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Diffstat (limited to 'media/libmedia/AudioTrack.cpp')
-rw-r--r--media/libmedia/AudioTrack.cpp348
1 files changed, 217 insertions, 131 deletions
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index b2c067b1290f..8529a8e46fa1 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -32,9 +32,9 @@
#include <media/AudioTrack.h>
#include <utils/Log.h>
-#include <utils/MemoryDealer.h>
-#include <utils/Parcel.h>
-#include <utils/IPCThreadState.h>
+#include <binder/MemoryDealer.h>
+#include <binder/Parcel.h>
+#include <binder/IPCThreadState.h>
#include <utils/Timers.h>
#include <cutils/atomic.h>
@@ -54,7 +54,7 @@ AudioTrack::AudioTrack(
int streamType,
uint32_t sampleRate,
int format,
- int channelCount,
+ int channels,
int frameCount,
uint32_t flags,
callback_t cbf,
@@ -62,7 +62,7 @@ AudioTrack::AudioTrack(
int notificationFrames)
: mStatus(NO_INIT)
{
- mStatus = set(streamType, sampleRate, format, channelCount,
+ mStatus = set(streamType, sampleRate, format, channels,
frameCount, flags, cbf, user, notificationFrames, 0);
}
@@ -70,7 +70,7 @@ AudioTrack::AudioTrack(
int streamType,
uint32_t sampleRate,
int format,
- int channelCount,
+ int channels,
const sp<IMemory>& sharedBuffer,
uint32_t flags,
callback_t cbf,
@@ -78,7 +78,7 @@ AudioTrack::AudioTrack(
int notificationFrames)
: mStatus(NO_INIT)
{
- mStatus = set(streamType, sampleRate, format, channelCount,
+ mStatus = set(streamType, sampleRate, format, channels,
0, flags, cbf, user, notificationFrames, sharedBuffer);
}
@@ -97,6 +97,7 @@ AudioTrack::~AudioTrack()
}
mAudioTrack.clear();
IPCThreadState::self()->flushCommands();
+ AudioSystem::releaseOutput(getOutput());
}
}
@@ -104,7 +105,7 @@ status_t AudioTrack::set(
int streamType,
uint32_t sampleRate,
int format,
- int channelCount,
+ int channels,
int frameCount,
uint32_t flags,
callback_t cbf,
@@ -121,11 +122,6 @@ status_t AudioTrack::set(
return INVALID_OPERATION;
}
- const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
- if (audioFlinger == 0) {
- LOGE("Could not get audioflinger");
- return NO_INIT;
- }
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
@@ -150,73 +146,82 @@ status_t AudioTrack::set(
if (format == 0) {
format = AudioSystem::PCM_16_BIT;
}
- if (channelCount == 0) {
- channelCount = 2;
+ if (channels == 0) {
+ channels = AudioSystem::CHANNEL_OUT_STEREO;
}
// validate parameters
- if (((format != AudioSystem::PCM_8_BIT) || sharedBuffer != 0) &&
- (format != AudioSystem::PCM_16_BIT)) {
+ if (!AudioSystem::isValidFormat(format)) {
LOGE("Invalid format");
return BAD_VALUE;
}
- if (channelCount != 1 && channelCount != 2) {
- LOGE("Invalid channel number");
+
+ // force direct flag if format is not linear PCM
+ if (!AudioSystem::isLinearPCM(format)) {
+ flags |= AudioSystem::OUTPUT_FLAG_DIRECT;
+ }
+
+ if (!AudioSystem::isOutputChannel(channels)) {
+ LOGE("Invalid channel mask");
return BAD_VALUE;
}
+ uint32_t channelCount = AudioSystem::popCount(channels);
- // Ensure that buffer depth covers at least audio hardware latency
- uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
- if (minBufCount < 2) minBufCount = 2;
+ audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
+ sampleRate, format, channels, (AudioSystem::output_flags)flags);
- // When playing from shared buffer, playback will start even if last audioflinger
- // block is partly filled.
