diff options
-rw-r--r-- | voip/jni/rtp/AudioGroup.cpp | 80 |
1 files changed, 48 insertions, 32 deletions
diff --git a/voip/jni/rtp/AudioGroup.cpp b/voip/jni/rtp/AudioGroup.cpp index cba1123..64db250 100644 --- a/voip/jni/rtp/AudioGroup.cpp +++ b/voip/jni/rtp/AudioGroup.cpp @@ -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; - LOGV("stream[%d] no data", mSocket); + 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; } - 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; + 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); } - 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); |