am dfd1484e: Merge "RTP: Adjust the jitter buffer to 512ms." into gingerbread (f6381ec1) · Commits · e / os / android_frameworks_base

voip/jni/rtp/AudioGroup.cpp

+35 −23

Original line number	Diff line number	Diff line
		@@ -57,9 +57,9 @@ int gRandom = -1;
		// a modulo operation on the index while accessing the array. However modulo can
		// be expensive on some platforms, such as ARM. Thus we round up the size of the
		// array to the nearest power of 2 and then use bitwise-and instead of modulo.
		// Currently we make it 256ms long and assume packet interval is 32ms or less.
		// The first 64ms is the place where samples get mixed. The rest 192ms is the
		// real jitter buffer. For a stream at 8000Hz it takes 4096 bytes. These numbers
		// Currently we make it 512ms long and assume packet interval is 40ms or less.
		// The first 80ms is the place where samples get mixed. The rest 432ms is the
		// real jitter buffer. For a stream at 8000Hz it takes 8192 bytes. These numbers
		// are chosen by experiments and each of them can be adjusted as needed.

		// Other notes:
		@@ -69,7 +69,11 @@ int gRandom = -1;
		// milliseconds. No floating points.
		// + If we cannot get enough CPU, we drop samples and simulate packet loss.
		// + Resampling is not done yet, so streams in one group must use the same rate.
		// For the first release we might only support 8kHz and 16kHz.
		// For the first release only 8000Hz is supported.

		#define BUFFER_SIZE 512
		#define HISTORY_SIZE 80
		#define MEASURE_PERIOD 2000

		class AudioStream
		{
		@@ -111,6 +115,7 @@ private:
		int mBufferMask;
		int mBufferHead;
		int mBufferTail;
		int mLatencyTimer;
		int mLatencyScore;

		uint16_t mSequence;
		@@ -159,12 +164,13 @@ bool AudioStream::set(int mode, int socket, sockaddr_storage *remote,
		mInterval = mSampleCount / mSampleRate;

		// Allocate jitter buffer.
		for (mBufferMask = 8192; mBufferMask < sampleRate; mBufferMask <<= 1);
		mBufferMask >>= 2;
		for (mBufferMask = 8; mBufferMask < mSampleRate; mBufferMask <<= 1);
		mBufferMask *= BUFFER_SIZE;
		mBuffer = new int16_t[mBufferMask];
		--mBufferMask;
		mBufferHead = 0;
		mBufferTail = 0;
		mLatencyTimer = 0;
		mLatencyScore = 0;

		// Initialize random bits.
		@@ -340,29 +346,35 @@ void AudioStream::decode(int tick)
		}

		// Make sure mBufferHead and mBufferTail are reasonable.
		if ((unsigned int)(tick + 256 - mBufferHead) > 1024) {
		mBufferHead = tick - 64;
		if ((unsigned int)(tick + BUFFER_SIZE - mBufferHead) > BUFFER_SIZE * 2) {
		mBufferHead = tick - HISTORY_SIZE;
		mBufferTail = mBufferHead;
		}

		if (tick - mBufferHead > 64) {
		if (tick - mBufferHead > HISTORY_SIZE) {
		// Throw away outdated samples.
		mBufferHead = tick - 64;
		mBufferHead = tick - HISTORY_SIZE;
		if (mBufferTail - mBufferHead < 0) {
		mBufferTail = mBufferHead;
		}
		}

		if (mBufferTail - tick <= 80) {
		mLatencyScore = tick;
		} else if (tick - mLatencyScore >= 5000) {
		// Reset the jitter buffer to 40ms if the latency keeps larger than 80ms
		// in the past 5s. This rarely happens, so let us just keep it simple.
		LOGD("stream[%d] latency control", mSocket);
		mBufferTail = tick + 40;
		// Adjust the jitter buffer if the latency keeps larger than two times of the
		// packet interval in the past two seconds.
		int score = mBufferTail - tick - mInterval * 2;
		if (mLatencyScore > score) {
		mLatencyScore = score;
		}
		if (mLatencyScore <= 0) {
		mLatencyTimer = tick;
		mLatencyScore = score;
		} else if (tick - mLatencyTimer >= MEASURE_PERIOD) {
		LOGD("stream[%d] reduces latency of %dms", mSocket, mLatencyScore);
		mBufferTail -= mLatencyScore;
		mLatencyTimer = tick;
		}

		if (mBufferTail - mBufferHead > 256 - mInterval) {
		if (mBufferTail - mBufferHead > BUFFER_SIZE - mInterval) {
		// Buffer overflow. Drop the packet.
		LOGD("stream[%d] buffer overflow", mSocket);
		recv(mSocket, &c, 1, MSG_DONTWAIT);
		@@ -413,17 +425,17 @@ void AudioStream::decode(int tick)
		}

		if (tick - mBufferTail > 0) {
		// Buffer underrun. Reset the jitter buffer to 40ms.
		// Buffer underrun. Reset the jitter buffer.
		LOGD("stream[%d] buffer underrun", mSocket);
		if (mBufferTail - mBufferHead <= 0) {
		mBufferHead = tick + 40;
		mBufferHead = tick + mInterval;
		mBufferTail = mBufferHead;
		} else {
		int tail = (tick + 40) * mSampleRate;
		int tail = (tick + mInterval) * mSampleRate;
		for (int i = mBufferTail * mSampleRate; i - tail < 0; ++i) {
		mBuffer[i & mBufferMask] = 0;
		}
		mBufferTail = tick + 40;
		mBufferTail = tick + mInterval;
		}
		}

		@@ -680,7 +692,7 @@ bool AudioGroup::NetworkThread::threadLoop()
		int count = 0;

		for (AudioStream *stream = chain; stream; stream = stream->mNext) {
		if (!stream->mTick \|\| tick - stream->mTick >= 0) {
		if (tick - stream->mTick >= 0) {
		stream->encode(tick, chain);
		}
		if (deadline - stream->mTick > 0) {