W3C home > Mailing lists > Public > public-audio@w3.org > October to December 2012

[Bug 17377] (AudioBufferSourceNodeResampling): AudioBufferSourceNode resampling

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
----------------------------------------------------------------------------
             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 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?

-- 
You are receiving this mail because:
You are on the CC list for the bug.
Received on Tuesday, 16 October 2012 20:27:53 UTC

This archive was generated by hypermail 2.3.1 : Tuesday, 6 January 2015 21:50:03 UTC