audiortp.cpp 16.2 KB
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/*
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 *  Copyright (C) 2004-2007 Savoir-Faire Linux inc.
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 *  Author: Emmanuel Milou <emmanuel.milou@savoirfairelinux.com>
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 *  Author: Alexandre Bourget <alexandre.bourget@savoirfairelinux.com>
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 *  Author: Yan Morin <yan.morin@savoirfairelinux.com>
 *  Author: Laurielle Lea <laurielle.lea@savoirfairelinux.com>
 *
 *  This program is free software; you can redistribute it and/or modify
 *  it under the terms of the GNU General Public License as published by
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 *  the Free Software Foundation; either version 3 of the License, or
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 *  (at your option) any later version.
 *
 *  This program is distributed in the hope that it will be useful,
 *  but WITHOUT ANY WARRANTY; without even the implied warranty of
 *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 *  GNU General Public License for more details.
 *
 *  You should have received a copy of the GNU General Public License
 *  along with this program; if not, write to the Free Software
 *   Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
 */

#include <cstdio>
#include <cstdlib>
#include <ccrtp/rtp.h>
#include <assert.h>
#include <string>
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#include <cstring>
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//#include <fstream>  // fstream + iostream pour fstream debugging..
//#include <iostream> // removeable...
#include <math.h>
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#include "../global.h"
#include "../manager.h"
#include "codecDescriptor.h"
#include "audiortp.h"
#include "audiolayer.h"
#include "ringbuffer.h"
#include "../user_cfg.h"
#include "../sipcall.h"
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#include <samplerate.h>
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////////////////////////////////////////////////////////////////////////////////
// AudioRtp                                                          
////////////////////////////////////////////////////////////////////////////////
AudioRtp::AudioRtp ()
{
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	_RTXThread = 0;
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}

AudioRtp::~AudioRtp (void) {
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	delete _RTXThread; _RTXThread = 0;
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}

int 
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AudioRtp::createNewSession (SIPCall *ca) {
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	ost::MutexLock m(_threadMutex);

	// something should stop the thread before...
	if ( _RTXThread != 0 ) { 
		_debug("! ARTP Failure: Thread already exists..., stopping it\n");
		delete _RTXThread; _RTXThread = 0;
		//return -1; 
	}

	// Start RTP Send/Receive threads
	_symmetric = Manager::instance().getConfigInt(SIGNALISATION,SYMMETRIC) ? true : false;
	_RTXThread = new AudioRtpRTX (ca, _symmetric);

	try {
		if (_RTXThread->start() != 0) {
			_debug("! ARTP Failure: unable to start RTX Thread\n");
			return -1;
		}
	} catch(...) {
		_debugException("! ARTP Failure: when trying to start a thread");
		throw;
	}
	return 0;
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}

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void
AudioRtp::closeRtpSession () {
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	ost::MutexLock m(_threadMutex);
	// This will make RTP threads finish.
	// _debug("Stopping AudioRTP\n");
	try {
		delete _RTXThread; _RTXThread = 0;
	} catch(...) {
		_debugException("! ARTP Exception: when stopping audiortp\n");
		throw;
	}
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}

////////////////////////////////////////////////////////////////////////////////
// AudioRtpRTX Class                                                          //
////////////////////////////////////////////////////////////////////////////////
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AudioRtpRTX::AudioRtpRTX (SIPCall *sipcall, bool sym)
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	// : _fstream("/tmp/audio.dat", std::ofstream::binary)
{
	setCancel(cancelDeferred);
	time = new ost::Time();
	_ca = sipcall;
	_sym = sym;
	// AudioRtpRTX should be close if we change sample rate

	_codecSampleRate = _ca->getAudioCodec()->getClockRate();
	
	// TODO: Change bind address according to user settings.
	// TODO: this should be the local ip not the external (router) IP
	std::string localipConfig = _ca->getLocalIp(); // _ca->getLocalIp();
	ost::InetHostAddress local_ip(localipConfig.c_str());

	if (!_sym) {
		_sessionRecv = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
		_sessionSend = new ost::RTPSession(local_ip, _ca->getLocalAudioPort());
		_session = NULL;
	} else {
		_session = new ost::SymmetricRTPSession (local_ip, _ca->getLocalAudioPort());
		_sessionRecv = NULL;
		_sessionSend = NULL;
	}

	// libsamplerate-related
	// Set the converter type for the upsampling and the downsampling
	// interpolator SRC_SINC_BEST_QUALITY
	_src_state_mic  = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);
	_src_state_spkr = src_new(SRC_SINC_BEST_QUALITY, 1, &_src_err);

