am 3cf71376: RTP: Send silence packets on idle streams for every second. (dc78e3fe) · Commits · e / os / android_frameworks_base

voip/jni/rtp/AudioGroup.cpp

+48 −32

Original line number	Diff line number	Diff line
		@@ -63,6 +63,14 @@ int gRandom = -1;
		// 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.

		// Originally a stream does not send packets when it is receive-only or there is
		// nothing to mix. However, this causes some problems with certain firewalls and
		// proxies. A firewall might remove a port mapping when there is no outgoing
		// packet for a preiod of time, and a proxy might wait for incoming packets from
		// both sides before start forwarding. To solve these problems, we send out a
		// silence packet on the stream for every second. It should be good enough to
		// keep the stream alive with relatively low resources.

		// Other notes:
		// + We use elapsedRealtime() to get the time. Since we use 32bit variables
		// instead of 64bit ones, comparison must be done by subtraction.
		@@ -110,7 +118,7 @@ private:
		int mSampleRate;
		int mSampleCount;
		int mInterval;
		int mLogThrottle;
		int mKeepAlive;

		int16_t *mBuffer;
		int mBufferMask;
		@@ -262,12 +270,8 @@ void AudioStream::encode(int tick, AudioStream *chain)
		++mSequence;
		mTimestamp += mSampleCount;

		if (mMode == RECEIVE_ONLY) {
		return;
		}

		// If there is an ongoing DTMF event, send it now.
		if (mDtmfEvent != -1) {
		if (mMode != RECEIVE_ONLY && mDtmfEvent != -1) {
		int duration = mTimestamp - mDtmfStart;
		// Make sure duration is reasonable.
		if (duration >= 0 && duration < mSampleRate * 100) {
		@@ -289,9 +293,17 @@ void AudioStream::encode(int tick, AudioStream *chain)
		mDtmfEvent = -1;
		}

		// It is time to mix streams.
		bool mixed = false;
		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 {
		// Mix all other streams.
		bool mixed = false;
		memset(buffer, 0, sizeof(buffer));
		while (chain) {
		if (chain != this &&
		@@ -300,16 +312,9 @@ void AudioStream::encode(int tick, AudioStream *chain)
		}
		chain = chain->mNext;
		}
		if (!mixed) {
		if ((mTick ^ mLogThrottle) >> 10) {
		mLogThrottle = mTick;
		LOGV("stream[%d] no data", mSocket);
		}
		return;
		}

		// Cook the packet and send it out.
		int16_t samples[mSampleCount];
		if (mixed) {
		// Saturate into 16 bits.
		for (int i = 0; i < mSampleCount; ++i) {
		int32_t sample = buffer[i];
		if (sample < -32768) {
		@@ -320,12 +325,23 @@ void AudioStream::encode(int tick, AudioStream *chain)
		}
		samples[i] = sample;
		}
		} else {
		if ((mTick ^ mKeepAlive) >> 10 == 0) {
		return;
		}
		mKeepAlive = mTick;
		memset(samples, 0, sizeof(samples));
		LOGV("stream[%d] no data", mSocket);
		}
		}

		if (!mCodec) {
		// Special case for device stream.
		send(mSocket, samples, sizeof(samples), MSG_DONTWAIT);
		return;
		}

		// Cook the packet and send it out.
		buffer[0] = htonl(mCodecMagic \| mSequence);
		buffer[1] = htonl(mTimestamp);
		buffer[2] = mSsrc;
		@@ -883,7 +899,7 @@ void add(JNIEnv *env, jobject thiz, jint mode,
		int codecType = -1;
		char codecName[16];
		int sampleRate = -1;
		sscanf(codecSpec, "%d %[^/]%*c%d", &codecType, codecName, &sampleRate);
		sscanf(codecSpec, "%d %15[^/]%*c%d", &codecType, codecName, &sampleRate);
		codec = newAudioCodec(codecName);
		int sampleCount = (codec ? codec->set(sampleRate, codecSpec) : -1);
		env->ReleaseStringUTFChars(jCodecSpec, codecSpec);