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M7350/base/media/libmedia/AudioTrack.cpp
T f9cc65cfda M7350v1_en_gpl
2024-09-09 08:52:07 +00:00

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/* //device/extlibs/pv/android/AudioTrack.cpp
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
** http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/
//#define LOG_NDEBUG 0
#define LOG_NDDEBUG 0
#define LOG_TAG "AudioTrack"
#include <stdint.h>
#include <sys/types.h>
#include <limits.h>
#include <sched.h>
#include <sys/resource.h>
#include <private/media/AudioTrackShared.h>
#include <media/AudioSystem.h>
#include <media/AudioTrack.h>
#include <utils/Log.h>
#include <binder/Parcel.h>
#include <binder/IPCThreadState.h>
#include <utils/Timers.h>
#include <cutils/atomic.h>
#define LIKELY( exp ) (__builtin_expect( (exp) != 0, true ))
#define UNLIKELY( exp ) (__builtin_expect( (exp) != 0, false ))
namespace android {
// ---------------------------------------------------------------------------
// static
status_t AudioTrack::getMinFrameCount(
int* frameCount,
int streamType,
uint32_t sampleRate)
{
if(streamType == AudioSystem::VOICE_CALL) {
LOGV("AudioTrack :: getMinFramecount voice call \n");
if(sampleRate == 8000) {
*frameCount = 160;
} else if (sampleRate == 16000) {
*frameCount = 320;
}
return NO_ERROR;
}
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
int afFrameCount;
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount) / afSampleRate);
if (minBufCount < 2) minBufCount = 2;
*frameCount = (sampleRate == 0) ? afFrameCount * minBufCount :
afFrameCount * minBufCount * sampleRate / afSampleRate;
return NO_ERROR;
}
// ---------------------------------------------------------------------------
AudioTrack::AudioTrack()
: mStatus(NO_INIT)
{
}
AudioTrack::AudioTrack(
int streamType,
uint32_t sampleRate,
int format,
int channels,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
: mStatus(NO_INIT)
{
mStatus = set(streamType, sampleRate, format, channels,
frameCount, flags, cbf, user, notificationFrames,
0, false, sessionId);
}
AudioTrack::AudioTrack(
int streamType,
uint32_t sampleRate,
int format,
int channels,
const sp<IMemory>& sharedBuffer,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
int sessionId)
: mStatus(NO_INIT)
{
mStatus = set(streamType, sampleRate, format, channels,
0, flags, cbf, user, notificationFrames,
sharedBuffer, false, sessionId);
}
AudioTrack::AudioTrack(
int streamType,
uint32_t sampleRate,
int format,
int channels,
uint32_t flags,
int sessionId,
int lpaSessionId)
: mStatus(NO_INIT), mAudioSession(-1)
{
mStatus = set(streamType, sampleRate, format, channels, flags, sessionId, lpaSessionId);
}
AudioTrack::~AudioTrack()
{
LOGV("AudioTrack dtor");
LOGV_IF(mSharedBuffer != 0, "Destructor sharedBuffer: %p", mSharedBuffer->pointer());
if (mStatus == NO_ERROR) {
// Make sure that callback function exits in the case where
// it is looping on buffer full condition in obtainBuffer().
// Otherwise the callback thread will never exit.
stop();
if (mAudioTrackThread != 0) {
mAudioTrackThread->requestExitAndWait();
mAudioTrackThread.clear();
}
if(mAudioTrack != NULL) {
mAudioTrack.clear();
}
if(mAudioSession >= 0) {
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger != 0) {
status_t status;
LOGV("Calling AudioFlinger::deleteSession");
audioFlinger->deleteSession();
} else {
LOGE("Could not get audioflinger");
}
AudioSystem::closeSession(mAudioSession);
mAudioSession = -1;
}
IPCThreadState::self()->flushCommands();
}
}
status_t AudioTrack::set(
int streamType,
uint32_t sampleRate,
int format,
int channels,
int frameCount,
uint32_t flags,
callback_t cbf,
void* user,
int notificationFrames,
const sp<IMemory>& sharedBuffer,
bool threadCanCallJava,
int sessionId)
{
LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
if (mAudioTrack != 0) {
LOGE("Track already in use");
return INVALID_OPERATION;
}
int afSampleRate;
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}
// handle default values first.
