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Multi-Way Conventional loudspeakers with crossovers

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Old 24th October 2012, 11:32 PM   #511
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Quote:
Originally Posted by flavio81 View Post
Well, because the analog part can be more expensive than the digital part, if it is to be realized correctly.
Yes, I think that is mostly the reason.

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Originally Posted by StigErik View Post
At 44100 Hz sample rate, 1 sample delay equals 7,7 mm. With high XO frequencies that might start to make a difference....
I have run into problems crossing circa 7KHz. At a 96KHz sampling rate, it's OK.
Moving the driver is best, if possible.
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Old 24th October 2012, 11:40 PM   #512
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As mentioned earlier this is quite a simplistic view of digital audio. By re-sampling, you can have any delay you like. Presumably a system that corrects automatically by applying the inverse impulse response or whatever, does this implicitly.
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Old 24th October 2012, 11:41 PM   #513
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Why extremes? Proper combinations of mechanical, analog and digital is the way to go.
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Old 25th October 2012, 12:05 AM   #514
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By re-sampling, you can have any delay you like.
Yes, of course. That's where the 96K I mentioned came from.
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Old 25th October 2012, 12:42 AM   #515
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Yes, of course. That's where the 96K I mentioned came from.
What I mean is you don't need a higher sample rate as such; merely (correctly) interpolate between your existing samples and you can still play back at 44.1 kHz with almost infinitesimally fine graduations of delay from the original.
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Old 25th October 2012, 01:14 AM   #516
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I would imagine that one can also do that with an all-pass filter to a certain extent using an IIR -bi-quad. As it stands the digital delay function block built into the analogue devices software for the DSP chip that I'm using doesn't have any delay option other then delaying in discrete steps with the smallest being 1 sample period. I get the impression that interpolating and resampling would require considerably more processing power then the simple delay that it provides. Of course the DSP chip does have ASRCs already built in, but they aren't able to work in the way described. I can use FIR filters with the chip but I know almost nothing about calculating their parameters.
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Old 25th October 2012, 01:36 AM   #517
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What delay steps are used by Renkus-Heinz in their steerable arrays, who knows?
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Old 25th October 2012, 01:54 AM   #518
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Quote:
Originally Posted by Charles Darwin View Post
May be one could use the oversampling mechanism to get shorter delays without going to massively higher SRs.

The XTA DP4xx series of speaker management systems for example uses a 96k SR but the delay is adjustable in 0.325µsec steps.
Unless there is another way of achieving this that I don't know anything about in my ignorance.
Delays in digital aren't restricted to integer multiples of the SR. Check out 'Fractional Delay Filters' if you'd like to learn more.
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Old 26th October 2012, 12:21 AM   #519
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What about the idea of scale-ability?

It is quite possible to get great results with an active system with nothing more than individual woofers, mids and tweets.

You won't find many stadium size speaker arrays that are all driven by a single amp!

James.
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Old 26th October 2012, 02:14 AM   #520
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It is not a problem with stadium size arrays. PA systems for small venues and especially stage monitors would benefit from DSP equalization, if not latency problem. It would be great to have stage monitors with absolutely flat frequency response and almost zero distortions.
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