
+ audio_stream_type_t streamType,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
const sp<IMemory>& sharedBuffer,
int sessionId,
+ bool isTimed,
status_t *status);
- AudioStreamOut* getOutput();
+ AudioStreamOut* getOutput() const;
AudioStreamOut* clearOutput();
virtual audio_stream_t* stream();
void suspend() { mSuspended++; }
void restore() { if (mSuspended) mSuspended--; }
- bool isSuspended() { return (mSuspended != 0); }
+ bool isSuspended() const { return (mSuspended != 0); }
virtual String8 getParameters(const String8& keys);
virtual void audioConfigChanged_l(int event, int param = 0);
virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
- int16_t *mixBuffer() { return mMixBuffer; };
+ int16_t *mixBuffer() const { return mMixBuffer; };
virtual void detachAuxEffect_l(int effectId);
status_t attachAuxEffect(const sp<AudioFlinger::PlaybackThread::Track> track,
@@ -753,25 +838,19 @@
virtual uint32_t hasAudioSession(int sessionId);
virtual uint32_t getStrategyForSession_l(int sessionId);
- void setStreamValid(int streamType, bool valid);
-
- struct stream_type_t {
- stream_type_t()
- : volume(1.0f),
- mute(false),
- valid(true)
- {
- }
- float volume;
- bool mute;
- bool valid;
- };
+ void setStreamValid(audio_stream_type_t streamType, bool valid);
protected:
int16_t* mMixBuffer;
int mSuspended;
int mBytesWritten;
+ private:
+ // mMasterMute is in both PlaybackThread and in AudioFlinger. When a
+ // PlaybackThread needs to find out if master-muted, it checks it's local
+ // copy rather than the one in AudioFlinger. This optimization saves a lock.
bool mMasterMute;
+ void setMasterMute_l(bool muted) { mMasterMute = muted; }
+ protected:
SortedVector< wp<Track> > mActiveTracks;
virtual int getTrackName_l() = 0;
@@ -780,6 +859,9 @@
virtual uint32_t idleSleepTimeUs() = 0;
virtual uint32_t suspendSleepTimeUs() = 0;
+ // Code snippets that are temporarily lifted up out of threadLoop() until the merge
+ void checkSilentMode_l();
+
private:
friend class AudioFlinger;
@@ -803,9 +885,9 @@
status_t dumpTracks(int fd, const Vector<String16>& args);
SortedVector< sp<Track> > mTracks;
- // mStreamTypes[] uses 1 additionnal stream type internally for the OutputTrack used by DuplicatingThread
+ // mStreamTypes[] uses 1 additional stream type internally for the OutputTrack used by DuplicatingThread
stream_type_t mStreamTypes[AUDIO_STREAM_CNT + 1];
- AudioStreamOut* mOutput;
+ AudioStreamOut *mOutput;
float mMasterVolume;
nsecs_t mLastWriteTime;
int mNumWrites;
@@ -817,19 +899,20 @@
public:
MixerThread (const sp<AudioFlinger>& audioFlinger,
AudioStreamOut* output,
- int id,
- uint32_t device);
+ audio_io_handle_t id,
+ uint32_t device,
+ type_t type = MIXER);
virtual ~MixerThread();
// Thread virtuals
virtual bool threadLoop();
- void invalidateTracks(int streamType);
+ void invalidateTracks(audio_stream_type_t streamType);
virtual bool checkForNewParameters_l();
virtual status_t dumpInternals(int fd, const Vector<String16>& args);
protected:
- uint32_t prepareTracks_l(const SortedVector< wp<Track> >& activeTracks,
+ mixer_state prepareTracks_l(const SortedVector< wp<Track> >& activeTracks,
Vector< sp<Track> > *tracksToRemove);
virtual int getTrackName_l();
virtual void deleteTrackName_l(int name);
@@ -837,15 +920,15 @@
virtual uint32_t suspendSleepTimeUs();
AudioMixer* mAudioMixer;
- uint32_t mPrevMixerStatus; // previous status (mixer_state) returned by
- // prepareTracks_l()
+ mixer_state mPrevMixerStatus; // previous status returned by prepareTracks_l()
};
class DirectOutputThread : public PlaybackThread {
public:
- DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device);
- ~DirectOutputThread();
+ DirectOutputThread (const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
+ audio_io_handle_t id, uint32_t device);
+ virtual ~DirectOutputThread();
// Thread virtuals
virtual bool threadLoop();
@@ -862,6 +945,8 @@
private:
void applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp);
+ // volumes last sent to audio HAL with stream->set_volume()
+ // FIXME use standard representation and names
float mLeftVolFloat;
float mRightVolFloat;
uint16_t mLeftVolShort;
@@ -870,8 +955,9 @@
class DuplicatingThread : public MixerThread {
public:
- DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread, int id);
- ~DuplicatingThread();
+ DuplicatingThread (const sp<AudioFlinger>& audioFlinger, MixerThread* mainThread,
+ audio_io_handle_t id);
+ virtual ~DuplicatingThread();
// Thread virtuals
virtual bool threadLoop();
@@ -889,38 +975,51 @@
uint32_t mWaitTimeMs;
};
- PlaybackThread *checkPlaybackThread_l(int output) const;
- MixerThread *checkMixerThread_l(int output) const;
- RecordThread *checkRecordThread_l(int input) const;
- float streamVolumeInternal(int stream) const { return mStreamTypes[stream].volume; }
- void audioConfigChanged_l(int event, int ioHandle, void *param2);
+ PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const;
+ MixerThread *checkMixerThread_l(audio_io_handle_t output) const;
+ RecordThread *checkRecordThread_l(audio_io_handle_t input) const;
+ // no range check, AudioFlinger::mLock held
+ bool streamMute_l(audio_stream_type_t stream) const
+ { return mStreamTypes[stream].mute; }
+ // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held
+ float streamVolume_l(audio_stream_type_t stream) const
+ { return mStreamTypes[stream].volume; }
+ void audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2);
+ // allocate an audio_io_handle_t, session ID, or effect ID
uint32_t nextUniqueId();
+
status_t moveEffectChain_l(int sessionId,
- AudioFlinger::PlaybackThread *srcThread,
- AudioFlinger::PlaybackThread *dstThread,
+ PlaybackThread *srcThread,
+ PlaybackThread *dstThread,
bool reRegister);
PlaybackThread *primaryPlaybackThread_l();
uint32_t primaryOutputDevice_l();
friend class AudioBuffer;
+ // server side of the client's IAudioTrack
class TrackHandle : public android::BnAudioTrack {
public:
TrackHandle(const sp<PlaybackThread::Track>& track);
virtual ~TrackHandle();
- virtual status_t start();
+ virtual sp<IMemory> getCblk() const;
+ virtual status_t start(pid_t tid);
virtual void stop();
virtual void flush();
virtual void mute(bool);
virtual void pause();
- virtual void setVolume(float left, float right);
- virtual sp<IMemory> getCblk() const;
virtual status_t attachAuxEffect(int effectId);
+ virtual status_t allocateTimedBuffer(size_t size,
+ sp<IMemory>* buffer);
+ virtual status_t queueTimedBuffer(const sp<IMemory>& buffer,
+ int64_t pts);
+ virtual status_t setMediaTimeTransform(const LinearTransform& xform,
+ int target);
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<PlaybackThread::Track> mTrack;
+ const sp<PlaybackThread::Track> mTrack;
};
friend class Client;
@@ -939,17 +1038,16 @@
// record track
class RecordTrack : public TrackBase {
public:
- RecordTrack(const wp<ThreadBase>& thread,
+ RecordTrack(RecordThread *thread,
const sp<Client>& client,
uint32_t sampleRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channelMask,
int frameCount,
- uint32_t flags,
int sessionId);
- ~RecordTrack();
+ virtual ~RecordTrack();
- virtual status_t start();
+ virtual status_t start(pid_t tid);
virtual void stop();
bool overflow() { bool tmp = mOverflow; mOverflow = false; return tmp; }
@@ -964,7 +1062,9 @@
RecordTrack(const RecordTrack&);
RecordTrack& operator = (const RecordTrack&);
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual status_t getNextBuffer(
+ AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
bool mOverflow;
};
@@ -974,33 +1074,34 @@
AudioStreamIn *input,
uint32_t sampleRate,
uint32_t channels,
- int id,
+ audio_io_handle_t id,
uint32_t device);
- ~RecordThread();
+ virtual ~RecordThread();
virtual bool threadLoop();
virtual status_t readyToRun();
virtual void onFirstRef();
- virtual status_t initCheck() const { return (mInput == 0) ? NO_INIT : NO_ERROR; }
+ virtual status_t initCheck() const { return (mInput == NULL) ? NO_INIT : NO_ERROR; }
sp<AudioFlinger::RecordThread::RecordTrack> createRecordTrack_l(
const sp<AudioFlinger::Client>& client,
uint32_t sampleRate,
- int format,
+ audio_format_t format,
int channelMask,
int frameCount,
- uint32_t flags,
int sessionId,
status_t *status);
status_t start(RecordTrack* recordTrack);
+ status_t start(RecordTrack* recordTrack, pid_t tid);
void stop(RecordTrack* recordTrack);
status_t dump(int fd, const Vector<String16>& args);
- AudioStreamIn* getInput();
+ AudioStreamIn* getInput() const;
AudioStreamIn* clearInput();
virtual audio_stream_t* stream();
- virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer);
+ virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer,
+ int64_t pts);
virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
virtual bool checkForNewParameters_l();
virtual String8 getParameters(const String8& keys);
@@ -1024,22 +1125,23 @@
int16_t *mRsmpInBuffer;
size_t mRsmpInIndex;
size_t mInputBytes;
- int mReqChannelCount;
- uint32_t mReqSampleRate;
+ const int mReqChannelCount;
+ const uint32_t mReqSampleRate;
ssize_t mBytesRead;
};
+ // server side of the client's IAudioRecord
class RecordHandle : public android::BnAudioRecord {
public:
RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack);
virtual ~RecordHandle();
- virtual status_t start();
- virtual void stop();
virtual sp<IMemory> getCblk() const;
+ virtual status_t start(pid_t tid);
+ virtual void stop();
virtual status_t onTransact(
uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags);
private:
- sp<RecordThread::RecordTrack> mRecordTrack;
+ const sp<RecordThread::RecordTrack> mRecordTrack;
};
//--- Audio Effect Management
@@ -1060,12 +1162,12 @@
// the attached track(s) to accumulate their auxiliary channel.
class EffectModule: public RefBase {
public:
- EffectModule(const wp<ThreadBase>& wThread,
+ EffectModule(ThreadBase *thread,
const wp<AudioFlinger::EffectChain>& chain,
effect_descriptor_t *desc,
int id,
int sessionId);
- ~EffectModule();
+ virtual ~EffectModule();
enum effect_state {
IDLE,
@@ -1077,7 +1179,7 @@
DESTROYED
};
- int id() { return mId; }
+ int id() const { return mId; }
void process();
void updateState();
status_t command(uint32_t cmdCode,
@@ -1089,18 +1191,18 @@
void reset_l();
status_t configure();
status_t init();
- uint32_t state() {
+ effect_state state() const {
return mState;
}
uint32_t status() {
return mStatus;
}
- int sessionId() {
+ int sessionId() const {
return mSessionId;
}
status_t setEnabled(bool enabled);
- bool isEnabled();
- bool isProcessEnabled();
+ bool isEnabled() const;
+ bool isProcessEnabled() const;
void setInBuffer(int16_t *buffer) { mConfig.inputCfg.buffer.s16 = buffer; }
int16_t *inBuffer() { return mConfig.inputCfg.buffer.s16; }
@@ -1108,10 +1210,10 @@
int16_t *outBuffer() { return mConfig.outputCfg.buffer.s16; }
void setChain(const wp<EffectChain>& chain) { mChain = chain; }
void setThread(const wp<ThreadBase>& thread) { mThread = thread; }
- wp<ThreadBase>& thread() { return mThread; }
+ const wp<ThreadBase>& thread() { return mThread; }
- status_t addHandle(sp<EffectHandle>& handle);
- void disconnect(const wp<EffectHandle>& handle, bool unpiniflast);
+ status_t addHandle(const sp<EffectHandle>& handle);
+ void disconnect(const wp<EffectHandle>& handle, bool unpinIfLast);
size_t removeHandle (const wp<EffectHandle>& handle);
effect_descriptor_t& desc() { return mDescriptor; }
@@ -1119,15 +1221,15 @@
status_t setDevice(uint32_t device);
status_t setVolume(uint32_t *left, uint32_t *right, bool controller);
- status_t setMode(uint32_t mode);
+ status_t setMode(audio_mode_t mode);
status_t start();
status_t stop();
void setSuspended(bool suspended);
- bool suspended();
+ bool suspended() const;
sp<EffectHandle> controlHandle();
- bool isPinned() { return mPinned; }
+ bool isPinned() const { return mPinned; }
void unPin() { mPinned = false; }
status_t dump(int fd, const Vector<String16>& args);
@@ -1146,7 +1248,7 @@
status_t start_l();
status_t stop_l();
- Mutex mLock; // mutex for process, commands and handles list protection
+mutable Mutex mLock; // mutex for process, commands and handles list protection
wp<ThreadBase> mThread; // parent thread
wp<EffectChain> mChain; // parent effect chain
int mId; // this instance unique ID
@@ -1154,9 +1256,10 @@
effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
effect_config_t mConfig; // input and output audio configuration
effect_handle_t mEffectInterface; // Effect module C API
- status_t mStatus; // initialization status
- uint32_t mState; // current activation state (effect_state)
+ status_t mStatus; // initialization status
+ effect_state mState; // current activation state
Vector< wp<EffectHandle> > mHandles; // list of client handles
+ // First handle in mHandles has highest priority and controls the effect module
uint32_t mMaxDisableWaitCnt; // maximum grace period before forcing an effect off after
// sending disable command.
uint32_t mDisableWaitCnt; // current process() calls count during disable period.
