auto import from //depot/cupcake/@135843
diff --git a/libs/audioflinger/AudioFlinger.cpp b/libs/audioflinger/AudioFlinger.cpp
new file mode 100644
index 0000000..92c40e9
--- /dev/null
+++ b/libs/audioflinger/AudioFlinger.cpp
@@ -0,0 +1,2474 @@
+/* //device/include/server/AudioFlinger/AudioFlinger.cpp
+**
+** Copyright 2007, 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 "AudioFlinger"
+//#define LOG_NDEBUG 0
+
+#include <math.h>
+#include <signal.h>
+#include <sys/time.h>
+#include <sys/resource.h>
+
+#include <utils/IServiceManager.h>
+#include <utils/Log.h>
+#include <utils/Parcel.h>
+#include <utils/IPCThreadState.h>
+#include <utils/String16.h>
+#include <utils/threads.h>
+
+#include <cutils/properties.h>
+
+#include <media/AudioTrack.h>
+#include <media/AudioRecord.h>
+
+#include <private/media/AudioTrackShared.h>
+
+#include <hardware_legacy/AudioHardwareInterface.h>
+
+#include "AudioMixer.h"
+#include "AudioFlinger.h"
+
+#ifdef WITH_A2DP
+#include "A2dpAudioInterface.h"
+#endif
+
+// ----------------------------------------------------------------------------
+// the sim build doesn't have gettid
+
+#ifndef HAVE_GETTID
+# define gettid getpid
+#endif
+
+// ----------------------------------------------------------------------------
+
+namespace android {
+
+//static const nsecs_t kStandbyTimeInNsecs = seconds(3);
+static const unsigned long kBufferRecoveryInUsecs = 2000;
+static const unsigned long kMaxBufferRecoveryInUsecs = 20000;
+static const float MAX_GAIN = 4096.0f;
+
+// retry counts for buffer fill timeout
+// 50 * ~20msecs = 1 second
+static const int8_t kMaxTrackRetries = 50;
+static const int8_t kMaxTrackStartupRetries = 50;
+
+static const int kStartSleepTime = 30000;
+static const int kStopSleepTime = 30000;
+
+// Maximum number of pending buffers allocated by OutputTrack::write()
+static const uint8_t kMaxOutputTrackBuffers = 5;
+
+
+#define AUDIOFLINGER_SECURITY_ENABLED 1
+
+// ----------------------------------------------------------------------------
+
+static bool recordingAllowed() {
+#ifndef HAVE_ANDROID_OS
+ return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+ if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+ bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO"));
+ if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO");
+ return ok;
+#else
+ if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO")))
+ LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest");
+ return true;
+#endif
+}
+
+static bool settingsAllowed() {
+#ifndef HAVE_ANDROID_OS
+ return true;
+#endif
+#if AUDIOFLINGER_SECURITY_ENABLED
+ if (getpid() == IPCThreadState::self()->getCallingPid()) return true;
+ bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"));
+ if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS");
+ return ok;
+#else
+ if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")))
+ LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest");
+ return true;
+#endif
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::AudioFlinger()
+ : BnAudioFlinger(),
+ mAudioHardware(0), mA2dpAudioInterface(0),
+ mA2dpEnabled(false), mA2dpEnabledReq(false),
+ mForcedSpeakerCount(0), mForcedRoute(0), mRouteRestoreTime(0), mMusicMuteSaved(false)
+{
+ mHardwareStatus = AUDIO_HW_IDLE;
+ mAudioHardware = AudioHardwareInterface::create();
+ mHardwareStatus = AUDIO_HW_INIT;
+ if (mAudioHardware->initCheck() == NO_ERROR) {
+ // open 16-bit output stream for s/w mixer
+ mHardwareStatus = AUDIO_HW_OUTPUT_OPEN;
+ status_t status;
+ AudioStreamOut *hwOutput = mAudioHardware->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ if (hwOutput) {
+ mHardwareMixerThread = new MixerThread(this, hwOutput, AudioSystem::AUDIO_OUTPUT_HARDWARE);
+ } else {
+ LOGE("Failed to initialize hardware output stream, status: %d", status);
+ }
+
+#ifdef WITH_A2DP
+ // Create A2DP interface
+ mA2dpAudioInterface = new A2dpAudioInterface();
+ AudioStreamOut *a2dpOutput = mA2dpAudioInterface->openOutputStream(AudioSystem::PCM_16_BIT, 0, 0, &status);
+ if (a2dpOutput) {
+ mA2dpMixerThread = new MixerThread(this, a2dpOutput, AudioSystem::AUDIO_OUTPUT_A2DP);
+ if (hwOutput) {
+ uint32_t frameCount = ((a2dpOutput->bufferSize()/a2dpOutput->frameSize()) * hwOutput->sampleRate()) / a2dpOutput->sampleRate();
+ MixerThread::OutputTrack *a2dpOutTrack = new MixerThread::OutputTrack(mA2dpMixerThread,
+ hwOutput->sampleRate(),
+ AudioSystem::PCM_16_BIT,
+ hwOutput->channelCount(),
+ frameCount);
+ mHardwareMixerThread->setOuputTrack(a2dpOutTrack);
+ }
+ } else {
+ LOGE("Failed to initialize A2DP output stream, status: %d", status);
+ }
+#endif
+
+ // FIXME - this should come from settings
+ setRouting(AudioSystem::MODE_NORMAL, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
+ setRouting(AudioSystem::MODE_RINGTONE, AudioSystem::ROUTE_SPEAKER, AudioSystem::ROUTE_ALL);
+ setRouting(AudioSystem::MODE_IN_CALL, AudioSystem::ROUTE_EARPIECE, AudioSystem::ROUTE_ALL);
+ setMode(AudioSystem::MODE_NORMAL);
+
+ setMasterVolume(1.0f);
+ setMasterMute(false);
+
+ // Start record thread
+ mAudioRecordThread = new AudioRecordThread(mAudioHardware);
+ if (mAudioRecordThread != 0) {
+ mAudioRecordThread->run("AudioRecordThread", PRIORITY_URGENT_AUDIO);
+ }
+ } else {
+ LOGE("Couldn't even initialize the stubbed audio hardware!");
+ }
+}
+
+AudioFlinger::~AudioFlinger()
+{
+ if (mAudioRecordThread != 0) {
+ mAudioRecordThread->exit();
+ mAudioRecordThread.clear();
+ }
+ mHardwareMixerThread.clear();
+ delete mAudioHardware;
+ // deleting mA2dpAudioInterface also deletes mA2dpOutput;
+#ifdef WITH_A2DP
+ mA2dpMixerThread.clear();
+ delete mA2dpAudioInterface;
+#endif
+}
+
+
+#ifdef WITH_A2DP
+void AudioFlinger::setA2dpEnabled(bool enable)
+{
+ LOGV_IF(enable, "set output to A2DP\n");
+ LOGV_IF(!enable, "set output to hardware audio\n");
+
+ mA2dpEnabledReq = enable;
+ mA2dpMixerThread->wakeUp();
+}
+#endif // WITH_A2DP
+
+bool AudioFlinger::streamForcedToSpeaker(int streamType)
+{
+ // NOTE that streams listed here must not be routed to A2DP by default:
+ // AudioSystem::routedToA2dpOutput(streamType) == false
+ return (streamType == AudioSystem::RING ||
+ streamType == AudioSystem::ALARM ||
+ streamType == AudioSystem::NOTIFICATION);
+}
+
+status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ 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);
+ }
+ }
+ }
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+
+status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "Hardware status: %d\n", mHardwareStatus);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+ snprintf(buffer, SIZE, "Permission Denial: "
+ "can't dump AudioFlinger from pid=%d, uid=%d\n",
+ IPCThreadState::self()->getCallingPid(),
+ IPCThreadState::self()->getCallingUid());
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::dump(int fd, const Vector<String16>& args)
+{
+ if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+ dumpPermissionDenial(fd, args);
+ } else {
+ AutoMutex lock(&mLock);
+
+ dumpClients(fd, args);
+ dumpInternals(fd, args);
+ mHardwareMixerThread->dump(fd, args);
+#ifdef WITH_A2DP
+ mA2dpMixerThread->dump(fd, args);
+#endif
+
+ // dump record client
+ if (mAudioRecordThread != 0) mAudioRecordThread->dump(fd, args);
+
+ if (mAudioHardware) {
+ mAudioHardware->dumpState(fd, args);
+ }
+ }
+ return NO_ERROR;
+}
+
+// IAudioFlinger interface
+
+
+sp<IAudioTrack> AudioFlinger::createTrack(
+ pid_t pid,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags,
+ const sp<IMemory>& sharedBuffer,
+ status_t *status)
+{
+ sp<MixerThread::Track> track;
+ sp<TrackHandle> trackHandle;
+ sp<Client> client;
+ wp<Client> wclient;
+ status_t lStatus;
+
+ if (streamType >= AudioSystem::NUM_STREAM_TYPES) {
+ LOGE("invalid stream type");
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
+ {
+ Mutex::Autolock _l(mLock);
+
+ wclient = mClients.valueFor(pid);
+
+ if (wclient != NULL) {
+ client = wclient.promote();
+ } else {
+ client = new Client(this, pid);
+ mClients.add(pid, client);
+ }
+#ifdef WITH_A2DP
+ if (isA2dpEnabled() && AudioSystem::routedToA2dpOutput(streamType)) {
+ track = mA2dpMixerThread->createTrack(client, streamType, sampleRate, format,
+ channelCount, frameCount, sharedBuffer, &lStatus);
+ } else
+#endif
+ {
+ track = mHardwareMixerThread->createTrack(client, streamType, sampleRate, format,
+ channelCount, frameCount, sharedBuffer, &lStatus);
+ }
+ if (track != NULL) {
+ trackHandle = new TrackHandle(track);
+ lStatus = NO_ERROR;
+ }
+ }
+
+Exit:
+ if(status) {
+ *status = lStatus;
+ }
+ return trackHandle;
+}
+
+uint32_t AudioFlinger::sampleRate(int output) const
+{
+#ifdef WITH_A2DP
+ if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+ return mA2dpMixerThread->sampleRate();
+ }
+#endif
+ return mHardwareMixerThread->sampleRate();
+}
+
+int AudioFlinger::channelCount(int output) const
+{
+#ifdef WITH_A2DP
+ if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+ return mA2dpMixerThread->channelCount();
+ }
+#endif
+ return mHardwareMixerThread->channelCount();
+}
+
+int AudioFlinger::format(int output) const
+{
+#ifdef WITH_A2DP
+ if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+ return mA2dpMixerThread->format();
+ }
+#endif
+ return mHardwareMixerThread->format();
+}
+
+size_t AudioFlinger::frameCount(int output) const
+{
+#ifdef WITH_A2DP
+ if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+ return mA2dpMixerThread->frameCount();
+ }
+#endif
+ return mHardwareMixerThread->frameCount();
+}
+
+uint32_t AudioFlinger::latency(int output) const
+{
+#ifdef WITH_A2DP
+ if (output == AudioSystem::AUDIO_OUTPUT_A2DP) {
+ return mA2dpMixerThread->latency();
+ }
+#endif
+ return mHardwareMixerThread->latency();
+}
+
+status_t AudioFlinger::setMasterVolume(float value)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ // when hw supports master volume, don't scale in sw mixer
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ if (mAudioHardware->setMasterVolume(value) == NO_ERROR) {
+ value = 1.0f;
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
+ mHardwareMixerThread->setMasterVolume(value);
+#ifdef WITH_A2DP
+ mA2dpMixerThread->setMasterVolume(value);
+#endif
+
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::setRouting(int mode, uint32_t routes, uint32_t mask)
+{
+ status_t err = NO_ERROR;
+
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if ((mode < AudioSystem::MODE_CURRENT) || (mode >= AudioSystem::NUM_MODES)) {
+ LOGW("Illegal value: setRouting(%d, %u, %u)", mode, routes, mask);
+ return BAD_VALUE;
+ }
+
+#ifdef WITH_A2DP
+ LOGD("setRouting %d %d %d, tid %d, calling tid %d\n", mode, routes, mask, gettid(), IPCThreadState::self()->getCallingPid());
+ if (mode == AudioSystem::MODE_NORMAL &&
+ (mask & AudioSystem::ROUTE_BLUETOOTH_A2DP)) {
+ AutoMutex lock(&mLock);
+
+ bool enableA2dp = false;
+ if (routes & AudioSystem::ROUTE_BLUETOOTH_A2DP) {
+ enableA2dp = true;
+ }
+ setA2dpEnabled(enableA2dp);
+ LOGV("setOutput done\n");
+ }
+#endif
+
+ // do nothing if only A2DP routing is affected
+ mask &= ~AudioSystem::ROUTE_BLUETOOTH_A2DP;
+ if (mask) {
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_GET_ROUTING;
+ uint32_t r;
+ err = mAudioHardware->getRouting(mode, &r);
+ if (err == NO_ERROR) {
+ r = (r & ~mask) | (routes & mask);
+ if (mode == AudioSystem::MODE_NORMAL ||
+ (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
+ mSavedRoute = r;
+ r |= mForcedRoute;
+ LOGV("setRouting mSavedRoute %08x mForcedRoute %08x\n", mSavedRoute, mForcedRoute);
+ }
+ mHardwareStatus = AUDIO_HW_SET_ROUTING;
+ err = mAudioHardware->setRouting(mode, r);
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
+ }
+ return err;
+}
+
+uint32_t AudioFlinger::getRouting(int mode) const
+{
+ uint32_t routes = 0;
+ if ((mode >= AudioSystem::MODE_CURRENT) && (mode < AudioSystem::NUM_MODES)) {
+ if (mode == AudioSystem::MODE_NORMAL ||
+ (mode == AudioSystem::MODE_CURRENT && getMode() == AudioSystem::MODE_NORMAL)) {
+ routes = mSavedRoute;
+ } else {
+ mHardwareStatus = AUDIO_HW_GET_ROUTING;
+ mAudioHardware->getRouting(mode, &routes);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ }
+ } else {
+ LOGW("Illegal value: getRouting(%d)", mode);
+ }
+ return routes;
+}
+
+status_t AudioFlinger::setMode(int mode)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ if ((mode < 0) || (mode >= AudioSystem::NUM_MODES)) {
+ LOGW("Illegal value: setMode(%d)", mode);
+ return BAD_VALUE;
+ }
+
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_MODE;
+ status_t ret = mAudioHardware->setMode(mode);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return ret;
+}
+
+int AudioFlinger::getMode() const
+{
+ int mode = AudioSystem::MODE_INVALID;
+ mHardwareStatus = AUDIO_HW_SET_MODE;
+ mAudioHardware->getMode(&mode);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return mode;
+}
+
+status_t AudioFlinger::setMicMute(bool state)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_HW_SET_MIC_MUTE;
+ status_t ret = mAudioHardware->setMicMute(state);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return ret;
+}
+
+bool AudioFlinger::getMicMute() const
+{
+ bool state = AudioSystem::MODE_INVALID;
+ mHardwareStatus = AUDIO_HW_GET_MIC_MUTE;
+ mAudioHardware->getMicMute(&state);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return state;
+}
+
+status_t AudioFlinger::setMasterMute(bool muted)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+ mHardwareMixerThread->setMasterMute(muted);
+#ifdef WITH_A2DP
+ mA2dpMixerThread->setMasterMute(muted);
+#endif
+ return NO_ERROR;
+}
+
+float AudioFlinger::masterVolume() const
+{
+ return mHardwareMixerThread->masterVolume();
+}
+
+bool AudioFlinger::masterMute() const
+{
+ return mHardwareMixerThread->masterMute();
+}
+
+status_t AudioFlinger::setStreamVolume(int stream, float value)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ return BAD_VALUE;
+ }
+
+ mHardwareMixerThread->setStreamVolume(stream, value);
+#ifdef WITH_A2DP
+ mA2dpMixerThread->setStreamVolume(stream, value);
+#endif
+
+ status_t ret = NO_ERROR;
+ if (stream == AudioSystem::VOICE_CALL ||
+ stream == AudioSystem::BLUETOOTH_SCO) {
+
+ if (stream == AudioSystem::VOICE_CALL) {
+ value = (float)AudioSystem::logToLinear(value)/100.0f;
+ } else { // (type == AudioSystem::BLUETOOTH_SCO)
+ value = 1.0f;
+ }
+
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_SET_VOICE_VOLUME;
+ ret = mAudioHardware->setVoiceVolume(value);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ }
+
+ return ret;
+}
+
+status_t AudioFlinger::setStreamMute(int stream, bool muted)
+{
+ // check calling permissions
+ if (!settingsAllowed()) {
+ return PERMISSION_DENIED;
+ }
+
+ if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ return BAD_VALUE;
+ }
+