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}

AudioRtpRTX::~AudioRtpRTX () {
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	_start.wait();

	try {
		this->terminate();
	} catch(...) {
		_debugException("! ARTP: Thread destructor didn't terminate correctly");
		throw;
	}
	//_debug("terminate audiortprtx ended...\n");
	_ca = 0;

	if (!_sym) {
		delete _sessionRecv; _sessionRecv = NULL;
		delete _sessionSend; _sessionSend = NULL;
	} else {
		delete _session;     _session = NULL;
	}

	delete [] _intBufferDown; _intBufferDown = NULL;
	delete [] _floatBufferUp; _floatBufferUp = NULL;
	delete [] _floatBufferDown; _floatBufferDown = NULL;
	delete [] _dataAudioLayer; _dataAudioLayer = NULL;

	delete [] _sendDataEncoded; _sendDataEncoded = NULL;
	delete [] _receiveDataDecoded; _receiveDataDecoded = NULL;

	delete time; time = NULL;

	// libsamplerate-related
	_src_state_mic  = src_delete(_src_state_mic);
	_src_state_spkr = src_delete(_src_state_spkr);
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}

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	void
AudioRtpRTX::initBuffers()
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{
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	int nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
	_dataAudioLayer = new SFLDataFormat[nbSamplesMax];
	_receiveDataDecoded = new int16[nbSamplesMax];
	_floatBufferDown  = new float32[nbSamplesMax];
	_floatBufferUp = new float32[nbSamplesMax];
	_sendDataEncoded = new unsigned char[nbSamplesMax];
	_intBufferDown = new int16[nbSamplesMax];
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}

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	void
AudioRtpRTX::initAudioRtpSession (void) 
{
	try {
		if (_ca == 0) { return; }

		//_debug("Init audio RTP session\n");
		ost::InetHostAddress remote_ip(_ca->getRemoteIp().c_str());
		if (!remote_ip) {
			_debug("! ARTP Thread Error: Target IP address [%s] is not correct!\n", _ca->getRemoteIp().data());
			return;
		}

		// Initialization
		if (!_sym) {
			_sessionRecv->setSchedulingTimeout (10000);
			_sessionRecv->setExpireTimeout(1000000);

			_sessionSend->setSchedulingTimeout(10000);
			_sessionSend->setExpireTimeout(1000000);
		} else {
			_session->setSchedulingTimeout(10000);
			_session->setExpireTimeout(1000000);
		}

		if (!_sym) {
			if ( !_sessionRecv->addDestination(remote_ip, (unsigned short) _ca->getRemoteAudioPort()) ) {
				_debug("AudioRTP Thread Error: could not connect to port %d\n",  _ca->getRemoteAudioPort());
				return;
			}
			if (!_sessionSend->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
				_debug("! ARTP Thread Error: could not connect to port %d\n",  _ca->getRemoteAudioPort());
				return;
			}

			AudioCodec* audiocodec = _ca->getAudioCodec();
			bool payloadIsSet = false;
			if (audiocodec) {
				if (audiocodec->hasDynamicPayload()) {
					payloadIsSet = _sessionRecv->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) audiocodec->getPayload(), audiocodec->getClockRate()));
				} else {
					payloadIsSet= _sessionRecv->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
					payloadIsSet = _sessionSend->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
				}
			}
			_sessionSend->setMark(true);
		} else {

			//_debug("AudioRTP Thread: Added session destination %s:%d\n", remote_ip.getHostname(), (unsigned short) _ca->getRemoteSdpAudioPort());

			if (!_session->addDestination (remote_ip, (unsigned short) _ca->getRemoteAudioPort())) {
				return;
			}

			AudioCodec* audiocodec = _ca->getAudioCodec();
			bool payloadIsSet = false;
			if (audiocodec) {
				if (audiocodec->hasDynamicPayload()) {
					payloadIsSet = _session->setPayloadFormat(ost::DynamicPayloadFormat((ost::PayloadType) audiocodec->getPayload(), audiocodec->getClockRate()));
				} else {
					payloadIsSet = _session->setPayloadFormat(ost::StaticPayloadFormat((ost::StaticPayloadType) audiocodec->getPayload()));
				}
			}
		}
	} catch(...) {
		_debugException("! ARTP Failure: initialisation failed");
		throw;
	}
}
	
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void
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AudioRtpRTX::sendSessionFromMic(int timestamp)
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{
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	// STEP:
	//   1. get data from mic
	//   2. convert it to int16 - good sample, good rate
	//   3. encode it
	//   4. send it
	try {
		int16* toSIP = NULL;

		timestamp += time->getSecond();
		if (_ca==0) { _debug(" !ARTP: No call associated (mic)\n"); return; } // no call, so we do nothing
		AudioLayer* audiolayer = Manager::instance().getAudioDriver();
		if (!audiolayer) { _debug(" !ARTP: No audiolayer available for mic\n"); return; }

