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Update docs to reflect new live stream name

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This commit is contained in:
Nick Mowen 2023-01-16 09:57:57 -07:00
parent 8f696731f9
commit 43f17de601
2 changed files with 31 additions and 0 deletions
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3
docs/docs/configuration/index.md
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@ -360,6 +360,9 @@ go2rtc:
# Optional: jsmpeg stream configuration for WebUI # Optional: jsmpeg stream configuration for WebUI
live: live:
# Optional: Set the name of the stream that should be used for live view
# in frigate WebUI. (default: name of camera)
stream_name: camera_name
# Optional: Set the height of the jsmpeg stream. (default: 720) # Optional: Set the height of the jsmpeg stream. (default: 720)
# This must be less than or equal to the height of the detect stream. Lower resolutions # This must be less than or equal to the height of the detect stream. Lower resolutions
# reduce bandwidth required for viewing the jsmpeg stream. Width is computed to match known aspect ratio. # reduce bandwidth required for viewing the jsmpeg stream. Width is computed to match known aspect ratio.

28
docs/docs/configuration/live.md
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@ -43,6 +43,34 @@ go2rtc:
test_cam: rtsp://192.168.1.5:554/live0 test_cam: rtsp://192.168.1.5:554/live0
``` ```
### Setting Stream For Live UI
There may be some cameras that you would prefer to use the sub stream for live view, but the main stream for recording. This can be done via `live -> stream_name`.
```yaml
go2rtc:
streams:
test_cam: ffmpeg:rtsp://192.168.1.5:554/live0#video=copy#audio=aac#audio=opus
test_cam_sub: ffmpeg:rtsp://192.168.1.5:554/substream#video=copy#audio=aac#audio=opus
cameras:
test_cam:
ffmpeg:
output_args:
record: preset-record-audio-copy
inputs:
- path: rtsp://127.0.0.1:8554/test_cam?video=copy&audio=aac # <--- the name here must match the name of the camera in restream
input_args: preset-rtsp-restream
roles:
- record
- path: rtsp://127.0.0.1:8554/test_cam_sub?video=copy&audio=aac # <--- the name here must match the name of the camera_sub in restream
input_args: preset-rtsp-restream
roles:
- detect
live:
stream_name: test_cam_sub
```
### WebRTC extra configuration: ### WebRTC extra configuration:
WebRTC works by creating a TCP or UDP connection on port `8555`. However, it requires additional configuration: WebRTC works by creating a TCP or UDP connection on port `8555`. However, it requires additional configuration: