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RTMP: Support enhanced RTMP specification for HEVC. v6.0.42 (#3495)

* RTMP: Support enhanced RTMP specification for HEVC,  v6.0.42.
* Player: Upgrade mpegts.js to support it.

Enhanced RTMP specification: https://github.com/veovera/enhanced-rtmp

First, start SRS `v6.0.42+` with HTTP-TS support:

```bash
./objs/srs -c conf/http.ts.live.conf
```

Then, you can use [OBS 29.1+](https://github.com/obsproject/obs-studio/releases) to push HEVC via RTMP.
Start OBS with the following settings in the `Settings > Stream` tab:

* Server: `rtmp://localhost/live`
* Stream Key: `livestream`
* Encoder: Please select the HEVC hardware encoder.

Finally, open the player http://localhost:8080/players/srs_player.html?stream=livestream.ts

Or use VLS or ffplay to play `http://localhost:8080/live/livestream.ts`

---------

Co-authored-by: chundonglinlin <chundonglinlin@163.com>
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Winlin 2023-04-08 09:18:10 +08:00 committed by GitHub
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@ -8,8 +8,9 @@ The changelog for SRS.
## SRS 6.0 Changelog
* v5.0, 2023-04-01, Merge [#3392](https://github.com/ossrs/srs/pull/3392): Support composited bridges for 1:N protocols converting. v6.0.41 (#3392)
* v5.0, 2023-04-01, Merge [#3458](https://github.com/ossrs/srs/pull/3450): API: Support HTTP basic authentication for API. v6.0.40 (#3458)
* v6.0, 2023-04-08, Merge [#3495](https://github.com/ossrs/srs/pull/3495): RTMP: Support enhanced RTMP specification for HEVC. v6.0.42 (#3495)
* v6.0, 2023-04-01, Merge [#3392](https://github.com/ossrs/srs/pull/3392): Support composited bridges for 1:N protocols converting. v6.0.41 (#3392)
* v6.0, 2023-04-01, Merge [#3458](https://github.com/ossrs/srs/pull/3450): API: Support HTTP basic authentication for API. v6.0.40 (#3458)
* v6.0, 2023-03-27, Merge [#3450](https://github.com/ossrs/srs/pull/3450): WebRTC: Error message carries the SDP when failed. v6.0.39 (#3450)
* v6.0, 2023-03-25, Merge [#3477](https://github.com/ossrs/srs/pull/3477): Remove unnecessary NULL check in srs_freep. v6.0.38 (#3477)
* v6.0, 2023-03-25, Merge [#3455](https://github.com/ossrs/srs/pull/3455): RTC: Call on_play before create session, for it might be freed for timeout. v6.0.37 (#3455)