Merge "MediaMetrics: Add thread-safety checking" into rvc-dev
diff --git a/media/codec2/components/vpx/C2SoftVpxEnc.cpp b/media/codec2/components/vpx/C2SoftVpxEnc.cpp
index ebc7a8f..74e105e 100644
--- a/media/codec2/components/vpx/C2SoftVpxEnc.cpp
+++ b/media/codec2/components/vpx/C2SoftVpxEnc.cpp
@@ -67,8 +67,9 @@
mLastTimestamp(0x7FFFFFFFFFFFFFFFull),
mSignalledOutputEos(false),
mSignalledError(false) {
- memset(mTemporalLayerBitrateRatio, 0, sizeof(mTemporalLayerBitrateRatio));
- mTemporalLayerBitrateRatio[0] = 100;
+ for (int i = 0; i < MAXTEMPORALLAYERS; i++) {
+ mTemporalLayerBitrateRatio[i] = 1.0f;
+ }
}
C2SoftVpxEnc::~C2SoftVpxEnc() {
@@ -123,7 +124,8 @@
mFrameRate = mIntf->getFrameRate_l();
mIntraRefresh = mIntf->getIntraRefresh_l();
mRequestSync = mIntf->getRequestSync_l();
- mTemporalLayers = mIntf->getTemporalLayers_l()->m.layerCount;
+ mLayering = mIntf->getTemporalLayers_l();
+ mTemporalLayers = mLayering->m.layerCount;
}
switch (mBitrateMode->value) {
@@ -225,6 +227,7 @@
mTemporalPattern[5] = kTemporalUpdateGoldenRefAltRef;
mTemporalPattern[6] = kTemporalUpdateLastRefAltRef;
mTemporalPattern[7] = kTemporalUpdateNone;
+ mTemporalLayerBitrateRatio[0] = mLayering->m.bitrateRatios[0];
mTemporalPatternLength = 8;
break;
case 3:
@@ -245,6 +248,8 @@
mTemporalPattern[5] = kTemporalUpdateNone;
mTemporalPattern[6] = kTemporalUpdateGoldenRefAltRef;
mTemporalPattern[7] = kTemporalUpdateNone;
+ mTemporalLayerBitrateRatio[0] = mLayering->m.bitrateRatios[0];
+ mTemporalLayerBitrateRatio[1] = mLayering->m.bitrateRatios[1];
mTemporalPatternLength = 8;
break;
default:
@@ -255,7 +260,7 @@
for (size_t i = 0; i < mCodecConfiguration->ts_number_layers; i++) {
mCodecConfiguration->ts_target_bitrate[i] =
mCodecConfiguration->rc_target_bitrate *
- mTemporalLayerBitrateRatio[i] / 100;
+ mTemporalLayerBitrateRatio[i];
}
if (mIntf->getSyncFramePeriod() >= 0) {
mCodecConfiguration->kf_max_dist = mIntf->getSyncFramePeriod();
diff --git a/media/codec2/components/vpx/C2SoftVpxEnc.h b/media/codec2/components/vpx/C2SoftVpxEnc.h
index 62ccd1b..5e34b8a 100644
--- a/media/codec2/components/vpx/C2SoftVpxEnc.h
+++ b/media/codec2/components/vpx/C2SoftVpxEnc.h
@@ -180,7 +180,7 @@
size_t mTemporalLayers;
// Temporal layer bitrare ratio in percentage
- uint32_t mTemporalLayerBitrateRatio[MAXTEMPORALLAYERS];
+ float_t mTemporalLayerBitrateRatio[MAXTEMPORALLAYERS];
// Temporal pattern type
TemporalPatternType mTemporalPatternType;
@@ -218,6 +218,7 @@
std::shared_ptr<C2StreamBitrateInfo::output> mBitrate;
std::shared_ptr<C2StreamBitrateModeTuning::output> mBitrateMode;
std::shared_ptr<C2StreamRequestSyncFrameTuning::output> mRequestSync;
+ std::shared_ptr<C2StreamTemporalLayeringTuning::output> mLayering;
C2_DO_NOT_COPY(C2SoftVpxEnc);
};
diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 214f888..06f66d3 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -32,19 +32,12 @@
#define RIDICULOUSLY_LARGE_FRAME_SIZE 4096
AudioEndpoint::AudioEndpoint()
- : mUpCommandQueue(nullptr)
- , mDataQueue(nullptr)
- , mFreeRunning(false)
+ : mFreeRunning(false)
, mDataReadCounter(0)
, mDataWriteCounter(0)
{
}
-AudioEndpoint::~AudioEndpoint() {
- delete mDataQueue;
- delete mUpCommandQueue;
-}
-
// TODO Consider moving to a method in RingBufferDescriptor
static aaudio_result_t AudioEndpoint_validateQueueDescriptor(const char *type,
const RingBufferDescriptor *descriptor) {
@@ -144,7 +137,7 @@
return AAUDIO_ERROR_INTERNAL;
}
- mUpCommandQueue = new FifoBuffer(
+ mUpCommandQueue = std::make_unique<FifoBuffer>(
descriptor->bytesPerFrame,
descriptor->capacityInFrames,
descriptor->readCounterAddress,
@@ -173,7 +166,7 @@
? &mDataWriteCounter
: descriptor->writeCounterAddress;
- mDataQueue = new FifoBuffer(
+ mDataQueue = std::make_unique<FifoBuffer>(
descriptor->bytesPerFrame,
descriptor->capacityInFrames,
readCounterAddress,
@@ -194,18 +187,15 @@
return mDataQueue->getEmptyRoomAvailable(wrappingBuffer);
}
-int32_t AudioEndpoint::getEmptyFramesAvailable()
-{
+int32_t AudioEndpoint::getEmptyFramesAvailable() {
return mDataQueue->getEmptyFramesAvailable();
}
-int32_t AudioEndpoint::getFullFramesAvailable(WrappingBuffer *wrappingBuffer)
-{
+int32_t AudioEndpoint::getFullFramesAvailable(WrappingBuffer *wrappingBuffer) {
return mDataQueue->getFullDataAvailable(wrappingBuffer);
}
-int32_t AudioEndpoint::getFullFramesAvailable()
-{
+int32_t AudioEndpoint::getFullFramesAvailable() {
return mDataQueue->getFullFramesAvailable();
}
@@ -217,29 +207,24 @@
mDataQueue->advanceReadIndex(deltaFrames);
}
-void AudioEndpoint::setDataReadCounter(fifo_counter_t framesRead)
-{
+void AudioEndpoint::setDataReadCounter(fifo_counter_t framesRead) {
mDataQueue->setReadCounter(framesRead);
}
-fifo_counter_t AudioEndpoint::getDataReadCounter()
-{
+fifo_counter_t AudioEndpoint::getDataReadCounter() const {
return mDataQueue->getReadCounter();
}
-void AudioEndpoint::setDataWriteCounter(fifo_counter_t framesRead)
-{
+void AudioEndpoint::setDataWriteCounter(fifo_counter_t framesRead) {
mDataQueue->setWriteCounter(framesRead);
}
-fifo_counter_t AudioEndpoint::getDataWriteCounter()
-{
+fifo_counter_t AudioEndpoint::getDataWriteCounter() const {
return mDataQueue->getWriteCounter();
}
int32_t AudioEndpoint::setBufferSizeInFrames(int32_t requestedFrames,
- int32_t *actualFrames)
-{
+ int32_t *actualFrames) {
if (requestedFrames < ENDPOINT_DATA_QUEUE_SIZE_MIN) {
requestedFrames = ENDPOINT_DATA_QUEUE_SIZE_MIN;
}
@@ -248,19 +233,17 @@
return AAUDIO_OK;
}
-int32_t AudioEndpoint::getBufferSizeInFrames() const
-{
+int32_t AudioEndpoint::getBufferSizeInFrames() const {
return mDataQueue->getThreshold();
}
-int32_t AudioEndpoint::getBufferCapacityInFrames() const
-{
+int32_t AudioEndpoint::getBufferCapacityInFrames() const {
return (int32_t)mDataQueue->getBufferCapacityInFrames();
}
void AudioEndpoint::dump() const {
- ALOGD("data readCounter = %lld", (long long) mDataQueue->getReadCounter());
- ALOGD("data writeCounter = %lld", (long long) mDataQueue->getWriteCounter());
+ ALOGD("data readCounter = %lld", (long long) getDataReadCounter());
+ ALOGD("data writeCounter = %lld", (long long) getDataWriteCounter());
}
void AudioEndpoint::eraseDataMemory() {
diff --git a/media/libaaudio/src/client/AudioEndpoint.h b/media/libaaudio/src/client/AudioEndpoint.h
index f5b67e8..484d917 100644
--- a/media/libaaudio/src/client/AudioEndpoint.h
+++ b/media/libaaudio/src/client/AudioEndpoint.h
@@ -35,7 +35,6 @@
public:
AudioEndpoint();
- virtual ~AudioEndpoint();
/**
* Configure based on the EndPointDescriptor_t.
@@ -67,11 +66,11 @@
*/
void setDataReadCounter(android::fifo_counter_t framesRead);
- android::fifo_counter_t getDataReadCounter();
+ android::fifo_counter_t getDataReadCounter() const;
void setDataWriteCounter(android::fifo_counter_t framesWritten);
- android::fifo_counter_t getDataWriteCounter();
+ android::fifo_counter_t getDataWriteCounter() const;
/**
* The result is not valid until after configure() is called.
@@ -94,8 +93,8 @@
void dump() const;
private:
- android::FifoBuffer *mUpCommandQueue;
- android::FifoBuffer *mDataQueue;
+ std::unique_ptr<android::FifoBuffer> mUpCommandQueue;
+ std::unique_ptr<android::FifoBuffer> mDataQueue;
bool mFreeRunning;
android::fifo_counter_t mDataReadCounter; // only used if free-running
android::fifo_counter_t mDataWriteCounter; // only used if free-running
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 6723ec9..f89cde7 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -58,7 +58,6 @@
AudioStreamInternal::AudioStreamInternal(AAudioServiceInterface &serviceInterface, bool inService)
: AudioStream()
, mClockModel()
- , mAudioEndpoint()
, mServiceStreamHandle(AAUDIO_HANDLE_INVALID)
, mInService(inService)
, mServiceInterface(serviceInterface)
@@ -74,7 +73,6 @@
aaudio_result_t AudioStreamInternal::open(const AudioStreamBuilder &builder) {
aaudio_result_t result = AAUDIO_OK;
- int32_t capacity;
int32_t framesPerBurst;
int32_t framesPerHardwareBurst;
AAudioStreamRequest request;
@@ -173,7 +171,8 @@
}
// Configure endpoint based on descriptor.
- result = mAudioEndpoint.configure(&mEndpointDescriptor, getDirection());
+ mAudioEndpoint = std::make_unique<AudioEndpoint>();
+ result = mAudioEndpoint->configure(&mEndpointDescriptor, getDirection());
if (result != AAUDIO_OK) {
goto error;
}
@@ -201,9 +200,10 @@
}
mFramesPerBurst = framesPerBurst; // only save good value
- capacity = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
- if (capacity < mFramesPerBurst || capacity > MAX_BUFFER_CAPACITY_IN_FRAMES) {
- ALOGE("%s - bufferCapacity out of range = %d", __func__, capacity);
+ mBufferCapacityInFrames = mEndpointDescriptor.dataQueueDescriptor.capacityInFrames;
+ if (mBufferCapacityInFrames < mFramesPerBurst
+ || mBufferCapacityInFrames > MAX_BUFFER_CAPACITY_IN_FRAMES) {
+ ALOGE("%s - bufferCapacity out of range = %d", __func__, mBufferCapacityInFrames);
result = AAUDIO_ERROR_OUT_OF_RANGE;
goto error;
}
@@ -230,7 +230,7 @@
}
const int32_t callbackBufferSize = mCallbackFrames * getBytesPerFrame();
- mCallbackBuffer = new uint8_t[callbackBufferSize];
+ mCallbackBuffer = std::make_unique<uint8_t[]>(callbackBufferSize);
}
// For debugging and analyzing the distribution of MMAP timestamps.
