am 9f0810e8: Merge "AudioFlinger: rename variables to clarify reference to... (10fde074) · Commits · e / os / android_frameworks_base

include/private/media/AudioTrackShared.h

+1 −1

Original line number	Diff line number	Diff line
		@@ -59,7 +59,7 @@ struct audio_track_cblk_t
		// 8 bit PCM data: in this case, mCblk->frameSize is based on a sample size of
		// 16 bit because data is converted to 16 bit before being stored in buffer
		uint32_t frameSize;
		uint8_t channels;
		uint8_t channelCount;
		uint8_t flowControlFlag; // underrun (out) or overrrun (in) indication
		uint8_t out; // out equals 1 for AudioTrack and 0 for AudioRecord
		uint8_t forceReady;

libs/audioflinger/AudioFlinger.cpp

+32 −31

Original line number	Diff line number	Diff line
		@@ -783,7 +783,7 @@ void AudioFlinger::removeClient_l(pid_t pid)
		AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id)
		: Thread(false),
		mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0),
		mFormat(0), mFrameSize(1), mStandby(false), mId(id), mExiting(false)
		mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false)
		{
		}

		@@ -816,7 +816,7 @@ uint32_t AudioFlinger::ThreadBase::sampleRate() const

		int AudioFlinger::ThreadBase::channelCount() const
		{
		return mChannelCount;
		return (int)mChannelCount;
		}

		int AudioFlinger::ThreadBase::format() const
		@@ -1064,7 +1064,7 @@ sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTra
		status_t lStatus;

		if (mType == DIRECT) {
		if (sampleRate != mSampleRate \|\| format != mFormat \|\| channelCount != mChannelCount) {
		if (sampleRate != mSampleRate \|\| format != mFormat \|\| channelCount != (int)mChannelCount) {
		LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p",
		sampleRate, format, channelCount, mOutput);
		lStatus = BAD_VALUE;
		@@ -1243,7 +1243,7 @@ void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
		switch (event) {
		case AudioSystem::OUTPUT_OPENED:
		case AudioSystem::OUTPUT_CONFIG_CHANGED:
		desc.channels = mChannelCount;
		desc.channels = mChannels;
		desc.samplingRate = mSampleRate;
		desc.format = mFormat;
		desc.frameCount = mFrameCount;
		@@ -1264,10 +1264,10 @@ void AudioFlinger::PlaybackThread::audioConfigChanged(int event, int param) {
		void AudioFlinger::PlaybackThread::readOutputParameters()
		{
		mSampleRate = mOutput->sampleRate();
		mChannelCount = AudioSystem::popCount(mOutput->channels());

		mChannels = mOutput->channels();
		mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
		mFormat = mOutput->format();
		mFrameSize = mOutput->frameSize();
		mFrameSize = (uint16_t)mOutput->frameSize();
		mFrameCount = mOutput->bufferSize() / mFrameSize;

		// FIXME - Current mixer implementation only supports stereo output: Always
		@@ -2342,7 +2342,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
		// clear all buffers
		mCblk->frameCount = frameCount;
		mCblk->sampleRate = sampleRate;
		mCblk->channels = (uint8_t)channelCount;
		mCblk->channelCount = (uint8_t)channelCount;
		if (sharedBuffer == 0) {
		mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
		memset(mBuffer, 0, frameCountchannelCountsizeof(int16_t));
		@@ -2366,7 +2366,7 @@ AudioFlinger::ThreadBase::TrackBase::TrackBase(
		// clear all buffers
		mCblk->frameCount = frameCount;
		mCblk->sampleRate = sampleRate;
		mCblk->channels = (uint8_t)channelCount;
		mCblk->channelCount = (uint8_t)channelCount;
		mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t);
		memset(mBuffer, 0, frameCountchannelCountsizeof(int16_t));
		// Force underrun condition to avoid false underrun callback until first data is
		@@ -2433,7 +2433,7 @@ int AudioFlinger::ThreadBase::TrackBase::sampleRate() const {
		}

		int AudioFlinger::ThreadBase::TrackBase::channelCount() const {
		return (int)mCblk->channels;
		return (int)mCblk->channelCount;
		}

		void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const {
		@@ -2445,9 +2445,9 @@ void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t f
		if (bufferStart < mBuffer \|\| bufferStart > bufferEnd \|\| bufferEnd > mBufferEnd \|\|
		((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) {
		LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \
		server %d, serverBase %d, user %d, userBase %d, channels %d",
		server %d, serverBase %d, user %d, userBase %d, channelCount %d",
		bufferStart, bufferEnd, mBuffer, mBufferEnd,
		cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channels);
		cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount);
		return 0;
		}

