Line length 100
Change-Id: Ib28fd7b9ce951a6933f006e7f8812ba617625530
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 9bdab2f..35bd431 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1118,7 +1118,8 @@
// removeClient_l() must be called with AudioFlinger::mLock held
void AudioFlinger::removeClient_l(pid_t pid)
{
- ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid());
+ ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(),
+ IPCThreadState::self()->getCallingPid());
mClients.removeItem(pid);
}
@@ -1221,7 +1222,8 @@
{
IoConfigEvent *ioEvent = new IoConfigEvent(event, param);
mConfigEvents.add(static_cast<ConfigEvent *>(ioEvent));
- ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param);
+ ALOGV("sendIoConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event,
+ param);
mWaitWorkCV.signal();
}
@@ -1250,7 +1252,8 @@
PrioConfigEvent *prioEvent = static_cast<PrioConfigEvent *>(event);
int err = requestPriority(prioEvent->pid(), prioEvent->tid(), prioEvent->prio());
if (err != 0) {
- ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d",
+ ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; "
+ "error %d",
prioEvent->prio(), prioEvent->pid(), prioEvent->tid(), err);
}
} break;
@@ -1667,7 +1670,8 @@
snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this);
result.append(buffer);
- snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime));
+ snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n",
+ ns2ms(systemTime() - mLastWriteTime));
result.append(buffer);
snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites);
result.append(buffer);
@@ -1797,7 +1801,7 @@
if (mType == DIRECT) {
if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) {
if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) {
- ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \""
+ ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x "
"for output %p with format %d",
sampleRate, format, channelMask, mOutput, mFormat);
lStatus = BAD_VALUE;
@@ -1965,7 +1969,8 @@
if (track->mainBuffer() != mMixBuffer) {
sp<EffectChain> chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
- ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId());
+ ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(),
+ track->sessionId());
chain->incActiveTrackCnt();
}
}
@@ -2031,7 +2036,8 @@
AudioSystem::OutputDescriptor desc;
void *param2 = NULL;
- ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param);
+ ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event,
+ param);
switch (event) {
case AudioSystem::OUTPUT_OPENED:
@@ -2039,7 +2045,8 @@
desc.channels = mChannelMask;
desc.samplingRate = mSampleRate;
desc.format = mFormat;
- desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t)
+ desc.frameCount = mNormalFrameCount; // FIXME see
+ // AudioFlinger::frameCount(audio_io_handle_t)
desc.latency = latency();
param2 = &desc;
break;
@@ -2068,7 +2075,8 @@
// Calculate size of normal mix buffer relative to the HAL output buffer size
double multiplier = 1.0;
- if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) {
+ if (mType == MIXER && (kUseFastMixer == FastMixer_Static ||
+ kUseFastMixer == FastMixer_Dynamic)) {
size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000;
size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000;
// round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer
@@ -2087,9 +2095,10 @@
multiplier = (double) maxNormalFrameCount / (double) mFrameCount;
}
} else {
- // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC
- // (it would be unusual for the normal mix buffer size to not be a multiple of fast
- // track, but we sometimes have to do this to satisfy the maximum frame count constraint)
+ // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL
+ // SRC (it would be unusual for the normal mix buffer size to not be a multiple of fast
+ // track, but we sometimes have to do this to satisfy the maximum frame count
+ // constraint)
// FIXME this rounding up should not be done if no HAL SRC
uint32_t truncMult = (uint32_t) multiplier;
if ((truncMult & 1)) {
@@ -2103,7 +2112,8 @@
mNormalFrameCount = multiplier * mFrameCount;
// round up to nearest 16 frames to satisfy AudioMixer
mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
- ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount);
+ ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount,
+ mNormalFrameCount);
delete[] mMixBuffer;
mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount];
@@ -2241,7 +2251,8 @@
return event->type() == AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE;
}
-void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove)
+void AudioFlinger::PlaybackThread::threadLoop_removeTracks(
+ const Vector< sp<Track> >& tracksToRemove)
{
size_t count = tracksToRemove.size();
if (CC_UNLIKELY(count)) {
@@ -2897,7 +2908,8 @@
} else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) {
memset (mMixBuffer, 0, mixBufferSize);
sleepTime = 0;
- ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start");
+ ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)),
+ "anticipated start");
}
// TODO add standby time extension fct of effect tail
}
@@ -3131,7 +3143,8 @@
if ((track->framesReady() >= minFrames) && track->isReady() &&
!track->isPaused() && !track->isTerminated())
{
- ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this);
+ ALOGVV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server,
+ this);
mixedTracks++;
@@ -3144,7 +3157,8 @@
if (chain != 0) {
tracksWithEffect++;
} else {
- ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d",
+ ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on "
+ "session %d",
name, track->sessionId());
}
}
@@ -3274,7 +3288,8 @@
chain->clearInputBuffer();
}
- ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this);
+ ALOGVV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user,
+ cblk->server, this);
if ((track->sharedBuffer() != 0) || track->isTerminated() ||
track->isStopped() || track->isPaused()) {
// We have consumed all the buffers of this track.
