- From: Olivier Thereaux <notifications@github.com>
- Date: Wed, 11 Sep 2013 07:29:50 -0700
- To: WebAudio/web-audio-api <web-audio-api@noreply.github.com>
- Message-ID: <WebAudio/web-audio-api/issues/131/24244407@github.com>
> [Original comment](https://www.w3.org/Bugs/Public/show_bug.cgi?id=17335#4) by redman on W3C Bugzilla. Wed, 05 Dec 2012 15:48:51 GMT (In reply to [comment #3](#issuecomment-24244398)) > (In reply to [comment #2](#issuecomment-24244392)) > > I think the level of detail in the new text is good. One thing that seems to > > be missing though is what the value is when t < time and t >= time + > > duration, respectively (most likely values[0] and values[N-1], respectively). > > > > Also, the expression "v(t) = values[N * (t - time) / duration]" is > > effectively nearest interpolation. Is that intended? Linear interpolation > > seems more logical. > > The idea is that the number of points in the Float32Array can be large so > that the curve data is effectively over-sampled and linear interpolation is > not necessary. > > Fixed: > https://dvcs.w3.org/hg/audio/rev/a658660f3174 That idea assumes the user creates a 'curve' that is itself sufficiently oversampled (has way too much data than needed). If you want to do it right you should provide an interpolator for undersampled cases and a bandlimiting filter for oversampled cases. In other words, you should do proper resampling with audio-rate parameters. Only if the data is played back at its original samplerate can you assume the data is a valid sample. Since it is an audio rate controller you should always see it as a signal and apply signal theory. --- Reply to this email directly or view it on GitHub: https://github.com/WebAudio/web-audio-api/issues/131#issuecomment-24244407
Received on Wednesday, 11 September 2013 14:35:24 UTC