if (streamType == AudioSystem::DEFAULT) {
streamType = AudioSystem::MUSIC;
}
if (sampleRate == 0) {
sampleRate = afSampleRate;
}
// these below should probably come from the audioFlinger too...
if (format == 0) {
format = AudioSystem::PCM_16_BIT;
}
if (channels == 0) {
channels = AudioSystem::CHANNEL_OUT_STEREO;
}
// validate parameters
if (!AudioSystem::isValidFormat(format)) {
LOGE("Invalid format");
return BAD_VALUE;
}
// 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);
audio_io_handle_t output = AudioSystem::getOutput((AudioSystem::stream_type)streamType,
sampleRate, format, channels, (AudioSystem::output_flags)flags);
if (output == 0) {
LOGE("Could not get audio output for stream type %d", streamType);
return BAD_VALUE;
}
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mSendLevel = 0;
mFrameCount = frameCount;
mNotificationFramesReq = notificationFrames;
mSessionId = sessionId;
mAuxEffectId = 0;
// create the IAudioTrack
status_t status = createTrack(streamType, sampleRate, format, channelCount,
frameCount, flags, sharedBuffer, output, true);
if (status != NO_ERROR) {
return status;
}
if (cbf != 0) {
mAudioTrackThread = new AudioTrackThread(*this, threadCanCallJava);
if (mAudioTrackThread == 0) {
LOGE("Could not create callback thread");
return NO_INIT;
}
}
mStatus = NO_ERROR;
mStreamType = streamType;
mFormat = format;
mChannels = channels;
mChannelCount = channelCount;
mSharedBuffer = sharedBuffer;
mMuted = false;
mActive = 0;
mCbf = cbf;
mUserData = user;
mLoopCount = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mFlags = flags;
mAudioSession = -1;
return NO_ERROR;
}
status_t AudioTrack::set(
int streamType,
uint32_t sampleRate,
int format,
int channels,
uint32_t flags,
int sessionId,
int lpaSessionId)
{
// handle default values first.
if (streamType == AudioSystem::DEFAULT) {
streamType = AudioSystem::MUSIC;
}
// these below should probably come from the audioFlinger too...
if (format == 0) {
format = AudioSystem::PCM_16_BIT;
}
// validate parameters
if (!AudioSystem::isValidFormat(format)) {
LOGE("Invalid format");
return BAD_VALUE;
}
// force direct flag if format is not linear PCM
if (!AudioSystem::isLinearPCM(format)) {
flags |= AudioSystem::OUTPUT_FLAG_DIRECT;
}
audio_io_handle_t output = AudioSystem::getSession((AudioSystem::stream_type)streamType,
format, (AudioSystem::output_flags)flags, lpaSessionId);
if (output == 0) {
LOGE("Could not get audio output for stream type %d", streamType);
return BAD_VALUE;
}
mVolume[LEFT] = 1.0f;
mVolume[RIGHT] = 1.0f;
mStatus = NO_ERROR;
mStreamType = streamType;
mFormat = format;
mChannels = 2;
mChannelCount = 2;
mSharedBuffer = NULL;
mMuted = false;
mActive = 0;
mCbf = NULL;
mNotificationFramesReq = 0;
mRemainingFrames = 0;
mUserData = NULL;
mLatency = 0;
mLoopCount = 0;
mMarkerPosition = 0;
mMarkerReached = false;
mNewPosition = 0;
mUpdatePeriod = 0;
mFlags = flags;
mAudioTrack = NULL;
mAudioSession = output;
mSessionId = sessionId;
mAuxEffectId = 0;
const sp<IAudioFlinger>& audioFlinger = AudioSystem::get_audio_flinger();
if (audioFlinger == 0) {
LOGE("Could not get audioflinger");
return NO_INIT;
}
status_t status;
audioFlinger->createSession(getpid(),
sampleRate,
