By default, caching is enabled during compilation, which means that data
is cached in Docker. This helps to avoid compiling third-party
dependency libraries. However, sometimes when updating third-party
libraries, it's necessary to disable caching to temporarily verify if
the pipeline can succeed. Therefore, a configure option should be added.
When this option is enabled, the compilation cache will not be used, and
all third-party libraries will be compiled from scratch.
---------
Co-authored-by: winlin <winlinvip@gmail.com>
* v6.0, 2023-10-21, Merge [#3847](https://github.com/ossrs/srs/pull/3847): WebRTC: TCP transport should use read_fully instead of read. v6.0.94 (#3847)
* v6.0, 2023-10-21, Merge [#3847](https://github.com/ossrs/srs/pull/3847): WebRTC: TCP transport should use read_fully instead of read. v6.0.94 (#3847)
* v6.0, 2023-10-20, Merge [#3846](https://github.com/ossrs/srs/pull/3846): Added system library option for ffmpeg, srtp, srt libraries. v6.0.93 (#3846)
* v6.0, 2023-10-20, Merge [#3846](https://github.com/ossrs/srs/pull/3846): Added system library option for ffmpeg, srtp, srt libraries. v6.0.93 (#3846)
@ -107,6 +108,7 @@ The changelog for SRS.
<aname="v5-changes"></a>
<aname="v5-changes"></a>
## SRS 5.0 Changelog
## SRS 5.0 Changelog
* v5.0, 2023-11-01, Merge [#3858](https://github.com/ossrs/srs/pull/3858): Support build without cache to test if actions fail. v5.0.196 (#3858)
* v5.0, 2023-10-21, Merge [#3847](https://github.com/ossrs/srs/pull/3847): WebRTC: TCP transport should use read_fully instead of read. v5.0.194 (#3847)
* v5.0, 2023-10-21, Merge [#3847](https://github.com/ossrs/srs/pull/3847): WebRTC: TCP transport should use read_fully instead of read. v5.0.194 (#3847)
* v5.0, 2023-10-20, Merge [#3846](https://github.com/ossrs/srs/pull/3846): Added system library option for ffmpeg, srtp, srt libraries. v5.0.193 (#3846)
* v5.0, 2023-10-20, Merge [#3846](https://github.com/ossrs/srs/pull/3846): Added system library option for ffmpeg, srtp, srt libraries. v5.0.193 (#3846)
<li>If you want low-latency live streaming (3-5 seconds), you can use HTTP-FLV with the <ahref="https://github.com/xqq/mpegts.js">flv.js</a> player for H5/MSE playback of HTTP-FLV.</li>
<li>If you want low-latency live streaming (3-5 seconds), you can use HTTP-TS with the <ahref="https://github.com/xqq/mpegts.js">mpegts.js</a> player for H5/MSE playback of HTTP-TS.</li>
<li>If you are not sensitive to latency (5-10 seconds) and want better cross-platform compatibility, you can use HLS with the <ahref="https://github.com/video-dev/hls.js/">hls.js</a> player for H5/MSE playback of HLS.</li>
<li>If you want ultra-low latency (within 1 second) and only need to support mainstream browsers, you can use WebRTC with the <ahref="rtc_player.html">RTC player</a>.</li>