Merge "aaudio: set errorCallback user data" into oc-dev
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 80aad2f..dfd5df7 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -1048,6 +1048,10 @@
bool haveVideo = false;
for (size_t i = 0; i < numTracks; ++i) {
sp<IMediaSource> source = extractor->getTrack(i);
+ if (source == nullptr) {
+ fprintf(stderr, "skip NULL track %zu, track count %zu.\n", i, numTracks);
+ continue;
+ }
const char *mime;
CHECK(source->getFormat()->findCString(
@@ -1110,6 +1114,10 @@
}
mediaSource = extractor->getTrack(i);
+ if (mediaSource == nullptr) {
+ fprintf(stderr, "skip NULL track %zu, total tracks %zu.\n", i, numTracks);
+ return -1;
+ }
}
}
diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index 8f9333a..2e1d240 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp
@@ -171,7 +171,8 @@
mWriter = new MPEG2TSWriter(
this, &MyConvertingStreamSource::WriteDataWrapper);
- for (size_t i = 0; i < extractor->countTracks(); ++i) {
+ size_t numTracks = extractor->countTracks();
+ for (size_t i = 0; i < numTracks; ++i) {
const sp<MetaData> &meta = extractor->getTrackMetaData(i);
const char *mime;
@@ -181,7 +182,12 @@
continue;
}
- CHECK_EQ(mWriter->addSource(extractor->getTrack(i)), (status_t)OK);
+ sp<IMediaSource> track = extractor->getTrack(i);
+ if (track == nullptr) {
+ fprintf(stderr, "skip NULL track %zu, total tracks %zu\n", i, numTracks);
+ continue;
+ }
+ CHECK_EQ(mWriter->addSource(track), (status_t)OK);
}
CHECK_EQ(mWriter->start(), (status_t)OK);
diff --git a/include/ndk/NdkImageReader.h b/include/ndk/NdkImageReader.h
index a158da9..e3600c2 100644
--- a/include/ndk/NdkImageReader.h
+++ b/include/ndk/NdkImageReader.h
@@ -307,22 +307,38 @@
* for the consumer usage. All other parameters and the return values are identical to those passed
* to {@line AImageReader_new}.
*
- * @param usage0 specifies how the consumer will access the AImage, using combination of the
- * AHARDWAREBUFFER_USAGE0 flags described in {@link hardware_buffer.h}.
- * Passing {@link AHARDWAREBUFFER_USAGE0_CPU_READ_OFTEN} is equivalent to calling
- * {@link AImageReader_new} with the same parameters. Note that consumers that do not
- * require CPU access to the buffer should omit {@link
- * AHARDWAREBUFFER_USAGE0_CPU_READ_OFTEN} to improve performance.
- * @param usage1 specifies how the consumer will access the AImage, using combination of the
- * AHARDWAREBUFFER_USAGE1 flags described in {@link hardware_buffer.h}.
+ * @param usage specifies how the consumer will access the AImage, using combination of the
+ * AHARDWAREBUFFER_USAGE flags described in {@link hardware_buffer.h}.
+ * Passing {@link AHARDWAREBUFFER_USAGE_CPU_READ_OFTEN} is equivalent to calling
+ * {@link AImageReader_new} with the same parameters.
+ *
+ * Note that not all format and usage flag combination is supported by the {@link AImageReader}.
+ * Below are the combinations supported by the {@link AImageReader}.
+ * <table>
+ * <tr>
+ * <th>Format</th>
+ * <th>Compatible usage flags</th>
+ * </tr>
+ * <tr>
+ * <td>non-{@link AIMAGE_FORMAT_PRIVATE PRIVATE} formats defined in {@link AImage.h}
+ * </td>
+ * <td>{@link AHARDWAREBUFFER_USAGE_CPU_READ_RARELY} or
+ * {@link AHARDWAREBUFFER_USAGE_CPU_READ_OFTEN}</td>
+ * </tr>
+ * <tr>
+ * <td>{@link AIMAGE_FORMAT_RGBA_8888}</td>
+ * <td>{@link AHARDWAREBUFFER_USAGE_VIDEO_ENCODE} or
+ * {@link AHARDWAREBUFFER_USAGE_GPU_SAMPLED_IMAGE}, or combined</td>
+ * </tr>
+ * </table>
*
* @see AImage
* @see AImageReader_new
* @see AHardwareBuffer
*/
media_status_t AImageReader_newWithUsage(
- int32_t width, int32_t height, int32_t format, uint64_t usage0,
- uint64_t usage1, int32_t maxImages, /*out*/ AImageReader** reader);
+ int32_t width, int32_t height, int32_t format, uint64_t usage, int32_t maxImages,
+ /*out*/ AImageReader** reader);
/*
* Acquire the next {@link AImage} from the image reader's queue asynchronously.
diff --git a/media/libaaudio/examples/write_sine/jni/Android.mk b/media/libaaudio/examples/write_sine/jni/Android.mk
index 5a884e1..0bda008 100644
--- a/media/libaaudio/examples/write_sine/jni/Android.mk
+++ b/media/libaaudio/examples/write_sine/jni/Android.mk
@@ -4,7 +4,8 @@
LOCAL_MODULE_TAGS := tests
LOCAL_C_INCLUDES := \
$(call include-path-for, audio-utils) \
- frameworks/av/media/libaaudio/include
+ frameworks/av/media/libaaudio/include \
+ frameworks/av/media/libaaudio/src
# NDK recommends using this kind of relative path instead of an absolute path.
LOCAL_SRC_FILES:= ../src/write_sine.cpp
diff --git a/media/libaaudio/examples/write_sine/src/SineGenerator.h b/media/libaaudio/examples/write_sine/src/SineGenerator.h
index f2eb984..64b772d 100644
--- a/media/libaaudio/examples/write_sine/src/SineGenerator.h
+++ b/media/libaaudio/examples/write_sine/src/SineGenerator.h
@@ -79,7 +79,7 @@
}
}
- double mAmplitude = 0.005; // unitless scaler
+ double mAmplitude = 0.05; // unitless scaler
double mPhase = 0.0;
double mPhaseIncrement = 440 * M_PI * 2 / 48000;
double mFrameRate = 48000;
diff --git a/media/libaaudio/examples/write_sine/src/write_sine.cpp b/media/libaaudio/examples/write_sine/src/write_sine.cpp
index df55c3f..9107a7c 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine.cpp
@@ -22,8 +22,8 @@
#include <aaudio/AAudio.h>
#include "SineGenerator.h"
-#define SAMPLE_RATE 48000
-#define NUM_SECONDS 5
+#define SAMPLE_RATE 48000
+#define NUM_SECONDS 15
#define NANOS_PER_MICROSECOND ((int64_t)1000)
#define NANOS_PER_MILLISECOND (NANOS_PER_MICROSECOND * 1000)
#define NANOS_PER_SECOND (NANOS_PER_MILLISECOND * 1000)
diff --git a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
index a7e32bd..cc0c3a4 100644
--- a/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
+++ b/media/libaaudio/examples/write_sine/src/write_sine_callback.cpp
@@ -26,7 +26,7 @@
#include <aaudio/AAudio.h>
#include "SineGenerator.h"
-#define NUM_SECONDS 5
+#define NUM_SECONDS 15
//#define SHARING_MODE AAUDIO_SHARING_MODE_EXCLUSIVE
#define SHARING_MODE AAUDIO_SHARING_MODE_SHARED
diff --git a/media/libaaudio/examples/write_sine/static/Android.mk b/media/libaaudio/examples/write_sine/static/Android.mk
index e4da6a8..3fee08a 100644
--- a/media/libaaudio/examples/write_sine/static/Android.mk
+++ b/media/libaaudio/examples/write_sine/static/Android.mk
@@ -4,6 +4,7 @@
LOCAL_MODULE_TAGS := examples
LOCAL_C_INCLUDES := \
$(call include-path-for, audio-utils) \
+ frameworks/av/media/libaaudio/src \
frameworks/av/media/libaaudio/include
# NDK recommends using this kind of relative path instead of an absolute path.
diff --git a/media/libaaudio/src/Android.mk b/media/libaaudio/src/Android.mk
index b5bb75f..f43c0ad 100644
--- a/media/libaaudio/src/Android.mk
+++ b/media/libaaudio/src/Android.mk
@@ -39,6 +39,7 @@
utility/FixedBlockAdapter.cpp \
utility/FixedBlockReader.cpp \
utility/FixedBlockWriter.cpp \
+ utility/LinearRamp.cpp \
fifo/FifoBuffer.cpp \
fifo/FifoControllerBase.cpp \
client/AudioEndpoint.cpp \
@@ -93,6 +94,7 @@
utility/FixedBlockAdapter.cpp \
utility/FixedBlockReader.cpp \
utility/FixedBlockWriter.cpp \
+ utility/LinearRamp.cpp \
fifo/FifoBuffer.cpp \
fifo/FifoControllerBase.cpp \
client/AudioEndpoint.cpp \
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index af4b93a..810751a 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -32,9 +32,10 @@
#include "binding/AAudioStreamConfiguration.h"
#include "binding/IAAudioService.h"
#include "binding/AAudioServiceMessage.h"
-#include "fifo/FifoBuffer.h"
-
#include "core/AudioStreamBuilder.h"
+#include "fifo/FifoBuffer.h"
+#include "utility/LinearRamp.h"
+
#include "AudioStreamInternal.h"
#define LOG_TIMESTAMPS 0
@@ -478,8 +479,9 @@
ALOGW("WARNING - processCommands() AAUDIO_SERVICE_EVENT_DISCONNECTED");
break;
case AAUDIO_SERVICE_EVENT_VOLUME:
- mVolume = message->event.dataDouble;
- ALOGD_IF(MYLOG_CONDITION, "processCommands() AAUDIO_SERVICE_EVENT_VOLUME %f", mVolume);
+ mVolumeRamp.setTarget((float) message->event.dataDouble);
+ ALOGD_IF(MYLOG_CONDITION, "processCommands() AAUDIO_SERVICE_EVENT_VOLUME %f",
+ message->event.dataDouble);
break;
default:
ALOGW("WARNING - processCommands() Unrecognized event = %d",
@@ -639,10 +641,10 @@
}
-// TODO this function needs a major cleanup.
aaudio_result_t AudioStreamInternal::writeNowWithConversion(const void *buffer,
int32_t numFrames) {
- // ALOGD_IF(MYLOG_CONDITION, "AudioStreamInternal::writeNowWithConversion(%p, %d)", buffer, numFrames);
+ // ALOGD_IF(MYLOG_CONDITION, "AudioStreamInternal::writeNowWithConversion(%p, %d)",
+ // buffer, numFrames);
WrappingBuffer wrappingBuffer;
uint8_t *source = (uint8_t *) buffer;
int32_t framesLeft = numFrames;
@@ -659,31 +661,67 @@
framesToWrite = framesAvailable;
}
int32_t numBytes = getBytesPerFrame() * framesToWrite;
- // TODO handle volume scaling
- if (getFormat() == mDeviceFormat) {
- // Copy straight through.
- memcpy(wrappingBuffer.data[partIndex], source, numBytes);
- } else if (getFormat() == AAUDIO_FORMAT_PCM_FLOAT
- && mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
- // Data conversion.
- AAudioConvert_floatToPcm16(
- (const float *) source,
- framesToWrite * getSamplesPerFrame(),
- (int16_t *) wrappingBuffer.data[partIndex]);
- } else if (getFormat() == AAUDIO_FORMAT_PCM_I16
- && mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT) {
- // Data conversion.
- AAudioConvert_pcm16ToFloat(
- (const int16_t *) source,
- framesToWrite * getSamplesPerFrame(),
- (float *) wrappingBuffer.data[partIndex]);
- } else {
- // TODO handle more conversions
- ALOGE("AudioStreamInternal::writeNowWithConversion() unsupported formats: %d, %d",
- getFormat(), mDeviceFormat);
- return AAUDIO_ERROR_UNEXPECTED_VALUE;
+ int32_t numSamples = framesToWrite * getSamplesPerFrame();
+ // Data conversion.
+ float levelFrom;
+ float levelTo;
+ bool ramping = mVolumeRamp.nextSegment(framesToWrite * getSamplesPerFrame(),
+ &levelFrom, &levelTo);
+ // The formats are validated when the stream is opened so we do not have to
+ // check for illegal combinations here.
+ if (getFormat() == AAUDIO_FORMAT_PCM_FLOAT) {
+ if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+ AAudio_linearRamp(
+ (const float *) source,
+ (float *) wrappingBuffer.data[partIndex],
+ framesToWrite,
+ getSamplesPerFrame(),
+ levelFrom,
+ levelTo);
+ } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
+ if (ramping) {
+ AAudioConvert_floatToPcm16(
+ (const float *) source,
+ (int16_t *) wrappingBuffer.data[partIndex],
+ framesToWrite,
+ getSamplesPerFrame(),
+ levelFrom,
+ levelTo);
+ } else {
+ AAudioConvert_floatToPcm16(
+ (const float *) source,
+ (int16_t *) wrappingBuffer.data[partIndex],
+ numSamples,
+ levelTo);
+ }
+ }
+ } else if (getFormat() == AAUDIO_FORMAT_PCM_I16) {
+ if (mDeviceFormat == AAUDIO_FORMAT_PCM_FLOAT) {
+ if (ramping) {
+ AAudioConvert_pcm16ToFloat(
+ (const int16_t *) source,
+ (float *) wrappingBuffer.data[partIndex],
+ framesToWrite,
+ getSamplesPerFrame(),
+ levelFrom,
+ levelTo);
+ } else {
+ AAudioConvert_pcm16ToFloat(
+ (const int16_t *) source,
+ (float *) wrappingBuffer.data[partIndex],
+ numSamples,
+ levelTo);
+ }
+ } else if (mDeviceFormat == AAUDIO_FORMAT_PCM_I16) {
+ AAudio_linearRamp(
+ (const int16_t *) source,
+ (int16_t *) wrappingBuffer.data[partIndex],
+ framesToWrite,
+ getSamplesPerFrame(),
+ levelFrom,
+ levelTo);
+ }
}
-
source += numBytes;
framesLeft -= framesToWrite;
} else {
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 8244311..e550ba3 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -22,11 +22,11 @@
#include "binding/IAAudioService.h"
#include "binding/AudioEndpointParcelable.h"
+#include "binding/AAudioServiceInterface.h"
#include "client/IsochronousClockModel.h"
#include "client/AudioEndpoint.h"
#include "core/AudioStream.h"
-
-#include "binding/AAudioServiceInterface.h"
+#include "utility/LinearRamp.h"
using android::sp;
using android::IAAudioService;
@@ -154,7 +154,7 @@
int64_t mLastFramesRead = 0; // used to prevent retrograde motion
int32_t mFramesPerBurst; // frames per HAL transfer
int32_t mXRunCount = 0; // how many underrun events?
- float mVolume = 1.0; // volume that the server told us to use
+ LinearRamp mVolumeRamp;
AAudioServiceInterface &mServiceInterface; // abstract interface to the service
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 1bb9e53..96fd427 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -32,6 +32,7 @@
// Arbitrary and somewhat generous number of bursts.
#define DEFAULT_BURSTS_PER_BUFFER_CAPACITY 8
+static const bool FAST_TRACKS_ENABLED = true;
/*
* Create a stream that uses the AudioTrack.
@@ -69,7 +70,9 @@
samplesPerFrame, channelMask);
// TODO add more performance options
- audio_output_flags_t flags = (audio_output_flags_t) AUDIO_OUTPUT_FLAG_FAST;
+ audio_output_flags_t flags = FAST_TRACKS_ENABLED
+ ? AUDIO_OUTPUT_FLAG_FAST
+ : AUDIO_OUTPUT_FLAG_NONE;
int32_t frameCount = builder.getBufferCapacity();
ALOGD("AudioStreamTrack::open(), requested buffer capacity %d", frameCount);
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index efbbfc5..5fa228a 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -27,6 +27,11 @@
using namespace android;
+// This is 3 dB, (10^(3/20)), to match the maximum headroom in AudioTrack for float data.
+// It is designed to allow occasional transient peaks.
+#define MAX_HEADROOM (1.41253754f)
+#define MIN_HEADROOM (0 - MAX_HEADROOM)
+
int32_t AAudioConvert_formatToSizeInBytes(aaudio_audio_format_t format) {
int32_t size = AAUDIO_ERROR_ILLEGAL_ARGUMENT;
switch (format) {
@@ -42,24 +47,153 @@
return size;
}
-// TODO This similar to a function in audio_utils. Consider using that instead.
-void AAudioConvert_floatToPcm16(const float *source, int32_t numSamples, int16_t *destination) {
+
+// TODO call clamp16_from_float function in primitives.h
+static inline int16_t clamp16_from_float(float f) {
+ /* Offset is used to expand the valid range of [-1.0, 1.0) into the 16 lsbs of the
+ * floating point significand. The normal shift is 3<<22, but the -15 offset
+ * is used to multiply by 32768.
