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Diffstat (limited to 'services/audioflinger/AudioFlinger.cpp')
-rw-r--r--services/audioflinger/AudioFlinger.cpp211
1 files changed, 106 insertions, 105 deletions
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index fad7087..bfa4a49 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -190,13 +190,13 @@ void AudioFlinger::onFirstRef()
continue;
ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i],
- mod->name, mod->id);
+ mod->name, mod->id);
mAudioHwDevs.push(dev);
if (mPrimaryHardwareDev == NULL) {
mPrimaryHardwareDev = dev;
ALOGI("Using '%s' (%s.%s) as the primary audio interface",
- mod->name, mod->id, audio_interfaces[i]);
+ mod->name, mod->id, audio_interfaces[i]);
}
}
@@ -515,7 +515,7 @@ sp<IAudioTrack> AudioFlinger::createTrack(
}
Exit:
- if(status) {
+ if (status != NULL) {
*status = lStatus;
}
return trackHandle;
@@ -610,7 +610,7 @@ status_t AudioFlinger::setMasterVolume(float value)
mMasterVolume = value;
mMasterVolumeSW = swmv;
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
+ mPlaybackThreads.valueAt(i)->setMasterVolume(swmv);
return NO_ERROR;
}
@@ -642,7 +642,7 @@ status_t AudioFlinger::setMode(audio_mode_t mode)
Mutex::Autolock _l(mLock);
mMode = mode;
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setMode(mode);
+ mPlaybackThreads.valueAt(i)->setMode(mode);
}
return ret;
@@ -693,7 +693,7 @@ status_t AudioFlinger::setMasterMute(bool muted)
// This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger
mMasterMute = muted;
for (size_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setMasterMute(muted);
+ mPlaybackThreads.valueAt(i)->setMasterMute(muted);
return NO_ERROR;
}
@@ -761,7 +761,7 @@ status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value,
if (thread == NULL) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
- mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
+ mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value);
}
} else {
thread->setStreamVolume(stream, value);
@@ -786,7 +786,7 @@ status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted)
AutoMutex lock(mLock);
mStreamTypes[stream].mute = muted;
for (uint32_t i = 0; i < mPlaybackThreads.size(); i++)
- mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
+ mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted);
return NO_ERROR;
}
@@ -1161,7 +1161,7 @@ void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param)
void AudioFlinger::ThreadBase::processConfigEvents()
{
mLock.lock();
- while(!mConfigEvents.isEmpty()) {
+ while (!mConfigEvents.isEmpty()) {
ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size());
ConfigEvent configEvent = mConfigEvents[0];
mConfigEvents.removeAt(0);
@@ -1345,13 +1345,13 @@ void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectCha
mSuspendedSessions.editValueAt(index);
for (size_t i = 0; i < sessionEffects.size(); i++) {
- sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
+ sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i);
for (int j = 0; j < desc->mRefCount; j++) {
if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) {
chain->setEffectSuspendedAll_l(true);
} else {
ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x",
- desc->mType.timeLow);
+ desc->mType.timeLow);
chain->setEffectSuspended_l(&desc->mType, true);
}
}
@@ -1386,7 +1386,7 @@ void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *ty
}
index = sessionEffects.indexOfKey(key);
- sp <SuspendedSessionDesc> desc;
+ sp<SuspendedSessionDesc> desc;
if (suspend) {
if (index >= 0) {
desc = sessionEffects.valueAt(index);
@@ -1659,14 +1659,14 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
// createTrack() was called by the client process.
