int OutputProcessor::packageAudio(unsigned char* inBuff, int inBuffLen,
unsigned char* outBuff, long int pts) {
if (audioPackager == 0) {
ELOG_DEBUG("No se ha inicializado el codec de output audio RTP");
return -1;
}
timeval time;
gettimeofday(&time, NULL);
long millis = (time.tv_sec * 1000) + (time.tv_usec / 1000);
RtpHeader head;
head.setSeqNumber(audioSeqnum_++);
// head.setTimestamp(millis*8);
head.setMarker(1);
if (pts==0){
// head.setTimestamp(audioSeqnum_*160);
head.setTimestamp(av_rescale(audioSeqnum_, (mediaInfo.audioCodec.sampleRate/1000), 1));
}else{
// head.setTimestamp(pts*8);
head.setTimestamp(av_rescale(pts, mediaInfo.audioCodec.sampleRate,1000));
}
head.setSSRC(44444);
head.setPayloadType(mediaInfo.rtpAudioInfo.PT);
// memcpy (rtpAudioBuffer_, &head, head.getHeaderLength());
// memcpy(&rtpAudioBuffer_[head.getHeaderLength()], inBuff, inBuffLen);
memcpy (outBuff, &head, head.getHeaderLength());
memcpy(&outBuff[head.getHeaderLength()], inBuff, inBuffLen);
// sink_->sendData(rtpBuffer_, l);
// rtpReceiver_->receiveRtpData(rtpBuffer_, (inBuffLen + RTP_HEADER_LEN));
return (inBuffLen+head.getHeaderLength());
}
void WebRtcConnection::writeSsrc(char* buf, int len, unsigned int ssrc) {
ELOG_DEBUG("LEN %d", len);
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
//if it is RTCP we check it it is a compound packet
if (chead->isRtcp()) {
char* movingBuf = buf;
int rtcpLength = 0;
int totalLength = 0;
do{
movingBuf+=rtcpLength;
RtcpHeader *chead= reinterpret_cast<RtcpHeader*>(movingBuf);
rtcpLength= (ntohs(chead->length)+1)*4;
totalLength+= rtcpLength;
ELOG_DEBUG("Is RTCP, prev SSRC %u, new %u, len %d ", chead->getSSRC(), ssrc, rtcpLength);
chead->ssrc=htonl(ssrc);
if (chead->packettype == RTCP_PS_Feedback_PT){
FirHeader *thefir = reinterpret_cast<FirHeader*>(movingBuf);
if (thefir->fmt == 4){ // It is a FIR Packet, we generate it
this->sendPLI();
}
}
} while(totalLength<len);
} else {
head->setSSRC(ssrc);
}
}
int OutputProcessor::packageVideo(unsigned char* inBuff, int buffSize, unsigned char* outBuff,
long int pts) {
if (videoPackager == 0) {
ELOG_DEBUG("No se ha inicailizado el codec de output vídeo RTP");
return -1;
}
// ELOG_DEBUG("To packetize %u", buffSize);
if (buffSize <= 0)
return -1;
RtpVP8Fragmenter frag(inBuff, buffSize, 1100);
bool lastFrame = false;
unsigned int outlen = 0;
timeval time;
gettimeofday(&time, NULL);
long millis = (time.tv_sec * 1000) + (time.tv_usec / 1000);
// timestamp_ += 90000 / mediaInfo.videoCodec.frameRate;
//int64_t pts = av_rescale(lastPts_, 1000000, (long int)video_time_base_);
do {
outlen = 0;
frag.getPacket(outBuff, &outlen, &lastFrame);
RtpHeader rtpHeader;
rtpHeader.setMarker(lastFrame?1:0);
rtpHeader.setSeqNumber(seqnum_++);
if (pts==0){
rtpHeader.setTimestamp(av_rescale(millis, 90000, 1000));
}else{
rtpHeader.setTimestamp(av_rescale(pts, 90000, 1000));
}
rtpHeader.setSSRC(55543);
rtpHeader.setPayloadType(100);
memcpy(rtpBuffer_, &rtpHeader, rtpHeader.getHeaderLength());
memcpy(&rtpBuffer_[rtpHeader.getHeaderLength()],outBuff, outlen);
int l = outlen + rtpHeader.getHeaderLength();
// sink_->sendData(rtpBuffer_, l);
rtpReceiver_->receiveRtpData(rtpBuffer_, l);
} while (!lastFrame);
return 0;
}
void WebRtcConnection::onTransportData(char* buf, int len, Transport *transport) {
if (audioSink_ == NULL && videoSink_ == NULL && fbSink_==NULL){
return;
}
// PROCESS STATS
if (this->statsListener_){ // if there is no listener we dont process stats
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
if (head->payloadtype != RED_90000_PT && head->payloadtype != PCMU_8000_PT)
thisStats_.processRtcpPacket(buf, len);
}
RtcpHeader* chead = reinterpret_cast<RtcpHeader*>(buf);
// DELIVER FEEDBACK (RR, FEEDBACK PACKETS)
if (chead->isFeedback()){
if (fbSink_ != NULL) {
fbSink_->deliverFeedback(buf,len);
}
} else {
// RTP or RTCP Sender Report
if (bundle_) {
// Check incoming SSRC
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
unsigned int recvSSRC;
if (chead->packettype == RTCP_Sender_PT) { //Sender Report
recvSSRC = chead->getSSRC();
}else{
recvSSRC = head->getSSRC();
}
// Deliver data
if (recvSSRC==this->getVideoSourceSSRC() || recvSSRC==this->getVideoSinkSSRC()) {
videoSink_->deliverVideoData(buf, len);
} else if (recvSSRC==this->getAudioSourceSSRC() || recvSSRC==this->getAudioSinkSSRC()) {
audioSink_->deliverAudioData(buf, len);
} else {
ELOG_ERROR("Unknown SSRC %u, localVideo %u, remoteVideo %u, ignoring", recvSSRC, this->getVideoSourceSSRC(), this->getVideoSinkSSRC());
}
} else if (transport->mediaType == AUDIO_TYPE) {
if (audioSink_ != NULL) {
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
// Firefox does not send SSRC in SDP
if (this->getAudioSourceSSRC() == 0) {
ELOG_DEBUG("Audio Source SSRC is %u", head->getSSRC());
this->setAudioSourceSSRC(head->getSSRC());
//this->updateState(TRANSPORT_READY, transport);
}
head->setSSRC(this->getAudioSinkSSRC());
audioSink_->deliverAudioData(buf, len);
}
} else if (transport->mediaType == VIDEO_TYPE) {
if (videoSink_ != NULL) {
RtpHeader *head = reinterpret_cast<RtpHeader*> (buf);
RtcpHeader *chead = reinterpret_cast<RtcpHeader*> (buf);
// Firefox does not send SSRC in SDP
if (this->getVideoSourceSSRC() == 0) {
unsigned int recvSSRC;
if (chead->packettype == RTCP_Sender_PT) { //Sender Report
recvSSRC = chead->getSSRC();
} else {
recvSSRC = head->getSSRC();
}
ELOG_DEBUG("Video Source SSRC is %u", recvSSRC);
this->setVideoSourceSSRC(recvSSRC);
//this->updateState(TRANSPORT_READY, transport);
}
// change ssrc for RTP packets, don't touch here if RTCP
if (chead->packettype != RTCP_Sender_PT) {
head->setSSRC(this->getVideoSinkSSRC());
}
videoSink_->deliverVideoData(buf, len);
}
}
}
}