+ */
+ static const float offset = (float)(3 << (22 - 15));
+ /* zero = (0x10f << 22) = 0x43c00000 (not directly used) */
+ static const int32_t limneg = (0x10f << 22) /*zero*/ - 32768; /* 0x43bf8000 */
+ static const int32_t limpos = (0x10f << 22) /*zero*/ + 32767; /* 0x43c07fff */
+
+ union {
+ float f;
+ int32_t i;
+ } u;
+
+ u.f = f + offset; /* recenter valid range */
+ /* Now the valid range is represented as integers between [limneg, limpos].
+ * Clamp using the fact that float representation (as an integer) is an ordered set.
+ */
+ if (u.i < limneg)
+ u.i = -32768;
+ else if (u.i > limpos)
+ u.i = 32767;
+ return u.i; /* Return lower 16 bits, the part of interest in the significand. */
+}
+
+// Same but without clipping.
+// Convert -1.0f to +1.0f to -32768 to +32767
+static inline int16_t floatToInt16(float f) {
+ static const float offset = (float)(3 << (22 - 15));
+ union {
+ float f;
+ int32_t i;
+ } u;
+ u.f = f + offset; /* recenter valid range */
+ return u.i; /* Return lower 16 bits, the part of interest in the significand. */
+}
+
+static float clipAndClampFloatToPcm16(float sample, float scaler) {
+ // Clip to valid range of a float sample to prevent excessive volume.
+ if (sample > MAX_HEADROOM) sample = MAX_HEADROOM;
+ else if (sample < MIN_HEADROOM) sample = MIN_HEADROOM;
+
+ // Scale and convert to a short.
+ float fval = sample * scaler;
+ return clamp16_from_float(fval);
+}
+
+void AAudioConvert_floatToPcm16(const float *source,
+ int16_t *destination,
+ int32_t numSamples,
+ float amplitude) {
+ float scaler = amplitude;
for (int i = 0; i < numSamples; i++) {
- float fval = source[i];
- fval += 1.0; // to avoid discontinuity at 0.0 caused by truncation
- fval *= 32768.0f;
- int32_t sample = (int32_t) fval;
- // clip to 16-bit range
- if (sample < 0) sample = 0;
- else if (sample > 0x0FFFF) sample = 0x0FFFF;
- sample -= 32768; // center at zero
- destination[i] = (int16_t) sample;
+ float sample = *source++;
+ *destination++ = clipAndClampFloatToPcm16(sample, scaler);
}
}
-void AAudioConvert_pcm16ToFloat(const int16_t *source, int32_t numSamples, float *destination) {
+void AAudioConvert_floatToPcm16(const float *source,
+ int16_t *destination,
+ int32_t numFrames,
+ int32_t samplesPerFrame,
+ float amplitude1,
+ float amplitude2) {
+ float scaler = amplitude1;
+ // divide by numFrames so that we almost reach amplitude2
+ float delta = (amplitude2 - amplitude1) / numFrames;
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
+ float sample = *source++;
+ *destination++ = clipAndClampFloatToPcm16(sample, scaler);
+ }
+ scaler += delta;
+ }
+}
+
+#define SHORT_SCALE 32768
+
+void AAudioConvert_pcm16ToFloat(const int16_t *source,
+ float *destination,
+ int32_t numSamples,
+ float amplitude) {
+ float scaler = amplitude / SHORT_SCALE;
for (int i = 0; i < numSamples; i++) {
- destination[i] = source[i] * (1.0f / 32768.0f);
+ destination[i] = source[i] * scaler;
+ }
+}
+
+// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
+void AAudioConvert_pcm16ToFloat(const int16_t *source,
+ float *destination,
+ int32_t numFrames,
+ int32_t samplesPerFrame,
+ float amplitude1,
+ float amplitude2) {
+ float scaler = amplitude1 / SHORT_SCALE;
+ float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
+ *destination++ = *source++ * scaler;
+ }
+ scaler += delta;
+ }
+}
+
+// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
+void AAudio_linearRamp(const float *source,
+ float *destination,
+ int32_t numFrames,
+ int32_t samplesPerFrame,
+ float amplitude1,
+ float amplitude2) {
+ float scaler = amplitude1;
+ float delta = (amplitude2 - amplitude1) / numFrames;
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
+ float sample = *source++;
+
+ // Clip to valid range of a float sample to prevent excessive volume.
+ if (sample > MAX_HEADROOM) sample = MAX_HEADROOM;
+ else if (sample < MIN_HEADROOM) sample = MIN_HEADROOM;
+
+ *destination++ = sample * scaler;
+ }
+ scaler += delta;
+ }
+}
+
+// This code assumes amplitude1 and amplitude2 are between 0.0 and 1.0
+void AAudio_linearRamp(const int16_t *source,
+ int16_t *destination,
+ int32_t numFrames,
+ int32_t samplesPerFrame,
+ float amplitude1,
+ float amplitude2) {
+ float scaler = amplitude1 / SHORT_SCALE;
+ float delta = (amplitude2 - amplitude1) / (SHORT_SCALE * (float) numFrames);
+ for (int frameIndex = 0; frameIndex < numFrames; frameIndex++) {
+ for (int sampleIndex = 0; sampleIndex < samplesPerFrame; sampleIndex++) {
+ // No need to clip because int16_t range is inherently limited.
+ float sample = *source++ * scaler;
+ *destination++ = floatToInt16(sample);
+ }
+ scaler += delta;
}
}
diff --git a/media/libaaudio/src/utility/AAudioUtilities.h b/media/libaaudio/src/utility/AAudioUtilities.h
index 3dc501e..0078cbb 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.h
+++ b/media/libaaudio/src/utility/AAudioUtilities.h
@@ -35,9 +35,120 @@
*/
aaudio_result_t AAudioConvert_androidToAAudioResult(android::status_t status);
-void AAudioConvert_floatToPcm16(const float *source, int32_t numSamples, int16_t *destination);
+/**
+ * Convert an array of floats to an array of int16_t.
+ *
+ * @param source
+ * @param destination
+ * @param numSamples number of values in the array
+ * @param amplitude level between 0.0 and 1.0
+ */
+void AAudioConvert_floatToPcm16(const float *source,
+ int16_t *destination,
+ int32_t numSamples,
+ float amplitude);
-void AAudioConvert_pcm16ToFloat(const int16_t *source, int32_t numSamples, float *destination);
+/**
+ * Convert floats to int16_t and scale by a linear ramp.
+ *
+ * The ramp stops just short of reaching amplitude2 so that the next
+ * ramp can start at amplitude2 without causing a discontinuity.
+ *
+ * @param source
+ * @param destination
+ * @param numFrames
+ * @param samplesPerFrame AKA number of channels
+ * @param amplitude1 level at start of ramp, between 0.0 and 1.0
+ * @param amplitude2 level past end of ramp, between 0.0 and 1.0
+ */
+void AAudioConvert_floatToPcm16(const float *source,
+ int16_t *destination,
+ int32_t numFrames,
+ int32_t samplesPerFrame,
+ float amplitude1,
+ float amplitude2);
+
+/**
+ * Convert int16_t array to float array ranging from -1.0 to +1.0.
+ * @param source
+ * @param destination
+ * @param numSamples
+ */
+//void AAudioConvert_pcm16ToFloat(const int16_t *source, int32_t numSamples,
+// float *destination);
+
+/**
+ *
+ * Convert int16_t array to float array ranging from +/- amplitude.
+ * @param source
+ * @param destination
+ * @param numSamples
+ * @param amplitude
+ */
+void AAudioConvert_pcm16ToFloat(const int16_t *source,
+ float *destination,
+ int32_t numSamples,
+ float amplitude);
+
+/**
+ * Convert floats to int16_t and scale by a linear ramp.
+ *
+ * The ramp stops just short of reaching amplitude2 so that the next
+ * ramp can start at amplitude2 without causing a discontinuity.
+ *
+ * @param source
+ * @param destination
+ * @param numFrames
+ * @param samplesPerFrame AKA number of channels
+ * @param amplitude1 level at start of ramp, between 0.0 and 1.0
+ * @param amplitude2 level at end of ramp, between 0.0 and 1.0
+ */
+void AAudioConvert_pcm16ToFloat(const int16_t *source,
+ float *destination,
+ int32_t numFrames,
+ int32_t samplesPerFrame,
+ float amplitude1,
+ float amplitude2);
+
+/**
+ * Scale floats by a linear ramp.
+ *
+ * The ramp stops just short of reaching amplitude2 so that the next
+ * ramp can start at amplitude2 without causing a discontinuity.
+ *
+ * @param source
+ * @param destination
+ * @param numFrames
+ * @param samplesPerFrame
+ * @param amplitude1
+ * @param amplitude2
+ */
+void AAudio_linearRamp(const float *source,
+ float *destination,
+ int32_t numFrames,
+ int32_t samplesPerFrame,
+ float amplitude1,
+ float amplitude2);
+
+/**
+ * Scale int16_t's by a linear ramp.
