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DIY DSP Active Speakers

edd9000

pfm Member
Since these are now close to finished, other than cosmetics, I thought I would post about them.

My initial goal was to build an active system, that was high efficiency using the Seos-12 CD horn.

Drivers were chosen by what came up on ebay, but I was looking for high efficiency, in the 100db range, decent response and high power handling.

To keep costs down a 3 way was chosen, this meant I could use the smaller horn, and I didn't need an 18" woofer capable of 1500hz.

I ended up with the Seos-12 horn, a precisions devices PD107 midrange and B&C 18PS76.

They are driven by my laptop running foobar, and using reaper as a host for the vst filters. Output is currently an 8 channel behringer soundcard, the fca610 which feeds my two Amcron PSA-2s and a Harman Kardon Citaction 24.

First some pictures of construction:
D300-9328_zps4618d642.jpg

D300-9333_zps6ebfd6fc.jpg

D300-9334_zps23f7316a.jpg


And as they are now, waiting final finish:
D300-9368_zps864d2597.jpg


The Design is loosely based on a JBL 4345, but I went with 3 separate boxes for future flexibility.
 
The crossovers are FIR, and designed in Rephase. Each driver was measured near field and the response used to build the crossover, here is an example of the mid:

import_zps6528a3e5.jpg

filter_zps170fae63.jpg


Each driver has its response corrected and the crossover filters saved. These are then loaded into Reaper using a convolver, in this case I am using voxengo pristine space.

Here is the result of the 3 drivers overlaid but not level matched against a full range measurement at 1m:
sum_zps60a19633.jpg


Full range sweep from 1.5m or showing completely linear phase:
response_zps6eff53e3.jpg


Impulse:
impulse_zps0dddbc19.jpg


Step response:
step_zpsf7eac0ea.jpg


Typical response at listening position:
inroom_zps69ef0cfe.jpg


Resulting response using DRC room correction:
withdrc_zpsb1b04599.jpg
 
I just measured the woofer slightly further back to make sure the ports output would be included.

The woofer enclosure and porting were designed in winisd, the port is tuned slightly low which creates a stepped response which is then flattened in the crossover filter.

Here is the before measurement:
bassraw_zps013e997b.jpg


You can see the phase switch from the port at around 40hz, and the step in response.

Here is the result after correction for response and phase in rephase:
bassafter_zps9726f1a1.jpg


I admit, none of this is especially precise as its done in room, so corrections were mostly based on the simulation more than the measurements, but the end result worked out.
 
Very good stuff, I love active DSP systems, so flexible. I just adjusted my target curve over the last few day. It now tips down similar to yours and I'm much happier with it overall although I might level the highs off a bit again. There is far more to be got out of considering how speaker drivers interact with each other and the room than the electronics in my recent experience. Very satisfying.

Stefan
 
I tried to make the speaker measure flat at 1m, and each driver flat in the near field. I rather like room correction, but I wanted the speakers to interact with the room as little as possibe, so correction wasn't needed. Drcdesigner lets you tailor the target curve but also the correction strength at various frequencies, I use it mostly in the bass, but I'm still experimenting.

I usualy add some bass boost,, 6db at 30hz, this isn't shown. It's then mostly flat to 8khz, and rolls off to around -3db at 18khz, the limit of my drivers/horns. Being a bbc/kef fan I often add a 2khz dip too.

I have found the bruel kjaer curve is the best place to start at listening position.
 
Wow am I glad I found this page via Google search! I am also considering the FCA610 as a way to split crossover frequencies to active speakers. Anything weird involved with getting this to work with Rephase/Reaper and the FCA610 or is it pretty straightforward? Smooth performance (set it and go) or do you need to reset/reboot anything a lot to continue operation over long periods of time?

How is the FCA610's sound quality?
 
Overall I am happy with the FCA610. It has some issues to be aware of:

It can sometimes crash, needing a power reset to make it work again, this hasn't happened using it, just when I leave it on and forget to unplug it after use.

It can crack and pop on settings changes and power up. I make sure to power off all my power amps while doing anything with the settings or turning it on.

I use rephrase to build the filters, and use them in a vst convolver hosted in reaper, its all fairly straight forward, if you want any more help just let me know.

The sound quality is fine, its not an expensive unit, using a loop back test and rightmark it measured well, low noise and thd. I don't really feel any need to change it for now.
 
Overall I am happy with the FCA610. It has some issues to be aware of:

It can sometimes crash, needing a power reset to make it work again, this hasn't happened using it, just when I leave it on and forget to unplug it after use.

It can crack and pop on settings changes and power up. I make sure to power off all my power amps while doing anything with the settings or turning it on.

I use rephrase to build the filters, and use them in a vst convolver hosted in reaper, its all fairly straight forward, if you want any more help just let me know.

The sound quality is fine, its not an expensive unit, using a loop back test and rightmark it measured well, low noise and thd. I don't really feel any need to change it for now.