		@@ -2532,7 +2532,7 @@ void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size)
		(mClient == NULL) ? getpid() : mClient->pid(),
		mStreamType,
		mFormat,
		mCblk->channels,
		mCblk->channelCount,
		mFrameCount,
		mState,
		mMute,
		@@ -2827,7 +2827,7 @@ void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size)
		snprintf(buffer, size, " %05d %03u %03u %04u %01d %05u %08x %08x\n",
		(mClient == NULL) ? getpid() : mClient->pid(),
		mFormat,
		mCblk->channels,
		mCblk->channelCount,
		mFrameCount,
		mState,
		mCblk->sampleRate,
		@@ -2856,8 +2856,8 @@ AudioFlinger::PlaybackThread::OutputTrack::OutputTrack(
		mCblk->volume[0] = mCblk->volume[1] = 0x1000;
		mOutBuffer.frameCount = 0;
		playbackThread->mTracks.add(this);
		LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channels %d mBufferEnd %p",
		mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channels, mBufferEnd);
		LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p",
		mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd);
		} else {
		LOGW("Error creating output track on thread %p", playbackThread);
		}
		@@ -2892,7 +2892,7 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr
		{
		Buffer *pInBuffer;
		Buffer inBuffer;
		uint32_t channels = mCblk->channels;
		uint32_t channelCount = mCblk->channelCount;
		bool outputBufferFull = false;
		inBuffer.frameCount = frames;
		inBuffer.i16 = data;
		@@ -2908,10 +2908,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr
		if (mBufferQueue.size() < kMaxOverFlowBuffers) {
		uint32_t startFrames = (mCblk->frameCount - frames);
		pInBuffer = new Buffer;
		pInBuffer->mBuffer = new int16_t[startFrames * channels];
		pInBuffer->mBuffer = new int16_t[startFrames * channelCount];
		pInBuffer->frameCount = startFrames;
		pInBuffer->i16 = pInBuffer->mBuffer;
		memset(pInBuffer->raw, 0, startFrames * channels * sizeof(int16_t));
		memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t));
		mBufferQueue.add(pInBuffer);
		} else {
		LOGW ("OutputTrack::write() %p no more buffers in queue", this);
		@@ -2949,12 +2949,12 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr
		}

		uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount;
		memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channels * sizeof(int16_t));
		memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t));
		mCblk->stepUser(outFrames);
		pInBuffer->frameCount -= outFrames;
		pInBuffer->i16 += outFrames * channels;
		pInBuffer->i16 += outFrames * channelCount;
		mOutBuffer.frameCount -= outFrames;
		mOutBuffer.i16 += outFrames * channels;
		mOutBuffer.i16 += outFrames * channelCount;

		if (pInBuffer->frameCount == 0) {
		if (mBufferQueue.size()) {
		@@ -2974,10 +2974,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr
		if (thread != 0 && !thread->standby()) {
		if (mBufferQueue.size() < kMaxOverFlowBuffers) {
		pInBuffer = new Buffer;
		pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channels];
		pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount];
		pInBuffer->frameCount = inBuffer.frameCount;
		pInBuffer->i16 = pInBuffer->mBuffer;
		memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channels * sizeof(int16_t));
		memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t));
		mBufferQueue.add(pInBuffer);
		LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size());
		} else {
		@@ -2993,10 +2993,10 @@ bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t fr
		if (mCblk->user < mCblk->frameCount) {
		frames = mCblk->frameCount - mCblk->user;
		pInBuffer = new Buffer;
		pInBuffer->mBuffer = new int16_t[frames * channels];
		pInBuffer->mBuffer = new int16_t[frames * channelCount];
		pInBuffer->frameCount = frames;
		pInBuffer->i16 = pInBuffer->mBuffer;
		memset(pInBuffer->raw, 0, frames * channels * sizeof(int16_t));
		memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t));
		mBufferQueue.add(pInBuffer);
		} else if (mActive) {
		stop();
		@@ -3371,7 +3371,7 @@ bool AudioFlinger::RecordThread::threadLoop()
		framesIn = framesOut;
		mRsmpInIndex += framesIn;
		framesOut -= framesIn;
		if (mChannelCount == mReqChannelCount \|\|
		if ((int)mChannelCount == mReqChannelCount \|\|
		mFormat != AudioSystem::PCM_16_BIT) {
		memcpy(dst, src, framesIn * mFrameSize);
		} else {
		@@ -3392,7 +3392,7 @@ bool AudioFlinger::RecordThread::threadLoop()
		}
		if (framesOut && mFrameCount == mRsmpInIndex) {
		if (framesOut == mFrameCount &&
		(mChannelCount == mReqChannelCount \|\| mFormat != AudioSystem::PCM_16_BIT)) {
		((int)mChannelCount == mReqChannelCount \|\| mFormat != AudioSystem::PCM_16_BIT)) {
		mBytesRead = mInput->read(buffer.raw, mInputBytes);
		framesOut = 0;
		} else {
		@@ -3696,7 +3696,7 @@ void AudioFlinger::RecordThread::audioConfigChanged(int event, int param) {
		switch (event) {
		case AudioSystem::INPUT_OPENED:
		case AudioSystem::INPUT_CONFIG_CHANGED:
		desc.channels = mChannelCount;
		desc.channels = mChannels;
		desc.samplingRate = mSampleRate;
		desc.format = mFormat;
		desc.frameCount = mFrameCount;
		@@ -3720,9 +3720,10 @@ void AudioFlinger::RecordThread::readInputParameters()
		mResampler = 0;

		mSampleRate = mInput->sampleRate();
		mChannelCount = AudioSystem::popCount(mInput->channels());
		mChannels = mInput->channels();
		mChannelCount = (uint16_t)AudioSystem::popCount(mChannels);
		mFormat = mInput->format();
		mFrameSize = mInput->frameSize();
		mFrameSize = (uint16_t)mInput->frameSize();
		mInputBytes = mInput->bufferSize();
		mFrameCount = mInputBytes / mFrameSize;
		mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount];

libs/audioflinger/AudioFlinger.h

+3 −2

Original line number	Diff line number	Diff line
		@@ -366,9 +366,10 @@ private:
		sp<AudioFlinger> mAudioFlinger;
		uint32_t mSampleRate;
		size_t mFrameCount;
		int mChannelCount;
		uint32_t mChannels;
		uint16_t mChannelCount;
		uint16_t mFrameSize;
		int mFormat;
		uint32_t mFrameSize;
		Condition mParamCond;
		Vector<String8> mNewParameters;
		status_t mParamStatus;