Consolidate libaudioclient buffers p1
Prepare for libaudioclient buffer refactor by updating
clients of libaudioclient to use new buffer interface.
libaudioclient
- Wrap existing buffers with new interface
- Modify internal calls to be compatible with wrap
AAudio
- Update to use new buffer interface
- Update record to use callback
TrackPlayerBase
- Used for SLES (in different repo). Update to use sp<>
ToneGenerator/MediaPlayerService/AudioPlayer
- Update to use new buffer interface
StageFright
- Update to new callback interface
- Update to use new buffer interface
Bug: 216175830 - shared buffer
Bug: 199156212 - callback interface
Test: atest AudioTrackTest AudioRecordTest
atest AudioTrackOffloadTest
OboeTester non-exclusive, non-MMAP, power-saving for both
AAudio and SLES, input and output
No-Typo-Check: Existing class members
Change-Id: Ib1241f2e530bc509b2d4dde956ec5188f2287994
diff --git a/cmds/stagefright/AudioPlayer.cpp b/cmds/stagefright/AudioPlayer.cpp
index a63bde6..6cddf47 100644
--- a/cmds/stagefright/AudioPlayer.cpp
+++ b/cmds/stagefright/AudioPlayer.cpp
@@ -453,7 +453,7 @@
}
size_t AudioPlayer::onMoreData(const AudioTrack::Buffer& buffer) {
- return fillBuffer(buffer.raw, buffer.size);
+ return fillBuffer(buffer.data(), buffer.size());
}
void AudioPlayer::onStreamEnd() {
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index 38f3c24..dd11169 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -76,6 +76,7 @@
// This takes advantage of them killing the stream when they see a size out of range.
// That is an undocumented behavior.
// TODO add to API in AudioRecord and AudioTrack
+ // TODO(b/216175830) cleanup size re-computation
const size_t SIZE_STOP_CALLBACKS = SIZE_MAX;
aaudio_data_callback_result_t callbackResult;
(void) checkForDisconnectRequest(true);
@@ -83,7 +84,7 @@
// Note that this code assumes an AudioTrack::Buffer is the same as
// AudioRecord::Buffer
// TODO define our own AudioBuffer and pass it from the subclasses.
- size_t written = buffer.size;
+ size_t written = buffer.size();
if (getState() == AAUDIO_STREAM_STATE_DISCONNECTED) {
ALOGW("%s() data, stream disconnected", __func__);
// This will kill the stream and prevent it from being restarted.
@@ -96,23 +97,23 @@
// caused by Legacy callbacks running after the track is "stopped".
written = 0;
} else {
- if (buffer.frameCount == 0) {
+ if (buffer.getFrameCount() == 0) {
ALOGW("%s() data, frameCount is zero", __func__);
return written;
}
// If the caller specified an exact size then use a block size adapter.
if (mBlockAdapter != nullptr) {
- int32_t byteCount = buffer.frameCount * getBytesPerDeviceFrame();
+ int32_t byteCount = buffer.getFrameCount() * getBytesPerDeviceFrame();
callbackResult = mBlockAdapter->processVariableBlock(
- static_cast<uint8_t*>(buffer.raw), byteCount);
+ buffer.data(), byteCount);
} else {
// Call using the AAudio callback interface.
- callbackResult = callDataCallbackFrames(static_cast<uint8_t *>(buffer.raw),
- buffer.frameCount);
+ callbackResult = callDataCallbackFrames(buffer.data(),
+ buffer.getFrameCount());
}
if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
- written = buffer.frameCount * getBytesPerDeviceFrame();
+ written = buffer.getFrameCount() * getBytesPerDeviceFrame();
} else {
if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
ALOGD("%s() callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
@@ -134,6 +135,70 @@
return written;
}
+// TODO (b/216175830) this method is duplicated in order to ease refactoring which will
+// reconsolidate.
+size_t AudioStreamLegacy::onMoreData(const android::AudioRecord::Buffer& buffer) {
+ // This illegal size can be used to tell AudioRecord or AudioTrack to stop calling us.
+ // This takes advantage of them killing the stream when they see a size out of range.
+ // That is an undocumented behavior.