channels,
&mSessionId,
&status);
if(status != NO_ERROR) {
LOGE("createSession returned with status %d", status);
}
/* Make the track active and start output */
android_atomic_or(1, &mActive);
AudioSystem::startOutput(output, (AudioSystem::stream_type)mStreamType);
LOGV("AudioTrack::set() - Started output(%d)",output);
return NO_ERROR;
}
status_t AudioTrack::initCheck() const
{
return mStatus;
}
// -------------------------------------------------------------------------
uint32_t AudioTrack::latency() const
{
return mLatency;
}
int AudioTrack::streamType() const
{
return mStreamType;
}
int AudioTrack::format() const
{
return mFormat;
}
int AudioTrack::channelCount() const
{
return mChannelCount;
}
uint32_t AudioTrack::frameCount() const
{
return mCblk->frameCount;
}
int AudioTrack::frameSize() const
{
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()
{
return mSharedBuffer;
}
// -------------------------------------------------------------------------
void AudioTrack::start()
{
if ( mAudioSession != -1 ) {
if ( NO_ERROR != AudioSystem::resumeSession(mAudioSession,
(AudioSystem::stream_type)mStreamType) )
{
LOGE("ResumeSession failed");
}
return;
}
sp<AudioTrackThread> t = mAudioTrackThread;
status_t status;
LOGD("start %p", this);
if (t != 0) {
if (t->exitPending()) {
if (t->requestExitAndWait() == WOULD_BLOCK) {
LOGE("AudioTrack::start called from thread");
return;
}
}
t->mLock.lock();
}
if (android_atomic_or(1, &mActive) == 0) {
mNewPosition = mCblk->server + mUpdatePeriod;
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
mCblk->flags &= ~CBLK_DISABLED_ON;
if (t != 0) {
t->run("AudioTrackThread", THREAD_PRIORITY_AUDIO_CLIENT);
} else {
setpriority(PRIO_PROCESS, 0, THREAD_PRIORITY_AUDIO_CLIENT);
}
if (mCblk->flags & CBLK_INVALID_MSK) {
LOGE("start() track %p invalidated, creating a new one", this);
// no need to clear the invalid flag as this cblk will not be used anymore
// force new track creation
status = DEAD_OBJECT;
} else {
status = mAudioTrack->start();
}
if (status == DEAD_OBJECT) {
LOGE("start() dead IAudioTrack: creating a new one");
mSessionId = 0;
status = createTrack(mStreamType, mCblk->sampleRate, mFormat, mChannelCount,
mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
if (status == NO_ERROR) {
status = mAudioTrack->start();
if (status == NO_ERROR) {
mNewPosition = mCblk->server + mUpdatePeriod;
}
}
}
if (status != NO_ERROR) {
LOGV("start() failed");
android_atomic_and(~1, &mActive);
if (t != 0) {
t->requestExit();
} else {
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
}
}
}
if (t != 0) {
t->mLock.unlock();
}
}
void AudioTrack::stop()
{
sp<AudioTrackThread> t = mAudioTrackThread;
LOGD("stop %p", this);
if (t != 0) {
t->mLock.lock();
}
if (android_atomic_and(~1, &mActive) == 1) {
if(mAudioTrack != NULL) {
mCblk->cv.signal();
mAudioTrack->stop();
// Cancel loops (If we are in the middle of a loop, playback
// would not stop until loopCount reaches 0).
setLoop(0, 0, 0);
// the playback head position will reset to 0, so if a marker is set, we need
// to activate it again
mMarkerReached = false;
// Force flush if a shared buffer is used otherwise audioflinger
// will not stop before end of buffer is reached.
