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clarifications and fixes for live go2rtc example (#8132)

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* clarifications and fixes for live go2rtc example

* fix
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Nicolas Mowen 2023-10-13 05:15:39 -06:00 committed by GitHub
parent 3869b274e2
commit 869bb2b177
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docs/docs/configuration/live.md
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@ -37,12 +37,12 @@ There may be some cameras that you would prefer to use the sub stream for live v
```yaml ```yaml
go2rtc: go2rtc:
streams: streams:
rtsp_cam: test_cam:
- rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio. - rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio.
- "ffmpeg:rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to opus - "ffmpeg:test_cam#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc
rtsp_cam_sub: test_cam_sub:
- rtsp://192.168.1.5:554/substream # <- stream which supports video & aac audio. - rtsp://192.168.1.5:554/substream # <- stream which supports video & aac audio.
- "ffmpeg:rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus - "ffmpeg:test_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc
cameras: cameras:
test_cam: test_cam:
@ -59,7 +59,7 @@ cameras:
roles: roles:
- detect - detect
live: live:
stream_name: rtsp_cam_sub stream_name: test_cam_sub
``` ```
### WebRTC extra configuration: ### WebRTC extra configuration: