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Fixes to AC3, MP3, 4+GiB MP4 and MP4 B-frames by Oswald de Bruin

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This commit is contained in:
Thulinma 2015-04-02 11:14:58 +02:00
parent f48cdb14a1
commit 514f0a9b2b
6 changed files with 83 additions and 45 deletions
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41
lib/rtp.cpp
View file

@ -107,24 +107,27 @@ namespace RTP {
}
}
void Packet::sendAAC(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel) {
void Packet::sendData(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel, std::string codec) {
/// \todo This function probably belongs in DMS somewhere.
data[1] |= 0x80;//setting the RTP marker bit to 1
/// \todo This 0x100000 value - What is it? Why is it hardcoded?
/// \todo The least significant 3 bits are used to signal some stuff from RFC 3640. Why do we send them always as 000?
*((int *)(data + getHsize())) = htonl(((payloadlen << 3) & 0x0010fff8) | 0x00100000);
memcpy(data + getHsize() + 4, payload, payloadlen);
callBack(socket, data, getHsize() + 4 + payloadlen, channel);
sentPackets++;
sentBytes += payloadlen;
increaseSequence();
}
void Packet::sendRaw(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel) {
/// \todo This function probably belongs in DMS somewhere.
data[1] |= 0x80;//setting the RTP marker bit to 1
memcpy(data + getHsize(), payload, payloadlen);
callBack(socket, data, getHsize() + payloadlen, channel);
long offsetLen = 0;
if (codec == "AAC"){
INFO_MSG("send AAC codec");
*((long *)(data + getHsize())) = htonl(((payloadlen << 3) & 0x0010fff8) | 0x00100000);
offsetLen = 4;
}else if (codec == "MP3"){
INFO_MSG("send MP3 codec");
*((long *)(data + getHsize())) = 0;//this is MBZ and Frag_Offset, which is always 0
offsetLen = 4;
}else if (codec == "AC3"){
INFO_MSG("send AC3 codec");
*((short *)(data + getHsize())) = htons(0x0001) ;//this is 6 bits MBZ, 2 bits FT = 0 = full frames and 8 bits saying we send 1 frame
offsetLen = 2;
}else{
INFO_MSG("send Raw");
}
memcpy(data + getHsize() + offsetLen, payload, payloadlen);
callBack(socket, data, getHsize() + offsetLen + payloadlen, channel);
sentPackets++;
sentBytes += payloadlen;
increaseSequence();
@ -162,12 +165,12 @@ namespace RTP {
((int *)rtcpData)[2] = htonl(2208988800UL + Util::epoch()); //epoch is in seconds
((int *)rtcpData)[3] = htonl((Util::getMS() % 1000) * 4294967.295);
if (metadata.tracks[tid].codec == "H264") {
if (metadata.tracks[tid].codec == "H264" || metadata.tracks[tid].codec == "MP3") {
((int *)rtcpData)[4] = htonl((ntpTime - 0) * 90000); //rtpts
} else if (metadata.tracks[tid].codec == "AAC") {
} else if (metadata.tracks[tid].codec == "AAC" || metadata.tracks[tid].codec == "AC3") {
((int *)rtcpData)[4] = htonl((ntpTime - 0) * metadata.tracks[tid].rate); //rtpts
} else {
DEBUG_MSG(DLVL_FAIL, "Unsupported codec");
DEBUG_MSG(DLVL_FAIL, "Unsupported codec: %s", metadata.tracks[tid].codec.c_str());
return;
}
//it should be the time packet was sent maybe, after all?

3
lib/rtp.h
View file

@ -45,8 +45,7 @@ namespace RTP {
void setTimestamp(unsigned int t);
void increaseSequence();
void sendH264(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel);
void sendAAC(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel);
void sendRaw(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel);
void sendData(void * socket, void callBack(void *, char *, unsigned int, unsigned int), const char * payload, unsigned int payloadlen, unsigned int channel, std::string codec);
void sendRTCP(long long & connectedAt, void * socket, unsigned int tid, DTSC::Meta & metadata, void callBack(void *, char *, unsigned int, unsigned int));

