Whitespace and indentation
Fix indentation to be multiple of 4.
Make it easier to search:
sp< not sp < to
"switch (...)" instead of "switch(...)" (also "if" and "while")
Remove redundant blank line at start or EOF.
Remove whitespace at end of line.
Remove extra blank lines where they don't add value.
Use git diff -b or -w to verify.
Change-Id: I966b7ba852faa5474be6907fb212f5e267c2874e
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 @@
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 @@
}
Exit:
- if(status) {
+ if (status != NULL) {
*status = lStatus;
}
return trackHandle;
@@ -610,7 +610,7 @@
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 @@
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 @@
// 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 @@
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 @@
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::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 @@
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 @@
}
index = sessionEffects.indexOfKey(key);
- sp <SuspendedSessionDesc> desc;
+ sp<SuspendedSessionDesc> desc;
if (suspend) {
if (index >= 0) {
desc = sessionEffects.valueAt(index);
@@ -1659,14 +1659,14 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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;
- }
+ 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) {
+ if (client != NULL) {
mCblkMemory = client->heap()->allocate(size);
if (mCblkMemory != 0) {
mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer());
@@ -3298,22 +3298,22 @@
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 @@
// 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 @@
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 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
{
// 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 @@
{
// 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 @@
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 @@
// 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 @@
}
}
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 @@
}
Exit:
- if(status) {
+ if (status != NULL) {
*status = lStatus;
}
return handle;
@@ -6226,7 +6227,7 @@
handle.clear();
}
- if(status) {
+ if (status != NULL) {
*status = lStatus;
}
return handle;
@@ -6296,7 +6297,7 @@
}
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::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 @@
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 @@
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 @@
}
}
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 @@
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);
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index f3c8dd2..0e4b24a 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -465,9 +465,9 @@
// ThreadBase mutex before processing the mixer and effects. This guarantees the
// integrity of the chains during the process.
// Also sets the parameter 'effectChains' to current value of mEffectChains.
- void lockEffectChains_l(Vector<sp <EffectChain> >& effectChains);
+ void lockEffectChains_l(Vector< sp<EffectChain> >& effectChains);
// unlock effect chains after process
- void unlockEffectChains(const Vector<sp<EffectChain> >& effectChains);
+ void unlockEffectChains(const Vector< sp<EffectChain> >& effectChains);
// set audio mode to all effect chains
void setMode(audio_mode_t mode);
// get effect module with corresponding ID on specified audio session
@@ -1056,7 +1056,7 @@
virtual uint32_t activeSleepTimeUs();
private:
- bool outputsReady(const SortedVector<sp<OutputTrack> > &outputTracks);
+ bool outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks);
protected:
// threadLoop snippets
virtual void threadLoop_mix();
@@ -1504,7 +1504,7 @@
uint32_t strategy() const { return mStrategy; }
void setStrategy(uint32_t strategy)
- { mStrategy = strategy; }
+ { mStrategy = strategy; }
// suspend effect of the given type
void setEffectSuspended_l(const effect_uuid_t *type,
@@ -1544,7 +1544,7 @@
wp<ThreadBase> mThread; // parent mixer thread
Mutex mLock; // mutex protecting effect list
- Vector<sp<EffectModule> > mEffects; // list of effect modules
+ Vector< sp<EffectModule> > mEffects; // list of effect modules
int mSessionId; // audio session ID
int16_t *mInBuffer; // chain input buffer
int16_t *mOutBuffer; // chain output buffer
diff --git a/services/audioflinger/AudioPolicyService.cpp b/services/audioflinger/AudioPolicyService.cpp
index d57326b..c23eb04 100644
--- a/services/audioflinger/AudioPolicyService.cpp
+++ b/services/audioflinger/AudioPolicyService.cpp
@@ -649,7 +649,7 @@
mLock.lock();
while (!exitPending())
{
- while(!mAudioCommands.isEmpty()) {
+ while (!mAudioCommands.isEmpty()) {
nsecs_t curTime = systemTime();
// commands are sorted by increasing time stamp: execute them from index 0 and up
if (mAudioCommands[0]->mTime <= curTime) {
@@ -693,16 +693,16 @@
delete data;
}break;
case SET_PARAMETERS: {
- ParametersData *data = (ParametersData *)command->mParam;
- ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
- data->mKeyValuePairs.string(), data->mIO);
- command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
- if (command->mWaitStatus) {
- command->mCond.signal();
- mWaitWorkCV.wait(mLock);
- }
- delete data;
- }break;
+ ParametersData *data = (ParametersData *)command->mParam;
+ ALOGV("AudioCommandThread() processing set parameters string %s, io %d",
+ data->mKeyValuePairs.string(), data->mIO);
+ command->mStatus = AudioSystem::setParameters(data->mIO, data->mKeyValuePairs);
+ if (command->mWaitStatus) {
+ command->mCond.signal();
+ mWaitWorkCV.wait(mLock);
+ }
+ delete data;
+ }break;
case SET_VOICE_VOLUME: {
VoiceVolumeData *data = (VoiceVolumeData *)command->mParam;
ALOGV("AudioCommandThread() processing set voice volume volume %f",
@@ -916,19 +916,19 @@
AudioParameter param = AudioParameter(data->mKeyValuePairs);
AudioParameter param2 = AudioParameter(data2->mKeyValuePairs);
for (size_t j = 0; j < param.size(); j++) {
- String8 key;
- String8 value;
- param.getAt(j, key, value);
- for (size_t k = 0; k < param2.size(); k++) {
- String8 key2;
- String8 value2;
- param2.getAt(k, key2, value2);
- if (key2 == key) {
- param2.remove(key2);
- ALOGV("Filtering out parameter %s", key2.string());
- break;
- }
- }
+ String8 key;
+ String8 value;
+ param.getAt(j, key, value);
+ for (size_t k = 0; k < param2.size(); k++) {
+ String8 key2;
+ String8 value2;
+ param2.getAt(k, key2, value2);
+ if (key2 == key) {
+ param2.remove(key2);
+ ALOGV("Filtering out parameter %s", key2.string());
+ break;
+ }
+ }
}
// if all keys have been filtered out, remove the command.
