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

@Joseph
Yes, that's true but unfortunately it's not specified how it's sample
accurate, playback time is specified in seconds, which are independant of
the number of samples, calling  performance.now()*1000 +
samples_in_last_buffer/samples_per_second is probably the most reliable way
to schedule the audio, but the javascript can get interrupted between
calling the time function and scheduling the audio causing cracks pops etc.
It's simpler to both implement and use if we either have a way to schedule
based on a delay in the format of number of samples played or a way to
queue buffers in a AudioBufferQueueNode or similar.


On Wed, Oct 30, 2013 at 6:29 AM, Joseph Berkovitz <joe@noteflight.com>wrote:

> I don't think the approach of this jsfiddle is kosher for two reasons,
> which are related.
>
> 1. Modifying a looped buffer that is playing in the main thread is
> necessarily going to be subject to jitter, even if there are no races per
> se.
> 2. Calling start(AC.currentTime) is inherently unreliable because you
> provide no time for the audio thread to schedule playback in advance.
>
> And it doesn't produce a clean tone on my machine :)
>
> @Patrick, if you schedule your buffers a safe amount ahead of the current
> time, and they are not being up/down sampled, you should get perfectly
> sample-accurate glitch-free playback in a compliant implementation.
>
>
> On Tue, Oct 29, 2013 at 2:55 PM, Srikumar K. S. <srikumarks@gmail.com>wrote:
>
>> 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*
>> President
>>
>> *Noteflight LLC*
>> Boston, Mass.
>> phone: +1 978 314 6271
>>        www.noteflight.com
>> "Your music, everywhere"
>>
>>
>>
>
>
> .            .       .    .  . ...Joe
>
> *Joe Berkovitz*
> President
>
> *Noteflight LLC*
> Boston, Mass.
> phone: +1 978 314 6271
> www.noteflight.com
> "Your music, everywhere"
>
>