Fully digital crossover

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I'm looking for a digital crossover that doesn't include AD and DA conversion and works up to 192kHz with 24bit.

What I want to achive is to connect the digital out on a soundcard to an external XO unit and get up to 4 filtered digital signals per channel that I can connect to separate dac's.

The optimal would be if it could be programed from a PC.

If anybody know of such an XO or feel capable to accept the challenge to construct one that can fit for DIY please let me know.
 
Hi Bitstrom,

take a look at this:
www.acourate.com

I am still checking it out, but at least it works!
All you need is a PC, their Linux on a stick and a soundcard.
I use a M-Audio 1010LT, so I have digital Input, but 8 analog outputs.

You could go and utilize the I2C inputs of the DACs, or prefereably a soundcard with digital (SP/DIF) outs (RME Hammerfall, for instance).

Best wishes
Thomas
 
Bitstrom said:
I'm looking for a digital crossover that doesn't include AD and DA conversion and works up to 192kHz with 24bit.

What I want to achive is to connect the digital out on a soundcard to an external XO unit and get up to 4 filtered digital signals per channel that I can connect to separate dac's.

The optimal would be if it could be programed from a PC.

If anybody know of such an XO or feel capable to accept the challenge to construct one that can fit for DIY please let me know.


I just bought a x-over, Behringer, DCX2496, you can download x-over data from PC via s/w Soundeasy. this is a desk top unit. not pc mounted.

M-Audio has a few models, which is mounted on the PC I think you can download x-over data from the PC. M-Audio is proven to be very good product and used by many audio enthusiasts.
 
Thank you all

My intention was to find a solotion that is not using a multichannel soundcard.

Except from the 192 kHz ability the dcx2496 tweaked with digital outputs is what comes closest to what I want to achive. But I read somewhere that the behringer emulates analog filters and therefore introduces phase shift into the signal.

But anyhow, thank you guys for trying to help.

Maybe I will eventually go for a dedicated pc with multiple soundcards since it seems to be a relatively cheap solution (depending on the software prices) for a to get a really high performing system.
 
Re: Thank you all

Bitstrom said:
My intention was to find a solotion that is not using a multichannel soundcard.


Why? If you're planning on using the PC as a source anyway, it seems like a natural avenue to explore.


Except from the 192 kHz ability the dcx2496 tweaked with digital outputs is what comes closest to what I want to achive. But I read somewhere that the behringer emulates analog filters and therefore introduces phase shift into the signal.

It sounds like the DEQX PDC is close to what you're looking for, although it's 3-way only and probably limited to 96kHz. It has digital inputs and optional digital outputs, and isn't limited to strictly IIR filters. Not cheap, though.



Maybe I will eventually go for a dedicated pc with multiple soundcards since it seems to be a relatively cheap solution (depending on the software prices) for a to get a really high performing system.

You want to avoid a multi-channel card, but would consider using multiple 2-channel cards? The short answer is "don't bother". Achieving sync across multiple cards is a pointless exercise in frustration given that very good multichannel cards are available for very modest prices.

Since what you're looking for doesn't exist, and you seem to find the 'obvious' approach unattractive, you might look at peufeu's ethernet-to-i2s-via-fpga project. He was away for a while and there wasn't much progress on it, but if/when he finishes it it'll be pretty close to what you're looking for.
 
I had a similar set of requirements which I have found a solution for (still testing, but seems to work).

I wanted to have 2 zones of audio from WMP11 with correction filters applied to each zone. I use DRC to generate correction filters for each zone (i.e. 2 rooms being corercted for).

I have the following setup:

2 x m-audio transit USB -> 2 x DAC-AH -> 2 x Power amps

Then, I use Convolver to apply the filters and to create 4 output channels from 2 input channels (you could apply your XO here).

