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206 commits

Author SHA1 Message Date
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
Haibo Chen
06f10b1894
fix coredump bug: caused by publish stream that codec is h.263 (#2505) 2021-08-04 17:06:55 +08:00
Haibo Chen
86c67f7d95 RTC: Support statistic for HTTP-API, HTTP-Callback and Security (#2483) v4.0.144 
* commit message for your changes. Lines starting

* Update srs_app_rtc_api.cpp

* add SrsRtcConnPlay and SrsRtcConnPublish, in enum SrsRtmpConnType

* Update srs_rtmp_stack.cpp

* Update srs_app_rtc_conn.cpp

* Update srs_app_rtc_api.cpp

* update utest

* Update srs_utest_app.cpp
 2021-07-24 08:08:35 +08:00
Haibo Chen
0efd7b1bbc
RTC: Support statistic for HTTP-API, HTTP-Callback and Security (#2483) 
* commit message for your changes. Lines starting

* Update srs_app_rtc_api.cpp

* add SrsRtcConnPlay and SrsRtcConnPublish, in enum SrsRtmpConnType

* Update srs_rtmp_stack.cpp

* Update srs_app_rtc_conn.cpp

* Update srs_app_rtc_api.cpp

* update utest

* Update srs_utest_app.cpp
 2021-07-24 08:05:10 +08:00
Winlin
c8a1e0f3da
Refine AUTHORS.txt to AUTHORS.md, etc. 5.0.8 (#2464) 
* Refine AUTHORS.txt to AUTHORS.md, etc. 5.0.8

* Update README.md

* Update README.md

* Refine format for AUTHORS.md
 2021-07-08 14:30:47 +08:00
Haibo Chen
90b7933dbb For #2403, fix padding packets for RTMP2RTC. 4.0.140. 
Calculate the correct payload_size which pure padding data, in the process of rtc2rtmp, make Chrome happy (#2461)

* Calculate the correct payload_size which pure padding data, in the process of rtc2rtmp, make Chrome happy

* make clear for magic number

make clear for magic number

* Update srs_app_rtc_source.cpp
 2021-07-08 14:27:51 +08:00
Haibo Chen
7eee9aa598
Calculate the correct payload_size which pure padding data, in the process of rtc2rtmp, make Chrome happy (#2461) 
* Calculate the correct payload_size which pure padding data, in the process of rtc2rtmp, make Chrome happy

* make clear for magic number

make clear for magic number

* Update srs_app_rtc_source.cpp
 2021-07-08 14:23:53 +08:00
winlin
15901cacee SquashSRS4: Use SPDX-License-Identifier: MIT. 5.0.3 2021-05-31 13:42:20 +08:00
winlin
3cd22b6e6e Use SPDX-License-Identifier: MIT. 4.0.124 2021-05-31 12:59:21 +08:00
winlin
f043a7eb48 SquashSRS4: Allow RTC play before publish. 2021-05-19 21:06:17 +08:00
root
d55af6be44 Fix #2362: Allow WebRTC to play before publishing, for GB28181 as such. 4.0.117 2021-05-19 20:54:22 +08:00
winlin
e3bca883e1 SuqashSRS4: Build SRT native 2021-05-16 16:14:00 +08:00
winlin
dae6dc5395 Rename SrsRtcStream* to SrsRtcSource*. 4.0.113 2021-05-15 12:33:02 +08:00