+ // TODO add to API in AudioRecord and AudioTrack
+ const size_t SIZE_STOP_CALLBACKS = SIZE_MAX;
+ aaudio_data_callback_result_t callbackResult;
+ (void) checkForDisconnectRequest(true);
+
+ // Note that this code assumes an AudioTrack::Buffer is the same as
+ // AudioRecord::Buffer
+ // TODO define our own AudioBuffer and pass it from the subclasses.
+ size_t written = buffer.size();
+ if (getState() == AAUDIO_STREAM_STATE_DISCONNECTED) {
+ ALOGW("%s() data, stream disconnected", __func__);
+ // This will kill the stream and prevent it from being restarted.
+ // That is OK because the stream is disconnected.
+ written = SIZE_STOP_CALLBACKS;
+ } else if (!mCallbackEnabled.load()) {
+ ALOGW("%s() no data because callback disabled, set size=0", __func__);
+ // Do NOT use SIZE_STOP_CALLBACKS here because that will kill the stream and
+ // prevent it from being restarted. This can occur because of a race condition
+ // caused by Legacy callbacks running after the track is "stopped".
+ written = 0;
+ } else {
+ if (buffer.getFrameCount() == 0) {
+ ALOGW("%s() data, frameCount is zero", __func__);
+ return written;
+ }
+
+ // If the caller specified an exact size then use a block size adapter.
+ if (mBlockAdapter != nullptr) {
+ int32_t byteCount = buffer.getFrameCount() * getBytesPerDeviceFrame();
+ callbackResult = mBlockAdapter->processVariableBlock(
+ buffer.data(), byteCount);
+ } else {
+ // Call using the AAudio callback interface.
+ callbackResult = callDataCallbackFrames(buffer.data(),
+ buffer.getFrameCount());
+ }
+ if (callbackResult == AAUDIO_CALLBACK_RESULT_CONTINUE) {
+ written = buffer.getFrameCount() * getBytesPerDeviceFrame();
+ } else {
+ if (callbackResult == AAUDIO_CALLBACK_RESULT_STOP) {
+ ALOGD("%s() callback returned AAUDIO_CALLBACK_RESULT_STOP", __func__);
+ } else {
+ ALOGW("%s() callback returned invalid result = %d",
+ __func__, callbackResult);
+ }
+ written = 0;
+ systemStopInternal();
+ // Disable the callback just in case the system keeps trying to call us.
+ mCallbackEnabled.store(false);
+ }
+
+ if (updateStateMachine() != AAUDIO_OK) {
+ forceDisconnect();
+ mCallbackEnabled.store(false);
+ }
+ }
+ return written;
+}
aaudio_result_t AudioStreamLegacy::checkForDisconnectRequest(bool errorCallbackEnabled) {
if (mRequestDisconnect.isRequested()) {
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
index c54d7e2..53f6e06 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.h
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h
@@ -17,9 +17,10 @@
#ifndef LEGACY_AUDIO_STREAM_LEGACY_H
#define LEGACY_AUDIO_STREAM_LEGACY_H
+#include <media/AudioRecord.h>
+#include <media/AudioSystem.h>
#include <media/AudioTimestamp.h>
#include <media/AudioTrack.h>
-#include <media/AudioSystem.h>
#include <aaudio/AAudio.h>
@@ -57,7 +58,8 @@
class AudioStreamLegacy : public AudioStream,
public FixedBlockProcessor,
- protected android::AudioTrack::IAudioTrackCallback {
+ protected android::AudioTrack::IAudioTrackCallback,
+ protected android::AudioRecord::IAudioRecordCallback {
public:
AudioStreamLegacy();
@@ -82,7 +84,11 @@
protected:
size_t onMoreData(const android::AudioTrack::Buffer& buffer) override;
+ // TODO (b/216175830) this method is duplicated in order to ease refactoring which will
+ // reconsolidate.
