I have a general question about real time audio anylysis with fmod. What would give the lowest latency for real time audio analysis?

From searching the forums I found these 2 solutions:
[list:21awen7u]sound->lock (something like the record to disk example)
create a custom dsp (record and than playback).[/list:u:21awen7u]

I think sound->lock would give lower latencies.

latency for custom dsp: fill record buffer -> fill playback buffer -> fill dsp buffer -> do analysis -> fille playback buffer
latency for sound->lock: fill record buffer -> sound->lock -> do analysis -> fill output buffer.

This would save filling the dsp buffer. Is this correct or am I completely wrong with my calculations?
I’d like to know what the experts think of that before I start coding. Thanks in advance.

Greets mylo

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