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Vmix Call audio streaming quality
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Joined: 3/26/2017(UTC) Posts: 23
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What are the specifications of the audio streaming quality settings for a Vmix call? What compression and EQ is done to the audio side of the call stream?
We are audio/music people and the quality of the audio side is (dare I say it) more important than the video :) (runs and hides) Is there anyway we can get Vmix callers audio rate up higher - to say at least 128kbps? - mp3 quality?
Thanks for all your help!
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Rank: Member
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Joined: 3/26/2017(UTC) Posts: 23
Was thanked: 3 time(s) in 2 post(s)
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So I know you're using WebRTC and the Opus Codec. I also know you're using the 48Khz sampling rate. The codec supports up to 510 kb/s ... so can we get this increased from ~30 (or whatever low amount it currently is)
Or at least have a setting where we can adjust that.
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1 user thanked ranz for this useful post.
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Rank: Advanced Member
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Joined: 7/1/2015(UTC) Posts: 1,151 Location: Houston TX Thanks: 319 times Was thanked: 263 time(s) in 233 post(s)
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Opus does not require that much bandwidth to deliver production quality audio. https://opus-codec.org/comparison/Opus at 32 kbps is great for voice. Opus at 64 kbps is better than 128 kbps MP3. Rates above 128 kbps are generally only used for multichannel streams. I'd be willing to bet than any audio quality issues you might be experiencing are related to issues of audio device, mic placement, level, etc. Not the codec or bit rate.
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