if (mSharedBuffer != 0) {
flush();
}
if (t != 0) {
t->requestExit();
} else {
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_NORMAL);
}
}
}
if (t != 0) {
t->mLock.unlock();
}
}
bool AudioTrack::stopped() const
{
return !mActive;
}
void AudioTrack::flush()
{
LOGV("flush");
// clear playback marker and periodic update counter
mMarkerPosition = 0;
mMarkerReached = false;
mUpdatePeriod = 0;
if (!mActive) {
mAudioTrack->flush();
// Release AudioTrack callback thread in case it was waiting for new buffers
// in AudioTrack::obtainBuffer()
mCblk->cv.signal();
}
}
void AudioTrack::pause()
{
LOGD("pause");
if ( mAudioSession != -1 ) {
if ( NO_ERROR != AudioSystem::pauseSession(mAudioSession,
(AudioSystem::stream_type)mStreamType) )
{
LOGE("PauseSession failed");
}
return;
}
if (android_atomic_and(~1, &mActive) == 1) {
mAudioTrack->pause();
}
}
void AudioTrack::mute(bool e)
{
mAudioTrack->mute(e);
mMuted = e;
}
bool AudioTrack::muted() const
{
return mMuted;
}
status_t AudioTrack::setVolume(float left, float right)
{
if (left > 1.0f || right > 1.0f) {
return BAD_VALUE;
}
mVolume[LEFT] = left;
mVolume[RIGHT] = right;
// write must be atomic
mCblk->volumeLR = (uint32_t(uint16_t(right * 0x1000)) << 16) | uint16_t(left * 0x1000);
return NO_ERROR;
}
void AudioTrack::getVolume(float* left, float* right)
{
if (left != NULL) {
*left = mVolume[LEFT];
}
if (right != NULL) {
*right = mVolume[RIGHT];
}
}
status_t AudioTrack::setAuxEffectSendLevel(float level)
{
LOGV("setAuxEffectSendLevel(%f)", level);
if (level > 1.0f) {
return BAD_VALUE;
}
mSendLevel = level;
mCblk->sendLevel = uint16_t(level * 0x1000);
return NO_ERROR;
}
void AudioTrack::getAuxEffectSendLevel(float* level)
{
if (level != NULL) {
*level = mSendLevel;
}
}
status_t AudioTrack::setSampleRate(int rate)
{
int afSamplingRate;
if (AudioSystem::getOutputSamplingRate(&afSamplingRate, mStreamType) != NO_ERROR) {
return NO_INIT;
}
// Resampler implementation limits input sampling rate to 2 x output sampling rate.
if (rate <= 0 || rate > afSamplingRate*2 ) return BAD_VALUE;
mCblk->sampleRate = rate;
return NO_ERROR;
}
uint32_t AudioTrack::getSampleRate()
{
return mCblk->sampleRate;
}
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) {
cblk->loopStart = UINT_MAX;
cblk->loopEnd = UINT_MAX;
cblk->loopCount = 0;
mLoopCount = 0;
return NO_ERROR;
}
if (loopStart >= loopEnd ||
loopEnd - loopStart > cblk->frameCount) {
LOGE("setLoop invalid value: loopStart %d, loopEnd %d, loopCount %d, framecount %d, user %d", loopStart, loopEnd, loopCount, cblk->frameCount, cblk->user);
return BAD_VALUE;
}
if ((mSharedBuffer != 0) && (loopEnd > cblk->frameCount)) {
LOGE("setLoop invalid value: loop markers beyond data: loopStart %d, loopEnd %d, framecount %d",
loopStart, loopEnd, cblk->frameCount);
return BAD_VALUE;
}
cblk->loopStart = loopStart;
cblk->loopEnd = loopEnd;
cblk->loopCount = loopCount;
mLoopCount = loopCount;
return NO_ERROR;
}
status_t AudioTrack::getLoop(uint32_t *loopStart, uint32_t *loopEnd, int *loopCount)
{
if (loopStart != 0) {
*loopStart = mCblk->loopStart;
}
if (loopEnd != 0) {
*loopEnd = mCblk->loopEnd;
}
if (loopCount != 0) {
if (mCblk->loopCount < 0) {
*loopCount = -1;
} else {
*loopCount = mCblk->loopCount;
}
}
return NO_ERROR;
}
status_t AudioTrack::setMarkerPosition(uint32_t marker)
{
if (mCbf == 0) return INVALID_OPERATION;
mMarkerPosition = marker;
mMarkerReached = false;
