Make record buffer in RecordThread variable format (5744661e) · Commits · e / os / android_frameworks_av

services/audioflinger/Threads.cpp

+8 −8

Original line number	Diff line number	Diff line
		@@ -5328,7 +5328,7 @@ AudioFlinger::RecordThread::~RecordThread()
		}
		mAudioFlinger->unregisterWriter(mFastCaptureNBLogWriter);
		mAudioFlinger->unregisterWriter(mNBLogWriter);
		delete[] mRsmpInBuffer;
		free(mRsmpInBuffer);
		}

		void AudioFlinger::RecordThread::onFirstRef()
		@@ -5561,7 +5561,7 @@ reacquire_wakelock:
		// 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 @@ reacquire_wakelock:
		// 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 @@ reacquire_wakelock:
		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 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
		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 @@ void AudioFlinger::RecordThread::readInputParameters_l()
		// 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 @@ void AudioFlinger::RecordThread::readInputParameters_l()
		// 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?

services/audioflinger/Threads.h

+1 −1

Original line number	Diff line number	Diff line
		@@ -1267,7 +1267,7 @@ private:
		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