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

From: Grant Galitz <grantgalitz@gmail.com>
Date: Mon, 11 Jul 2011 02:48:40 -0400
Message-ID: <CAD8zUBaLo60MfM43CwkksL6x51p94bpB5cwOk855D37PJneW7w@mail.gmail.com>
To: Jussi Kalliokoski <jussi.kalliokoski@gmail.com>, public-audio@w3.org
This problem with the clicking an popping from a realtime API that doesn't
allow the developer to push samples ahead of time is seriously affecting
audio testing with the JavaScript GameBoy Advance emulator. It has DMA audio
that needs to be explicitly timed and has resampling and mixing that has to
be done in JS still for the web audio support. :/

On Mon, Jul 11, 2011 at 2:43 AM, Grant Galitz <grantgalitz@gmail.com> wrote:

> 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 06:49:16 UTC

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