Android/frameworks_base
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Merge "Whitespace"

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This commit is contained in:
Glenn Kasten
2012-03-21 07:22:40 -07:00
committed by Android (Google) Code Review
parent 85b51b8ae3 f743e1f6ab
commit 32d72b2f53
6 changed files with 28 additions and 23 deletions
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2
include/common_time/local_clock.h
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@@ -28,7 +28,7 @@ namespace android {
class LocalClock {
public:
LocalClock();
LocalClock();
bool initCheck();

5
include/media/AudioTrack.h
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@@ -169,7 +169,7 @@ public:
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 @@ public:
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 @@ protected:
// 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,

3
media/libmedia/AudioTrack.cpp
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@@ -194,6 +194,7 @@ status_t AudioTrack::set(
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 @@ status_t AudioTrack::set(
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;

33
services/audioflinger/AudioFlinger.cpp
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@@ -1589,7 +1589,7 @@ void AudioFlinger::PlaybackThread::onFirstRef()
}
// 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,
@@ -2338,7 +2338,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
size_t tracksWithEffect = 0;
float masterVolume = mMasterVolume;
bool masterMute = mMasterMute;
bool masterMute = mMasterMute;
if (masterMute) {
masterVolume = 0;
@@ -2377,7 +2377,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
// +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
@@ -2515,6 +2515,7 @@ AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTrac
// 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
@@ -3373,19 +3374,19 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
}
} 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;
}
}

2
services/audioflinger/AudioMixer.cpp
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@@ -132,7 +132,7 @@ void AudioMixer::deleteTrackName(int name)
invalidateState(1<<name);
}
if (track.resampler != NULL) {
// delete the resampler
// delete the resampler
delete track.resampler;
track.resampler = NULL;
track.sampleRate = mSampleRate;

6
voip/jni/rtp/AudioGroup.cpp
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@@ -809,9 +809,9 @@ bool AudioGroup::DeviceThread::threadLoop()
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;
}