+#ifdef WITH_A2DP
+ mA2dpMixerThread->setStreamMute(stream, muted);
+#endif
+ if (stream == AudioSystem::MUSIC)
+ {
+ AutoMutex lock(&mHardwareLock);
+ if (mForcedRoute != 0)
+ mMusicMuteSaved = muted;
+ else
+ mHardwareMixerThread->setStreamMute(stream, muted);
+ } else {
+ mHardwareMixerThread->setStreamMute(stream, muted);
+ }
+
+
+
+ return NO_ERROR;
+}
+
+float AudioFlinger::streamVolume(int stream) const
+{
+ if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ return 0.0f;
+ }
+ return mHardwareMixerThread->streamVolume(stream);
+}
+
+bool AudioFlinger::streamMute(int stream) const
+{
+ if (uint32_t(stream) >= AudioSystem::NUM_STREAM_TYPES) {
+ return true;
+ }
+
+ if (stream == AudioSystem::MUSIC && mForcedRoute != 0)
+ {
+ return mMusicMuteSaved;
+ }
+ return mHardwareMixerThread->streamMute(stream);
+}
+
+bool AudioFlinger::isMusicActive() const
+{
+ #ifdef WITH_A2DP
+ if (isA2dpEnabled()) {
+ return mA2dpMixerThread->isMusicActive();
+ }
+ #endif
+ return mHardwareMixerThread->isMusicActive();
+}
+
+status_t AudioFlinger::setParameter(const char* key, const char* value)
+{
+ status_t result, result2;
+ AutoMutex lock(mHardwareLock);
+ mHardwareStatus = AUDIO_SET_PARAMETER;
+
+ LOGV("setParameter() key %s, value %s, tid %d, calling tid %d", key, value, gettid(), IPCThreadState::self()->getCallingPid());
+ result = mAudioHardware->setParameter(key, value);
+ if (mA2dpAudioInterface) {
+ result2 = mA2dpAudioInterface->setParameter(key, value);
+ if (result2)
+ result = result2;
+ }
+ mHardwareStatus = AUDIO_HW_IDLE;
+ return result;
+}
+
+size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount)
+{
+ return mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
+}
+
+void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client)
+{
+
+ LOGV("registerClient() %p, tid %d, calling tid %d", client.get(), gettid(), IPCThreadState::self()->getCallingPid());
+ Mutex::Autolock _l(mLock);
+
+ sp<IBinder> binder = client->asBinder();
+ if (mNotificationClients.indexOf(binder) < 0) {
+ LOGV("Adding notification client %p", binder.get());
+ binder->linkToDeath(this);
+ mNotificationClients.add(binder);
+ client->a2dpEnabledChanged(isA2dpEnabled());
+ }
+}
+
+void AudioFlinger::binderDied(const wp<IBinder>& who) {
+
+ LOGV("binderDied() %p, tid %d, calling tid %d", who.unsafe_get(), gettid(), IPCThreadState::self()->getCallingPid());
+ Mutex::Autolock _l(mLock);
+
+ IBinder *binder = who.unsafe_get();
+
+ if (binder != NULL) {
+ int index = mNotificationClients.indexOf(binder);
+ if (index >= 0) {
+ LOGV("Removing notification client %p", binder);
+ mNotificationClients.removeAt(index);
+ }
+ }
+}
+
+void AudioFlinger::handleOutputSwitch()
+{
+ if (mA2dpEnabled != mA2dpEnabledReq)
+ {
+ Mutex::Autolock _l(mLock);
+
+ if (mA2dpEnabled != mA2dpEnabledReq)
+ {
+ mA2dpEnabled = mA2dpEnabledReq;
+ SortedVector < sp<MixerThread::Track> > tracks;
+ SortedVector < wp<MixerThread::Track> > activeTracks;
+
+ // We hold mA2dpMixerThread mLock already
+ Mutex::Autolock _l(mHardwareMixerThread->mLock);
+
+ // Transfer tracks playing on MUSIC stream from one mixer to the other
+ if (mA2dpEnabled) {
+ mHardwareMixerThread->getTracks(tracks, activeTracks);
+ mA2dpMixerThread->putTracks(tracks, activeTracks);
+ } else {
+ mA2dpMixerThread->getTracks(tracks, activeTracks);
+ mHardwareMixerThread->putTracks(tracks, activeTracks);
+ }
+
+ // Notify AudioSystem of the A2DP activation/deactivation
+ size_t size = mNotificationClients.size();
+ for (size_t i = 0; i < size; i++) {
+ sp<IBinder> binder = mNotificationClients.itemAt(i).promote();
+ if (binder != NULL) {
+ LOGV("Notifying output change to client %p", binder.get());
+ sp<IAudioFlingerClient> client = interface_cast<IAudioFlingerClient> (binder);
+ client->a2dpEnabledChanged(mA2dpEnabled);
+ }
+ }
+
+ mHardwareMixerThread->wakeUp();
+ }
+ }
+}
+
+void AudioFlinger::removeClient(pid_t pid)
+{
+ LOGV("removeClient() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
+ Mutex::Autolock _l(mLock);
+ mClients.removeItem(pid);
+}
+
+void AudioFlinger::wakeUp()
+{
+ mHardwareMixerThread->wakeUp();
+#ifdef WITH_A2DP
+ mA2dpMixerThread->wakeUp();
+#endif // WITH_A2DP
+}
+
+bool AudioFlinger::isA2dpEnabled() const
+{
+ return mA2dpEnabledReq;
+}
+
+void AudioFlinger::handleForcedSpeakerRoute(int command)
+{
+ switch(command) {
+ case ACTIVE_TRACK_ADDED:
+ {
+ AutoMutex lock(mHardwareLock);
+ if (mForcedSpeakerCount++ == 0) {
+ mRouteRestoreTime = 0;
+ mMusicMuteSaved = mHardwareMixerThread->streamMute(AudioSystem::MUSIC);
+ if (mForcedRoute == 0 && !(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
+ LOGV("Route forced to Speaker ON %08x", mSavedRoute | AudioSystem::ROUTE_SPEAKER);
+ mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, true);
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ mAudioHardware->setMasterVolume(0);
+ usleep(mHardwareMixerThread->latency()*1000);
+ mHardwareStatus = AUDIO_HW_SET_ROUTING;
+ mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute | AudioSystem::ROUTE_SPEAKER);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ // delay track start so that audio hardware has time to siwtch routes
+ usleep(kStartSleepTime);
+ mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME;
+ mAudioHardware->setMasterVolume(mHardwareMixerThread->masterVolume());
+ mHardwareStatus = AUDIO_HW_IDLE;
+ }
+ mForcedRoute = AudioSystem::ROUTE_SPEAKER;
+ }
+ LOGV("mForcedSpeakerCount incremented to %d", mForcedSpeakerCount);
+ }
+ break;
+ case ACTIVE_TRACK_REMOVED:
+ {
+ AutoMutex lock(mHardwareLock);
+ if (mForcedSpeakerCount > 0){
+ if (--mForcedSpeakerCount == 0) {
+ mRouteRestoreTime = systemTime() + milliseconds(kStopSleepTime/1000);
+ }
+ LOGV("mForcedSpeakerCount decremented to %d", mForcedSpeakerCount);
+ } else {
+ LOGE("mForcedSpeakerCount is already zero");
+ }
+ }
+ break;
+ case CHECK_ROUTE_RESTORE_TIME:
+ case FORCE_ROUTE_RESTORE:
+ if (mRouteRestoreTime) {
+ AutoMutex lock(mHardwareLock);
+ if (mRouteRestoreTime &&
+ (systemTime() > mRouteRestoreTime || command == FORCE_ROUTE_RESTORE)) {
+ mHardwareMixerThread->setStreamMute(AudioSystem::MUSIC, mMusicMuteSaved);
+ mForcedRoute = 0;
+ if (!(mSavedRoute & AudioSystem::ROUTE_SPEAKER)) {
+ mHardwareStatus = AUDIO_HW_SET_ROUTING;
+ mAudioHardware->setRouting(AudioSystem::MODE_NORMAL, mSavedRoute);
+ mHardwareStatus = AUDIO_HW_IDLE;
+ LOGV("Route forced to Speaker OFF %08x", mSavedRoute);
+ }
+ mRouteRestoreTime = 0;
+ }
+ }
+ break;
+ }
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int outputType)
+ : Thread(false),
+ mAudioFlinger(audioFlinger), mAudioMixer(0), mOutput(output), mOutputType(outputType),
+ mSampleRate(0), mFrameCount(0), mChannelCount(0), mFormat(0), mMixBuffer(0),
+ mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mStandby(false),
+ mInWrite(false)
+{
+ mSampleRate = output->sampleRate();
+ mChannelCount = output->channelCount();
+
+ // FIXME - Current mixer implementation only supports stereo output
+ if (mChannelCount == 1) {
+ LOGE("Invalid audio hardware channel count");
+ }
+
+ mFormat = output->format();
+ mFrameCount = output->bufferSize() / output->channelCount() / sizeof(int16_t);
+ mAudioMixer = new AudioMixer(mFrameCount, output->sampleRate());
+
+ // FIXME - Current mixer implementation only supports stereo output: Always
+ // Allocate a stereo buffer even if HW output is mono.