		AudioCodec* audiocodec = _ca->getAudioCodec();
		if (!audiocodec) { _debug(" !ARTP: No audiocodec available for mic\n"); return; }

		// we have to get 20ms of data from the mic *20/1000 = /50
		int maxBytesToGet = _layerSampleRate * _layerFrameSize * sizeof(SFLDataFormat) / 1000;

		// available bytes inside ringbuffer
		int availBytesFromMic = audiolayer->canGetMic();

		// take the lowest
		int bytesAvail = (availBytesFromMic < maxBytesToGet) ? availBytesFromMic : maxBytesToGet;

		// Get bytes from micRingBuffer to data_from_mic
		int nbSample = audiolayer->getMic(_dataAudioLayer, bytesAvail) / sizeof(SFLDataFormat);
		int nb_sample_up = nbSample;
		int nbSamplesMax = _layerFrameSize * audiocodec->getClockRate() / 1000;
		
		nbSample = reSampleData(audiocodec->getClockRate(), nb_sample_up, DOWN_SAMPLING);	
		
		toSIP = _intBufferDown;
		
		if ( nbSample < nbSamplesMax - 10 ) { // if only 10 is missing, it's ok
			// fill end with 0...
			//_debug("begin: %p, nbSample: %d\n", toSIP, nbSample);
			memset(toSIP + nbSample, 0, (nbSamplesMax-nbSample)*sizeof(int16));
			nbSample = nbSamplesMax;
		}
		//_debug("AR: Nb sample: %d int, [0]=%d [1]=%d [2]=%d\n", nbSample, toSIP[0], toSIP[1], toSIP[2]);

		// for the mono: range = 0 to RTP_FRAME2SEND * sizeof(int16)
		// codecEncode(char *dest, int16* src, size in bytes of the src)
		int compSize = audiocodec->codecEncode(_sendDataEncoded, toSIP, nbSample*sizeof(int16));

		// encode divise by two
		// Send encoded audio sample over the network
		if (compSize > nbSamplesMax) { _debug("! ARTP: %d should be %d\n", compSize, nbSamplesMax);}
		if (!_sym) {
			_sessionSend->putData(timestamp, _sendDataEncoded, compSize);
		} else {
			_session->putData(timestamp, _sendDataEncoded, compSize);
		}
		toSIP = NULL;
	} catch(...) {
		_debugException("! ARTP: sending failed");
		throw;
	}
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}

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	void
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AudioRtpRTX::receiveSessionForSpkr (int& countTime)
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{
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	if (_ca == 0) { return; }
	try {
		AudioLayer* audiolayer = Manager::instance().getAudioDriver();
		if (!audiolayer) { return; }

		const ost::AppDataUnit* adu = NULL;
		// Get audio data stream

		if (!_sym) {
			adu = _sessionRecv->getData(_sessionRecv->getFirstTimestamp());
		} else {
			adu = _session->getData(_session->getFirstTimestamp());
		}
		if (adu == NULL) {
			return;
		}

		int payload = adu->getType(); // codec type
		unsigned char* data  = (unsigned char*)adu->getData(); // data in char
		unsigned int size    = adu->getSize(); // size in char
		  

		// Decode data with relevant codec
		AudioCodec* audiocodec = _ca->getCodecMap().getCodec((CodecType)payload);
		_codecSampleRate = audiocodec->getClockRate();
		int max = (int)(_codecSampleRate * _layerFrameSize);

		if ( size > max ) {
                        _debug("We have received from RTP a packet larger than expected: %s VS %s\n", size, max);
                        _debug("The packet size has been cropped\n");
                        size=max;
                }


		if (audiocodec != NULL) {
			int expandedSize = audiocodec->codecDecode(_receiveDataDecoded, data, size);
			//buffer _receiveDataDecoded ----> short int or int16, coded on 2 bytes
			int nbInt16 = expandedSize / sizeof(int16);
			//nbInt16 represents the number of samples we just decoded
			if (nbInt16 > max) {
				_debug("We have decoded an RTP packet larger than expected: %s VS %s. Cropping.\n", nbInt16, max);
				nbInt16=max;
			}

			SFLDataFormat* toAudioLayer;
			int nbSample = nbInt16;

			// Do sample rate conversion
			int nb_sample_down = nbSample;
			nbSample = reSampleData(_codecSampleRate , nb_sample_down, UP_SAMPLING);
#ifdef DATAFORMAT_IS_FLOAT
			toAudioLayer = _floatBufferUp;
#else
			toAudioLayer = _dataAudioLayer;
#endif


			audiolayer->putMain(toAudioLayer, nbSample * sizeof(SFLDataFormat));

			// Notify (with a beep) an incoming call when there is already a call 
			countTime += time->getSecond();
			if (Manager::instance().incomingCallWaiting() > 0) {
				countTime = countTime % 500; // more often...
				if (countTime == 0) {
					Manager::instance().notificationIncomingCall();
				}
			}