+ *
+ * The ramp stops just short of reaching amplitude2 so that the next
+ * ramp can start at amplitude2 without causing a discontinuity.
+ *
+ * @param source
+ * @param destination
+ * @param numFrames
+ * @param samplesPerFrame
+ * @param amplitude1
+ * @param amplitude2
+ */
+void AAudio_linearRamp(const int16_t *source,
+ int16_t *destination,
+ int32_t numFrames,
+ int32_t samplesPerFrame,
+ float amplitude1,
+ float amplitude2);
/**
* Calculate the number of bytes and prevent numeric overflow.
diff --git a/media/libaaudio/src/utility/LinearRamp.cpp b/media/libaaudio/src/utility/LinearRamp.cpp
new file mode 100644
index 0000000..1714bbf
--- /dev/null
+++ b/media/libaaudio/src/utility/LinearRamp.cpp
@@ -0,0 +1,50 @@
+/*
+ * Copyright 2017 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 "LinearRamp.h"
+
+bool LinearRamp::isRamping() {
+ float target = mTarget.load();
+ if (target != mLevelTo) {
+ // Update target. Continue from previous level.
+ mLevelTo = target;
+ mRemaining = mLengthInFrames;
+ return true;
+ } else {
+ return mRemaining > 0;
+ }
+}
+
+bool LinearRamp::nextSegment(int32_t frames, float *levelFrom, float *levelTo) {
+ bool ramping = isRamping();
+ *levelFrom = mLevelFrom;
+ if (ramping) {
+ float level;
+ if (frames >= mRemaining) {
+ level = mLevelTo;
+ mRemaining = 0;
+ } else {
+ // Interpolate to a point along the full ramp.
+ level = mLevelFrom + (frames * (mLevelTo - mLevelFrom) / mRemaining);
+ mRemaining -= frames;
+ }
+ mLevelFrom = level; // for next ramp
+ *levelTo = level;
+ } else {
+ *levelTo = mLevelTo;
+ }
+ return ramping;
+}
\ No newline at end of file
diff --git a/media/libaaudio/src/utility/LinearRamp.h b/media/libaaudio/src/utility/LinearRamp.h
new file mode 100644
index 0000000..ff09dce
--- /dev/null
+++ b/media/libaaudio/src/utility/LinearRamp.h
@@ -0,0 +1,97 @@
+/*
+ * Copyright 2017 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 AAUDIO_LINEAR_RAMP_H
+#define AAUDIO_LINEAR_RAMP_H
+
+#include <atomic>
+#include <stdint.h>
+
+/**
+ * Generate segments along a linear ramp.
+ * The ramp target can be updated from another thread.
+ * When the target is updated, a new ramp is started from the current position.
+ *
+ * The first ramp starts at 0.0.
+ *
+ */
+class LinearRamp {
+public:
+ LinearRamp() {
+ mTarget.store(1.0f);
+ }
+
+ void setLengthInFrames(int32_t frames) {
+ mLengthInFrames = frames;
+ }
+
+ int32_t getLengthInFrames() {
+ return mLengthInFrames;
+ }
+
+ /**
+ * This may be called by another thread.
+ * @param target
+ */
+ void setTarget(float target) {
+ mTarget.store(target);
+ }
+
+ float getTarget() {
+ return mTarget.load();
+ }
+
+ /**
+ * Force the nextSegment to start from this level.
+ *
+ * WARNING: this can cause a discontinuity if called while the ramp is being used.
+ * Only call this when setting the initial ramp.
+ *
+ * @param level
+ */
+ void forceCurrent(float level) {
+ mLevelFrom = level;
+ mLevelTo = level; // forces a ramp if it does not match target
+ }
+
+ float getCurrent() {
+ return mLevelFrom;
+ }
+
+ /**
+ * Get levels for next ramp segment.
+ *
+ * @param frames number of frames in the segment
+ * @param levelFrom pointer to starting amplitude
+ * @param levelTo pointer to ending amplitude
+ * @return true if ramp is still moving towards the target
+ */
+ bool nextSegment(int32_t frames, float *levelFrom, float *levelTo);
+
+private:
+
+ bool isRamping();
+
+ std::atomic<float> mTarget;
+
+ int32_t mLengthInFrames = 48000 / 50; // 20 msec at 48000 Hz
+ int32_t mRemaining = 0;
+ float mLevelFrom = 0.0f;
+ float mLevelTo = 0.0f;
+};
+
+
+#endif //AAUDIO_LINEAR_RAMP_H
diff --git a/media/libaaudio/tests/Android.mk b/media/libaaudio/tests/Android.mk
index 06c9364..01360b1 100644
--- a/media/libaaudio/tests/Android.mk
+++ b/media/libaaudio/tests/Android.mk
@@ -35,3 +35,15 @@
LOCAL_STATIC_LIBRARIES := libaaudio
LOCAL_MODULE := test_block_adapter
include $(BUILD_NATIVE_TEST)
+
+include $(CLEAR_VARS)
+LOCAL_C_INCLUDES := \
+ $(call include-path-for, audio-utils) \
+ frameworks/av/media/libaaudio/include \
+ frameworks/av/media/libaaudio/src
+LOCAL_SRC_FILES:= test_linear_ramp.cpp
+LOCAL_SHARED_LIBRARIES := libaudioclient libaudioutils libbinder \
+ libcutils liblog libmedia libutils
+LOCAL_STATIC_LIBRARIES := libaaudio
+LOCAL_MODULE := test_linear_ramp
+include $(BUILD_NATIVE_TEST)
diff --git a/media/libaaudio/tests/test_linear_ramp.cpp b/media/libaaudio/tests/test_linear_ramp.cpp
new file mode 100644
index 0000000..5c53982
--- /dev/null
+++ b/media/libaaudio/tests/test_linear_ramp.cpp
@@ -0,0 +1,108 @@
+/*
+ * Copyright (C) 2017 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 <iostream>
+
+#include <gtest/gtest.h>
+
+#include "utility/AAudioUtilities.h"
+#include "utility/LinearRamp.h"
+
+
+TEST(test_linear_ramp, linear_ramp_segments) {
+ LinearRamp ramp;
+ const float source[4] = {1.0f, 1.0f, 1.0f, 1.0f };
+ float destination[4] = {1.0f, 1.0f, 1.0f, 1.0f };
+
+ float levelFrom = -1.0f;
+ float levelTo = -1.0f;
+ ramp.setLengthInFrames(8);
+ ramp.setTarget(8.0f);
+
+ ASSERT_EQ(8, ramp.getLengthInFrames());
+
+ bool ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
+ ASSERT_EQ(1, ramping);
+ ASSERT_EQ(0.0f, levelFrom);
+ ASSERT_EQ(4.0f, levelTo);
+
+ AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
+ ASSERT_EQ(0.0f, destination[0]);
+ ASSERT_EQ(1.0f, destination[1]);
+ ASSERT_EQ(2.0f, destination[2]);
+ ASSERT_EQ(3.0f, destination[3]);
+
+ ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
+ ASSERT_EQ(1, ramping);
+ ASSERT_EQ(4.0f, levelFrom);
+ ASSERT_EQ(8.0f, levelTo);
+
+ AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
+ ASSERT_EQ(4.0f, destination[0]);
+ ASSERT_EQ(5.0f, destination[1]);
+ ASSERT_EQ(6.0f, destination[2]);
+ ASSERT_EQ(7.0f, destination[3]);
+
+ ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
+ ASSERT_EQ(0, ramping);
+ ASSERT_EQ(8.0f, levelFrom);
+ ASSERT_EQ(8.0f, levelTo);
+
+ AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
+ ASSERT_EQ(8.0f, destination[0]);
+ ASSERT_EQ(8.0f, destination[1]);
+ ASSERT_EQ(8.0f, destination[2]);
+ ASSERT_EQ(8.0f, destination[3]);
+
+};
+
+
+TEST(test_linear_ramp, linear_ramp_forced) {
+ LinearRamp ramp;
+ const float source[4] = {1.0f, 1.0f, 1.0f, 1.0f };
+ float destination[4] = {1.0f, 1.0f, 1.0f, 1.0f };
+
+ float levelFrom = -1.0f;
+ float levelTo = -1.0f;
+ ramp.setLengthInFrames(4);
+ ramp.setTarget(8.0f);
+ ramp.forceCurrent(4.0f);
+ ASSERT_EQ(4.0f, ramp.getCurrent());
+
+ bool ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
+ ASSERT_EQ(1, ramping);
+ ASSERT_EQ(4.0f, levelFrom);
+ ASSERT_EQ(8.0f, levelTo);
+
+ AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
+ ASSERT_EQ(4.0f, destination[0]);
+ ASSERT_EQ(5.0f, destination[1]);
+ ASSERT_EQ(6.0f, destination[2]);
+ ASSERT_EQ(7.0f, destination[3]);
+
+ ramping = ramp.nextSegment(4, &levelFrom, &levelTo);
+ ASSERT_EQ(0, ramping);
+ ASSERT_EQ(8.0f, levelFrom);
+ ASSERT_EQ(8.0f, levelTo);
+
+ AAudio_linearRamp(source, destination, 4, 1, levelFrom, levelTo);
+ ASSERT_EQ(8.0f, destination[0]);
+ ASSERT_EQ(8.0f, destination[1]);
+ ASSERT_EQ(8.0f, destination[2]);
+ ASSERT_EQ(8.0f, destination[3]);
+
+};
+
diff --git a/media/libaudiohal/Android.mk b/media/libaudiohal/Android.mk
index 68a1f7b..e592169 100644
--- a/media/libaudiohal/Android.mk
+++ b/media/libaudiohal/Android.mk
@@ -5,30 +5,37 @@
LOCAL_SHARED_LIBRARIES := \
libcutils \
liblog \
- libutils
+ libutils \
+ libhardware
+
+LOCAL_SRC_FILES := \
+ DeviceHalLocal.cpp \
+ DevicesFactoryHalHybrid.cpp \
+ DevicesFactoryHalLocal.cpp \
+ StreamHalLocal.cpp
+
+LOCAL_CFLAGS := -Wall -Werror
ifeq ($(USE_LEGACY_LOCAL_AUDIO_HAL), true)
