Re: [linux-audio-dev] analog latency

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Subject: Re: [linux-audio-dev] analog latency
From: Iain Sandoe (iain_AT_sandoe.co.uk)
Date: Sun Jun 11 2000 - 09:46:46 EEST


>Benno Senoner wrote:
> On Sun, 11 Jun 2000, Paul Barton-Davis wrote:
>> just a quick little FYI:
>>
>> i was reading the "manual" that came with some new Tannoy monitors I
>> got a few weeks ago. there is a comment in the back to the effect
>> that:
>>
>> * the filters on almost every analog EQ have a delay time of
>> 25usec
>> * with only a few exceptions, all equalizers run the filters
>> in series.
>>
>> So much for the naive belief that analog systems have no latency.
>>
>> OK, not much :)
>>
>> --p
>
> "in series ..." = In practice ?
>
> Does that mean a 16band EQ = 16 x 25usec = 400usec = 0.4msec ?

It seems a quite unlikely that each band would have exactly the same
'delay'.

I haven't seen the original 'manual' that Paul refers to, but presumably
they are talking about a frequency dependent phase shift in the analogue
filters... and equating the linear portion of dPhi/dOmega to a time delay.

If you replicate the same filter as a 'digital' version - you will get the
same 'delay' - *in addition* to any acquisition/replay delays.

If you 'do it better digitally' (e.g. FIR or FFT-based) your implementation
delays may well exceed those. Again, these appear *serially* with the
acquisition & replay delays.

There may be some algorithmic structure that allows you to do some trick
that is more efficient that can be achieved economically (or with low noise)
in the analogue domain... but I don't think you can make a useful filter
without some *real* delay - causality won't allow it.

Iain.


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