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WebRTC: Fix no audio and video issue for Firefox. (#3079) v4.0.268

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* Remove extern SrsPps* duplicate declarations

* fix(rtmp2rtc): fix video payload type for rtmp to rtc bridge (#3041)

* Revert changes not belongs to this PR.

* Fix naming issue, follow SRS style.

* Use srs_assert instead of assert.

* Fix firefox no audio issue.

Co-authored-by: winlin <winlin@vip.126.com>
This commit is contained in:
johzzy 2022-11-21 22:01:01 +08:00 • committed by winlin
parent 7d782ee8c9
commit e529536563
4 changed files with 14 additions and 7 deletions
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1
trunk/doc/CHANGELOG.md
View file

@ -8,6 +8,7 @@ The changelog for SRS.
## SRS 4.0 Changelog
* v4.0, 2022-11-22, Pick [#3079](https://github.com/ossrs/srs/issues/3079): WebRTC: Fix no audio and video issue for Firefox. v4.0.268
* v4.0, 2022-10-10, For [#2901](https://github.com/ossrs/srs/issues/2901): Edge: Fast disconnect and reconnect. v4.0.267
* v4.0, 2022-09-27, For [#3167](https://github.com/ossrs/srs/issues/3167): WebRTC: Refine sequence jitter algorithm. v4.0.266
* v4.0, 2022-09-16, For [#3179](https://github.com/ossrs/srs/issues/3179): WebRTC: Make sure the same m-lines order for offer and answer. v4.0.265

16
trunk/src/app/srs_app_rtc_source.cpp
View file

@ -722,6 +722,8 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
if (!descs.empty()) {
audio_ssrc = descs.at(0)->ssrc_;
}
// Note we must use the PT of source, see https://github.com/ossrs/srs/pull/3079
audio_payload_type_ = descs.empty() ? kAudioPayloadType : descs.front()->media_->pt_;
}
// video track ssrc
@ -730,6 +732,8 @@ SrsRtcFromRtmpBridger::SrsRtcFromRtmpBridger(SrsRtcSource* source)
if (!descs.empty()) {
video_ssrc = descs.at(0)->ssrc_;
}
// Note we must use the PT of source, see https://github.com/ossrs/srs/pull/3079
video_payload_type_ = descs.empty() ? kVideoPayloadType : descs.front()->media_->pt_;
}
}
@ -900,7 +904,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_opus(SrsAudioFrame* audio, SrsRtpPack
{
srs_error_t err = srs_success;
pkt->header.set_payload_type(kAudioPayloadType);
pkt->header.set_payload_type(audio_payload_type_);
pkt->header.set_ssrc(audio_ssrc);
pkt->frame_type = SrsFrameTypeAudio;
pkt->header.set_marker(true);
@ -1044,7 +1048,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_stap_a(SrsRtcSource* source, SrsShare
return srs_error_new(ERROR_RTC_RTP_MUXER, "sps/pps empty");
}
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_payload_type(video_payload_type_);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->nalu_type = (SrsAvcNaluType)kStapA;
@ -1126,7 +1130,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
SrsRtpPacket* pkt = new SrsRtpPacket();
pkts.push_back(pkt);
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_payload_type(video_payload_type_);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->nalu_type = (SrsAvcNaluType)first_nalu_type;
@ -1160,7 +1164,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_nalus(SrsSharedPtrMessage* msg, const
SrsRtpPacket* pkt = new SrsRtpPacket();
pkts.push_back(pkt);
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_payload_type(video_payload_type_);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->nalu_type = (SrsAvcNaluType)kFuA;
@ -1190,7 +1194,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_single_nalu(SrsSharedPtrMessage* msg,
SrsRtpPacket* pkt = new SrsRtpPacket();
pkts.push_back(pkt);
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_payload_type(video_payload_type_);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);
@ -1223,7 +1227,7 @@ srs_error_t SrsRtcFromRtmpBridger::package_fu_a(SrsSharedPtrMessage* msg, SrsSam
SrsRtpPacket* pkt = new SrsRtpPacket();
pkts.push_back(pkt);
pkt->header.set_payload_type(kVideoPayloadType);
pkt->header.set_payload_type(video_payload_type_);
pkt->header.set_ssrc(video_ssrc);
pkt->frame_type = SrsFrameTypeVideo;
pkt->header.set_sequence(video_sequence++);

2
trunk/src/app/srs_app_rtc_source.hpp
View file

@ -261,6 +261,8 @@ private:
uint16_t video_sequence;
uint32_t audio_ssrc;
uint32_t video_ssrc;
uint8_t audio_payload_type_;
uint8_t video_payload_type_;
public:
SrsRtcFromRtmpBridger(SrsRtcSource* source);
virtual ~SrsRtcFromRtmpBridger();

2
trunk/src/core/srs_core_version4.hpp
View file

@ -9,6 +9,6 @@
#define VERSION_MAJOR 4
#define VERSION_MINOR 0
#define VERSION_REVISION 267
#define VERSION_REVISION 268
#endif