@@ -3368,7 +3383,8 @@
if (track->mainBuffer() != mMixBuffer) {
chain = getEffectChain_l(track->sessionId());
if (chain != 0) {
- ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId());
+ ALOGV("stopping track on chain %p for session Id: %d", chain.get(),
+ track->sessionId());
chain->decActiveTrackCnt();
}
}
@@ -3381,7 +3397,8 @@
// mix buffer must be cleared if all tracks are connected to an
// effect chain as in this case the mixer will not write to
// mix buffer and track effects will accumulate into it
- if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) {
+ if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) ||
+ (mixedTracks == 0 && fastTracks > 0)) {
// FIXME as a performance optimization, should remember previous zero status
memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t));
}
@@ -3995,7 +4012,8 @@
AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger,
AudioFlinger::MixerThread* mainThread, audio_io_handle_t id)
- : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(), DUPLICATING),
+ : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->outDevice(),
+ DUPLICATING),
mWaitTimeMs(UINT_MAX)
{
addOutputTrack(mainThread);
@@ -4116,18 +4134,21 @@
}
-bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks)
+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();
if (thread == 0) {
- ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get());
+ ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p",
+ outputTracks[i].get());
return false;
}
PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
// see note at standby() declaration
if (playbackThread->standby() && !playbackThread->isSuspended()) {
- ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get());
+ ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(),
+ thread.get());
return false;
}
}
@@ -4174,7 +4195,8 @@
// mChannelCount
// mChannelMask
{
- ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size());
+ 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);
@@ -4335,7 +4357,8 @@
const sp<IMemory>& sharedBuffer,
int sessionId,
IAudioFlinger::track_flags_t flags)
- : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId),
+ : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer,
+ sessionId),
mMute(false),
mFillingUpStatus(FS_INVALID),
// mRetryCount initialized later when needed
@@ -4354,7 +4377,8 @@
if (mCblk != NULL) {
// NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of
// 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack
- mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t);
+ mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) :
+ sizeof(uint8_t);
// to avoid leaking a track name, do not allocate one unless there is an mCblk
mName = thread->getTrackName_l(channelMask, sessionId);
mCblk->mName = mName;
@@ -4379,7 +4403,8 @@
thread->mFastTrackAvailMask &= ~(1 << i);
}
}
- ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid());
+ ALOGV("Track constructor name %d, calling pid %d", mName,
+ IPCThreadState::self()->getCallingPid());
}
AudioFlinger::PlaybackThread::Track::~Track()
@@ -4421,8 +4446,8 @@
/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
{
- result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB "
- " Server User Main buf Aux Buf Flags Underruns\n");
+ result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate "
+ "L dB R dB Server User Main buf Aux Buf Flags Underruns\n");
}
void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
@@ -4649,7 +4674,8 @@
// and then to STOPPED and reset() when presentation is complete
mState = STOPPING_1;
}
- ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread);
+ ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName,
+ playbackThread);
}
if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) {
thread->mLock.unlock();
@@ -5408,7 +5434,8 @@
}
// AudioBufferProvider interface
-status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts)
+status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer,
+ int64_t pts)
{
audio_track_cblk_t* cblk = this->cblk();
uint32_t framesAvail;
@@ -5600,7 +5627,8 @@
mOutBuffer.frameCount = pInBuffer->frameCount;
nsecs_t startTime = systemTime();
if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) {
- ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get());
+ ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this,
+ mThread.unsafe_get());
outputBufferFull = true;
break;
}
@@ -5612,7 +5640,8 @@
}
}
- uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
+ uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount :
+ pInBuffer->frameCount;
memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
mCblk->stepUser(outFrames);
pInBuffer->frameCount -= outFrames;
@@ -5625,7 +5654,8 @@
mBufferQueue.removeAt(0);
delete [] pInBuffer->mBuffer;
delete pInBuffer;
- ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
+ ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this,
+ mThread.unsafe_get(), mBufferQueue.size());
} else {
break;
}
@@ -5641,11 +5671,14 @@
pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
pInBuffer->frameCount = inBuffer.frameCount;
pInBuffer->i16 = pInBuffer->mBuffer;
- memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
+ memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount *
+ sizeof(int16_t));
mBufferQueue.add(pInBuffer);
- ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
+ ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this,
+ mThread.unsafe_get(), mBufferQueue.size());
} else {
- ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this);
+ ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
+ mThread.unsafe_get(), this);
}
}
}
@@ -5670,7 +5703,8 @@
return outputBufferFull;
}
-status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
+status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(
+ AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs)
{
int active;
status_t result;
@@ -5934,13 +5968,14 @@
*sessionId = lSessionId;
}
}
- // create new record track. The record track uses one track in mHardwareMixerThread by convention.