To connect the 2 devices into a single virtual device, i use Virtual Audio Cable as the sound device in WMP, under ASIO4ALL and use VSTHost to host the ConvolverVST plugin. Then, in VSTHost, i set the device to ASIO4ALL and the input to the 2xVAC channels and the output to the 2x2 m-audio channels (4 channels out). i see no reason why you could not connect any number of output devices in this way. regarding drift, i have not noticed any, but I am feeding to 2 different rooms so is hard to tell. I believe VSTHost has the ability to apply delays to specific channels. I wonder if the fact that I have 2 identical USB audio devices connected to the same powered USB hub alleviates. I'm still experimenting, but wanted to share.
 
The DEQX PDC looks like a really nice solution, except the price :bawling:. So I guess I will have to wait for a cheaper competitor to appear since this is the evident future of audio reproduction.

About the syncronisation of more then one soundcard I think it is possible to solve with delay settings.

I think "hjb" is having the solution for the time beeing. I have been looking at a similar setup but hoped to find a simpler or at least more integrated way to go. Winamp with some plugins will do the trick, but at the cost of a rather advanced setup.

The sync problem could be an real issue though if it comes to a 2,5 way speaker were I guess it can be important to have the woofers play very close in time.

For now I will go to pray that some big brands to get into this market with sinking prices as a result.
 
Hi Bit,
you've asked for 192ks/S.
What sources have you found that can make use of that?

Some designers are looking at upgrading the DCX2496 to 24192.
This may take a while before they publish.
If you're patient that mod could be forthcoming just as the market starts to think about providing a higher sampling rate.

What is the bit rate in the audio channels of HDTV?
 
hjb said:
I had a similar set of requirements which I have found a solution for (still testing, but seems to work).

I wanted to have 2 zones of audio from WMP11 with correction filters applied to each zone. I use DRC to generate correction filters for each zone (i.e. 2 rooms being corercted for).

I have the following setup:

2 x m-audio transit USB -> 2 x DAC-AH -> 2 x Power amps

Then, I use Convolver to apply the filters and to create 4 output channels from 2 input channels (you could apply your XO here).

To connect the 2 devices into a single virtual device, i use Virtual Audio Cable as the sound device in WMP, under ASIO4ALL and use VSTHost to host the ConvolverVST plugin. Then, in VSTHost, i set the device to ASIO4ALL and the input to the 2xVAC channels and the output to the 2x2 m-audio channels (4 channels out). i see no reason why you could not connect any number of output devices in this way. regarding drift, i have not noticed any, but I am feeding to 2 different rooms so is hard to tell. I believe VSTHost has the ability to apply delays to specific channels. I wonder if the fact that I have 2 identical USB audio devices connected to the same powered USB hub alleviates. I'm still experimenting, but wanted to share.


Very nice, most people seem to rely on an application called console and a sound card driver that allows for internal routing, this rules out a number of soundcards!

I checked the home page for VAC, they reportedly support vista but no floating point formats. I was under the impressiion that vista audio is floating point.

It would be cool to have a system like yours running with media center under vista.
 
I have a design that uses the AD1940 as the DSP and CS8416 receiver chip that would do what you are after. Mostly. I also use an analog devices CODEC, as I am after anaogue output rather than digital.

If you ae after implementation of FIR filters, rather than IIR, then the AD1940 is not a good solution for you. The memory on the AD1940 is not large enough, and (as far as I can tell) the architecture of the AD1940 is not targeted at FIR filters.

That said, the AD1940 is brilliant for IIR filters, implementation of delays (the RAM is useable for true time delay independently for each output) and - well a heap of other stuff.

In my implementation I take the digital audio outputs straight to the D/A inputs off the CODEC. You would need to drive a digital output, I assume SPDIF or whatever your application needs. No big deal there.

If you are looking to build something (rather than buy a pre builtt box) then I am happy to provide you the schematics and software I use on the controller.

The second last version is here somewhere:

http://www.users.bigpond.com/anne.phil/default.htm


Phil
 
googlyone said:
If you are looking to build something (rather than buy a pre builtt box) then I am happy to provide you the schematics and software I use on the controller.

The second last version is here somewhere:

http://www.users.bigpond.com/anne.phil/default.htm

Phil

GrEaT�@project. Very nice to know real implementation of AD1940.
Now I'm playing with TI's 6713 DSK.
Hard to code that 6713 and compiler is very sensitive:(
 
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