if (!mStreamTypes[streamType].valid) {
ALOGW("createTrack_l() on thread %p: invalidating track on stream %d",
- this, streamType);
+ this, streamType);
android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags);
}
}
lStatus = NO_ERROR;
Exit:
- if(status) {
+ if (status) {
*status = lStatus;
}
return track;
@@ -2645,10 +2645,10 @@ bool AudioFlinger::MixerThread::checkForNewParameters_l()
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
- mStandby = true;
- mBytesWritten = 0;
- status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+ mOutput->stream->common.standby(&mOutput->stream->common);
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
}
if (status == NO_ERROR && reconfig) {
@@ -2902,7 +2902,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix()
size_t count = mFrameCount * mChannelCount;
uint8_t *src = (uint8_t *)mMixBuffer + count-1;
int16_t *dst = mMixBuffer + count-1;
- while(count--) {
+ while (count--) {
*dst-- = (int16_t)(*src--^0x80) << 8;
}
}
@@ -2955,7 +2955,7 @@ void AudioFlinger::DirectOutputThread::threadLoop_mix()
size_t count = mFrameCount * mChannelCount;
int16_t *src = mMixBuffer;
uint8_t *dst = (uint8_t *)mMixBuffer;
- while(count--) {
+ while (count--) {
*dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80;
}
}
@@ -3014,10 +3014,10 @@ bool AudioFlinger::DirectOutputThread::checkForNewParameters_l()
status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
if (!mStandby && status == INVALID_OPERATION) {
- mOutput->stream->common.standby(&mOutput->stream->common);
- mStandby = true;
- mBytesWritten = 0;
- status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
+ mOutput->stream->common.standby(&mOutput->stream->common);
+ mStandby = true;
+ mBytesWritten = 0;
+ status = mOutput->stream->common.set_parameters(&mOutput->stream->common,
keyValuePair.string());
}
if (status == NO_ERROR && reconfig) {
@@ -3208,7 +3208,7 @@ void AudioFlinger::DuplicatingThread::updateWaitTime_l()
bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks)
{
for (size_t i = 0; i < outputTracks.size(); i++) {
- sp <ThreadBase> thread = outputTracks[i]->thread().promote();
+ sp<ThreadBase> thread = outputTracks[i]->thread().promote();
if (thread == 0) {
ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
return false;
@@ -3264,14 +3264,14 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
// ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize);
- size_t size = sizeof(audio_track_cblk_t);
- uint8_t channelCount = popcount(channelMask);
- size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
- if (sharedBuffer == 0) {
- size += bufferSize;
- }
-
- if (client != NULL) {
+ size_t size = sizeof(audio_track_cblk_t);
+ uint8_t channelCount = popcount(channelMask);
+ size_t bufferSize = frameCount*channelCount*sizeof(int16_t);
+ if (sharedBuffer == 0) {
+ size += bufferSize;
+ }
+
+ if (client != NULL) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
@@ -3298,22 +3298,22 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
client->heap()->dump("AudioTrack");
return;
}
- } else {
- mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
- // construct the shared structure in-place.
- new(mCblk) audio_track_cblk_t();
- // clear all buffers
- mCblk->frameCount = frameCount;
- mCblk->sampleRate = sampleRate;
- mChannelCount = channelCount;
- mChannelMask = channelMask;
- mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
- memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
- // Force underrun condition to avoid false underrun callback until first data is
- // written to buffer (other flags are cleared)
- mCblk->flags = CBLK_UNDERRUN_ON;
- mBufferEnd = (uint8_t *)mBuffer + bufferSize;
- }
+ } else {
+ mCblk = (audio_track_cblk_t *)(new uint8_t[size]);
+ // construct the shared structure in-place.
+ new(mCblk) audio_track_cblk_t();
+ // clear all buffers
+ mCblk->frameCount = frameCount;
+ mCblk->sampleRate = sampleRate;
+ mChannelCount = channelCount;
+ mChannelMask = channelMask;
+ mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
+ memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t));
+ // Force underrun condition to avoid false underrun callback until first data is
+ // written to buffer (other flags are cleared)
+ mCblk->flags = CBLK_UNDERRUN_ON;
+ mBufferEnd = (uint8_t *)mBuffer + bufferSize;
+ }
}
AudioFlinger::ThreadBase::TrackBase::~TrackBase()
@@ -3491,22 +3491,22 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
// AudioBufferProvider interface
status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
- AudioBufferProvider::Buffer* buffer, int64_t pts)
