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Fwd: Concerning the gap-less output of real-time generated audio in JavaScript

From: Grant Galitz <grantgalitz@gmail.com>
Date: Mon, 11 Jul 2011 02:43:44 -0400
Message-ID: <CAD8zUBYYq21oNxPgkDcDOootuu7++Lj5FfyQEhyBAqtEnExgMg@mail.gmail.com>
To: public-audio@w3.org
---------- Forwarded message ----------
From: Grant Galitz <grantgalitz@gmail.com>
Date: Mon, Jul 11, 2011 at 2:43 AM
Subject: Re: Concerning the gap-less output of real-time generated audio in
To: Jussi Kalliokoski <jussi.kalliokoski@gmail.com>

I agree it should be a mixed callback/write based API that allows the
developer to provide samples ahead of time. I do this with the exposed APIs
for my XAudioJS lib for thin wrapping mozAudio, web audio, a flash fallback,
and wav pcm data uri generation together.

On Mon, Jul 11, 2011 at 2:38 AM, Jussi Kalliokoski <
jussi.kalliokoski@gmail.com> wrote:

> Hello all, I'll jump in on this.
> On Mon, Jul 11, 2011 at 9:11 AM, Grant Galitz <grantgalitz@gmail.com>wrote:
>> I'll briefly compare the mozilla audio data api and the web audio api and
>> run through this list of what can be improved upon in web audio.
>> - Web Audio does not allow resampling, this is a major thorn in probably a
>> couple people's butts, because I have to do this in JavaScript manually. If
>> there is a security concern for bottlenecking, then I'd assume we could
>> throw in some implementation-side limitations on the number of concurrent
>> supposed resampling nodes that could be run at the same time.
> It most certainly is! However I disagree that there should be a resampling
> node, this is a simple matter and has a simple solution employed in most if
> not all the client side audio APIs: being able to select the sample rate.
> And you're right, I don't think it's going to raise much respect amongst
> existing audio devs if you can't even choose the sample rate for yourself.
> But I believe Chris knows this already.
>> - Web Audio forces the JavaScript developer to maintain an audio buffer in
>> JavaScript. This applies for audio that cannot be timed to the web audio
>> callback, such as an app timed by setInterval that has to produce x samples
>> every x milliseconds. The Mozilla Audio Data API allows the JS developer to
>> push samples to the browser and let the browser manage the buffer on its
>> own. The callback grabbing x number of samples every call is not a buffer on
>> its own, that's the callback sampling the whole buffer of what I'm talking
>> about. Buffer ring management in JavaScript takes up some CPU load and it
>> would always be better in my opinion to let the browser manage such a task.
> This is a good point as well. But IMO a more useful approach would be to
> have the callback API and then an alternate write call that mixes the
> written buffers into the buffers provided by the callback (if provided) and
> another one that writes ahead of time, pushing away callbacks. And please,
> don't make the developer handle the tail, like in mozAudio. Something like
> node.write(buffer, channelCount = 2, sampleRate = [context default]) and
> node.writeAhead( -||- );
>> - "The callback method knows how often to fire," this is a fallacy, even
>> flash falls for this issue and can produce clicks and pops on real-time
>> generated audio (Even their docs hint at this). This is because by the time
>> the callback API figures out a delay, its buffering may be premature due to
>> previous calculations and may as a result gap the audio. It is imperative
>> you let the developer control the buffering process, since only the
>> developer would truly know how much buffering is needed. Web Audio in chrome
>> gaps out for instance when we're drawing to a canvas stretched to fullscreen
>> and a canvas op takes a few milliseconds to perform, to a reasonable person
>> this would seem inappropriate. This ties in basically with the previous
>> point of letting the browser manage the buffer passed to it, and allowing
>> the JS developer to buffer ahead of time rather than having a real-time
>> thread try to play catch-up with an inherently bad plan.
> You're right, if the callback blocks for longer than the buffer length gets
> played, pops and cracks are inevitable, but... That's the case of digital
> real time audio, no matter what platform. Having a write-only API in this
> case is not an option.
>> - Building up on the last point, in order to achieve ahead-of-time
>> buffering, I believe it would be wise to either introduce a stub function
>> that allows samples to be added at any time without waiting for a callback,
>> just like mozWriteAudio, OR to allow the callback method to be called when
>> buffering reaches a specified low point *specified* by the developer. This
>> low point is not how many samples are to be sent to the browser each
>> callback, but lets the API know WHEN to fire the callback, with the firing
>> being at a certain number of samples before buffer empty.
>> I hope we can use some or all of these points listed in providing a proper
>> API for real-time generated audio output in JavaScript in a 21st century
>> browser. :D
> ;)
> Jussi
Received on Monday, 11 July 2011 06:44:12 UTC

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