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rtmp: support additional enhanced RTMP features (#4168) (#4321) (#4954)

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* support reading AV1, VP9, H265, Opus, AC-3, G711, LPCM
* support reading multiple video or audio tracks at once
This commit is contained in:
Alessandro Ros
2025-09-11 23:18:46 +02:00
committed by GitHub
parent 2be164acd6
commit 9318107779
31 changed files with 3029 additions and 649 deletions
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docs/2-usage/3-read.md
+13 -1
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@@ -9,7 +9,7 @@ Live streams can be read from the server with the following protocols and codecs
| [SRT](#srt) | | H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video | Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3 |
| [WebRTC](#webrtc) | WHEP | AV1, VP9, VP8, H265, H264 | Opus, G722, G711 (PCMA, PCMU) |
| [RTSP](#rtsp) | UDP, UDP-Multicast, TCP, RTSPS | AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec | Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec |
| [RTMP](#rtmp) | RTMP, RTMPS, Enhanced RTMP | H264 | MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3) |
| [RTMP](#rtmp) | RTMP, RTMPS, Enhanced RTMP | AV1, VP9, H265, H264 | Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G711 (PCMA, PCMU), LPCM |
| [HLS](#hls) | Low-Latency HLS, MP4-based HLS, legacy HLS | AV1, VP9, H265, H264 | Opus, MPEG-4 Audio (AAC) |
We provide instructions for reading with the following software:
@@ -182,6 +182,18 @@ The RTSP protocol supports several underlying transport protocols, each with its
ffmpeg -rtsp_transport tcp -i rtsp://localhost:8554/mystream -c copy output.mp4
```
FFmpeg can also read a stream with RTMP:
```sh
ffmpeg -i rtmp://localhost/mystream -c copy output.mp4
```
In order to read AV1, VP9, H265, Opus, AC3 tracks and in order to read multiple video or audio tracks, the `-rtmp_enhanced_codecs` flag must be present:
```sh
ffmpeg -rtmp_enhanced_codecs ac-3,av01,avc1,ec-3,fLaC,hvc1,.mp3,mp4a,Opus,vp09 -i rtmp://localhost/mystream -c copy output.mp4
```
### GStreamer
GStreamer can read a stream from the server in several ways (RTSP, RTMP, HLS, WebRTC with WHEP, SRT). The recommended one consists in reading with [RTSP](#rtsp):