return NO_ERROR;
}
status_t AudioTrack::getMarkerPosition(uint32_t *marker)
{
if (marker == 0) return BAD_VALUE;
*marker = mMarkerPosition;
return NO_ERROR;
}
status_t AudioTrack::setPositionUpdatePeriod(uint32_t updatePeriod)
{
if (mCbf == 0) return INVALID_OPERATION;
uint32_t curPosition;
getPosition(&curPosition);
mNewPosition = curPosition + updatePeriod;
mUpdatePeriod = updatePeriod;
return NO_ERROR;
}
status_t AudioTrack::getPositionUpdatePeriod(uint32_t *updatePeriod)
{
if (updatePeriod == 0) return BAD_VALUE;
*updatePeriod = mUpdatePeriod;
return NO_ERROR;
}
status_t AudioTrack::setPosition(uint32_t position)
{
Mutex::Autolock _l(mCblk->lock);
if (!stopped()) return INVALID_OPERATION;
if (position > mCblk->user) return BAD_VALUE;
mCblk->server = position;
mCblk->flags |= CBLK_FORCEREADY_ON;
return NO_ERROR;
}
status_t AudioTrack::getPosition(uint32_t *position)
{
if (position == 0) return BAD_VALUE;
*position = mCblk->server;
return NO_ERROR;
}
status_t AudioTrack::reload()
{
if (!stopped()) return INVALID_OPERATION;
flush();
mCblk->stepUser(mCblk->frameCount);
return NO_ERROR;
}
audio_io_handle_t AudioTrack::getOutput()
{
return AudioSystem::getOutput((AudioSystem::stream_type)mStreamType,
mCblk->sampleRate, mFormat, mChannels, (AudioSystem::output_flags)mFlags);
}
int AudioTrack::getSessionId()
{
return mSessionId;
}
status_t AudioTrack::attachAuxEffect(int effectId)
{
LOGV("attachAuxEffect(%d)", effectId);
status_t status = mAudioTrack->attachAuxEffect(effectId);
if (status == NO_ERROR) {
mAuxEffectId = effectId;
}
return status;
}
// -------------------------------------------------------------------------
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,
bool enforceFrameCount)
{
status_t status;
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;
}
int afFrameCount;
if (AudioSystem::getOutputFrameCount(&afFrameCount, streamType) != NO_ERROR) {
return NO_INIT;
}
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
}
mNotificationFramesAct = mNotificationFramesReq;
if (!AudioSystem::isLinearPCM(format)) {
if (sharedBuffer != 0) {
frameCount = sharedBuffer->size();
}
} else {
// Ensure that buffer depth covers at least audio hardware latency
uint32_t minBufCount = afLatency / ((1000 * afFrameCount)/afSampleRate);
if (minBufCount < 2) minBufCount = 2;
int minFrameCount = (afFrameCount*sampleRate*minBufCount)/afSampleRate;
if (sharedBuffer == 0) {
if (frameCount == 0) {
frameCount = minFrameCount;
}
if (mNotificationFramesAct == 0) {
mNotificationFramesAct = frameCount/2;
}
// Make sure that application is notified with sufficient margin
// before underrun
if (mNotificationFramesAct > (uint32_t)frameCount/2) {
mNotificationFramesAct = frameCount/2;
}
if (frameCount < minFrameCount) {
if (enforceFrameCount) {
LOGE("Invalid buffer size: minFrameCount %d, frameCount %d", minFrameCount, frameCount);
return BAD_VALUE;
} else {
frameCount = minFrameCount;
}
}
} 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);
}
}
LOGD("Request AudioFlinger to create track");
sp<IAudioTrack> track = audioFlinger->createTrack(getpid(),
streamType,
sampleRate,
format,
channelCount,
frameCount,
((uint16_t)flags) << 16,
sharedBuffer,
output,
&mSessionId,
&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->flags |= CBLK_DIRECTION_OUT;
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(mCblk->frameCount);
