In the scenario of converting WebRTC to RTMP, this conversion will not
proceed until an SenderReport is received; for reference, see:
https://github.com/ossrs/srs/pull/2470.
Thus, if HTTP-FLV streaming is attempted before the SR is received, the
FLV Header will contain only audio, devoid of video content.
This error can be resolved by disabling `guess_has_av` in the
configuration file, since we can guarantee that both audio and video are
present in the test cases.
However, in the original regression tests, the
`TestRtcPublish_HttpFlvPlay` test case contains a bug:
5a404c089b/trunk/3rdparty/srs-bench/srs/rtc_test.go (L2421-L2424)
The test would pass when `hasAudio` is true and `hasVideo` is false,
which is actually incorrect. Therefore, it has been revised so that the
test now only passes if both values are true.
---------
Co-authored-by: Copilot <175728472+Copilot@users.noreply.github.com>
Co-authored-by: winlin <winlinvip@gmail.com>
1. Add live benchmark support in srs-bench, which only connects and
disconnects without any media transport, to test source creation and
disposal and verify source memory leaks.
2. SmartPtr: Support cleanup of HTTP-FLV stream. Unregister the HTTP-FLV
handler for the pattern and clean up the objects and resources.
3. Support benchmarking RTMP/SRT with srs-bench by integrating the gosrt
and oryx RTMP libraries.
4. Refine SRT and RTC sources by using a timer to clean up the sources,
following the same strategy as the Live source.
---------
Co-authored-by: Haibo Chen <495810242@qq.com>
Co-authored-by: Jacob Su <suzp1984@gmail.com>
1. Enable blackbox test for each PR and push.
2. Refine Makefile and README for srs-bench.
3. Add blackbox using FFmpeg and ffprobe.
4. Add blackbox basic test for RTMP stream.
5. Add blackbox basic test for HTTP-FLV stream.
6. Fix utest rand seed issue.
PICK 2141d220b4
1. Build regression test tool for gb28181.
2. Run regression test for gb28181.
3. Format go code and eliminate logs.
4. Change base docker to ubuntu20.
PICK 7750bdae10
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.
