Merge "libstagefright: Aggregate buffers in audio source" (d30bcef9) · Commits · e / os / android_frameworks_av

include/media/stagefright/AudioSource.h

+9 −2

Original line number	Diff line number	Diff line
		@@ -53,6 +53,7 @@ struct AudioSource : public MediaSource, public MediaBufferObserver {
		MediaBuffer *buffer, const ReadOptions options = NULL);

		status_t dataCallback(const AudioRecord::Buffer& buffer);
		virtual void onEvent(int event, void* info);
		virtual void signalBufferReturned(MediaBuffer *buffer);

		protected:
		@@ -60,7 +61,9 @@ protected:

		private:
		enum {
		kMaxBufferSize = 2048,
		//This max buffer size is derived from aggregation of audio
		//buffers for max duration 80 msec with 48K sampling rate.
		kMaxBufferSize = 30720,

		// After the initial mute, we raise the volume linearly
		// over kAutoRampDurationUs.
		@@ -75,6 +78,11 @@ private:
		Condition mFrameAvailableCondition;
		Condition mFrameEncodingCompletionCondition;

		AudioRecord::Buffer mTempBuf;
		uint32_t mPrevPosition;
		uint32_t mAllocBytes;
		int32_t mAudioSessionId;
		AudioRecord::transfer_type mTransferMode;
		sp<AudioRecord> mRecord;
		status_t mInitCheck;
		bool mStarted;
		@@ -91,7 +99,6 @@ private:
		int64_t mAutoRampStartUs;

		List<MediaBuffer * > mBuffersReceived;

		void trackMaxAmplitude(int16_t *data, int nSamples);

		// This is used to raise the volume from mute to the

media/libstagefright/AudioSource.cpp

+138 −20

Original line number	Diff line number	Diff line
		@@ -31,23 +31,12 @@
		#include <cutils/properties.h>
		#include <system/audio.h>

		#define AUDIO_RECORD_DEFAULT_BUFFER_DURATION 20
		namespace android {

		static void AudioRecordCallbackFunction(int event, void user, void info) {
		AudioSource source = (AudioSource ) user;
		switch (event) {
		case AudioRecord::EVENT_MORE_DATA: {
		source->dataCallback(((AudioRecord::Buffer ) info));
		break;
		}
		case AudioRecord::EVENT_OVERRUN: {
		ALOGW("AudioRecord reported overrun!");
		break;
		}
		default:
		// does nothing
		break;
		}
		source->onEvent(event, info);
		}

		AudioSource::AudioSource(
		@@ -80,6 +69,60 @@ AudioSource::AudioSource(
		bufCount++;
		}

		mTempBuf.size = 0;
		mTempBuf.frameCount = 0;
		mTempBuf.i16 = (short*)NULL;
		mPrevPosition = 0;
		mAudioSessionId = -1;
		mAllocBytes = 0;
		mTransferMode = AudioRecord::TRANSFER_CALLBACK;

		//decide whether to use callback or event pos callback
		//use position marker only for PCM 16 bit and mono or stereo capture
		//and if input source is camera
		if((mFormat == AUDIO_FORMAT_PCM_16_BIT) &&
		((channelCount == 1) \|\| (channelCount == 2)) &&
		(inputSource == AUDIO_SOURCE_CAMCORDER)) {

		//Need audioSession Id in the extended audio record constructor
		//where the transfer mode can be specified
		mAudioSessionId = AudioSystem::newAudioUniqueId();
		AudioSystem::acquireAudioSessionId(mAudioSessionId, -1);

		mRecord = new AudioRecord(
		inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
		audio_channel_in_mask_from_count(channelCount),
		(size_t) (bufCount * frameCount),
		AudioRecordCallbackFunction,
		this,
		frameCount /notificationFrames/,
		mAudioSessionId,
		AudioRecord::TRANSFER_SYNC,
		AUDIO_INPUT_FLAG_NONE);

		int buffDuration = AUDIO_RECORD_DEFAULT_BUFFER_DURATION;
		char propValue[PROPERTY_VALUE_MAX];
		if (property_get("audio.record.buffer.duration", propValue, NULL)) {
		if (atoi(propValue) < AUDIO_RECORD_DEFAULT_BUFFER_DURATION)
		buffDuration = AUDIO_RECORD_DEFAULT_BUFFER_DURATION;
		else
		buffDuration = atoi(propValue);
		}
		else
		buffDuration = AUDIO_RECORD_DEFAULT_BUFFER_DURATION;

