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| author | Glenn Kasten <gkasten@google.com> | 2013-08-06 07:40:12 -0700 |
|---|---|---|
| committer | Glenn Kasten <gkasten@google.com> | 2013-08-06 08:48:29 -0700 |
| commit | 70949c47fbae3f836d15f040551d7631be3ed7c2 (patch) | |
| tree | 4bed0b2a693f915ca1d92233664dec9d49e20e5a /services/audioflinger/Threads.cpp | |
| parent | 77baebfe89c99415a0ee4c2ea87bfd8a0f9e51f1 (diff) | |
| download | frameworks_av-70949c47fbae3f836d15f040551d7631be3ed7c2.zip frameworks_av-70949c47fbae3f836d15f040551d7631be3ed7c2.tar.gz frameworks_av-70949c47fbae3f836d15f040551d7631be3ed7c2.tar.bz2 | |
Pull up mBufferSize from RecordThread to ThreadBase
Now both playback and record threads have a HAL buffer size for fast
tracks, and a potentially larger-sized buffer for normal tracks.
Change-Id: I6f57b85ecece3cbd4e9e64b8c41c3352f7656912
Diffstat (limited to 'services/audioflinger/Threads.cpp')
| -rw-r--r-- | services/audioflinger/Threads.cpp | 11 |
1 files changed, 7 insertions, 4 deletions
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp index f0c27c3..2b67a02 100644 --- a/services/audioflinger/Threads.cpp +++ b/services/audioflinger/Threads.cpp @@ -266,8 +266,8 @@ AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio : Thread(false /*canCallJava*/), mType(type), mAudioFlinger(audioFlinger), - // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, and mFormat are - // set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() + // mSampleRate, mFrameCount, mChannelMask, mChannelCount, mFrameSize, mFormat, mBufferSize + // are set by PlaybackThread::readOutputParameters() or RecordThread::readInputParameters() mParamStatus(NO_ERROR), mStandby(false), mOutDevice(outDevice), mInDevice(inDevice), mAudioSource(AUDIO_SOURCE_DEFAULT), mId(id), @@ -423,6 +423,8 @@ void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) result.append(buffer); snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); result.append(buffer); + snprintf(buffer, SIZE, "HAL buffer size: %u bytes\n", mBufferSize); + result.append(buffer); snprintf(buffer, SIZE, "Channel Count: %u\n", mChannelCount); result.append(buffer); snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); @@ -1559,7 +1561,8 @@ void AudioFlinger::PlaybackThread::readOutputParameters() mFormat); } mFrameSize = audio_stream_frame_size(&mOutput->stream->common); - mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; + mBufferSize = mOutput->stream->common.get_buffer_size(&mOutput->stream->common); + mFrameCount = mBufferSize / mFrameSize; if (mFrameCount & 15) { ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", mFrameCount); @@ -4159,7 +4162,7 @@ AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, ) : ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD), mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), - // mRsmpInIndex and mBufferSize set by readInputParameters() + // mRsmpInIndex set by readInputParameters() mReqChannelCount(popcount(channelMask)), mReqSampleRate(sampleRate) // mBytesRead is only meaningful while active, and so is cleared in start() |