34
src/input/input_mp4.cpp
View file

@ -71,6 +71,7 @@ namespace Mist {
sttsBox.clear();
sttsBox.read(tmp);
}else if (stblBoxType == "ctts"){
///\todo this box should not have to be read, since its information is taken from the DTSH
tmp = std::string(stblLoopPeek.asBox() ,stblLoopPeek.boxedSize());
cttsBox.clear();
cttsBox.read(tmp);
@ -341,6 +342,8 @@ namespace Mist {
MP4::STSZ stszBox;
MP4::STCO stcoBox;
MP4::STSC stscBox;
MP4::CTTS cttsBox;//optional ctts box
bool hasCTTS = false;
for (uint32_t m = 0; m < ((MP4::STBL&)minfLoopPeek).getContentCount(); m++){
tmp = std::string(((MP4::STBL&)minfLoopPeek).getContent(m).asBox(),((MP4::STBL&)minfLoopPeek).getContent(m).boxedSize());
std::string stboxRead = tmp;
@ -356,6 +359,9 @@ namespace Mist {
stcoBox.read(stboxRead);
}else if (stblBoxType == "stsc"){
stscBox.read(stboxRead);
}else if (stblBoxType == "ctts"){
cttsBox.read(stboxRead);
hasCTTS = true;
}else if (stblBoxType == "stsd"){
//check for codec in here
MP4::Box & tmpBox = ((MP4::STSD&)stblLoopPeek).getEntry(0);
@ -390,18 +396,15 @@ namespace Mist {
myMeta.tracks[trackNo].codec = "AC3";
}else{
MP4::Box esds = ((MP4::AudioSampleEntry&)tmpBox).getCodecBox();
if (((MP4::ESDS&)esds).isAAC()){
myMeta.tracks[trackNo].codec = "AAC";
myMeta.tracks[trackNo].init = ((MP4::ESDS&)esds).getInitData();
}else{
myMeta.tracks[trackNo].codec = "MP3";
}
myMeta.tracks[trackNo].codec = ((MP4::ESDS&)esds).getCodec();
myMeta.tracks[trackNo].init = ((MP4::ESDS&)esds).getInitData();
}
myMeta.tracks[trackNo].size = 16;///\todo this might be nice to calculate from mp4 file;
//get Visual sample entry -> esds -> startcodes
}else{
myMeta.tracks.erase(trackNo);
}
}
}//rof stbl
uint64_t totaldur = 0;///\todo note: set this to begin time
@ -415,6 +418,10 @@ namespace Mist {
//change to for over all samples
unsigned int stcoIndex = 0;
unsigned int stscIndex = 0;
unsigned int cttsIndex = 0;//current ctts Index we are reading
unsigned int cttsEntryRead = 0;//current part of ctts we are reading
MP4::CTTSEntry cttsEntry;
unsigned int fromSTCOinSTSC = 0;
long long unsigned int tempOffset;
bool stcoIs64 = (stcoBox.getType() == "co64");
@ -479,6 +486,19 @@ namespace Mist {
tempSTTS = sttsBox.getSTTSEntry(entryNo);
}
}
//set time offset
if (hasCTTS){
cttsEntry = cttsBox.getCTTSEntry(cttsIndex);
cttsEntryRead++;
if (cttsEntryRead >= cttsEntry.sampleCount){
cttsIndex++;
cttsEntryRead = 0;
}
BsetPart.timeOffset = (cttsEntry.sampleOffset * 1000)/timeScale;
}else{
BsetPart.timeOffset = 0;
}
//set size, that's easy
BsetPart.size = stszBox.getEntrySize(sampleIndex);
//trackid
@ -523,7 +543,7 @@ namespace Mist {
int tmp = fread(data, it->size, 1, inFile);
if (tmp == 1){
//add data
myMeta.update(it->time, 1, it->trackID, it->size, it->bpos, it->keyframe);
myMeta.update(it->time, it->timeOffset, it->trackID, it->size, it->bpos, it->keyframe);
}else{
INFO_MSG("fread did not return 1, bpos: %llu size: %llu keyframe: %d error: %s", it->bpos, it->size, it->keyframe, strerror(errno));
return false;

1
src/input/input_mp4.h
View file

@ -44,6 +44,7 @@ namespace Mist {
long long unsigned int bpos;
long long unsigned int size;
long long unsigned int stcoNr;
long unsigned int timeOffset;
bool keyframe;
};

