commit 70949c47fbae3f836d15f040551d7631be3ed7c2
author Glenn Kasten <gkasten@google.com> Tue Aug 06 07:40:12 2013 -0700
committer Glenn Kasten <gkasten@google.com> Tue Aug 06 08:48:29 2013 -0700
tree 4bed0b2a693f915ca1d92233664dec9d49e20e5a
parent 77baebfe89c99415a0ee4c2ea87bfd8a0f9e51f1

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

services/audioflinger/Threads.cpp

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 @@
     :   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 @@
     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 @@
                 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 @@
                                          ) :
     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()