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Commit Graph

6547 Commits (4b09a7d686432cf18ad20028b8a232df3f068ad0)

Author SHA1 Message Date
winlin 4b09a7d686 Configure: Reorder the functions, nothing changed. 2 years ago
winlin 5559ac25fe Refine configure to guess OS automatically. v5.0.121 
1. Guess for macOS and cygwin64.
2. Refine options for configure.
 2 years ago
winlin 6299dee1b6 Update new authors. 2 years ago
winlin 07a9a005d5 Refine default config file for SRS. v5.0.120 
1. Docker use srs.conf and env variables.
2. Show help if run SRS without any options.
3. Do not guess config file, use whatever from user.
 2 years ago
winlin ae3b367487 Asan: Only link by statically for asan. 2 years ago
winlin 87a2ef100a Script: Discover version from code. 2 years ago
winlin 8a0ac8e3a1 FLV: Fix bug for header flag gussing. v5.0.119 (#939) 2 years ago
winlin 386bb41f63 Script: Fix configure help bug. 2 years ago
winlin 37867533cd MP3: Convert RTMP(MP3) to WebRTC(OPUS). v5.0.118 (#296) (#3340) 2 years ago
winlin 1c5788c638 MP3: Support decode mp3 by FFmpeg natively. (#296) (#3340) 2 years ago
winlin 95defe6dad MP3: Support dump stream information. v5.0.117 (#296) (#3339) 2 years ago
winlin 23b7939574 Actions: Fix GitHub actions warnings. 2 years ago
winlin f6e0b1c894 MP3: Support mp3 for RTMP/HLS/HTTP-FLV/HTTP-TS/HLS etc. v5.0.116 2 years ago
winlin 0a49638f54 MP3: Add config examples for MP3. #296 2 years ago
winlin 05d7400cd5 Merge branch v4.0.269 into 5.0release 
1. MP3: Fix bug for TS or HLS with mp3 codec. v4.0.269 (#296) (#3333)
 2 years ago
Winlin 577cd299e1
MP3: Fix bug for TS or HLS with mp3 codec. v4.0.269 (#296) (#3333) 
* MP3: Fix bug for TS or HLS with mp3 codec. v4.0.269 (#296)

1. Refresh HLS audio codec if changed in stream.
2. Refresh TS audio codec if changed in stream.
3. Fix mp3 demux bug in SrsFormat::audio_mp3_demux.
4. Use 3(MPEG1) not 4(MPEG2) as PMT stream type, follow FFmpeg.
5. MP3: Update utest for mp3 sample parsing.
6. MP3: Ignore empty frame sample.
7. UTest: Fix utest failed, do not copy files.
 2 years ago
winlin 518c25aec3 Print version and signature to stdout. 2 years ago
winlin 5dcd6637e3 Fix #3328: Docker: Avoiding duplicated copy files. v5.0.115 2 years ago
Winlin 6f3d6b9b65
GB: Refine lazy object GC. v5.0.114 (#3321) 
* GB: Refine lazy object GC.

1. Remove gc_set_creator_wrapper, pass by resource constructor.
2. Remove SRS_LAZY_WRAPPER_GENERATOR macro, use template directly.
3. Remove interfaces ISrsGbSipConn and ISrsGbSipConnWrapper.
4. Remove ISrsGbMediaConn and ISrsGbMediaConnWrapper.

* GC: Refine wrapper constructor.

* GB: Refine lazy object GC. v5.0.114
 2 years ago
ChenGH 7eaee46f1f
Asan: Support parse asan symbol backtrace log. v5.0.113 (#3324) 
* asan: support parse asan symbol log

* asan: refine srs_parse_asan_backtrace_symbols error code

* asan: Refine code, extract asan log to error file.

Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
john 09a96175e8
SRT: Fix srt to rtmp crash when sps or pps empty. v5.0.112 (#3323) 
* SRT: fix crash when sps/pps empty. v5.0.112

Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
Winlin 56040cab42
GB28181: Fix memory overlap for small packets. v5.0.111 (#3315) 2 years ago
Winlin a36cb57949
FLV: Support set default has_av and disable guessing. v5.0.110 (#3311) 
* FLV: Support set default has_av and disable guessing. v5.0.110

1. Support config default has_audio and has_video.
2. Support disable guessing has_audio or has_video.

* FLV: Reset to false if start to guess has_av.

