So are y'all planning to deprecate and remove `getOutputTimestamp` once this ships? If so, that sounds reasonable.
I note Chris asked y'all to file for a webkit signal upthread. Did that happen? On Tuesday, November 30, 2021 at 12:54:17 AM UTC+1 hongchan wrote: > I agree that outputLatency is better in terms of both usability and > interoperability. > > FYI, the counter shows getOutputTimestamp is about 0.001% and there are no > partners who are using this API: > > https://chromestatus.com/metrics/feature/popularity#AudioContextGetOutputTimestamp > > > On Mon, Nov 22, 2021 at 3:00 PM Chris Cunningham <[email protected]> > wrote: > >> Re: privacy discussion, Hongchan and I scheduled to meet with Paul Jensen >> next week. >> >> Also, I recently determined that this new WebAudio API (outputLatency) is >> largely redundant with an existing API (getOutputTimestamp()). >> https://github.com/WebAudio/web-audio-api/issues/2461 >> >> Chrome has already shipped the latter API, so I believe this new API >> doesn't add potential for additional fingerprinting. Shipping both APIs is >> still a good idea for the sake of inter-op. >> >> On Friday, November 19, 2021 at 8:45:28 AM UTC-8 hongchan wrote: >> >>> Thanks for the clarification, Yoav! >>> >>> I hope the mitigation approach in the previous comment makes sense from >>> the privacy perspective, but it will hurt the usability/value of this API >>> significantly. >>> For the privacy-focused discussion, should I use other venues rather >>> than this I2S thread? >>> >>> Cheers, >>> Hongchan >>> >>> >>> On Thu, Nov 18, 2021 at 10:35 PM Yoav Weiss <[email protected]> >>> wrote: >>> >>>> Apologies, I misread the discussion. A spec issue is indeed not needed, >>>> as this is already covered. >>>> >>>> From my perspective, since this exposes active entropy, the fact that >>>> it exposes more data than the (passively exposed) platform is not >>>> prohibitive on its own. >>>> At the same time, it'd be good to work with privacy-focused folks to >>>> make sure we expose as little information as needed to maintain the API's >>>> functionality, but not more. >>>> >>>> On Fri, Nov 19, 2021 at 4:57 AM Chris Cunningham < >>>> [email protected]> wrote: >>>> >>>>> > If the reported value is incorrect, the A/V synchronization won't be >>>>> possible. (Perhaps chcunningham@ can say more about this use case.) >>>>> >>>>> The typical video player design drives the clock with audio. In short, >>>>> you send buffers of audio to the OS, use the latency value to know when >>>>> those buffers actually reach the user's ears, and choose a video frame >>>>> accordingly. Apps using WebCodecs will design apps this way (example >>>>> here <https://github.com/chcunningham/wc-talk#simple_video_playerhtml> >>>>> ). >>>>> >>>>> As Mike notes, the latency value can vary by quite a lot depending on >>>>> the device (huge differences for bluetooth vs wired), so it's critical to >>>>> have the platform tell you when this changes. In terms of precision, we >>>>> can >>>>> _probably_ do some rounding like to the nearest millisecond. Higher >>>>> values >>>>> may lead to issues; user studies >>>>> <https://en.wikipedia.org/wiki/Audio-to-video_synchronization#Recommendations> >>>>> >>>>> have shown sensitivity to fairly low values of drift. >>>>> >>>>> Aside: we may eventually desire a display latency value for similar >>>>> reasons. >>>>> >>>>> On