		//set to update position after frames worth of buffduration time for 16 bits
		mAllocBytes = ((sizeof(uint8_t) * frameCount * 2 * channelCount));
		ALOGI("AudioSource in TRANSFER_SYNC with duration %d ms", buffDuration);
		mTempBuf.i16 = (short*) malloc(mAllocBytes);
		if (mTempBuf.i16 == NULL) {
		mAllocBytes = 0;
		mInitCheck = NO_MEMORY;
		}
		mTransferMode = AudioRecord::TRANSFER_SYNC;
		mRecord->setPositionUpdatePeriod((sampleRate * buffDuration)/1000);
		} else {
		//Sound recorder and VOIP use cases does NOT use aggregation
		mRecord = new AudioRecord(
		inputSource, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
		audio_channel_in_mask_from_count(channelCount),
		@@ -87,7 +130,12 @@ AudioSource::AudioSource(
		AudioRecordCallbackFunction,
		this,
		frameCount /notificationFrames/);
		ALOGI("AudioSource in TRANSFER_CALLBACK");
		mTransferMode = AudioRecord::TRANSFER_CALLBACK;
		}

		mInitCheck = mRecord->initCheck();

		mAutoRampStartUs = kAutoRampStartUs;
		uint32_t playbackLatencyMs = 0;
		if (AudioSystem::getOutputLatency(&playbackLatencyMs,
		@@ -100,6 +148,7 @@ AudioSource::AudioSource(
		} else {
		mInitCheck = status;
		}
		ALOGV("mInitCheck %d", mInitCheck);
		}

		AudioSource::AudioSource( audio_source_t inputSource, const sp<MetaData>& meta )
		@@ -137,6 +186,14 @@ AudioSource::AudioSource( audio_source_t inputSource, const sp<MetaData>& meta )
		AudioRecordCallbackFunction,
		this);
		mInitCheck = mRecord->initCheck();
		mTempBuf.size = 0;
		mTempBuf.frameCount = 0;
		mTempBuf.i16 = (short*)NULL;
		mPrevPosition = 0;
		mAudioSessionId = -1;
		mAllocBytes = 0;
		mTransferMode = AudioRecord::TRANSFER_CALLBACK;

		}

		AudioSource::~AudioSource() {
		@@ -223,6 +280,16 @@ status_t AudioSource::reset() {
		waitOutstandingEncodingFrames_l();
		releaseQueuedFrames_l();

		if (mTransferMode == AudioRecord::TRANSFER_SYNC) {
		if(mAudioSessionId != -1)
		AudioSystem::releaseAudioSessionId(mAudioSessionId, -1);

		mAudioSessionId = -1;
		if(mTempBuf.i16) {
		free(mTempBuf.i16);
		mTempBuf.i16 = (short*)NULL;
		}
		}
		return OK;
		}

		@@ -408,6 +475,57 @@ status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
		return OK;
		}

		void AudioSource::onEvent(int event, void* info) {

		switch (event) {
		case AudioRecord::EVENT_MORE_DATA: {
		dataCallback(((AudioRecord::Buffer ) info));
		break;
		}
		case AudioRecord::EVENT_NEW_POS: {
		uint32_t position = 0;
		mRecord->getPosition(&position);
		size_t framestoRead = position - mPrevPosition;
		size_t bytestoRead = (framestoRead * 2 * mRecord->channelCount());
		if(bytestoRead <=0 \|\| bytestoRead > mAllocBytes) {
		//try to read only max
		ALOGI("greater than allocated size in callback, adjusting size");
		bytestoRead = mAllocBytes;
		framestoRead = (mAllocBytes / (2 * mRecord->channelCount()));
		}

		if(mTempBuf.i16 && framestoRead > 0) {
		//read only if you have valid data
		size_t bytesRead = mRecord->read(mTempBuf.i16, framestoRead);
		size_t framesRead = 0;
		ALOGV("event_new_pos, new pos %d, frames to read %d\n", \
		position, framestoRead);
		ALOGV("bytes read = %d \n", bytesRead);
		if(bytesRead > 0){
		framesRead = (bytesRead / (2 * mRecord->channelCount()));
		mPrevPosition += framesRead;
		mTempBuf.size = bytesRead;
		mTempBuf.frameCount = framesRead;
		dataCallback(mTempBuf);
		} else {
		ALOGE("EVENT_NEW_POS did not return any data");
		}
		} else {
		ALOGE("Init error");
		}
		break;
		}
		case AudioRecord::EVENT_OVERRUN: {
		ALOGW("AudioRecord reported overrun!");
		break;
		}
		default:
		// does nothing
		break;
		}
		return;
		}

		void AudioSource::queueInputBuffer_l(MediaBuffer *buffer, int64_t timeUs) {
		if (mRecPaused) {
		if (!mBuffersReceived.empty()) {