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How to approach audio latency?
Rank: Member
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Joined: 4/12/2020(UTC) Posts: 11 Location: London Thanks: 2 times
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My aim is to use vMix to speak to a person via Skype and demonstrate pre-recorded material to them.
This means that there are three audio streams: my voice, the person speaking via Skype and the pre-recorded material on the PC.
vMix generates latency: if I monitor my own voice via vMix I hear everything but my voice is delayed. If I monitor my self 'live' from the mixer I don't get to hear the pre-recorded material or the other side of the conversation. How do I square this circle?
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Rank: Advanced Member
Groups: Registered
Joined: 7/19/2017(UTC) Posts: 45 Thanks: 11 times Was thanked: 14 time(s) in 7 post(s)
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Audio buses. Pure and simple.
Assign the pre-recorded material and the other person to a bus, this will be your monitoring bus. (A Mix-minus, as in you're receiving everything mixed minus your voice)
You really shouldn't need to monitor your own voice, for the reason you discovered. Latency. It will always throw you off.
Conversely, you should send your guest on Skype every bit of audio minus their voice. Assign you, and the material to another bus, and send that as your outgoing audio.
You shouldn't ever send "master" to any guest, because the chance of them hearing themselves back get higher. (You never want that)
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1 user thanked NewsworthyVision for this useful post.
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How to approach audio latency?
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