Make record buffer in RecordThread variable format
Change-Id: Id4bb9b973eeea16946fba3bc084c7ac270d9fa33
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b30fd20..373d643 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -5328,7 +5328,7 @@
}
mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter);
mAudioFlinger->unregisterWriter(mNBLogWriter);
- delete[] mRsmpInBuffer;
+ free(mRsmpInBuffer);
}
void AudioFlinger::RecordThread::onFirstRef()
@@ -5561,7 +5561,7 @@
// If an NBAIO source is present, use it to read the normal capture's data
if (mPipeSource != 0) {
size_t framesToRead = mBufferSize / mFrameSize;
- framesRead = mPipeSource->read(&mRsmpInBuffer[rear * mChannelCount],
+ framesRead = mPipeSource->read((uint8_t*)mRsmpInBuffer + rear * mFrameSize,
framesToRead, AudioBufferProvider::kInvalidPTS);
if (framesRead == 0) {
// since pipe is non-blocking, simulate blocking input
@@ -5570,7 +5570,7 @@
// otherwise use the HAL / AudioStreamIn directly
} else {
ssize_t bytesRead = mInput->stream->read(mInput->stream,
- &mRsmpInBuffer[rear * mChannelCount], mBufferSize);
+ (uint8_t*)mRsmpInBuffer + rear * mFrameSize, mBufferSize);
if (bytesRead < 0) {
framesRead = bytesRead;
} else {
@@ -5590,13 +5590,13 @@
ALOG_ASSERT(framesRead > 0);
if (mTeeSink != 0) {
- (void) mTeeSink->write(&mRsmpInBuffer[rear * mChannelCount], framesRead);
+ (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead);
}
// If destination is non-contiguous, we now correct for reading past end of buffer.
{
size_t part1 = mRsmpInFramesP2 - rear;
if ((size_t) framesRead > part1) {
- memcpy(mRsmpInBuffer, &mRsmpInBuffer[mRsmpInFramesP2 * mChannelCount],
+ memcpy(mRsmpInBuffer, (uint8_t*)mRsmpInBuffer + mRsmpInFramesP2 * mFrameSize,
(framesRead - part1) * mFrameSize);
}
}
@@ -6214,7 +6214,7 @@
return NOT_ENOUGH_DATA;
}
- buffer->raw = recordThread->mRsmpInBuffer + front * recordThread->mChannelCount;
+ buffer->raw = (uint8_t*)recordThread->mRsmpInBuffer + front * recordThread->mFrameSize;
buffer->frameCount = part1;
mRsmpInUnrel = part1;
return NO_ERROR;
@@ -6588,7 +6588,7 @@
// Note this is independent of the maximum downsampling ratio permitted for capture.
mRsmpInFrames = mFrameCount * 7;
mRsmpInFramesP2 = roundup(mRsmpInFrames);
- delete[] mRsmpInBuffer;
+ free(mRsmpInBuffer);
// TODO optimize audio capture buffer sizes ...
// Here we calculate the size of the sliding buffer used as a source
@@ -6598,7 +6598,7 @@
// The current value is higher than necessary. However it should not add to latency.
// Over-allocate beyond mRsmpInFramesP2 to permit a HAL read past end of buffer
- mRsmpInBuffer = new int16_t[(mRsmpInFramesP2 + mFrameCount - 1) * mChannelCount];
+ (void)posix_memalign(&mRsmpInBuffer, 32, (mRsmpInFramesP2 + mFrameCount - 1) * mFrameSize);
// AudioRecord mSampleRate and mChannelCount are constant due to AudioRecord API constraints.
// But if thread's mSampleRate or mChannelCount changes, how will that affect active tracks?
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 27bc56b..ed2e4a1 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1267,7 +1267,7 @@
Condition mStartStopCond;
// resampler converts input at HAL Hz to output at AudioRecord client Hz
- int16_t *mRsmpInBuffer; // see new[] for details on the size
+ void *mRsmpInBuffer; //
size_t mRsmpInFrames; // size of resampler input in frames
size_t mRsmpInFramesP2;// size rounded up to a power-of-2