mirror of
https://github.com/blakeblackshear/frigate.git
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c3b313a70d
* Initial audio classification model implementation * fix mypy * Keep audio labelmap local * Cleanup * Start adding config for audio * Add the detector * Add audio detection process keypoints * Build out base config * Load labelmap correctly * Fix config bugs * Start audio process * Fix startup issues * Try to cleanup restarting * Add ffmpeg input args * Get audio detection working * Save event to db * End events if not heard for 30 seconds * Use not heard config * Stop ffmpeg when shutting down * Fixes * End events correctly * Use api instead of event queue to save audio events * Get events working * Close threads when stop event is sent * remove unused * Only start audio process if at least one camera is enabled * Add const for float * Cleanup labelmap * Add audio icon in frontend * Add ability to toggle audio with mqtt * Set initial audio value * Fix audio enabling * Close logpipe * Isort * Formatting * Fix web tests * Fix web tests * Handle cases where args are a string * Remove log * Cleanup process close * Use correct field * Simplify if statement * Use var for localhost * Add audio detectors docs * Add restream docs to mention audio detection * Add full config docs * Fix links to other docs --------- Co-authored-by: Jason Hunter <hunterjm@gmail.com>
146 lines
6.4 KiB
Markdown
146 lines
6.4 KiB
Markdown
---
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id: restream
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title: Restream
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---
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## RTSP
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Frigate can restream your video feed as an RTSP feed for other applications such as Home Assistant to utilize it at `rtsp://<frigate_host>:8554/<camera_name>`. Port 8554 must be open. [This allows you to use a video feed for detection in Frigate and Home Assistant live view at the same time without having to make two separate connections to the camera](#reduce-connections-to-camera). The video feed is copied from the original video feed directly to avoid re-encoding. This feed does not include any annotation by Frigate.
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Frigate uses [go2rtc](https://github.com/AlexxIT/go2rtc/tree/v1.5.0) to provide its restream and MSE/WebRTC capabilities. The go2rtc config is hosted at the `go2rtc` in the config, see [go2rtc docs](https://github.com/AlexxIT/go2rtc/tree/v1.5.0#configuration) for more advanced configurations and features.
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:::note
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You can access the go2rtc webUI at `http://frigate_ip:5000/live/webrtc` which can be helpful to debug as well as provide useful information about your camera streams.
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:::
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### Birdseye Restream
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Birdseye RTSP restream can be enabled at `birdseye -> restream` and accessed at `rtsp://<frigate_host>:8554/birdseye`. Enabling the restream will cause birdseye to run 24/7 which may increase CPU usage somewhat.
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### Securing Restream With Authentication
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The go2rtc restream can be secured with RTSP based username / password authentication. Ex:
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```yaml
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go2rtc:
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rtsp:
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username: "admin"
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password: "pass"
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streams:
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...
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```
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**NOTE:** This does not apply to localhost requests, there is no need to provide credentials when using the restream as a source for frigate cameras.
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## RTMP (Deprecated)
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In previous Frigate versions RTMP was used for re-streaming. RTMP has disadvantages however including being incompatible with H.265, high bitrates, and certain audio codecs. RTMP is deprecated and it is recommended to move to the new restream role.
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## Reduce Connections To Camera
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Some cameras only support one active connection or you may just want to have a single connection open to the camera. The RTSP restream allows this to be possible.
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### With Single Stream
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One connection is made to the camera. One for the restream, `detect` and `record` connect to the restream.
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```yaml
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go2rtc:
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streams:
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rtsp_cam: # <- for RTSP streams
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- rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio
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- "ffmpeg:rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus)
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http_cam: # <- for other streams
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- http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_main.bcs&user=user&password=password # <- stream which supports video & aac audio
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- "ffmpeg:http_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus)
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cameras:
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rtsp_cam:
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ffmpeg:
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output_args:
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record: preset-record-generic-audio-copy
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inputs:
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- path: rtsp://127.0.0.1:8554/rtsp_cam # <--- the name here must match the name of the camera in restream
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input_args: preset-rtsp-restream
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roles:
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- record
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- detect
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- audio # <- only necessary if audio detection is enabled
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http_cam:
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ffmpeg:
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output_args:
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record: preset-record-generic-audio-copy
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inputs:
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- path: rtsp://127.0.0.1:8554/http_cam # <--- the name here must match the name of the camera in restream
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input_args: preset-rtsp-restream
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roles:
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- record
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- detect
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- audio # <- only necessary if audio detection is enabled
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```
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### With Sub Stream
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Two connections are made to the camera. One for the sub stream, one for the restream, `record` connects to the restream.
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```yaml
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go2rtc:
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streams:
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rtsp_cam:
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- rtsp://192.168.1.5:554/live0 # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
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- "ffmpeg:rtsp_cam#audio=opus" # <- copy of the stream which transcodes audio to opus
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rtsp_cam_sub:
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- rtsp://192.168.1.5:554/substream # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
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- "ffmpeg:rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus
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http_cam:
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- http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_main.bcs&user=user&password=password # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
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- "ffmpeg:http_cam#audio=opus" # <- copy of the stream which transcodes audio to opus
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http_cam_sub:
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- http://192.168.50.155/flv?port=1935&app=bcs&stream=channel0_ext.bcs&user=user&password=password # <- stream which supports video & aac audio. This is only supported for rtsp streams, http must use ffmpeg
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- "ffmpeg:http_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus
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cameras:
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rtsp_cam:
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ffmpeg:
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output_args:
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record: preset-record-generic-audio-copy
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inputs:
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- path: rtsp://127.0.0.1:8554/rtsp_cam # <--- the name here must match the name of the camera in restream
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input_args: preset-rtsp-restream
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roles:
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- record
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- path: rtsp://127.0.0.1:8554/rtsp_cam_sub # <--- the name here must match the name of the camera_sub in restream
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input_args: preset-rtsp-restream
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roles:
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- audio # <- only necessary if audio detection is enabled
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- detect
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http_cam:
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ffmpeg:
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output_args:
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record: preset-record-generic-audio-copy
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inputs:
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- path: rtsp://127.0.0.1:8554/http_cam # <--- the name here must match the name of the camera in restream
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input_args: preset-rtsp-restream
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roles:
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- record
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- path: rtsp://127.0.0.1:8554/http_cam_sub # <--- the name here must match the name of the camera_sub in restream
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input_args: preset-rtsp-restream
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roles:
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- audio # <- only necessary if audio detection is enabled
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- detect
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```
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## Advanced Restream Configurations
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The [exec](https://github.com/AlexxIT/go2rtc/tree/v1.5.0#source-exec) source in go2rtc can be used for custom ffmpeg commands. An example is below:
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NOTE: The output will need to be passed with two curly braces `{{output}}`
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```yaml
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go2rtc:
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streams:
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stream1: exec:ffmpeg -hide_banner -re -stream_loop -1 -i /media/BigBuckBunny.mp4 -c copy -rtsp_transport tcp -f rtsp {{output}}
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```
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