@@ -1187,8 +1290,10 @@
uint32_t *replySize,
void *pReplyData);
virtual void disconnect();
- virtual void disconnect(bool unpiniflast);
- virtual sp<IMemory> getCblk() const;
+ private:
+ void disconnect(bool unpinIfLast);
+ public:
+ virtual sp<IMemory> getCblk() const { return mCblkMemory; }
virtual status_t onTransact(uint32_t code, const Parcel& data,
Parcel* reply, uint32_t flags);
@@ -1204,13 +1309,13 @@
uint32_t replySize,
void *pReplyData);
void setEnabled(bool enabled);
- bool enabled() { return mEnabled; }
+ bool enabled() const { return mEnabled; }
// Getters
- int id() { return mEffect->id(); }
- int priority() { return mPriority; }
- bool hasControl() { return mHasControl; }
- sp<EffectModule> effect() { return mEffect; }
+ int id() const { return mEffect->id(); }
+ int priority() const { return mPriority; }
+ bool hasControl() const { return mHasControl; }
+ sp<EffectModule> effect() const { return mEffect; }
void dump(char* buffer, size_t size);
@@ -1222,7 +1327,7 @@
sp<EffectModule> mEffect; // pointer to controlled EffectModule
sp<IEffectClient> mEffectClient; // callback interface for client notifications
- sp<Client> mClient; // client for shared memory allocation
+ /*const*/ sp<Client> mClient; // client for shared memory allocation, see disconnect()
sp<IMemory> mCblkMemory; // shared memory for control block
effect_param_cblk_t* mCblk; // control block for deferred parameter setting via shared memory
uint8_t* mBuffer; // pointer to parameter area in shared memory
@@ -1242,7 +1347,8 @@
class EffectChain: public RefBase {
public:
EffectChain(const wp<ThreadBase>& wThread, int sessionId);
- ~EffectChain();
+ EffectChain(ThreadBase *thread, int sessionId);
+ virtual ~EffectChain();
// special key used for an entry in mSuspendedEffects keyed vector
// corresponding to a suspend all request.
@@ -1264,7 +1370,7 @@
status_t addEffect_l(const sp<EffectModule>& handle);
size_t removeEffect_l(const sp<EffectModule>& handle);
- int sessionId() { return mSessionId; }
+ int sessionId() const { return mSessionId; }
void setSessionId(int sessionId) { mSessionId = sessionId; }
sp<EffectModule> getEffectFromDesc_l(effect_descriptor_t *descriptor);
@@ -1272,32 +1378,32 @@
sp<EffectModule> getEffectFromType_l(const effect_uuid_t *type);
bool setVolume_l(uint32_t *left, uint32_t *right);
void setDevice_l(uint32_t device);
- void setMode_l(uint32_t mode);
+ void setMode_l(audio_mode_t mode);
void setInBuffer(int16_t *buffer, bool ownsBuffer = false) {
mInBuffer = buffer;
mOwnInBuffer = ownsBuffer;
}
- int16_t *inBuffer() {
+ int16_t *inBuffer() const {
return mInBuffer;
}
void setOutBuffer(int16_t *buffer) {
mOutBuffer = buffer;
}
- int16_t *outBuffer() {
+ int16_t *outBuffer() const {
return mOutBuffer;
}
void incTrackCnt() { android_atomic_inc(&mTrackCnt); }
void decTrackCnt() { android_atomic_dec(&mTrackCnt); }
- int32_t trackCnt() { return mTrackCnt;}
+ int32_t trackCnt() const { return mTrackCnt;}
void incActiveTrackCnt() { android_atomic_inc(&mActiveTrackCnt);
mTailBufferCount = mMaxTailBuffers; }
void decActiveTrackCnt() { android_atomic_dec(&mActiveTrackCnt); }
- int32_t activeTrackCnt() { return mActiveTrackCnt;}
+ int32_t activeTrackCnt() const { return mActiveTrackCnt;}
- uint32_t strategy() { return mStrategy; }
+ uint32_t strategy() const { return mStrategy; }
void setStrategy(uint32_t strategy)
{ mStrategy = strategy; }
@@ -1328,7 +1434,8 @@
// get a list of effect modules to suspend when an effect of the type
// passed is enabled.
- Vector< sp<EffectModule> > getSuspendEligibleEffects();
+ void getSuspendEligibleEffects(Vector< sp<EffectModule> > &effects);
+
// get an effect module if it is currently enable
sp<EffectModule> getEffectIfEnabled(const effect_uuid_t *type);
// true if the effect whose descriptor is passed can be suspended
@@ -1359,54 +1466,121 @@
KeyedVector< int, sp<SuspendedEffectDesc> > mSuspendedEffects;
};
+ // AudioStreamOut and AudioStreamIn are immutable, so their fields are const.
+ // For emphasis, we could also make all pointers to them be "const *",
+ // but that would clutter the code unnecessarily.
+
struct AudioStreamOut {
- audio_hw_device_t *hwDev;
- audio_stream_out_t *stream;
+ audio_hw_device_t* const hwDev;
+ audio_stream_out_t* const stream;
AudioStreamOut(audio_hw_device_t *dev, audio_stream_out_t *out) :
hwDev(dev), stream(out) {}
};
struct AudioStreamIn {
- audio_hw_device_t *hwDev;
- audio_stream_in_t *stream;
+ audio_hw_device_t* const hwDev;
+ audio_stream_in_t* const stream;
AudioStreamIn(audio_hw_device_t *dev, audio_stream_in_t *in) :
hwDev(dev), stream(in) {}
};
+ // for mAudioSessionRefs only
struct AudioSessionRef {
- int sessionid;
- pid_t pid;
+ // FIXME rename parameter names when fields get "m" prefix
+ AudioSessionRef(int sessionid_, pid_t pid_) :
+ sessionid(sessionid_), pid(pid_), cnt(1) {}
+ const int sessionid;
+ const pid_t pid;
int cnt;
};
friend class RecordThread;
friend class PlaybackThread;
+ enum master_volume_support {
+ // MVS_NONE:
+ // Audio HAL has no support for master volume, either setting or
+ // getting. All master volume control must be implemented in SW by the
+ // AudioFlinger mixing core.
+ MVS_NONE,
+
+ // MVS_SETONLY:
+ // Audio HAL has support for setting master volume, but not for getting
+ // master volume (original HAL design did not include a getter).
+ // AudioFlinger needs to keep track of the last set master volume in
+ // addition to needing to set an initial, default, master volume at HAL
+ // load time.
+ MVS_SETONLY,
+
+ // MVS_FULL:
+ // Audio HAL has support both for setting and getting master volume.
+ // AudioFlinger should send all set and get master volume requests
+ // directly to the HAL.
+ MVS_FULL,
+ };
+
mutable Mutex mLock;
- DefaultKeyedVector< pid_t, wp<Client> > mClients;
+ DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client()
mutable Mutex mHardwareLock;
- audio_hw_device_t* mPrimaryHardwareDev;
+
+ // These two fields are immutable after onFirstRef(), so no lock needed to access
+ audio_hw_device_t* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL
Vector<audio_hw_device_t*> mAudioHwDevs;
- mutable int mHardwareStatus;
+
+ enum hardware_call_state {
+ AUDIO_HW_IDLE = 0,
+ AUDIO_HW_INIT,
+ AUDIO_HW_OUTPUT_OPEN,
+ AUDIO_HW_OUTPUT_CLOSE,
+ AUDIO_HW_INPUT_OPEN,
+ AUDIO_HW_INPUT_CLOSE,
+ AUDIO_HW_STANDBY,
+ AUDIO_HW_SET_MASTER_VOLUME,
+ AUDIO_HW_GET_ROUTING,
+ AUDIO_HW_SET_ROUTING,
+ AUDIO_HW_GET_MODE,
+ AUDIO_HW_SET_MODE,
+ AUDIO_HW_GET_MIC_MUTE,
+ AUDIO_HW_SET_MIC_MUTE,
+ AUDIO_SET_VOICE_VOLUME,
+ AUDIO_SET_PARAMETER,
+ AUDIO_HW_GET_INPUT_BUFFER_SIZE,
+ AUDIO_HW_GET_MASTER_VOLUME,
+ };
+
+ mutable hardware_call_state mHardwareStatus; // for dump only
- DefaultKeyedVector< int, sp<PlaybackThread> > mPlaybackThreads;
- PlaybackThread::stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
+ DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads;
+ stream_type_t mStreamTypes[AUDIO_STREAM_CNT];
+
+ // both are protected by mLock
float mMasterVolume;
+ float mMasterVolumeSW;
+ master_volume_support mMasterVolumeSupportLvl;
bool mMasterMute;
- DefaultKeyedVector< int, sp<RecordThread> > mRecordThreads;
+ DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads;
DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients;
- volatile int32_t mNextUniqueId;
- uint32_t mMode;
+ volatile int32_t mNextUniqueId; // updated by android_atomic_inc
+ audio_mode_t mMode;
bool mBtNrecIsOff;
+ // protected by mLock
Vector<AudioSessionRef*> mAudioSessionRefs;
+
+ float masterVolume_l() const;
+ float masterVolumeSW_l() const { return mMasterVolumeSW; }
+ bool masterMute_l() const { return mMasterMute; }
+
+private:
+ sp<Client> registerPid_l(pid_t pid); // always returns non-0
+
};
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 9dda256..020d62a 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioMixer.cpp
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -18,6 +18,7 @@
#define LOG_TAG "AudioMixer"
//#define LOG_NDEBUG 0
+#include <assert.h>
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
@@ -27,240 +28,236 @@
#include <utils/Log.h>
#include <cutils/bitops.h>
+#include <cutils/compiler.h>
#include <system/audio.h>
+#include <audio_utils/primitives.h>
+#include <common_time/local_clock.h>
+#include <common_time/cc_helper.h>
+
#include "AudioMixer.h"
namespace android {
-// ----------------------------------------------------------------------------
-
-static inline int16_t clamp16(int32_t sample)
-{
- if ((sample>>15) ^ (sample>>31))
- sample = 0x7FFF ^ (sample>>31);
- return sample;
-}
// ----------------------------------------------------------------------------
AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate)
- : mActiveTrack(0), mTrackNames(0), mSampleRate(sampleRate)
+ : mTrackNames(0), mSampleRate(sampleRate)
{
+ // AudioMixer is not yet capable of multi-channel beyond stereo
+ assert(2 == MAX_NUM_CHANNELS);
+
+ LocalClock lc;
+
mState.enabledTracks= 0;
mState.needsChanged = 0;
mState.frameCount = frameCount;
- mState.outputTemp = 0;
- mState.resampleTemp = 0;
mState.hook = process__nop;
+ mState.outputTemp = NULL;
+ mState.resampleTemp = NULL;
+ // mState.reserved
track_t* t = mState.tracks;
- for (int i=0 ; i<32 ; i++) {
+ for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
t->needs = 0;
t->volume[0] = UNITY_GAIN;
t->volume[1] = UNITY_GAIN;
+ // no initialization needed
+ // t->prevVolume[0]
+ // t->prevVolume[1]
t->volumeInc[0] = 0;
t->volumeInc[1] = 0;
t->auxLevel = 0;
t->auxInc = 0;
+ // no initialization needed
+ // t->prevAuxLevel
+ // t->frameCount
t->channelCount = 2;
- t->enabled = 0;
+ t->enabled = false;
t->format = 16;
t->channelMask = AUDIO_CHANNEL_OUT_STEREO;
- t->buffer.raw = 0;
- t->bufferProvider = 0;
- t->hook = 0;
- t->resampler = 0;
+ t->bufferProvider = NULL;
+ t->buffer.raw = NULL;
+ // t->buffer.frameCount
+ t->hook = NULL;
+ t->in = NULL;
+ t->resampler = NULL;
t->sampleRate = mSampleRate;
- t->in = 0;
t->mainBuffer = NULL;
t->auxBuffer = NULL;
+ t->localTimeFreq = lc.getLocalFreq();
t++;
}
}
- AudioMixer::~AudioMixer()
- {
- track_t* t = mState.tracks;
