- From: Alex Gouaillard <alex.gouaillard@temasys.com.sg>
- Date: Thu, 16 Jan 2014 16:28:09 +0800
- To: Silvia Pfeiffer <silviapfeiffer1@gmail.com>
- Cc: Alexey Aylarov <alexey@zingaya.com>, Rob Manson <robman@mob-labs.com>, "public-webrtc@w3.org" <public-webrtc@w3.org>
so it fits well the first part "Spec (in progress)", but it was not listed. On Thu, Jan 16, 2014 at 3:15 PM, Silvia Pfeiffer <silviapfeiffer1@gmail.com> wrote: > Correct - this is the state of affairs on Chrome. However, I don't > think that's a spec problem, but rather an implementation issue. There > are bugs in place and they are being worked on. > Silvia. > > On Thu, Jan 16, 2014 at 6:12 PM, Alex Gouaillard > <alex.gouaillard@temasys.com.sg> wrote: >> @alexey, if i remember correctly, the problem was about getting a >> stream from PC, so a remote stream, and then plug it in web audio. For >> video, I'm pretty sure it is even less advanced than audio. >> >> >> >> On Thu, Jan 16, 2014 at 2:08 PM, Alexey Aylarov <alexey@zingaya.com> wrote: >>> Actually you can send a stream from WebAudio to PeerConnection in Chrome. For example, you can open mp3 file and send it together with microphone stream via PeerConnection or without mic stream as well. Chrome has some other issues related to that which prevent client conference mixing, but Chrome team knows about that and will fix it in the future. In Firefox it's worse - WebAudio isn't connected to WebRTC yet, so you can't send a stream from WebAudio to PeerConnection there, I filled the bug and Mozilla team knows about the problem, but it has rather low priority at the moment. >>> >>> Alexey >>> >>>> On Jan 16, 2014, at 3:53, Rob Manson <robman@mob-labs.com> wrote: >>>> >>>> Hey Justin, >>>> >>>> thanks for the summary...makes it much easier to make sure I'm not re-raising something I missed in the "long" threads 8) >>>> >>>> I'd also like to add that "there's no way to add programmatically generated streams to peer connections". I'll assume you'll probably want to put that in "Spec (v2)". >>>> >>>> At the moment the only streams that can be used are the ones generated by gUM. It would also be useful to be able to add post-processed streams (or probably more accurately MediaStreamTracks e.g. use face tracking to mask a persons face, or use object detection to highlight a specific object, etc.). At the moment the only way to send this data to the remote client is via another channel (e.g. DC or WS)...and sync'ing is definitely an issue there. >>>> >>>> For example we would definitely like to be able to share the Augmented Web video streams that we can now create. >>>> See one example here http://youtu.be/OJHgBSRJNJY >>>> >>>> This also goes for Web Audio API generated audio tracks too. >>>> >>>> NOTE: The generation of these streams/tracks is obviously outside the webrtc spec scope. >>>> >>>> roBman >>>> >>>> >>>>> On 16/01/14 10:18 AM, Justin Uberti wrote: >>>>> Thanks to everyone who posted about what is missing in WebRTC. I attempted to collate the results below, sorted into either "spec" or "implementation" categories. >>>>> >>>>> Basically, I think the key things that are causing trouble are being actively worked on both in this WG and in implementations; we are on track to resolve these problems, hopefully in the next few months. >>>>> >>>>> Full list below: >>>>> >>>>> Spec (in progress) >>>>> >>>>> * >>>>> >>>>> Bad error notifications >>>>> >>>>> * >>>>> >>>>> Lower image resolution without stopping the stream (RTCRtpSender >>>>> or MST.applyConstraints) >>>>> >>>>> * >>>>> >>>>> API for capping bandwidth (RTCRtpSender) >>>>> >>>>> * >>>>> >>>>> Recording of streams (MediaStreamRecorder) >>>>> >>>>> * >>>>> >>>>> More debugging of candidate pair states (getStats) >>>>> >>>>> * >>>>> >>>>> Determine type of candidate (getStats) >>>>> >>>>> * >>>>> >>>>> List all the DCs on a PC (TBD if we need this or not) >>>>> >>>>> >>>>> Spec (v2) >>>>> >>>>> * >>>>> >>>>> Too attached for SDP, O/A >>>>> >>>>> * >>>>> >>>>> TURN auth failure does not cause an error >>>>> >>>>> * >>>>> >>>>> Better control of video mute behavior >>>>> >>>>> * >>>>> >>>>> Screen sharing without extensions (maybe) >>>>> >>>>> >>>>> Spec (future) >>>>> >>>>> * >>>>> >>>>> Access PeerConnection from Web Workers >>>>> >>>>> * >>>>> >>>>> Keep PeerConnection across reload/navigation >>>>> >>>>> >>>>> Implementations >>>>> >>>>> * >>>>> >>>>> Stable multi-stream support (working on this, some spec dependencies) >>>>> >>>>> * >>>>> >>>>> NAT/FW traversal, connection stability issues (Chrome working on >>>>> this in Q1) >>>>> >>>>> * >>>>> >>>>> AEC performance issues (Chrome working on this in Q1) >>>>> >>>>> * >>>>> >>>>> BWE and handling of low-bandwidth situations (Chrome working on >>>>> this in Q1) >>>>> >>>>> * >>>>> >>>>> Not all ICE states implemented/ICE never goes to failed (Chrome >>>>> working on this in Q1) >>>>> (Chrome: https://code.google.com/p/webrtc/issues/detail?id=1414) >>>>> >>>>> >>>>> Nontechnical >>>>> >>>>> * WebRTC support in other browsers (IE, Safari) >>>> >>>> -- >>>> Rob >>>> >>>> Checkout my new book - Getting started with WebRTC >>>> http://www.packtpub.com/getting-started-with-webrtc/book >>>> >>>> CEO & co-founder >>>> http://mob-labs.com >>>> >>>> Chair of the W3C Augmented Web Community Group >>>> http://www.w3.org/community/ar >>>> >>>> Invited Expert with the ISO, Khronos Group & W3C >>>> >>>> >>> >>
Received on Thursday, 16 January 2014 08:28:38 UTC