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Try to fix two blackbox test: 1. TestSlow_SrtPublish_HttpTsPlay_HEVC_Basic: fixed by enlarge the wait time from 3 seconds to 4 before run ffprobe task, which will record the stream by ffmpeg first. 2 TestSlow_SrtPublish_HlsPlay_HEVC_Basic: fixed by wait 16 seconds before run ffprobe task. About the 2 case: it seems ridiculous to add 16 seconds delay before run ffprobe task. > So, I start #4088 to check the github action workflow process, I start this branch from a very earlier version `6.0.113 (srs/core/srs_core_version6.hpp)`, what I found it that, inside `SRS blackbox-test`, the srs version `6.0.128`, the latest version, was printed. That's really wired. I confirmed this is not the SRS source code's problem, check https://github.com/suzp1984/srs/actions/runs/9511600525/job/26218025559 the srs code 6.0.113 was checkout and running actions based on them, still met same problem. --------- 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 | ||
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