RTP cleanup + RTSP analyser + RTP MPEG2/MP2/HEVC support
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Thulinma
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9 changed files with 1651 additions and 1216 deletions
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@ -1,214 +1,36 @@
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#pragma once
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#include "output.h"
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#include <mist/socket.h>
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#include <mist/rtp.h>
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#include <mist/http_parser.h>
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#include <mist/encode.h>
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#include <mist/h264.h>
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#include <mist/http_parser.h>
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#include <mist/rtp.h>
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#include <mist/sdp.h>
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#include <mist/socket.h>
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namespace Mist {
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///Structure used to keep track of selected tracks.
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class RTPTrack {
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public:
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Socket::UDPConnection data;
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Socket::UDPConnection rtcp;
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RTP::Packet pack;
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long long rtcpSent;
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uint64_t firstTime;
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int channel;/// Channel number, used in TCP sending
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uint64_t packCount;
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uint16_t rtpSeq;
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std::map<uint16_t, RTP::Packet> packBuffer;
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uint32_t cPort;
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std::string transportString;
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std::string control;
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std::string fmtp;
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std::string spsData;
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std::string ppsData;
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uint64_t fpsTime;
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double fpsMeta;
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double fps;
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RTPTrack(){
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rtcpSent = 0;
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channel = -1;
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firstTime = 0;
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packCount = 0;
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cPort = 0;
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rtpSeq = 0;
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fpsTime = 0;
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fpsMeta = 0;
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fps = 0;
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}
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std::string getParamString(const std::string & param) const{
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if (!fmtp.size()){return "";}
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size_t pos = fmtp.find(param);
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if (pos == std::string::npos){return "";}
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pos += param.size()+1;
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size_t ePos = fmtp.find_first_of(" ;", pos);
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return fmtp.substr(pos, ePos-pos);
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}
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uint64_t getParamInt(const std::string & param) const{
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return atoll(getParamString(param).c_str());
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}
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std::string mediaDescription(const DTSC::Track & trk){
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std::stringstream mediaDesc;
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if (trk.codec == "H264") {
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MP4::AVCC avccbox;
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avccbox.setPayload(trk.init);
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mediaDesc << "m=video 0 RTP/AVP 97\r\n"
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"a=rtpmap:97 H264/90000\r\n"
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"a=cliprect:0,0," << trk.height << "," << trk.width << "\r\n"
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"a=framesize:97 " << trk.width << '-' << trk.height << "\r\n"
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"a=fmtp:97 packetization-mode=1;profile-level-id="
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<< std::hex << std::setw(2) << std::setfill('0') << (int)trk.init.data()[1] << std::dec << "E0"
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<< std::hex << std::setw(2) << std::setfill('0') << (int)trk.init.data()[3] << std::dec << ";"
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"sprop-parameter-sets="
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<< Encodings::Base64::encode(std::string(avccbox.getSPS(), avccbox.getSPSLen()))
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<< ","
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<< Encodings::Base64::encode(std::string(avccbox.getPPS(), avccbox.getPPSLen()))
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<< "\r\n"
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"a=framerate:" << ((double)trk.fpks)/1000.0 << "\r\n"
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"a=control:track" << trk.trackID << "\r\n";
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} else if (trk.codec == "AAC") {
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mediaDesc << "m=audio 0 RTP/AVP 96" << "\r\n"
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"a=rtpmap:96 mpeg4-generic/" << trk.rate << "/" << trk.channels << "\r\n"
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"a=fmtp:96 streamtype=5; profile-level-id=15; config=";
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for (unsigned int i = 0; i < trk.init.size(); i++) {
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mediaDesc << std::hex << std::setw(2) << std::setfill('0') << (int)trk.init[i] << std::dec;
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}
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//these values are described in RFC 3640
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mediaDesc << "; mode=AAC-hbr; SizeLength=13; IndexLength=3; IndexDeltaLength=3;\r\n"
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"a=control:track" << trk.trackID << "\r\n";
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}else if (trk.codec == "MP3") {
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mediaDesc << "m=" << trk.type << " 0 RTP/AVP 14" << "\r\n"
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"a=rtpmap:14 MPA/90000/" << trk.channels << "\r\n"
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"a=control:track" << trk.trackID << "\r\n";
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}else if ( trk.codec == "AC3") {
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mediaDesc << "m=audio 0 RTP/AVP 100" << "\r\n"
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"a=rtpmap:100 AC3/" << trk.rate << "/" << trk.channels << "\r\n"
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"a=control:track" << trk.trackID << "\r\n";
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}else if ( trk.codec == "ALAW") {
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if (trk.channels == 1 && trk.rate == 8000){
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mediaDesc << "m=audio 0 RTP/AVP 8" << "\r\n";
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}else{
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mediaDesc << "m=audio 0 RTP/AVP 101" << "\r\n";
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mediaDesc << "a=rtpmap:101 PCMA/" << trk.rate << "/" << trk.channels << "\r\n";
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}
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mediaDesc << "a=control:track" << trk.trackID << "\r\n";
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}else if ( trk.codec == "PCM") {
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if (trk.size == 16 && trk.channels == 2 && trk.rate == 44100){
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mediaDesc << "m=audio 0 RTP/AVP 10" << "\r\n";
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} else if (trk.size == 16 && trk.channels == 1 && trk.rate == 44100){
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mediaDesc << "m=audio 0 RTP/AVP 11" << "\r\n";
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}else{
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mediaDesc << "m=audio 0 RTP/AVP 103" << "\r\n";
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mediaDesc << "a=rtpmap:103 L" << trk.size << "/" << trk.rate << "/" << trk.channels << "\r\n";
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}
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mediaDesc << "a=control:track" << trk.trackID << "\r\n";
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}else if ( trk.codec == "opus") {
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mediaDesc << "m=audio 0 RTP/AVP 102" << "\r\n"
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"a=rtpmap:102 opus/" << trk.rate << "/" << trk.channels << "\r\n"
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"a=control:track" << trk.trackID << "\r\n";
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}
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return mediaDesc.str();
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}
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bool parseTransport(const std::string & transport, const std::string & host, const std::string & source, const DTSC::Track & trk){
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unsigned int SSrc = rand();
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if (trk.codec == "H264") {
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pack = RTP::Packet(97, 1, 0, SSrc);
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}else if(trk.codec == "AAC"){
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pack = RTP::Packet(96, 1, 0, SSrc);
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}else if(trk.codec == "AC3"){
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pack = RTP::Packet(100, 1, 0, SSrc);
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}else if(trk.codec == "MP3"){
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pack = RTP::Packet(14, 1, 0, SSrc);
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}else if(trk.codec == "ALAW"){
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if (trk.channels == 1 && trk.rate == 8000){
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pack = RTP::Packet(8, 1, 0, SSrc);
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}else{
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pack = RTP::Packet(101, 1, 0, SSrc);
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}
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}else if ( trk.codec == "PCM") {
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if (trk.size == 16 && trk.channels == 2 && trk.rate == 44100){
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pack = RTP::Packet(10, 1, 0, SSrc);
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} else if (trk.size == 16 && trk.channels == 1 && trk.rate == 44100){
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pack = RTP::Packet(11, 1, 0, SSrc);
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}else{
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pack = RTP::Packet(103, 1, 0, SSrc);
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}
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}else if(trk.codec == "opus"){
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pack = RTP::Packet(102, 1, 0, SSrc);
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}else{
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ERROR_MSG("Unsupported codec %s for RTSP on track %u", trk.codec.c_str(), trk.trackID);
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return false;
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}
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if (transport.find("TCP") != std::string::npos) {
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std::string chanE = transport.substr(transport.find("interleaved=") + 12, (transport.size() - transport.rfind('-') - 1)); //extract channel ID
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channel = atol(chanE.c_str());
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rtcpSent = 0;
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transportString = transport;
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} else {
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channel = -1;
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size_t port_loc = transport.rfind("client_port=") + 12;
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cPort = atol(transport.substr(port_loc, transport.rfind('-') - port_loc).c_str());
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uint32_t portA, portB;
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//find available ports locally;
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int sendbuff = 4*1024*1024;
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data.SetDestination(host, cPort);
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portA = data.bind(0);
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setsockopt(data.getSock(), SOL_SOCKET, SO_SNDBUF, &sendbuff, sizeof(sendbuff));
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rtcp.SetDestination(host, cPort + 1);
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portB = rtcp.bind(0);
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setsockopt(rtcp.getSock(), SOL_SOCKET, SO_SNDBUF, &sendbuff, sizeof(sendbuff));
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std::stringstream tStr;
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tStr << "RTP/AVP/UDP;unicast;client_port=" << cPort << '-' << cPort + 1 << ";";
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if (source.size()){
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tStr << "source=" << source << ";";
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}
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tStr << "server_port=" << portA << "-" << portB << ";ssrc=" << std::hex << SSrc << std::dec;
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transportString = tStr.str();
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INFO_MSG("Transport string: %s", transportString.c_str());
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}
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return true;
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}
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std::string rtpInfo(const DTSC::Track & trk, const std::string & source, uint64_t currentTime){
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unsigned int timeMultiplier = 1;
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timeMultiplier = ((double)trk.rate / 1000.0);
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if (trk.codec == "H264") {
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timeMultiplier = 90;
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}
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std::stringstream rInfo;
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rInfo << "url=" << source << "/track" << trk.trackID << ";"; //get the current url, not localhost
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rInfo << "sequence=" << pack.getSequence() << ";rtptime=" << currentTime * timeMultiplier;
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return rInfo.str();
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}
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};
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class OutRTSP : public Output {
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public:
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OutRTSP(Socket::Connection & myConn);
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static void init(Util::Config * cfg);
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void sendNext();
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void onRequest();
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void requestHandler();
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bool onFinish();
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private:
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void parseSDP(const std::string & sdp);
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long long connectedAt;///< The timestamp the connection was made, as reference point for RTCP packets.
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std::map<int, RTPTrack> tracks;///< List of selected tracks with RTSP-specific session data.
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unsigned int pausepoint;///< Position to pause at, when reached
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HTTP::Parser HTTP_R, HTTP_S;
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std::string source;
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uint64_t lastTimeSync;
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bool expectTCP;
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bool handleTCP();
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void handleUDP();
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void handleIncomingRTP(const uint64_t track, const RTP::Packet & pkt);
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void h264MultiParse(uint64_t ts, const uint64_t track, char * buffer, const uint32_t len);
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void h264Packet(uint64_t ts, const uint64_t track, const char * buffer, const uint32_t len, bool isKey);
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void updateH264Init(uint64_t trackNo);
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namespace Mist{
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class OutRTSP : public Output{
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public:
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OutRTSP(Socket::Connection &myConn);
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static void init(Util::Config *cfg);
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void sendNext();
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void onRequest();
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void requestHandler();
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bool onFinish();
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void incomingPacket(const DTSC::Packet &pkt);
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private:
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long long connectedAt; ///< The timestamp the connection was made, as reference point for RTCP
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///packets.
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unsigned int pausepoint; ///< Position to pause at, when reached
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SDP::State sdpState;
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HTTP::Parser HTTP_R, HTTP_S;
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std::string source;
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uint64_t lastTimeSync;
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bool expectTCP;
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bool handleTCP();
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void handleUDP();
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};
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}
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typedef Mist::OutRTSP mistOut;
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