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Re: [web-audio-api] (AudioBufferSourceNodeResampling): AudioBufferSourceNode resampling (#118)

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 J├Ągenstedt
> 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?

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Received on Wednesday, 11 September 2013 14:30:29 UTC

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