+ AudioBufferProvider::Buffer* buffer, int64_t pts)
{
- audio_track_cblk_t* cblk = this->cblk();
- uint32_t framesReady;
- uint32_t framesReq = buffer->frameCount;
+ audio_track_cblk_t* cblk = this->cblk();
+ uint32_t framesReady;
+ uint32_t framesReq = buffer->frameCount;
- // Check if last stepServer failed, try to step now
- if (mStepServerFailed) {
- if (!step()) goto getNextBuffer_exit;
- ALOGV("stepServer recovered");
- mStepServerFailed = false;
- }
+ // Check if last stepServer failed, try to step now
+ if (mStepServerFailed) {
+ if (!step()) goto getNextBuffer_exit;
+ ALOGV("stepServer recovered");
+ mStepServerFailed = false;
+ }
- framesReady = cblk->framesReady();
+ framesReady = cblk->framesReady();
- if (CC_LIKELY(framesReady)) {
+ if (CC_LIKELY(framesReady)) {
uint32_t s = cblk->server;
uint32_t bufferEnd = cblk->serverBase + cblk->frameCount;
@@ -3518,21 +3518,21 @@ status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(
framesReq = bufferEnd - s;
}
- buffer->raw = getBuffer(s, framesReq);
- if (buffer->raw == NULL) goto getNextBuffer_exit;
+ buffer->raw = getBuffer(s, framesReq);
+ if (buffer->raw == NULL) goto getNextBuffer_exit;
- buffer->frameCount = framesReq;
+ buffer->frameCount = framesReq;
return NO_ERROR;
- }
+ }
getNextBuffer_exit:
- buffer->raw = NULL;
- buffer->frameCount = 0;
- ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
- return NOT_ENOUGH_DATA;
+ buffer->raw = NULL;
+ buffer->frameCount = 0;
+ ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get());
+ return NOT_ENOUGH_DATA;
}
-uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{
+uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const {
return mCblk->framesReady();
}
@@ -3684,8 +3684,8 @@ status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId)
status_t status = DEAD_OBJECT;
sp<ThreadBase> thread = mThread.promote();
if (thread != 0) {
- PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
- status = playbackThread->attachAuxEffect(this, EffectId);
+ PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
+ status = playbackThread->attachAuxEffect(this, EffectId);
}
return status;
}
@@ -4104,14 +4104,14 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
mOverflow(false)
{
if (mCblk != NULL) {
- ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
- if (format == AUDIO_FORMAT_PCM_16_BIT) {
- mCblk->frameSize = mChannelCount * sizeof(int16_t);
- } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
- mCblk->frameSize = mChannelCount * sizeof(int8_t);
- } else {
- mCblk->frameSize = sizeof(int8_t);
- }
+ ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer);
+ if (format == AUDIO_FORMAT_PCM_16_BIT) {
+ mCblk->frameSize = mChannelCount * sizeof(int16_t);
+ } else if (format == AUDIO_FORMAT_PCM_8_BIT) {
+ mCblk->frameSize = mChannelCount * sizeof(int8_t);
+ } else {
+ mCblk->frameSize = sizeof(int8_t);
+ }
}
}
@@ -4130,7 +4130,7 @@ status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvi
uint32_t framesAvail;
uint32_t framesReq = buffer->frameCount;
- // Check if last stepServer failed, try to step now
+ // Check if last stepServer failed, try to step now
if (mStepServerFailed) {
if (!step()) goto getNextBuffer_exit;
ALOGV("stepServer recovered");
@@ -4976,7 +4976,7 @@ Exit:
status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid)
{
ALOGV("RecordThread::start tid=%d", tid);
- sp <ThreadBase> strongMe = this;
+ sp<ThreadBase> strongMe = this;
status_t status = NO_ERROR;
{
AutoMutex lock(mLock);
@@ -5029,7 +5029,7 @@ startError:
void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) {
ALOGV("RecordThread::stop");
- sp <ThreadBase> strongMe = this;
+ sp<ThreadBase> strongMe = this;
{
AutoMutex lock(mLock);
if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) {
@@ -5196,8 +5196,9 @@ bool AudioFlinger::RecordThread::checkForNewParameters_l()
if (status == NO_ERROR) {
status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
if (status == INVALID_OPERATION) {
- mInput->stream->common.standby(&mInput->stream->common);
- status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
+ mInput->stream->common.standby(&mInput->stream->common);
+ status = mInput->stream->common.set_parameters(&mInput->stream->common,
+ keyValuePair.string());
}
if (reconfig) {
if (status == BAD_VALUE &&
@@ -5285,8 +5286,8 @@ void AudioFlinger::RecordThread::readInputParameters()
if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2)
{
int channelCount;
- // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid
- // stereo to mono post process as the resampler always outputs stereo.
+ // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid
+ // stereo to mono post process as the resampler always outputs stereo.
if (mChannelCount == 1 && mReqChannelCount == 2) {
channelCount = 1;
} else {
@@ -5460,7 +5461,7 @@ status_t AudioFlinger::closeOutput(audio_io_handle_t output)
{
// keep strong reference on the playback thread so that
// it is not destroyed while exit() is executed
- sp <PlaybackThread> thread;
+ sp<PlaybackThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkPlaybackThread_l(output);
@@ -5613,7 +5614,7 @@ status_t AudioFlinger::closeInput(audio_io_handle_t input)
{
// keep strong reference on the record thread so that
// it is not destroyed while exit() is executed
- sp <RecordThread> thread;
+ sp<RecordThread> thread;
{
Mutex::Autolock _l(mLock);
thread = checkRecordThread_l(input);
@@ -5746,7 +5747,7 @@ void AudioFlinger::purgeStaleEffects_l() {
AudioSessionRef *ref = mAudioSessionRefs.itemAt(k);
if (ref->mSessionid == sessionid) {
ALOGV(" session %d still exists for %d with %d refs",
- sessionid, ref->mPid, ref->mCnt);
+ sessionid, ref->mPid, ref->mCnt);
found = true;
break;
}
@@ -5979,7 +5980,7 @@ sp<IEffect> AudioFlinger::createEffect(pid_t pid,
// because of code checking output when entering the function.
// Note: io is never 0 when creating an effect on an input
if (io == 0) {
- // look for the thread where the specified audio session is present
+ // look for the thread where the specified audio session is present
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
io = mPlaybackThreads.keyAt(i);
@@ -5987,12 +5988,12 @@ sp<IEffect> AudioFlinger::createEffect(pid_t pid,
}
}
if (io == 0) {
- for (size_t i = 0; i < mRecordThreads.size(); i++) {
- if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
- io = mRecordThreads.keyAt(i);
- break;
- }
- }
+ for (size_t i = 0; i < mRecordThreads.size(); i++) {
+ if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) {
+ io = mRecordThreads.keyAt(i);
+ break;
+ }
+ }
}
// If no output thread contains the requested session ID, default to
// first output. The effect chain will be moved to the correct output
@@ -6023,7 +6024,7 @@ sp<IEffect> AudioFlinger::createEffect(pid_t pid,
}
Exit:
- if(status) {
+ if (status != NULL) {
*status = lStatus;
}
return handle;
@@ -6226,7 +6227,7 @@ Exit:
handle.clear();
}
- if(status) {
+ if (status != NULL) {
*status = lStatus;
}
return handle;
@@ -6296,7 +6297,7 @@ void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) {
}
void AudioFlinger::ThreadBase::lockEffectChains_l(
- Vector<sp <AudioFlinger::EffectChain> >& effectChains)
+ Vector< sp<AudioFlinger::EffectChain> >& effectChains)
{
effectChains = mEffectChains;
for (size_t i = 0; i < mEffectChains.size(); i++) {
@@ -6305,7 +6306,7 @@ void AudioFlinger::ThreadBase::lockEffectChains_l(
}
void AudioFlinger::ThreadBase::unlockEffectChains(
- const Vector<sp <AudioFlinger::EffectChain> >& effectChains)
+ const Vector< sp<AudioFlinger::EffectChain> >& effectChains)
{
for (size_t i = 0; i < effectChains.size(); i++) {
effectChains[i]->unlock();
@@ -6481,7 +6482,7 @@ status_t AudioFlinger::PlaybackThread::attachAuxEffect_l(
void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId)
{
- for (size_t i = 0; i < mTracks.size(); ++i) {
+ for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (track->auxEffectId() == effectId) {
attachAuxEffect_l(track, 0);
@@ -7265,7 +7266,7 @@ AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect,
if (mCblk != NULL) {
new(mCblk) effect_param_cblk_t();
mBuffer = (uint8_t *)mCblk + bufOffset;
- }
+ }
} else {
ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
return;
@@ -8032,7 +8033,7 @@ void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModul
}
}
ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x",
- effect->desc().type.timeLow);
+ effect->desc().type.timeLow);
sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
// if effect is requested to suspended but was not yet enabled, supend it now.
if (desc->mEffect == 0) {
@@ -8045,7 +8046,7 @@ void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModul
return;
}
ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x",
- effect->desc().type.timeLow);
+ effect->desc().type.timeLow);
sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index);
desc->mEffect.clear();
effect->setSuspended(false);
generated by cgit v1.1 at 2024-05-18 07:29:47 +0000