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 UTC