- for (int i=0 ; i<32 ; i++) {
- delete t->resampler;
- t++;
- }
- delete [] mState.outputTemp;
- delete [] mState.resampleTemp;
- }
-
- int AudioMixer::getTrackName()
- {
- uint32_t names = mTrackNames;
- uint32_t mask = 1;
- int n = 0;
- while (names & mask) {
- mask <<= 1;
- n++;
+AudioMixer::~AudioMixer()
+{
+ track_t* t = mState.tracks;
+ for (unsigned i=0 ; i < MAX_NUM_TRACKS ; i++) {
+ delete t->resampler;
+ t++;
}
- if (mask) {
+ delete [] mState.outputTemp;
+ delete [] mState.resampleTemp;
+}
+
+int AudioMixer::getTrackName()
+{
+ uint32_t names = ~mTrackNames;
+ if (names != 0) {
+ int n = __builtin_ctz(names);
ALOGV("add track (%d)", n);
- mTrackNames |= mask;
+ mTrackNames |= 1 << n;
return TRACK0 + n;
}
return -1;
- }
+}
- void AudioMixer::invalidateState(uint32_t mask)
- {
+void AudioMixer::invalidateState(uint32_t mask)
+{
if (mask) {
mState.needsChanged |= mask;
mState.hook = process__validate;
}
}
- void AudioMixer::deleteTrackName(int name)
- {
+void AudioMixer::deleteTrackName(int name)
+{
name -= TRACK0;
- if (uint32_t(name) < MAX_NUM_TRACKS) {
- ALOGV("deleteTrackName(%d)", name);
- track_t& track(mState.tracks[ name ]);
- if (track.enabled != 0) {
- track.enabled = 0;
- invalidateState(1<<name);
- }
- if (track.resampler) {
- // delete the resampler
- delete track.resampler;
- track.resampler = 0;
- track.sampleRate = mSampleRate;
- invalidateState(1<<name);
- }
- track.volumeInc[0] = 0;
- track.volumeInc[1] = 0;
- mTrackNames &= ~(1<<name);
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ ALOGV("deleteTrackName(%d)", name);
+ track_t& track(mState.tracks[ name ]);
+ if (track.enabled) {
+ track.enabled = false;
+ invalidateState(1<<name);
}
- }
-
-status_t AudioMixer::enable(int name)
-{
- switch (name) {
- case MIXING: {
- if (mState.tracks[ mActiveTrack ].enabled != 1) {
- mState.tracks[ mActiveTrack ].enabled = 1;
- ALOGV("enable(%d)", mActiveTrack);
- invalidateState(1<<mActiveTrack);
- }
- } break;
- default:
- return NAME_NOT_FOUND;
+ if (track.resampler != NULL) {
+ // delete the resampler
+ delete track.resampler;
+ track.resampler = NULL;
+ track.sampleRate = mSampleRate;
+ invalidateState(1<<name);
}
- return NO_ERROR;
+ track.volumeInc[0] = 0;
+ track.volumeInc[1] = 0;
+ mTrackNames &= ~(1<<name);
}
-status_t AudioMixer::disable(int name)
+void AudioMixer::enable(int name)
{
- switch (name) {
- case MIXING: {
- if (mState.tracks[ mActiveTrack ].enabled != 0) {
- mState.tracks[ mActiveTrack ].enabled = 0;
- ALOGV("disable(%d)", mActiveTrack);
- invalidateState(1<<mActiveTrack);
- }
- } break;
- default:
- return NAME_NOT_FOUND;
+ name -= TRACK0;
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ track_t& track = mState.tracks[name];
+
+ if (!track.enabled) {
+ track.enabled = true;
+ ALOGV("enable(%d)", name);
+ invalidateState(1 << name);
}
- return NO_ERROR;
}
-status_t AudioMixer::setActiveTrack(int track)
+void AudioMixer::disable(int name)
{
- if (uint32_t(track-TRACK0) >= MAX_NUM_TRACKS) {
- return BAD_VALUE;
+ name -= TRACK0;
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ track_t& track = mState.tracks[name];
+
+ if (track.enabled) {
+ track.enabled = false;
+ ALOGV("disable(%d)", name);
+ invalidateState(1 << name);
}
- mActiveTrack = track - TRACK0;
- return NO_ERROR;
}
-status_t AudioMixer::setParameter(int target, int name, void *value)
+void AudioMixer::setParameter(int name, int target, int param, void *value)
{
+ name -= TRACK0;
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ track_t& track = mState.tracks[name];
+
int valueInt = (int)value;
int32_t *valueBuf = (int32_t *)value;
switch (target) {
- case TRACK:
- if (name == CHANNEL_MASK) {
- uint32_t mask = (uint32_t)value;
- if (mState.tracks[ mActiveTrack ].channelMask != mask) {
- uint8_t channelCount = popcount(mask);
- if ((channelCount <= MAX_NUM_CHANNELS) && (channelCount)) {
- mState.tracks[ mActiveTrack ].channelMask = mask;
- mState.tracks[ mActiveTrack ].channelCount = channelCount;
- ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
- invalidateState(1<<mActiveTrack);
- return NO_ERROR;
- }
- } else {
- return NO_ERROR;
- }
- }
- if (name == MAIN_BUFFER) {
- if (mState.tracks[ mActiveTrack ].mainBuffer != valueBuf) {
- mState.tracks[ mActiveTrack ].mainBuffer = valueBuf;
- ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
- invalidateState(1<<mActiveTrack);
- }
- return NO_ERROR;
- }
- if (name == AUX_BUFFER) {
- if (mState.tracks[ mActiveTrack ].auxBuffer != valueBuf) {
- mState.tracks[ mActiveTrack ].auxBuffer = valueBuf;
- ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
- invalidateState(1<<mActiveTrack);
- }
- return NO_ERROR;
- }
- break;
- case RESAMPLE:
- if (name == SAMPLE_RATE) {
- if (valueInt > 0) {
- track_t& track = mState.tracks[ mActiveTrack ];
- if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
- ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
- uint32_t(valueInt));
- invalidateState(1<<mActiveTrack);
- }
- return NO_ERROR;
+ case TRACK:
+ switch (param) {
+ case CHANNEL_MASK: {
+ uint32_t mask = (uint32_t)value;
+ if (track.channelMask != mask) {
+ uint8_t channelCount = popcount(mask);
+ assert((channelCount <= MAX_NUM_CHANNELS) && (channelCount));
+ track.channelMask = mask;
+ track.channelCount = channelCount;
+ ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", mask);
+ invalidateState(1 << name);
}
- }
- if (name == RESET) {
- track_t& track = mState.tracks[ mActiveTrack ];
- track.resetResampler();
- invalidateState(1<<mActiveTrack);
- return NO_ERROR;
+ } break;
+ case MAIN_BUFFER:
+ if (track.mainBuffer != valueBuf) {
+ track.mainBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+ invalidateState(1 << name);
+ }
+ break;
+ case AUX_BUFFER:
+ if (track.auxBuffer != valueBuf) {
+ track.auxBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+ invalidateState(1 << name);
+ }
+ break;
+ default:
+ // bad param
+ assert(false);
}
break;
+
+ case RESAMPLE:
+ switch (param) {
+ case SAMPLE_RATE:
+ assert(valueInt > 0);
+ if (track.setResampler(uint32_t(valueInt), mSampleRate)) {
+ ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
+ uint32_t(valueInt));
+ invalidateState(1 << name);
+ }
+ break;
+ case RESET:
+ track.resetResampler();
+ invalidateState(1 << name);
+ break;
+ default:
+ // bad param
+ assert(false);
+ }
+ break;
+
case RAMP_VOLUME:
case VOLUME:
- if ((uint32_t(name-VOLUME0) < MAX_NUM_CHANNELS)) {
- track_t& track = mState.tracks[ mActiveTrack ];
- if (track.volume[name-VOLUME0] != valueInt) {
+ switch (param) {
+ case VOLUME0:
+ case VOLUME1:
+ if (track.volume[param-VOLUME0] != valueInt) {
ALOGV("setParameter(VOLUME, VOLUME0/1: %04x)", valueInt);
- track.prevVolume[name-VOLUME0] = track.volume[name-VOLUME0] << 16;
- track.volume[name-VOLUME0] = valueInt;
+ track.prevVolume[param-VOLUME0] = track.volume[param-VOLUME0] << 16;
+ track.volume[param-VOLUME0] = valueInt;
if (target == VOLUME) {
- track.prevVolume[name-VOLUME0] = valueInt << 16;
- track.volumeInc[name-VOLUME0] = 0;
+ track.prevVolume[param-VOLUME0] = valueInt << 16;
+ track.volumeInc[param-VOLUME0] = 0;
} else {
- int32_t d = (valueInt<<16) - track.prevVolume[name-VOLUME0];
+ int32_t d = (valueInt<<16) - track.prevVolume[param-VOLUME0];
int32_t volInc = d / int32_t(mState.frameCount);
- track.volumeInc[name-VOLUME0] = volInc;
+ track.volumeInc[param-VOLUME0] = volInc;
if (volInc == 0) {
- track.prevVolume[name-VOLUME0] = valueInt << 16;
+ track.prevVolume[param-VOLUME0] = valueInt << 16;
}
}
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
- return NO_ERROR;
- } else if (name == AUXLEVEL) {
- track_t& track = mState.tracks[ mActiveTrack ];
+ break;
+ case AUXLEVEL:
+ //assert(0 <= valueInt && valueInt <= MAX_GAIN_INT);
if (track.auxLevel != valueInt) {
ALOGV("setParameter(VOLUME, AUXLEVEL: %04x)", valueInt);
track.prevAuxLevel = track.auxLevel << 16;
@@ -276,13 +273,19 @@
track.prevAuxLevel = valueInt << 16;
}
}
- invalidateState(1<<mActiveTrack);
+ invalidateState(1 << name);
}
- return NO_ERROR;
+ break;
+ default:
+ // bad param
+ assert(false);
}
break;
+
+ default:
+ // bad target
+ assert(false);
}
- return BAD_VALUE;
}
bool AudioMixer::track_t::setResampler(uint32_t value, uint32_t devSampleRate)
@@ -290,9 +293,10 @@
if (value!=devSampleRate || resampler) {
if (sampleRate != value) {
sampleRate = value;
- if (resampler == 0) {
+ if (resampler == NULL) {
resampler = AudioResampler::create(
format, channelCount, devSampleRate);
+ resampler->setLocalTimeFreq(localTimeFreq);
}
return true;
}
@@ -300,22 +304,10 @@
return false;
}
-bool AudioMixer::track_t::doesResample() const
-{
- return resampler != 0;
-}
-
-void AudioMixer::track_t::resetResampler()
-{
- if (resampler != 0) {
- resampler->reset();
- }
-}
-
inline
void AudioMixer::track_t::adjustVolumeRamp(bool aux)
{
- for (int i=0 ; i<2 ; i++) {
+ for (uint32_t i=0 ; i<MAX_NUM_CHANNELS ; i++) {
if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
volumeInc[i] = 0;
@@ -331,39 +323,31 @@
}
}
-size_t AudioMixer::track_t::getUnreleasedFrames()
-{
- if (resampler != NULL) {
- return resampler->getUnreleasedFrames();
- }
- return 0;
-}
-
-size_t AudioMixer::getUnreleasedFrames(int name)
+size_t AudioMixer::getUnreleasedFrames(int name) const
{
name -= TRACK0;
if (uint32_t(name) < MAX_NUM_TRACKS) {
- track_t& track(mState.tracks[name]);
- return track.getUnreleasedFrames();
+ return mState.tracks[name].getUnreleasedFrames();
}
return 0;
}
-status_t AudioMixer::setBufferProvider(AudioBufferProvider* buffer)
+void AudioMixer::setBufferProvider(int name, AudioBufferProvider* buffer)
{
- mState.tracks[ mActiveTrack ].bufferProvider = buffer;
- return NO_ERROR;
+ name -= TRACK0;
+ assert(uint32_t(name) < MAX_NUM_TRACKS);
+ mState.tracks[name].bufferProvider = buffer;
}
-void AudioMixer::process()
+void AudioMixer::process(int64_t pts)
{
- mState.hook(&mState);
+ mState.hook(&mState, pts);
}
-void AudioMixer::process__validate(state_t* state)
+void AudioMixer::process__validate(state_t* state, int64_t pts)
{
ALOGW_IF(!state->needsChanged,
"in process__validate() but nothing's invalid");
@@ -386,9 +370,9 @@
// compute everything we need...