+ mMixBuffer = new int16_t[mFrameCount * 2];
+ memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t));
+}
+
+AudioFlinger::MixerThread::~MixerThread()
+{
+ delete [] mMixBuffer;
+ delete mAudioMixer;
+}
+
+status_t AudioFlinger::MixerThread::dump(int fd, const Vector<String16>& args)
+{
+ dumpInternals(fd, args);
+ dumpTracks(fd, args);
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::MixerThread::dumpTracks(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "Output %d mixer thread tracks\n", mOutputType);
+ result.append(buffer);
+ result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+ for (size_t i = 0; i < mTracks.size(); ++i) {
+ wp<Track> wTrack = mTracks[i];
+ if (wTrack != 0) {
+ sp<Track> track = wTrack.promote();
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
+ }
+ }
+ }
+
+ snprintf(buffer, SIZE, "Output %d mixer thread active tracks\n", mOutputType);
+ result.append(buffer);
+ result.append(" Name Clien Typ Fmt Chn Buf S M F SRate LeftV RighV Serv User\n");
+ for (size_t i = 0; i < mActiveTracks.size(); ++i) {
+ wp<Track> wTrack = mTracks[i];
+ if (wTrack != 0) {
+ sp<Track> track = wTrack.promote();
+ if (track != 0) {
+ track->dump(buffer, SIZE);
+ result.append(buffer);
+ }
+ }
+ }
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+
+ snprintf(buffer, SIZE, "Output %d mixer thread internals\n", mOutputType);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames());
+ result.append(buffer);
+ snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+ result.append(buffer);
+ snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite);
+ result.append(buffer);
+ snprintf(buffer, SIZE, "standby: %d\n", mStandby);
+ result.append(buffer);
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+// Thread virtuals
+bool AudioFlinger::MixerThread::threadLoop()
+{
+ unsigned long sleepTime = kBufferRecoveryInUsecs;
+ int16_t* curBuf = mMixBuffer;
+ Vector< sp<Track> > tracksToRemove;
+ size_t enabledTracks = 0;
+ nsecs_t standbyTime = systemTime();
+ size_t mixBufferSize = mFrameCount*mChannelCount*sizeof(int16_t);
+ nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 2;
+
+#ifdef WITH_A2DP
+ bool outputTrackActive = false;
+#endif
+
+ do {
+ enabledTracks = 0;
+ { // scope for the mLock
+
+ Mutex::Autolock _l(mLock);
+
+#ifdef WITH_A2DP
+ if (mOutputType == AudioSystem::AUDIO_OUTPUT_A2DP) {
+ mAudioFlinger->handleOutputSwitch();
+ }
+ if (mOutputTrack != NULL && !mAudioFlinger->isA2dpEnabled()) {
+ if (outputTrackActive) {
+ mOutputTrack->stop();
+ outputTrackActive = false;
+ }
+ }
+#endif
+
+ const SortedVector< wp<Track> >& activeTracks = mActiveTracks;
+
+ // put audio hardware into standby after short delay
+ if UNLIKELY(!activeTracks.size() && systemTime() > standbyTime) {
+ // wait until we have something to do...
+ LOGV("Audio hardware entering standby, output %d\n", mOutputType);
+// mAudioFlinger->mHardwareStatus = AUDIO_HW_STANDBY;
+ if (!mStandby) {
+ mOutput->standby();
+ mStandby = true;
+ }
+
+#ifdef WITH_A2DP
+ if (outputTrackActive) {
+ mOutputTrack->stop();
+ outputTrackActive = false;
+ }
+#endif
+ if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
+ mAudioFlinger->handleForcedSpeakerRoute(FORCE_ROUTE_RESTORE);
+ }
+// mHardwareStatus = AUDIO_HW_IDLE;
+ // we're about to wait, flush the binder command buffer
+ IPCThreadState::self()->flushCommands();
+ mWaitWorkCV.wait(mLock);
+ LOGV("Audio hardware exiting standby, output %d\n", mOutputType);
+
+ if (mMasterMute == false) {
+ char value[PROPERTY_VALUE_MAX];
+ property_get("ro.audio.silent", value, "0");
+ if (atoi(value)) {
+ LOGD("Silence is golden");
+ setMasterMute(true);
+ }
+ }
+
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ continue;
+ }
+
+ // Forced route to speaker is handled by hardware mixer thread
+ if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
+ mAudioFlinger->handleForcedSpeakerRoute(CHECK_ROUTE_RESTORE_TIME);
+ }
+
+ // find out which tracks need to be processed
+ size_t count = activeTracks.size();
+ for (size_t i=0 ; i<count ; i++) {
+ sp<Track> t = activeTracks[i].promote();
+ if (t == 0) continue;
+
+ 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());
+ if (cblk->framesReady() && (track->isReady() || track->isStopped()) &&
+ !track->isPaused())
+ {
+ //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server);
+
+ // compute volume for this track
+ int16_t left, right;
+ if (track->isMuted() || mMasterMute || track->isPausing()) {
+ left = right = 0;
+ if (track->isPausing()) {
+ LOGV("paused(%d)", track->name());
+ track->setPaused();
+ }
+ } else {
+ float typeVolume = mStreamTypes[track->type()].volume;
+ float v = mMasterVolume * typeVolume;
+ float v_clamped = v * cblk->volume[0];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ left = int16_t(v_clamped);
+ v_clamped = v * cblk->volume[1];
+ if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN;
+ right = int16_t(v_clamped);
+ }
+
+ // XXX: these things DON'T need to be done each time
+ mAudioMixer->setBufferProvider(track);
+ mAudioMixer->enable(AudioMixer::MIXING);
+
+ int param;
+ if ( track->mFillingUpStatus == Track::FS_FILLED) {
+ // no ramp for the first volume setting
+ track->mFillingUpStatus = Track::FS_ACTIVE;
+ if (track->mState == TrackBase::RESUMING) {
+ track->mState = TrackBase::ACTIVE;
+ param = AudioMixer::RAMP_VOLUME;
+ } else {
+ param = AudioMixer::VOLUME;
+ }
+ } else {
+ param = AudioMixer::RAMP_VOLUME;
+ }
+ mAudioMixer->setParameter(param, AudioMixer::VOLUME0, left);
+ mAudioMixer->setParameter(param, AudioMixer::VOLUME1, right);
+ mAudioMixer->setParameter(
+ AudioMixer::TRACK,
+ AudioMixer::FORMAT, track->format());
+ mAudioMixer->setParameter(
+ AudioMixer::TRACK,
+ AudioMixer::CHANNEL_COUNT, track->channelCount());
+ mAudioMixer->setParameter(
+ AudioMixer::RESAMPLE,
+ AudioMixer::SAMPLE_RATE,
+ int(cblk->sampleRate));
+
+ // reset retry count
+ track->mRetryCount = kMaxTrackRetries;
+ enabledTracks++;
+ } else {
+ //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server);
+ if (track->isStopped()) {
+ track->reset();
+ }
+ if (track->isTerminated() || track->isStopped() || track->isPaused()) {
+ // We have consumed all the buffers of this track.
+ // Remove it from the list of active tracks.
+ LOGV("remove(%d) from active list", track->name());
+ tracksToRemove.add(track);
+ } else {
+ // No buffers for this track. Give it a few chances to
+ // fill a buffer, then remove it from active list.
+ if (--(track->mRetryCount) <= 0) {
+ LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name());
+ tracksToRemove.add(track);
+ }
+ }
+ // LOGV("disable(%d)", track->name());
+ mAudioMixer->disable(AudioMixer::MIXING);
+ }
+ }
+
+ // remove all the tracks that need to be...
+ count = tracksToRemove.size();
+ if (UNLIKELY(count)) {
+ for (size_t i=0 ; i<count ; i++) {
+ const sp<Track>& track = tracksToRemove[i];
+ removeActiveTrack(track);
+ if (track->isTerminated()) {
+ mTracks.remove(track);
+ deleteTrackName(track->mName);
+ }
+ }
+ }
+ }
+
+ if (LIKELY(enabledTracks)) {
+ // mix buffers...
+ mAudioMixer->process(curBuf);
+
+#ifdef WITH_A2DP
+ if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
+ if (!outputTrackActive) {
+ LOGV("starting output track in mixer for output %d", mOutputType);
+ mOutputTrack->start();
+ outputTrackActive = true;
+ }
+ mOutputTrack->write(curBuf, mFrameCount);
+ }
+#endif
+
+ // output audio to hardware
+ mLastWriteTime = systemTime();
+ mInWrite = true;
+ mOutput->write(curBuf, mixBufferSize);
+ mNumWrites++;
+ mInWrite = false;
+ mStandby = false;
+ nsecs_t temp = systemTime();
+ standbyTime = temp + kStandbyTimeInNsecs;
+ nsecs_t delta = temp - mLastWriteTime;
+ if (delta > maxPeriod) {
+ LOGW("write blocked for %llu msecs", ns2ms(delta));
+ mNumDelayedWrites++;
+ }
+ sleepTime = kBufferRecoveryInUsecs;
+ } else {
+#ifdef WITH_A2DP
+ if (mOutputTrack != NULL && mAudioFlinger->isA2dpEnabled()) {
+ if (outputTrackActive) {
+ mOutputTrack->write(curBuf, 0);
+ if (mOutputTrack->bufferQueueEmpty()) {
+ mOutputTrack->stop();
+ outputTrackActive = false;
+ } else {
+ standbyTime = systemTime() + kStandbyTimeInNsecs;
+ }
+ }
+ }
+#endif
+ // There was nothing to mix this round, which means all
+ // active tracks were late. Sleep a little bit to give
+ // them another chance. If we're too late, the audio
+ // hardware will zero-fill for us.
+ //LOGV("no buffers - usleep(%lu)", sleepTime);
+ usleep(sleepTime);
+ if (sleepTime < kMaxBufferRecoveryInUsecs) {
+ sleepTime += kBufferRecoveryInUsecs;
+ }
+ }
+
+ // finally let go of all our tracks, without the lock held
+ // since we can't guarantee the destructors won't acquire that
+ // same lock.