- if (sharedBuffer != 0 && minBufCount > 1) {
- minBufCount--;
+ if (output == 0) {
+ LOGE("Could not get audio output for stream type %d", streamType);
+ return BAD_VALUE;
}
- int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
-
- if (sharedBuffer == 0) {
- if (frameCount == 0) {
- frameCount = minFrameCount;
- }
- if (notificationFrames == 0) {
- notificationFrames = frameCount/2;
- }
- // Make sure that application is notified with sufficient margin
- // before underrun
- if (notificationFrames > frameCount/2) {
- notificationFrames = frameCount/2;
+ if (!AudioSystem::isLinearPCM(format)) {
+ if (sharedBuffer != 0) {
+ frameCount = sharedBuffer->size();
}
} else {
- // Ensure that buffer alignment matches channelcount
- if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
- LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
- return BAD_VALUE;
- }
- frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
- }
+ // Ensure that buffer depth covers at least audio hardware latency
+ uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
+ if (minBufCount < 2) minBufCount = 2;
- if (frameCount < minFrameCount) {
- LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
- return BAD_VALUE;
+ int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
+
+ if (sharedBuffer == 0) {
+ if (frameCount == 0) {
+ frameCount = minFrameCount;
+ }
+ if (notificationFrames == 0) {
+ notificationFrames = frameCount/2;
+ }
+ // Make sure that application is notified with sufficient margin
+ // before underrun
+ if (notificationFrames > frameCount/2) {
+ notificationFrames = frameCount/2;
+ }
+ if (frameCount < minFrameCount) {
+ LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
+ return BAD_VALUE;
+ }
+ } else {
+ // Ensure that buffer alignment matches channelcount
+ if (((uint32_t)sharedBuffer->pointer() & (channelCount | 1)) != 0) {
+ LOGE("Invalid buffer alignement: address %p, channelCount %d", sharedBuffer->pointer(), channelCount);
+ return BAD_VALUE;
+ }
+ frameCount = sharedBuffer->size()/channelCount/sizeof(int16_t);
+ }
}
- // create the track
- status_t status;
- sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
- streamType, sampleRate, format, channelCount, frameCount, flags, sharedBuffer, &status);
+ mVolume[LEFT] = 1.0f;
+ mVolume[RIGHT] = 1.0f;
+ // create the IAudioTrack
+ status_t status = createTrack(streamType, sampleRate, format, channelCount,
+ frameCount, flags, sharedBuffer, output);
- if (track == 0) {
- LOGE("AudioFlinger could not create track, status: %d", status);
+ if (status != NO_ERROR) {
return status;
}
- sp<IMemory> cblk = track->getCblk();
- if (cblk == 0) {
- LOGE("Could not get control block");
- return NO_INIT;
- }
+
if (cbf != 0) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
if (mAudioTrackThread == 0) {
@@ -227,24 +232,9 @@ status_t AudioTrack::set(
mStatus = NO_ERROR;
- mAudioTrack = track;
- mCblkMemory = cblk;
- mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
- mCblk->out = 1;
- // Update buffer size in case it has been limited by AudioFlinger during track creation
- mFrameCount = mCblk->frameCount;
- if (sharedBuffer == 0) {
- mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
- } else {
- mCblk->buffers = sharedBuffer->pointer();
- // Force buffer full condition as data is already present in shared memory
- mCblk->stepUser(mFrameCount);
- }
- mCblk->volume[0] = mCblk->volume[1] = 0x1000;
- mVolume[LEFT] = 1.0f;
- mVolume[RIGHT] = 1.0f;
mStreamType = streamType;
mFormat = format;
+ mChannels = channels;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
@@ -259,6 +249,7 @@ status_t AudioTrack::set(
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
+ mFlags = flags;
return NO_ERROR;
}
@@ -297,7 +288,11 @@ uint32_t AudioTrack::frameCount() const
int AudioTrack::frameSize() const
{
- return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+ if (AudioSystem::isLinearPCM(mFormat)) {
+ return channelCount()*((format() == AudioSystem::PCM_8_BIT) ? sizeof(uint8_t) : sizeof(int16_t));
