Implemented RTSP Receiver Report
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parent
364190034d
commit
df6735e92f
5 changed files with 86 additions and 24 deletions
38
lib/rtp.cpp
38
lib/rtp.cpp
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@ -4,6 +4,7 @@
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#include "timing.h"
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#include "bitfields.h"
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#include "mpeg.h"
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#include "sdp.h"
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#include <arpa/inet.h>
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namespace RTP{
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@ -245,17 +246,18 @@ namespace RTP{
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increaseSequence();
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}
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void Packet::sendRTCP(long long &connectedAt, void *socket, unsigned int tid,
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void Packet::sendRTCP_SR(long long &connectedAt, void *socket, unsigned int tid,
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DTSC::Meta &metadata,
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void callBack(void *, char *, unsigned int, unsigned int)){
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void *rtcpData = malloc(32);
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char *rtcpData = (char*)malloc(32);
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if (!rtcpData){
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FAIL_MSG("Could not allocate 32 bytes. Something is seriously messed up.");
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return;
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}
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((int *)rtcpData)[0] = htonl(0x80C80006);
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((int *)rtcpData)[1] = htonl(getSSRC());
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// unsigned int tid = packet["trackid"].asInt();
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rtcpData[0] = 0x80;//version 2, no padding, zero receiver reports
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rtcpData[1] = 200;//sender report
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Bit::htobs(rtcpData+2, 6);//6 4-byte words follow the header
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Bit::htobl(rtcpData+4, getSSRC());//set source identifier
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// timestamp in ms
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double ntpTime = 2208988800UL + Util::epoch() + (Util::getMS() % 1000) / 1000.0;
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if (startRTCP < 1 && startRTCP > -1){startRTCP = ntpTime;}
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@ -276,6 +278,32 @@ namespace RTP{
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free(rtcpData);
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}
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void Packet::sendRTCP_RR(long long &connectedAt, SDP::Track & sTrk, unsigned int tid,
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DTSC::Meta &metadata,
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void callBack(void *, char *, unsigned int, unsigned int)){
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char *rtcpData = (char*)malloc(32);
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if (!rtcpData){
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FAIL_MSG("Could not allocate 32 bytes. Something is seriously messed up.");
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return;
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}
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if (!(sTrk.lostCurrent + sTrk.packCurrent)){sTrk.packCurrent++;}
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rtcpData[0] = 0x81;//version 2, no padding, one receiver report
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rtcpData[1] = 201;//receiver report
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Bit::htobs(rtcpData+2, 7);//7 4-byte words follow the header
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Bit::htobl(rtcpData+4, sTrk.mySSRC);//set receiver identifier
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Bit::htobl(rtcpData+8, sTrk.theirSSRC);//set source identifier
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rtcpData[12] = (sTrk.lostCurrent * 255) / (sTrk.lostCurrent + sTrk.packCurrent); //fraction lost since prev RR
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Bit::htob24(rtcpData+13, sTrk.lostTotal); //cumulative packets lost since start
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Bit::htobl(rtcpData+16, sTrk.rtpSeq | (sTrk.packTotal & 0xFFFF0000ul)); //highest sequence received
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Bit::htobl(rtcpData+20, 0); /// \TODO jitter (diff in timestamp vs packet arrival)
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Bit::htobl(rtcpData+24, 0); /// \TODO last SR (middle 32 bits of last SR or zero)
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Bit::htobl(rtcpData+28, 0); /// \TODO delay since last SR in 2b seconds + 2b fraction
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callBack(&(sTrk.rtcp), (char *)rtcpData, 32, 0);
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sTrk.lostCurrent = 0;
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sTrk.packCurrent = 0;
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free(rtcpData);
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}
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Packet::Packet(){
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managed = false;
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data = 0;
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@ -15,6 +15,10 @@
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#include <string>
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#include <vector>
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namespace SDP{
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class Track;
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};
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/// This namespace holds all RTP-parsing and sending related functionality.
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namespace RTP{
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@ -59,7 +63,9 @@ namespace RTP{
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void sendData(void *socket, void callBack(void *, char *, unsigned int, unsigned int),
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const char *payload, unsigned int payloadlen, unsigned int channel,
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std::string codec);
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void sendRTCP(long long &connectedAt, void *socket, unsigned int tid, DTSC::Meta &metadata,
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void sendRTCP_SR(long long &connectedAt, void *socket, unsigned int tid, DTSC::Meta &metadata,
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void callBack(void *, char *, unsigned int, unsigned int));
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void sendRTCP_RR(long long &connectedAt, SDP::Track & sTrk, unsigned int tid, DTSC::Meta &metadata,
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void callBack(void *, char *, unsigned int, unsigned int));
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Packet();
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32
lib/sdp.cpp
32
lib/sdp.cpp
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@ -15,9 +15,14 @@ namespace SDP{
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packCount = 0;
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cPort = 0;
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rtpSeq = 0;
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lostTotal = 0;
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lostCurrent = 0;
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packTotal = 0;
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packCurrent = 0;
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fpsTime = 0;
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fpsMeta = 0;
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fps = 0;
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mySSRC = rand();
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}
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/// Extracts a particular parameter from the fmtp string. fmtp member must be set before calling.
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@ -175,35 +180,34 @@ namespace SDP{
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/// \return True if successful, false otherwise.