+ tracksToRemove.clear();
+ } while (true);
+
+ return false;
+}
+
+status_t AudioFlinger::MixerThread::readyToRun()
+{
+ if (mSampleRate == 0) {
+ LOGE("No working audio driver found.");
+ return NO_INIT;
+ }
+ LOGI("AudioFlinger's thread ready to run for output %d", mOutputType);
+ return NO_ERROR;
+}
+
+void AudioFlinger::MixerThread::onFirstRef()
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+
+ snprintf(buffer, SIZE, "Mixer Thread for output %d", mOutputType);
+
+ run(buffer, ANDROID_PRIORITY_URGENT_AUDIO);
+}
+
+
+sp<AudioFlinger::MixerThread::Track> AudioFlinger::MixerThread::createTrack(
+ const sp<AudioFlinger::Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer,
+ status_t *status)
+{
+ sp<Track> track;
+ status_t lStatus;
+
+ // Resampler implementation limits input sampling rate to 2 x output sampling rate.
+ if (sampleRate > MAX_SAMPLE_RATE || sampleRate > mSampleRate*2) {
+ LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate);
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
+ {
+ Mutex::Autolock _l(mLock);
+
+ if (mSampleRate == 0) {
+ LOGE("Audio driver not initialized.");
+ lStatus = NO_INIT;
+ goto Exit;
+ }
+
+ track = new Track(this, client, streamType, sampleRate, format,
+ channelCount, frameCount, sharedBuffer);
+ if (track->getCblk() == NULL) {
+ track.clear();
+ lStatus = NO_MEMORY;
+ goto Exit;
+ }
+ mTracks.add(track);
+ lStatus = NO_ERROR;
+ }
+
+Exit:
+ if(status) {
+ *status = lStatus;
+ }
+ return track;
+}
+
+void AudioFlinger::MixerThread::getTracks(
+ SortedVector < sp<Track> >& tracks,
+ SortedVector < wp<Track> >& activeTracks)
+{
+ size_t size = mTracks.size();
+ LOGV ("MixerThread::getTracks() for output %d, mTracks.size %d, mActiveTracks.size %d", mOutputType, mTracks.size(), mActiveTracks.size());
+ for (size_t i = 0; i < size; i++) {
+ sp<Track> t = mTracks[i];
+ if (AudioSystem::routedToA2dpOutput(t->mStreamType)) {
+ tracks.add(t);
+ int j = mActiveTracks.indexOf(t);
+ if (j >= 0) {
+ t = mActiveTracks[j].promote();
+ if (t != NULL) {
+ activeTracks.add(t);
+ }
+ }
+ }
+ }
+
+ size = activeTracks.size();
+ for (size_t i = 0; i < size; i++) {
+ removeActiveTrack(activeTracks[i]);
+ }
+
+ size = tracks.size();
+ for (size_t i = 0; i < size; i++) {
+ sp<Track> t = tracks[i];
+ mTracks.remove(t);
+ deleteTrackName(t->name());
+ }
+}
+
+void AudioFlinger::MixerThread::putTracks(
+ SortedVector < sp<Track> >& tracks,
+ SortedVector < wp<Track> >& activeTracks)
+{
+
+ LOGV ("MixerThread::putTracks() for output %d, tracks.size %d, activeTracks.size %d", mOutputType, tracks.size(), activeTracks.size());
+
+ size_t size = tracks.size();
+ for (size_t i = 0; i < size ; i++) {
+ sp<Track> t = tracks[i];
+ int name = getTrackName();
+
+ if (name < 0) return;
+
+ t->mName = name;
+ t->mMixerThread = this;
+ mTracks.add(t);
+
+ int j = activeTracks.indexOf(t);
+ if (j >= 0) {
+ addActiveTrack(t);
+ }
+ }
+}
+
+uint32_t AudioFlinger::MixerThread::sampleRate() const
+{
+ return mSampleRate;
+}
+
+int AudioFlinger::MixerThread::channelCount() const
+{
+ return mChannelCount;
+}
+
+int AudioFlinger::MixerThread::format() const
+{
+ return mFormat;
+}
+
+size_t AudioFlinger::MixerThread::frameCount() const
+{
+ return mFrameCount;
+}
+
+uint32_t AudioFlinger::MixerThread::latency() const
+{
+ if (mOutput) {
+ return mOutput->latency();
+ }
+ else {
+ return 0;
+ }
+}
+
+status_t AudioFlinger::MixerThread::setMasterVolume(float value)
+{
+ mMasterVolume = value;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::MixerThread::setMasterMute(bool muted)
+{
+ mMasterMute = muted;
+ return NO_ERROR;
+}
+
+float AudioFlinger::MixerThread::masterVolume() const
+{
+ return mMasterVolume;
+}
+
+bool AudioFlinger::MixerThread::masterMute() const
+{
+ return mMasterMute;
+}
+
+status_t AudioFlinger::MixerThread::setStreamVolume(int stream, float value)
+{
+ mStreamTypes[stream].volume = value;
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::MixerThread::setStreamMute(int stream, bool muted)
+{
+ mStreamTypes[stream].mute = muted;
+ return NO_ERROR;
+}
+
+float AudioFlinger::MixerThread::streamVolume(int stream) const
+{
+ return mStreamTypes[stream].volume;
+}
+
+bool AudioFlinger::MixerThread::streamMute(int stream) const
+{
+ return mStreamTypes[stream].mute;
+}
+
+bool AudioFlinger::MixerThread::isMusicActive() const
+{
+ size_t count = mActiveTracks.size();
+ for (size_t i = 0 ; i < count ; ++i) {
+ sp<Track> t = mActiveTracks[i].promote();
+ if (t == 0) continue;
+ Track* const track = t.get();
+ if (t->mStreamType == AudioSystem::MUSIC)
+ return true;
+ }
+ return false;
+}
+
+status_t AudioFlinger::MixerThread::addTrack(const sp<Track>& track)
+{
+ status_t status = ALREADY_EXISTS;
+ Mutex::Autolock _l(mLock);
+
+ // here the track could be either new, or restarted
+ // in both cases "unstop" the track
+ if (track->isPaused()) {
+ track->mState = TrackBase::RESUMING;
+ LOGV("PAUSED => RESUMING (%d)", track->name());
+ } else {
+ track->mState = TrackBase::ACTIVE;
+ LOGV("? => ACTIVE (%d)", track->name());
+ }
+ // set retry count for buffer fill
+ track->mRetryCount = kMaxTrackStartupRetries;
+ if (mActiveTracks.indexOf(track) < 0) {
+ // the track is newly added, make sure it fills up all its
+ // buffers before playing. This is to ensure the client will
+ // effectively get the latency it requested.
+ track->mFillingUpStatus = Track::FS_FILLING;
+ track->mResetDone = false;
+ addActiveTrack(track);
+ status = NO_ERROR;
+ }
+
+ LOGV("mWaitWorkCV.broadcast");
+ mWaitWorkCV.broadcast();
+
+ return status;
+}
+
+void AudioFlinger::MixerThread::removeTrack(wp<Track> track, int name)
+{
+ Mutex::Autolock _l(mLock);
+ sp<Track> t = track.promote();
+ if (t!=NULL && (t->mState <= TrackBase::STOPPED)) {
+ remove_track_l(track, name);
+ }
+}
+
+void AudioFlinger::MixerThread::remove_track_l(wp<Track> track, int name)
+{
+ sp<Track> t = track.promote();
+ if (t!=NULL) {
+ t->reset();
+ }
+ deleteTrackName(name);
+ removeActiveTrack(track);
+ mWaitWorkCV.broadcast();
+}
+
+void AudioFlinger::MixerThread::destroyTrack(const sp<Track>& track)
+{
+ // NOTE: We're acquiring a strong reference on the track before
+ // acquiring the lock, this is to make sure removing it from
+ // mTracks won't cause the destructor to be called while the lock is
+ // held (note that technically, 'track' could be a reference to an item
+ // in mTracks, which is why we need to do this).
+ sp<Track> keep(track);
+ Mutex::Autolock _l(mLock);
+ track->mState = TrackBase::TERMINATED;
+ if (mActiveTracks.indexOf(track) < 0) {
+ LOGV("remove track (%d) and delete from mixer", track->name());
+ mTracks.remove(track);
+ deleteTrackName(keep->name());
+ }
+}
+
+
+void AudioFlinger::MixerThread::addActiveTrack(const wp<Track>& t)
+{
+ mActiveTracks.add(t);
+
+ // Force routing to speaker for certain stream types
+ // The forced routing to speaker is managed by hardware mixer
+ if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
+ sp<Track> track = t.promote();
+ if (track == NULL) return;
+
+ if (streamForcedToSpeaker(track->type())) {
+ mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_ADDED);
+ }
+ }
+}
+
+void AudioFlinger::MixerThread::removeActiveTrack(const wp<Track>& t)
+{
+ mActiveTracks.remove(t);
+
+ // Force routing to speaker for certain stream types
+ // The forced routing to speaker is managed by hardware mixer
+ if (mOutputType == AudioSystem::AUDIO_OUTPUT_HARDWARE) {
+ sp<Track> track = t.promote();
+ if (track == NULL) return;
+
+ if (streamForcedToSpeaker(track->type())) {
+ mAudioFlinger->handleForcedSpeakerRoute(ACTIVE_TRACK_REMOVED);
+ }
+ }
+}
+
+int AudioFlinger::MixerThread::getTrackName()
+{
+ return mAudioMixer->getTrackName();
+}
+
+void AudioFlinger::MixerThread::deleteTrackName(int name)
+{
+ mAudioMixer->deleteTrackName(name);
+}
+
+size_t AudioFlinger::MixerThread::getOutputFrameCount()
+{
+ return mOutput->bufferSize() / mOutput->channelCount() / sizeof(int16_t);
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::TrackBase::TrackBase(
+ const sp<MixerThread>& mixerThread,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags,
+ const sp<IMemory>& sharedBuffer)
+ : RefBase(),
+ mMixerThread(mixerThread),
+ mClient(client),
+ mStreamType(streamType),
+ mFrameCount(0),
+ mState(IDLE),
+ mClientTid(-1),
+ mFormat(format),
+ mFlags(flags & ~SYSTEM_FLAGS_MASK)
+{
+ mName = mixerThread->getTrackName();
+ LOGV("TrackBase contructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ if (mName < 0) {
+ LOGE("no more track names availlable");
+ return;
+ }
+
+ LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
+
+ // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
+ size_t size = sizeof(audio_track_cblk_t);
+ size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
+ if (sharedBuffer == 0) {
+ size += bufferSize;
+ }
+
+ if (client != NULL) {
+ 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.