21
src/output/output_dash_mp4.cpp
View file

@ -113,7 +113,7 @@ namespace Mist {
hev1Box.setCLAP(hvccBox);
stsdBox.setEntry(hev1Box, 0);
}
if (myMeta.tracks[tid].codec == "AAC"){
if (myMeta.tracks[tid].codec == "AAC" || myMeta.tracks[tid].codec == "MP3"){
MP4::AudioSampleEntry ase;
ase.setCodec("mp4a");
ase.setDataReferenceIndex(1);
@ -284,7 +284,7 @@ namespace Mist {
if (myMeta.tracks[tid].codec == "H264" || myMeta.tracks[tid].codec == "HEVC"){
tfhdBox.setTrackID(1);
}
if (myMeta.tracks[tid].codec == "AAC"){
if (myMeta.tracks[tid].codec == "AAC" || myMeta.tracks[tid].codec == "AC3" || myMeta.tracks[tid].codec == "MP3"){
tfhdBox.setFlags(MP4::tfhdSampleFlag);
tfhdBox.setTrackID(1);
tfhdBox.setDefaultSampleFlags(MP4::isKeySample);
@ -347,7 +347,7 @@ namespace Mist {
i++;
}
}
if (myMeta.tracks[tid].codec == "AAC"){
if (myMeta.tracks[tid].codec == "AAC" || myMeta.tracks[tid].codec == "AC3" || myMeta.tracks[tid].codec == "MP3"){
trunBox.setFlags(MP4::trundataOffset | MP4::trunsampleSize | MP4::trunsampleDuration);
trunBox.setDataOffset(88 + (8 * myMeta.tracks[tid].keys[keyNum].getParts()) + 8);
for (int j = 0; j < myMeta.tracks[tid].keys[keyNum].getParts(); j++){
@ -434,6 +434,7 @@ namespace Mist {
int lastAudTime = 0;
int audKeys = 0;
int audInitTrack = 0;
///\todo Dash automatically selects the last audio and video track for manifest, maybe make this expandable/selectable?
for (std::map<unsigned int, DTSC::Track>::iterator it = myMeta.tracks.begin(); it != myMeta.tracks.end(); it ++){
if (it->second.lastms > lastTime){
lastTime = it->second.lastms;
@ -448,7 +449,7 @@ namespace Mist {
vidKeys = it->second.keys.size();
vidInitTrack = it->first;
}
if (it->second.codec == "AAC" && it->second.lastms > lastAudTime){
if ((it->second.codec == "AAC" || it->second.codec == "MP3" || it->second.codec == "AC3")&& it->second.lastms > lastAudTime){
lastAudTime = it->second.lastms;
audKeys = it->second.keys.size();
audInitTrack = it->first;
@ -518,10 +519,17 @@ namespace Mist {
r << " </SegmentTemplate>" << std::endl;
for (std::map<unsigned int, DTSC::Track>::iterator it = myMeta.tracks.begin(); it != myMeta.tracks.end(); it++){
if (it->second.codec == "AAC"){
if (it->second.codec == "AAC" || it->second.codec == "MP3" || it->second.codec == "AC3"){
r << " <Representation ";
r << "id=\"" << it->first << "\" ";
r << "codecs=\"mp4a.40.2\" ";
// (see RFC6381): sample description entry , ObjectTypeIndication [MP4RA, RFC], ObjectTypeIndication [MP4A ISO/IEC 14496-3:2009]
if (it->second.codec == "AAC" ){
r << "codecs=\"mp4a.40.2\" ";
}else if (it->second.codec == "MP3" ){
r << "codecs=\"mp4a.40.34\" ";
}else if (it->second.codec == "AC3" ){
r << "codecs=\"ec-3\" ";
}
r << "audioSamplingRate=\"" << it->second.rate << "\" ";
r << "bandwidth=\"" << it->second.bps << "\">" << std::endl;
r << " <AudioChannelConfiguration schemeIdUri=\"urn:mpeg:dash:23003:3:audio_channel_configuration:2011\" value=\"" << it->second.channels << "\" />" << std::endl;
@ -547,6 +555,7 @@ namespace Mist {
capa["codecs"][0u][0u].append("HEVC");
capa["codecs"][0u][1u].append("AAC");
capa["codecs"][0u][1u].append("AC3");
capa["codecs"][0u][1u].append("MP3");
capa["methods"][0u]["handler"] = "http";
capa["methods"][0u]["type"] = "dash/video/mp4";
capa["methods"][0u]["priority"] = 8ll;

