Re: Intent to ship WebRTC RTCRtpReceiver contributing and synchronization sources

2017-12-07 Thread Nico Grunbaum

Answers can be found inline.

-Nico

On 12/7/17 12:45 PM, Boris Zbarsky wrote:

On 12/6/17 3:22 PM, Nico Grunbaum wrote:
It is in nightly now, and we intend to ship these interfaces in 59. 
Chrome shipped a partial implementation[3] of 
getContributingSources() in 59. Edge has partial support[4] for 
getContributingSources().


OK.  Neither of those ships getSynchronizationSources(), right?

Correct those are not shipping in Chrome or Edge yet. Chrome has issued 
an intent to ship: 
https://groups.google.com/a/chromium.org/forum/#!topic/blink-dev/I39cDWKyMJo 
.  I have not found any such signals with regards to Edge.
Testing: WPT are disabled, as the tests are themselves not 
functional[5] and there is ongoing spec discussion as how to fix 
them[6].


Do we have other (non-wpt) tests for this running in our CI?


Yes, we have coverage via mochitests.
Have we done any checking of how interoperable what we're proposing to 
ship is with what the other UAs have shipped?
No, not yet, and I agree with your statements below on why this is 
important to complete.


I just filed a few issues on the spec here, because some parts of it 
are not making sense to me:


* https://github.com/w3c/webrtc-pc/issues/1688
* https://github.com/w3c/webrtc-pc/issues/1689
* https://github.com/w3c/webrtc-pc/issues/1690

For the first two, it's a question about the API shape that is being 
specified.  For the last one, it's an issue where the spec is not 
clear enough to implement interoperably, from what I can tell, and 
testing to see what various shipping implementations do might be 
interesting.


Yes, the last one in particular is an interesting point and an open 
issue https://github.com/w3c/webrtc-pc/issues/1497. I looks to be an 
2018 interim topic.  There also appears to be discussion of this issue 
on the Chrome bug for getContributingSources, 
https://bugs.chromium.org/p/chromium/issues/detail?id=625208#c11


FWIW Our implementation makes sure that timestamp field is comparable to 
Date.now() so that it will also be comparable to timestamps in 
getStats(), https://www.w3.org/TR/webrtc/#dom-rtcstats-timestamp .

-Boris
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Re: Intent to ship WebRTC RTCRtpReceiver contributing and synchronization sources

2017-12-06 Thread Nico Grunbaum

Thank you for the link to the template Boris.

Intent to ship WebRTC RTCRtpReceiver contributing and synchronization 
sources


Summary: RTCRtpReceiver's[1] getSynchronizationSources() method allows 
WebRTC applications to monitor volume levels of call participants 
without having to use the costly getStats()[2] call. RTCRtpReceiver's 
getContributingSources() method enables WebRTC applications to monitor 
the volume level of participants whose audio is being mixed upstream if 
the mixer includes this data in the RTP extension headers.


It is in nightly now, and we intend to ship these interfaces in 59. 
Chrome shipped a partial implementation[3] of getContributingSources() 
in 59. Edge has partial support[4] for getContributingSources().


Bug to turn on by default: 
https://bugzilla.mozilla.org/show_bug.cgi?id=1363667

Link to standard: https://w3c.github.io/webrtc-pc/#rtcrtpreceiver-interface
Testing: WPT are disabled, as the tests are themselves not functional[5] 
and there is ongoing spec discussion as how to fix them[6].


[1] https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver
[2] https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getstats()
[3] https://bugs.chromium.org/p/chromium/issues/detail?id=703122&desc=2
[4] https://msdn.microsoft.com/en-us/library/mt502507(v=vs.85).aspx
[5] https://bugzilla.mozilla.org/show_bug.cgi?id=1419900
[6] https://github.com/w3c/webrtc-pc/pull/1668#issuecomment-346951615

-- Nico Grunbaum


On 12/6/17 5:42 AM, Boris Zbarsky wrote:

Would you mind filling out the other parts of the intent templates at
https://wiki.mozilla.org/WebAPI/ExposureGuidelines#Intent_to_Implement 
and https://wiki.mozilla.org/WebAPI/ExposureGuidelines#Intent_to_Ship ?


At the very least, what is the status in other implementations and 
what is the testing situation?


-Boris
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Intent to ship WebRTC RTCRtpReceiver contributing and synchronization sources

2017-12-05 Thread Nico Grunbaum
Background: RTCRtpReceiver's[1] getSynchronizationSources() method 
allows WebRTC applications to monitor volume levels of call participants 
without having to use the costly getStats()[2] call. RTCRtpReceiver's 
getContributingSources() method enables WebRTC applications to monitor 
the volume level of participants whose audio is being mixed upstream if 
the mixer includes this data in the RTP extension headers.


We intend to ship these interfaces in 59, and they are on nightly now.

Tracking: https://bugzilla.mozilla.org/show_bug.cgi?id=1363667

[1] https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver
[2] https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getstats()

Cheers,
Nico Grunbaum
-. --.

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Re: More ThinkStation P710 Nvidia tips (was Re: Faster gecko builds with IceCC on Mac and Linux)

2017-11-02 Thread Nico Grunbaum
For rr I have an i7 desktop with a base clock of 4.0 Ghz, and for 
building I use icecc to distribute the load (or rather I will be again 
when bug 1412240[0] is closed).  The i9 series has lower base clocks 
(2.8 Ghz, and 2.6Ghz for the top SKUs)[1], but high boost clocks of 4.2 
Ghz.  If I were to switch over to an i9 for everything, would I see a 
notable difference in performance in rr?


-Nico

[0] https://bugzilla.mozilla.org/show_bug.cgi?id=1412240 Build failure 
in libavutil (missing atomic definitions), when building with clang and 
icecc


[1] 
https://ark.intel.com/products/series/123588/Intel-Core-X-series-Processors


On 10/27/17 7:50 PM, Robert O'Callahan wrote:

BTW can someone forward this entire thread to their friends at AMD so AMD
will fix their CPUs to run rr? They're tantalizingly close :-/.

Rob


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