+ size_t onMoreData(const android::AudioRecord::Buffer& buffer) override;
void onNewIAudioTrack() override;
+ void onNewIAudioRecord() override { onNewIAudioTrack(); }
aaudio_result_t getBestTimestamp(clockid_t clockId,
int64_t *framePosition,
int64_t *timeNanoseconds,
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index df7d4cf..988d097 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -37,10 +37,6 @@
using namespace android;
using namespace aaudio;
-static void sCallbackWrapper(int event, void* userData, void* info) {
- static_cast<AudioStreamRecord*>(userData)->processCallback(event, info);
-}
-
AudioStreamRecord::AudioStreamRecord()
: AudioStreamLegacy()
, mFixedBlockWriter(*this)
@@ -128,13 +124,11 @@
uint32_t notificationFrames = 0;
// Setup the callback if there is one.
- AudioRecord::legacy_callback_t callback = nullptr;
- void *callbackData = nullptr;
+ sp<AudioRecord::IAudioRecordCallback> callback;
AudioRecord::transfer_type streamTransferType = AudioRecord::transfer_type::TRANSFER_SYNC;
if (builder.getDataCallbackProc() != nullptr) {
streamTransferType = AudioRecord::transfer_type::TRANSFER_CALLBACK;
- callback = sCallbackWrapper;
- callbackData = this;
+ callback = sp<AudioRecord::IAudioRecordCallback>::fromExisting(this);
}
mCallbackBufferSize = builder.getFramesPerDataCallback();
@@ -181,7 +175,6 @@
channelMask,
frameCount,
callback,
- callbackData,
notificationFrames,
false /*threadCanCallJava*/,
sessionId,
@@ -354,24 +347,6 @@
}
}
-void AudioStreamRecord::processCallback(int event, void *info) {
- switch (event) {
- case AudioRecord::EVENT_MORE_DATA:
- {
- AudioTrack::Buffer *audioBuffer = static_cast<AudioTrack::Buffer *>(info);
- audioBuffer->size = onMoreData(*audioBuffer);
- break;
- }
- // Stream got rerouted so we disconnect.
- case AudioRecord::EVENT_NEW_IAUDIORECORD:
- onNewIAudioTrack();
- break;
- default:
- break;
- }
- return;
-}
-
aaudio_result_t AudioStreamRecord::requestStart_l()
{
if (mAudioRecord.get() == nullptr) {
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index ebd488a..0c37fb5 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -266,7 +266,7 @@
size_t onMoreData(const AudioRecord::Buffer& buffer) override {
AudioRecord::Buffer copy = buffer;
mCallback(AudioRecord::EVENT_MORE_DATA, mData, ©);
- return copy.size;
+ return copy.size();
}
void onOverrun() override { mCallback(AudioRecord::EVENT_OVERRUN, mData, nullptr); }
@@ -1131,7 +1131,7 @@
}
if (mTransfer != TRANSFER_OBTAIN) {
audioBuffer->frameCount = 0;
- audioBuffer->size = 0;
+ audioBuffer->mSize = 0;
audioBuffer->raw = NULL;
if (nonContig != NULL) {
*nonContig = 0;
@@ -1214,7 +1214,7 @@
} while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
- audioBuffer->size = buffer.mFrameCount * mServerFrameSize;
+ audioBuffer->mSize = buffer.mFrameCount * mServerFrameSize;
audioBuffer->raw = buffer.mRaw;
audioBuffer->sequence = oldSequence;
if (nonContig != NULL) {
@@ -1291,7 +1291,7 @@
size_t bytesRead = audioBuffer.frameCount * mFrameSize;
memcpy_by_audio_format(buffer, mFormat, audioBuffer.raw, mServerConfig.format,
- audioBuffer.size / mServerSampleSize);
+ audioBuffer.mSize / mServerSampleSize);
buffer = ((char *) buffer) + bytesRead;
userSize -= bytesRead;
read += bytesRead;
@@ -1497,15 +1497,15 @@
if (mServerConfig.format != mFormat) {
buffer = &mFormatConversionBuffer;
buffer->frameCount = audioBuffer.frameCount;
- buffer->size = buffer->frameCount * mFrameSize;
+ buffer->mSize = buffer->frameCount * mFrameSize;
buffer->sequence = audioBuffer.sequence;
memcpy_by_audio_format(buffer->raw, mFormat, audioBuffer.raw,
- mServerConfig.format, audioBuffer.size / mServerSampleSize);
+ mServerConfig.format, audioBuffer.size() / mServerSampleSize);
}
- const size_t reqSize = buffer->size;
+ const size_t reqSize = buffer->size();
const size_t readSize = callback->onMoreData(*buffer);
- buffer->size = readSize;
+ buffer->mSize = readSize;
// Validate on returned size
if (ssize_t(readSize) < 0 || readSize > reqSize) {
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index bdf3147..919d6d2 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -293,7 +293,7 @@
size_t onMoreData(const AudioTrack::Buffer & buffer) override {
AudioTrack::Buffer copy = buffer;
mCallback(AudioTrack::EVENT_MORE_DATA, mData, static_cast<void*>(©));
- return copy.size;
+ return copy.size();
}
void onUnderrun() override {
mCallback(AudioTrack::EVENT_UNDERRUN, mData, nullptr);
@@ -319,7 +319,7 @@
size_t onCanWriteMoreData(const AudioTrack::Buffer & buffer) override {
AudioTrack::Buffer copy = buffer;
mCallback(AudioTrack::EVENT_CAN_WRITE_MORE_DATA, mData, static_cast<void*>(©));
- return copy.size;
+ return copy.size();
}
};
}
@@ -2194,7 +2194,7 @@
}
if (mTransfer != TRANSFER_OBTAIN) {
audioBuffer->frameCount = 0;
- audioBuffer->size = 0;
+ audioBuffer->mSize = 0;
audioBuffer->raw = NULL;
if (nonContig != NULL) {
*nonContig = 0;
@@ -2286,7 +2286,7 @@
} while (((status == DEAD_OBJECT) || (status == NOT_ENOUGH_DATA)) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
- audioBuffer->size = buffer.mFrameCount * mFrameSize;
+ audioBuffer->mSize = buffer.mFrameCount * mFrameSize;
audioBuffer->raw = buffer.mRaw;
audioBuffer->sequence = oldSequence;
if (nonContig != NULL) {
@@ -2302,7 +2302,7 @@
return;
}
- size_t stepCount = audioBuffer->size / mFrameSize;
+ size_t stepCount = audioBuffer->mSize / mFrameSize;
if (stepCount == 0) {
return;
}
@@ -2382,8 +2382,8 @@
return ssize_t(err);
}
- size_t toWrite = audioBuffer.size;
- memcpy(audioBuffer.i8, buffer, toWrite);
+ size_t toWrite = audioBuffer.size();
+ memcpy(audioBuffer.raw, buffer, toWrite);
buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
@@ -2740,11 +2740,11 @@
}
}
- size_t reqSize = audioBuffer.size;
+ size_t reqSize = audioBuffer.size();
if (mTransfer == TRANSFER_SYNC_NOTIF_CALLBACK) {
// when notifying client it can write more data, pass the total size that can be
// written in the next write() call, since it's not passed through the callback
- audioBuffer.size += nonContig;
+ audioBuffer.mSize += nonContig;
}
const size_t writtenSize = (mTransfer == TRANSFER_CALLBACK)
? callback->onMoreData(audioBuffer)
@@ -2809,7 +2809,7 @@
}
// releaseBuffer reads from audioBuffer.size
- audioBuffer.size = writtenSize;
+ audioBuffer.mSize = writtenSize;
size_t releasedFrames = writtenSize / mFrameSize;
audioBuffer.frameCount = releasedFrames;
diff --git a/media/libaudioclient/ToneGenerator.cpp b/media/libaudioclient/ToneGenerator.cpp
index cd3eacb..9b43f3c 100644
--- a/media/libaudioclient/ToneGenerator.cpp
+++ b/media/libaudioclient/ToneGenerator.cpp
@@ -1329,7 +1329,7 @@
//
// Input:
// buffer An buffer object containing a pointer which we will fill with
-// buffer.size bytes.
+// buffer.size() bytes.
//
// Output:
// The number of bytes we successfully wrote.
@@ -1337,16 +1337,16 @@
////////////////////////////////////////////////////////////////////////////////
size_t ToneGenerator::onMoreData(const AudioTrack::Buffer& buffer) {
- int16_t *lpOut = buffer.i16;
- uint32_t lNumSmp = (buffer.size / sizeof(int16_t) < UINT32_MAX) ?
