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

From: Robert O'Callahan <robert@ocallahan.org>
Date: Tue, 18 Jun 2013 11:23:10 +1200
Message-ID: <CAOp6jLagySTPJU5UfbtSUOszyRvtg+LpSfxZK5Tp48bGY63YxA@mail.gmail.com>
To: Kevin Gadd <kevin.gadd@gmail.com>
Cc: Jukka Jylänki <jujjyl@gmail.com>, "public-audio@w3.org" <public-audio@w3.org>
On Tue, Jun 18, 2013 at 10:33 AM, Kevin Gadd <kevin.gadd@gmail.com> wrote:

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

No, all Web Audio time values refer to a timeline where paused streams have
been padded with silence.

I actually agree that at some point we should provide pausing as a Web
Audio feature and allow node timelines to diverge. I just don't think this
feature justifies that.

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

I guess that's true for some of the AudioBufferSourceNode attributes. But
that's a very limited case.

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Received on Monday, 17 June 2013 23:23:37 UTC

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