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RTP: Send silence packets on idle streams for every second.

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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.

Bug: 3119690
Change-Id: Ib9c55e5dddfba28928bd9b376832b68bda24c0e4
This commit is contained in:
Chia-chi Yeh
2011-01-04 19:10:06 +08:00
parent 8f49c025ca
commit 3cf7137642
1 changed files with 48 additions and 32 deletions
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80
voip/jni/rtp/AudioGroup.cpp
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@@ -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) {
@@ -289,43 +293,55 @@ void AudioStream::encode(int tick, AudioStream *chain)
mDtmfEvent = -1;
}
// It is time to mix streams.
bool mixed = false;
int32_t buffer[mSampleCount + 3];
memset(buffer, 0, sizeof(buffer));
while (chain) {
if (chain != this &&
chain->mix(buffer, tick - mInterval, tick, mSampleRate)) {
mixed = true;
int16_t samples[mSampleCount];
if (mMode == RECEIVE_ONLY) {
if ((mTick ^ mKeepAlive) >> 10 == 0) {
return;
}
chain = chain->mNext;
}
if (!mixed) {
if ((mTick ^ mLogThrottle) >> 10) {
mLogThrottle = mTick;
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 &&
chain->mix(buffer, tick - mInterval, tick, mSampleRate)) {
mixed = true;
}
chain = chain->mNext;
}
if (mixed) {
// Saturate into 16 bits.
for (int i = 0; i < mSampleCount; ++i) {
int32_t sample = buffer[i];
if (sample < -32768) {
sample = -32768;
}
if (sample > 32767) {
sample = 32767;
}
samples[i] = sample;
}
} else {
if ((mTick ^ mKeepAlive) >> 10 == 0) {
return;
}
mKeepAlive = mTick;
memset(samples, 0, sizeof(samples));
LOGV("stream[%d] no data", mSocket);
}
return;
}
// Cook the packet and send it out.
int16_t samples[mSampleCount];
for (int i = 0; i < mSampleCount; ++i) {
int32_t sample = buffer[i];
if (sample < -32768) {
sample = -32768;
}
if (sample > 32767) {
sample = 32767;
}
samples[i] = sample;
}
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);