int countActiveTracks = 0;
- int all16BitsStereoNoResample = 1;
- int resampling = 0;
- int volumeRamp = 0;
+ bool all16BitsStereoNoResample = true;
+ bool resampling = false;
+ bool volumeRamp = false;
uint32_t en = state->enabledTracks;
while (en) {
const int i = 31 - __builtin_clz(en);
@@ -405,7 +389,7 @@
}
if (t.volumeInc[0]|t.volumeInc[1]) {
- volumeRamp = 1;
+ volumeRamp = true;
} else if (!t.doesResample() && t.volumeRL == 0) {
n |= NEEDS_MUTE_ENABLED;
}
@@ -415,16 +399,16 @@
t.hook = track__nop;
} else {
if ((n & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
- all16BitsStereoNoResample = 0;
+ all16BitsStereoNoResample = false;
}
if ((n & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
- all16BitsStereoNoResample = 0;
- resampling = 1;
+ all16BitsStereoNoResample = false;
+ resampling = true;
t.hook = track__genericResample;
} else {
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
t.hook = track__16BitsMono;
- all16BitsStereoNoResample = 0;
+ all16BitsStereoNoResample = false;
}
if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_2){
t.hook = track__16BitsStereo;
@@ -447,11 +431,11 @@
} else {
if (state->outputTemp) {
delete [] state->outputTemp;
- state->outputTemp = 0;
+ state->outputTemp = NULL;
}
if (state->resampleTemp) {
delete [] state->resampleTemp;
- state->resampleTemp = 0;
+ state->resampleTemp = NULL;
}
state->hook = process__genericNoResampling;
if (all16BitsStereoNoResample && !volumeRamp) {
@@ -467,115 +451,33 @@
countActiveTracks, state->enabledTracks,
all16BitsStereoNoResample, resampling, volumeRamp);
- state->hook(state);
+ state->hook(state, pts);
- // Now that the volume ramp has been done, set optimal state and
- // track hooks for subsequent mixer process
- if (countActiveTracks) {
- int allMuted = 1;
- uint32_t en = state->enabledTracks;
- while (en) {
- const int i = 31 - __builtin_clz(en);
- en &= ~(1<<i);
- track_t& t = state->tracks[i];
- if (!t.doesResample() && t.volumeRL == 0)
- {
- t.needs |= NEEDS_MUTE_ENABLED;
- t.hook = track__nop;
- } else {
- allMuted = 0;
- }
- }
- if (allMuted) {
- state->hook = process__nop;
- } else if (all16BitsStereoNoResample) {
- if (countActiveTracks == 1) {
- state->hook = process__OneTrack16BitsStereoNoResampling;
- }
- }
- }
-}
-
-static inline
-int32_t mulAdd(int16_t in, int16_t v, int32_t a)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smlabb %[out], %[in], %[v], %[a] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v), [a]"r"(a)
- : );
- return out;
-#else
- return a + in * int32_t(v);
-#endif
-}
-
-static inline
-int32_t mul(int16_t in, int16_t v)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- asm( "smulbb %[out], %[in], %[v] \n"
- : [out]"=r"(out)
- : [in]"%r"(in), [v]"r"(v)
- : );
- return out;
-#else
- return in * int32_t(v);
-#endif
-}
-
-static inline
-int32_t mulAddRL(int left, uint32_t inRL, uint32_t vRL, int32_t a)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- if (left) {
- asm( "smlabb %[out], %[inRL], %[vRL], %[a] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
- : );
- } else {
- asm( "smlatt %[out], %[inRL], %[vRL], %[a] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL), [a]"r"(a)
- : );
+ // Now that the volume ramp has been done, set optimal state and
+ // track hooks for subsequent mixer process
+ if (countActiveTracks) {
+ bool allMuted = true;
+ uint32_t en = state->enabledTracks;
+ while (en) {
+ const int i = 31 - __builtin_clz(en);
+ en &= ~(1<<i);
+ track_t& t = state->tracks[i];
+ if (!t.doesResample() && t.volumeRL == 0)
+ {
+ t.needs |= NEEDS_MUTE_ENABLED;
+ t.hook = track__nop;
+ } else {
+ allMuted = false;
+ }
+ }
+ if (allMuted) {
+ state->hook = process__nop;
+ } else if (all16BitsStereoNoResample) {
+ if (countActiveTracks == 1) {
+ state->hook = process__OneTrack16BitsStereoNoResampling;
+ }
+ }
}
- return out;
-#else
- if (left) {
- return a + int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
- } else {
- return a + int16_t(inRL>>16) * int16_t(vRL>>16);
- }
-#endif
-}
-
-static inline
-int32_t mulRL(int left, uint32_t inRL, uint32_t vRL)
-{
-#if defined(__arm__) && !defined(__thumb__)
- int32_t out;
- if (left) {
- asm( "smulbb %[out], %[inRL], %[vRL] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL)
- : );
- } else {
- asm( "smultt %[out], %[inRL], %[vRL] \n"
- : [out]"=r"(out)
- : [inRL]"%r"(inRL), [vRL]"r"(vRL)
- : );
- }
- return out;
-#else
- if (left) {
- return int16_t(inRL&0xFFFF) * int16_t(vRL&0xFFFF);
- } else {
- return int16_t(inRL>>16) * int16_t(vRL>>16);
- }
-#endif
}
@@ -591,13 +493,13 @@
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
volumeRampStereo(t, out, outFrameCount, temp, aux);
} else {
volumeStereo(t, out, outFrameCount, temp, aux);
}
} else {
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
t->resampler->setVolume(UNITY_GAIN, UNITY_GAIN);
memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
t->resampler->resample(temp, outFrameCount, t->bufferProvider);
@@ -628,7 +530,7 @@
// (vl + vlInc*frameCount)/65536.0f, frameCount);
// ramp volume
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
int32_t va = t->prevAuxLevel;
const int32_t vaInc = t->auxInc;
int32_t l;
@@ -655,7 +557,7 @@
}
t->prevVolume[0] = vl;
t->prevVolume[1] = vr;
- t->adjustVolumeRamp((aux != NULL));
+ t->adjustVolumeRamp(aux != NULL);
}
void AudioMixer::volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
@@ -663,8 +565,8 @@
const int16_t vl = t->volume[0];
const int16_t vr = t->volume[1];
- if UNLIKELY(aux != NULL) {
- const int16_t va = (int16_t)t->auxLevel;
+ if (CC_UNLIKELY(aux != NULL)) {
+ const int16_t va = t->auxLevel;
do {
int16_t l = (int16_t)(*temp++ >> 12);
int16_t r = (int16_t)(*temp++ >> 12);
@@ -688,13 +590,13 @@
void AudioMixer::track__16BitsStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
- int16_t const *in = static_cast<int16_t const *>(t->in);
+ const int16_t *in = static_cast<const int16_t *>(t->in);
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
int32_t l;
int32_t r;
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
@@ -727,7 +629,7 @@
const uint32_t vrl = t->volumeRL;
const int16_t va = (int16_t)t->auxLevel;
do {
- uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
@@ -739,7 +641,7 @@
}
} else {
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
@@ -765,7 +667,7 @@
else {
const uint32_t vrl = t->volumeRL;
do {
- uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
out[0] = mulAddRL(1, rl, vrl, out[0]);
out[1] = mulAddRL(0, rl, vrl, out[1]);
@@ -778,11 +680,11 @@
void AudioMixer::track__16BitsMono(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
{
- int16_t const *in = static_cast<int16_t const *>(t->in);
+ const int16_t *in = static_cast<int16_t const *>(t->in);
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1]|t->auxInc)) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
int32_t va = t->prevAuxLevel;
@@ -825,7 +727,7 @@
}
} else {
// ramp gain
- if UNLIKELY(t->volumeInc[0]|t->volumeInc[1]) {
+ if (CC_UNLIKELY(t->volumeInc[0]|t->volumeInc[1])) {
int32_t vl = t->prevVolume[0];
int32_t vr = t->prevVolume[1];
const int32_t vlInc = t->volumeInc[0];
@@ -862,21 +764,8 @@
t->in = in;
}
-void AudioMixer::ditherAndClamp(int32_t* out, int32_t const *sums, size_t c)
-{
- for (size_t i=0 ; i<c ; i++) {
- int32_t l = *sums++;
- int32_t r = *sums++;
- int32_t nl = l >> 12;
- int32_t nr = r >> 12;
- l = clamp16(nl);
- r = clamp16(nr);
- *out++ = (r<<16) | (l & 0xFFFF);
- }
-}
-
// no-op case
-void AudioMixer::process__nop(state_t* state)
+void AudioMixer::process__nop(state_t* state, int64_t pts)
{
uint32_t e0 = state->enabledTracks;
size_t bufSize = state->frameCount * sizeof(int16_t) * MAX_NUM_CHANNELS;
@@ -891,7 +780,7 @@
i = 31 - __builtin_clz(e2);
e2 &= ~(1<<i);
track_t& t2 = state->tracks[i];
- if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<i);
}
}
@@ -906,8 +795,10 @@
size_t outFrames = state->frameCount;
while (outFrames) {
t1.buffer.frameCount = outFrames;
- t1.bufferProvider->getNextBuffer(&t1.buffer);
- if (!t1.buffer.raw) break;
+ int64_t outputPTS = calculateOutputPTS(
+ t1, pts, state->frameCount - outFrames);
+ t1.bufferProvider->getNextBuffer(&t1.buffer, outputPTS);
+ if (t1.buffer.raw == NULL) break;
outFrames -= t1.buffer.frameCount;
t1.bufferProvider->releaseBuffer(&t1.buffer);
}
@@ -916,7 +807,7 @@
}
// generic code without resampling
-void AudioMixer::process__genericNoResampling(state_t* state)
+void AudioMixer::process__genericNoResampling(state_t* state, int64_t pts)
{
int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
@@ -928,7 +819,7 @@
e0 &= ~(1<<i);
track_t& t = state->tracks[i];
t.buffer.frameCount = state->frameCount;
- t.bufferProvider->getNextBuffer(&t.buffer);
+ t.bufferProvider->getNextBuffer(&t.buffer, pts);
t.frameCount = t.buffer.frameCount;
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
@@ -949,7 +840,7 @@
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
- if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
@@ -966,23 +857,25 @@
track_t& t = state->tracks[i];
size_t outFrames = BLOCKSIZE;
int32_t *aux = NULL;
- if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+ if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
aux = t.auxBuffer + numFrames;
}
while (outFrames) {
size_t inFrames = (t.frameCount > outFrames)?outFrames:t.frameCount;
if (inFrames) {
- (t.hook)(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
+ t.hook(&t, outTemp + (BLOCKSIZE-outFrames)*MAX_NUM_CHANNELS, inFrames, state->resampleTemp, aux);
t.frameCount -= inFrames;
outFrames -= inFrames;
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
aux += inFrames;
}
}
if (t.frameCount == 0 && outFrames) {
t.bufferProvider->releaseBuffer(&t.buffer);
t.buffer.frameCount = (state->frameCount - numFrames) - (BLOCKSIZE - outFrames);
- t.bufferProvider->getNextBuffer(&t.buffer);
+ int64_t outputPTS = calculateOutputPTS(
+ t, pts, numFrames + (BLOCKSIZE - outFrames));
+ t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
if (t.in == NULL) {
enabledTracks &= ~(1<<i);
@@ -1010,9 +903,10 @@
}
- // generic code with resampling
-void AudioMixer::process__genericResampling(state_t* state)
+// generic code with resampling
+void AudioMixer::process__genericResampling(state_t* state, int64_t pts)
{
+ // this const just means that local variable outTemp doesn't change
int32_t* const outTemp = state->outputTemp;
const size_t size = sizeof(int32_t) * MAX_NUM_CHANNELS * state->frameCount;
@@ -1030,7 +924,7 @@
j = 31 - __builtin_clz(e2);
e2 &= ~(1<<j);
track_t& t2 = state->tracks[j];
- if UNLIKELY(t2.mainBuffer != t1.mainBuffer) {
+ if (CC_UNLIKELY(t2.mainBuffer != t1.mainBuffer)) {
e1 &= ~(1<<j);
}
}
@@ -1042,7 +936,7 @@
e1 &= ~(1<<i);
track_t& t = state->tracks[i];
int32_t *aux = NULL;
- if UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED) {
+ if (CC_UNLIKELY((t.needs & NEEDS_AUX__MASK) == NEEDS_AUX_ENABLED)) {
aux = t.auxBuffer;
}
@@ -1050,23 +944,25 @@
// acquire/release the buffers because it's done by
// the resampler.
if ((t.needs & NEEDS_RESAMPLE__MASK) == NEEDS_RESAMPLE_ENABLED) {
- (t.hook)(&t, outTemp, numFrames, state->resampleTemp, aux);
+ t.resampler->setPTS(pts);
+ t.hook(&t, outTemp, numFrames, state->resampleTemp, aux);
} else {
size_t outFrames = 0;
while (outFrames < numFrames) {
t.buffer.frameCount = numFrames - outFrames;
- t.bufferProvider->getNextBuffer(&t.buffer);
+ int64_t outputPTS = calculateOutputPTS(t, pts, outFrames);
+ t.bufferProvider->getNextBuffer(&t.buffer, outputPTS);
t.in = t.buffer.raw;
// t.in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
if (t.in == NULL) break;
- if UNLIKELY(aux != NULL) {
+ if (CC_UNLIKELY(aux != NULL)) {
aux += outFrames;
}
- (t.hook)(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
+ t.hook(&t, outTemp + outFrames*MAX_NUM_CHANNELS, t.buffer.frameCount, state->resampleTemp, aux);
outFrames += t.buffer.frameCount;
t.bufferProvider->releaseBuffer(&t.buffer);
}
@@ -1077,9 +973,15 @@
}
// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state,
+ int64_t pts)
{
+ // This method is only called when state->enabledTracks has exactly
+ // one bit set. The asserts below would verify this, but are commented out
+ // since the whole point of this method is to optimize performance.
