Re: How to play back synthesized 22kHz audio in a glitch-free manner?

If the Web Audio API had an explicit support for buffer queueing/stitching
with AudioBufferSourceNodes, and the user could give that contract to the
Web Audio impl with the 'startImmediatelyAfter' function, then the
implementation could perform audio resampling on the stream as a whole, and
not to discontinuous source nodes individually.

If it was possible to specify the playback frequency in
ScriptProcessorNode, I guess I would be satisfied, although being able to
queue sourcenodes as back-to-back buffers would be a more flexible feature
due to variable-rate decoding and playback opportunities.


2013/6/17 Robert O'Callahan <robert@ocallahan.org>

> On Tue, Jun 18, 2013 at 1:36 AM, Jukka Jylänki <jujjyl@gmail.com> wrote:
>
>> With the start()-scheduled approach, In Firefox nightly, the audio
>> stutters on all sampling rates. In Chrome, audio does not give glitches on
>> 48kHz or 96kHz. This sample
>> https://dl.dropboxusercontent.com/u/40949268/emcc/bugs/webaudio_only_sdl_beep.htmlruns through the various sampling rates to allow you to test as well.
>>
>
> OK. Thanks for the testcase. Obviously we (Firefox) have a general
> scheduling bug. The resampling issue is deeper, however. Perhaps it's not
> surprising that if we resample a fixed-duration tone and then repeat it
> using a series of separate AudioBufferSourceNodes, you don't get a
> continuous tone.
>
> Two approaches here might work:
> 1) specify a sample rate for ScriptProcessorNode and resample its inputs
> and outputs automatically
> 2) allow specifying a sample rate for an entire AudioContext at creation
> time and use either ScriptProcessorNode or AudioBufferSourceNode.
>
> Rob
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