- From: Brad Isbell via GitHub <sysbot+gh@w3.org>
- Date: Tue, 19 Sep 2023 21:42:58 +0000
- To: public-webrtc-logs@w3.org
`sampleRate` is required in a lot of my usage of the API, where collaborators on a common broadcast stream need to match sample rate. For example, one radio show host may be streaming at 48 kHz. The next radio show host for hour #2 may have a default sample rate of 44.1 kHz, but we can get a 48 kHz stream for them by requesting it with constraints. (Of course this is sometimes transparently resampled, but without it we're forced to resample in JS which isn't efficient and adds extra latency.) And, a link for reference... fingerprinting concerns around sample rate were previously discussed over on the Web Audio API: https://github.com/WebAudio/web-audio-api/issues/1457 Regarding `sampleSize`, I'd agree completely with @padenot about problems with the current API. We have actually been trying to move to float32 audio interfaces, but there are compatibility issues from the OS, to the audio API the user agent uses, all the way through the user agent. Even once we get those issues ironed out, there's no way to know if it was 32-bit integer samples, or one of the float32 implementations. -- GitHub Notification of comment by bradisbell Please view or discuss this issue at https://github.com/w3c/mediacapture-main/issues/958#issuecomment-1726541903 using your GitHub account -- Sent via github-notify-ml as configured in https://github.com/w3c/github-notify-ml-config
Received on Tuesday, 19 September 2023 21:43:00 UTC