@@ -239,7 +239,7 @@
// You can use this offset to reduce glitching.
// You can also use this offset to force glitching. By iterating over multiple
// values you can reveal the distribution of the hardware timing jitter.
- if (mAudioEndpoint.isFreeRunning()) { // MMAP?
+ if (mAudioEndpoint->isFreeRunning()) { // MMAP?
int32_t offsetMicros = (getDirection() == AAUDIO_DIRECTION_OUTPUT)
? AAudioProperty_getOutputMMapOffsetMicros()
: AAudioProperty_getInputMMapOffsetMicros();
@@ -251,7 +251,7 @@
mTimeOffsetNanos = offsetMicros * AAUDIO_NANOS_PER_MICROSECOND;
}
- setBufferSize(capacity / 2); // Default buffer size to match Q
+ setBufferSize(mBufferCapacityInFrames / 2); // Default buffer size to match Q
setState(AAUDIO_STREAM_STATE_OPEN);
@@ -279,8 +279,12 @@
mServiceStreamHandle = AAUDIO_HANDLE_INVALID;
mServiceInterface.closeStream(serviceStreamHandle);
- delete[] mCallbackBuffer;
- mCallbackBuffer = nullptr;
+ mCallbackBuffer.reset();
+
+ // Update local frame counters so we can query them after releasing the endpoint.
+ getFramesRead();
+ getFramesWritten();
+ mAudioEndpoint.reset();
result = mEndPointParcelable.close();
aaudio_result_t result2 = AudioStream::release_l();
return (result != AAUDIO_OK) ? result : result2;
@@ -539,7 +543,7 @@
case AAUDIO_SERVICE_EVENT_DISCONNECTED:
// Prevent hardware from looping on old data and making buzzing sounds.
if (getDirection() == AAUDIO_DIRECTION_OUTPUT) {
- mAudioEndpoint.eraseDataMemory();
+ mAudioEndpoint->eraseDataMemory();
}
result = AAUDIO_ERROR_DISCONNECTED;
setState(AAUDIO_STREAM_STATE_DISCONNECTED);
@@ -565,7 +569,10 @@
while (result == AAUDIO_OK) {
AAudioServiceMessage message;
- if (mAudioEndpoint.readUpCommand(&message) != 1) {
+ if (!mAudioEndpoint) {
+ break;
+ }
+ if (mAudioEndpoint->readUpCommand(&message) != 1) {
break; // no command this time, no problem
}
switch (message.what) {
@@ -593,7 +600,10 @@
while (result == AAUDIO_OK) {
AAudioServiceMessage message;
- if (mAudioEndpoint.readUpCommand(&message) != 1) {
+ if (!mAudioEndpoint) {
+ break;
+ }
+ if (mAudioEndpoint->readUpCommand(&message) != 1) {
break; // no command this time, no problem
}
switch (message.what) {
@@ -626,7 +636,7 @@
const char * fifoName = "aaRdy";
ATRACE_BEGIN(traceName);
if (ATRACE_ENABLED()) {
- int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();
+ int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
ATRACE_INT(fifoName, fullFrames);
}
@@ -655,7 +665,7 @@
if (timeoutNanoseconds == 0) {
break; // don't block
} else if (wakeTimeNanos != 0) {
- if (!mAudioEndpoint.isFreeRunning()) {
+ if (!mAudioEndpoint->isFreeRunning()) {
// If there is software on the other end of the FIFO then it may get delayed.
// So wake up just a little after we expect it to be ready.
wakeTimeNanos += mWakeupDelayNanos;
@@ -680,12 +690,12 @@
ALOGW("processData(): past deadline by %d micros",
(int)((wakeTimeNanos - deadlineNanos) / AAUDIO_NANOS_PER_MICROSECOND));
mClockModel.dump();
- mAudioEndpoint.dump();
+ mAudioEndpoint->dump();
break;
}
if (ATRACE_ENABLED()) {
- int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();
+ int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
ATRACE_INT(fifoName, fullFrames);
int64_t sleepForNanos = wakeTimeNanos - currentTimeNanos;
ATRACE_INT("aaSlpNs", (int32_t)sleepForNanos);
@@ -697,7 +707,7 @@
}
if (ATRACE_ENABLED()) {
- int32_t fullFrames = mAudioEndpoint.getFullFramesAvailable();
+ int32_t fullFrames = mAudioEndpoint->getFullFramesAvailable();
ATRACE_INT(fifoName, fullFrames);
}
@@ -731,11 +741,15 @@
adjustedFrames = std::min(maximumSize, adjustedFrames);
}
- // Clip against the actual size from the endpoint.
- int32_t actualFrames = 0;
- mAudioEndpoint.setBufferSizeInFrames(maximumSize, &actualFrames);
- // actualFrames should be <= actual maximum size of endpoint
- adjustedFrames = std::min(actualFrames, adjustedFrames);
+ if (mAudioEndpoint) {
+ // Clip against the actual size from the endpoint.
+ int32_t actualFrames = 0;
+ // Set to maximum size so we can write extra data when ready in order to reduce glitches.
+ // The amount we keep in the buffer is controlled by mBufferSizeInFrames.