+ } else {
+ return sizeof(uint8_t);
+ }
}
sp<IMemory>& AudioTrack::sharedBuffer()
@@ -323,15 +318,27 @@ void AudioTrack::start()
}
if (android_atomic_or(1, &mActive) == 0) {
- mNewPosition = mCblk->server + mUpdatePeriod;
- mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
- mCblk->waitTimeMs = 0;
- if (t != 0) {
- t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
+ audio_io_handle_t output = AudioTrack::getOutput();
+ status_t status = mAudioTrack->start();
+ if (status == DEAD_OBJECT) {
+ LOGV("start() dead IAudioTrack: creating a new one");
+ status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
+ mFrameCount, mFlags, mSharedBuffer, output);
+ }
+ if (status == NO_ERROR) {
+ AudioSystem::startOutput(output, (AudioSystem::stream_type)mStreamType);
+ mNewPosition = mCblk->server + mUpdatePeriod;
+ mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
+ mCblk->waitTimeMs = 0;
+ if (t != 0) {
+ t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
+ } else {
+ setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+ }
} else {
- setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
+ LOGV("start() failed");
+ android_atomic_and(~1, &mActive);
}
- mAudioTrack->start();
}
if (t != 0) {
@@ -367,6 +374,7 @@ void AudioTrack::stop()
} else {
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
}
+ AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
}
if (t != 0) {
@@ -382,12 +390,12 @@ bool AudioTrack::stopped() const
void AudioTrack::flush()
{
LOGV("flush");
-
+
// clear playback marker and periodic update counter
mMarkerPosition = 0;
mMarkerReached = false;
mUpdatePeriod = 0;
-
+
if (!mActive) {
mAudioTrack->flush();
@@ -403,6 +411,7 @@ void AudioTrack::pause()
if (android_atomic_and(~1, &mActive) == 1) {
mActive = 0;
mAudioTrack->pause();
+ AudioSystem::stopOutput(getOutput(), (AudioSystem::stream_type)mStreamType);
}
}
@@ -455,7 +464,6 @@ status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount
{
audio_track_cblk_t* cblk = mCblk;
-
Mutex::Autolock _l(cblk->lock);
if (loopCount == 0) {
@@ -476,7 +484,7 @@ status_t AudioTrack::setLoop(uint32_t loopStart, uint32_t loopEnd, int loopCount
LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
loopStart, loopEnd, mFrameCount);
return BAD_VALUE;
- }
+ }
cblk->loopStart = loopStart;
cblk->loopEnd = loopEnd;
@@ -555,7 +563,7 @@ status_t AudioTrack::setPosition(uint32_t position)
mCblk->server = position;
mCblk->forceReady = 1;
-
+
return NO_ERROR;
}
@@ -571,7 +579,7 @@ status_t AudioTrack::getPosition(uint32_t *position)
status_t AudioTrack::reload()
{
if (!stopped()) return INVALID_OPERATION;
-
+
flush();
mCblk->stepUser(mFrameCount);
@@ -579,12 +587,75 @@ status_t AudioTrack::reload()
return NO_ERROR;
}
+audio_io_handle_t AudioTrack::getOutput()
+{
+ return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType,
+ mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
+}
+
// -------------------------------------------------------------------------
+status_t AudioTrack::createTrack(
+ int streamType,
+ uint32_t sampleRate,
+ int format,
+ int channelCount,
+ int frameCount,
+ uint32_t flags,
+ const sp<IMemory>& sharedBuffer,
+ audio_io_handle_t output)
+{
+ status_t status;
+ const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
+ if (audioFlinger == 0) {
+ LOGE("Could not get audioflinger");
+ return NO_INIT;
+ }
+
+ sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
+ streamType,
+ sampleRate,
+ format,
+ channelCount,
+ frameCount,
+ ((uint16_t)flags) << 16,
+ sharedBuffer,
+ output,
+ &status);
+
+ if (track == 0) {
+ LOGE("AudioFlinger could not create track, status: %d", status);
+ return status;
+ }
+ sp<IMemory> cblk = track->getCblk();
+ if (cblk == 0) {
+ LOGE("Could not get control block");
+ return NO_INIT;
+ }
+ mAudioTrack.clear();
+ mAudioTrack = track;
+ mCblkMemory.clear();
+ mCblkMemory = cblk;
+ mCblk = static_cast<audio_track_cblk_t*>(cblk->pointer());
+ mCblk->out = 1;