// otherwise, update the key value pairs
@@ -1020,7 +1020,7 @@
ALOGE("startTone: illegal tone requested (%d)", tone);
if (stream != AUDIO_STREAM_VOICE_CALL)
ALOGE("startTone: illegal stream (%d) requested for tone %d", stream,
- tone);
+ tone);
mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING,
AUDIO_STREAM_VOICE_CALL);
return 0;
diff --git a/services/audioflinger/AudioPolicyService.h b/services/audioflinger/AudioPolicyService.h
index 7119b90..9ed905d 100644
--- a/services/audioflinger/AudioPolicyService.h
+++ b/services/audioflinger/AudioPolicyService.h
@@ -311,8 +311,8 @@
mutable Mutex mLock; // prevents concurrent access to AudioPolicy manager functions changing
// device connection state or routing
- sp <AudioCommandThread> mAudioCommandThread; // audio commands thread
- sp <AudioCommandThread> mTonePlaybackThread; // tone playback thread
+ sp<AudioCommandThread> mAudioCommandThread; // audio commands thread
+ sp<AudioCommandThread> mTonePlaybackThread; // tone playback thread
struct audio_policy_device *mpAudioPolicyDev;
struct audio_policy *mpAudioPolicy;
KeyedVector< audio_source_t, InputSourceDesc* > mInputSources;
diff --git a/services/audioflinger/AudioResampler.cpp b/services/audioflinger/AudioResampler.cpp
index 398ba0b..fbb54cf 100644
--- a/services/audioflinger/AudioResampler.cpp
+++ b/services/audioflinger/AudioResampler.cpp
@@ -227,7 +227,7 @@
mX0L = mBuffer.i16[mBuffer.frameCount*2-2];
mX0R = mBuffer.i16[mBuffer.frameCount*2-1];
provider->releaseBuffer(&mBuffer);
- // mBuffer.frameCount == 0 now so we reload a new buffer
+ // mBuffer.frameCount == 0 now so we reload a new buffer
}
int16_t *in = mBuffer.i16;
diff --git a/services/audioflinger/AudioResampler.h b/services/audioflinger/AudioResampler.h
index 9deb796..1610e00 100644
--- a/services/audioflinger/AudioResampler.h
+++ b/services/audioflinger/AudioResampler.h
@@ -33,7 +33,7 @@
// HIGH_QUALITY: fixed multi-tap FIR (e.g. 48KHz->44.1KHz)
// NOTE: high quality SRC will only be supported for
// certain fixed rate conversions. Sample rate cannot be
- // changed dynamically.
+ // changed dynamically.
enum src_quality {
DEFAULT=0,
LOW_QUALITY=1,
diff --git a/services/audioflinger/AudioResamplerCubic.h b/services/audioflinger/AudioResamplerCubic.h
index b72b62a..892785a 100644
--- a/services/audioflinger/AudioResamplerCubic.h
+++ b/services/audioflinger/AudioResamplerCubic.h
@@ -55,7 +55,7 @@
p->y1 = p->y2;
p->y2 = p->y3;
p->y3 = in;
- p->a = (3 * (p->y1 - p->y2) - p->y0 + p->y3) >> 1;
+ p->a = (3 * (p->y1 - p->y2) - p->y0 + p->y3) >> 1;
p->b = (p->y2 << 1) + p->y0 - (((5 * p->y1 + p->y3)) >> 1);
p->c = (p->y2 - p->y0) >> 1;
}
diff --git a/services/audioflinger/AudioResamplerSinc.cpp b/services/audioflinger/AudioResamplerSinc.cpp
index d373c08..76662d8 100644
--- a/services/audioflinger/AudioResamplerSinc.cpp
+++ b/services/audioflinger/AudioResamplerSinc.cpp
@@ -222,7 +222,7 @@
} else {
read<CHANNELS>(impulse, phaseFraction, mBuffer.i16, inputIndex);
}
- }
+ }
}
int16_t *in = mBuffer.i16;
const size_t frameCount = mBuffer.frameCount;
@@ -247,7 +247,7 @@
if (inputIndex >= frameCount)
break; // need a new buffer
read<CHANNELS>(impulse, phaseFraction, in, inputIndex);
- } else if(phaseIndex == 2) { // maximum value
+ } else if (phaseIndex == 2) { // maximum value
inputIndex++;
if (inputIndex >= frameCount)
break; // 0 frame available, 2 frames needed