+ mAudioEndpoint->setBufferSizeInFrames(maximumSize, &actualFrames);
+ // actualFrames should be <= actual maximum size of endpoint
+ adjustedFrames = std::min(actualFrames, adjustedFrames);
+ }
mBufferSizeInFrames = adjustedFrames;
ALOGV("%s(%d) returns %d", __func__, requestedFrames, adjustedFrames);
@@ -747,7 +761,7 @@
}
int32_t AudioStreamInternal::getBufferCapacity() const {
- return mAudioEndpoint.getBufferCapacityInFrames();
+ return mBufferCapacityInFrames;
}
int32_t AudioStreamInternal::getFramesPerBurst() const {
@@ -760,5 +774,5 @@
}
bool AudioStreamInternal::isClockModelInControl() const {
- return isActive() && mAudioEndpoint.isFreeRunning() && mClockModel.isRunning();
+ return isActive() && mAudioEndpoint->isFreeRunning() && mClockModel.isRunning();
}
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 095f30c..61591b3 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -155,7 +155,8 @@
IsochronousClockModel mClockModel; // timing model for chasing the HAL
- AudioEndpoint mAudioEndpoint; // source for reads or sink for writes
+ std::unique_ptr<AudioEndpoint> mAudioEndpoint; // source for reads or sink for writes
+
aaudio_handle_t mServiceStreamHandle; // opaque handle returned from service
int32_t mFramesPerBurst = MIN_FRAMES_PER_BURST; // frames per HAL transfer
@@ -164,7 +165,7 @@
// Offset from underlying frame position.
int64_t mFramesOffsetFromService = 0; // offset for timestamps
- uint8_t *mCallbackBuffer = nullptr;
+ std::unique_ptr<uint8_t[]> mCallbackBuffer;
int32_t mCallbackFrames = 0;
// The service uses this for SHARED mode.
@@ -178,6 +179,9 @@
float mStreamVolume = 1.0f;
+ int64_t mLastFramesWritten = 0;
+ int64_t mLastFramesRead = 0;
+
private:
/*
* Asynchronous write with data conversion.
@@ -207,6 +211,8 @@
int32_t mDeviceChannelCount = 0;
int32_t mBufferSizeInFrames = 0; // local threshold to control latency
+ int32_t mBufferCapacityInFrames = 0;
+
};
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 9684ee4..9fa2e40 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -42,8 +42,8 @@
AudioStreamInternalCapture::~AudioStreamInternalCapture() {}
void AudioStreamInternalCapture::advanceClientToMatchServerPosition() {
- int64_t readCounter = mAudioEndpoint.getDataReadCounter();
- int64_t writeCounter = mAudioEndpoint.getDataWriteCounter();
+ int64_t readCounter = mAudioEndpoint->getDataReadCounter();
+ int64_t writeCounter = mAudioEndpoint->getDataWriteCounter();
// Bump offset so caller does not see the retrograde motion in getFramesRead().
int64_t offset = readCounter - writeCounter;
@@ -53,7 +53,7 @@
// Force readCounter to match writeCounter.
// This is because we cannot change the write counter in the hardware.
- mAudioEndpoint.setDataReadCounter(writeCounter);
+ mAudioEndpoint->setDataReadCounter(writeCounter);
}
// Write the data, block if needed and timeoutMillis > 0
@@ -86,7 +86,7 @@
}
// If we have gotten this far then we have at least one timestamp from server.
- if (mAudioEndpoint.isFreeRunning()) {
+ if (mAudioEndpoint->isFreeRunning()) {
//ALOGD("AudioStreamInternalCapture::processDataNow() - update remote counter");
// Update data queue based on the timing model.
// Jitter in the DSP can cause late writes to the FIFO.
@@ -95,7 +95,7 @@
// that the DSP could have written the data.
int64_t estimatedRemoteCounter = mClockModel.convertLatestTimeToPosition(currentNanoTime);
// TODO refactor, maybe use setRemoteCounter()
- mAudioEndpoint.setDataWriteCounter(estimatedRemoteCounter);
+ mAudioEndpoint->setDataWriteCounter(estimatedRemoteCounter);
}
// This code assumes that we have already received valid timestamps.
@@ -108,8 +108,8 @@
// If the capture buffer is full beyond capacity then consider it an overrun.
// For shared streams, the xRunCount is passed up from the service.
- if (mAudioEndpoint.isFreeRunning()
- && mAudioEndpoint.getFullFramesAvailable() > mAudioEndpoint.getBufferCapacityInFrames()) {
+ if (mAudioEndpoint->isFreeRunning()
+ && mAudioEndpoint->getFullFramesAvailable() > mAudioEndpoint->getBufferCapacityInFrames()) {
mXRunCount++;
if (ATRACE_ENABLED()) {
ATRACE_INT("aaOverRuns", mXRunCount);
@@ -143,7 +143,7 @@
// Calculate frame position based off of the readCounter because
// the writeCounter might have just advanced in the background,
// causing us to sleep until a later burst.
- int64_t nextPosition = mAudioEndpoint.getDataReadCounter() + mFramesPerBurst;
+ int64_t nextPosition = mAudioEndpoint->getDataReadCounter() + mFramesPerBurst;
wakeTime = mClockModel.convertPositionToLatestTime(nextPosition);
}
break;
@@ -166,7 +166,7 @@
uint8_t *destination = (uint8_t *) buffer;
int32_t framesLeft = numFrames;
- mAudioEndpoint.getFullFramesAvailable(&wrappingBuffer);
+ mAudioEndpoint->getFullFramesAvailable(&wrappingBuffer);
// Read data in one or two parts.
for (int partIndex = 0; framesLeft > 0 && partIndex < WrappingBuffer::SIZE; partIndex++) {
@@ -208,26 +208,29 @@
}
int32_t framesProcessed = numFrames - framesLeft;
- mAudioEndpoint.advanceReadIndex(framesProcessed);
+ mAudioEndpoint->advanceReadIndex(framesProcessed);
//ALOGD("readNowWithConversion() returns %d", framesProcessed);
return framesProcessed;
}
int64_t AudioStreamInternalCapture::getFramesWritten() {
- const int64_t framesWrittenHardware = isClockModelInControl()
- ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
- : mAudioEndpoint.getDataWriteCounter();
- // Add service offset and prevent retrograde motion.
- mLastFramesWritten = std::max(mLastFramesWritten,
- framesWrittenHardware + mFramesOffsetFromService);
+ if (mAudioEndpoint) {
+ const int64_t framesWrittenHardware = isClockModelInControl()
+ ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
+ : mAudioEndpoint->getDataWriteCounter();
+ // Add service offset and prevent retrograde motion.
