Audio quality - avoid automatic input level control

Hi developers,
I’m not a developer, I’m a user, with special needing.
I’m a musician, pianist and piano teacher. I tried to work online, doing lesson or merely listening my students sometimes, using most common social networks : Whatsapp, Skype, WeChat and so on; first I need absolute synchronized audio and video, and I saw that it’s one of the most feature of this platform.
On second, I would find a way to avoid the automatic input level control on the device (in my case, android smartphone), because this feature compromises very badly the dynamic range of audio stream; I’m working with classic music and I need to well select the difference between a ‘pianissimo’ and a ‘mezzoforte’.
I tried to search about but without results, I’m not a developer, but I think this problem stays behind your code, but I would ask if I can avoid it using, for example, a PC (I’m a Linux user) instead of a smartphone, or what do I need to avoid it using an Android or Apple phone.
I thank you in advance for your patience, hoping to bring all you an interesting subject of discussion.
My greetings

Hi Dario,

You can disable automatic gain control by appending this to your URL: “#config.disableAGC=true”. This should work on Chrome on the desktop, but I don’t think it will work on mobile. Perhaps someone else can chime in if they know of a way to pass config params on mobile.

Also see here for more audio processing options you can disable this way:

To pass more than one option you need to separate by ampersand, e.g.:

Hope this helps,


Passing the https link with params in the join room field should work, at least it used to work. We were using that for testing using torture.

I thank you very much; you were opening a window I was searching for long time; I’m using Firefox instead of Chrome, but I can use it as well; unfortunately the most common devices to be used for my activity would be mobile devices, but it’s a good starting point anyhow.
Where may I get more details about the quality of the audio received (bandwidth, dynamic range)? is it a feature of WebRTC or there is a codec operating some processes ?