# Use audiohal directly w/o hwbinder middleware.
# This is for performance comparison and debugging only.
-LOCAL_SRC_FILES := \
- DeviceHalLocal.cpp \
- DevicesFactoryHalLocal.cpp \
+LOCAL_SRC_FILES += \
EffectBufferHalLocal.cpp \
- EffectHalLocal.cpp \
EffectsFactoryHalLocal.cpp \
- StreamHalLocal.cpp
+ EffectHalLocal.cpp
LOCAL_SHARED_LIBRARIES += \
- libeffects \
- libhardware
+ libeffects
+
+LOCAL_CFLAGS += -DUSE_LEGACY_LOCAL_AUDIO_HAL
else # if !USE_LEGACY_LOCAL_AUDIO_HAL
-LOCAL_SRC_FILES := \
+LOCAL_SRC_FILES += \
ConversionHelperHidl.cpp \
- HalDeathHandlerHidl.cpp \
+ HalDeathHandlerHidl.cpp \
DeviceHalHidl.cpp \
DevicesFactoryHalHidl.cpp \
EffectBufferHalHidl.cpp \
@@ -60,6 +67,4 @@
LOCAL_MODULE := libaudiohal
-LOCAL_CFLAGS := -Wall -Werror
-
include $(BUILD_SHARED_LIBRARY)
diff --git a/media/libaudiohal/DevicesFactoryHalHidl.cpp b/media/libaudiohal/DevicesFactoryHalHidl.cpp
index fc2645e..31da263 100644
--- a/media/libaudiohal/DevicesFactoryHalHidl.cpp
+++ b/media/libaudiohal/DevicesFactoryHalHidl.cpp
@@ -33,11 +33,6 @@
namespace android {
-// static
-sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
- return new DevicesFactoryHalHidl();
-}
-
DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
mDevicesFactory = IDevicesFactory::getService();
if (mDevicesFactory != 0) {
diff --git a/media/libaudiohal/DevicesFactoryHalHidl.h b/media/libaudiohal/DevicesFactoryHalHidl.h
index a26dec1..e2f1ad1 100644
--- a/media/libaudiohal/DevicesFactoryHalHidl.h
+++ b/media/libaudiohal/DevicesFactoryHalHidl.h
@@ -36,7 +36,7 @@
virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
private:
- friend class DevicesFactoryHalInterface;
+ friend class DevicesFactoryHalHybrid;
sp<IDevicesFactory> mDevicesFactory;
diff --git a/media/libaudiohal/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/DevicesFactoryHalHybrid.cpp
new file mode 100644
index 0000000..454b03b
--- /dev/null
+++ b/media/libaudiohal/DevicesFactoryHalHybrid.cpp
@@ -0,0 +1,54 @@
+/*
+ * Copyright (C) 2017 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 "DevicesFactoryHalHybrid"
+//#define LOG_NDEBUG 0
+
+#include "DevicesFactoryHalHybrid.h"
+#include "DevicesFactoryHalLocal.h"
+#ifndef USE_LEGACY_LOCAL_AUDIO_HAL
+#include "DevicesFactoryHalHidl.h"
+#endif
+
+namespace android {
+
+// static
+sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
+ return new DevicesFactoryHalHybrid();
+}
+
+DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
+ : mLocalFactory(new DevicesFactoryHalLocal()),
+ mHidlFactory(
+#ifdef USE_LEGACY_LOCAL_AUDIO_HAL
+ nullptr
+#else
+ new DevicesFactoryHalHidl()
+#endif
+ ) {
+}
+
+DevicesFactoryHalHybrid::~DevicesFactoryHalHybrid() {
+}
+
+status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
+ if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0) {
+ return mHidlFactory->openDevice(name, device);
+ }
+ return mLocalFactory->openDevice(name, device);
+}
+
+} // namespace android
diff --git a/media/libaudiohal/DevicesFactoryHalHybrid.h b/media/libaudiohal/DevicesFactoryHalHybrid.h
new file mode 100644
index 0000000..abd57d6
--- /dev/null
+++ b/media/libaudiohal/DevicesFactoryHalHybrid.h
@@ -0,0 +1,47 @@
+/*
+ * Copyright (C) 2017 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_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
+#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
+
+#include <media/audiohal/DevicesFactoryHalInterface.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+class DevicesFactoryHalHybrid : public DevicesFactoryHalInterface
+{
+ public:
+ // Opens a device with the specified name. To close the device, it is
+ // necessary to release references to the returned object.
+ virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
+
+ private:
+ friend class DevicesFactoryHalInterface;
+
+ // Can not be constructed directly by clients.
+ DevicesFactoryHalHybrid();
+
+ virtual ~DevicesFactoryHalHybrid();
+
+ sp<DevicesFactoryHalInterface> mLocalFactory;
+ sp<DevicesFactoryHalInterface> mHidlFactory;
+};
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
diff --git a/media/libaudiohal/DevicesFactoryHalLocal.cpp b/media/libaudiohal/DevicesFactoryHalLocal.cpp
index cd9a9e7..13a9acd 100644
--- a/media/libaudiohal/DevicesFactoryHalLocal.cpp
+++ b/media/libaudiohal/DevicesFactoryHalLocal.cpp
@@ -27,11 +27,6 @@
namespace android {
-// static
-sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
- return new DevicesFactoryHalLocal();
-}
-
static status_t load_audio_interface(const char *if_name, audio_hw_device_t **dev)
{
const hw_module_t *mod;
diff --git a/media/libaudiohal/DevicesFactoryHalLocal.h b/media/libaudiohal/DevicesFactoryHalLocal.h
index 58ce4ff..b9d18ab 100644
--- a/media/libaudiohal/DevicesFactoryHalLocal.h
+++ b/media/libaudiohal/DevicesFactoryHalLocal.h
@@ -33,7 +33,7 @@
virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
private:
- friend class DevicesFactoryHalInterface;
+ friend class DevicesFactoryHalHybrid;
// Can not be constructed directly by clients.
DevicesFactoryHalLocal() {}
diff --git a/media/libaudiohal/EffectBufferHalHidl.cpp b/media/libaudiohal/EffectBufferHalHidl.cpp
index ef4097a..8b5201b 100644
--- a/media/libaudiohal/EffectBufferHalHidl.cpp
+++ b/media/libaudiohal/EffectBufferHalHidl.cpp
@@ -47,7 +47,7 @@
status_t EffectBufferHalInterface::mirror(
void* external, size_t size, sp<EffectBufferHalInterface>* buffer) {
sp<EffectBufferHalInterface> tempBuffer = new EffectBufferHalHidl(size);
- status_t result = reinterpret_cast<EffectBufferHalHidl*>(tempBuffer.get())->init();
+ status_t result = static_cast<EffectBufferHalHidl*>(tempBuffer.get())->init();
if (result == OK) {
tempBuffer->setExternalData(external);
*buffer = tempBuffer;
diff --git a/media/libaudiohal/EffectHalHidl.cpp b/media/libaudiohal/EffectHalHidl.cpp
index 7d61443..b49b975 100644
--- a/media/libaudiohal/EffectHalHidl.cpp
+++ b/media/libaudiohal/EffectHalHidl.cpp
@@ -209,8 +209,8 @@
status_t EffectHalHidl::setProcessBuffers() {
Return<Result> ret = mEffect->setProcessBuffers(
- reinterpret_cast<EffectBufferHalHidl*>(mInBuffer.get())->hidlBuffer(),
- reinterpret_cast<EffectBufferHalHidl*>(mOutBuffer.get())->hidlBuffer());
+ static_cast<EffectBufferHalHidl*>(mInBuffer.get())->hidlBuffer(),
+ static_cast<EffectBufferHalHidl*>(mOutBuffer.get())->hidlBuffer());
if (ret.isOk() && ret == Result::OK) {
mBuffersChanged = false;
return OK;
diff --git a/media/libaudiohal/EffectHalHidl.h b/media/libaudiohal/EffectHalHidl.h
index 0d011aa..6ffdaf1 100644
--- a/media/libaudiohal/EffectHalHidl.h
+++ b/media/libaudiohal/EffectHalHidl.h
@@ -58,6 +58,9 @@
// Free resources on the remote side.
virtual status_t close();
+ // Whether it's a local implementation.
