Re: Continuous playback of javascript synthesized consecutive audio buffers causes audio artifacts when no buffer starving occurs.

Ah I see. 

We've seen some discussion regarding the modifiability of buffers that have 
been assigned to source nodes (i.e. the "race condition wars"). This jsfiddle
(http://jsfiddle.net/gy5AU/) shows code that modifies a single 22050Hz
buffer periodically to generate a glitch-free (up to setTimeout's jitter) 
440Hz sine tone.

This produces a clean tone on my machine unlike the original beep.html
test case ... but should this work at all?

-Kumar

On 30 Oct, 2013, at 1:38 am, Joseph Berkovitz <joe@noteflight.com> wrote:

> If I'm not mistaken I think the original issue here was sequencing audio buffers whose playback rate was not equal to the AC sample rate, and were thus being up- or down-sampled for playback. In this case one gets aliasing problems at the splice point between successive buffers, and the spec is silent about this sort of thing. These aliasing issues do not arise when the playback rate and the AC sample rate are equal.
> 
> I don't think the buffer queue model below would address this (although it's very useful in its own right).
> 
> On Oct 29, 2013, at 3:18 PM, Srikumar K. S. <srikumarks@gmail.com> wrote:
> 
>>> It would allow for pre-synthesized audio playback in a glitch free manner.
>>> 
>> I'm not sure whether this is to address an implementation bug or a spec shortcoming. The concept of a buffer queue can already be expressed using the AudioBufferSourceNode. Whether it works without glitches in current implementations is likely not a spec shortcoming .. unless it is impossible to create such an implementation, which I don't think is the case.
>> 
>> For instance, see the buffer_queue model at https://github.com/srikumarks/steller/blob/master/src/models/buffer_queue.js. The example code there asks for a function to be called when the queue runs low, but it can sequence buffers passed to the "enqueue" method. 
>> 
>> Reproducing the 440Hz sine tone example here -
>> 
>>    var ac = new AudioContext;
>>    var sh = new org.anclab.steller.Scheduler(ac);
>>    var q = sh.models.buffer_queue();
>>    q.connect(ac.destination);
>>    q.on('low', function () {
>>        var phase = 0.0, dphase = 2.0 * Math.PI * 440.0 / 44100.0;
>>        return function (lowEvent, q) {
>>            var audioBuffer = q.createBuffer(1, 1024);
>>            var chan = audioBuffer.getChannelData(0);
>>            var i;
>>            for (i = 0; i < 1024; ++i) {
>>                chan[i] = 0.2 * Math.sin(phase);
>>                phase += dphase;
>>            }
>>            q.enqueue(audioBuffer);
>>        };
>>    }());
>>    q.start(ac.currentTime);
>> 
>> -Kumar
>> 
>> On 30 Oct, 2013, at 12:25 am, Patrick Martin <patrick.martin.r@gmail.com> wrote:
>> 
>>> It would allow for pre-synthesized audio playback in a glitch free manner.
>>> 
>>> On Oct 20, 2013 10:06 PM, "Robert O'Callahan" <robert@ocallahan.org> wrote:
>>> On Mon, Oct 21, 2013 at 6:35 AM, Srikumar Karaikudi Subramanian <srikumarks@gmail.com> wrote:
>>> What advantage might such an AudioBufferSequenceNode have over a ScriptProcessorNode with a queue processing render function?
>>> 
>>> It would probably have the advantage of not being susceptible to small amounts of main-thread jank.
>>> 
>>> Rob
>>> -- 
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>> 
> 
> .            .       .    .  . ...Joe
> 
> Joe Berkovitz
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