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RTP library fixes and improvements.

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- Fixes marker bit bug in H264 output over RTP-based protocols
- Fixes RTCP packets going over the wrong channel in TCP-based RTSP
- Fixes RTSP RTCP packets having wrong timestamps
- Fixes payload padding parser bug in H264, H265, VP8 and VP9 RTP-based inputs.
- Cleans up WebRTC's needless use of thisPacket.getTime() to use thisTime instead
This commit is contained in:
Thulinma 2023-06-14 13:46:49 +02:00
parent 132e59db51
commit a1232d56af
5 changed files with 43 additions and 25 deletions
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34
lib/rtp.cpp
View file

@ -22,6 +22,11 @@ namespace RTP{
}
unsigned int Packet::getPayloadSize() const{
// If there is more padding than content, ignore the packet
if (getHsize() + (getPadding() ? data[maxDataLen - 1] : 0) >= maxDataLen){
WARN_MSG("Packet has more padding than payload; ignoring packet");
return 0;
}
return maxDataLen - getHsize() - (getPadding() ? data[maxDataLen - 1] : 0);
}
@ -273,7 +278,6 @@ namespace RTP{
void Packet::sendH264(void *socket, void callBack(void *, const char *, size_t, uint8_t),
const char *payload, uint32_t payloadlen, uint32_t channel, bool lastOfAccesUnit){
if ((payload[0] & 0x1F) == 12){return;}
/// \todo This function probably belongs in DMS somewhere.
if (payloadlen + getHsize() + 2 <= maxDataLen){
data[1] &= 0x7F; // setting the RTP marker bit to 0
@ -446,12 +450,22 @@ namespace RTP{
unsigned int payloadlen, unsigned int channel, std::string codec){
if (codec == "H264"){
unsigned long sent = 0;
const char * lastPtr = 0;
size_t lastLen = 0;
while (sent < payloadlen){
unsigned long nalSize = ntohl(*((unsigned long *)(payload + sent)));
sendH264(socket, callBack, payload + sent + 4, nalSize, channel,
(sent + nalSize + 4) >= payloadlen ? true : false);
// Since we skip filler data, we need to delay sending by one NAL unit to reliably
// detect the end of the access unit.
if ((payload[sent + 4] & 0x1F) != 12){
// If we have a pointer stored, we know it's not the last one, so send it as non-last.
if (lastPtr){sendH264(socket, callBack, lastPtr, lastLen, channel, false);}
lastPtr = payload + sent + 4;
lastLen = nalSize;
}
sent += nalSize + 4;
}
// Still a pointer stored? That means it was the last one. Mark it as such and send.
if (lastPtr){sendH264(socket, callBack, lastPtr, lastLen, channel, true);}
return;
}
if (codec == "VP8"){
@ -512,7 +526,7 @@ namespace RTP{
increaseSequence();
}
void Packet::sendRTCP_SR(void *socket, void callBack(void *, const char *, size_t, uint8_t)){
void Packet::sendRTCP_SR(void *socket, uint8_t channel, void callBack(void *, const char *, size_t, uint8_t)){
char *rtcpData = (char *)malloc(32);
if (!rtcpData){
FAIL_MSG("Could not allocate 32 bytes. Something is seriously messed up.");
@ -529,7 +543,7 @@ namespace RTP{
//*((int *)(rtcpData+16) ) = htonl(getTimeStamp());//rtpts
Bit::htobl(rtcpData + 20, sentPackets); // packet
Bit::htobl(rtcpData + 24, sentBytes); // octet
callBack(socket, (char *)rtcpData, 28, 0);
callBack(socket, (char *)rtcpData, 28, channel);
free(rtcpData);
}
@ -867,6 +881,10 @@ namespace RTP{
/// Adds an RTP packet to the converter, outputting DTSC packets and/or updating init data,
/// as-needed.
void toDTSC::addRTP(const RTP::Packet &pkt){
if (pkt.getPayloadType() >= 72 && pkt.getPayloadType() <= 76){
INFO_MSG("RTCP packet, ignoring for decoding");
return;
}
if (codec.empty()){
MEDIUM_MSG("Unknown codec - ignoring RTP packet.");
return;
@ -908,17 +926,17 @@ namespace RTP{
// From here on, there is codec-specific parsing. We call handler functions for each codec,
// except for the trivial codecs.
if (codec == "H264"){
return handleH264(msTime, pl, plSize, missed, (pkt.getPadding() == 1) ? true : false);
return handleH264(msTime, pl, plSize, missed, false);
}
if (codec == "AAC"){return handleAAC(msTime, pl, plSize);}
if (codec == "MP2" || codec == "MP3"){return handleMP2(msTime, pl, plSize);}
if (codec == "HEVC"){return handleHEVC(msTime, pl, plSize, missed);}
if (codec == "MPEG2"){return handleMPEG2(msTime, pl, plSize);}
if (codec == "VP8"){
return handleVP8(msTime, pl, plSize, missed, (pkt.getPadding() == 1) ? true : false);
return handleVP8(msTime, pl, plSize, missed, false);
}
if (codec == "VP9"){
return handleVP8(msTime, pl, plSize, missed, (pkt.getPadding() == 1) ? true : false);
return handleVP8(msTime, pl, plSize, missed, false);
}
// Trivial codecs just fill a packet with raw data and continue. Easy peasy, lemon squeezy.
if (codec == "ALAW" || codec == "opus" || codec == "PCM" || codec == "ULAW"){

