- From: <bugzilla@jessica.w3.org>
- Date: Tue, 16 Oct 2012 20:27:50 +0000
- To: public-audio@w3.org
- Message-ID: <bug-17377-5429-I7g89FSyGl@http.www.w3.org/Bugs/Public/>
https://www.w3.org/Bugs/Public/show_bug.cgi?id=17377
Chris Rogers <crogers@google.com> changed:
What |Removed |Added
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Status|NEW |ASSIGNED
CC| |crogers@google.com
--- Comment #1 from Chris Rogers <crogers@google.com> ---
(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?
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Received on Tuesday, 16 October 2012 20:27:53 UTC