28
src/output/output_rtsp.cpp
View file

@ -87,15 +87,15 @@ namespace Mist {
callBack = sendTCP;
}
if(myMeta.tracks[tid].codec == "AAC"){
tracks[tid].rtpPacket.setTimestamp(timestamp * ((double) myMeta.tracks[tid].rate / 1000.0));
tracks[tid].rtpPacket.sendAAC(socket, callBack, dataPointer, dataLen, tracks[tid].channel);
if(myMeta.tracks[tid].codec == "MP3"){
tracks[tid].rtpPacket.setTimestamp(timestamp * 90);
tracks[tid].rtpPacket.sendData(socket, callBack, dataPointer, dataLen, tracks[tid].channel, "MP3");
return;
}
if(myMeta.tracks[tid].codec == "MP3" || myMeta.tracks[tid].codec == "AC3"){
if( myMeta.tracks[tid].codec == "AC3" || myMeta.tracks[tid].codec == "AAC"){
tracks[tid].rtpPacket.setTimestamp(timestamp * ((double) myMeta.tracks[tid].rate / 1000.0));
tracks[tid].rtpPacket.sendRaw(socket, callBack, dataPointer, dataLen, tracks[tid].channel);
tracks[tid].rtpPacket.sendData(socket, callBack, dataPointer, dataLen, tracks[tid].channel,myMeta.tracks[tid].codec);
return;
}
@ -220,6 +220,7 @@ namespace Mist {
//loop over all tracks, add them to the SDP.
/// \todo Make sure this works correctly for multibitrate streams.
for (std::map<unsigned int, DTSC::Track>::iterator objIt = myMeta.tracks.begin(); objIt != myMeta.tracks.end(); objIt ++) {
INFO_MSG("Codec: %s", objIt->second.codec.c_str());
if (objIt->second.codec == "H264") {
MP4::AVCC avccbox;
avccbox.setPayload(objIt->second.init);
@ -248,11 +249,12 @@ namespace Mist {
transportString << "; mode=AAC-hbr; SizeLength=13; IndexLength=3; IndexDeltaLength=3;\r\n"
"a=control:track" << objIt->second.trackID << "\r\n";
}else if (objIt->second.codec == "MP3") {
transportString << "m=" << objIt->second.type << " 0 RTP/AVP 96" << "\r\n"
"a=rtpmap:14 MPA/" << objIt->second.rate << "/" << objIt->second.channels << "\r\n"
//"a=fmtp:96 streamtype=5; profile-level-id=15;";
//these values are described in RFC 3640
//transportString << " mode=AAC-hbr; SizeLength=13; IndexLength=3; IndexDeltaLength=3;\r\n"
transportString << "m=" << objIt->second.type << " 0 RTP/AVP 14" << "\r\n"
"a=rtpmap:14 MPA/90000/" << objIt->second.channels << "\r\n"
"a=control:track" << objIt->second.trackID << "\r\n";
}else if ( objIt->second.codec == "AC3") {
transportString << "m=" << objIt->second.type << " 0 RTP/AVP 100" << "\r\n"
"a=rtpmap:100 AC3/" << objIt->second.rate << "/" << objIt->second.channels << "\r\n"
"a=control:track" << objIt->second.trackID << "\r\n";
}
}//for tracks iterator
@ -268,8 +270,12 @@ namespace Mist {
unsigned int SSrc = rand();
if (myMeta.tracks[trId].codec == "H264") {
tracks[trId].rtpPacket = RTP::Packet(97, 1, 0, SSrc);
}else if(myMeta.tracks[trId].codec == "AAC" || myMeta.tracks[trId].codec == "MP3"){
}else if(myMeta.tracks[trId].codec == "AAC"){
tracks[trId].rtpPacket = RTP::Packet(96, 1, 0, SSrc);
}else if(myMeta.tracks[trId].codec == "AC3"){
tracks[trId].rtpPacket = RTP::Packet(100, 1, 0, SSrc);
}else if(myMeta.tracks[trId].codec == "MP3"){
tracks[trId].rtpPacket = RTP::Packet(14, 1, 0, SSrc);
}else{
DEBUG_MSG(DLVL_FAIL,"Unsupported codec for RTSP: %s",myMeta.tracks[trId].codec.c_str());
}