+ new(mCblk) audio_track_cblk_t();
+ // clear all buffers
+ mCblk->frameCount = frameCount;
+ mCblk->sampleRate = sampleRate;
+ mCblk->channels = channelCount;
+ if (sharedBuffer == 0) {
+ mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+ memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
+ // Force underrun condition to avoid false underrun callback until first data is
+ // written to buffer
+ mCblk->flowControlFlag = 1;
+ } else {
+ mBuffer = sharedBuffer->pointer();
+ }
+ mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+ }
+ } else {
+ LOGE("not enough memory for AudioTrack size=%u", size);
+ client->heap()->dump("AudioTrack");
+ return;
+ }
+ } else {
+ mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
+ if (mCblk) { // construct the shared structure in-place.
+ new(mCblk) audio_track_cblk_t();
+ // clear all buffers
+ mCblk->frameCount = frameCount;
+ mCblk->sampleRate = sampleRate;
+ mCblk->channels = channelCount;
+ mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+ memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
+ // Force underrun condition to avoid false underrun callback until first data is
+ // written to buffer
+ mCblk->flowControlFlag = 1;
+ mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+ }
+ }
+}
+
+AudioFlinger::MixerThread::TrackBase::~TrackBase()
+{
+ if (mCblk) {
+ mCblk->~audio_track_cblk_t(); // destroy our shared-structure.
+ }
+ mCblkMemory.clear(); // and free the shared memory
+ mClient.clear();
+}
+
+void AudioFlinger::MixerThread::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ buffer->raw = 0;
+ mFrameCount = buffer->frameCount;
+ step();
+ buffer->frameCount = 0;
+}
+
+bool AudioFlinger::MixerThread::TrackBase::step() {
+ bool result;
+ audio_track_cblk_t* cblk = this->cblk();
+
+ result = cblk->stepServer(mFrameCount);
+ if (!result) {
+ LOGV("stepServer failed acquiring cblk mutex");
+ mFlags |= STEPSERVER_FAILED;
+ }
+ return result;
+}
+
+void AudioFlinger::MixerThread::TrackBase::reset() {
+ audio_track_cblk_t* cblk = this->cblk();
+
+ cblk->user = 0;
+ cblk->server = 0;
+ cblk->userBase = 0;
+ cblk->serverBase = 0;
+ mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK);
+ LOGV("TrackBase::reset");
+}
+
+sp<IMemory> AudioFlinger::MixerThread::TrackBase::getCblk() const
+{
+ return mCblkMemory;
+}
+
+int AudioFlinger::MixerThread::TrackBase::sampleRate() const {
+ return mCblk->sampleRate;
+}
+
+int AudioFlinger::MixerThread::TrackBase::channelCount() const {
+ return mCblk->channels;
+}
+
+void* AudioFlinger::MixerThread::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
+ audio_track_cblk_t* cblk = this->cblk();
+ int16_t *bufferStart = (int16_t *)mBuffer + (offset-cblk->serverBase)*cblk->channels;
+ int16_t *bufferEnd = bufferStart + frames * cblk->channels;
+
+ // Check validity of returned pointer in case the track control block would have been corrupted.
+ if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd) {
+ LOGW("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 bufferStart;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::Track::Track(
+ const sp<MixerThread>& mixerThread,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ const sp<IMemory>& sharedBuffer)
+ : TrackBase(mixerThread, client, streamType, sampleRate, format, channelCount, frameCount, 0, sharedBuffer)
+{
+ mVolume[0] = 1.0f;
+ mVolume[1] = 1.0f;
+ mMute = false;
+ mSharedBuffer = sharedBuffer;
+}
+
+AudioFlinger::MixerThread::Track::~Track()
+{
+ wp<Track> weak(this); // never create a strong ref from the dtor
+ mState = TERMINATED;
+ mMixerThread->removeTrack(weak, mName);
+}
+
+void AudioFlinger::MixerThread::Track::destroy()
+{
+ mMixerThread->destroyTrack(this);
+}
+
+void AudioFlinger::MixerThread::Track::dump(char* buffer, size_t size)
+{
+ snprintf(buffer, size, " %5d %5d %3u %3u %3u %3u %1d %1d %1d %5u %5u %5u %04x %04x\n",
+ mName - AudioMixer::TRACK0,
+ (mClient == NULL) ? getpid() : mClient->pid(),
+ mStreamType,
+ mFormat,
+ mCblk->channels,
+ mFrameCount,
+ mState,
+ mMute,
+ mFillingUpStatus,
+ mCblk->sampleRate,
+ mCblk->volume[0],
+ mCblk->volume[1],
+ mCblk->server,
+ mCblk->user);
+}
+
+status_t AudioFlinger::MixerThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ 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 (!step()) goto getNextBuffer_exit;
+ LOGV("stepServer recovered");
+ mFlags &= ~TrackBase::STEPSERVER_FAILED;
+ }
+
+ framesReady = cblk->framesReady();
+
+ if (LIKELY(framesReady)) {
+ uint32_t s = cblk->server;
+ uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
+
+ bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd;
+ if (framesReq > framesReady) {
+ framesReq = framesReady;
+ }
+ if (s + framesReq > bufferEnd) {
+ framesReq = bufferEnd - s;
+ }
+
+ buffer->raw = getBuffer(s, framesReq);
+ if (buffer->raw == 0) goto getNextBuffer_exit;
+
+ buffer->frameCount = framesReq;
+ return NO_ERROR;
+ }
+
+getNextBuffer_exit:
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+}
+
+bool AudioFlinger::MixerThread::Track::isReady() const {
+ if (mFillingUpStatus != FS_FILLING) return true;
+
+ if (mCblk->framesReady() >= mCblk->frameCount ||
+ mCblk->forceReady) {
+ mFillingUpStatus = FS_FILLED;
+ mCblk->forceReady = 0;
+ LOGV("Track::isReady() track %d for output %d", mName, mMixerThread->mOutputType);
+ return true;
+ }
+ return false;
+}
+
+status_t AudioFlinger::MixerThread::Track::start()
+{
+ LOGV("start(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
+ mMixerThread->addTrack(this);
+ return NO_ERROR;
+}
+
+void AudioFlinger::MixerThread::Track::stop()
+{
+ LOGV("stop(%d), calling thread %d for output %d", mName, IPCThreadState::self()->getCallingPid(), mMixerThread->mOutputType);
+ Mutex::Autolock _l(mMixerThread->mLock);
+ if (mState > STOPPED) {
+ mState = STOPPED;
+ // If the track is not active (PAUSED and buffers full), flush buffers
+ if (mMixerThread->mActiveTracks.indexOf(this) < 0) {
+ reset();
+ }
+ LOGV("(> STOPPED) => STOPPED (%d)", mName);
+ }
+}
+
+void AudioFlinger::MixerThread::Track::pause()
+{
+ LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid());
+ Mutex::Autolock _l(mMixerThread->mLock);
+ if (mState == ACTIVE || mState == RESUMING) {
+ mState = PAUSING;
+ LOGV("ACTIVE/RESUMING => PAUSING (%d)", mName);
+ }
+}
+
+void AudioFlinger::MixerThread::Track::flush()
+{
+ LOGV("flush(%d)", mName);
+ Mutex::Autolock _l(mMixerThread->mLock);
+ if (mState != STOPPED && mState != PAUSED && mState != PAUSING) {
+ return;
+ }
+ // No point remaining in PAUSED state after a flush => go to
+ // STOPPED state
+ mState = STOPPED;
+
+ // NOTE: reset() will reset cblk->user and cblk->server with
+ // the risk that at the same time, the AudioMixer is trying to read
+ // data. In this case, getNextBuffer() would return a NULL pointer
+ // as audio buffer => the AudioMixer code MUST always test that pointer
+ // returned by getNextBuffer() is not NULL!
