Whitespace
Fix indentation, and add blank lines in key places for clarity
Change-Id: I57a0a8142394f83203161aa9b8aa9276abf3ed7c
diff --git a/include/common_time/local_clock.h b/include/common_time/local_clock.h
index 845d1c21..384c3de 100644
--- a/include/common_time/local_clock.h
+++ b/include/common_time/local_clock.h
@@ -28,7 +28,7 @@
class LocalClock {
public:
- LocalClock();
+ LocalClock();
bool initCheck();
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
index 552e829..ad27a1e 100644
--- a/include/media/AudioTrack.h
+++ b/include/media/AudioTrack.h
@@ -169,7 +169,7 @@
callback_t cbf = 0,
void* user = 0,
int notificationFrames = 0,
- int sessionId = 0);
+ int sessionId = 0);
/* Creates an audio track and registers it with AudioFlinger. With this constructor,
* the PCM data to be rendered by AudioTrack is passed in a shared memory buffer
@@ -215,7 +215,7 @@
int notificationFrames = 0,
const sp<IMemory>& sharedBuffer = 0,
bool threadCanCallJava = false,
- int sessionId = 0);
+ int sessionId = 0);
/* Result of constructing the AudioTrack. This must be checked
@@ -468,6 +468,7 @@
// body of AudioTrackThread::threadLoop()
bool processAudioBuffer(const sp<AudioTrackThread>& thread);
+
status_t createTrack_l(audio_stream_type_t streamType,
uint32_t sampleRate,
audio_format_t format,
diff --git a/media/libmedia/AudioTrack.cpp b/media/libmedia/AudioTrack.cpp
index 048be1d..ebb28cd 100644
--- a/media/libmedia/AudioTrack.cpp
+++ b/media/libmedia/AudioTrack.cpp
@@ -194,6 +194,7 @@
if (AudioSystem::getOutputSamplingRate(&afSampleRate, streamType) != NO_ERROR) {
return NO_INIT;
}
+
uint32_t afLatency;
if (AudioSystem::getOutputLatency(&afLatency, streamType) != NO_ERROR) {
return NO_INIT;
@@ -203,9 +204,11 @@
if (streamType == AUDIO_STREAM_DEFAULT) {
streamType = AUDIO_STREAM_MUSIC;
}
+
if (sampleRate == 0) {
sampleRate = afSampleRate;
}
+
// these below should probably come from the audioFlinger too...
if (format == AUDIO_FORMAT_DEFAULT) {
format = AUDIO_FORMAT_PCM_16_BIT;
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index d83d19a..86bb7f9 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1588,7 +1588,7 @@
}
// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held
-sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
+sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l(
const sp<AudioFlinger::Client>& client,
audio_stream_type_t streamType,
uint32_t sampleRate,
@@ -2337,7 +2337,7 @@
size_t tracksWithEffect = 0;
float masterVolume = mMasterVolume;
- bool masterMute = mMasterMute;
+ bool masterMute = mMasterMute;
if (masterMute) {
masterVolume = 0;
@@ -2376,7 +2376,7 @@
// +1 for rounding and +1 for additional sample needed for interpolation
minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1;
// add frames already consumed but not yet released by the resampler
- // because cblk->framesReady() will include these frames
+ // because cblk->framesReady() will include these frames
minFrames += mAudioMixer->getUnreleasedFrames(track->name());
// the minimum track buffer size is normally twice the number of frames necessary
// to fill one buffer and the resampler should not leave more than one buffer worth
@@ -2514,6 +2514,7 @@
// reset retry count
track->mRetryCount = kMaxTrackRetries;
+
// If one track is ready, set the mixer ready if:
// - the mixer was not ready during previous round OR
// - no other track is not ready
@@ -3372,19 +3373,19 @@
}
} 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;
+ // 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;
}
}
diff --git a/services/audioflinger/AudioMixer.cpp b/services/audioflinger/AudioMixer.cpp
index 1ec238b..6e761ba 100644
--- a/services/audioflinger/AudioMixer.cpp
+++ b/services/audioflinger/AudioMixer.cpp
@@ -132,7 +132,7 @@
invalidateState(1<<name);
}
if (track.resampler != NULL) {
- // delete the resampler
+ // delete the resampler
delete track.resampler;
track.resampler = NULL;
track.sampleRate = mSampleRate;
diff --git a/voip/jni/rtp/AudioGroup.cpp b/voip/jni/rtp/AudioGroup.cpp
index 1139577..b9bbd16 100644
--- a/voip/jni/rtp/AudioGroup.cpp
+++ b/voip/jni/rtp/AudioGroup.cpp
@@ -809,9 +809,9 @@
AudioTrack track;
AudioRecord record;
if (track.set(AUDIO_STREAM_VOICE_CALL, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_CHANNEL_OUT_MONO, output) != NO_ERROR || record.set(
- AUDIO_SOURCE_VOICE_COMMUNICATION, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
- AUDIO_CHANNEL_IN_MONO, input) != NO_ERROR) {
+ AUDIO_CHANNEL_OUT_MONO, output) != NO_ERROR ||
+ record.set(AUDIO_SOURCE_VOICE_COMMUNICATION, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
+ AUDIO_CHANNEL_IN_MONO, input) != NO_ERROR) {
ALOGE("cannot initialize audio device");
return false;
}