+ virtual bool isLocal() const { return false; }
+
uint64_t effectId() const { return mEffectId; }
static void effectDescriptorToHal(
diff --git a/media/libaudiohal/EffectHalLocal.h b/media/libaudiohal/EffectHalLocal.h
index b499462..693fb50 100644
--- a/media/libaudiohal/EffectHalLocal.h
+++ b/media/libaudiohal/EffectHalLocal.h
@@ -48,6 +48,9 @@
// Free resources on the remote side.
virtual status_t close();
+ // Whether it's a local implementation.
+ virtual bool isLocal() const { return true; }
+
effect_handle_t handle() const { return mHandle; }
private:
diff --git a/media/libaudiohal/StreamHalLocal.cpp b/media/libaudiohal/StreamHalLocal.cpp
index b25e518..05800a0 100644
--- a/media/libaudiohal/StreamHalLocal.cpp
+++ b/media/libaudiohal/StreamHalLocal.cpp
@@ -79,11 +79,13 @@
}
status_t StreamHalLocal::addEffect(sp<EffectHalInterface> effect) {
+ LOG_ALWAYS_FATAL_IF(!effect->isLocal(), "Only local effects can be added for a local stream");
return mStream->add_audio_effect(mStream,
static_cast<EffectHalLocal*>(effect.get())->handle());
}
status_t StreamHalLocal::removeEffect(sp<EffectHalInterface> effect) {
+ LOG_ALWAYS_FATAL_IF(!effect->isLocal(), "Only local effects can be removed for a local stream");
return mStream->remove_audio_effect(mStream,
static_cast<EffectHalLocal*>(effect.get())->handle());
}
@@ -162,7 +164,7 @@
// correctly the case when the callback is invoked while StreamOutHalLocal's destructor is
// already running, because the destructor is invoked after the refcount has been atomically
// decremented.
- wp<StreamOutHalLocal> weakSelf(reinterpret_cast<StreamOutHalLocal*>(cookie));
+ wp<StreamOutHalLocal> weakSelf(static_cast<StreamOutHalLocal*>(cookie));
sp<StreamOutHalLocal> self = weakSelf.promote();
if (self == 0) return 0;
sp<StreamOutHalInterfaceCallback> callback = self->mCallback.promote();
diff --git a/media/libaudiohal/include/EffectHalInterface.h b/media/libaudiohal/include/EffectHalInterface.h
index 7f9a6fd..92622aa 100644
--- a/media/libaudiohal/include/EffectHalInterface.h
+++ b/media/libaudiohal/include/EffectHalInterface.h
@@ -52,6 +52,9 @@
// Free resources on the remote side.
virtual status_t close() = 0;
+ // Whether it's a local implementation.
+ virtual bool isLocal() const = 0;
+
protected:
// Subclasses can not be constructed directly by clients.
EffectHalInterface() {}
diff --git a/media/libmedia/include/media/IMediaExtractor.h b/media/libmedia/include/media/IMediaExtractor.h
index cf1b9fb..ab40f53 100644
--- a/media/libmedia/include/media/IMediaExtractor.h
+++ b/media/libmedia/include/media/IMediaExtractor.h
@@ -34,6 +34,9 @@
DECLARE_META_INTERFACE(MediaExtractor);
virtual size_t countTracks() = 0;
+ // This function could return NULL IMediaSource even when index is within the
+ // track count returned by countTracks, since it's possible the track is malformed
+ // and it's not detected during countTracks call.
virtual sp<IMediaSource> getTrack(size_t index) = 0;
enum GetTrackMetaDataFlags {
diff --git a/media/libmediaplayerservice/nuplayer/GenericSource.cpp b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
index fb1f42b..22b09d4 100644
--- a/media/libmediaplayerservice/nuplayer/GenericSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/GenericSource.cpp
@@ -379,6 +379,11 @@
source.get(), mFd, (long long)mOffset, (long long)mLength);
if (source.get() != nullptr) {
mDataSource = DataSource::CreateFromIDataSource(source);
+ if (mDataSource != nullptr) {
+ // Close the local file descriptor as it is not needed anymore.
+ close(mFd);
+ mFd = -1;
+ }
} else {
ALOGW("extractor service cannot make data source");
}
@@ -390,7 +395,9 @@
ALOGD("FileSource local");
mDataSource = new FileSource(mFd, mOffset, mLength);
}
-
+ // TODO: close should always be done on mFd, see the lines following
+ // DataSource::CreateFromIDataSource above,
+ // and the FileSource constructor should dup the mFd argument as needed.
mFd = -1;
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
index d048777..0d4c730 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.cpp
@@ -202,7 +202,8 @@
mPaused(false),
mPausedByClient(true),
mPausedForBuffering(false),
- mIsDrmProtected(false) {
+ mIsDrmProtected(false),
+ mDataSourceType(DATA_SOURCE_TYPE_NONE) {
clearFlushComplete();
}
@@ -225,6 +226,7 @@
msg->setObject("source", new StreamingSource(notify, source));
msg->post();
+ mDataSourceType = DATA_SOURCE_TYPE_STREAM;
}
static bool IsHTTPLiveURL(const char *url) {
@@ -258,10 +260,12 @@
if (IsHTTPLiveURL(url)) {
source = new HTTPLiveSource(notify, httpService, url, headers);
ALOGV("setDataSourceAsync HTTPLiveSource %s", url);
+ mDataSourceType = DATA_SOURCE_TYPE_HTTP_LIVE;
} else if (!strncasecmp(url, "rtsp://", 7)) {
source = new RTSPSource(
notify, httpService, url, headers, mUIDValid, mUID);
ALOGV("setDataSourceAsync RTSPSource %s", url);
+ mDataSourceType = DATA_SOURCE_TYPE_RTSP;
} else if ((!strncasecmp(url, "http://", 7)
|| !strncasecmp(url, "https://", 8))
&& ((len >= 4 && !strcasecmp(".sdp", &url[len - 4]))
@@ -269,6 +273,7 @@
source = new RTSPSource(
notify, httpService, url, headers, mUIDValid, mUID, true);
ALOGV("setDataSourceAsync RTSPSource http/https/.sdp %s", url);
+ mDataSourceType = DATA_SOURCE_TYPE_RTSP;
} else {
ALOGV("setDataSourceAsync GenericSource %s", url);
@@ -282,6 +287,9 @@
} else {
ALOGE("Failed to set data source!");
}
+
+ // regardless of success/failure
+ mDataSourceType = DATA_SOURCE_TYPE_GENERIC_URL;
}
msg->setObject("source", source);
msg->post();
@@ -307,6 +315,7 @@
msg->setObject("source", source);
msg->post();
+ mDataSourceType = DATA_SOURCE_TYPE_GENERIC_FD;
}
void NuPlayer::setDataSourceAsync(const sp<DataSource> &dataSource) {
@@ -323,6 +332,7 @@
msg->setObject("source", source);
msg->post();
+ mDataSourceType = DATA_SOURCE_TYPE_MEDIA;
}
status_t NuPlayer::getDefaultBufferingSettings(
@@ -2651,6 +2661,32 @@
}
}
+const char *NuPlayer::getDataSourceType() {
+ switch (mDataSourceType) {
+ case DATA_SOURCE_TYPE_HTTP_LIVE:
+ return "HTTPLive";
+
+ case DATA_SOURCE_TYPE_RTSP:
+ return "RTSP";
+
+ case DATA_SOURCE_TYPE_GENERIC_URL:
+ return "GenURL";
+
+ case DATA_SOURCE_TYPE_GENERIC_FD:
+ return "GenFD";
+
+ case DATA_SOURCE_TYPE_MEDIA:
+ return "Media";
+
+ case DATA_SOURCE_TYPE_STREAM:
+ return "Stream";
+
+ case DATA_SOURCE_TYPE_NONE:
+ default:
+ return "None";