+ mLastFramesWritten = std::max(mLastFramesWritten,
+ framesWrittenHardware + mFramesOffsetFromService);
+ }
return mLastFramesWritten;
}
int64_t AudioStreamInternalCapture::getFramesRead() {
- int64_t frames = mAudioEndpoint.getDataReadCounter() + mFramesOffsetFromService;
- //ALOGD("getFramesRead() returns %lld", (long long)frames);
- return frames;
+ if (mAudioEndpoint) {
+ mLastFramesRead = mAudioEndpoint->getDataReadCounter() + mFramesOffsetFromService;
+ }
+ return mLastFramesRead;
}
// Read data from the stream and pass it to the callback for processing.
@@ -243,7 +246,7 @@
int64_t timeoutNanos = calculateReasonableTimeout(mCallbackFrames);
// This is a BLOCKING READ!
- result = read(mCallbackBuffer, mCallbackFrames, timeoutNanos);
+ result = read(mCallbackBuffer.get(), mCallbackFrames, timeoutNanos);
if ((result != mCallbackFrames)) {
ALOGE("callbackLoop: read() returned %d", result);
if (result >= 0) {
@@ -255,7 +258,7 @@
}
// Call application using the AAudio callback interface.
- callbackResult = maybeCallDataCallback(mCallbackBuffer, mCallbackFrames);
+ callbackResult = maybeCallDataCallback(mCallbackBuffer.get(), mCallbackFrames);
if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
ALOGD("%s(): callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.h b/media/libaaudio/src/client/AudioStreamInternalCapture.h
index 294dbaf..6436a53 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.h
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.h
@@ -68,8 +68,6 @@
* @return frames written or negative error
*/
aaudio_result_t readNowWithConversion(void *buffer, int32_t numFrames);
-
- int64_t mLastFramesWritten = 0; // used to prevent retrograde motion
};
} /* namespace aaudio */
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 536009a..1303daf 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -87,8 +87,8 @@
}
void AudioStreamInternalPlay::advanceClientToMatchServerPosition() {
- int64_t readCounter = mAudioEndpoint.getDataReadCounter();
- int64_t writeCounter = mAudioEndpoint.getDataWriteCounter();
+ int64_t readCounter = mAudioEndpoint->getDataReadCounter();
+ int64_t writeCounter = mAudioEndpoint->getDataWriteCounter();
// Bump offset so caller does not see the retrograde motion in getFramesRead().
int64_t offset = writeCounter - readCounter;
@@ -98,7 +98,7 @@
// Force writeCounter to match readCounter.
// This is because we cannot change the read counter in the hardware.
- mAudioEndpoint.setDataWriteCounter(readCounter);
+ mAudioEndpoint->setDataWriteCounter(readCounter);
}
void AudioStreamInternalPlay::onFlushFromServer() {
@@ -135,11 +135,11 @@
// If we have gotten this far then we have at least one timestamp from server.
// If a DMA channel or DSP is reading the other end then we have to update the readCounter.
- if (mAudioEndpoint.isFreeRunning()) {
+ if (mAudioEndpoint->isFreeRunning()) {
// Update data queue based on the timing model.
int64_t estimatedReadCounter = mClockModel.convertTimeToPosition(currentNanoTime);
// ALOGD("AudioStreamInternal::processDataNow() - estimatedReadCounter = %d", (int)estimatedReadCounter);
- mAudioEndpoint.setDataReadCounter(estimatedReadCounter);
+ mAudioEndpoint->setDataReadCounter(estimatedReadCounter);
}
if (mNeedCatchUp.isRequested()) {
@@ -151,7 +151,7 @@
// If the read index passed the write index then consider it an underrun.
// For shared streams, the xRunCount is passed up from the service.
- if (mAudioEndpoint.isFreeRunning() && mAudioEndpoint.getFullFramesAvailable() < 0) {
+ if (mAudioEndpoint->isFreeRunning() && mAudioEndpoint->getFullFramesAvailable() < 0) {
mXRunCount++;
if (ATRACE_ENABLED()) {
ATRACE_INT("aaUnderRuns", mXRunCount);
@@ -170,7 +170,7 @@
// Sleep if there is too much data in the buffer.
// Calculate an ideal time to wake up.
if (wakeTimePtr != nullptr
- && (mAudioEndpoint.getFullFramesAvailable() >= getBufferSize())) {
+ && (mAudioEndpoint->getFullFramesAvailable() >= getBufferSize())) {
// By default wake up a few milliseconds from now. // TODO review
int64_t wakeTime = currentNanoTime + (1 * AAUDIO_NANOS_PER_MILLISECOND);
aaudio_stream_state_t state = getState();
@@ -188,7 +188,7 @@
{
// Sleep until the readCounter catches up and we only have
// the getBufferSize() frames of data sitting in the buffer.
- int64_t nextReadPosition = mAudioEndpoint.getDataWriteCounter() - getBufferSize();
+ int64_t nextReadPosition = mAudioEndpoint->getDataWriteCounter() - getBufferSize();
wakeTime = mClockModel.convertPositionToTime(nextReadPosition);
}
break;
@@ -210,7 +210,7 @@
uint8_t *byteBuffer = (uint8_t *) buffer;
int32_t framesLeft = numFrames;
- mAudioEndpoint.getEmptyFramesAvailable(&wrappingBuffer);
+ mAudioEndpoint->getEmptyFramesAvailable(&wrappingBuffer);
// Write data in one or two parts.
int partIndex = 0;
@@ -236,24 +236,28 @@
partIndex++;
}
int32_t framesWritten = numFrames - framesLeft;
- mAudioEndpoint.advanceWriteIndex(framesWritten);
+ mAudioEndpoint->advanceWriteIndex(framesWritten);
return framesWritten;
}
int64_t AudioStreamInternalPlay::getFramesRead() {
- const int64_t framesReadHardware = isClockModelInControl()
- ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
- : mAudioEndpoint.getDataReadCounter();
- // Add service offset and prevent retrograde motion.
