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Fix #3181: SRT & WebRTC: Use SrsRawH264Stream to mux SPS/PPS.

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hondaxiao 2022-09-20 21:10:44 +08:00 • committed by winlin
parent f974c7c8b0
commit 4acb246c57
7 changed files with 50 additions and 51 deletions
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44
trunk/src/app/srs_app_srt_source.cpp
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@ -434,30 +434,28 @@ srs_error_t SrsRtmpFromSrtBridge::check_sps_pps_change(SrsTsMessage* msg)
// ts tbn to flv tbn.
uint32_t dts = (uint32_t)(msg->dts / 90);
//type_codec1 + avc_type + composition time + fix header + count of sps + len of sps + sps + count of pps + len of pps + pps
int nb_payload = 1 + 1 + 3 + 5 + 1 + 2 + sps_.size() + 1 + 2 + pps_.size();
std::string sh;
SrsRawH264Stream* avc = new SrsRawH264Stream();
SrsAutoFree(SrsRawH264Stream, avc);
if ((err = avc->mux_sequence_header(sps_, pps_, sh)) != srs_success) {
return srs_error_wrap(err, "mux sequence header");
}
// h264 packet to flv packet.
char* flv = NULL;
int nb_flv = 0;
if ((err = avc->mux_avc2flv(sh, SrsVideoAvcFrameTypeKeyFrame, SrsVideoAvcFrameTraitSequenceHeader, dts, dts, &flv, &nb_flv)) != srs_success) {
return srs_error_wrap(err, "avc to flv");
}
SrsMessageHeader header;
header.initialize_video(nb_flv, dts, 1);
SrsCommonMessage rtmp;
rtmp.header.initialize_video(nb_payload, dts, 1);
rtmp.create_payload(nb_payload);
rtmp.size = nb_payload;
SrsBuffer payload(rtmp.payload, rtmp.size);
//TODO: call api
payload.write_1bytes(0x17);// type(4 bits): key frame; code(4bits): avc
payload.write_1bytes(0x0); // avc_type: sequence header
payload.write_1bytes(0x0); // composition time
payload.write_1bytes(0x0);
payload.write_1bytes(0x0);
payload.write_1bytes(0x01); // version
payload.write_1bytes(sps_[1]);
payload.write_1bytes(sps_[2]);
payload.write_1bytes(sps_[3]);
payload.write_1bytes(0xff);
payload.write_1bytes(0xe1);
payload.write_2bytes(sps_.size());
payload.write_bytes((char*)sps_.data(), sps_.size());
payload.write_1bytes(0x01);
payload.write_2bytes(pps_.size());
payload.write_bytes((char*)pps_.data(), pps_.size());
if ((err = rtmp.create(&header, flv, nb_flv)) != srs_success) {
return srs_error_wrap(err, "create rtmp");
}
if ((err = live_source_->on_video(&rtmp)) != srs_success) {
return srs_error_wrap(err, "srt to rtmp sps/pps");
}