clarifications and fixes for live go2rtc example

This commit is contained in:
Nicolas Mowen 2023-10-11 14:56:13 -06:00 committed by GitHub
parent dcafcc1320
commit ab1b762ddd
No known key found for this signature in database
GPG Key ID: 4AEE18F83AFDEB23

View File

@ -37,12 +37,12 @@ There may be some cameras that you would prefer to use the sub stream for live v
```yaml
go2rtc:
streams:
rtsp_cam:
test_cam:
- 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
rtsp_cam_sub:
- "ffmpeg:rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc
test_cam_sub:
- 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:rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus for webrtc
cameras:
test_cam:
@ -59,7 +59,7 @@ cameras:
roles:
- detect
live:
stream_name: rtsp_cam_sub
stream_name: test_cam_sub
```
### WebRTC extra configuration: