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Re: Reviewing the Web Audio API (from webrtc)

From: Chris Rogers <crogers@google.com>
Date: Wed, 28 Mar 2012 22:32:40 -0700
Message-ID: <CA+EzO0=ZMsyv=RzBGMzSu+UhKaHvs=ZYd_kidTONPT4mqbCLgg@mail.gmail.com>
To: robert@ocallahan.org
Cc: "Wei, James" <james.wei@intel.com>, "public-audio@w3.org" <public-audio@w3.org>, public-webrtc@w3.org
On Wed, Mar 28, 2012 at 10:10 PM, Robert O'Callahan <robert@ocallahan.org>wrote:

> On Thu, Mar 29, 2012 at 3:33 PM, Chris Rogers <crogers@google.com> wrote:
>> HTMLMediaElements already have a mechanism for synchronization using the
>> HTML5 MediaController API.  Live stream (live camera/mic or remote peers)
>> MediaStreams would maintain synchronization (I assume you mean audio/video
>> sync in this case).  The Web Audio API would just be used to apply effects,
>> not changing the synchronization.
> Can you explain how audio/video sync would account for the latency
> introduced by Web Audio processing? Have you found a way to do this
> automatically?

None of the built-in Web Audio processing algorithms have any appreciable
latency which would perceptibly affect audio/video sync.  We're talking
about 3ms or less here.  In terms of irritation, network latency is of
vastly more concern for WebRTC applications.

Received on Thursday, 29 March 2012 05:33:12 UTC

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