- mLastFramesRead = std::max(mLastFramesRead, framesReadHardware + mFramesOffsetFromService);
+ if (mAudioEndpoint) {
+ const int64_t framesReadHardware = isClockModelInControl()
+ ? mClockModel.convertTimeToPosition(AudioClock::getNanoseconds())
+ : mAudioEndpoint->getDataReadCounter();
+ // Add service offset and prevent retrograde motion.
+ mLastFramesRead = std::max(mLastFramesRead, framesReadHardware + mFramesOffsetFromService);
+ }
return mLastFramesRead;
}
int64_t AudioStreamInternalPlay::getFramesWritten() {
- const int64_t framesWritten = mAudioEndpoint.getDataWriteCounter()
- + mFramesOffsetFromService;
- return framesWritten;
+ if (mAudioEndpoint) {
+ mLastFramesWritten = mAudioEndpoint->getDataWriteCounter()
+ + mFramesOffsetFromService;
+ }
+ return mLastFramesWritten;
}
@@ -268,11 +272,11 @@
// result might be a frame count
while (mCallbackEnabled.load() && isActive() && (result >= 0)) {
// Call application using the AAudio callback interface.
- callbackResult = maybeCallDataCallback(mCallbackBuffer, mCallbackFrames);
+ callbackResult = maybeCallDataCallback(mCallbackBuffer.get(), mCallbackFrames);
if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
// Write audio data to stream. This is a BLOCKING WRITE!
- result = write(mCallbackBuffer, mCallbackFrames, timeoutNanos);
+ result = write(mCallbackBuffer.get(), mCallbackFrames, timeoutNanos);
if ((result != mCallbackFrames)) {
if (result >= 0) {
// Only wrote some of the frames requested. Must have timed out.
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.h b/media/libaaudio/src/client/AudioStreamInternalPlay.h
index cab2942..2e93157 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.h
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.h
@@ -92,8 +92,6 @@
aaudio_result_t writeNowWithConversion(const void *buffer,
int32_t numFrames);
- int64_t mLastFramesRead = 0; // used to prevent retrograde motion
-
AAudioFlowGraph mFlowGraph;
};
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index f51db70..4252dfd 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -111,6 +111,33 @@
return AAUDIO_ERROR_INVALID_STATE;
}
+ switch (getState()) {
+ // Is this a good time to start?
+ case AAUDIO_STREAM_STATE_OPEN:
+ case AAUDIO_STREAM_STATE_PAUSING:
+ case AAUDIO_STREAM_STATE_PAUSED:
+ case AAUDIO_STREAM_STATE_STOPPING:
+ case AAUDIO_STREAM_STATE_STOPPED:
+ case AAUDIO_STREAM_STATE_FLUSHED:
+ break; // Proceed with starting.
+
+ // Already started?
+ case AAUDIO_STREAM_STATE_STARTING:
+ case AAUDIO_STREAM_STATE_STARTED:
+ ALOGW("%s() stream was already started, state = %s", __func__,
+ AudioGlobal_convertStreamStateToText(getState()));
+ return AAUDIO_ERROR_INVALID_STATE;
+
+ // Don't start when the stream is dead!
+ case AAUDIO_STREAM_STATE_DISCONNECTED:
+ case AAUDIO_STREAM_STATE_CLOSING:
+ case AAUDIO_STREAM_STATE_CLOSED:
+ default:
+ ALOGW("%s() stream is dead, state = %s", __func__,
+ AudioGlobal_convertStreamStateToText(getState()));
+ return AAUDIO_ERROR_INVALID_STATE;
+ }
+
aaudio_result_t result = requestStart();
if (result == AAUDIO_OK) {
// We only call this for logging in "dumpsys audio". So ignore return code.
@@ -156,8 +183,8 @@
case AAUDIO_STREAM_STATE_CLOSING:
case AAUDIO_STREAM_STATE_CLOSED:
default:
- ALOGW("safePause() stream not running, state = %s",
- AudioGlobal_convertStreamStateToText(getState()));
+ ALOGW("%s() stream not running, state = %s",
+ __func__, AudioGlobal_convertStreamStateToText(getState()));
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -268,6 +295,11 @@
if (mState == AAUDIO_STREAM_STATE_CLOSED) {
ALOGE("%s(%d) tried to set to %d but already CLOSED", __func__, getId(), state);
+ // Once CLOSING, we can only move to CLOSED state.
+ } else if (mState == AAUDIO_STREAM_STATE_CLOSING
+ && state != AAUDIO_STREAM_STATE_CLOSED) {
+ ALOGE("%s(%d) tried to set to %d but already CLOSING", __func__, getId(), state);
+
// Once DISCONNECTED, we can only move to CLOSING or CLOSED state.
} else if (mState == AAUDIO_STREAM_STATE_DISCONNECTED
&& !(state == AAUDIO_STREAM_STATE_CLOSING
diff --git a/media/libaaudio/tests/test_various.cpp b/media/libaaudio/tests/test_various.cpp
index 5bb1046..1c26615 100644
--- a/media/libaaudio/tests/test_various.cpp
+++ b/media/libaaudio/tests/test_various.cpp
@@ -28,15 +28,20 @@
// Callback function that does nothing.
aaudio_data_callback_result_t NoopDataCallbackProc(
- AAudioStream *stream,
- void *userData,
+ AAudioStream * stream,
+ void * /* userData */,
void *audioData,
int32_t numFrames
) {
- (void) stream;
- (void) userData;
- (void) audioData;
- (void) numFrames;
+ int channels = AAudioStream_getChannelCount(stream);
+ int numSamples = channels * numFrames;
+ bool allZeros = true;
+ float * const floatData = reinterpret_cast<float *>(audioData);
+ for (int i = 0; i < numSamples; i++) {
+ allZeros &= (floatData[i] == 0.0f);
+ floatData[i] = 0.0f;
+ }
+ EXPECT_TRUE(allZeros);
return AAUDIO_CALLBACK_RESULT_CONTINUE;
}
@@ -56,6 +61,7 @@
nullptr);
AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
AAudioStreamBuilder_setSharingMode(aaudioBuilder, sharingMode);
+ AAudioStreamBuilder_setFormat(aaudioBuilder, AAUDIO_FORMAT_PCM_FLOAT);
// Create an AAudioStream using the Builder.
