Thanks for clarifying ekr! I hadn't read the full details in the bug and
github pages.
On Thu, Dec 7, 2017 at 8:57 PM, Eric Rescorla wrote:
> Can you explain why you think this is an increased fingerprinting surface?
> The data in question here is the audio level of *incoming* media, and as
> t
On 12/7/17 4:47 PM, Nico Grunbaum wrote:
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
Thank you for the link!
Yes, we have coverage via mochitests.
Great.
No, not ye
Can you explain why you think this is an increased fingerprinting surface?
The data in question here is the audio level of *incoming* media, and as
the bug indicates, there are other ways to obtain it.
-Ekr
On Thu, Dec 7, 2017 at 3:41 PM, Tanvi Vyas wrote:
> Is there a pref to turn this added
Is there a pref to turn this added functionality off? That way users who
are worried about their fingerprint can change the about:config pref? Or
perhaps it can be disabled when privacy.resistFingerprinting is set to true?
Thanks!
~Tanvi
On Tue, Dec 5, 2017 at 9:19 PM, Nico Grunbaum wrote:
>
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] f
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 getSynchronizationSou
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 us
On 12/6/17 12:19 AM, Nico Grunbaum wrote:
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-getstat
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 lev
9 matches
Mail list logo