+ //assert(0 != state->enabledTracks);
const int i = 31 - __builtin_clz(state->enabledTracks);
+ //assert((1 << i) == state->enabledTracks);
const track_t& t = state->tracks[i];
AudioBufferProvider::Buffer& b(t.buffer);
@@ -1092,8 +994,9 @@
const uint32_t vrl = t.volumeRL;
while (numFrames) {
b.frameCount = numFrames;
- t.bufferProvider->getNextBuffer(&b);
- int16_t const *in = b.i16;
+ int64_t outputPTS = calculateOutputPTS(t, pts, out - t.mainBuffer);
+ t.bufferProvider->getNextBuffer(&b, outputPTS);
+ const int16_t *in = b.i16;
// in == NULL can happen if the track was flushed just after having
// been enabled for mixing.
@@ -1105,11 +1008,11 @@
}
size_t outFrames = b.frameCount;
- if (UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
+ if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN || uint32_t(vr) > UNITY_GAIN)) {
// volume is boosted, so we might need to clamp even though
// we process only one track.
do {
- uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
@@ -1120,7 +1023,7 @@
} while (--outFrames);
} else {
do {
- uint32_t rl = *reinterpret_cast<uint32_t const *>(in);
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
in += 2;
int32_t l = mulRL(1, rl, vrl) >> 12;
int32_t r = mulRL(0, rl, vrl) >> 12;
@@ -1132,10 +1035,12 @@
}
}
+#if 0
// 2 tracks is also a common case
// NEVER used in current implementation of process__validate()
// only use if the 2 tracks have the same output buffer
-void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state)
+void AudioMixer::process__TwoTracks16BitsStereoNoResampling(state_t* state,
+ int64_t pts)
{
int i;
uint32_t en = state->enabledTracks;
@@ -1149,12 +1054,12 @@
const track_t& t1 = state->tracks[i];
AudioBufferProvider::Buffer& b1(t1.buffer);
- int16_t const *in0;
+ const int16_t *in0;
const int16_t vl0 = t0.volume[0];
const int16_t vr0 = t0.volume[1];
size_t frameCount0 = 0;
- int16_t const *in1;
+ const int16_t *in1;
const int16_t vl1 = t1.volume[0];
const int16_t vr1 = t1.volume[1];
size_t frameCount1 = 0;
@@ -1162,14 +1067,16 @@
//FIXME: only works if two tracks use same buffer
int32_t* out = t0.mainBuffer;
size_t numFrames = state->frameCount;
- int16_t const *buff = NULL;
+ const int16_t *buff = NULL;
while (numFrames) {
if (frameCount0 == 0) {
b0.frameCount = numFrames;
- t0.bufferProvider->getNextBuffer(&b0);
+ int64_t outputPTS = calculateOutputPTS(t0, pts,
+ out - t0.mainBuffer);
+ t0.bufferProvider->getNextBuffer(&b0, outputPTS);
if (b0.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
@@ -1183,14 +1090,16 @@
}
if (frameCount1 == 0) {
b1.frameCount = numFrames;
- t1.bufferProvider->getNextBuffer(&b1);
+ int64_t outputPTS = calculateOutputPTS(t1, pts,
+ out - t0.mainBuffer);
+ t1.bufferProvider->getNextBuffer(&b1, outputPTS);
if (b1.i16 == NULL) {
if (buff == NULL) {
buff = new int16_t[MAX_NUM_CHANNELS * state->frameCount];
}
in1 = buff;
b1.frameCount = numFrames;
- } else {
+ } else {
in1 = b1.i16;
}
frameCount1 = b1.frameCount;
@@ -1225,11 +1134,18 @@
}
}
- if (buff != NULL) {
- delete [] buff;
- }
+ delete [] buff;
+}
+#endif
+
+int64_t AudioMixer::calculateOutputPTS(const track_t& t, int64_t basePTS,
+ int outputFrameIndex)
+{
+ if (AudioBufferProvider::kInvalidPTS == basePTS)
+ return AudioBufferProvider::kInvalidPTS;
+
+ return basePTS + ((outputFrameIndex * t.localTimeFreq) / t.sampleRate);
}
// ----------------------------------------------------------------------------
}; // namespace android
-
diff --git a/services/audioflinger/AudioMixer.h b/services/audioflinger/AudioMixer.h
index 0137185..b210212 100644
--- a/services/audioflinger/AudioMixer.h
+++ b/services/audioflinger/AudioMixer.h
@@ -1,4 +1,4 @@
-/* //device/include/server/AudioFlinger/AudioMixer.h
+/*
**
** Copyright 2007, The Android Open Source Project
**
@@ -28,17 +28,12 @@
// ----------------------------------------------------------------------------
-#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
-#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
-
-// ----------------------------------------------------------------------------
-
class AudioMixer
{
public:
AudioMixer(size_t frameCount, uint32_t sampleRate);
- ~AudioMixer();
+ /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed
static const uint32_t MAX_NUM_TRACKS = 32;
static const uint32_t MAX_NUM_CHANNELS = 2;
@@ -47,11 +42,10 @@
enum { // names
- // track units (32 units)
+ // track names (MAX_NUM_TRACKS units)
TRACK0 = 0x1000,
- // enable/disable
- MIXING = 0x2000,
+ // 0x2000 is unused
// setParameter targets
TRACK = 0x3000,
@@ -65,33 +59,31 @@
FORMAT = 0x4001,
MAIN_BUFFER = 0x4002,
AUX_BUFFER = 0x4003,
- // for TARGET RESAMPLE
+ // for target RESAMPLE
SAMPLE_RATE = 0x4100,
RESET = 0x4101,
- // for TARGET VOLUME (8 channels max)
+ // for target RAMP_VOLUME and VOLUME (8 channels max)
VOLUME0 = 0x4200,
VOLUME1 = 0x4201,
AUXLEVEL = 0x4210,
};
+ // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS
int getTrackName();
void deleteTrackName(int name);
- status_t enable(int name);
- status_t disable(int name);
+ void enable(int name);
+ void disable(int name);
- status_t setActiveTrack(int track);
- status_t setParameter(int target, int name, void *value);
+ void setParameter(int name, int target, int param, void *value);
- status_t setBufferProvider(AudioBufferProvider* bufferProvider);
- void process();
+ void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
+ void process(int64_t pts);
uint32_t trackNames() const { return mTrackNames; }
- static void ditherAndClamp(int32_t* out, int32_t const *sums, size_t c);
-
- size_t getUnreleasedFrames(int name);
+ size_t getUnreleasedFrames(int name) const;
private:
@@ -119,15 +111,9 @@
NEEDS_AUX_ENABLED = 0x00010000,
};
- static inline int32_t applyVolume(int32_t in, int32_t v) {
- return in * v;
- }
-
-
struct state_t;
struct track_t;
- typedef void (*mix_t)(state_t* state);
typedef void (*hook_t)(track_t* t, int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
static const int BLOCKSIZE = 16; // 4 cache lines
@@ -135,41 +121,58 @@
uint32_t needs;
union {
- int16_t volume[2]; // [0]3.12 fixed point
+ int16_t volume[MAX_NUM_CHANNELS]; // [0]3.12 fixed point
int32_t volumeRL;
};
- int32_t prevVolume[2];
+ int32_t prevVolume[MAX_NUM_CHANNELS];
- int32_t volumeInc[2];
- int32_t auxLevel;
+ // 16-byte boundary
+
+ int32_t volumeInc[MAX_NUM_CHANNELS];
int32_t auxInc;
int32_t prevAuxLevel;
+ // 16-byte boundary
+
+ int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
uint16_t frameCount;
- uint8_t channelCount : 4;
- uint8_t enabled : 1;
- uint8_t reserved0 : 3;
- uint8_t format;
- uint32_t channelMask;
+ uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+ uint8_t format; // always 16
+ uint16_t enabled; // actually bool
+ uint32_t channelMask; // currently under-used
AudioBufferProvider* bufferProvider;
- mutable AudioBufferProvider::Buffer buffer;
+
+ // 16-byte boundary
+
+ mutable AudioBufferProvider::Buffer buffer; // 8 bytes
hook_t hook;
- void const* in; // current location in buffer
+ const void* in; // current location in buffer
+
+ // 16-byte boundary
AudioResampler* resampler;
uint32_t sampleRate;
int32_t* mainBuffer;
int32_t* auxBuffer;
+ // 16-byte boundary
+
+ uint64_t localTimeFreq;
+
+ int64_t padding;
+
+ // 16-byte boundary
+
bool setResampler(uint32_t sampleRate, uint32_t devSampleRate);
- bool doesResample() const;
- void resetResampler();
+ bool doesResample() const { return resampler != NULL; }
+ void resetResampler() { if (resampler != NULL) resampler->reset(); }
void adjustVolumeRamp(bool aux);
- size_t getUnreleasedFrames();
+ size_t getUnreleasedFrames() const { return resampler != NULL ?
