- 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 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? -- 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