go-webrtc
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keroserene
Add support for sending and receiving audio, and all that entails.
My motivation for implementing this only included receiving and sending audio (as a back-end service would do), not recording or playing audio from/to microphone and speakers (as a front-end app might). Therefore, I stuck with a fake AudioDeviceModule [0]. Down the road, it might be desirable to allow real AudioDeviceModules.
addresses #7
[0] Another reason for using a fake AudioDeviceModule was so I could disable its capture and push of audio, which audio ended up being interleaved with the audio from sent tracks, corrupting it on the receiving end. There must be a better way to fix this, but I couldn't find it.
Any progress on this? I am ready to start putting in more A/V support, and I would prefer to go with this approach if it is approved.
Any progress on this?
It seems reasonable but I haven't looked at it in too much detail. Suffice it to say, it's probably safe to build on.
I've been working on a new project based on this and it's been working well so far. I'll see if I can clean up some of the code and contribute back to this.
