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OPUS is great and is the best audio codec hands down.
But sometimes we need other codecs. For example, legacy HLS (browser) players can only play AAC audio.
go2rtc can be used for that:
streams:
my_cam:
- rtsp://thingino:thingino@192.168.1.31/ch1
- ffmpeg:my_cam#audio=aacBut this will invoke ffmpeg on the receiving end, which consumes CPU resources and (probably) adds some latency.
It would be great if we could have different audio codecs per channel, so that I could do something like this in go2rtc:
streams:
my_cam:
# ch1 would be set to send OPUS
- rtsp://thingino:thingino@192.168.1.31/ch1
# copy AAC audio track from my_cam_hd
- rtsp://127.0.0.1:8554/my_cam_hd?audio=aac
my_cam_hd:
# ch0 would be set to send AAC
- rtsp://thingino:thingino@192.168.1.31/ch0
# copy OPUS audio track from my_cam
- rtsp://127.0.0.1:8554/my_cam?audio=opusWhich would not require any transcoding. :)
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