+ resampler->getUnreleasedFrames() : 0; };
};
// pad to 32-bytes to fill cache line
@@ -177,14 +180,14 @@
uint32_t enabledTracks;
uint32_t needsChanged;
size_t frameCount;
- mix_t hook;
+ void (*hook)(state_t* state, int64_t pts); // one of process__*, never NULL
int32_t *outputTemp;
int32_t *resampleTemp;
int32_t reserved[2];
- track_t tracks[32]; __attribute__((aligned(32)));
+ track_t tracks[MAX_NUM_TRACKS]; __attribute__((aligned(32)));
};
- int mActiveTrack;
+ // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
uint32_t mTrackNames;
const uint32_t mSampleRate;
@@ -199,12 +202,19 @@
static void volumeRampStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
static void volumeStereo(track_t* t, int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
- static void process__validate(state_t* state);
- static void process__nop(state_t* state);
- static void process__genericNoResampling(state_t* state);
- static void process__genericResampling(state_t* state);
- static void process__OneTrack16BitsStereoNoResampling(state_t* state);
- static void process__TwoTracks16BitsStereoNoResampling(state_t* state);
+ static void process__validate(state_t* state, int64_t pts);
+ static void process__nop(state_t* state, int64_t pts);
+ static void process__genericNoResampling(state_t* state, int64_t pts);
+ static void process__genericResampling(state_t* state, int64_t pts);
+ static void process__OneTrack16BitsStereoNoResampling(state_t* state,
+ int64_t pts);
+#if 0
+ static void process__TwoTracks16BitsStereoNoResampling(state_t* state,
+ int64_t pts);
+#endif
+
+ static int64_t calculateOutputPTS(const track_t& t, int64_t basePTS,
+ int outputFrameIndex);
};
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index 6be669b..987b039 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -30,8 +30,8 @@
#include <utils/String16.h>
#include <utils/threads.h>
#include "AudioPolicyService.h"
+#include "ServiceUtilities.h"
#include <cutils/properties.h>
-#include <dlfcn.h>
#include <hardware_legacy/power.h>
#include <media/AudioEffect.h>
#include <media/EffectsFactoryApi.h>
@@ -44,18 +44,11 @@
namespace android {
-static const char *kDeadlockedString = "AudioPolicyService may be deadlocked\n";
-static const char *kCmdDeadlockedString = "AudioPolicyService command thread may be deadlocked\n";
+static const char kDeadlockedString[] = "AudioPolicyService may be deadlocked\n";
+static const char kCmdDeadlockedString[] = "AudioPolicyService command thread may be deadlocked\n";
static const int kDumpLockRetries = 50;
-static const int kDumpLockSleep = 20000;
-
-static bool checkPermission() {
- if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
- bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
- if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
- return ok;
-}
+static const int kDumpLockSleepUs = 20000;
namespace {
extern struct audio_policy_service_ops aps_ops;
@@ -123,19 +116,7 @@
// release audio pre processing resources
for (size_t i = 0; i < mInputSources.size(); i++) {
- InputSourceDesc *source = mInputSources.valueAt(i);
- Vector <EffectDesc *> effects = source->mEffects;
- for (size_t j = 0; j < effects.size(); j++) {
- delete effects[j]->mName;
- Vector <effect_param_t *> params = effects[j]->mParams;
- for (size_t k = 0; k < params.size(); k++) {
- delete params[k];
- }
- params.clear();
- delete effects[j];
- }
- effects.clear();
- delete source;
+ delete mInputSources.valueAt(i);
}
mInputSources.clear();
@@ -145,9 +126,9 @@
}
mInputs.clear();
- if (mpAudioPolicy && mpAudioPolicyDev)
+ if (mpAudioPolicy != NULL && mpAudioPolicyDev != NULL)
mpAudioPolicyDev->destroy_audio_policy(mpAudioPolicyDev, mpAudioPolicy);
- if (mpAudioPolicyDev)
+ if (mpAudioPolicyDev != NULL)
audio_policy_dev_close(mpAudioPolicyDev);
}
@@ -158,7 +139,7 @@
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (!audio_is_output_device(device) && !audio_is_input_device(device)) {
@@ -186,15 +167,15 @@
device_address);
}
-status_t AudioPolicyService::setPhoneState(int state)
+status_t AudioPolicyService::setPhoneState(audio_mode_t state)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (state < 0 || state >= AUDIO_MODE_CNT) {
+ if (uint32_t(state) >= AUDIO_MODE_CNT) {
return BAD_VALUE;
}
@@ -208,26 +189,13 @@
return NO_ERROR;
}
-status_t AudioPolicyService::setRingerMode(uint32_t mode, uint32_t mask)
-{
- if (mpAudioPolicy == NULL) {
- return NO_INIT;
- }
- if (!checkPermission()) {
- return PERMISSION_DENIED;
- }
-
- mpAudioPolicy->set_ringer_mode(mpAudioPolicy, mode, mask);
- return NO_ERROR;
-}
-
status_t AudioPolicyService::setForceUse(audio_policy_force_use_t usage,
audio_policy_forced_cfg_t config)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
if (usage < 0 || usage >= AUDIO_POLICY_FORCE_USE_CNT) {
@@ -255,7 +223,7 @@
audio_io_handle_t AudioPolicyService::getOutput(audio_stream_type_t stream,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_policy_output_flags_t flags)
{
@@ -301,9 +269,9 @@
mpAudioPolicy->release_output(mpAudioPolicy, output);
}
-audio_io_handle_t AudioPolicyService::getInput(int inputSource,
+audio_io_handle_t AudioPolicyService::getInput(audio_source_t inputSource,
uint32_t samplingRate,
- uint32_t format,
+ audio_format_t format,
uint32_t channels,
audio_in_acoustics_t acoustics,
int audioSession)
@@ -311,6 +279,10 @@
if (mpAudioPolicy == NULL) {
return 0;
}
+ // already checked by client, but double-check in case the client wrapper is bypassed
+ if (uint32_t(inputSource) >= AUDIO_SOURCE_CNT) {
+ return 0;
+ }
Mutex::Autolock _l(mLock);
audio_io_handle_t input = mpAudioPolicy->get_input(mpAudioPolicy, inputSource, samplingRate,
format, channels, acoustics);
@@ -319,7 +291,7 @@
return input;
}
// create audio pre processors according to input source
- ssize_t index = mInputSources.indexOfKey((audio_source_t)inputSource);
+ ssize_t index = mInputSources.indexOfKey(inputSource);
if (index < 0) {
return input;
}
@@ -398,40 +370,58 @@
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return BAD_VALUE;
}
mpAudioPolicy->init_stream_volume(mpAudioPolicy, stream, indexMin, indexMax);
return NO_ERROR;
}
-status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream, int index)
+status_t AudioPolicyService::setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (!checkPermission()) {
+ if (!settingsAllowed()) {
return PERMISSION_DENIED;
}
- if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return BAD_VALUE;
}
- return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index);
+ if (mpAudioPolicy->set_stream_volume_index_for_device) {
+ return mpAudioPolicy->set_stream_volume_index_for_device(mpAudioPolicy,
+ stream,
+ index,
+ device);
+ } else {
+ return mpAudioPolicy->set_stream_volume_index(mpAudioPolicy, stream, index);
+ }
}
-status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream, int *index)
+status_t AudioPolicyService::getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device)
{
if (mpAudioPolicy == NULL) {
return NO_INIT;
}
- if (stream < 0 || stream >= AUDIO_STREAM_CNT) {
+ if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
return BAD_VALUE;
}
- return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index);
+ if (mpAudioPolicy->get_stream_volume_index_for_device) {
+ return mpAudioPolicy->get_stream_volume_index_for_device(mpAudioPolicy,
+ stream,
+ index,
+ device);
+ } else {
+ return mpAudioPolicy->get_stream_volume_index(mpAudioPolicy, stream, index);
+ }
}
uint32_t AudioPolicyService::getStrategyForStream(audio_stream_type_t stream)
@@ -487,7 +477,7 @@
return mpAudioPolicy->set_effect_enabled(mpAudioPolicy, id, enabled);
}
-bool AudioPolicyService::isStreamActive(int stream, uint32_t inPastMs) const
+bool AudioPolicyService::isStreamActive(audio_stream_type_t stream, uint32_t inPastMs) const
{
if (mpAudioPolicy == NULL) {
return 0;
@@ -534,7 +524,7 @@
}
void AudioPolicyService::binderDied(const wp<IBinder>& who) {
- ALOGW("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(),
+ ALOGW("binderDied() %p, tid %d, calling pid %d", who.unsafe_get(), gettid(),
IPCThreadState::self()->getCallingPid());
}
@@ -546,7 +536,7 @@
locked = true;
break;
}
- usleep(kDumpLockSleep);
+ usleep(kDumpLockSleepUs);
}
return locked;
}
@@ -570,7 +560,7 @@
status_t AudioPolicyService::dump(int fd, const Vector<String16>& args)
{
- if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+ if (!dumpAllowed()) {
dumpPermissionDenial(fd);
} else {
bool locked = tryLock(mLock);
@@ -580,10 +570,10 @@
}
dumpInternals(fd);
- if (mAudioCommandThread != NULL) {
+ if (mAudioCommandThread != 0) {
mAudioCommandThread->dump(fd);
}
- if (mTonePlaybackThread != NULL) {
+ if (mTonePlaybackThread != 0) {
mTonePlaybackThread->dump(fd);
}
@@ -614,8 +604,7 @@
{
Vector<sp<AudioEffect> > fxVector = inputDesc->mEffects;
for (size_t i = 0; i < fxVector.size(); i++) {
- sp<AudioEffect> fx = fxVector.itemAt(i);
- fx->setEnabled(enabled);
+ fxVector.itemAt(i)->setEnabled(enabled);
}
}
@@ -641,7 +630,7 @@
release_wake_lock(mName.string());
}
mAudioCommands.clear();
- if (mpToneGenerator != NULL) delete mpToneGenerator;
+ delete mpToneGenerator;
}
void AudioPolicyService::AudioCommandThread::onFirstRef()
@@ -674,8 +663,7 @@
ToneData *data = (ToneData *)command->mParam;
ALOGV("AudioCommandThread() processing start tone %d on stream %d",
data->mType, data->mStream);
- if (mpToneGenerator != NULL)
- delete mpToneGenerator;
+ delete mpToneGenerator;
mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
mpToneGenerator->startTone(data->mType);
delete data;
@@ -767,7 +755,7 @@
snprintf(buffer, SIZE, "- Commands:\n");
result = String8(buffer);
result.append(" Command Time Wait pParam\n");
- for (int i = 0; i < (int)mAudioCommands.size(); i++) {
+ for (size_t i = 0; i < mAudioCommands.size(); i++) {
mAudioCommands[i]->dump(buffer, SIZE);
result.append(buffer);
}
@@ -782,7 +770,8 @@
return NO_ERROR;
}
-void AudioPolicyService::AudioCommandThread::startToneCommand(int type, int stream)
+void AudioPolicyService::AudioCommandThread::startToneCommand(ToneGenerator::tone_type type,
+ audio_stream_type_t stream)
{
AudioCommand *command = new AudioCommand();
command->mCommand = START_TONE;
@@ -809,9 +798,9 @@
mWaitWorkCV.signal();
}
-status_t AudioPolicyService::AudioCommandThread::volumeCommand(int stream,
+status_t AudioPolicyService::AudioCommandThread::volumeCommand(audio_stream_type_t stream,
float volume,
- int output,
+ audio_io_handle_t output,
int delayMs)
{
status_t status = NO_ERROR;
@@ -841,7 +830,7 @@
return status;
}
-status_t AudioPolicyService::AudioCommandThread::parametersCommand(int ioHandle,
+status_t AudioPolicyService::AudioCommandThread::parametersCommand(audio_io_handle_t ioHandle,
const char *keyValuePairs,
int delayMs)
{
@@ -900,7 +889,7 @@
// insertCommand_l() must be called with mLock held
void AudioPolicyService::AudioCommandThread::insertCommand_l(AudioCommand *command, int delayMs)
{
- ssize_t i;
+ ssize_t i; // not size_t because i will count down to -1
Vector <AudioCommand *> removedCommands;
command->mTime = systemTime() + milliseconds(delayMs);
@@ -1011,7 +1000,7 @@
const char *keyValuePairs,
int delayMs)
{
- mAudioCommandThread->parametersCommand((int)ioHandle, keyValuePairs,
+ mAudioCommandThread->parametersCommand(ioHandle, keyValuePairs,
delayMs);
}
@@ -1020,8 +1009,8 @@
audio_io_handle_t output,
int delayMs)
{
- return (int)mAudioCommandThread->volumeCommand((int)stream, volume,
- (int)output, delayMs);
+ return (int)mAudioCommandThread->volumeCommand(stream, volume,
+ output, delayMs);
}
int AudioPolicyService::startTone(audio_policy_tone_t tone,
@@ -1052,7 +1041,7 @@
// Audio pre-processing configuration
// ----------------------------------------------------------------------------
-const char *AudioPolicyService::kInputSourceNames[AUDIO_SOURCE_CNT -1] = {
+/*static*/ const char * const AudioPolicyService::kInputSourceNames[AUDIO_SOURCE_CNT -1] = {
MIC_SRC_TAG,
VOICE_UL_SRC_TAG,
VOICE_DL_SRC_TAG,
@@ -1152,7 +1141,7 @@
if (param == NULL && value == NULL) {
// try to parse simple parameter form {int int}
param = root->first_child;
- if (param) {
+ if (param != NULL) {
// Note: that a pair of random strings is read as 0 0
int *ptr = (int *)fx_param->data;
int *ptr2 = (int *)((char *)param + sizeof(effect_param_t));
@@ -1241,7 +1230,7 @@
node = node->next;
continue;
}
- EffectDesc *effect = new EffectDesc(*effects[i]);
+ EffectDesc *effect = new EffectDesc(*effects[i]); // deep copy
loadEffectParameters(node, effect->mParams);
ALOGV("loadInputSource() adding effect %s uuid %08x", effect->mName, effect->mUuid.timeLow);
source->mEffects.add(effect);
@@ -1292,11 +1281,7 @@
ALOGW("loadEffect() invalid uuid %s", node->value);
return NULL;
}
- EffectDesc *effect = new EffectDesc();
- effect->mName = strdup(root->name);
- memcpy(&effect->mUuid, &uuid, sizeof(effect_uuid_t));
-
- return effect;