+ }
+ }
+
// Modular DRM begin
status_t NuPlayer::prepareDrm(const uint8_t uuid[16], const Vector<uint8_t> &drmSessionId)
{
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayer.h b/media/libmediaplayerservice/nuplayer/NuPlayer.h
index d542749..c69835f 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayer.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayer.h
@@ -93,6 +93,8 @@
status_t prepareDrm(const uint8_t uuid[16], const Vector<uint8_t> &drmSessionId);
status_t releaseDrm();
+ const char *getDataSourceType();
+
protected:
virtual ~NuPlayer();
@@ -236,6 +238,18 @@
sp<ICrypto> mCrypto;
bool mIsDrmProtected;
+ typedef enum {
+ DATA_SOURCE_TYPE_NONE,
+ DATA_SOURCE_TYPE_HTTP_LIVE,
+ DATA_SOURCE_TYPE_RTSP,
+ DATA_SOURCE_TYPE_GENERIC_URL,
+ DATA_SOURCE_TYPE_GENERIC_FD,
+ DATA_SOURCE_TYPE_MEDIA,
+ DATA_SOURCE_TYPE_STREAM,
+ } DATA_SOURCE_TYPE;
+
+ std::atomic<DATA_SOURCE_TYPE> mDataSourceType;
+
inline const sp<DecoderBase> &getDecoder(bool audio) {
return audio ? mAudioDecoder : mVideoDecoder;
}
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
index 01008b4..0c06976 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDriver.cpp
@@ -53,6 +53,7 @@
static const char *kPlayerPlaying = "android.media.mediaplayer.playingMs";
static const char *kPlayerError = "android.media.mediaplayer.err";
static const char *kPlayerErrorCode = "android.media.mediaplayer.errcode";
+static const char *kPlayerDataSourceType = "android.media.mediaplayer.dataSource";
NuPlayerDriver::NuPlayerDriver(pid_t pid)
@@ -570,6 +571,8 @@
mAnalyticsItem->setInt64(kPlayerDuration, duration_ms);
mAnalyticsItem->setInt64(kPlayerPlaying, (mPlayingTimeUs+500)/1000 );
+
+ mAnalyticsItem->setCString(kPlayerDataSourceType, mPlayer->getDataSourceType());
}
diff --git a/media/libstagefright/NuMediaExtractor.cpp b/media/libstagefright/NuMediaExtractor.cpp
index e3ca516..51f1ba3 100644
--- a/media/libstagefright/NuMediaExtractor.cpp
+++ b/media/libstagefright/NuMediaExtractor.cpp
@@ -297,6 +297,10 @@
sp<IMediaSource> source = mImpl->getTrack(index);
+ if (source == nullptr) {
+ return ERROR_MALFORMED;
+ }
+
status_t ret = source->start();
if (ret != OK) {
return ret;
diff --git a/media/ndk/NdkImage.cpp b/media/ndk/NdkImage.cpp
index 60312da..6d28d1b 100644
--- a/media/ndk/NdkImage.cpp
+++ b/media/ndk/NdkImage.cpp
@@ -31,12 +31,10 @@
#define ALIGN(x, mask) ( ((x) + (mask) - 1) & ~((mask) - 1) )
-AImage::AImage(AImageReader* reader, int32_t format, uint64_t usage0, uint64_t usage1,
- BufferItem* buffer, int64_t timestamp,
- int32_t width, int32_t height, int32_t numPlanes) :
- mReader(reader), mFormat(format), mUsage0(usage0), mUsage1(usage1),
- mBuffer(buffer), mLockedBuffer(nullptr), mTimestamp(timestamp),
- mWidth(width), mHeight(height), mNumPlanes(numPlanes) {
+AImage::AImage(AImageReader* reader, int32_t format, uint64_t usage, BufferItem* buffer,
+ int64_t timestamp, int32_t width, int32_t height, int32_t numPlanes) :
+ mReader(reader), mFormat(format), mUsage(usage), mBuffer(buffer), mLockedBuffer(nullptr),
+ mTimestamp(timestamp), mWidth(width), mHeight(height), mNumPlanes(numPlanes) {
}
// Can only be called by free() with mLock hold
@@ -178,9 +176,9 @@
return AMEDIA_ERROR_INVALID_OBJECT;
}
- if ((mUsage0 & AHARDWAREBUFFER_USAGE_CPU_READ_OFTEN) == 0) {
+ if ((mUsage & AHARDWAREBUFFER_USAGE_CPU_READ_OFTEN) == 0) {
ALOGE("%s: AImage %p does not have any software read usage bits set, usage=%" PRIu64 "",
- __FUNCTION__, this, mUsage0);
+ __FUNCTION__, this, mUsage);
return AMEDIA_IMGREADER_CANNOT_LOCK_IMAGE;
}
@@ -191,13 +189,10 @@
auto lockedBuffer = std::make_unique<CpuConsumer::LockedBuffer>();
- uint64_t producerUsage;
- uint64_t consumerUsage;
- android_hardware_HardwareBuffer_convertToGrallocUsageBits(
- &producerUsage, &consumerUsage, mUsage0, mUsage1);
+ uint64_t grallocUsage = android_hardware_HardwareBuffer_convertToGrallocUsageBits(mUsage);
status_t ret =
- lockImageFromBuffer(mBuffer, consumerUsage, mBuffer->mFence->dup(), lockedBuffer.get());
+ lockImageFromBuffer(mBuffer, grallocUsage, mBuffer->mFence->dup(), lockedBuffer.get());
if (ret != OK) {
ALOGE("%s: AImage %p failed to lock, error=%d", __FUNCTION__, this, ret);
return AMEDIA_IMGREADER_CANNOT_LOCK_IMAGE;
diff --git a/media/ndk/NdkImagePriv.h b/media/ndk/NdkImagePriv.h
index 1fcb495..e9073d5 100644
--- a/media/ndk/NdkImagePriv.h
+++ b/media/ndk/NdkImagePriv.h
@@ -32,9 +32,8 @@
// TODO: this only supports ImageReader
struct AImage {
- AImage(AImageReader* reader, int32_t format, uint64_t usage0, uint64_t usage1,
- BufferItem* buffer, int64_t timestamp,
- int32_t width, int32_t height, int32_t numPlanes);
+ AImage(AImageReader* reader, int32_t format, uint64_t usage, BufferItem* buffer,
+ int64_t timestamp, int32_t width, int32_t height, int32_t numPlanes);
// free all resources while keeping object alive. Caller must obtain reader lock
void close() { close(-1); }
@@ -75,8 +74,7 @@
// When reader is close, AImage will only accept close API call
wp<AImageReader> mReader;
const int32_t mFormat;
- const uint64_t mUsage0; // AHARDWAREBUFFER_USAGE0* flags.
- const uint64_t mUsage1; // AHARDWAREBUFFER_USAGE1* flags.
+ const uint64_t mUsage; // AHARDWAREBUFFER_USAGE_* flags.
BufferItem* mBuffer;
std::unique_ptr<CpuConsumer::LockedBuffer> mLockedBuffer;
const int64_t mTimestamp;
diff --git a/media/ndk/NdkImageReader.cpp b/media/ndk/NdkImageReader.cpp
index 7e41d28..5d1a20b 100644
--- a/media/ndk/NdkImageReader.cpp
+++ b/media/ndk/NdkImageReader.cpp
@@ -238,14 +238,12 @@
AImageReader::AImageReader(int32_t width,
int32_t height,
int32_t format,
- uint64_t usage0,
- uint64_t usage1,
+ uint64_t usage,
int32_t maxImages)
: mWidth(width),
mHeight(height),
mFormat(format),
- mUsage0(usage0),
- mUsage1(usage1),
+ mUsage(usage),
mMaxImages(maxImages),
mNumPlanes(getNumPlanesForFormat(format)),
mFrameListener(new FrameListener(this)),
@@ -256,20 +254,14 @@
PublicFormat publicFormat = static_cast<PublicFormat>(mFormat);
mHalFormat = android_view_Surface_mapPublicFormatToHalFormat(publicFormat);
mHalDataSpace = android_view_Surface_mapPublicFormatToHalDataspace(publicFormat);
-
- uint64_t producerUsage;
- uint64_t consumerUsage;
- android_hardware_HardwareBuffer_convertToGrallocUsageBits(
- &producerUsage, &consumerUsage, mUsage0, mUsage1);
- // Strip out producerUsage here.