+ reset();
+}
+
+void AudioFlinger::MixerThread::Track::reset()
+{
+ // Do not reset twice to avoid discarding data written just after a flush and before
+ // the audioflinger thread detects the track is stopped.
+ if (!mResetDone) {
+ TrackBase::reset();
+ // Force underrun condition to avoid false underrun callback until first data is
+ // written to buffer
+ mCblk->flowControlFlag = 1;
+ mCblk->forceReady = 0;
+ mFillingUpStatus = FS_FILLING;
+ mResetDone = true;
+ }
+}
+
+void AudioFlinger::MixerThread::Track::mute(bool muted)
+{
+ mMute = muted;
+}
+
+void AudioFlinger::MixerThread::Track::setVolume(float left, float right)
+{
+ mVolume[0] = left;
+ mVolume[1] = right;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::RecordTrack::RecordTrack(
+ const sp<MixerThread>& mixerThread,
+ const sp<Client>& client,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags)
+ : TrackBase(mixerThread, client, streamType, sampleRate, format,
+ channelCount, frameCount, flags, 0),
+ mOverflow(false)
+{
+}
+
+AudioFlinger::MixerThread::RecordTrack::~RecordTrack()
+{
+ mMixerThread->deleteTrackName(mName);
+}
+
+status_t AudioFlinger::MixerThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ 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 (!step()) goto getNextBuffer_exit;
+ LOGV("stepServer recovered");
+ mFlags &= ~TrackBase::STEPSERVER_FAILED;
+ }
+
+ framesAvail = cblk->framesAvailable_l();
+
+ if (LIKELY(framesAvail)) {
+ uint32_t s = cblk->server;
+ uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
+
+ if (framesReq > framesAvail) {
+ framesReq = framesAvail;
+ }
+ if (s + framesReq > bufferEnd) {
+ framesReq = bufferEnd - s;
+ }
+
+ buffer->raw = getBuffer(s, framesReq);
+ if (buffer->raw == 0) goto getNextBuffer_exit;
+
+ buffer->frameCount = framesReq;
+ return NO_ERROR;
+ }
+
+getNextBuffer_exit:
+ buffer->raw = 0;
+ buffer->frameCount = 0;
+ return NOT_ENOUGH_DATA;
+}
+
+status_t AudioFlinger::MixerThread::RecordTrack::start()
+{
+ return mMixerThread->mAudioFlinger->startRecord(this);
+}
+
+void AudioFlinger::MixerThread::RecordTrack::stop()
+{
+ mMixerThread->mAudioFlinger->stopRecord(this);
+ TrackBase::reset();
+ // Force overerrun condition to avoid false overrun callback until first data is
+ // read from buffer
+ mCblk->flowControlFlag = 1;
+}
+
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::MixerThread::OutputTrack::OutputTrack(
+ const sp<MixerThread>& mixerThread,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount)
+ : Track(mixerThread, NULL, AudioSystem::SYSTEM, sampleRate, format, channelCount, frameCount, NULL),
+ mOutputMixerThread(mixerThread)
+{
+
+ mCblk->out = 1;
+ mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ mCblk->volume[0] = mCblk->volume[1] = 0x1000;
+ mOutBuffer.frameCount = 0;
+ mCblk->bufferTimeoutMs = 10;
+
+ LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p",
+ mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd);
+
+}
+
+AudioFlinger::MixerThread::OutputTrack::~OutputTrack()
+{
+ stop();
+}
+
+status_t AudioFlinger::MixerThread::OutputTrack::start()
+{
+ status_t status = Track::start();
+
+ mRetryCount = 127;
+ return status;
+}
+
+void AudioFlinger::MixerThread::OutputTrack::stop()
+{
+ Track::stop();
+ clearBufferQueue();
+ mOutBuffer.frameCount = 0;
+}
+
+void AudioFlinger::MixerThread::OutputTrack::write(int16_t* data, uint32_t frames)
+{
+ Buffer *pInBuffer;
+ Buffer inBuffer;
+ uint32_t channels = mCblk->channels;
+
+ inBuffer.frameCount = frames;
+ inBuffer.i16 = data;
+
+ if (mCblk->user == 0) {
+ if (mOutputMixerThread->isMusicActive()) {
+ mCblk->forceReady = 1;
+ LOGV("OutputTrack::start() force ready");
+ } else if (mCblk->frameCount > frames){
+ if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
+ uint32_t startFrames = (mCblk->frameCount - frames);
+ LOGV("OutputTrack::start() write %d frames", startFrames);
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[startFrames * channels];
+ pInBuffer->frameCount = startFrames;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ } else {
+ LOGW ("OutputTrack::write() no more buffers");
+ }
+ }
+ }
+
+ while (1) {
+ // First write pending buffers, then new data
+ if (mBufferQueue.size()) {
+ pInBuffer = mBufferQueue.itemAt(0);
+ } else {
+ pInBuffer = &inBuffer;
+ }
+
+ if (pInBuffer->frameCount == 0) {
+ break;
+ }
+
+ if (mOutBuffer.frameCount == 0) {
+ mOutBuffer.frameCount = pInBuffer->frameCount;
+ if (obtainBuffer(&mOutBuffer) == (status_t)AudioTrack::NO_MORE_BUFFERS) {
+ break;
+ }
+ }
+
+ uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
+ memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t));
+ mCblk->stepUser(outFrames);
+ pInBuffer->frameCount -= outFrames;
+ pInBuffer->i16 += outFrames * channels;
+ mOutBuffer.frameCount -= outFrames;
+ mOutBuffer.i16 += outFrames * channels;
+
+ if (pInBuffer->frameCount == 0) {
+ if (mBufferQueue.size()) {
+ mBufferQueue.removeAt(0);
+ delete [] pInBuffer->mBuffer;
+ delete pInBuffer;
+ } else {
+ break;
+ }
+ }
+ }
+
+ // If we could not write all frames, allocate a buffer and queue it for next time.
+ if (inBuffer.frameCount) {
+ if (mBufferQueue.size() < kMaxOutputTrackBuffers) {
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels];
+ pInBuffer->frameCount = inBuffer.frameCount;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ } else {
+ LOGW("OutputTrack::write() no more buffers");
+ }
+ }
+
+ // Calling write() with a 0 length buffer, means that no more data will be written:
+ // If no more buffers are pending, fill output track buffer to make sure it is started
+ // by output mixer.
+ if (frames == 0 && mBufferQueue.size() == 0 && mCblk->user < mCblk->frameCount) {
+ frames = mCblk->frameCount - mCblk->user;
+ pInBuffer = new Buffer;
+ pInBuffer->mBuffer = new int16_t[frames * channels];
+ pInBuffer->frameCount = frames;
+ pInBuffer->i16 = pInBuffer->mBuffer;
+ memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
+ mBufferQueue.add(pInBuffer);
+ }
+
+}
+
+status_t AudioFlinger::MixerThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer)
+{
+ int active;
+ int timeout = 0;
+ status_t result;
+ audio_track_cblk_t* cblk = mCblk;
+ uint32_t framesReq = buffer->frameCount;
+
+ LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server);
+ buffer->frameCount = 0;
+
+ uint32_t framesAvail = cblk->framesAvailable();
+
+ if (framesAvail == 0) {
+ return AudioTrack::NO_MORE_BUFFERS;
+ }
+
+ if (framesReq > framesAvail) {
+ framesReq = framesAvail;
+ }
+
+ uint32_t u = cblk->user;
+ uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
+
+ if (u + framesReq > bufferEnd) {
+ framesReq = bufferEnd - u;
+ }
+
+ buffer->frameCount = framesReq;
+ buffer->raw = (void *)cblk->buffer(u);
+ return NO_ERROR;
+}
+
+
+void AudioFlinger::MixerThread::OutputTrack::clearBufferQueue()
+{
+ size_t size = mBufferQueue.size();
+ Buffer *pBuffer;
+
+ for (size_t i = 0; i < size; i++) {
+ pBuffer = mBufferQueue.itemAt(i);
+ delete [] pBuffer->mBuffer;
+ delete pBuffer;
+ }
+ mBufferQueue.clear();
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid)
+ : RefBase(),
+ mAudioFlinger(audioFlinger),
+ mMemoryDealer(new MemoryDealer(1024*1024)),
+ mPid(pid)
+{
+ // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer
+}
+
+AudioFlinger::Client::~Client()
+{
+ mAudioFlinger->removeClient(mPid);
+}
+
+const sp<MemoryDealer>& AudioFlinger::Client::heap() const
+{
+ return mMemoryDealer;
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::MixerThread::Track>& track)
+ : BnAudioTrack(),
+ mTrack(track)
+{
+}
+
+AudioFlinger::TrackHandle::~TrackHandle() {
+ // just stop the track on deletion, associated resources
+ // will be freed from the main thread once all pending buffers have
+ // been played. Unless it's not in the active track list, in which
+ // case we free everything now...