- buffer.size / sizeof(int16_t) : UINT32_MAX;
- if (buffer.size == 0) return 0;
+ int16_t *lpOut = reinterpret_cast<int16_t*>(buffer.data());
+ uint32_t lNumSmp = (buffer.size() / sizeof(int16_t) < UINT32_MAX) ?
+ buffer.size() / sizeof(int16_t) : UINT32_MAX;
+ if (buffer.size() == 0) return 0;
// We will write to the entire buffer unless we are stopped, then we return
// 0 at loop end
size_t bytesWritten = lNumSmp * sizeof(int16_t);
// Clear output buffer: WaveGenerator accumulates into lpOut buffer
- memset(lpOut, 0, buffer.size);
+ memset(lpOut, 0, buffer.size());
while (lNumSmp) {
unsigned int lReqSmp = lNumSmp < mProcessSize*2 ? lNumSmp : mProcessSize;
diff --git a/media/libaudioclient/TrackPlayerBase.cpp b/media/libaudioclient/TrackPlayerBase.cpp
index 188f321..4fc1c44 100644
--- a/media/libaudioclient/TrackPlayerBase.cpp
+++ b/media/libaudioclient/TrackPlayerBase.cpp
@@ -33,11 +33,14 @@
doDestroy();
}
-void TrackPlayerBase::init(AudioTrack* pat, player_type_t playerType, audio_usage_t usage,
- audio_session_t sessionId) {
+void TrackPlayerBase::init(const sp<AudioTrack>& pat,
+ const sp<AudioTrack::IAudioTrackCallback>& callback,
+ player_type_t playerType, audio_usage_t usage,
+ audio_session_t sessionId) {
PlayerBase::init(playerType, usage, sessionId);
mAudioTrack = pat;
if (mAudioTrack != 0) {
+ mCallbackHandle = callback;
mSelfAudioDeviceCallback = new SelfAudioDeviceCallback(*this);
mAudioTrack->addAudioDeviceCallback(mSelfAudioDeviceCallback);
mAudioTrack->setPlayerIId(mPIId); // set in PlayerBase::init().
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
index 4c89249..169a6a7 100644
--- a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -355,7 +355,7 @@
audioBuffer.frameCount = static_cast<size_t>(mFdp.ConsumeIntegral<uint32_t>());
record->obtainBuffer(&audioBuffer, waitCount, &nonContig);
bool blocking = false;
- record->read(audioBuffer.raw, audioBuffer.size, blocking);
+ record->read(audioBuffer.data(), audioBuffer.size(), blocking);
record->getInputFramesLost();
record->getFlags();
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index 3cfcbf3..faea716 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -40,7 +40,6 @@
struct audio_track_cblk_t;
class AudioRecordClientProxy;
-
// ----------------------------------------------------------------------------
class AudioRecord : public AudioSystem::AudioDeviceCallback
@@ -70,15 +69,21 @@
class Buffer
{
+ friend AudioRecord;
public:
- // FIXME use m prefix
+ size_t size() const { return mSize; }
+ size_t getFrameCount() const { return frameCount; }
+ uint8_t* data() const { return ui8; }
+ // Leaving public for now to assist refactoring. This class will
+ // be replaced.
size_t frameCount; // number of sample frames corresponding to size;
// on input to obtainBuffer() it is the number of frames desired
// on output from obtainBuffer() it is the number of available
// frames to be read
// on input to releaseBuffer() it is currently ignored
- size_t size; // input/output in bytes == frameCount * frameSize
+ private:
+ size_t mSize; // input/output in bytes == frameCount * frameSize
// on input to obtainBuffer() it is ignored
// on output from obtainBuffer() it is the number of available
// bytes to be read, which is frameCount * frameSize
@@ -90,7 +95,7 @@
union {
void* raw;
int16_t* i16; // signed 16-bit
- int8_t* i8; // unsigned 8-bit, offset by 0x80
+ uint8_t* ui8; // unsigned 8-bit, offset by 0x80
// input to obtainBuffer(): unused, output: pointer to buffer
};
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index 16e10b5..d777124 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -95,34 +95,36 @@
class Buffer
{
+ friend AudioTrack;
public:
- // FIXME use m prefix
+ size_t size() const { return mSize; }
+ size_t getFrameCount() const { return frameCount; }
+ uint8_t * data() const { return ui8; }
+ // Leaving public for now to ease refactoring. This class will be
+ // replaced
size_t frameCount; // number of sample frames corresponding to size;
// on input to obtainBuffer() it is the number of frames desired,
// on output from obtainBuffer() it is the number of available
// [empty slots for] frames to be filled
// on input to releaseBuffer() it is currently ignored
-
- size_t size; // input/output in bytes == frameCount * frameSize
+ private:
+ size_t mSize; // input/output in bytes == frameCount * frameSize
// on input to obtainBuffer() it is ignored
// on output from obtainBuffer() it is the number of available
// [empty slots for] bytes to be filled,
// which is frameCount * frameSize
// on input to releaseBuffer() it is the number of bytes to
// release
- // FIXME This is redundant with respect to frameCount. Consider
- // removing size and making frameCount the primary field.