- mHalUsage = android_convertGralloc1To0Usage(0, consumerUsage);
+ mHalUsage = android_hardware_HardwareBuffer_convertToGrallocUsageBits(mUsage);
sp<IGraphicBufferProducer> gbProducer;
sp<IGraphicBufferConsumer> gbConsumer;
BufferQueue::createBufferQueue(&gbProducer, &gbConsumer);
- String8 consumerName = String8::format("ImageReader-%dx%df%xu%" PRIu64 "u%" PRIu64 "m%d-%d-%d",
- mWidth, mHeight, mFormat, mUsage0, mUsage1, mMaxImages, getpid(),
+ String8 consumerName = String8::format("ImageReader-%dx%df%xu%" PRIu64 "m%d-%d-%d",
+ mWidth, mHeight, mFormat, mUsage, mMaxImages, getpid(),
createProcessUniqueId());
mBufferItemConsumer =
@@ -445,10 +437,10 @@
}
if (mHalFormat == HAL_PIXEL_FORMAT_BLOB) {
- *image = new AImage(this, mFormat, mUsage0, mUsage1, buffer, buffer->mTimestamp,
+ *image = new AImage(this, mFormat, mUsage, buffer, buffer->mTimestamp,
readerWidth, readerHeight, mNumPlanes);
} else {
- *image = new AImage(this, mFormat, mUsage0, mUsage1, buffer, buffer->mTimestamp,
+ *image = new AImage(this, mFormat, mUsage, buffer, buffer->mTimestamp,
bufferWidth, bufferHeight, mNumPlanes);
}
mAcquiredImages.push_back(*image);
@@ -587,12 +579,12 @@
/*out*/AImageReader** reader) {
ALOGV("%s", __FUNCTION__);
return AImageReader_newWithUsage(
- width, height, format, AHARDWAREBUFFER_USAGE_CPU_READ_OFTEN, 0, maxImages, reader);
+ width, height, format, AHARDWAREBUFFER_USAGE_CPU_READ_OFTEN, maxImages, reader);
}
EXPORT
media_status_t AImageReader_newWithUsage(
- int32_t width, int32_t height, int32_t format, uint64_t usage0, uint64_t usage1,
+ int32_t width, int32_t height, int32_t format, uint64_t usage,
int32_t maxImages, /*out*/ AImageReader** reader) {
ALOGV("%s", __FUNCTION__);
@@ -626,7 +618,7 @@
}
AImageReader* tmpReader = new AImageReader(
- width, height, format, usage0, usage1, maxImages);
+ width, height, format, usage, maxImages);
if (tmpReader == nullptr) {
ALOGE("%s: AImageReader allocation failed", __FUNCTION__);
return AMEDIA_ERROR_UNKNOWN;
diff --git a/media/ndk/NdkImageReaderPriv.h b/media/ndk/NdkImageReaderPriv.h
index a233ec8..35af169 100644
--- a/media/ndk/NdkImageReaderPriv.h
+++ b/media/ndk/NdkImageReaderPriv.h
@@ -55,8 +55,7 @@
AImageReader(int32_t width,
int32_t height,
int32_t format,
- uint64_t usage0,
- uint64_t usage1,
+ uint64_t usage,
int32_t maxImages);
~AImageReader();
@@ -117,8 +116,7 @@
const int32_t mWidth;
const int32_t mHeight;
const int32_t mFormat;
- const uint64_t mUsage0; // AHARDWAREBUFFER_USAGE0* flags.
- const uint64_t mUsage1; // AHARDWAREBUFFER_USAGE1* flags.
+ const uint64_t mUsage; // AHARDWAREBUFFER_USAGE_* flags.
const int32_t mMaxImages;
// TODO(jwcai) Seems completely unused in AImageReader class.
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index fb303bf..2bdb008 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -211,8 +211,7 @@
if (!requestQueueRet.isOk()) {
ALOGE("Transaction error when getting request metadata fmq: %s, not use it",
requestQueueRet.description().c_str());
- queue = nullptr;
- // Don't use the queue onwards.
+ return DEAD_OBJECT;
}
auto resultQueueRet = session->getCaptureResultMetadataQueue(
[&queue = mResultMetadataQueue](const auto& descriptor) {
@@ -226,8 +225,7 @@
if (!resultQueueRet.isOk()) {
ALOGE("Transaction error when getting result metadata queue from camera session: %s",
resultQueueRet.description().c_str());
- mResultMetadataQueue = nullptr;
- // Don't use the queue onwards.
+ return DEAD_OBJECT;
}
mInterface = std::make_unique<HalInterface>(session, queue);
@@ -3013,7 +3011,7 @@
// Unknown template ID
return BAD_VALUE;
}
- mHidlSession->constructDefaultRequestSettings(id,
+ auto err = mHidlSession->constructDefaultRequestSettings(id,
[&status, &requestTemplate]
(common::V1_0::Status s, const device::V3_2::CameraMetadata& request) {
status = s;
@@ -3035,7 +3033,12 @@
}
}
});
- res = CameraProviderManager::mapToStatusT(status);
+ if (!err.isOk()) {
+ ALOGE("%s: Transaction error: %s", __FUNCTION__, err.description().c_str());
+ res = DEAD_OBJECT;
+ } else {
+ res = CameraProviderManager::mapToStatusT(status);
+ }
}
return res;
}
@@ -3109,12 +3112,17 @@
HalStreamConfiguration finalConfiguration;
common::V1_0::Status status;
- mHidlSession->configureStreams(requestedConfiguration,
+ auto err = mHidlSession->configureStreams(requestedConfiguration,
[&status, &finalConfiguration]
(common::V1_0::Status s, const HalStreamConfiguration& halConfiguration) {
finalConfiguration = halConfiguration;
status = s;
});
+ if (!err.isOk()) {
+ ALOGE("%s: Transaction error: %s", __FUNCTION__, err.description().c_str());
+ return DEAD_OBJECT;
+ }
+
if (status != common::V1_0::Status::OK ) {
return CameraProviderManager::mapToStatusT(status);
}
@@ -3300,12 +3308,15 @@
captureRequest->fmqSettingsSize = 0u;
}
}
- mHidlSession->processCaptureRequest(captureRequests, cachesToRemove,
+ auto err = mHidlSession->processCaptureRequest(captureRequests, cachesToRemove,
[&status, &numRequestProcessed] (auto s, uint32_t n) {
status = s;
*numRequestProcessed = n;
});
-
+ if (!err.isOk()) {
+ ALOGE("%s: Transaction error: %s", __FUNCTION__, err.description().c_str());
+ return DEAD_OBJECT;
+ }
if (status == common::V1_0::Status::OK && *numRequestProcessed != batchSize) {
ALOGE("%s: processCaptureRequest returns OK but processed %d/%zu requests",
__FUNCTION__, *numRequestProcessed, batchSize);
@@ -3343,7 +3354,13 @@
if (mHal3Device != nullptr) {
res = mHal3Device->ops->flush(mHal3Device);
} else {
- res = CameraProviderManager::mapToStatusT(mHidlSession->flush());
+ auto err = mHidlSession->flush();
+ if (!err.isOk()) {
+ ALOGE("%s: Transaction error: %s", __FUNCTION__, err.description().c_str());
+ res = DEAD_OBJECT;
+ } else {
+ res = CameraProviderManager::mapToStatusT(err);
+ }
}
return res;
}
@@ -3369,7 +3386,11 @@
if (mHal3Device != nullptr) {
mHal3Device->common.close(&mHal3Device->common);
} else {
- mHidlSession->close();
+ auto err = mHidlSession->close();
+ // Interface will be dead shortly anyway, so don't log errors
+ if (!err.isOk()) {
+ res = DEAD_OBJECT;
+ }
}
return res;
}
diff --git a/services/mediadrm/Android.mk b/services/mediadrm/Android.mk
index 1d5fa07..fa3a02b 100644
--- a/services/mediadrm/Android.mk
+++ b/services/mediadrm/Android.mk
@@ -43,8 +43,9 @@
# TODO: Some legacy DRM plugins only support 32-bit. They need to be migrated to
# 64-bit. (b/18948909) Once all of a device's legacy DRM plugins support 64-bit,
-# that device can turn on ENABLE_MEDIADRM_64 to build this service as 64-bit.
-ifneq ($(ENABLE_MEDIADRM_64), true)
+# that device can turn on TARGET_ENABLE_MEDIADRM_64 to build this service as
+# 64-bit.
+ifneq ($(TARGET_ENABLE_MEDIADRM_64), true)
LOCAL_32_BIT_ONLY := true
endif
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index b197798..d3e182a 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -46,6 +46,7 @@
// Use 2 for "double buffered"
#define BUFFER_SIZE_IN_BURSTS 2
+#define BURSTS_PER_MIX_LOOP 1
// The mStreamInternal will use a service interface that does not go through Binder.
AAudioServiceEndpoint::AAudioServiceEndpoint(AAudioService &audioService)