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

Isn't that already true in the presence of sources like <audio> or <video>,
or WebRTC streams, where pausing can occur?

Furthermore, isn't this true for AudioBufferSourceNode once playbackRate is
applied?

If 'all nodes operate on the same timescale' is an intended design rule, I
think that's reasonable - maybe even a very valuable rule. But it doesn't
seem to be one right now from my layman's reading of the spec.

-kg


On Mon, Jun 17, 2013 at 3:29 PM, Robert O'Callahan <robert@ocallahan.org>wrote:

> On Tue, Jun 18, 2013 at 10:15 AM, Kevin Gadd <kevin.gadd@gmail.com> wrote:
>
>> Could one simply define a ResamplerNode/PlaybackRateAdjustmentNode? Then,
>> in cases where you want to stitch together smaller buffers and adjust the
>> playback rate of all of them, you give them all the resampler node as a
>> shared destination.
>>
>
> Correct me if I'm wrong, but doesn't that mean different nodes operate on
> different timescales? It would be better to avoid that if we can.
>
> Rob
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Received on Monday, 17 June 2013 22:34:31 UTC