+ // Update buffer size in case it has been limited by AudioFlinger during track creation
+ mFrameCount = mCblk->frameCount;
+ if (sharedBuffer == 0) {
+ mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t);
+ } else {
+ mCblk->buffers = sharedBuffer->pointer();
+ // Force buffer full condition as data is already present in shared memory
+ mCblk->stepUser(mFrameCount);
+ }
+
+ mCblk->volumeLR = (int32_t(int16_t(mVolume[LEFT] * 0x1000)) << 16) | int16_t(mVolume[RIGHT] * 0x1000);
+
+ return NO_ERROR;
+}
+
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
int active;
- int timeout = 0;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
@@ -596,19 +667,22 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
uint32_t framesAvail = cblk->framesAvailable();
if (framesAvail == 0) {
- Mutex::Autolock _l(cblk->lock);
+ cblk->lock.lock();
goto start_loop_here;
while (framesAvail == 0) {
active = mActive;
if (UNLIKELY(!active)) {
LOGV("Not active and NO_MORE_BUFFERS");
+ cblk->lock.unlock();
return NO_MORE_BUFFERS;
}
- if (UNLIKELY(!waitCount))
+ if (UNLIKELY(!waitCount)) {
+ cblk->lock.unlock();
return WOULD_BLOCK;
- timeout = 0;
+ }
+
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
- if (__builtin_expect(result!=NO_ERROR, false)) {
+ if (__builtin_expect(result!=NO_ERROR, false)) {
cblk->waitTimeMs += waitTimeMs;
if (cblk->waitTimeMs >= cblk->bufferTimeoutMs) {
// timing out when a loop has been set and we have already written upto loop end
@@ -616,16 +690,25 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
if (cblk->user < cblk->loopEnd) {
LOGW( "obtainBuffer timed out (is the CPU pegged?) %p "
"user=%08x, server=%08x", this, cblk->user, cblk->server);
- //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
+ //unlock cblk mutex before calling mAudioTrack->start() (see issue #1617140)
cblk->lock.unlock();
- mAudioTrack->start();
+ result = mAudioTrack->start();
+ if (result == DEAD_OBJECT) {
+ LOGW("obtainBuffer() dead IAudioTrack: creating a new one");
+ result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
+ mFrameCount, mFlags, mSharedBuffer, getOutput());
+ if (result == NO_ERROR) {
+ cblk = mCblk;
+ cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
+ }
+ }
cblk->lock.lock();
- timeout = 1;
}
cblk->waitTimeMs = 0;
}
-
+
if (--waitCount == 0) {
+ cblk->lock.unlock();
return TIMED_OUT;
}
}
@@ -633,10 +716,11 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
start_loop_here:
framesAvail = cblk->framesAvailable_l();
}
+ cblk->lock.unlock();
}
cblk->waitTimeMs = 0;
-
+
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
@@ -648,17 +732,16 @@ status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
framesReq = bufferEnd - u;
}
- LOGW_IF(timeout,
- "*** SERIOUS WARNING *** obtainBuffer() timed out "
- "but didn't need to be locked. We recovered, but "
- "this shouldn't happen (user=%08x, server=%08x)", cblk->user, cblk->server);
-
- audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
- audioBuffer->channelCount= mChannelCount;
- audioBuffer->format = AudioSystem::PCM_16_BIT;
- audioBuffer->frameCount = framesReq;
- audioBuffer->size = framesReq*mChannelCount*sizeof(int16_t);
- audioBuffer->raw = (int8_t *)cblk->buffer(u);
+ audioBuffer->flags = mMuted ? Buffer::MUTE : 0;
+ audioBuffer->channelCount = mChannelCount;
+ audioBuffer->frameCount = framesReq;
+ audioBuffer->size = framesReq * cblk->frameSize;
+ if (AudioSystem::isLinearPCM(mFormat)) {
+ audioBuffer->format = AudioSystem::PCM_16_BIT;
+ } else {
+ audioBuffer->format = mFormat;
+ }
+ audioBuffer->raw = (int8_t *)cblk->buffer(u);
active = mActive;
return active ? status_t(NO_ERROR) : status_t(STOPPED);
}
@@ -690,10 +773,8 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize)
Buffer audioBuffer;
do {
- audioBuffer.frameCount = userSize/mChannelCount;
- if (mFormat == AudioSystem::PCM_16_BIT) {
- audioBuffer.frameCount >>= 1;
- }
+ audioBuffer.frameCount = userSize/frameSize();
+
// Calling obtainBuffer() with a negative wait count causes
// an (almost) infinite wait time.
