Merge "Add lost frame handling in AudioSource" into gingerbread (23796e3b) · Commits · e / os / android_frameworks_base

include/media/stagefright/AudioSource.h

+2 −3

Original line number	Diff line number	Diff line
		@@ -57,11 +57,10 @@ private:

		bool mCollectStats;
		bool mTrackMaxAmplitude;
		int64_t mTotalReadTimeUs;
		int64_t mTotalReadBytes;
		int64_t mTotalReads;
		int64_t mStartTimeUs;
		int16_t mMaxAmplitude;
		int64_t mPrevSampleTimeUs;
		int64_t mNumLostFrames;

		MediaBufferGroup *mGroup;

media/libstagefright/AudioSource.cpp

+92 −45

Original line number	Diff line number	Diff line
		@@ -26,8 +26,7 @@
		#include <media/stagefright/MediaDefs.h>
		#include <media/stagefright/MetaData.h>
		#include <cutils/properties.h>
		#include <sys/time.h>
		#include <time.h>
		#include <stdlib.h>

		namespace android {

		@@ -35,9 +34,8 @@ AudioSource::AudioSource(
		int inputSource, uint32_t sampleRate, uint32_t channels)
		: mStarted(false),
		mCollectStats(false),
		mTotalReadTimeUs(0),
		mTotalReadBytes(0),
		mTotalReads(0),
		mPrevSampleTimeUs(0),
		mNumLostFrames(0),
		mGroup(NULL) {

		LOGV("sampleRate: %d, channels: %d", sampleRate, channels);
		@@ -110,10 +108,7 @@ status_t AudioSource::stop() {
		mStarted = false;

		if (mCollectStats) {
		LOGI("%lld reads: %.2f bps in %lld us",
		mTotalReads,
		(mTotalReadBytes * 8000000.0) / mTotalReadTimeUs,
		mTotalReadTimeUs);
		LOGI("Total lost audio frames: %lld", mNumLostFrames);
		}

		return OK;
		@@ -129,67 +124,113 @@ sp<MetaData> AudioSource::getFormat() {
		return meta;
		}

		/*
		* Returns -1 if frame skipping request is too long.
		* Returns 0 if there is no need to skip frames.
		* Returns 1 if we need to skip frames.
		*/
		static int skipFrame(int64_t timestampUs,
		const MediaSource::ReadOptions *options) {

		int64_t skipFrameUs;
		if (!options \|\| !options->getSkipFrame(&skipFrameUs)) {
		return 0;
		}

		if (skipFrameUs <= timestampUs) {
		return 0;
		}

		// Safe guard against the abuse of the kSkipFrame_Option.
		if (skipFrameUs - timestampUs >= 1E6) {
		LOGE("Frame skipping requested is way too long: %lld us",
		skipFrameUs - timestampUs);

		return -1;
		}

		LOGV("skipFrame: %lld us > timestamp: %lld us",
		skipFrameUs, timestampUs);

		return 1;

		}

		status_t AudioSource::read(
		MediaBuffer *out, const ReadOptions options) {
		*out = NULL;
		++mTotalReads;

		MediaBuffer *buffer;
		CHECK_EQ(mGroup->acquire_buffer(&buffer), OK);

		int err = 0;
		while (mStarted) {

		uint32_t numFramesRecorded;
		mRecord->getPosition(&numFramesRecorded);
		int64_t latency = mRecord->latency() * 1000;

		int64_t readTime = systemTime() / 1000;

		if (numFramesRecorded == 0) {
		if (numFramesRecorded == 0 && mPrevSampleTimeUs == 0) {
		// Initial delay
		if (mStartTimeUs > 0) {
		mStartTimeUs = readTime - mStartTimeUs;
		mStartTimeUs = systemTime() / 1000 - mStartTimeUs;
		} else {
		mStartTimeUs += latency;
		// Assume latency is constant.
		mStartTimeUs += mRecord->latency() * 1000;
		}
		mPrevSampleTimeUs = mStartTimeUs;
		}

		uint32_t sampleRate = mRecord->getSampleRate();

