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Re: Reviewing the Web Audio API (from webrtc)

From: Robert O'Callahan <robert@ocallahan.org>
Date: Thu, 29 Mar 2012 21:58:16 +1300
Message-ID: <CAOp6jLa4UShFqAeOqWHFj8w0VfMWOoMGM5kstmUDQqJEMGa+OA@mail.gmail.com>
To: Chris Rogers <crogers@google.com>
Cc: "Wei, James" <james.wei@intel.com>, "public-audio@w3.org" <public-audio@w3.org>, public-webrtc@w3.org
On Thu, Mar 29, 2012 at 6:32 PM, Chris Rogers <crogers@google.com> wrote:

> None of the built-in Web Audio processing algorithms have any appreciable
> latency which would perceptibly affect audio/video sync.


OK, but there are processing algorithms that necessarily have significant
latency, like this one:
http://people.mozilla.org/~roc/stream-demos/video-with-extra-track-and-effect.html

We're talking about 3ms or less here.  In terms of irritation, network
> latency is of vastly more concern for WebRTC applications.


That depends on the application. WebRTC APIs can be used for more than just
interactive chat. For example, an application could pull an audio and video
stream from some source, take a user's commentary in a stream from the
microphone, mix them with a ducking effect, and stream the resulting audio
and video out to a set of peers. The latency might be too high for
interaction, but just fine for a "live broadcast".

Rob
-- 
“You have heard that it was said, ‘Love your neighbor and hate your enemy.’
But I tell you, love your enemies and pray for those who persecute you,
that you may be children of your Father in heaven. ... If you love those
who love you, what reward will you get? Are not even the tax collectors
doing that? And if you greet only your own people, what are you doing more
than others?" [Matthew 5:43-47]
Received on Thursday, 29 March 2012 08:58:46 GMT

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