Hello. I ran into a similar issue when trying to retrieve a remote audio stream via PeerConnection, filter it through web audio, and then send the resulting stream to another peer. I know I may be asking for too much conjecture, but do you have any kind of time frame for when this issue might be fixed? Alternatively, is there a good workaround I can try (note: it has to be in Firefox)? Thank you.
On Thursday, January 9, 2014 11:21:50 AM UTC-5, Randell Jesup wrote: > On 1/9/2014 8:29 AM, [email protected] wrote: > > > I am developing a multi peer audio conferencing software (of sorts) and > > since using the Web Audio API handles playback better (faster) than an HTML > > 5 audio element. I would like to use it. Firefox is currently supporting > > Web Audio (AudioContext) and so I am working on migrating my software over > > to playing back both local and remote streams using AudioContext for > > Firefox (because Chrome does not support "createMediaStreamSource" on > > remote streams... yet) > > > > > > Additionally, for those browsers that support WebRTC and Web Audio we give > > them the possibility to use effects (such as distortion and reverb) on > > their stream. Here is where I am running into problems and I can't find > > exactly where the problem lies... so I'm having difficulty in coming up > > with a solution. > > > > > > If a peer using Chrome(31) sends a stream that has effects applied to it > > using Web Audio both Firefox(26-29) and Chrome can playback the stream. If > > a peer is using Firefox and applies effects using Web Audio neither can > > playback the stream. > > > > > > There is no apparent difference in SDP offers/answers. The only real > > difference, that I discovered, is that the Firefox stream taken directly > > from getUserMedia is labeled as [object LocalMediaStream] and if I pass it > > through AudioContext it is then labeled as [object MediaStream]... even > > after using "addStream" to set it as my local stream when setting up the > > peer connection. > > > > To reduce latency and avoid latency buildup in clock rate mismatch cases > > between the internal streams and the microphone input, audio going from > > a getUserMedia MediaStream (LocalMediaStream) to a PeerConnection > > 'bypasses' the MediaStreamGraph itself (PeerConnection uses direct > > notification of data being added to the stream as its source, instead of > > the processed output of the stream). This needs to be fixed for cases > > where the stream added to the PeerConnection isn't a LocalMediaStream, > > though that also means that there's a possibility that microphone > > sources processed with WebAudio will drift (more delay or occasional > > underflows/glitches), which also needs to be resolved with resampling > > (bug 908834 and 818822) > > > > I filed a bug to track this, bug 958090. Note that while it can be > > fixed on it's own, as mentioned it will cause problems when used until > > the resampling bugs are fixed. > > > > -- > > Randell Jesup, Mozilla _______________________________________________ dev-media mailing list [email protected] https://lists.mozilla.org/listinfo/dev-media
