- From: Olivier Thereaux <notifications@github.com>
- Date: Wed, 11 Sep 2013 07:29:41 -0700
- To: WebAudio/web-audio-api <web-audio-api@noreply.github.com>
- Message-ID: <WebAudio/web-audio-api/issues/118/24244306@github.com>
> [Original comment](https://www.w3.org/Bugs/Public/show_bug.cgi?id=17377#0) by Chris Rogers on W3C Bugzilla. Tue, 16 Oct 2012 20:27:50 GMT (In reply to comment #0) > Audio-ISSUE-92 (AudioBufferSourceNodeResampling): AudioBufferSourceNode > resampling [Web Audio API] > > http://www.w3.org/2011/audio/track/issues/92 > > Raised by: Philip Jägenstedt > On product: Web Audio API > > AudioBufferSourceNode has a playbackRate attribute which will require > interpolation/resampling of some kind. However, it is not defined how to > resample. Possibly it should be as close as possible to an ideal resampling, > in which case that should be stated. Alternatively, it could be possible to > specify which kind of resampling to perform via an attribute: nearest, > linear, cubic, sinc, etc... > > It also needs to be defined what should be done about folding when the net > result of the sample rates and playback rate is a downsampling, if anything. I agree that a .resamplingType attribute could be defined, but holding off on that for now, since it's something which can later be added. In the mean-time, I think we should specify that the default is "linear". Does that seem ok? --- Reply to this email directly or view it on GitHub: https://github.com/WebAudio/web-audio-api/issues/118#issuecomment-24244306
Received on Wednesday, 11 September 2013 14:30:29 UTC