ASSERT_EQ(AAUDIO_OK,
@@ -69,12 +75,33 @@
EXPECT_EQ(AAUDIO_OK, AAudioStream_requestStop(aaudioStream));
EXPECT_EQ(AAUDIO_OK, AAudioStream_release(aaudioStream));
- aaudio_stream_state_t state = AAudioStream_getState(aaudioStream);
- EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, state);
+ EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
// We should be able to call this again without crashing.
EXPECT_EQ(AAUDIO_OK, AAudioStream_release(aaudioStream));
- state = AAudioStream_getState(aaudioStream);
+ EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
+
+ // We expect these not to crash.
+ AAudioStream_setBufferSizeInFrames(aaudioStream, 0);
+ AAudioStream_setBufferSizeInFrames(aaudioStream, 99999999);
+
+ // We should NOT be able to start or change a stream after it has been released.
+ EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, AAudioStream_requestStart(aaudioStream));
+ EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
+ EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, AAudioStream_requestPause(aaudioStream));
+ EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
+ EXPECT_EQ(AAUDIO_ERROR_INVALID_STATE, AAudioStream_requestStop(aaudioStream));
+ EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, AAudioStream_getState(aaudioStream));
+
+ // Does this crash?
+ EXPECT_LT(0, AAudioStream_getFramesRead(aaudioStream));
+ EXPECT_LT(0, AAudioStream_getFramesWritten(aaudioStream));
+
+ // Verify Closing State. Does this crash?
+ aaudio_stream_state_t state = AAUDIO_STREAM_STATE_UNKNOWN;
+ EXPECT_EQ(AAUDIO_OK, AAudioStream_waitForStateChange(aaudioStream,
+ AAUDIO_STREAM_STATE_UNKNOWN, &state,
+ 500 * NANOS_PER_MILLISECOND));
EXPECT_EQ(AAUDIO_STREAM_STATE_CLOSING, state);
EXPECT_EQ(AAUDIO_OK, AAudioStream_close(aaudioStream));
@@ -114,6 +141,7 @@
// Request stream properties.
AAudioStreamBuilder_setDataCallback(aaudioBuilder, NoopDataCallbackProc, nullptr);
AAudioStreamBuilder_setPerformanceMode(aaudioBuilder, perfMode);
+ AAudioStreamBuilder_setFormat(aaudioBuilder, AAUDIO_FORMAT_PCM_FLOAT);
// Create an AAudioStream using the Builder.
ASSERT_EQ(AAUDIO_OK, AAudioStreamBuilder_openStream(aaudioBuilder, &aaudioStream));
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 4762b63..0d20f20 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -156,6 +156,7 @@
aidl_interface {
name: "capture_state_listener-aidl",
+ unstable: true,
local_include_dir: "aidl",
srcs: [
"aidl/android/media/ICaptureStateListener.aidl",
diff --git a/media/libeffects/config/Android.bp b/media/libeffects/config/Android.bp
index 8476f82..8493e30 100644
--- a/media/libeffects/config/Android.bp
+++ b/media/libeffects/config/Android.bp
@@ -15,6 +15,7 @@
"libtinyxml2",
"libutils",
"libmedia_helper",
+ "libcutils",
],
header_libs: ["libaudio_system_headers"],
diff --git a/media/libeffects/config/include/media/EffectsConfig.h b/media/libeffects/config/include/media/EffectsConfig.h
index ef10e0d..57d4dd7 100644
--- a/media/libeffects/config/include/media/EffectsConfig.h
+++ b/media/libeffects/config/include/media/EffectsConfig.h
@@ -35,11 +35,6 @@
/** Default path of effect configuration file. Relative to DEFAULT_LOCATIONS. */
constexpr const char* DEFAULT_NAME = "audio_effects.xml";
-/** Default path of effect configuration file.
- * The /vendor partition is the recommended one, the others are deprecated.
- */
-constexpr const char* DEFAULT_LOCATIONS[] = {"/odm/etc", "/vendor/etc", "/system/etc"};
-
/** Directories where the effect libraries will be search for. */
constexpr const char* LD_EFFECT_LIBRARY_PATH[] =
#ifdef __LP64__
diff --git a/media/libeffects/config/src/EffectsConfig.cpp b/media/libeffects/config/src/EffectsConfig.cpp
index 85fbf11..26eaaf8 100644
--- a/media/libeffects/config/src/EffectsConfig.cpp
+++ b/media/libeffects/config/src/EffectsConfig.cpp
@@ -27,6 +27,7 @@
#include <media/EffectsConfig.h>
#include <media/TypeConverter.h>
+#include <system/audio_config.h>
using namespace tinyxml2;
@@ -338,7 +339,7 @@
return parseWithPath(path);
}
- for (const std::string& location : DEFAULT_LOCATIONS) {
+ for (const std::string& location : audio_get_configuration_paths()) {
std::string defaultPath = location + '/' + DEFAULT_NAME;
if (access(defaultPath.c_str(), R_OK) != 0) {
continue;
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 1df7c88..62a86e7 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -33,6 +33,7 @@
aidl_interface {
name: "resourcemanager_aidl_interface",
+ unstable: true,
local_include_dir: "aidl",
srcs: [
"aidl/android/media/IResourceManagerClient.aidl",
diff --git a/media/libmediatranscoding/Android.bp b/media/libmediatranscoding/Android.bp
index 7468426..f948bd8 100644
--- a/media/libmediatranscoding/Android.bp
+++ b/media/libmediatranscoding/Android.bp
@@ -17,6 +17,7 @@
// AIDL interfaces of MediaTranscoding.
