Merge "RTP: Update parameters for larger packet intervals." (6d8b9b84) · Commits · e / os / android_frameworks_base

voip/jni/rtp/AudioGroup.cpp

+40 −44

Original line number	Diff line number	Diff line
		@@ -28,6 +28,7 @@
		#include <arpa/inet.h>
		#include <netinet/in.h>

		// #define LOG_NDEBUG 0
		#define LOG_TAG "AudioGroup"
		#include <cutils/atomic.h>
		#include <cutils/properties.h>
		@@ -62,9 +63,9 @@ 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 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
		// Currently we make it 2048ms long and assume packet interval is 50ms or less.
		// The first 100ms is the place where samples get mixed. The rest is the real
		// jitter buffer. For a stream at 8000Hz it takes 32 kilobytes. These numbers
		// are chosen by experiments and each of them can be adjusted as needed.

		// Originally a stream does not send packets when it is receive-only or there is
		@@ -84,9 +85,11 @@ int gRandom = -1;
		// + Resampling is not done yet, so streams in one group must use the same rate.
		// For the first release only 8000Hz is supported.

		#define BUFFER_SIZE 512
		#define HISTORY_SIZE 80
		#define MEASURE_PERIOD 2000
		#define BUFFER_SIZE 2048
		#define HISTORY_SIZE 100
		#define MEASURE_BASE 100
		#define MEASURE_PERIOD 5000
		#define DTMF_PERIOD 200

		class AudioStream
		{
		@@ -278,7 +281,7 @@ void AudioStream::encode(int tick, AudioStream *chain)
		if (mMode != RECEIVE_ONLY && mDtmfEvent != -1) {
		int duration = mTimestamp - mDtmfStart;
		// Make sure duration is reasonable.
		if (duration >= 0 && duration < mSampleRate * 100) {
		if (duration >= 0 && duration < mSampleRate * DTMF_PERIOD) {
		duration += mSampleCount;
		int32_t buffer[4] = {
		htonl(mDtmfMagic \| mSequence),
		@@ -286,7 +289,7 @@ void AudioStream::encode(int tick, AudioStream *chain)
		mSsrc,
		htonl(mDtmfEvent \| duration),
		};
		if (duration >= mSampleRate * 100) {
		if (duration >= mSampleRate * DTMF_PERIOD) {
		buffer[3] \|= htonl(1 << 23);
		mDtmfEvent = -1;
		}
		@@ -298,26 +301,20 @@ void AudioStream::encode(int tick, AudioStream *chain)
		}

		int32_t buffer[mSampleCount + 3];
		int16_t samples[mSampleCount];
		if (mMode == RECEIVE_ONLY) {
		if ((mTick ^ mKeepAlive) >> 10 == 0) {
		return;
		}
		mKeepAlive = mTick;
		memset(samples, 0, sizeof(samples));
		} else {
		bool data = false;
		if (mMode != RECEIVE_ONLY) {
		// Mix all other streams.
		bool mixed = false;
		memset(buffer, 0, sizeof(buffer));
		while (chain) {
		if (chain != this &&
		chain->mix(buffer, tick - mInterval, tick, mSampleRate)) {
		mixed = true;
		if (chain != this) {
		data \|= chain->mix(buffer, tick - mInterval, tick, mSampleRate);
		}
		chain = chain->mNext;
		}
		}

		if (mixed) {
		int16_t samples[mSampleCount];
		if (data) {
		// Saturate into 16 bits.
		for (int i = 0; i < mSampleCount; ++i) {
		int32_t sample = buffer[i];
		@@ -335,6 +332,8 @@ void AudioStream::encode(int tick, AudioStream *chain)
		}
		mKeepAlive = mTick;
		memset(samples, 0, sizeof(samples));

		if (mMode != RECEIVE_ONLY) {
		LOGV("stream[%d] no data", mSocket);
		}
		}
		@@ -380,19 +379,16 @@ void AudioStream::decode(int tick)
		}
		}

		// 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) {
		// Adjust the jitter buffer if the latency keeps larger than the threshold
		// in the measurement period.
		int score = mBufferTail - tick - MEASURE_BASE;
		if (mLatencyScore > score \|\| mLatencyScore <= 0) {
		mLatencyScore = score;
		}
		if (mLatencyScore <= 0) {
		mLatencyTimer = tick;
		mLatencyScore = score;
		} else if (tick - mLatencyTimer >= MEASURE_PERIOD) {
		LOGV("stream[%d] reduces latency of %dms", mSocket, mLatencyScore);
		mBufferTail -= mLatencyScore;
		mLatencyTimer = tick;
		mLatencyScore = -1;
		}

		int count = (BUFFER_SIZE - (mBufferTail - mBufferHead)) * mSampleRate;