+ return new EffectDesc(root->name, uuid);
}
status_t AudioPolicyService::loadEffects(cnode *root, Vector <EffectDesc *>& effects)
@@ -1348,13 +1333,13 @@
static audio_io_handle_t aps_open_output(void *service,
uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
uint32_t *pLatencyMs,
audio_policy_output_flags_t flags)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
@@ -1368,7 +1353,7 @@
audio_io_handle_t output2)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
@@ -1378,7 +1363,7 @@
static int aps_close_output(void *service, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
return af->closeOutput(output);
@@ -1387,7 +1372,7 @@
static int aps_suspend_output(void *service, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
@@ -1398,7 +1383,7 @@
static int aps_restore_output(void *service, audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return PERMISSION_DENIED;
}
@@ -1409,12 +1394,12 @@
static audio_io_handle_t aps_open_input(void *service,
uint32_t *pDevices,
uint32_t *pSamplingRate,
- uint32_t *pFormat,
+ audio_format_t *pFormat,
uint32_t *pChannels,
- uint32_t acoustics)
+ audio_in_acoustics_t acoustics)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL) {
+ if (af == 0) {
ALOGW("%s: could not get AudioFlinger", __func__);
return 0;
}
@@ -1426,7 +1411,7 @@
static int aps_close_input(void *service, audio_io_handle_t input)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
return af->closeInput(input);
@@ -1436,7 +1421,7 @@
audio_io_handle_t output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
return af->setStreamOutput(stream, output);
@@ -1447,10 +1432,10 @@
audio_io_handle_t dst_output)
{
sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
- if (af == NULL)
+ if (af == 0)
return PERMISSION_DENIED;
- return af->moveEffects(session, (int)src_output, (int)dst_output);
+ return af->moveEffects(session, src_output, dst_output);
}
static char * aps_get_parameters(void *service, audio_io_handle_t io_handle,
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index d898a53..679fd30 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -19,6 +19,7 @@
#include <cutils/misc.h>
#include <cutils/config_utils.h>
+#include <utils/String8.h>
#include <utils/Vector.h>
#include <utils/SortedVector.h>
#include <binder/BinderService.h>
@@ -31,8 +32,6 @@
namespace android {
-class String8;
-
// ----------------------------------------------------------------------------
class AudioPolicyService :
@@ -59,13 +58,12 @@
virtual audio_policy_dev_state_t getDeviceConnectionState(
audio_devices_t device,
const char *device_address);
- virtual status_t setPhoneState(int state);
- virtual status_t setRingerMode(uint32_t mode, uint32_t mask);
+ virtual status_t setPhoneState(audio_mode_t state);
virtual status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config);
virtual audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage);
virtual audio_io_handle_t getOutput(audio_stream_type_t stream,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
audio_policy_output_flags_t flags =
AUDIO_POLICY_OUTPUT_FLAG_INDIRECT);
@@ -76,12 +74,12 @@
audio_stream_type_t stream,
int session = 0);
virtual void releaseOutput(audio_io_handle_t output);
- virtual audio_io_handle_t getInput(int inputSource,
+ virtual audio_io_handle_t getInput(audio_source_t inputSource,
uint32_t samplingRate = 0,
- uint32_t format = AUDIO_FORMAT_DEFAULT,
+ audio_format_t format = AUDIO_FORMAT_DEFAULT,
uint32_t channels = 0,
audio_in_acoustics_t acoustics =
- (audio_in_acoustics_t)0,
+ (audio_in_acoustics_t)0 /*AUDIO_IN_ACOUSTICS_NONE*/,
int audioSession = 0);
virtual status_t startInput(audio_io_handle_t input);
virtual status_t stopInput(audio_io_handle_t input);
@@ -89,8 +87,12 @@
virtual status_t initStreamVolume(audio_stream_type_t stream,
int indexMin,
int indexMax);
- virtual status_t setStreamVolumeIndex(audio_stream_type_t stream, int index);
- virtual status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index);
+ virtual status_t setStreamVolumeIndex(audio_stream_type_t stream,
+ int index,
+ audio_devices_t device);
+ virtual status_t getStreamVolumeIndex(audio_stream_type_t stream,
+ int *index,
+ audio_devices_t device);
virtual uint32_t getStrategyForStream(audio_stream_type_t stream);
virtual uint32_t getDevicesForStream(audio_stream_type_t stream);
@@ -103,7 +105,7 @@
int id);
virtual status_t unregisterEffect(int id);
virtual status_t setEffectEnabled(int id, bool enabled);
- virtual bool isStreamActive(int stream, uint32_t inPastMs = 0) const;
+ virtual bool isStreamActive(audio_stream_type_t stream, uint32_t inPastMs = 0) const;
virtual status_t queryDefaultPreProcessing(int audioSession,
effect_descriptor_t *descriptors,
@@ -169,10 +171,13 @@
virtual bool threadLoop();
void exit();
- void startToneCommand(int type = 0, int stream = 0);
+ void startToneCommand(ToneGenerator::tone_type type,
+ audio_stream_type_t stream);
void stopToneCommand();
- status_t volumeCommand(int stream, float volume, int output, int delayMs = 0);
- status_t parametersCommand(int ioHandle, const char *keyValuePairs, int delayMs = 0);
+ status_t volumeCommand(audio_stream_type_t stream, float volume,
+ audio_io_handle_t output, int delayMs = 0);
+ status_t parametersCommand(audio_io_handle_t ioHandle,
+ const char *keyValuePairs, int delayMs = 0);
status_t voiceVolumeCommand(float volume, int delayMs = 0);
void insertCommand_l(AudioCommand *command, int delayMs = 0);
@@ -196,20 +201,20 @@
class ToneData {
public:
- int mType; // tone type (START_TONE only)
- int mStream; // stream type (START_TONE only)
+ ToneGenerator::tone_type mType; // tone type (START_TONE only)
+ audio_stream_type_t mStream; // stream type (START_TONE only)
};
class VolumeData {
public:
- int mStream;
+ audio_stream_type_t mStream;
float mVolume;
- int mIO;
+ audio_io_handle_t mIO;
};
class ParametersData {
public:
- int mIO;
+ audio_io_handle_t mIO;
String8 mKeyValuePairs;
};
@@ -228,8 +233,33 @@
class EffectDesc {
public:
- EffectDesc() {}
- virtual ~EffectDesc() {}
+ EffectDesc(const char *name, const effect_uuid_t& uuid) :
+ mName(strdup(name)),
+ mUuid(uuid) { }
+ EffectDesc(const EffectDesc& orig) :
+ mName(strdup(orig.mName)),
+ mUuid(orig.mUuid) {
+ // deep copy mParams
+ for (size_t k = 0; k < orig.mParams.size(); k++) {
+ effect_param_t *origParam = orig.mParams[k];
+ // psize and vsize are rounded up to an int boundary for allocation
+ size_t origSize = sizeof(effect_param_t) +
+ ((origParam->psize + 3) & ~3) +
+ ((origParam->vsize + 3) & ~3);
+ effect_param_t *dupParam = (effect_param_t *) malloc(origSize);
+ memcpy(dupParam, origParam, origSize);
+ // This works because the param buffer allocation is also done by
+ // multiples of 4 bytes originally. In theory we should memcpy only
+ // the actual param size, that is without rounding vsize.
+ mParams.add(dupParam);
+ }
+ }
+ /*virtual*/ ~EffectDesc() {
+ free(mName);
+ for (size_t k = 0; k < mParams.size(); k++) {
+ free(mParams[k]);
+ }
+ }
char *mName;
effect_uuid_t mUuid;
Vector <effect_param_t *> mParams;
@@ -238,7 +268,11 @@
class InputSourceDesc {
public:
InputSourceDesc() {}
- virtual ~InputSourceDesc() {}
+ /*virtual*/ ~InputSourceDesc() {
+ for (size_t j = 0; j < mEffects.size(); j++) {
+ delete mEffects[j];
+ }
+ }
Vector <EffectDesc *> mEffects;
};
@@ -251,7 +285,7 @@
Vector< sp<AudioEffect> >mEffects;
};
- static const char *kInputSourceNames[AUDIO_SOURCE_CNT -1];
+ static const char * const kInputSourceNames[AUDIO_SOURCE_CNT -1];
void setPreProcessorEnabled(InputDesc *inputDesc, bool enabled);
status_t loadPreProcessorConfig(const char *path);
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 4586b54..398ba0b 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -23,8 +23,10 @@
#include <cutils/log.h>
#include <cutils/properties.h>
#include "AudioResampler.h"
+#if 0
#include "AudioResamplerSinc.h"
#include "AudioResamplerCubic.h"
+#endif
#ifdef __arm__
#include <machine/cpu-features.h>
@@ -99,6 +101,7 @@
ALOGV("Create linear Resampler");
resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate);
break;
+#if 0
case MED_QUALITY:
ALOGV("Create cubic Resampler");
resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate);
@@ -107,6 +110,7 @@
ALOGV("Create sinc Resampler");
resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate);
break;
+#endif
}
// initialize resampler
@@ -118,7 +122,8 @@
int32_t sampleRate) :
mBitDepth(bitDepth), mChannelCount(inChannelCount),
mSampleRate(sampleRate), mInSampleRate(sampleRate), mInputIndex(0),
- mPhaseFraction(0) {
+ mPhaseFraction(0), mLocalTimeFreq(0),
+ mPTS(AudioBufferProvider::kInvalidPTS) {
// sanity check on format
if ((bitDepth != 16) ||(inChannelCount < 1) || (inChannelCount > 2)) {
ALOGE("Unsupported sample format, %d bits, %d channels", bitDepth,
@@ -130,12 +135,6 @@
mVolume[0] = mVolume[1] = 0;
mBuffer.frameCount = 0;
- // save format for quick lookup
- if (inChannelCount == 1) {
- mFormat = MONO_16_BIT;
- } else {
- mFormat = STEREO_16_BIT;
- }
}
AudioResampler::~AudioResampler() {
@@ -152,6 +151,23 @@
mVolume[1] = right;
}
+void AudioResampler::setLocalTimeFreq(uint64_t freq) {
+ mLocalTimeFreq = freq;
+}
+
+void AudioResampler::setPTS(int64_t pts) {
+ mPTS = pts;
+}
+
+int64_t AudioResampler::calculateOutputPTS(int outputFrameIndex) {
+
+ if (mPTS == AudioBufferProvider::kInvalidPTS) {
+ return AudioBufferProvider::kInvalidPTS;
+ } else {
+ return mPTS + ((outputFrameIndex * mLocalTimeFreq) / mSampleRate);
+ }
+}
+
void AudioResampler::reset() {
mInputIndex = 0;
mPhaseFraction = 0;
@@ -190,7 +206,7 @@
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+ // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
@@ -198,12 +214,13 @@
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
goto resampleStereo16_exit;
}
- // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
@@ -217,7 +234,7 @@
// handle boundary case
while (inputIndex == 0) {
- // ALOGE("boundary case\n");
+ // ALOGE("boundary case");
out[outputIndex++] += vl * Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vr * Interp(mX0R, in[1], phaseFraction);
Advance(&inputIndex, &phaseFraction, phaseIncrement);
@@ -226,7 +243,7 @@
}
// process input samples
- // ALOGE("general case\n");
+ // ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
@@ -248,7 +265,7 @@
Advance(&inputIndex, &phaseFraction, phaseIncrement);
}
- // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
@@ -265,7 +282,7 @@
}
}
- // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleStereo16_exit:
// save state
@@ -286,19 +303,20 @@
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d\n",
+ // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
// outFrameCount, inputIndex, phaseFraction, phaseIncrement);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
while (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL) {
mInputIndex = inputIndex;
mPhaseFraction = phaseFraction;
goto resampleMono16_exit;
}
- // ALOGE("New buffer fetched: %d frames\n", mBuffer.frameCount);
+ // ALOGE("New buffer fetched: %d frames", mBuffer.frameCount);
if (mBuffer.frameCount > inputIndex) break;
inputIndex -= mBuffer.frameCount;
@@ -310,7 +328,7 @@
// handle boundary case
while (inputIndex == 0) {
- // ALOGE("boundary case\n");
+ // ALOGE("boundary case");
int32_t sample = Interp(mX0L, in[0], phaseFraction);
out[outputIndex++] += vl * sample;
out[outputIndex++] += vr * sample;
@@ -320,7 +338,7 @@
}
// process input samples
- // ALOGE("general case\n");
+ // ALOGE("general case");
#ifdef ASM_ARM_RESAMP1 // asm optimisation for ResamplerOrder1
if (inputIndex + 2 < mBuffer.frameCount) {
@@ -343,7 +361,7 @@
}
- // ALOGE("loop done - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("loop done - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
// if done with buffer, save samples
if (inputIndex >= mBuffer.frameCount) {
@@ -359,7 +377,7 @@
}
}
- // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d\n", outputIndex, inputIndex);
+ // ALOGE("output buffer full - outputIndex=%d, inputIndex=%d", outputIndex, inputIndex);
resampleMono16_exit:
// save state
@@ -390,6 +408,7 @@
* phaseFraction : phase fraction for next interpolation
*
*******************************************************************/
+__attribute__((noinline))
void AudioResamplerOrder1::AsmMono16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
@@ -500,6 +519,7 @@
* phaseFraction : phase fraction for next interpolation
*
*******************************************************************/
+__attribute__((noinline))
void AudioResamplerOrder1::AsmStereo16Loop(int16_t *in, int32_t* maxOutPt, int32_t maxInIdx,
size_t &outputIndex, int32_t* out, size_t &inputIndex, int32_t vl, int32_t vr,
uint32_t &phaseFraction, uint32_t phaseIncrement)
@@ -600,6 +620,5 @@
// ----------------------------------------------------------------------------