}
mCblk->volumeLR = (uint32_t(uint16_t(mVolume[RIGHT] * 0x1000)) << 16) | uint16_t(mVolume[LEFT] * 0x1000);
mCblk->sendLevel = uint16_t(mSendLevel * 0x1000);
mAudioTrack->attachAuxEffect(mAuxEffectId);
mCblk->bufferTimeoutMs = MAX_STARTUP_TIMEOUT_MS;
mCblk->waitTimeMs = 0;
mRemainingFrames = mNotificationFramesAct;
mLatency = afLatency + (1000*mCblk->frameCount) / sampleRate;
return NO_ERROR;
}
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, int32_t waitCount)
{
int active;
status_t result;
audio_track_cblk_t* cblk = mCblk;
uint32_t framesReq = audioBuffer->frameCount;
uint32_t waitTimeMs = (waitCount < 0) ? cblk->bufferTimeoutMs : WAIT_PERIOD_MS;
audioBuffer->frameCount = 0;
audioBuffer->size = 0;
uint32_t framesAvail = cblk->framesAvailable();
if (framesAvail == 0) {
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)) {
cblk->lock.unlock();
return WOULD_BLOCK;
}
if (!(cblk->flags & CBLK_INVALID_MSK)) {
result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs));
}
if (cblk->flags & CBLK_INVALID_MSK) {
LOGW("obtainBuffer() track %p invalidated, creating a new one", this);
// no need to clear the invalid flag as this cblk will not be used anymore
cblk->lock.unlock();
goto create_new_track;
}
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
// is a normal condition: no need to wake AudioFlinger up.
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)
cblk->lock.unlock();
result = mAudioTrack->start();
if (result == DEAD_OBJECT) {
LOGE("obtainBuffer() dead IAudioTrack: creating a new one");
create_new_track:
mSessionId = 0;
result = createTrack(mStreamType, cblk->sampleRate, mFormat, mChannelCount,
mFrameCount, mFlags, mSharedBuffer, getOutput(), false);
if (result == NO_ERROR) {
cblk = mCblk;
cblk->bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
mAudioTrack->start();
}
}
cblk->lock.lock();
}
cblk->waitTimeMs = 0;
}
if (--waitCount == 0) {
cblk->lock.unlock();
return TIMED_OUT;
}
}
// read the server count again
start_loop_here:
framesAvail = cblk->framesAvailable_l();
}
cblk->lock.unlock();
}
// restart track if it was disabled by audioflinger due to previous underrun
if (cblk->flags & CBLK_DISABLED_MSK) {
cblk->flags &= ~CBLK_DISABLED_ON;
LOGW("obtainBuffer() track %p disabled, restarting", this);
mAudioTrack->start();
}
cblk->waitTimeMs = 0;
if (framesReq > framesAvail) {
framesReq = framesAvail;
}
uint32_t u = cblk->user;
uint32_t bufferEnd = cblk->userBase + cblk->frameCount;
if (u + framesReq > bufferEnd) {
framesReq = bufferEnd - 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);
}
void AudioTrack::releaseBuffer(Buffer* audioBuffer)
{
audio_track_cblk_t* cblk = mCblk;
cblk->stepUser(audioBuffer->frameCount);
}
// -------------------------------------------------------------------------
ssize_t AudioTrack::write(const void* buffer, size_t userSize)
{
if (mSharedBuffer != 0) return INVALID_OPERATION;
if (ssize_t(userSize) < 0) {
// sanity-check. user is most-likely passing an error code.
LOGE("AudioTrack::write(buffer=%p, size=%u (%d)",
buffer, userSize, userSize);
return BAD_VALUE;
}
LOGV("write %p: %d bytes, mActive=%d", this, userSize, mActive);
ssize_t written = 0;
const int8_t *src = (const int8_t *)buffer;
Buffer audioBuffer;
do {
audioBuffer.frameCount = userSize/frameSize();
// Calling obtainBuffer() with a negative wait count causes
// an (almost) infinite wait time.
