- From: Peter Thatcher <pthatcher@google.com>
- Date: Thu, 3 Sep 2015 21:30:48 -0700
- To: Cullen Jennings <fluffy@iii.ca>
- Cc: "public-webrtc@w3.org" <public-webrtc@w3.org>, Harald Tveit Alvestrand <harald@alvestrand.no>
- Message-ID: <CAJrXDUFJxkW2gHjbsZJBX8iw+o8Sh6prwzLMP7s5Y4VhetWqDg@mail.gmail.com>
By the way, if it helps, this is the best we had at the time when we gave up: dictionary RTCRtpContributingSource { double packetTime; unsigned int csrc; int audioLevel; { partial interface RTCRtpReceiver { // A read rather than an event because it would fire so frequently. sequence<RTCRtpContributingSource> getContributingSources(); } On Thu, Sep 3, 2015 at 9:28 PM, Peter Thatcher <pthatcher@google.com> wrote: > #4 came up in ORTC, and we went through a number of ideas and designs, and > in the end, it was a complex API to get right, and all of the developers > involved that were asking for it basically said "this isn't that important; > we can live without it". > > So, I agree with Harald's conclusion on #4. And all of this conclusions, > for that matter. > > > As for it being a MUST in the API, the chairs made a clear position that > if something is going to be part of 1.0, then it MUST have a PR by the time > of the f2f. Is there such a PR for CSRCs? > > > On Thu, Sep 3, 2015 at 9:56 AM, Cullen Jennings <fluffy@iii.ca> wrote: > >> >> I think #4 is MUST have and there was consensus to put it in. I don't see >> consensus to remove it. This is critical for building conferencing systems >> where there is rapid indication of which parties are adding sound to the >> conference. This is a feature many developers ask for. >> >> I think that we must have a way to have Client-to-mixer level sent by in >> RTP. If the default is to always offer this in the SDP and accept it if >> offered, and do it if negotiated in SDP, then I don't think that a controls >> surface to turn that on and off is needed. (people can mangle SDP if they >> want to turn it off). But if we decide the default is not to always offer >> this, then I think the #276 is needed. >> >> I don't care too much about #6 but it pretty trivial and don't have any >> objections to it going in >> >> >> > On Sep 1, 2015, at 10:03 AM, Harald Alvestrand <harald@alvestrand.no> >> wrote: >> > >> > We have 3 issues that relate to CSRC information in our tracker. >> > We've had debate about them - most recently from July 31 to August 18 - >> > but very few people have contributed, and there seems to be no strong >> > consensus. >> > >> > The tracker issues are: >> > >> > #4 Need API to read the CSRC on received tracks >> > #6 Need API to receive mixer to client audio level information for a >> track >> > #276 Control to turn on Client-to-Mixer Audio Level in offers >> > >> > The contributors and their (no doubt mischaracterized) positions from >> > the August mail thread are roughly: >> > >> > - Bernard Aboba has called CSRC information "not a must have" >> > - Cullen Jennings has called CSRC information "useful for a bunch of >> things" >> > - Inaki Baz Castillo has said that "the usecase is **really** big" >> > - Harald Alvestrand (me) has said that "I don't regard this as an >> > important use case". >> > >> > There was also a spate of debate about this in February, with a few more >> > participants. >> > >> > Given that: >> > >> > a) there doesn't seem to be a consensus that CSRC information is >> important >> > b) we're trying to limit the scope of 1.0 features that we don't have >> > consensus to add >> > c) generation of volume information is hard to retrofit later (old >> > clients wouldn't generate it) >> > d) new interfaces for retrieving information are not so hard to retrofit >> > later >> > >> > I'm suggesting the following disposition for these issues: >> > >> > #4: We don't add any such API in WebRTC 1.0 >> > #6: We don't add any such API in WebRTC 1.0. >> > #276: We find a convenient place to say that implemementations MUST use >> > this feature (that is, send volume information) if negotiated >> successfully. >> > We don't add any API for turning this functionality off in WebRTC 1.0. >> > >> > This will give us a situation where volume information is flowing to the >> > mixer, and the mixer can take advantage of it if desired - and at a >> > later stage, we can add APIs for reading the information that already >> > flows in the system on incoming flows. >> > >> > Does this seem like an appropriate disposition given where we are? >> > >> > Harald >> > >> > >> > >> >> >> >
Received on Friday, 4 September 2015 04:31:56 UTC