Thu, Nov 18, 2021 at 12:45 PM Hongchan Choi <[email protected]> >>>>> wrote: >>>>> >>>>>> Hi Yoav, >>>>>> >>>>>> Could you clarify what needs to be discussed from the spec side? The >>>>>> feature (and its behavior) is properly specified, and its privacy aspect >>>>>> is >>>>>> already reviewed by W3C: >>>>>> https://github.com/WebAudio/web-audio-api/issues/1498 (outputLatency) >>>>>> https://github.com/WebAudio/web-audio-api/issues/1495 (general >>>>>> security/privacy consideration) >>>>>> >>>>>> I thought the request from this thread is to devise a Chrome-specific >>>>>> mitigation, not reopening the spec discussion. Please correct me if I'm >>>>>> mistaken. >>>>>> >>>>>> Cheers, >>>>>> Hongchan >>>>>> >>>>>> >>>>>> On Thu, Nov 18, 2021 at 12:37 PM Yoav Weiss <[email protected]> >>>>>> wrote: >>>>>> >>>>>>> Hey folks! >>>>>>> >>>>>>> It seems like there are some things to figure out here before >>>>>>> shipping. Could y'all file a spec issue to discuss this with the >>>>>>> broader >>>>>>> community? >>>>>>> >>>>>>> On Tuesday, November 16, 2021 at 6:16:14 PM UTC+1 hongchan wrote: >>>>>>> >>>>>>>> Thanks for the feedback, Mike. This is really helpful! >>>>>>>> >>>>>>>> Mitigation strategies include adding jitter (dithering) and >>>>>>>>> quantization so that the exact skew is incorrectly reported >>>>>>>>> >>>>>>>> >>>>>>>> This bit refers to the randomization mechanism deployed by other >>>>>>>> browsers, but this approach defeats the purpose of this specific API >>>>>>>> and it >>>>>>>> is also argued in the spec: >>>>>>>> >>>>>>>> Note however that most audio systems aim for low latency, to >>>>>>>>> synchronise the audio generated by WebAudio to other audio or video >>>>>>>>> sources >>>>>>>>> or to visual cues (for example in a game, or an audio recording or >>>>>>>>> music >>>>>>>>> making environment). Excessive latency decreases usability and may be >>>>>>>>> an >>>>>>>>> accessibility issue. >>>>>>>> >>>>>>>> >>>>>>>> If the reported value is incorrect, the A/V synchronization won't >>>>>>>> be possible. (Perhaps chcunningham@ can say more about this use case.) >>>>>>>> >>>>>>>> One mitigation idea: we can do some research to collect values >>>>>>>> reported by major platforms/devices and devise a reasonable "bucket >>>>>>>> system" >>>>>>>> for them. >>>>>>>> >>>>>>>> >>>>>>>> On Tue, Nov 16, 2021 at 8:14 AM Mike Taylor <[email protected]> >>>>>>>> wrote: >>>>>>>> >>>>>>>>> I suppose it would be nice to do some testing to verify the idea >>>>>>>>> that there's a small number of platform + audio device pairs. I was >>>>>>>>> hoping Mozilla's >>>>>>>>> hardware survey <https://data.firefox.com/dashboard/hardware> >>>>>>>>> might have something on that, but it doesn't track speakers. >>>>>>>>> >>>>>>>>> Just from some simple testing with the following script (thanks >>>>>>>>> dmcardle@ for the snippet) in Firefox on macOS by 3 people, we have >>>>>>>>> very >>>>>>>>> different values that change dramatically when you unplug a headset: >>>>>>>>> >>>>>>>>> ``` >>>>>>>>> (function getNextSample() { >>>>>>>>> console.log((new AudioContext).outputLatency); >>>>>>>>> setTimeout(getNextSample, 10); >>>>>>>>> })(); >>>>>>>>> ``` >>>>>>>>> Here's the values on my macbook, using Firefox Nightly: >>>>>>>>> bose headset plugged in: 0.005895833 >>>>>>>>> take headset out: 0.013083333 >>>>>>>>> plug bose headset back in: 0.005875 (different, but stable) >>>>>>>>> >>>>>>>>> https://www.w3.org/TR/webaudio/#priv-sec mentions some mitigation >>>>>>>>> strategies - do we have a plan to apply any of them? >>>>>>>>> >>>>>>>>> On 11/15/21 11:53 AM, Ajay Rahatekar wrote: >>>>>>>>> >>>>>>>>> The small number of platform + audio output device combination >>>>>>>>> i.e. a large number of users with the same platform and output >>>>>>>>> device (users on macOS using airpods) would likely be a natural >>>>>>>>> obfuscator. Not sure if we have data about the distribution. >>>>>>>>> >>>>>>>>> On Mon, Nov 15, 2021 at 8:05 AM Mike Taylor <[email protected]> >>>>>>>>> wrote: >>>>>>>>> >>>>>>>>>> On 11/15/21 10:56 AM, Hongchan Choi wrote: >>>>>>>>>> >>>>>>>>>> On Mon, Nov 15, 2021 at 6:49 AM Mike Taylor <[email protected]> >>>>>>>>>> wrote: >>>>>>>>>> >>>>>>>>>>> On 11/15/21 1:47 AM, Yoav Weiss wrote: >>>>>>>>>>> >>>>>>>>>>> On Fri, Nov 12, 2021 at 5:33 PM 'Ajay Rahatekar' via blink-dev < >>>>>>>>>>> [email protected]> wrote: >>>>>>>>>>> >>>>>>>>>>>> >>>>>>>>>>>> TAG review status >>>>>>>>>>>> >>>>>>>>>>>> Completed: Web Audio API specification is W3C Recommendation. >>>>>>>>>>>> >>>>>>>>>>>> Risks >>>>>>>>>>>> >>>>>>>>>>>> There is a risk of the feature being used for fingerprinting. >>>>>>>>>>>> However outputLatency is the buffer size of the >>>>>>>>>>>> platform-provided audio callback, so the value is inherently >>>>>>>>>>>> platform-specific. That said, the majority of the platform audio >>>>>>>>>>>> buffer >>>>>>>>>>>> size is widely known. (MacOS = 128 frames, Windows = 10ms, Android >>>>>>>>>>>> = 96 >>>>>>>>>>>> frames, etc) >>>>>>>>>>>> >>>>>>>>>>>> This feature does not expose more than what you can >>>>>>>>>>>> query/infer from the UA string. >>>>>>>>>>>> >>>>>>>>>>> >>>>>>>>>>> Would that exposure map cleanly to UA-Platform >>>>>>>>>>> <https://wicg.github.io/ua-client-hints/#sec-ch-ua-platform>, >>>>>>>>>>> which is considered low-entropy and exposed by default? Or would it >>>>>>>>>>> add >>>>>>>>>>> more than that? >>>>>>>>>>> >>>>>>>>>>> /cc +Mike Taylor >>>>>>>>>>> >>>>>>>>>>>> Looking at >>>>>>>>>>> https://www.w3.org/TR/webaudio/#dom-audiocontext-outputlatency, >>>>>>>>>>> it states that it depends on the platform _and_ the hardware output >>>>>>>>>>> device. >>>>>>>>>>> If I use an app using outputLatency with speaker A, then switch to >>>>>>>>>>> speaker >>>>>>>>>>> B, will the outputLatency remain the same? >>>>>>>>>>> >>>>>>>>>> >>>>>>>>>> The specification says: If the audio output device is changed the >>>>>>>>>> outputLatency attribute value will be updated accordingly. >>>>>>>>>> >>>>>>>>>> So the answer is no. The value will change accordingly. >>>>>>>>>> >>>>>>>>>> If that's the case, then outputLatency can reveal more entropy >>>>>>>>>> than just platform alone, right? It would be useful to know what >>>>>>>>>> this looks >>>>>>>>>> like in practice, or what mitigations we might be able to apply >>>>>>>>>> depending >>>>>>>>>> on the size of these latency differences. >>>>>>>>>> >>>>>>>>> >>>>>>>>> -- You received this message because you are subscribed to the Google Groups "blink-dev" group. To unsubscribe from this group and stop receiving emails from it, send an email to [email protected]. To view this discussion on the web visit https://groups.google.com/a/chromium.org/d/msgid/blink-dev/745f266b-caf5-4ba1-8429-9c289b4209b0n%40chromium.org.
