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24235d8b6a
1. After enabling FFmpeg opus, the transcoding time for each opus packet is around 4ms. 2. To speed up case execution, our test publisher sends 400 opus packets at intervals of 1ms. 3. After the publisher starts, wait for 30ms, then the player starts. 4. Due to the lengthy processing time for each opus packet, SRS continuously receives packets from the publisher, so it doesn't switch coroutines and can't accept the player's connection. 5. Only after all opus packets are processed will it accept the player connection. Therefore, the player doesn't receive any data, leading to the failure of the case. --------- Co-authored-by: winlin <winlinvip@gmail.com> |
1 year ago | |
|---|---|---|
| .. | ||
| ffmpeg-4-fit | 2 years ago | |
| gperftools-2-fit | 3 years ago | |
| gprof | 5 years ago | |
| gtest-fit | 3 years ago | |
| httpx-static | 1 year ago | |
| libsrtp-2-fit | 3 years ago | |
| openssl-1.1-fit | 3 years ago | |
| patches | 1 year ago | |
| signaling | 2 years ago | |
| srs-bench | 1 year ago | |
| srt-1-fit | 1 year ago | |
| st-srs | 2 years ago | |
| README.md | 1 year ago | |
| openssl-OpenSSL_1_0_2u.tar.gz | 5 years ago | |
| opus-1.3.1.tar.gz | 5 years ago | |
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