WebRTC consultation help
Budget: $50 – $0 AUD
Looking for an expert consultant to help solve an issue.
I have a team of people working on a WebRTC based app for me. It uses the AntMedia servers and broadcasts realtime audio from one mobile device (iOS and Android) to multiple mobile devices.
I am trying to better align the variation in latency between devices. So I need to delay WebRTC audio playback by up to 150ms on some devices.
So essentially we want to buffer the audio stream on the output device, client side.
So far no luck looking at the WebRTC framework to intercept audio, might we need to be looking at the C++ level? Help appreciated. - Steve
I have a team of people working on a WebRTC based app for me. It uses the AntMedia servers and broadcasts realtime audio from one mobile device (iOS and Android) to multiple mobile devices.
I am trying to better align the variation in latency between devices. So I need to delay WebRTC audio playback by up to 150ms on some devices.
So essentially we want to buffer the audio stream on the output device, client side.
So far no luck looking at the WebRTC framework to intercept audio, might we need to be looking at the C++ level? Help appreciated. - Steve