Rendezvous Mic Gain Auto-Adjusts?

Does Rendezvous use automatic gain control with WebRTC?  Can this be disabled?

I record a podcast & Youtube series using Rendezvous with 2 other participants.  They both have different audio setups and both highlight issues with automatic gain control.  Both require a change of -6dB to reduce their levels to something suitable for recording.

One participant has a Blue Yeti mic with its own gain control on a Mac.  I've had him reduce the gain on his end multiple times.  No matter how low it goes, I still need to bring his level down that 6dB on my end with Audio Mixer.  For best audio quality, I'd like to have him set the gain on his end and make no adjustments on my end.  But it seems that WebRTC makes this impossible.

The other participant uses a Blue Snowball on a PC and moves around a lot, changing the distance to his mic.  His mic gain is definitely auto-adjusting, which produces terrible quality for a second or two every time he moves closer to the mic.  Audio quality is poor when WebRTC increases the gain too far and the -6dB change doesn't help this.

Any help with this would be appreciated.

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  • In Wirecast Pro have you considered using the FX Filters?
    Limiter, Loudness, Dynamics depending on the issue.

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  • CraigS 

    Yes, I've used some of the filters.  That was mostly unsuccessful and produced worse audio, probably because I don't yet know how to implement them.  I'm still attempting to learn more about them.  But they are a risk to my recording.

    Filters just seem like a workaround and don't actually address the original problems created by automatic gain control.  I'd like to modify the audio as little as possible, not more.

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  • Robert Pierce WebRTC wouldn't have any audio controls built in that we can control. It's really just the source processed. You'd have to either implement on the receiving side (the filters for example) or on the send side (the guest location).

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  • CraigS Are you suggesting that the browser (Chrome) or OS is causing the auto-gain adjustments?

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  • Robert Pierce said:
    Does Rendezvous use automatic gain control with WebRTC?  Can this be disabled? 

    The answer to that is a big NO. I use Rendezvous every time I use wirecast, and there is no auto gain control.

    BUT: There is a trap that can get you into trouble. Some applications on the client computer DO automatically adjust microphone gain. I've found some that even do it when they are in the background. For example - I know that Skype plays with volume levels.

    Regarding audio enhancements. I use two plugins all the time:

    1. Dynamics: Start with defaults, and change Limiter to -1db, and Gate Threshold to -35db.
    2. Noise Reduction: All default values

    Just to share - There are two big problems with audio plugins:

    • They are shot specific. If I have one audio input appearing in lots of shots, I have to go to each and every one to apply and configure the same settings again and again.
    • Cannot be configured in preview: If you are doing Rendezvous - and a guest joins after the start of a broadcast, you cannot configure plugins before the guest goes live... which means your guest sounds crap for the first say 60 seconds unless you get them to pre-join before the broadcast. Unfortunately, there is no workaround.
    • If you put the above two points together, its even worse. You have a guest join after broadcast starts. You send the first shot live. You configure the audio perfectly. Then, you want to change to another shot with the same guest - and perhaps another guest in the same shot. You have to re apply the plugins again.

    Conclusion: Audio is one of the biggest weaknesses in Wirecast. I know there are feature requests in for various enhancements. I understand that some of the issues are planned to be tweaked in V12. Time will tell I guess.

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  • Greg Kuhnert 

    I've seen others reporting that Skype changes gain and have followed the recommended steps to prevent that from happening.  Both Windows and MacOS have some hardware setting to change.  But that didnt affect anything.  The Windows machine is brand new and doesn't even have Skype or other chat programs installed, so its not that.  I guess I need to keep digging.

    Thnx for the suggestions on filter settings.

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  • Just found this in Chrome flags: #enable-webrtc-hybrid-agc 

    I'm going to test with that disabled.

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  • Robert Pierce It'll be interesting to see if that has any impact. Although, from your previous description, AGC is default on when Chrome uses webrtc.

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