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218 commits

Author SHA1 Message Date
chundonglinlin
3fa4f66648
WebRTC: Support config the bitrate of transcoding AAC to Opus. v5.0.167, v6.0.60 (#3515) 
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Co-authored-by: john <hondaxiao@tencent.com>
 2023-07-18 11:09:50 +08:00
Winlin
7f997b39ae
WHIP: Add OBS support, ensuring compatibility with a unique SDP. v5.0.158, v6.0.51 (#3581) 
1. Ignore SDP GROUP LS.
2. Support ice in global session info.
3. Support audio codec "OPUS" or "opus".

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Co-authored-by: Johnny <hellojinqiang@gmail.com>
 2023-06-15 12:11:31 +08:00
Winlin
dcd02fe69c
Support composited bridges for 1:N protocols converting. v6.0.41 (#3392) 
Co-authored-by: john <hondaxiao@tencent.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
 2023-04-01 21:34:59 +08:00
Haibo Chen
4a5f479a0c
GB: Support H.265 for GB28181 (#3408) 
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: stone <bluestn@163.com>
Co-authored-by: Winlin <winlin@vip.126.com>
 2023-02-14 14:28:41 +08:00
john
7922057467
RTC: fix rtc publisher pli cid (#3318) 
* RTC: fix rtc publisher pli cid
* RTC: log bridge request keyframe
* Update release v6.0.19 v5.0.138

