Method rxFrame

SIP/SIPBase.cpp:1159–1239

GSM::AudioFrame *SipRtp::rxFrame()
{
	if(getSipState()!=Active) {
		LOG(DEBUG) <<"skip"<<LOGVAR(getSipState());
		return 0; 
	}
	int more = 0;

	// The buffer size is:
	//  For GSM, 260 bits + 4 bit header is 33 bytes.
	//  For TFO (3GPP 28.062 5.2.2) is 40 bytes.
	//  For AMR 'compact transport format', variable size, max is 244 bits + 10 bit header??
	// (pat) We will not support TFO for years, if ever.
	// TODO: Make the rxFrame buffer big enough (160?) for G.711.  But we dont support that yet.
	const int maxsize = 33;

	// (pat 8-2013) I tried rtp_session_get_current_recv_ts but it just doesnt work; returns garbage.
	// It is frightening how much we depend on the extremely buggy RTP library.
	//uint32_t suggestedRxTime = rtp_session_get_current_recv_ts(mSession);

	// (pat 8-2013) The RTP library has a scheduling mode and a blocking mode whereby it blocks the thread
	// until the specified time has elapsed, which is 20ms.
	// But after a discontinuity in the send data stream the RTP scheduler seems to get confused and rxFrame
	// returns 0 for long periods of time.
	// A discontinuity can be artificially generated at the time of writing using an in-call-SMS,
	// which currently blocks TCH while the current thread runs SACCH, a bug to be fixed,
	// however discontinuities could occur for other reasons so the code should not break when one occurs.
	// I added this code controlled by rtpUseRealTime so we could handle the real elapsed time ourselves and not block.
	// This code below suffers no such instability, so we are using it.  Someday if we switch RTP libraries,
	// someone can try turning off rtpUseRealTime - but you have to TEST DISCONTINUITIES, which is hard,
	// so dont just turn this off, try a test call, and pronounce it fixed.
	if (rtpUseRealTime) {
		struct timeval tv;
		gettimeofday(&tv,NULL);
		uint64_t realTime = (uint64_t)tv.tv_sec * (1000 * 1000) + (uint64_t)tv.tv_usec;		// time in usecs.
		if (mRxRealTime == 0) {
			// First time, init for next pass through...
			devassert(mRxTime == 0);
			mRxRealTime = realTime;
		} else {
			uint64_t delay = realTime - mRxRealTime;		// total elapsed time in usecs.
			uint64_t delayInFrames = delay / (1000 * 20);  // 20ms per frame	// elapsed time in frames.
			unsigned proposedRxTime = delayInFrames * 160;	// elapsed time in RTP 'time' units.  (160 / 8000 bitrate == 20 msecs)
			LOG(DEBUG) << format("realTime=%.3f delay=%.3f delayInFrames=%u RxTime=%u proposed=%u",
				realTime/1e6,delay/1e6,(unsigned)delayInFrames,mRxTime,proposedRxTime);
			if (proposedRxTime <= mRxTime) {
				LOG(DEBUG) <<"skip, insufficient time passed";
				return NULL;
			}
			// We will snag the next frame with a number equal or higher than this.  If there are none available, we get NULL.
			// When there is a discontinuity of missing frames, we get a bunch of NULL frames during the discontinuity, then
			// things pick up normally again.
			mRxTime += 160;
		}
	}

	GSM::AudioFrame *result = new GSM::AudioFrame(maxsize);  // Make it big enough for anything we might support.
	int ret = rtp_session_recv_with_ts(mSession, result->begin(), maxsize, mRxTime, &more);