		// Insert null frames when lost frames are detected.
		int64_t timestampUs = mPrevSampleTimeUs;
		uint32_t numLostBytes = mRecord->getInputFramesLost() << 1;
		#if 0
		// Simulate lost frames
		numLostBytes = ((rand() * 1.0 / RAND_MAX)) * kMaxBufferSize;
		numLostBytes &= 0xFFFFFFFE; // Alignment request

		// Reduce the chance to lose
		if (rand() * 1.0 / RAND_MAX >= 0.05) {
		numLostBytes = 0;
		}
		#endif
		if (numLostBytes > 0) {
		// Not expect too many lost frames!
		CHECK(numLostBytes <= kMaxBufferSize);

		ssize_t n = 0;
		timestampUs += (1000000LL * numLostBytes >> 1) / sampleRate;
		CHECK(timestampUs > mPrevSampleTimeUs);
		if (mCollectStats) {
		n = mRecord->read(buffer->data(), buffer->size());
		int64_t endTime = systemTime() / 1000;
		mTotalReadTimeUs += (endTime - readTime);
		if (n >= 0) {
		mTotalReadBytes += n;
		mNumLostFrames += (numLostBytes >> 1);
		}
		} else {
		n = mRecord->read(buffer->data(), buffer->size());
		if ((err = skipFrame(timestampUs, options)) == -1) {
		buffer->release();
		return UNKNOWN_ERROR;
		} else if (err != 0) {
		continue;
		}
		memset(buffer->data(), 0, numLostBytes);
		buffer->set_range(0, numLostBytes);
		buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
		mPrevSampleTimeUs = timestampUs;
		*out = buffer;
		return OK;
		}

		ssize_t n = mRecord->read(buffer->data(), buffer->size());
		if (n < 0) {
		buffer->release();
		buffer = NULL;

		return (status_t)n;
		}

		uint32_t sampleRate = mRecord->getSampleRate();
		int64_t timestampUs = (1000000LL * numFramesRecorded) / sampleRate +
		mStartTimeUs;
		int64_t skipFrameUs;
		if (!options \|\| !options->getSkipFrame(&skipFrameUs)) {
		skipFrameUs = timestampUs; // Don't skip frame
		}

		if (skipFrameUs > timestampUs) {
		// Safe guard against the abuse of the kSkipFrame_Option.
		if (skipFrameUs - timestampUs >= 1E6) {
		LOGE("Frame skipping requested is way too long: %lld us",
		skipFrameUs - timestampUs);
		int64_t recordDurationUs = (1000000LL * n >> 1) / sampleRate;
		timestampUs += recordDurationUs;
		if ((err = skipFrame(timestampUs, options)) == -1) {
		buffer->release();
		return UNKNOWN_ERROR;
		}
		LOGV("skipFrame: %lld us > timestamp: %lld us, samples %d",
		skipFrameUs, timestampUs, numFramesRecorded);
		} else if (err != 0) {
		continue;
		}

		@@ -197,7 +238,13 @@ status_t AudioSource::read(
		trackMaxAmplitude((int16_t *) buffer->data(), n >> 1);
		}

		buffer->meta_data()->setInt64(kKeyTime, timestampUs);
		buffer->meta_data()->setInt64(kKeyTime, mPrevSampleTimeUs);
		CHECK(timestampUs > mPrevSampleTimeUs);
		if (mNumLostFrames == 0) {
		CHECK_EQ(mPrevSampleTimeUs,
		mStartTimeUs + (1000000LL * numFramesRecorded) / sampleRate);
		}
		mPrevSampleTimeUs = timestampUs;
		LOGV("initial delay: %lld, sample rate: %d, timestamp: %lld",
		mStartTimeUs, sampleRate, timestampUs);