status_t err = obtainBuffer(&audioBuffer, -1);
if (err < 0) {
// out of buffers, return #bytes written
if (err == status_t(NO_MORE_BUFFERS))
break;
return ssize_t(err);
}
size_t toWrite;
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
int count = toWrite;
int16_t *dst = (int16_t *)(audioBuffer.i8);
while(count--) {
*dst++ = (int16_t)(*src++^0x80) << 8;
}
} else {
toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, src, toWrite);
src += toWrite;
}
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
} while (userSize);
return written;
}
// -------------------------------------------------------------------------
bool AudioTrack::processAudioBuffer(const sp<AudioTrackThread>& thread)
{
Buffer audioBuffer;
uint32_t frames;
size_t writtenSize;
// Manage underrun callback
if (mActive && (mCblk->framesReady() == 0)) {
LOGV("Underrun user: %x, server: %x, flags %04x", mCblk->user, mCblk->server, mCblk->flags);
if ((mCblk->flags & CBLK_UNDERRUN_MSK) == CBLK_UNDERRUN_OFF) {
mCbf(EVENT_UNDERRUN, mUserData, 0);
if (mCblk->server == mCblk->frameCount) {
mCbf(EVENT_BUFFER_END, mUserData, 0);
}
mCblk->flags |= CBLK_UNDERRUN_ON;
if (mSharedBuffer != 0) return false;
}
}
// Manage loop end callback
while (mLoopCount > mCblk->loopCount) {
int loopCount = -1;
mLoopCount--;
if (mLoopCount >= 0) loopCount = mLoopCount;
mCbf(EVENT_LOOP_END, mUserData, (void *)&loopCount);
}
// Manage marker callback
if (!mMarkerReached && (mMarkerPosition > 0)) {
if (mCblk->server >= mMarkerPosition) {
mCbf(EVENT_MARKER, mUserData, (void *)&mMarkerPosition);
mMarkerReached = true;
}
}
// Manage new position callback
if (mUpdatePeriod > 0) {
while (mCblk->server >= mNewPosition) {
mCbf(EVENT_NEW_POS, mUserData, (void *)&mNewPosition);
mNewPosition += mUpdatePeriod;
}
}
// If Shared buffer is used, no data is requested from client.
if (mSharedBuffer != 0) {
frames = 0;
} else {
frames = mRemainingFrames;
}
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).
status_t err = obtainBuffer(&audioBuffer, 1);
if (err < NO_ERROR) {
if (err != TIMED_OUT) {
LOGE_IF(err != status_t(NO_MORE_BUFFERS), "Error obtaining an audio buffer, giving up.");
return false;
}
break;
}
if (err == status_t(STOPPED)) return false;
// 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 && !(mFlags & AudioSystem::OUTPUT_FLAG_DIRECT)) {
audioBuffer.size >>= 1;
}
size_t reqSize = audioBuffer.size;
mCbf(EVENT_MORE_DATA, mUserData, &audioBuffer);
writtenSize = audioBuffer.size;
// Sanity check on returned size
if (ssize_t(writtenSize) <= 0) {
// The callback is done filling buffers
// Keep this thread going to handle timed events and
// still try to get more data in intervals of WAIT_PERIOD_MS
// but don't just loop and block the CPU, so wait
usleep(WAIT_PERIOD_MS*1000);
break;
}
if (writtenSize > reqSize) writtenSize = reqSize;
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;
int16_t *dst = audioBuffer.i16 + writtenSize-1;
while(count--) {
*dst-- = (int16_t)(*src--^0x80) << 8;
}
writtenSize <<= 1;
}
audioBuffer.size = writtenSize;
// 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);
}
while (frames);
if (frames == 0) {
mRemainingFrames = mNotificationFramesAct;
} else {
mRemainingFrames = frames;
}
return true;
}
status_t AudioTrack::dump(int fd, const Vector<String16>& args) const
{
const size_t SIZE = 256;
char buffer[SIZE];
String8 result;
result.append(" AudioTrack::dump\n");
snprintf(buffer, 255, " stream type(%d), left - right volume(%f, %f)\n", mStreamType, mVolume[0], mVolume[1]);
result.append(buffer);
snprintf(buffer, 255, " format(%d), channel count(%d), frame count(%d)\n", mFormat, mChannelCount, mCblk->frameCount);
result.append(buffer);
