Cheap DSP crossover

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dave

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I have decided that I need a crossover for my subs, with equalisation. Brief rummaging has so far found:

Behringer DCX2496LE 2 in to 6 self contained £225
T.racks DSP204 2 into 4 self contained, £225
T.racks DSP 4x4 mini 4 in 4 out. PC interface £95
Mini DSP 2x4HD 2 into 4 PC interface £250
DAP DCP-26 MKII 2 into 6 self contained £225

Any thoughts on these? T.racks don't seem to be used much outside Germany, but its suggested they may sound a bit better (than the Behringer), but it is difficult to tell. The 4x4 is tempting <$, there is also a version with class D amps, but I don't like things that need to plug in to PCs to be useful. I will probably get the Behringer, because I like the self contained and its interface that draws little graphs. I have an older one already, and I know it's not the best sounding, but it should be ok for bass......
 
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tuga

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I have decided that I need a crossover for my subs, with equalisation. Brief rummaging has so far found:

Behringer DCX2496LE 2 in to 6 self contained £225
T.racks DSP204 2 into 4 self contained, £225
T.racks DSP 4x4 mini 4 in 4 out. PC interface £95
Mini DSP 2x4HD 2 into 4 PC interface £250
DAP DCP-26 MKII 2 into 6 self contained £225

Any thoughts on these? T.racks don't seem to be used much outside Germany, but its suggested they may sound a bit better (than the Behringer), but it is difficult to tell. The 4x4 is tempting <$, there is also a version with class D amps, but I don't like things that need to plug in to PCs to be useful. I will probably get the Behringer, because I like the self contained and its interface that draws little graphs. I have an older one already, and I know it's not the best sounding, but it should be ok for bass......

I would only use one that can DSProcess at 48-bit or above and st least 96kHz.
If they all do, perhaps consider whether you would like to use a PC.
If you already do...someone else will have to help.
 
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dave

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I would only use one that can DSProcess at 48-bit or above and st least 96kHz.
If they all do, perhaps consider whether you would like to use a PC.
If you already do...someone else will have to help.
Looks like they are all 32 bits.
 

Shadders

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I would only use one that can DSProcess at 48-bit or above and st least 96kHz.
If they all do, perhaps consider whether you would like to use a PC.
If you already do...someone else will have to help.
Hi,
It does not matter about this if the source is 16bit or 24bit. If the processor is 32bit fixed point, it will have an accumulator of 56bits or greater, so no overflow issues.

DAC's with filters use fixed point processing too, so there is no impact on the sound if 32bit DSP engines are used.

Regards,
Shadders.
 

tuga

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Hi,
It does not matter about this if the source is 16bit or 24bit. If the processor is 32bit fixed point, it will have an accumulator of 56bits or greater, so no overflow issues.

DAC's with filters use fixed point processing too, so there is no impact on the sound if 32bit DSP engines are used.

Regards,
Shadders.

I trust HQPlayer's Jussi Laako view on the subject which is that "even 64-bit floating point is not always enough".
 

Shadders

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I trust HQPlayer's Jussi Laako view on the subject which is that "even 64-bit floating point is not always enough".
Hi,
I did not look at the link.

I do recall you posted a link to a study where digital volume controls can have an effect when truncating - i though i had downloaded the paper - but could not find - do you have that link if you remember ? Thanks.

If the sound is 24bit signed integer, then floating point or fixed point 32bit does not offer much benefit, if the final output is 24bit signed integer based. The error will always be 1LSB at worst for any filter in the DAC, and for 24bit this is not audible. What may be audible is the filter response based on its topology, FIR, low latency (small number of taps), stop band attenuation etc.

Regards,
Shadders.
 

tuga

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Hi,
I did not look at the link.

I do recall you posted a link to a study where digital volume controls can have an effect when truncating - i though i had downloaded the paper - but could not find - do you have that link if you remember ? Thanks.

If the sound is 24bit signed integer, then floating point or fixed point 32bit does not offer much benefit, if the final output is 24bit signed integer based. The error will always be 1LSB at worst for any filter in the DAC, and for 24bit this is not audible. What may be audible is the filter response based on its topology, FIR, low latency (small number of taps), stop band attenuation etc.

Regards,
Shadders.

Hi Shadders, this is the link to the piece you are looking for:


Like I said, I trust Jussi Laako who is communications software engineer (he worked for Nokia and Intel) and has been developping audio upsampling/DSP software for a couple of decades. I have read many of his comments and seen his measurements.
 
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Shadders

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Hi Shadders, this is the link to the piece you are looking for:


Like I said, I trust Jussi Laako who is communications software engineer (he worked for Nokia and Intel) and has been developping audio upsampling/DSP software for a couple of decades. I have read many of his comments and seen his measurements.
Hi Tuga,
Thanks - much appreciated.

I did finally read the comments - effectively Miska is using DSP for audio processing with multiple cascaded operations and mixing where effects are added and wide ranges of gain. So his analysis is essentially the more bits the better.

With Dave's requirement for a crossover, then depending how you implement it, the worst case is attenuation of the incoming data (16bit/24bit) by a few dB to ensure that at the crossover point, should the filters combine to add gain, that gain does not cause data overflow. Essentially a crossover using Linkwitz Riley filters correctly configured, will be 0dB gain across the entire audio band, so no overflow, and only attenuation based on the filter rolloffs.

Regards,
Shadders.
 
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dave

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With Dave's requirement for a crossover, then depending how you implement it, the worst case is attenuation of the incoming data (16bit/24bit) by a few dB to ensure that at the crossover point, should the filters combine to add gain, that gain does not cause data overflow. Essentially a crossover using Linkwitz Riley filters correctly configured, will be 0dB gain across the entire audio band, so no overflow, and only attenuation based on the filter rolloffs.
Thanks guys, I am doing crossover and modest EQ for 4 subs, so most of the bits will be thrown away :). I have ordered a DCX2496LE, should turn up today.
 

tuga

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Ric
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Thanks guys, I am doing crossover and modest EQ for 4 subs, so most of the bits will be thrown away :). I have ordered a DCX2496LE, should turn up today.
Ok, I thought that would be high-passing the main speakers as well. In that case I think that you will be fine.
 

dave

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Ok, I thought that would be high-passing the main speakers as well. In that case I think that you will be fine.
No I am much more low tech than that, I have some 10n polystyrene capacitors in series to the input of the mid / tweeter amp.
 
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