Audio Unit Latency
Audio Unit Latency
- Subject: Audio Unit Latency
- From: Brian Davies <email@hidden>
- Date: Thu, 31 May 2012 21:55:06 +1000
Am new to core audio but not to DSP.
Am testing an Audio Effect Unit which has latency in the DSP. Subclassed AUEffectBase, implemented GetLatency(), GetTailTIme(), and SupportsTail(). First two return the latency, SupportsTail returns true.
In one audio editor it seems to work as expected (yet to do extensive testing), in another the tail bytes are ignored and the file shortened (so the leading zeros are correctly discarded). Have done some checking on method calls by the hosts, and find that BOTH call GetLatency() and GetTailTIme() (repeatedly) but NEITHER ever calls SupportsTail(). Am I missing something in the documentation? It stresses the need to implement GetTailTIme (). Is there anything else I need to do to allow DSP which has simple latency?
My interpretation is that if N samples starting at M need to be processed, and the output is L samples behind the input, then the host should send in N+L samples starting at M (possibly zero padded for the last L) and discard the first L output samples, effectively time-shifting the N output samples back to the correct time.
Are my assumptions correct? Surely there are many working effect units with latency?
Brian Davies
email@hidden
www.kagi.com/clickrepair
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