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Re: [web-audio-api] OfflineAudioContext and ScriptProcessorNodes (#69)

From: Olivier Thereaux <notifications@github.com>
Date: Wed, 11 Sep 2013 07:29:39 -0700
To: WebAudio/web-audio-api <web-audio-api@noreply.github.com>
Message-ID: <WebAudio/web-audio-api/issues/69/24244251@github.com>
> [Original comment](https://www.w3.org/Bugs/Public/show_bug.cgi?id=22723#17) by Ehsan Akhgari [:ehsan] on W3C Bugzilla. Mon, 22 Jul 2013 17:38:57 GMT

(In reply to [comment #17](#issuecomment-24244243))
> (In reply to [comment #16](#issuecomment-24244239))
> > (In reply to [comment #15](#issuecomment-24244234))
> > > (In reply to [comment #7](#issuecomment-24244176))
> > > > If every node used this "wait for all inputs before running" logic, then
> > > > script nodes with buffer sizes greater than 128 need not impose a delay in
> > > > their signal paths. 
> > > 
> > > I just realized a subtlety in this. If a script processor node's
> > > onaudioprocess reads computed values from AudioParams, then the perceived
> > > k-rate for those AudioParams will be determined by the block size set for
> > > the script node and not the fixed 128-sample-block in the spec. Not only
> > > that, it will look like a filter-type script node (with input and output) is
> > > prescient and anticipates animated AudioParams, because the the
> > > onaudioprocess will only get to run once enough input chunks have
> > > accumulated, meaning the values of some of these k-rate AudioParams could
> > > already have advanced to a time corresponding to the end of the script
> > > node's buffer duration.
> > 
> > No, according to the spec the implementation must do 128-frame block
> > processing all the time, which means that for example if we have 1024 frames
> > to fill up for a ScriptProcessorNode, we need to call the block processing
> > code 8 times, and each k-rate AudioParam will be sampled at the beginning of
> > each block.
> 
> That holds only for the native nodes, doesn't it?

No, that's true for all nodes.

> With the real-time
> context, script processor nodes with buffer sizes > 128 (which is all the
> time) already have a lower k-rate than the native nodes if they read
> computed values of AudioParams within their onaudioprocess callbacks.

I'm not sure what you mean here.  How do you "sample" the AudioParam value inside the audioprocess event handler?

> Anyway, to ensure that the k-rate is uniform at least during offline
> processing, it looks like the only way is to raise onaudioprocess events for
> each 128-sample-frame block. The event dispatcher better put up some
> performance :)

Doing that violates the current spec, and I think would be a very bad idea.

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Received on Wednesday, 11 September 2013 14:34:26 UTC

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