In dockerfile, we can set the default RTC candidate to env:
```
ENV SRS_RTC_SERVER_CANDIDATE=\$CANDIDATE
CMD ["./objs/srs", "-e"]
```
When starts a docker container, user can setup the candidate by env:
```
docker run --rm -it --env CANDIDATE=1.2.3.4 ossrs/srs:5
```
We should parse the content of SRS_RTC_SERVER_CANDIDATE as env variable name and parse it again.
---------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: pengfei.ma <pengfei.ma@ctechm.com>
For WebRTC, SRS expect the h.264 codec is:
```
a=rtpmap:106 H264/90000
a=fmtp:106 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f
```
But sometimes, the device does not support the profile, for example only bellow:
```
a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e033
a=fmtp:122 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=420033
a=fmtp:121 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640033
a=fmtp:120 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0033
```
So we should warning user about the profile missmatch, because it might not work.
----------
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
Co-authored-by: LiPeng <lipeng19811218@gmail.com>
For some use scenario, the publisher is invited when player want to view the stream:
1. Publisher connect to system, but does not publish any stream to SRS yet.
2. Player connect to system and start to request the stream.
3. System notifies publisher to publish stream to SRS.
4. Player play the stream from SRS.
Please notice that `system` means your business system, not SRS.
This is what we called `on-demand-live-streaming`, so when the last player stop to view the stream, what happends?
1. System needs to notify publisher to stop publish.
2. Or, SRS disconnect the publisher when idle(the last player stops playing).
This PR is for the solution 2, so that the cleanup is very simple, your system does not need to notify publisher to stop publish, because SRS has already disconnected the publihser.
---------
Co-authored-by: winlin <winlin@vip.126.com>
Co-authored-by: chundonglinlin <chundonglinlin@163.com>
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.
PICK 07a9a005d5
1. SRT: Fix srt to rtmp crash when sps or pps empty. v5.0.112 (#3323)
2. GB28181: Fix memory overlap for small packets. v5.0.111 (#3315)
3. FLV: Support set default has_av and disable guessing. v5.0.110 (#3311)
4. FLV: Drop packet if header flag is not matched. v5.0.109 (#3306)
5. FLV: Reset has_audio or has_video if only sequence header. (#3310)
* 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.
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.
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.
* 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>
* 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>
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.
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.
1. Update TS video codec to HEVC during streaming.
2. Return error when HEVC is disabled.
3. Parse HEVC NALU type by SrsHevcNaluTypeParse.
4. Show message when codec change for TS.
Co-authored-by: runner365 <shi.weibd@hotmail.com>
1. Support configure with --h265=on.
2. Parse HEVC(H.265) from FLV or RTMP packet.
3. Support HEVC over RTMP or HTTP-FLV.
Co-authored-by: runner365 <shi.weibd@hotmail.com>
* 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>
* 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>
* 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>
1. Support cygwin by '--cygwin64=on'
2. Detect cygwin automatically.
3. Disalbe sanitizer, srt and srtp with openssl.
4. Disable multiple threads, use single threads.
5. Support utest for cygwin64.
6. Query features for windows by API.
7. Disable stat APIs for cygwin.
8. Use ST select event driver.
Co-authored-by: wenjie.zhao <740936897@qq.com>
* MP4: Fix memory leak when error.
* Kernel: Support free global objects for utest.
* HTTP: Fix memory leak when error.
* MP4: Support more sample rate for audio.
* RTMP: Support free field for utest.
* UTest: Support address sanitizer.
* Exporter: metrics support cpu gauge.
* Exporter: metrics support memory and uname..
* Exporter: Ignore error when uname fail.
Co-authored-by: winlin <winlin@vip.126.com>
1. Support Java/log4j log level text.
2. Support configuring by `--log-new-level=on` which is enabled by default.
3. Support `--log-new-level=off` to use SRS 4.0 log level for compatibility.
01. Support GB config as StreamCaster.
02. Support disable GB by --gb28181=off.
03. Add utests for SIP examples.
04. Wireshark plugin to decode TCP/9000 as rtp.rfc4571
05. Support MPEGPS program stream codec.
06. Add utest for PS stream codec.
07. Decode MPEGPS packet stream.
08. Carry RTP and PS packet as helper in PS message.
09. Support recover from error mode.
10. Support process by a pack of PS/TS messages.
11. Add statistic for recovered and msgs dropped.
12. Recover from err position fastly.
13. Define state machine for GB session.
14. Bind context to GB session.
15. Re-invite when media disconnected.
16. Update GitHub actions with GB28181.
17. Support parse CANDIDATE by env or pip.
18. Support mux GB28181 to RTMP.
19. Support regression test by srs-bench.