2
lib/rtp.h
View file

@ -105,7 +105,7 @@ namespace RTP{
const char *payload, unsigned int payloadlen, unsigned int channel);
void sendData(void *socket, void callBack(void *, const char *, size_t, uint8_t), const char *payload,
unsigned int payloadlen, unsigned int channel, std::string codec);
void sendRTCP_SR(void *socket, void callBack(void *, const char *, size_t, uint8_t));
void sendRTCP_SR(void *socket, uint8_t channel, void callBack(void *, const char *, size_t, uint8_t));
void sendRTCP_RR(SDP::Track &sTrk, void callBack(void *, const char *, size_t, uint8_t));
Packet();

20
src/output/output_rtsp.cpp
View file

@ -148,19 +148,9 @@ namespace Mist{
if (sdpState.tracks[thisIdx].channel == -1){// UDP connection
socket = &sdpState.tracks[thisIdx].data;
callBack = sendUDP;
if (Util::bootSecs() != sdpState.tracks[thisIdx].rtcpSent){
sdpState.tracks[thisIdx].pack.setTimestamp(timestamp * SDP::getMultiplier(&M, thisIdx));
sdpState.tracks[thisIdx].rtcpSent = Util::bootSecs();
sdpState.tracks[thisIdx].pack.sendRTCP_SR(&sdpState.tracks[thisIdx].rtcp, sendUDP);
}
}else{
socket = &myConn;
callBack = sendTCP;
if (Util::bootSecs() != sdpState.tracks[thisIdx].rtcpSent){
sdpState.tracks[thisIdx].pack.setTimestamp(timestamp * SDP::getMultiplier(&M, thisIdx));
sdpState.tracks[thisIdx].rtcpSent = Util::bootSecs();
sdpState.tracks[thisIdx].pack.sendRTCP_SR(socket, sendTCP);
}
}
uint64_t offset = thisPacket.getInt("offset");
@ -168,6 +158,16 @@ namespace Mist{
sdpState.tracks[thisIdx].pack.sendData(socket, callBack, dataPointer, dataLen,
sdpState.tracks[thisIdx].channel, meta.getCodec(thisIdx));
if (Util::bootSecs() != sdpState.tracks[thisIdx].rtcpSent){
if (sdpState.tracks[thisIdx].channel == -1){// UDP connection
sdpState.tracks[thisIdx].pack.sendRTCP_SR(&sdpState.tracks[thisIdx].rtcp, 0, callBack);
}else{
sdpState.tracks[thisIdx].pack.sendRTCP_SR(socket, sdpState.tracks[thisIdx].channel, callBack);
}
sdpState.tracks[thisIdx].rtcpSent = Util::bootSecs();
}
static uint64_t lastAnnounce = Util::bootSecs();
if (reqUrl.size() && lastAnnounce + 5 < Util::bootSecs()){
INFO_MSG("Sending announce");

2
src/output/output_sdp.cpp
View file

@ -170,7 +170,7 @@ namespace Mist{
if (Util::bootSecs() != sdpState.tracks[thisIdx].rtcpSent){
sdpState.tracks[thisIdx].pack.setTimestamp(timestamp * SDP::getMultiplier(&M, thisIdx));
sdpState.tracks[thisIdx].rtcpSent = Util::bootSecs();
sdpState.tracks[thisIdx].pack.sendRTCP_SR(&sdpState.tracks[thisIdx].rtcp, sendUDP);
sdpState.tracks[thisIdx].pack.sendRTCP_SR(&sdpState.tracks[thisIdx].rtcp, sdpState.tracks[thisIdx].channel, sendUDP);
}
}else{
FAIL_MSG("RTP SDP output does not support TCP. No data will be sent to the target address");

10
src/output/output_webrtc.cpp
View file

@ -1712,8 +1712,8 @@ namespace Mist{
onIdle();
}
if (M.getLive() && stayLive && lastTimeSync + 666 < thisPacket.getTime()){
lastTimeSync = thisPacket.getTime();
if (M.getLive() && stayLive && lastTimeSync + 666 < thisTime){
lastTimeSync = thisTime;
if (liveSeek()){return;}
}
@ -1760,9 +1760,9 @@ namespace Mist{
// This checks if we have a whole integer multiplier, and if so,
// ensures only integer math is used to prevent rounding errors
if (mult == (uint64_t)mult){
rtcTrack.rtpPacketizer.setTimestamp(thisPacket.getTime() * (uint64_t)mult);
rtcTrack.rtpPacketizer.setTimestamp(thisTime * (uint64_t)mult);
}else{
rtcTrack.rtpPacketizer.setTimestamp(thisPacket.getTime() * mult);
rtcTrack.rtpPacketizer.setTimestamp(thisTime * mult);
}
bool isKeyFrame = thisPacket.getFlag("keyframe");
@ -1798,7 +1798,7 @@ namespace Mist{
//If this track hasn't sent yet, actually sent
if (mustSendSR.count(thisIdx)){
mustSendSR.erase(thisIdx);
rtcTrack.rtpPacketizer.sendRTCP_SR((void *)&udp, onRTPPacketizerHasRTCPDataCallback);
rtcTrack.rtpPacketizer.sendRTCP_SR((void *)&udp, 0, onRTPPacketizerHasRTCPDataCallback);
}
}