snprintf(buffer, 255, " sample rate(%d), status(%d), muted(%d)\n", (mCblk == 0) ? 0 : mCblk->sampleRate, mStatus, mMuted);
result.append(buffer);
snprintf(buffer, 255, " active(%d), latency (%d)\n", mActive, mLatency);
result.append(buffer);
::write(fd, result.string(), result.size());
return NO_ERROR;
}
// =========================================================================
AudioTrack::AudioTrackThread::AudioTrackThread(AudioTrack& receiver, bool bCanCallJava)
: Thread(bCanCallJava), mReceiver(receiver)
{
}
bool AudioTrack::AudioTrackThread::threadLoop()
{
return mReceiver.processAudioBuffer(this);
}
status_t AudioTrack::AudioTrackThread::readyToRun()
{
return NO_ERROR;
}
void AudioTrack::AudioTrackThread::onFirstRef()
{
}
// =========================================================================
audio_track_cblk_t::audio_track_cblk_t()
: lock(Mutex::SHARED), cv(Condition::SHARED), user(0), server(0),
userBase(0), serverBase(0), buffers(0), frameCount(0),
loopStart(UINT_MAX), loopEnd(UINT_MAX), loopCount(0), volumeLR(0),
flags(0), sendLevel(0)
{
}
uint32_t audio_track_cblk_t::stepUser(uint32_t frameCount)
{
uint32_t u = this->user;
u += frameCount;
// Ensure that user is never ahead of server for AudioRecord
if (flags & CBLK_DIRECTION_MSK) {
// If stepServer() has been called once, switch to normal obtainBuffer() timeout period
if (bufferTimeoutMs == MAX_STARTUP_TIMEOUT_MS-1) {
bufferTimeoutMs = MAX_RUN_TIMEOUT_MS;
}
} else if (u > this->server) {
LOGW("stepServer occured after track reset");
u = this->server;
}
if (u >= userBase + this->frameCount) {
userBase += this->frameCount;
}
this->user = u;
// Clear flow control error condition as new data has been written/read to/from buffer.
flags &= ~CBLK_UNDERRUN_MSK;
return u;
}
bool audio_track_cblk_t::stepServer(uint32_t frameCount)
{
// the code below simulates lock-with-timeout
// we MUST do this to protect the AudioFlinger server
// as this lock is shared with the client.
status_t err;
err = lock.tryLock();
if (err == -EBUSY) { // just wait a bit
usleep(1000);
err = lock.tryLock();
}
if (err != NO_ERROR) {
// probably, the client just died.
return false;
}
uint32_t s = this->server;
s += frameCount;
if (flags & CBLK_DIRECTION_MSK) {
// 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_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
// we have s > u
if (s > this->user) {
LOGW("stepServer occured after track reset");
s = this->user;
}
}
if (s >= loopEnd) {
LOGW_IF(s > loopEnd, "stepServer: s %u > loopEnd %u", s, loopEnd);
s = loopStart;
if (--loopCount == 0) {
loopEnd = UINT_MAX;
loopStart = UINT_MAX;
}
}
if (s >= serverBase + this->frameCount) {
serverBase += this->frameCount;
}
this->server = s;
cv.signal();
lock.unlock();
return true;
}
void* audio_track_cblk_t::buffer(uint32_t offset) const
{
return (int8_t *)this->buffers + (offset - userBase) * this->frameSize;
}
uint32_t audio_track_cblk_t::framesAvailable()
{
Mutex::Autolock _l(lock);
return framesAvailable_l();
}
uint32_t audio_track_cblk_t::framesAvailable_l()
{
uint32_t u = this->user;
uint32_t s = this->server;
if (flags & CBLK_DIRECTION_MSK) {
uint32_t limit = (s < loopStart) ? s : loopStart;
return limit + frameCount - u;
} else {
return frameCount + u - s;
}
}
uint32_t audio_track_cblk_t::framesReady()
{
uint32_t u = this->user;
uint32_t s = this->server;
if (flags & CBLK_DIRECTION_MSK) {
if (u < loopEnd) {
return u - s;
} else {
Mutex::Autolock _l(lock);
if (loopCount >= 0) {
return (loopEnd - loopStart)*loopCount + u - s;
} else {
return UINT_MAX;
}
}
} else {
return s - u;
}
}
// -------------------------------------------------------------------------
}; // namespace android
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