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bool Track::parseTransport(const std::string &transport, const std::string &host,
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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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pack = RTP::Packet(97, 1, 0, mySSRC);
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}else if (trk.codec == "HEVC"){
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pack = RTP::Packet(104, 1, 0, SSrc);
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pack = RTP::Packet(104, 1, 0, mySSRC);
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}else if (trk.codec == "MPEG2"){
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pack = RTP::Packet(32, 1, 0, SSrc);
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pack = RTP::Packet(32, 1, 0, mySSRC);
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}else if (trk.codec == "AAC"){
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pack = RTP::Packet(96, 1, 0, SSrc);
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pack = RTP::Packet(96, 1, 0, mySSRC);
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}else if (trk.codec == "AC3"){
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pack = RTP::Packet(100, 1, 0, SSrc);
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pack = RTP::Packet(100, 1, 0, mySSRC);
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}else if (trk.codec == "MP3" || trk.codec == "MP2"){
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pack = RTP::Packet(14, 1, 0, SSrc);
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pack = RTP::Packet(14, 1, 0, mySSRC);
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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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pack = RTP::Packet(8, 1, 0, mySSRC);
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}else{
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pack = RTP::Packet(101, 1, 0, SSrc);
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pack = RTP::Packet(101, 1, 0, mySSRC);
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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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pack = RTP::Packet(10, 1, 0, mySSRC);
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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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pack = RTP::Packet(11, 1, 0, mySSRC);
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}else{
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pack = RTP::Packet(103, 1, 0, SSrc);
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pack = RTP::Packet(103, 1, 0, mySSRC);
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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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pack = RTP::Packet(102, 1, 0, mySSRC);
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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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@ -231,7 +235,7 @@ namespace SDP{
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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()){tStr << "source=" << source << ";";}
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tStr << "server_port=" << portA << "-" << portB << ";ssrc=" << std::hex << SSrc << std::dec;
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tStr << "server_port=" << portA << "-" << portB << ";ssrc=" << std::hex << mySSRC << 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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@ -19,6 +19,8 @@ namespace SDP{
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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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int32_t lostTotal, lostCurrent;
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uint32_t packTotal, packCurrent;
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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; /// Current transport string.
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@ -26,6 +28,7 @@ namespace SDP{
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std::string fmtp; /// fmtp string, used by getParamString / getParamInt
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std::string spsData;
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std::string ppsData;
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uint32_t mySSRC, theirSSRC;
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h265::initData hevcInfo;
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uint64_t fpsTime;
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double fpsMeta;
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@ -119,7 +119,7 @@ namespace Mist{
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callBack = sendUDP;
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if (Util::epoch() / 5 != sdpState.tracks[tid].rtcpSent){
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sdpState.tracks[tid].rtcpSent = Util::epoch() / 5;
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sdpState.tracks[tid].pack.sendRTCP(connectedAt, &sdpState.tracks[tid].rtcp, tid, myMeta,
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sdpState.tracks[tid].pack.sendRTCP_SR(connectedAt, &sdpState.tracks[tid].rtcp, tid, myMeta,
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sendUDP);
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}
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}else{
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@ -381,20 +381,28 @@ namespace Mist{
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myConn.addDown(s.data_len);
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RTP::Packet pack(s.data, s.data_len);
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if (!it->second.rtpSeq){it->second.rtpSeq = pack.getSequence();}
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// packet is very early - assume dropped after 10 packets
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while ((int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())) < -10){
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// packet is very early - assume dropped after 30 packets
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while ((int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())) < -30){
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WARN_MSG("Giving up on packet %u", it->second.rtpSeq);
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++(it->second.rtpSeq);
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++(it->second.lostTotal);
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++(it->second.lostCurrent);
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++(it->second.packTotal);
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++(it->second.packCurrent);
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// send any buffered packets we may have
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while (it->second.packBuffer.count(it->second.rtpSeq)){
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sdpState.handleIncomingRTP(it->first, pack);
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++(it->second.rtpSeq);
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++(it->second.packTotal);
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++(it->second.packCurrent);
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}
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}
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// send any buffered packets we may have
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while (it->second.packBuffer.count(it->second.rtpSeq)){
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sdpState.handleIncomingRTP(it->first, pack);
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++(it->second.rtpSeq);
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++(it->second.packTotal);
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++(it->second.packCurrent);
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}
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// packet is slightly early - buffer it
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if (((int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())) < 0)){
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// packet is late
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if ((int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())) > 0){
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// negative difference?
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--(it->second.lostTotal);
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--(it->second.lostCurrent);
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++(it->second.packTotal);
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++(it->second.packCurrent);
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WARN_MSG("Dropped a packet that arrived too late! (%d packets difference)",
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(int16_t)(((uint16_t)it->second.rtpSeq) - ((uint16_t)pack.getSequence())));
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return;
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if (it->second.rtpSeq == pack.getSequence()){
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sdpState.handleIncomingRTP(it->first, pack);
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++(it->second.rtpSeq);
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++(it->second.packTotal);
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++(it->second.packCurrent);
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if (!it->second.theirSSRC){
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it->second.theirSSRC = pack.getSSRC();
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}
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}
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}
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if (Util::epoch() / 5 != it->second.rtcpSent){
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it->second.rtcpSent = Util::epoch() / 5;
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it->second.pack.sendRTCP_RR(connectedAt, it->second, it->first, myMeta, sendUDP);
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}
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}
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}
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}
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