+ mTrack->destroy();
+}
+
+status_t AudioFlinger::TrackHandle::start() {
+ return mTrack->start();
+}
+
+void AudioFlinger::TrackHandle::stop() {
+ mTrack->stop();
+}
+
+void AudioFlinger::TrackHandle::flush() {
+ mTrack->flush();
+}
+
+void AudioFlinger::TrackHandle::mute(bool e) {
+ mTrack->mute(e);
+}
+
+void AudioFlinger::TrackHandle::pause() {
+ 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::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ return BnAudioTrack::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+sp<IAudioRecord> AudioFlinger::openRecord(
+ pid_t pid,
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags,
+ status_t *status)
+{
+ sp<AudioRecordThread> thread;
+ sp<MixerThread::RecordTrack> recordTrack;
+ sp<RecordHandle> recordHandle;
+ sp<Client> client;
+ wp<Client> wclient;
+ AudioStreamIn* input = 0;
+ int inFrameCount;
+ size_t inputBufferSize;
+ status_t lStatus;
+
+ // check calling permissions
+ if (!recordingAllowed()) {
+ lStatus = PERMISSION_DENIED;
+ goto Exit;
+ }
+
+ if (uint32_t(streamType) >= AudioRecord::NUM_STREAM_TYPES) {
+ LOGE("invalid stream type");
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
+ if (sampleRate > MAX_SAMPLE_RATE) {
+ LOGE("Sample rate out of range");
+ lStatus = BAD_VALUE;
+ goto Exit;
+ }
+
+ if (mAudioRecordThread == 0) {
+ LOGE("Audio record thread not started");
+ lStatus = NO_INIT;
+ goto Exit;
+ }
+
+
+ // Check that audio input stream accepts requested audio parameters
+ inputBufferSize = mAudioHardware->getInputBufferSize(sampleRate, format, channelCount);
+ if (inputBufferSize == 0) {
+ lStatus = BAD_VALUE;
+ LOGE("Bad audio input parameters: sampling rate %u, format %d, channels %d", sampleRate, format, channelCount);
+ goto Exit;
+ }
+
+ // add client to list
+ {
+ Mutex::Autolock _l(mLock);
+ wclient = mClients.valueFor(pid);
+ if (wclient != NULL) {
+ client = wclient.promote();
+ } else {
+ client = new Client(this, pid);
+ mClients.add(pid, client);
+ }
+ }
+
+ // frameCount must be a multiple of input buffer size
+ inFrameCount = inputBufferSize/channelCount/sizeof(short);
+ frameCount = ((frameCount - 1)/inFrameCount + 1) * inFrameCount;
+
+ // create new record track and pass to record thread
+ recordTrack = new MixerThread::RecordTrack(mHardwareMixerThread, client, streamType, sampleRate,
+ format, channelCount, frameCount, flags);
+ if (recordTrack->getCblk() == NULL) {
+ recordTrack.clear();
+ lStatus = NO_MEMORY;
+ goto Exit;
+ }
+
+ // return to handle to client
+ recordHandle = new RecordHandle(recordTrack);
+ lStatus = NO_ERROR;
+
+Exit:
+ if (status) {
+ *status = lStatus;
+ }
+ return recordHandle;
+}
+
+status_t AudioFlinger::startRecord(MixerThread::RecordTrack* recordTrack) {
+ if (mAudioRecordThread != 0) {
+ return mAudioRecordThread->start(recordTrack);
+ }
+ return NO_INIT;
+}
+
+void AudioFlinger::stopRecord(MixerThread::RecordTrack* recordTrack) {
+ if (mAudioRecordThread != 0) {
+ mAudioRecordThread->stop(recordTrack);
+ }
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::MixerThread::RecordTrack>& recordTrack)
+ : BnAudioRecord(),
+ mRecordTrack(recordTrack)
+{
+}
+
+AudioFlinger::RecordHandle::~RecordHandle() {
+ stop();
+}
+
+status_t AudioFlinger::RecordHandle::start() {
+ LOGV("RecordHandle::start()");
+ return mRecordTrack->start();
+}
+
+void AudioFlinger::RecordHandle::stop() {
+ LOGV("RecordHandle::stop()");
+ 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)
+{
+ return BnAudioRecord::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+
+AudioFlinger::AudioRecordThread::AudioRecordThread(AudioHardwareInterface* audioHardware) :
+ mAudioHardware(audioHardware),
+ mActive(false)
+{
+}
+
+AudioFlinger::AudioRecordThread::~AudioRecordThread()
+{
+}
+
+bool AudioFlinger::AudioRecordThread::threadLoop()
+{
+ LOGV("AudioRecordThread: start record loop");
+ AudioBufferProvider::Buffer buffer;
+ int inBufferSize = 0;
+ int inFrameCount = 0;
+ AudioStreamIn* input = 0;
+
+ mActive = 0;
+
+ // start recording
+ while (!exitPending()) {
+ if (!mActive) {
+ mLock.lock();
+ if (!mActive && !exitPending()) {
+ LOGV("AudioRecordThread: loop stopping");
+ if (input) {
+ delete input;
+ input = 0;
+ }
+ mRecordTrack.clear();
+ mStopped.signal();
+
+ mWaitWorkCV.wait(mLock);
+
+ LOGV("AudioRecordThread: loop starting");
+ if (mRecordTrack != 0) {
+ input = mAudioHardware->openInputStream(mRecordTrack->format(),
+ mRecordTrack->channelCount(),
+ mRecordTrack->sampleRate(),
+ &mStartStatus,
+ (AudioSystem::audio_in_acoustics)(mRecordTrack->mFlags >> 16));
+ if (input != 0) {
+ inBufferSize = input->bufferSize();
+ inFrameCount = inBufferSize/input->frameSize();
+ }
+ } else {
+ mStartStatus = NO_INIT;
+ }
+ if (mStartStatus !=NO_ERROR) {
+ LOGW("record start failed, status %d", mStartStatus);
+ mActive = false;
+ mRecordTrack.clear();
+ }
+ mWaitWorkCV.signal();
+ }
+ mLock.unlock();
+ } else if (mRecordTrack != 0) {
+
+ buffer.frameCount = inFrameCount;
+ if (LIKELY(mRecordTrack->getNextBuffer(&buffer) == NO_ERROR)) {
+ LOGV("AudioRecordThread read: %d frames", buffer.frameCount);
+ ssize_t bytesRead = input->read(buffer.raw, inBufferSize);
+ if (bytesRead < 0) {
+ LOGE("Error reading audio input");
+ sleep(1);
+ }
+ mRecordTrack->releaseBuffer(&buffer);
+ mRecordTrack->overflow();
+ }
+
+ // client isn't retrieving buffers fast enough
+ else {
+ if (!mRecordTrack->setOverflow())
+ LOGW("AudioRecordThread: buffer overflow");
+ // Release the processor for a while before asking for a new buffer.
+ // This will give the application more chance to read from the buffer and
+ // clear the overflow.
+ usleep(5000);
+ }
+ }
+ }
+
+
+ if (input) {
+ delete input;
+ }
+ mRecordTrack.clear();
+
+ return false;
+}
+
+status_t AudioFlinger::AudioRecordThread::start(MixerThread::RecordTrack* recordTrack)
+{
+ LOGV("AudioRecordThread::start");
+ AutoMutex lock(&mLock);
+ mActive = true;
+ // If starting the active track, just reset mActive in case a stop
+ // was pending and exit
+ if (recordTrack == mRecordTrack.get()) return NO_ERROR;
+
+ if (mRecordTrack != 0) return -EBUSY;
+
+ mRecordTrack = recordTrack;
+
+ // signal thread to start
+ LOGV("Signal record thread");
+ mWaitWorkCV.signal();
+ mWaitWorkCV.wait(mLock);
+ LOGV("Record started, status %d", mStartStatus);
+ return mStartStatus;
+}
+
+void AudioFlinger::AudioRecordThread::stop(MixerThread::RecordTrack* recordTrack) {
+ LOGV("AudioRecordThread::stop");
+ AutoMutex lock(&mLock);
+ if (mActive && (recordTrack == mRecordTrack.get())) {
+ mActive = false;
+ mStopped.wait(mLock);
+ }
+}
+
+void AudioFlinger::AudioRecordThread::exit()
+{
+ LOGV("AudioRecordThread::exit");
+ {
+ AutoMutex lock(&mLock);
+ requestExit();
+ mWaitWorkCV.signal();
+ }
+ requestExitAndWait();
+}
+
+status_t AudioFlinger::AudioRecordThread::dump(int fd, const Vector<String16>& args)
+{
+ const size_t SIZE = 256;
+ char buffer[SIZE];
+ String8 result;
+ pid_t pid = 0;
+
+ if (mRecordTrack != 0 && mRecordTrack->mClient != 0) {
+ snprintf(buffer, SIZE, "Record client pid: %d\n", mRecordTrack->mClient->pid());
+ result.append(buffer);
+ } else {
+ result.append("No record client\n");
+ }
+ write(fd, result.string(), result.size());
+ return NO_ERROR;
+}
+
+status_t AudioFlinger::onTransact(
+ uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags)
+{
+ return BnAudioFlinger::onTransact(code, data, reply, flags);
+}
+
+// ----------------------------------------------------------------------------
+void AudioFlinger::instantiate() {
+ defaultServiceManager()->addService(
+ String16("media.audio_flinger"), new AudioFlinger());
+}
+
+}; // namespace android