Re: [webrtc-stats] Stats API should require additional permission / user opt-in (#550)

1)
- What granularity is needed to be able to tell anything more useful than "there is or isn't audio being produced right now"? getStats() gives you aggregate counters, so the best you can do is to say that in an interval between two getStats() calls your average packet size was X bytes and the average audio energy was Y. In Chrome, the minimum interval you could achieve is 50 ms due to caching. Would mandating a caching time mitigate the problem?
- Unless we're talking about isolated streams, the RTCPeerConnection can only process tracks you already have access to. You can use WebAudio or you can read pixels of a canvas or other APIs, not to mention you're sending the tracks somewhere, so the other endpoint can do whatever it wants (including communicating back with the JS to tell it whatever the result of its analysis). So we there's already, directly or indirectly, access on a byte level.

Re: @fippo: getSynchronizationSources() can poll much more frequently. It will tell you audioLevel of packets (only received packets but you could do a loopback if you could indirectly say something about sent packets too). But again, why not use WebAudio?

In either case, I don't mean to make the argument "there is another API that is even worse" as an excuse for us to do something bad. My question is: Is this really a problem with these APIs or is this an objection to having granted access to tracks in the first place, which is usable with a large number of APIs?

2)

Codec capabilities and encoder/decoder implementation strings are a valid fingerprinting concern.

What would be a way to mitigate these concerns? Adding a prompt on a per-API basis fails to address how confusing a "do you want to grant access to getStats?" would be to a normal user. Would hardware and media related privacy concerns be best addressed with a prompt of larger scope?

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Received on Wednesday, 19 February 2020 08:44:54 UTC