aidl_interface {
name: "mediatranscoding_aidl_interface",
+ unstable: true,
local_include_dir: "aidl",
srcs: [
"aidl/android/media/IMediaTranscodingService.aidl",
diff --git a/media/libstagefright/codecs/mp3dec/src/pvmp3_framedecoder.cpp b/media/libstagefright/codecs/mp3dec/src/pvmp3_framedecoder.cpp
index df6cd03..a5c7f5e 100644
--- a/media/libstagefright/codecs/mp3dec/src/pvmp3_framedecoder.cpp
+++ b/media/libstagefright/codecs/mp3dec/src/pvmp3_framedecoder.cpp
@@ -659,20 +659,12 @@
huffcodetab *pHuff;
pVars = (tmp3dec_file *)pMem;
-
- pVars->num_channels = 0;
+ memset(pVars, 0, sizeof(*pVars));
pExt->totalNumberOfBitsUsed = 0;
pExt->inputBufferCurrentLength = 0;
pExt->inputBufferUsedLength = 0;
- pVars->mainDataStream.offset = 0;
-
- pv_memset((void*)pVars->mainDataBuffer,
- 0,
- BUFSIZE*sizeof(*pVars->mainDataBuffer));
-
-
pVars->inputStream.pBuffer = pExt->pInputBuffer;
/*
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index aaa28bc..d5272bc 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -152,10 +152,16 @@
bool AudioOutputDescriptor::setVolume(float volumeDb,
VolumeSource volumeSource,
const StreamTypeVector &/*streams*/,
- const DeviceTypeSet& /*deviceTypes*/,
+ const DeviceTypeSet& deviceTypes,
uint32_t delayMs,
bool force)
{
+
+ if (!supportedDevices().containsDeviceAmongTypes(deviceTypes)) {
+ ALOGV("%s output ID %d unsupported device %s",
+ __func__, getId(), toString(deviceTypes).c_str());
+ return false;
+ }
// We actually change the volume if:
// - the float value returned by computeVolume() changed
// - the force flag is set
diff --git a/services/audiopolicy/engine/config/include/EngineConfig.h b/services/audiopolicy/engine/config/include/EngineConfig.h
index 7f5ed5e..5d22c24 100644
--- a/services/audiopolicy/engine/config/include/EngineConfig.h
+++ b/services/audiopolicy/engine/config/include/EngineConfig.h
@@ -31,9 +31,6 @@
/** Default path of audio policy usages configuration file. */
constexpr char DEFAULT_PATH[] = "/vendor/etc/audio_policy_engine_configuration.xml";
-/** Directories where the effect libraries will be search for. */
-constexpr const char* POLICY_USAGE_LIBRARY_PATH[] = {"/odm/etc/", "/vendor/etc/", "/system/etc/"};
-
using AttributesVector = std::vector<audio_attributes_t>;
using StreamVector = std::vector<audio_stream_type_t>;
diff --git a/services/audiopolicy/engine/config/src/EngineConfig.cpp b/services/audiopolicy/engine/config/src/EngineConfig.cpp
index 7f8cdd9..4842cb2 100644
--- a/services/audiopolicy/engine/config/src/EngineConfig.cpp
+++ b/services/audiopolicy/engine/config/src/EngineConfig.cpp
@@ -21,6 +21,7 @@
#include <cutils/properties.h>
#include <media/TypeConverter.h>
#include <media/convert.h>
+#include <system/audio_config.h>
#include <utils/Log.h>
#include <libxml/parser.h>
#include <libxml/xinclude.h>
@@ -693,9 +694,6 @@
return deserializeLegacyVolumeCollection(doc, cur, volumeGroups, nbSkippedElements);
}
-static const char *kConfigLocationList[] = {"/odm/etc", "/vendor/etc", "/system/etc"};
-static const int kConfigLocationListSize =
- (sizeof(kConfigLocationList) / sizeof(kConfigLocationList[0]));
static const int gApmXmlConfigFilePathMaxLength = 128;
static constexpr const char *apmXmlConfigFileName = "audio_policy_configuration.xml";
@@ -715,9 +713,9 @@
fileNames.push_back(apmXmlConfigFileName);
for (const char* fileName : fileNames) {
- for (int i = 0; i < kConfigLocationListSize; i++) {
+ for (const auto& path : audio_get_configuration_paths()) {
snprintf(audioPolicyXmlConfigFile, sizeof(audioPolicyXmlConfigFile),
- "%s/%s", kConfigLocationList[i], fileName);
+ "%s/%s", path.c_str(), fileName);
ret = parseLegacyVolumeFile(audioPolicyXmlConfigFile, volumeGroups);
if (ret == NO_ERROR) {
return ret;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 3d16977..f1c2ab5 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -47,6 +47,7 @@
#include <media/AudioParameter.h>
#include <private/android_filesystem_config.h>
#include <system/audio.h>
+#include <system/audio_config.h>
#include "AudioPolicyManager.h"
#include <Serializer.h>
#include "TypeConverter.h"
@@ -4367,12 +4368,6 @@
return mAudioPortGeneration++;
}
-// Treblized audio policy xml config will be located in /odm/etc or /vendor/etc.
-static const char *kConfigLocationList[] =
- {"/odm/etc", "/vendor/etc", "/system/etc"};
-static const int kConfigLocationListSize =
- (sizeof(kConfigLocationList) / sizeof(kConfigLocationList[0]));
-
static status_t deserializeAudioPolicyXmlConfig(AudioPolicyConfig &config) {
char audioPolicyXmlConfigFile[AUDIO_POLICY_XML_CONFIG_FILE_PATH_MAX_LENGTH];
std::vector<const char*> fileNames;
@@ -4394,9 +4389,9 @@
fileNames.push_back(AUDIO_POLICY_XML_CONFIG_FILE_NAME);
for (const char* fileName : fileNames) {
- for (int i = 0; i < kConfigLocationListSize; i++) {
+ for (const auto& path : audio_get_configuration_paths()) {
snprintf(audioPolicyXmlConfigFile, sizeof(audioPolicyXmlConfigFile),
- "%s/%s", kConfigLocationList[i], fileName);
+ "%s/%s", path.c_str(), fileName);
ret = deserializeAudioPolicyFile(audioPolicyXmlConfigFile, &config);
if (ret == NO_ERROR) {
config.setSource(audioPolicyXmlConfigFile);