status_t err = obtainBuffer(&audioBuffer, -1);
@@ -705,7 +786,8 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize)
}
size_t toWrite;
- if (mFormat == AudioSystem::PCM_8_BIT) {
+
+ if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
toWrite = audioBuffer.size>>1;
// 8 to 16 bit conversion
@@ -714,7 +796,7 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize)
while(count--) {
*dst++ = (int16_t)(*src++^0x80) << 8;
}
- }else {
+ } else {
toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, src, toWrite);
src += toWrite;
@@ -742,13 +824,13 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
if (mCblk->flowControlFlag == 0) {
mCbf(EVENT_UNDERRUN, mUserData, 0);
if (mCblk->server == mCblk->frameCount) {
- mCbf(EVENT_BUFFER_END, mUserData, 0);
+ mCbf(EVENT_BUFFER_END, mUserData, 0);
}
mCblk->flowControlFlag = 1;
if (mSharedBuffer != 0) return false;
}
}
-
+
// Manage loop end callback
while (mLoopCount > mCblk->loopCount) {
int loopCount = -1;
@@ -767,7 +849,7 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
}
// Manage new position callback
- if(mUpdatePeriod > 0) {
+ if (mUpdatePeriod > 0) {
while (mCblk->server >= mNewPosition) {
mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
mNewPosition += mUpdatePeriod;
@@ -784,10 +866,10 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
do {
audioBuffer.frameCount = frames;
-
- // Calling obtainBuffer() with a wait count of 1
- // limits wait time to WAIT_PERIOD_MS. This prevents from being
- // stuck here not being able to handle timed events (position, markers, loops).
+
+ // Calling obtainBuffer() with a wait count of 1
+ // limits wait time to WAIT_PERIOD_MS. This prevents from being
+ // stuck here not being able to handle timed events (position, markers, loops).
status_t err = obtainBuffer(&audioBuffer, 1);
if (err < NO_ERROR) {
if (err != TIMED_OUT) {
@@ -801,7 +883,7 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
// Divide buffer size by 2 to take into account the expansion
// due to 8 to 16 bit conversion: the callback must fill only half
// of the destination buffer
- if (mFormat == AudioSystem::PCM_8_BIT) {
+ if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
audioBuffer.size >>= 1;
}
@@ -820,7 +902,7 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
}
if (writtenSize > reqSize) writtenSize = reqSize;
- if (mFormat == AudioSystem::PCM_8_BIT) {
+ if (mFormat == AudioSystem::PCM_8_BIT && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
// 8 to 16 bit conversion
const int8_t *src = audioBuffer.i8 + writtenSize-1;
int count = writtenSize;
@@ -832,7 +914,11 @@ bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
}
audioBuffer.size = writtenSize;
- audioBuffer.frameCount = writtenSize/mChannelCount/sizeof(int16_t);
+ // NOTE: mCblk->frameSize is not equal to AudioTrack::frameSize() for
+ // 8 bit PCM data: in this case, mCblk->frameSize is based on a sampel size of
+ // 16 bit.
+ audioBuffer.frameCount = writtenSize/mCblk->frameSize;
+
frames -= audioBuffer.frameCount;
releaseBuffer(&audioBuffer);
@@ -891,7 +977,7 @@ void AudioTrack::AudioTrackThread::onFirstRef()
// =========================================================================
audio_track_cblk_t::audio_track_cblk_t()
- : user(0), server(0), userBase(0), serverBase(0), buffers(0), frameCount(0),
+ : lock(Mutex::SHARED), user(0), server(0), userBase(0), serverBase(0), buffers(0), frameCount(0),
loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0), flowControlFlag(1), forceReady(0)
{
}
@@ -948,8 +1034,8 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount)
// Mark that we have read the first buffer so that next time stepUser() is called
// we switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS) {
- bufferTimeoutMs = MAX_RUN_TIMEOUT_MS - 1;
- }
+ bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS - 1;
+ }
// It is possible that we receive a flush()
// while the mixer is processing a block: in this case,
// stepServer() is called After the flush() has reset u & s and
@@ -981,7 +1067,7 @@ bool audio_track_cblk_t::stepServer(uint32_t frameCount)
void* audio_track_cblk_t::buffer(uint32_t offset) const
{
- return (int16_t *)this->buffers + (offset-userBase)*this->channels;
+ return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
}
uint32_t audio_track_cblk_t::framesAvailable()
generated by cgit v1.2.3-59-g8ed1b (git 2.39.0) at 2025-11-25 09:33:58 +0000