union {
void* raw;
int16_t* i16; // signed 16-bit
- int8_t* i8; // unsigned 8-bit, offset by 0x80
+ uint8_t* ui8; // unsigned 8-bit, offset by 0x80
}; // input to obtainBuffer(): unused, output: pointer to buffer
uint32_t sequence; // IAudioTrack instance sequence number, as of obtainBuffer().
// It is set by obtainBuffer() and confirmed by releaseBuffer().
// Not "user-serviceable".
- // TODO Consider sp<IMemory> instead, or in addition to this.
};
/* As a convenience, if a callback is supplied, a handler thread
diff --git a/media/libaudioclient/include/media/TrackPlayerBase.h b/media/libaudioclient/include/media/TrackPlayerBase.h
index 80124b8..fe88116 100644
--- a/media/libaudioclient/include/media/TrackPlayerBase.h
+++ b/media/libaudioclient/include/media/TrackPlayerBase.h
@@ -28,8 +28,8 @@
explicit TrackPlayerBase();
virtual ~TrackPlayerBase();
- void init(AudioTrack* pat, player_type_t playerType, audio_usage_t usage,
- audio_session_t sessionId);
+ void init(const sp<AudioTrack>& pat, const sp<AudioTrack::IAudioTrackCallback>& callback,
+ player_type_t playerType, audio_usage_t usage, audio_session_t sessionId);
virtual void destroy();
//IPlayer implementation
@@ -66,8 +66,8 @@
// volume coming from the player volume API
float mPlayerVolumeL, mPlayerVolumeR;
-
- sp<SelfAudioDeviceCallback> mSelfAudioDeviceCallback;
+ sp<AudioTrack::IAudioTrackCallback> mCallbackHandle;
+ sp<SelfAudioDeviceCallback> mSelfAudioDeviceCallback;
};
} // namespace android
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index c7a7a3a..b512982 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -2658,7 +2658,7 @@
return 0;
}
size_t actualSize = (*me->mCallback)(
- me.get(), buffer.raw, buffer.size, me->mCallbackCookie,
+ me.get(), buffer.data(), buffer.size(), me->mCallbackCookie,
CB_EVENT_FILL_BUFFER);
// Log when no data is returned from the callback.
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index b6acdc8..faaae3f 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -36,21 +36,9 @@
using content::AttributionSourceState;
-static void AudioRecordCallbackFunction(int event, void *user, void *info) {
- AudioSource *source = (AudioSource *) user;
- switch (event) {
- case AudioRecord::EVENT_MORE_DATA: {
- source->dataCallback(*((AudioRecord::Buffer *) info));
- break;
- }
- case AudioRecord::EVENT_OVERRUN: {
- ALOGW("AudioRecord reported overrun!");
- break;
- }
- default:
- // does nothing
- break;
- }
+
+void AudioSource::onOverrun() {
+ ALOGW("AudioRecord reported overrun!");
}
AudioSource::AudioSource(
@@ -129,8 +117,7 @@
audio_channel_in_mask_from_count(channelCount),
attributionSource,
(size_t) (bufCount * frameCount),
- AudioRecordCallbackFunction,
- this,
+ wp<AudioRecord::IAudioRecordCallback>::fromExisting(this),
frameCount /*notificationFrames*/,
AUDIO_SESSION_ALLOCATE,
AudioRecord::TRANSFER_DEFAULT,
@@ -359,7 +346,7 @@
return;
}
-status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
+size_t AudioSource::onMoreData(const AudioRecord::Buffer& audioBuffer) {
int64_t timeUs, position, timeNs;
ExtendedTimestamp ts;
ExtendedTimestamp::Location location;
@@ -384,21 +371,21 @@
ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);
Mutex::Autolock autoLock(mLock);
+
if (!mStarted) {
ALOGW("Spurious callback from AudioRecord. Drop the audio data.");
- return OK;
+ return audioBuffer.size();
}
- const size_t bufferSize = audioBuffer.size;
// Drop retrieved and previously lost audio data.