Co-authored-by: Winlin <winlin@vip.126.com>
Co-authored-by: ChenGH <chengh_math@126.com>
 2023-01-19 10:49:17 +08:00
winlin
c46ef81ff2 SRS5: Update license date to 2023. v5.0.123 
PICK 72f8ed4916
 2023-01-01 08:56:20 +08:00
winlin
6875876349 SRS5: MP3: Convert RTMP(MP3) to WebRTC(OPUS). v5.0.118 (#296) (#3340) 
PICK 37867533cd
 2022-12-26 18:06:38 +08:00
johzzy
6eb10afca2
WebRTC: Fix no audio and video issue for Firefox. (#3079) 
* 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>
 2022-11-21 22:01:01 +08:00
winlin
c12deded98 GB28181: Fix bug for parsing GB to RTC. 2022-10-07 19:47:34 +08:00
winlin
378bffa34f Micro changes and refines. 2022-09-30 17:57:48 +08:00
winlin
912cd6a59c Merge branch '4.0release' into develop 2022-09-28 17:47:51 +08:00
winlin
8bd8c1146d WebRTC: Eliminate unused debugging log. 2022-09-28 17:46:50 +08:00
winlin
0c6d30861b Merge branch '4.0release' into develop 2022-09-27 14:53:23 +08:00
winlin
386b92e9ab For #3167: WebRTC: Refine sequence jitter algorithm. v4.0.266 2022-09-27 14:53:05 +08:00
hondaxiao
4acb246c57 Fix #3181: SRT & WebRTC: Use SrsRawH264Stream to mux SPS/PPS. 2022-09-22 14:55:55 +08:00
winlin
79358673ef Merge branch '4.0release' into develop 2022-09-03 18:13:11 +08:00
winlin
34196ea7f7 Fix #3167: WebRTC: Play stucked when republish. v4.0.260 2022-09-03 17:14:32 +08:00
winlin
783aea7ac3 Fix #1405: Support guessing IBMF first. v5.0.58 2022-09-01 19:28:51 +08:00
winlin
d117145b95 Update date from 2021 to 2022. 2022-06-20 19:22:25 +08:00
winlin
e09daa2d4b SRT: Change bridges to bridge. 2022-06-14 20:05:09 +08:00
hondaxiao
e13d16439e SRT: support rtmp to srt 2022-06-14 20:02:22 +08:00
winlin
fa78cf3354 Prefix with srs_protocol in protocol directory. 2022-06-09 20:26:58 +08:00
winlin
f1840b87e5 Fix typo, change bridger to bridge. 2022-06-09 19:35:07 +08:00
winlin
d5c86dc5fa Switch LICENSE from MIT to **MIT or MulanPSL-2.0**. v5.0.21 2022-01-13 18:40:22 +08:00
winlin
93aa0eb5ba Squash: Fix bugs 2022-01-13 18:26:28 +08:00
chundonglinlin
7580341a1e
LiveSource/RtcSource: Refine fetch for external exposed interface. (#2873) 2022-01-13 11:43:32 +08:00
winlin
716e578a19 Squash: Fix bugs 2021-12-26 17:30:51 +08:00
chundonglinlin
3188c772b1
RTC: Eliminate duplicate assignment for video packet frame type (#2803) 
Co-authored-by: zhangjunqin1 <zhangjunqin@jd.com>
 2021-12-21 08:32:17 +08:00
winlin
f05e67e1a6 Squash: Fix bugs 2021-12-13 09:24:16 +08:00
john
7c353b5986 RTC: Fix memory leak when replace rtp packet in cache. (#2771). v4.0.205 
* fix memory leak when replace rtp packet in cache.
 2021-12-07 09:11:01 +08:00
winlin
8576fa7052 Squash: Merge v4.0.203 2021-12-04 11:21:35 +08:00
john
f3c4023c25
Fix bugs for RTC2RTMP. (#2768) 
1. Cache IDR frame's rtp timestamp instead of avsync timestamp.
2. Cache clock rate calculate by sender report.
3. Using srs_rtp_seq_distance instead of direct minus.
4. Add utest of av timestamp sync when duplicated sender report.
 2021-12-04 11:15:02 +08:00
johzzy
ff8657e1c5 RTC: Fix crash when pkt->payload() if pkt is nullptr (#2751). v4.0.199 2021-11-25 07:36:12 +08:00
johzzy
a862573220
RTC: Fix crash when pkt->payload() if pkt is nullptr (#2751) 2021-11-25 07:33:41 +08:00
winlin
5f85d405e7 Squash: Merge #2721, #2729 2021-11-13 19:36:43 +08:00
john
469bd8cfe2
RTC: check audio track exist when negotiate (#2729) 2021-11-13 19:09:45 +08:00
john
878833bb95
Rtc2Rtmp: Using rtp timestamp to distinguish different video frame (#2721) 2021-11-09 07:35:00 +08:00
winlin
b874d9c9ba Squash: Merge SRS 4.0, regression test for RTMP. 2021-10-12 08:36:24 +08:00
winlin
71ed6e5dc5 RTC: Refine config, aac to rtmp_to_rtc, bframe to keep_bframe. v4.0.174 2021-10-11 22:14:45 +08:00
winlin
5042117034 Squash: Merge SRS 4.0 2021-10-07 21:10:44 +08:00
ChenGH
7a4de9ffe7
Fix #2415, refine dtls fragment and rtp payload size (#2652) 2021-10-07 21:05:30 +08:00
winlin
7d3ec991e1 Squash: Merge SRS 4.0 2021-09-26 17:12:55 +08:00
winlin
ad4b648ed2 For #2545, Refine code with space lines. 2021-09-26 17:07:59 +08:00
johzzy
ee23e3abed fix some crash in rtc. (#2545) 2021-09-26 17:04:00 +08:00
johzzy
dc778020fc
fix some crash in rtc. (#2545) 2021-09-26 17:01:53 +08:00
winlin
f01c9638f1 Support http callback on_play/stop. 5.0.12 2021-09-23 13:38:04 +08:00
zozobreakzou
46adcfb6c9
[rtc] *Fix Fua package bug(payload size minus one). (#2618) 
* This can cause webrtc video PacketBuffer assemble corrupt when (nal size - 1) % 1300 == 0
* issues about webrtc all caused by this bug
 2021-09-23 11:10:16 +08:00
winlin
85620a34f5 Squash: Fix rtc to rtmp sync timestamp using sender report. #2470 2021-08-17 07:25:03 +08:00
john
ea8cff6163
RTC: Fix rtc to rtmp sync timestamp using sender report. (#2470) 
* fix annotation spell failed

* RTC to RTMP using SenderReport to sync av timestamp

* update pion/webrtc versio from v3.0.4 -> v3.0.13, auto config sender/receiver report

* Add rtc push flv play regression test

* Add unit test of ntp and av sync time

* Take flag CXX to makefile of utest

* Add annotation about rtc unit test

* Fix compiler error in C++98

* Add FFmpeg log callback funciton.
 2021-08-17 06:32:35 +08:00
Haibo Chen
529b89a29e Fix #2504 coredump bug: caused by publish stream that codec is h.263 (#2505) 2021-08-04 17:14:41 +08:00