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

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
>> -- 
>> Jtehsauts  tshaei dS,o n" Wohfy  Mdaon  yhoaus  eanuttehrotraiitny  eovni le atrhtohu gthot sf oirng iyvoeu rs ihnesa.r"t sS?o  Whhei csha iids  teoa stiheer :p atroa lsyazye,d  'mYaonu,r  "sGients  uapr,e  tfaokreg iyvoeunr, 'm aotr  atnod  sgaoy ,h o'mGee.t"  uTph eann dt hwea lmka'n?  gBoutt  uIp  waanndt  wyeonut  thoo mken.o w  
> 

.            .       .    .  . ...Joe

Joe Berkovitz
President

Noteflight LLC
Boston, Mass.
phone: +1 978 314 6271
www.noteflight.com
"Your music, everywhere"