Excellent, thanks for the feedback! Amazing that Behringer doesn't have any software of their own to make this an active crossover/DSP engine but I am guessing they don't want to cannibalize their DEQ/DCX market, though the FCA610 assumes the buyer has a capable PC.
 
Excellent work. I have been looking at digital crossovers for quite a while as I have had a similar project which has been on the go for a few years. At the moment the individual speakers are driven using an analogue xover which is fine. The xover has a pot for each channel and also one for each individual speaker. So if I fancy a bit more/less bass or trable its easy. It's a single ended system, but I would like to go balanced and I would imagine the digital xover is more accurate.

Edd: Did you consider alternative sound cards? eg the M-Audio Profire 610. I wondered whether any came with software that would do the job.

Aidan
 
The Behringer does come with audio workstation software which could host the VST crossovers, but I have not tried it. I like Reaper, its hugely flexible and available at a great price for individual use, and a fully functional demo.

I looked at other sound cards, anything that supports ASIO should work. I went with the behringer because it has 8 balanced analogue outputs, its usb, and had a master volume control. All of which the m-audio does but it is only firewire which I don't have on my laptop.

Digital crossovers are more flexible, its possible to use linear phase, and alter many aspects on the fly. You can also experiment with many different slopes, much steeper than offered with analogue.

Using an active crossover as a tone control isn't the best idea. When you raise or lower the level of an individual driver you also end up changing the crossover point, which can alter the behaviour of the speaker, either in the polar response or just how it blends. On the digital side, its very easy to build and measure a flat crossover, then just use another filter in front to control tone, I use a house curve target for this.
 
This has been a nice thread, I'd have never thought of doing it like this rather than how I've done it with a mini sharc. We use catch up tv with ours as well so the delay introduced by such huge FIR filters as Edd uses would be to much for me but a great way of doing things if it's music only.

Stefan
 
I only use huge FIR filters because I can. It would be easy to build filters with lower tap counts or even use IR filters for non critical use such as catch up TV.

The main issue is software like Foobar, happily outputs its audio to reaper, everything else I have to use the loopback built into the laptops soundcard which is awkward and sometimes glitchy. I have tried vb audio cable and others with the same problems, sometimes it works, sometimes it doesn't. It not a real issue for me as 99% of my listening is from foobar or vinyl, but its not ideal.
 
I only use huge FIR filters because I can. It would be easy to build filters with lower tap counts or even use IR filters for non critical use such as catch up TV.

The main issue is software like Foobar, happily outputs its audio to reaper, everything else I have to use the loopback built into the laptops soundcard which is awkward and sometimes glitchy. I have tried vb audio cable and others with the same problems, sometimes it works, sometimes it doesn't. It not a real issue for me as 99% of my listening is from foobar or vinyl, but its not ideal.

Do you need to use the loopback workaround to play browser sources like Pandora or Youtube or movies via VLC? What about the Spotify client, have you tried that one? Thanks!
 
Do you need to use the loopback workaround to play browser sources like Pandora or Youtube or movies via VLC? What about the Spotify client, have you tried that one? Thanks!

Anything that wont support an ASIO output, so anything in the browser needs to use the loopback. I have it stable now on my setup using a separate portable install of reaper using ASIO4ALL so I can take the loopback driver of my sound card (its called "stereo mix" under recording devices, and is on most realtek soundcards) and the output of the Behringer. Its not ideal, but I only use it now and then.

I'm not sure about VLC, I will have a look later.

This is all windows based, there may be other options with a mac.

There are soundcards with this feature in the drivers, such as focusrite.
 
I'm considering a digital LP/HP filter for my sub. Also, potentially, I want to be able to alter phase response and put in delays. Would the solutions mentioned so far fit the bill? I would like a choice of analogue and digital outputs. I'm after digital input, or something computer-based (since I stream everything).
 
In what way would you like to alter the phase response? do you mean the group delay of the subwoofer for linear phase?

If you just mean the general phase that is altered by changing the delay.

For a general sub crossover I would look at the minidsp units unless you use a pc as your primary playback source.
 
I believe it came with traktion, since I had already been using reaper I have not tried it.

In theory, any VST host will work, its just a matter of getting the routing right and how to stack the VST plugins.

The main reason I like reaper is the rearoute asio driver, I can use it as an output in foobar, and reaper can use it as an input to then run all the processing.

Jriver I believe will host VSTs, and do most of what I have been doing, but again, I have not tried it, my preference is for free/cheap software tools :)
 
In what way would you like to alter the phase response? do you mean the group delay of the subwoofer for linear phase?

If you just mean the general phase that is altered by changing the delay.

For a general sub crossover I would look at the minidsp units unless you use a pc as your primary playback source.
I would like complete control over the phase behaviour (but I would need a way of measuring phase too ...) but let's say for now delays would be sufficient: is there anything out there you recommend?

Many thanks and sorry for the delayed (!) response.
 


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