if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
(void) mRecord->getInputFramesLost();
- int64_t receievedFrames = bufferSize / mRecord->frameSize();
+ int64_t receievedFrames = audioBuffer.size() / mRecord->frameSize();
ALOGV("Drop audio data(%" PRId64 " frames) at %" PRId64 "/%" PRId64 " us",
receievedFrames, timeUs, mStartTimeUs);
mNumFramesSkipped += receievedFrames;
- return OK;
+ return audioBuffer.size();
}
if (mStopSystemTimeUs != -1 && timeUs >= mStopSystemTimeUs) {
@@ -406,7 +393,7 @@
(long long)timeUs, (long long)mStopSystemTimeUs);
mNoMoreFramesToRead = true;
mFrameAvailableCondition.signal();
- return OK;
+ return audioBuffer.size();
}
if (mNumFramesReceived == 0 && mPrevSampleTimeUs == 0) {
@@ -427,7 +414,7 @@
}
CHECK_EQ(numLostBytes & 1, 0u);
- CHECK_EQ(audioBuffer.size & 1, 0u);
+ CHECK_EQ(audioBuffer.size() & 1, 0u);
if (numLostBytes > 0) {
// Loss of audio frames should happen rarely; thus the LOGW should
// not cause a logging spam
@@ -449,17 +436,17 @@
queueInputBuffer_l(lostAudioBuffer, timeUs);
}
- if (audioBuffer.size == 0) {
+ if (audioBuffer.size() == 0) {
ALOGW("Nothing is available from AudioRecord callback buffer");
- return OK;
+ return audioBuffer.size();
}
- MediaBuffer *buffer = new MediaBuffer(bufferSize);
+ MediaBuffer *buffer = new MediaBuffer(audioBuffer.size());
memcpy((uint8_t *) buffer->data(),
- audioBuffer.i16, audioBuffer.size);
- buffer->set_range(0, bufferSize);
+ audioBuffer.data(), audioBuffer.size());
+ buffer->set_range(0, audioBuffer.size());
queueInputBuffer_l(buffer, timeUs);
- return OK;
+ return audioBuffer.size();
}
void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
diff --git a/media/libstagefright/include/media/stagefright/AudioSource.h b/media/libstagefright/include/media/stagefright/AudioSource.h
index 43d50f1..5e84977 100644
--- a/media/libstagefright/include/media/stagefright/AudioSource.h
+++ b/media/libstagefright/include/media/stagefright/AudioSource.h
@@ -35,7 +35,9 @@
class AudioRecord;
-struct AudioSource : public MediaSource, public MediaBufferObserver {
+struct AudioSource : public MediaSource,
+ public MediaBufferObserver,
+ public AudioRecord::IAudioRecordCallback {
// Note that the "channels" parameter _is_ the number of channels,
// _not_ a bitmask of audio_channels_t constants.
AudioSource(
@@ -74,7 +76,6 @@
MediaBufferBase **buffer, const ReadOptions *options = NULL);
virtual status_t setStopTimeUs(int64_t stopTimeUs);
- status_t dataCallback(const AudioRecord::Buffer& buffer);
virtual void signalBufferReturned(MediaBufferBase *buffer);
status_t setInputDevice(audio_port_handle_t deviceId);
@@ -142,6 +143,10 @@
void waitOutstandingEncodingFrames_l();
status_t reset();
+ // IAudioRecordCallback implementation
+ size_t onMoreData(const AudioRecord::Buffer&) override;
+ void onOverrun() override;
+
AudioSource(const AudioSource &);
AudioSource &operator=(const AudioSource &);