10000 Track live and local audio and convert to buffer · Issue #110 · stasel/WebRTC · GitHub
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Track live and local audio and convert to buffer #110

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TayyabAli652 opened this issue Dec 10, 2024 · 2 comments
Open

Track live and local audio and convert to buffer #110

TayyabAli652 opened this issue Dec 10, 2024 · 2 comments

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@TayyabAli652
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Hello @stasel,

I’m using your library and encountering an issue. I need to continuously monitor both local and remote audio streams as someone speaks, convert the audio into a buffer, and send it over a WebSocket. However, I couldn’t find a way to achieve this with the library.

One potential solution I came across was to use addSink on the audio track, but it seems that addSink is not available in this library. Could you please guide me on how I can implement this functionality?

Thank you!
@TayyabAli652
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@stasel any update on this?

@stasel
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stasel commented Mar 5, 2025

Hi @TayyabAli652 ,
Unfortunately I cannot provide an answer on the internal workings of the library.
It is worth asking the people on the official WebRTC group to get some answers

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