media: Use audio timestamp for recording. (4f998cde) · Commits · e / os / android_frameworks_av

include/media/stagefright/AudioSource.h

+2 −0

Original line number	Diff line number	Diff line
		@@ -89,6 +89,8 @@ private:
		int64_t mPrevSampleTimeUs;
		int64_t mInitialReadTimeUs;
		int64_t mNumFramesReceived;
		int64_t mNumFramesSkipped;
		int64_t mNumFramesLost;
		int64_t mNumClientOwnedBuffers;

		List<MediaBuffer * > mBuffersReceived;

media/libstagefright/AudioSource.cpp

+30 −11

Original line number	Diff line number	Diff line
		@@ -62,6 +62,8 @@ AudioSource::AudioSource(
		mPrevSampleTimeUs(0),
		mInitialReadTimeUs(0),
		mNumFramesReceived(0),
		mNumFramesSkipped(0),
		mNumFramesLost(0),
		mNumClientOwnedBuffers(0) {
		ALOGV("sampleRate: %u, outSampleRate: %u, channelCount: %u",
		sampleRate, outSampleRate, channelCount);
		@@ -295,11 +297,27 @@ void AudioSource::signalBufferReturned(MediaBuffer *buffer) {
		}

		status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
		int64_t timeUs = systemTime() / 1000ll;
		int64_t timeUs, position, timeNs;
		ExtendedTimestamp ts;
		ExtendedTimestamp::Location location;
		const int32_t usPerSec = 1000000;

		if (mRecord->getTimestamp(&ts) == OK &&
		ts.getBestTimestamp(&position, &timeNs, ExtendedTimestamp::TIMEBASE_MONOTONIC,
		&location) == OK) {
		// Use audio timestamp.
		timeUs = timeNs / 1000 -
		(position - mNumFramesSkipped -
		mNumFramesReceived + mNumFramesLost) * usPerSec / mSampleRate;
		} else {
		// This should not happen in normal case.
		ALOGW("Failed to get audio timestamp, fallback to use systemclock");
		timeUs = systemTime() / 1000ll;
		// Estimate the real sampling time of the 1st sample in this buffer
		// from AudioRecord's latency. (Apply this adjustment first so that
		// the start time logic is not affected.)
		timeUs -= mRecord->latency() * 1000LL;
		}

		ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);
		Mutex::Autolock autoLock(mLock);
		@@ -308,10 +326,15 @@ status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
		return OK;
		}

		const size_t bufferSize = audioBuffer.size;

		// Drop retrieved and previously lost audio data.
		if (mNumFramesReceived == 0 && timeUs < mStartTimeUs) {
		(void) mRecord->getInputFramesLost();
		ALOGV("Drop audio data at %" PRId64 "/%" PRId64 " us", timeUs, mStartTimeUs);
		int64_t receievedFrames = bufferSize / mRecord->frameSize();
		ALOGV("Drop audio data(%" PRId64 " frames) at %" PRId64 "/%" PRId64 " us",
		receievedFrames, timeUs, mStartTimeUs);
		mNumFramesSkipped += receievedFrames;
		return OK;
		}

		@@ -320,11 +343,7 @@ status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
		// Initial delay
		if (mStartTimeUs > 0) {
		mStartTimeUs = timeUs - mStartTimeUs;
		} else {
		// Assume latency is constant.
		mStartTimeUs += mRecord->latency() * 1000;
		}

		mPrevSampleTimeUs = mStartTimeUs;
		}

		@@ -354,6 +373,7 @@ status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
		MediaBuffer *lostAudioBuffer = new MediaBuffer(bufferSize);
		memset(lostAudioBuffer->data(), 0, bufferSize);
		lostAudioBuffer->set_range(0, bufferSize);
		mNumFramesLost += bufferSize / mRecord->frameSize();
		queueInputBuffer_l(lostAudioBuffer, timeUs);
		}

		@@ -362,7 +382,6 @@ status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) {
		return OK;
		}

		const size_t bufferSize = audioBuffer.size;
		MediaBuffer *buffer = new MediaBuffer(bufferSize);
		memcpy((uint8_t *) buffer->data(),
		audioBuffer.i16, audioBuffer.size);