a6d14eb09a
try to fix #4428. ## Cause rtmp do not support hevc, rtmp enhanced do. ## How to reproduce 1. start srs. `./objs/srs -c conf/srt.conf` 2. publish hevc (h.265) stream to srs by srt. `ffmpeg -re -i ./doc/source.flv -c:v libx265 -crf 28 -preset medium -c:a copy -pes_payload_size 0 -f mpegts 'srt://127.0.0.1:10080?streamid=#!::r=live/livestream,m=publish'` 3. probe the rtmp stream `ffprobe rtmp://localhost/live/livestream` ## About the Failed BlackBox test The failed blackbox test: `TestSlow_SrtPublish_RtmpPlay_HEVC_Basic` `TestSlow_SrtPublish_HttpFlvPlay_HEVC_Basic` ### Cause: The ffmpeg 5 is used to record a piece of video (DRV), the ffmpeg will transcode the enhanced flv format to TS format, but ffmpeg 5 don't support enhanced rtmp (or flv) in this case. The solution is to replace the ffmpeg to version 7 in those 2 test cases. ### why not upgrade ffmpeg to version 7? The black tests dependency on ffmpeg 5 will fail, and there are a few of them are not easy to resolve in ffmpeg 7. --------- Co-authored-by: winlin <winlinvip@gmail.com> |
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|---|---|---|
| .. | ||
| ffmpeg-4-fit | ||
| gperftools-2-fit | ||
| gprof | ||
| gtest-fit | ||
| httpx-static | ||
| libsrtp-2-fit | ||
| openssl-1.1-fit | ||
| patches | ||
| signaling | ||
| srs-bench | ||
| srs-docs | ||
| srt-1-fit | ||
| st-srs | ||
| openssl-OpenSSL_1_0_2u.tar.gz | ||
| opus-1.3.1.tar.gz | ||
| README.md | ||
http-parser-2.1.zip
- for srs to support http callback.
- https://github.com/nodejs/http-parser
- https://github.com/ossrs/http-parser
- https://ossrs.net/lts/zh-cn/license#http-parser
nginx-1.5.7.zip
- http://nginx.org/
- for srs to support hls streaming.
srt-1-fit srt-1.5.3.tar.gz
openssl-1.1-fit openssl-1.1.1l.tar.gz
openssl-1.1.0e.zip openssl-OpenSSL_1_0_2u.tar.gz
- http://www.openssl.org/source/openssl-1.1.0e.tar.gz
- openssl for SRS(with-ssl) RTMP complex handshake to delivery h264+aac stream.
- SRTP depends on openssl 1.0.*, so we use both ssl versions.
- https://ossrs.net/lts/zh-cn/license#openssl
libsrtp-2.3.0.tar.gz
- For WebRTC, SRTP to encrypt and decrypt RTP.
- https://github.com/cisco/libsrtp/releases/tag/v2.3.0
ffmpeg-4.2.tar.gz opus-1.3.1.tar.gz
- http://ffmpeg.org/releases/ffmpeg-4.2.tar.gz
- https://github.com/xiph/opus/releases/tag/v1.3.1
- To support RTMP/WebRTC transcoding.
- https://ossrs.net/lts/zh-cn/license#ffmpeg
gtest-fit
- google test framework.
- https://github.com/google/googletest/releases/tag/release-1.11.0
gperftools-2-fit
- gperf tools for performance benchmark.
- https://github.com/gperftools/gperftools/releases/tag/gperftools-2.9.1
st-srs st-1.9.zip state-threads state-threads-1.9.1.tar.gz
- Patched ST from https://github.com/ossrs/state-threads
- https://ossrs.net/lts/zh-cn/license#state-threads
JSON
USRSCTP
links:
- state-threads: https://github.com/ossrs/state-threads
- x264: ftp://ftp.videolan.org/pub/videolan/x264/snapshots/x264-snapshot-20131129-2245-stable.tar.bz2
- lame: http://nchc.dl.sourceforge.net/project/lame/lame/3.99/lame-3.99.5.tar.gz
- yasm: http://www.tortall.net/projects/yasm/releases/yasm-1.2.0.tar.gz
- speex: http://downloads.xiph.org/releases/speex/speex-1.2rc1.tar.gz