Can users determine if audio has glitched?

Is there a way under the current API for users to determine if audio has glitched (in particular, because their ScriptProcessorNode was too slow)?

This would be very useful, because then programs could increase buffer size or decrease audio quality based on whether or not the script processor was completing on time.

The only way I can think to do this currently would be to inspect the playbackTime attribute on the AudioProcessingEvent and make sure that's it's increasing by exactly one buffer each call.  This seems a bit fragile to me, as perhaps there's other reasons why the web audio graph might decide to skip a buffer of script processing.


Received on Monday, 10 June 2013 22:15:07 UTC