-}
-; // namespace android
+} // namespace android
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index ffa690a..9deb796 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -49,12 +49,16 @@
virtual void init() = 0;
virtual void setSampleRate(int32_t inSampleRate);
virtual void setVolume(int16_t left, int16_t right);
+ virtual void setLocalTimeFreq(uint64_t freq);
+
+ // set the PTS of the next buffer output by the resampler
+ virtual void setPTS(int64_t pts);
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider) = 0;
virtual void reset();
- virtual size_t getUnreleasedFrames() { return mInputIndex; }
+ virtual size_t getUnreleasedFrames() const { return mInputIndex; }
protected:
// number of bits for phase fraction - 30 bits allows nearly 2x downsampling
@@ -66,16 +70,17 @@
// multiplier to calculate fixed point phase increment
static const double kPhaseMultiplier = 1L << kNumPhaseBits;
- enum format {MONO_16_BIT, STEREO_16_BIT};
AudioResampler(int bitDepth, int inChannelCount, int32_t sampleRate);
// prevent copying
AudioResampler(const AudioResampler&);
AudioResampler& operator=(const AudioResampler&);
- int32_t mBitDepth;
- int32_t mChannelCount;
- int32_t mSampleRate;
+ int64_t calculateOutputPTS(int outputFrameIndex);
+
+ const int32_t mBitDepth;
+ const int32_t mChannelCount;
+ const int32_t mSampleRate;
int32_t mInSampleRate;
AudioBufferProvider::Buffer mBuffer;
union {
@@ -83,10 +88,11 @@
uint32_t mVolumeRL;
};
int16_t mTargetVolume[2];
- format mFormat;
size_t mInputIndex;
int32_t mPhaseIncrement;
uint32_t mPhaseFraction;
+ uint64_t mLocalTimeFreq;
+ int64_t mPTS;
};
// ----------------------------------------------------------------------------
diff --git a/services/audioflinger/AudioResamplerCubic.cpp b/services/audioflinger/AudioResamplerCubic.cpp
index 47205ba..18e59e9 100644
--- a/services/audioflinger/AudioResamplerCubic.cpp
+++ b/services/audioflinger/AudioResamplerCubic.cpp
@@ -65,7 +65,7 @@
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL)
return;
// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
@@ -95,11 +95,12 @@
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL)
goto save_state; // ugly, but efficient
in = mBuffer.i16;
- // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
// advance sample state
@@ -130,10 +131,10 @@
// fetch first buffer
if (mBuffer.frameCount == 0) {
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer, mPTS);
if (mBuffer.raw == NULL)
return;
- // ALOGW("New buffer: offset=%p, frames=%d\n", mBuffer.raw, mBuffer.frameCount);
+ // ALOGW("New buffer: offset=%p, frames=%d", mBuffer.raw, mBuffer.frameCount);
}
int16_t *in = mBuffer.i16;
@@ -160,7 +161,8 @@
inputIndex = 0;
provider->releaseBuffer(&mBuffer);
mBuffer.frameCount = inFrameCount;
- provider->getNextBuffer(&mBuffer);
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
if (mBuffer.raw == NULL)
goto save_state; // ugly, but efficient
// ALOGW("New buffer: offset=%p, frames=%dn", mBuffer.raw, mBuffer.frameCount);
@@ -181,4 +183,3 @@
// ----------------------------------------------------------------------------
}
; // namespace android
-
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index 9e5e254..d373c08 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -199,33 +199,33 @@
size_t outputSampleCount = outFrameCount * 2;
size_t inFrameCount = (outFrameCount*mInSampleRate)/mSampleRate;
- AudioBufferProvider::Buffer& buffer(mBuffer);
while (outputIndex < outputSampleCount) {
// buffer is empty, fetch a new one
- while (buffer.frameCount == 0) {
- buffer.frameCount = inFrameCount;
- provider->getNextBuffer(&buffer);
- if (buffer.raw == NULL) {
+ while (mBuffer.frameCount == 0) {
+ mBuffer.frameCount = inFrameCount;
+ provider->getNextBuffer(&mBuffer,
+ calculateOutputPTS(outputIndex / 2));
+ if (mBuffer.raw == NULL) {
goto resample_exit;
}
const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
if (phaseIndex == 1) {
// read one frame
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
} else if (phaseIndex == 2) {
// read 2 frames
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
inputIndex++;
if (inputIndex >= mBuffer.frameCount) {
inputIndex -= mBuffer.frameCount;
- provider->releaseBuffer(&buffer);
+ provider->releaseBuffer(&mBuffer);
} else {
- read<CHANNELS>(impulse, phaseFraction, buffer.i16, inputIndex);
+ read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
}
}
}
- int16_t *in = buffer.i16;
- const size_t frameCount = buffer.frameCount;
+ int16_t *in = mBuffer.i16;
+ const size_t frameCount = mBuffer.frameCount;
// Always read-in the first samples from the input buffer
int16_t* head = impulse + halfNumCoefs*CHANNELS;
@@ -264,7 +264,7 @@
// if done with buffer, save samples
if (inputIndex >= frameCount) {
inputIndex -= frameCount;
- provider->releaseBuffer(&buffer);
+ provider->releaseBuffer(&mBuffer);
}
}
@@ -284,7 +284,7 @@
**/
void AudioResamplerSinc::read(
int16_t*& impulse, uint32_t& phaseFraction,
- int16_t const* in, size_t inputIndex)
+ const int16_t* in, size_t inputIndex)
{
const uint32_t phaseIndex = phaseFraction >> kNumPhaseBits;
impulse += CHANNELS;
@@ -302,7 +302,7 @@
template<int CHANNELS>
void AudioResamplerSinc::filterCoefficient(
- int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples)
+ int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples)
{
// compute the index of the coefficient on the positive side and
// negative side
@@ -317,9 +317,9 @@
l = 0;
r = 0;
- int32_t const* coefs = mFirCoefs;
- int16_t const *sP = samples;
- int16_t const *sN = samples+CHANNELS;
+ const int32_t* coefs = mFirCoefs;
+ const int16_t *sP = samples;
+ const int16_t *sN = samples+CHANNELS;
for (unsigned int i=0 ; i<halfNumCoefs/4 ; i++) {
interpolate<CHANNELS>(l, r, coefs+indexP, lerpP, sP);
interpolate<CHANNELS>(l, r, coefs+indexN, lerpN, sN);
@@ -339,13 +339,13 @@
template<int CHANNELS>
void AudioResamplerSinc::interpolate(
int32_t& l, int32_t& r,
- int32_t const* coefs, int16_t lerp, int16_t const* samples)
+ const int32_t* coefs, int16_t lerp, const int16_t* samples)
{
int32_t c0 = coefs[0];
int32_t c1 = coefs[1];
int32_t sinc = mulAdd(lerp, (c1-c0)<<1, c0);
if (CHANNELS == 2) {
- uint32_t rl = *reinterpret_cast<uint32_t const*>(samples);
+ uint32_t rl = *reinterpret_cast<const uint32_t*>(samples);
l = mulAddRL(1, rl, sinc, l);
r = mulAddRL(0, rl, sinc, r);
} else {
@@ -355,4 +355,3 @@
// ----------------------------------------------------------------------------
}; // namespace android
-
diff --git a/services/audioflinger/AudioResamplerSinc.h b/services/audioflinger/AudioResamplerSinc.h
index e6cb90b..f0a07b8 100644
--- a/services/audioflinger/AudioResamplerSinc.h
+++ b/services/audioflinger/AudioResamplerSinc.h
@@ -31,7 +31,7 @@
public:
AudioResamplerSinc(int bitDepth, int inChannelCount, int32_t sampleRate);
- ~AudioResamplerSinc();
+ virtual ~AudioResamplerSinc();
virtual void resample(int32_t* out, size_t outFrameCount,
AudioBufferProvider* provider);
@@ -44,22 +44,22 @@
template<int CHANNELS>
inline void filterCoefficient(
- int32_t& l, int32_t& r, uint32_t phase, int16_t const *samples);
+ int32_t& l, int32_t& r, uint32_t phase, const int16_t *samples);
template<int CHANNELS>
inline void interpolate(
int32_t& l, int32_t& r,
- int32_t const* coefs, int16_t lerp, int16_t const* samples);
+ const int32_t* coefs, int16_t lerp, const int16_t* samples);
template<int CHANNELS>
inline void read(int16_t*& impulse, uint32_t& phaseFraction,
- int16_t const* in, size_t inputIndex);
+ const int16_t* in, size_t inputIndex);
int16_t *mState;
int16_t *mImpulse;
int16_t *mRingFull;
- int32_t const * mFirCoefs;
+ const int32_t * mFirCoefs;
static const int32_t mFirCoefsDown[];
static const int32_t mFirCoefsUp[];
diff --git a/services/audioflinger/ServiceUtilities.cpp b/services/audioflinger/ServiceUtilities.cpp
new file mode 100644
index 0000000..6a58852
--- /dev/null
+++ b/services/audioflinger/ServiceUtilities.cpp
@@ -0,0 +1,55 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <binder/IPCThreadState.h>
+#include <binder/IServiceManager.h>
+#include <binder/PermissionCache.h>
+#include "ServiceUtilities.h"
+
+namespace android {
+
+// This optimization assumes mediaserver process doesn't fork, which it doesn't
+const pid_t getpid_cached = getpid();
+
+bool recordingAllowed() {
+ if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
+ static const String16 sRecordAudio("android.permission.RECORD_AUDIO");
+ // don't use PermissionCache; this is not a system permission
+ bool ok = checkCallingPermission(sRecordAudio);
+ if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO");
+ return ok;
+}
+
+bool settingsAllowed() {
+ if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
+ static const String16 sAudioSettings("android.permission.MODIFY_AUDIO_SETTINGS");
+ // don't use PermissionCache; this is not a system permission
+ bool ok = checkCallingPermission(sAudioSettings);
+ if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
+ return ok;
+}
+
+bool dumpAllowed() {
+ // don't optimize for same pid, since mediaserver never dumps itself
+ static const String16 sDump("android.permission.DUMP");
+ // OK to use PermissionCache; this is a system permission
+ bool ok = PermissionCache::checkCallingPermission(sDump);
+ // convention is for caller to dump an error message to fd instead of logging here
+ //if (!ok) ALOGE("Request requires android.permission.DUMP");
+ return ok;
+}
+
+} // namespace android
diff --git a/include/media/thread_init.h b/services/audioflinger/ServiceUtilities.h
similarity index 73%
rename from include/media/thread_init.h
rename to services/audioflinger/ServiceUtilities.h
index 2feac86..f77ec5b 100644
--- a/include/media/thread_init.h
+++ b/services/audioflinger/ServiceUtilities.h
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2008 The Android Open Source Project
+ * Copyright (C) 2012 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,11 +14,14 @@
* limitations under the License.
*/
-#ifndef THREAD_INIT_H
-#define THREAD_INIT_H
+#include <unistd.h>
-bool InitializeForThread();
-void UninitializeForThread();
+namespace android {
-#endif /* THREAD_INIT_H*/
-
+extern const pid_t getpid_cached;
+
+bool recordingAllowed();
+bool settingsAllowed();
+bool dumpAllowed();
+
+}
diff --git a/services/camera/libcameraservice/CameraHardwareInterface.h b/services/camera/libcameraservice/CameraHardwareInterface.h
index 34087b5..87a0802 100644
--- a/services/camera/libcameraservice/CameraHardwareInterface.h
+++ b/services/camera/libcameraservice/CameraHardwareInterface.h
@@ -21,8 +21,6 @@
#include <binder/MemoryBase.h>
#include <binder/MemoryHeapBase.h>
#include <utils/RefBase.h>
-#include <surfaceflinger/ISurface.h>
-#include <ui/android_native_buffer.h>
#include <ui/GraphicBuffer.h>
#include <camera/Camera.h>
#include <camera/CameraParameters.h>
@@ -635,6 +633,12 @@
return native_window_set_crop(a, &crop);
}
+ static int __set_timestamp(struct preview_stream_ops *w,
+ int64_t timestamp) {
+ ANativeWindow *a = anw(w);
+ return native_window_set_buffers_timestamp(a, timestamp);
+ }
+
static int __set_usage(struct preview_stream_ops* w, int usage)
{
ANativeWindow *a = anw(w);
@@ -664,6 +668,7 @@
mHalPreviewWindow.nw.set_buffer_count = __set_buffer_count;
mHalPreviewWindow.nw.set_buffers_geometry = __set_buffers_geometry;
mHalPreviewWindow.nw.set_crop = __set_crop;
+ mHalPreviewWindow.nw.set_timestamp = __set_timestamp;
mHalPreviewWindow.nw.set_usage = __set_usage;
mHalPreviewWindow.nw.set_swap_interval = __set_swap_interval;
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index 918f31e..adf1d49 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -29,10 +29,10 @@
#include <cutils/atomic.h>
#include <cutils/properties.h>
#include <gui/SurfaceTextureClient.h>
+#include <gui/Surface.h>
#include <hardware/hardware.h>
#include <media/AudioSystem.h>
#include <media/mediaplayer.h>
-#include <surfaceflinger/ISurface.h>
#include <utils/Errors.h>
#include <utils/Log.h>
#include <utils/String16.h>
@@ -351,7 +351,7 @@
// Enable zoom, error, focus, and metadata messages by default
enableMsgType(CAMERA_MSG_ERROR | CAMERA_MSG_ZOOM | CAMERA_MSG_FOCUS |
- CAMERA_MSG_PREVIEW_METADATA);
+ CAMERA_MSG_PREVIEW_METADATA | CAMERA_MSG_FOCUS_MOVE);
// Callback is disabled by default
mPreviewCallbackFlag = CAMERA_FRAME_CALLBACK_FLAG_NOOP;
diff --git a/services/camera/tests/CameraServiceTest/CameraServiceTest.cpp b/services/camera/tests/CameraServiceTest/CameraServiceTest.cpp
index 1055538..e417b79 100644
--- a/services/camera/tests/CameraServiceTest/CameraServiceTest.cpp
+++ b/services/camera/tests/CameraServiceTest/CameraServiceTest.cpp
@@ -22,7 +22,6 @@
#include <sys/types.h>
#include <sys/wait.h>
#include <unistd.h>
-#include <surfaceflinger/ISurface.h>
#include <camera/Camera.h>
#include <camera/CameraParameters.h>
#include <ui/GraphicBuffer.h>