		} else {
			countTime += time->getSecond();
		}

		delete adu; adu = NULL;
	} catch(...) {
		_debugException("! ARTP: receiving failed");
		throw;
	}
}
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int 
AudioRtpRTX::reSampleData(int sampleRate_codec, int nbSamples, int status)
{
	if(status==UP_SAMPLING)
		return upSampleData(sampleRate_codec, nbSamples);
	else if(status==DOWN_SAMPLING)
		return downSampleData(sampleRate_codec, nbSamples);
	else
		return 0;
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}

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////////////////////////////////////////////////////////////////////
//////////// RESAMPLING FUNCTIONS /////////////////////////////////
//////////////////////////////////////////////////////////////////

	int
AudioRtpRTX::upSampleData(int sampleRate_codec, int nbSamples)
{
	double upsampleFactor = (double) _layerSampleRate / sampleRate_codec;
	int nbSamplesMax = (int) (_layerSampleRate * _layerFrameSize /1000);
	if( upsampleFactor != 1 )
	{
		SRC_DATA src_data;
		src_data.data_in = _floatBufferDown;
		src_data.data_out = _floatBufferUp;
		src_data.input_frames = nbSamples;
		src_data.output_frames = (int) floor(upsampleFactor * nbSamples);
		src_data.src_ratio = upsampleFactor;
		src_data.end_of_input = 0; // More data will come
		src_short_to_float_array(_receiveDataDecoded, _floatBufferDown, nbSamples);
		src_process(_src_state_spkr, &src_data);
		nbSamples  = ( src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;		
		src_float_to_short_array(_floatBufferUp, _dataAudioLayer, nbSamples);
	}

	return nbSamples;
}	

	int
AudioRtpRTX::downSampleData(int sampleRate_codec, int nbSamples)
{
	double downsampleFactor = (double) sampleRate_codec / _layerSampleRate;
	int nbSamplesMax = (int) (sampleRate_codec * _layerFrameSize / 1000);
	if ( downsampleFactor != 1)
	{
		SRC_DATA src_data;	
		src_data.data_in = _floatBufferUp;
		src_data.data_out = _floatBufferDown;
		src_data.input_frames = nbSamples;
		src_data.output_frames = (int) floor(downsampleFactor * nbSamples);
		src_data.src_ratio = downsampleFactor;
		src_data.end_of_input = 0; // More data will come
		src_short_to_float_array(_dataAudioLayer, _floatBufferUp, nbSamples);
		src_process(_src_state_mic, &src_data);
		nbSamples  = ( src_data.output_frames_gen > nbSamplesMax) ? nbSamplesMax : src_data.output_frames_gen;
		src_float_to_short_array(_floatBufferDown, _intBufferDown, nbSamples);
	}
	return nbSamples;

}

//////////////////////// END RESAMPLING //////////////////////////////////////////////////////

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void
AudioRtpRTX::run () {
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	//mic, we receive from soundcard in stereo, and we send encoded
	//encoding before sending
	AudioLayer *audiolayer = Manager::instance().getAudioDriver();

	_layerFrameSize = audiolayer->getFrameSize(); // en ms
	_layerSampleRate = audiolayer->getSampleRate();	
	initBuffers();
	int step = (int)(_layerFrameSize * _codecSampleRate / 1000);

	try {
		// Init the session
		initAudioRtpSession();

		// start running the packet queue scheduler.
		//_debug("AudioRTP Thread started\n");
		if (!_sym) {
			_sessionRecv->startRunning();
			_sessionSend->startRunning();
		} else {
			_session->startRunning();
			//_debug("Session is now: %d active\n", _session->isActive());
		}

		int timestamp = 0; // for mic
		int countTime = 0; // for receive
		TimerPort::setTimer(_layerFrameSize);

		audiolayer->flushMic();
		audiolayer->startStream();
		_start.post();
		_debug("- ARTP Action: Start\n");
		while (!testCancel()) {
			////////////////////////////
			// Send session
			////////////////////////////
			sendSessionFromMic(timestamp);
			timestamp += step;
			////////////////////////////
			// Recv session
			////////////////////////////
			receiveSessionForSpkr(countTime);

			// Let's wait for the next transmit cycle
			Thread::sleep(TimerPort::getTimer());
			TimerPort::incTimer(_layerFrameSize); // 'frameSize' ms
		}
		//_debug("stop stream for audiortp loop\n");
		audiolayer->stopStream();
	} catch(std::exception &e) {
		_start.post();
		_debug("! ARTP: Stop %s\n", e.what());
		throw;
	} catch(...) {
		_start.post();
		_debugException("* ARTP Action: Stop");
		throw;
	}
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}


// EOF