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Update live.md
Placed `ffmpeg:http_cam#audio=opus` in quotes so it doesn't appear as commented out in docs.
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@ -24,10 +24,10 @@ 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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- "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 http 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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- "ffmpeg:http_cam#audio=opus" # <- copy of the stream which transcodes audio to the missing codec (usually will be opus)
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```
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### Setting Stream For Live UI
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@ -39,10 +39,10 @@ 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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- "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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- "ffmpeg:rtsp_cam_sub#audio=opus" # <- copy of the stream which transcodes audio to opus
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cameras:
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test_cam:
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