+ // create new record track.
+ // The record track uses one track in mHardwareMixerThread by convention.
recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask,
frameCount, lSessionId, flags, tid, &lStatus);
}
if (lStatus != NO_ERROR) {
- // remove local strong reference to Client before deleting the RecordTrack so that the Client
- // destructor is called by the TrackBase destructor with mLock held
+ // remove local strong reference to Client before deleting the RecordTrack so that the
+ // Client destructor is called by the TrackBase destructor with mLock held
client.clear();
recordTrack.clear();
goto Exit;
@@ -5959,7 +5994,8 @@
// ----------------------------------------------------------------------------
-AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
+AudioFlinger::RecordHandle::RecordHandle(
+ const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack)
: BnAudioRecord(),
mRecordTrack(recordTrack)
{
@@ -5974,7 +6010,8 @@
return mRecordTrack->getCblk();
}
-status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) {
+status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event,
+ int triggerSession) {
ALOGV("RecordHandle::start()");
return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession);
}
@@ -6122,7 +6159,8 @@
size_t framesIn = mFrameCount - mRsmpInIndex;
if (framesIn) {
int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize;
- int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize;
+ int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) *
+ mActiveTrack->mCblk->frameSize;
if (framesIn > framesOut)
framesIn = framesOut;
mRsmpInIndex += framesIn;
@@ -6143,7 +6181,8 @@
if (framesOut && mFrameCount == mRsmpInIndex) {
void *readInto;
if (framesOut == mFrameCount &&
- ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
+ ((int)mChannelCount == mReqChannelCount ||
+ mFormat != AUDIO_FORMAT_PCM_16_BIT)) {
readInto = buffer.raw;
framesOut = 0;
} else {
@@ -6177,12 +6216,14 @@
if (mChannelCount == 1 && mReqChannelCount == 1) {
framesOut >>= 1;
}
- mResampler->resample(mRsmpOutBuffer, framesOut, this /* AudioBufferProvider* */);
- // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer()
- // are 32 bit aligned which should be always true.
+ mResampler->resample(mRsmpOutBuffer, framesOut,
+ this /* AudioBufferProvider* */);
+ // ditherAndClamp() works as long as all buffers returned by
+ // mActiveTrack->getNextBuffer() are 32 bit aligned which should be always true.
if (mChannelCount == 2 && mReqChannelCount == 1) {
ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut);
- // the resampler always outputs stereo samples: do post stereo to mono conversion
+ // the resampler always outputs stereo samples:
+ // do post stereo to mono conversion
downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer,
framesOut);
} else {
@@ -6656,7 +6697,8 @@
status = BAD_VALUE;
} else {
mInDevice = value;
- // disable AEC and NS if the device is a BT SCO headset supporting those pre processings
+ // disable AEC and NS if the device is a BT SCO headset supporting those
+ // pre processings
if (mTracks.size() > 0) {
bool suspend = audio_is_bluetooth_sco_device(mInDevice) &&
mAudioFlinger->btNrecIsOff();
@@ -6678,7 +6720,8 @@
mAudioSource = (audio_source_t)value;
}
if (status == NO_ERROR) {
- status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string());
+ status = mInput->stream->common.set_parameters(&mInput->stream->common,
+ keyValuePair.string());
if (status == INVALID_OPERATION) {
inputStandBy();
status = mInput->stream->common.set_parameters(&mInput->stream->common,
@@ -6688,8 +6731,10 @@
if (status == BAD_VALUE &&
reqFormat == mInput->stream->common.get_format(&mInput->stream->common) &&
reqFormat == AUDIO_FORMAT_PCM_16_BIT &&
- ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) &&
- popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 &&
+ ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common)
+ <= (2 * reqSamplingRate)) &&
+ popcount(mInput->stream->common.get_channels(&mInput->stream->common))
+ <= FCC_2 &&
(reqChannelCount <= FCC_2)) {
status = NO_ERROR;
}
@@ -6783,7 +6828,8 @@
mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN);
mRsmpOutBuffer = new int32_t[mFrameCount * 2];
- // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples
+ // optmization: if mono to mono, alter input frame count as if we were inputing
+ // stereo samples
if (mChannelCount == 1 && mReqChannelCount == 1) {
mFrameCount >>= 1;
}
@@ -7010,7 +7056,8 @@
&outStream);
mHardwareStatus = AUDIO_HW_IDLE;
- ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d",
+ ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, "
+ "Channels %x, status %d",
outStream,
config.sample_rate,
config.format,
@@ -7063,7 +7110,8 @@
MixerThread *thread2 = checkMixerThread_l(output2);
if (thread1 == NULL || thread2 == NULL) {
- ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2);
+ ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1,
+ output2);
return 0;
}
@@ -7098,7 +7146,8 @@
if (thread->type() == ThreadBase::MIXER) {
for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) {
- DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
+ DuplicatingThread *dupThread =
+ (DuplicatingThread *)mPlaybackThreads.valueAt(i).get();
dupThread->removeOutputTrack((MixerThread *)thread.get());
}
}
@@ -7185,16 +7234,17 @@
status = inHwHal->open_input_stream(inHwHal, id, *pDevices, &config,
&inStream);
- ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d",
+ ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, "
+ "status %d",
inStream,
config.sample_rate,
config.format,
config.channel_mask,
status);
- // If the input could not be opened with the requested parameters and we can handle the conversion internally,
- // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo
- // or stereo to mono conversions on 16 bit PCM inputs.
+ // If the input could not be opened with the requested parameters and we can handle the
+ // conversion internally, try to open again with the proposed parameters. The AudioFlinger can
+ // resample the input and do mono to stereo or stereo to mono conversions on 16 bit PCM inputs.
if (status == BAD_VALUE &&
reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT &&
(config.sample_rate <= 2 * reqSamplingRate) &&
@@ -8072,7 +8122,8 @@
for (size_t i = 0; i < mTracks.size(); ++i) {
sp<Track> track = mTracks[i];
if (session == track->sessionId()) {
- ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer);
+ ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(),
+ buffer);
track->setMainBuffer(buffer);
chain->incTrackCnt();
}
@@ -8914,12 +8965,15 @@
result.append("\t\tDescriptor:\n");
snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion,
- mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2],
+ mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],
+ mDescriptor.uuid.node[2],
mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n",
- mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion,
- mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2],
+ mDescriptor.type.timeLow, mDescriptor.type.timeMid,
+ mDescriptor.type.timeHiAndVersion,
+ mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],
+ mDescriptor.type.node[2],
mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]);
result.append(buffer);
snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n",
@@ -9003,7 +9057,8 @@
mBuffer = (uint8_t *)mCblk + bufOffset;
}
} else {
- ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t));
+ ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE +
+ sizeof(effect_param_cblk_t));
return;
}
}
@@ -9130,8 +9185,9 @@
// handle commands that are not forwarded transparently to effect engine
if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) {
- // No need to trylock() here as this function is executed in the binder thread serving a particular client process:
- // no risk to block the whole media server process or mixer threads is we are stuck here
+ // No need to trylock() here as this function is executed in the binder thread serving a
+ // particular client process: no risk to block the whole media server process or mixer
+ // threads if we are stuck here
Mutex::Autolock _l(mCblk->lock);
if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE ||
mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) {
@@ -9271,7 +9327,8 @@
}
// getEffectFromDesc_l() must be called with ThreadBase::mLock held
-sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor)
+sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(
+ effect_descriptor_t *descriptor)
{
size_t size = mEffects.size();
@@ -9430,7 +9487,8 @@
// check invalid effect chaining combinations
if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE ||
iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) {
- ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name);
+ ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s",
+ desc.name, d.name);
return INVALID_OPERATION;
}
// remember position of first insert effect and by default
@@ -9481,7 +9539,8 @@
}
mEffects.insertAt(effect, idx_insert);
- ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert);
+ ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this,
+ idx_insert);
}
effect->configure();
return NO_ERROR;
@@ -9512,7 +9571,8 @@
}
}
mEffects.removeAt(i);
- ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i);
+ ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(),
+ this, i);
break;
}
}
@@ -9736,7 +9796,8 @@
for (size_t i = 0; i < types.size(); i++) {
setEffectSuspended_l(types[i], false);
}
- ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index));
+ ALOGV("setEffectSuspendedAll_l() remove entry for %08x",
+ mSuspendedEffects.keyAt(index));
mSuspendedEffects.removeItem((int)kKeyForSuspendAll);
}
}
@@ -9762,7 +9823,8 @@
return true;
}
-void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects)
+void AudioFlinger::EffectChain::getSuspendEligibleEffects(
+ Vector< sp<AudioFlinger::EffectModule> > &effects)
{
effects.clear();
for (size_t i = 0; i < mEffects.size(); i++) {