* FLV: Add regression test for FLV header av metadata.
 2 years ago
Winlin 4551200e95
FLV: Drop packet if header flag is not matched. v5.0.109 (#3306) 
1. Ignore audo or video packets if FLV header disable it.
2. Run: Add regression test config and run for IDEA.
3. Test: Refine regression test to allow no audio/video for FLV
4. Config: Whether drop packet if not match header.
 2 years ago
Winlin 35185cf844
FLV: Reset has_audio or has_video if only sequence header. (#3310) 
1. Reset has_audio if got some video frames but no audio frames.
2. Reset has_video if got some audio frames but no video frames.
3. Note that audio/video frames are not sequence header.
 2 years ago
john d1bc155c8b
DASH: Fix dash crash bug when writing file. v5.0.108 (#3301) 
Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
john bbe333d3ca
SRT: Support SRT to RTMP to WebRTC. v5.0.107 (#3296) 
* SRT: Support SRT to RTMP to WebRTC. v5.0.107

Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
Haibo Chen c5a0c5947f
API: Parse fragment of URI. v5.0.106 (#3295) 
* parse fragment of uri
* adapt FMLE URL: 'rtmp://ip/app/app2#k=v/stream', then add more test case

Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
winlin 0e550d496b Cygwin: Enable gb28181 for Windows. v5.0.105 2 years ago
chengh 8be4c8e334 Asan: Set asan loging callback. v5.0.104 2 years ago
winlin 41769308d2 GB28181: Enable GB for CentOS 7 package. v5.0.103 2 years ago
winlin 4b5ae7b3d2 Package script support extra options. v5.0.102 2 years ago
winlin e86e0c8999 Disable CLS and APM by default. v5.0.101 2 years ago
mapengfei53 c7b7921712
Config: Add utest for configuring with ENV variables. v5.0.100 (#3284) 
* Config: Add utest for configuring with ENV variables.

Co-authored-by: pengfei.ma <pengfei.ma@ctechm.com>
Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
stone a4d9e45545
Live: Fix bug for gop cache limits. v5.0.99 (#3289) 
* bugfix: setting srt bridge to rtmp gop cache limit while SrsMpegtsSrtConn::acquire_publish 

* setting http_stream gop cache limit while SrsHttpStreamServer::hijack

* if gop_cache_max_frames_ == 0, don't enable the got cache max frames limit

Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
Winlin e83fc2388b
Docker: Remove CentOS 6 support. (#3287) 
1. Remove CentOS 6 for test and utest.
2. Statically build FFmpeg, no so depends.
 2 years ago
Winlin 5cadfff2e5
SRT: Support transform tlpkdrop to tlpktdrop. 5.0.98 (#3279) 2 years ago
Winlin fdbfe59784
Config: Add ENV tips for config. 5.0.97 (#3278) 2 years ago
john 271afbbf82
DASH: Fix number mode bug to make it run. v5.0.96 (#3240) 
* Add utc time utility
* Fix calculate duration in fmp4
* Refine dash code, use segment template timeline
* Shrink m4s file and cleanup
* Support play by dash.js
* Use SegmentTemplate timeline mode with $Number$

Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
Winlin af192d6184
For #3176: GB28181: Error and logging for HEVC. v5.0.95 (#3276) 
1. Parse video codec from PSM packet.
2. Return error and logging if HEVC packet.
3. Ignore invalid AVC NALUs, drop AVC AUD and SEI.
4. Disconnect TCP connection if HEVC.
 2 years ago
winlin f10412d289 Asan: Fix utest bug. 2 years ago
Winlin 13918ed81f
For #3236: Live: Change gop cache limits to 2500. v5.0.94 (#3273) 2 years ago
stone ec76512e42
Live: Limit cached max frames by gop_cache_max_frames (#3236) 
* add gop_cache_max_frames

* Live: Limit cached max frames by gop_cache_max_frames. v5.0.93

Co-authored-by: wanglei <wanglei@unicloud.com>
Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
winlin 4ada0bc629 Asan: Cleanup for testing for asan. 2 years ago
winlin cdbebb3729 Merge branch '4.0release' into develop 2 years ago
johzzy e529536563 WebRTC: Fix no audio and video issue for Firefox. (#3079) v4.0.268 
* 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>
 2 years ago
Winlin b72ad85502
Asan: Check libasan and show tips. v5.0.92 (#3266) 2 years ago
ChenGH 6b130d4205
Asan: Try to fix st_memory_leak for asan check (#3264) 
* asan: try to fix st_memory_leak for asan check

* asan: srs_st_unit should be call in hybrid server stop

* Rename st_uninit to st_destroy. v5.0.91

Co-authored-by: winlin <winlin@vip.126.com>
 2 years ago
chengh 6fa17aa3f8 ST: Support st_destroy to free resources for asan. 2 years ago
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>
 2 years ago