Kuei Yang Wang said:
Yes, this is what I said a few posts above. I also said that you are shooting in the dark unless you have a way of confirming your targets. Some posters do have measuring equipment and hopefully know how to take advantage of them. Let's say you are shooting for a nice 2nd order LR low pass filter at 1400Hz on your woofer. How do you know after measuring the driver through the Behringer how far off the target you really are? You have to load your measured response in a program like Leap, SoundEasy, lspCad or Filter Shop go to the optimizer window, set the target and eyeball the difference. Then apply a different crossover slope or EQ and re-measure, import the new measurement, go to the optimizer see how far off you are, and so on. If you have SMAART you can do this fairly quickly provided you load a target curve first. Then you have to repeat this procedure for all the drivers and move on to time aligning them. Then you listen to the speaker and see if you like what you hear. If for example the tweeter is straining a bit, you set a new goal and start all over again. That's how you would do it using good engineering practices. Off course you can just tweak the knobs untill you hear what you like and be done. I said before, nothing wrong with that approach. After all it's your speaker. If it sounds good to you, that's all it matters.
You are wrong there. LspCAD pro was first to provide MLS measurement, crossover design and simulation through a multichannel sound card. SoundEasy followed. Now you can use d-player, a free utility to do the simulation of a passive crossover. Hell, if you don't mind having to turn on your computer to listen to your favorite tunes, you can use those programs as your crossover. Much more flexible than a hardware box and cheaper too.
If concert PA sound is what you are after, then yes that approach would be OK. But don't claim you have a world class hi-fi loudspeaker if you go that route. You won't impress me there. I work for Clair Brothers, if you are into live sound then you know who that is. We use proprietary DSP's that are capable of much more than the Behringers and DBXes and I would choose those over an analog crossover ONLY IF the price was of no concern. For all the other boxes, I'll go with analog until DSP can actually do more. If I got a DSP processor that could write in it's memory any transfer function I can come up with in lspCAD in addition to delays and some tricky transient perfect crossovers that are not possible in analog world I would spend the money.
Calsod? You are in the wrong century. We don't use DOS anymore. I only know a couple of brilliant engineers that use it, but not because it's so good. It's because they know it inside out and are comfortable using it for last 10 years. Besides, what if Calsod optimizes your filter slopes to something in-between Butterworth and Linkwitz/Reiley? Or second order filter with Q of 1.8? No such preset slopes in the box, and you have to use up two or more bands of EQ to approximate it. Then you are left with no more DSP horse power to correct for other anomalies (in the Behringer. Other DSP boxes might have more filters available).
Again, not since the end of last century. THAT makes an audiophile quality VCA. Certainly it's distortion is on order of magnitude lower than the effects of a poor crossover.
Keep in mind that if you use a CD player's digital out, you feed the Behringer 16 bit resolution, at best 96dB S/N. The processing can only decimate it. You can play games with numbers, but you can't gain what's not there to begin with.
If you have a low res picture of a tree, it doesn't matter how you upsample it in Photoshop, you still won't be able to count the leaves on it. The same goes for digital audio. Your output converters can be 24 bit, but if you feed it equivalent of 14 bit resolution, that's what they're going to (truthfully) reproduce. Attenuate your 16 bit signal in digital domain by 12 dB and that's what you get- 14 bit resolution.
I'm not against digital crossovers in principle. They are just too stupid still. This will change soon enough. I just want to point out that digital processing is not artifact-free either. The simple textbook type algorithms don't work all that well. Good DSP engineers write some very sophisticated code to minimise ringing, aliasing, truncating etc. Who knows how good those algorithms in the Behringer really are?
Koinichiwa,
I "hold up" the Mike and look at my PC Screen. My PC being especially moved to the living/listening room for that. Software is freeware. PC I need to surf web. PC is Double hight Pizza Box size, screen 15" LCD. Smart, small & powerfull at 2.4GHz P4, 80gb HD, DVD/CD-RW Drive, 512MB RAM. Gawd, I remember thinking a 420MB HARDDISK was an AWFULLY BIG piece of storage - now my PC has more male sheep than that, hell, my grapics card has more RAM than my first Windoze NT Workstation which was a pretty penny back in the summe rof 94....
Hardly freeware, is it?
Actually, my current PC is a LOT more expensive than the Behringer hardware box but more importantly, even with a good Soundcard (which pushes the cost of the Behringer Box) it sounds worse in loop through mode than my (modified) behringer 8024 Digital EQ. Moreover, despite being from the manufacturer designed to be for a computer REALLY quiet it is a lot noisier (mechanically) than the background noise in my listening room.
So - cheaper? MAYBE but unlikely (unless quality really plays no role). Better or even equal on sonics? Hardly AND you have to keep the box in a different room and run bleeding long cables. And of course what happens if I want to surf the web and listen to music at the same time?
I don't dispute that a fully PC based solution is more flexible. But how flexible do you really need?
I'm not trying to. I run 30++ Year old 15" Tannoy Coaxials with 300B SE Amp's. Oh, and a digital EQ for room correction. That system, for monitoring or just playbak takes some beating, both from the High End and Pro-Audio side of things.... But (other than the size of the corner horn/reflex enclosures the system is not at all impressive. The music usually is though.
Hmm. In the REAL world a lot of people bolt a lot of drivers onto a plan Baffle like to drive them active and get good sound. Doing this analogue is no small job. I did stuff like that in the 1980's, full feedback loop around the woofers, current feed on midrange & tweeter, subtractive X-Overs and all that Jazz. In the end of the day my passive EV based "Pseudo Klipsch Cornwall" (build from an EV Kit) where a lot better and the 12" East German made Studio Coaxials where better in a lot of ways except dynamics and level handling, both driven from old V69 Style push-pull Valve Amps.
What does that tell us? Just because it is "newer" it ain't neccesarily better. When we got an early Yamaha Digital X-O for out large PA system it knocked the stuffing out of my own design LR active X-Over and that was really not bad at all - we took it on sound only over anything else we could have afforded and put the spare cash into more drivers...
Ahhm. I don't use DOS anymore either. But I have Calsod and it is jolly good at what it does. The user interface is non existent, but still better than a lot of early Unix or Mainframe stuff, so no sweat here. Of course, you cannot expect a modern WIMP (WIMP - Windoze Icon Menu Pointer) guy to use real mans software, but those stripped down things work surprisingly well. And I for one already have invested the money for the SW AND the time to learn it.... ;-)
BTW, under every GOS (Graphic Operating System) lurks a COS (Charater based Operating System). And no matter what platform, there are still loads of things that are much quicker, more reliable and controllable on a system prompt....
Graphical Shells are for looking good, not for getting work done.
Well, I have been using for 7 and yes, I know it inside out and it has no ******** "wizzards", "helpers", "graphic shells" and other superfluous junk that gets in the way of actually getting results.
Pull out one of the trusty parametric EQ section in the Behringer, loop it in and set it suitably. I will agree that it is possible to do better, but there is the question between very good sonically and "technically perfect". Also, if Calsod tells me I need a 2nd order filter with a Q of 1.8 I know it is really telling me that I have made wrong assumptions to start with.
BUT you trade applying a high Q "shelving" parametric for a reduced slope in the main X-Over and so you get most of your processing power back. It's not an issue from where I stand. Just set the damn thing right and get on with it.
Okay. THAT makes a lot of stuff and yes, their VCA's are a little better than those they functionally emulate. Subjectively ist much of a likeness though.
Of course (BTW, CD is limited to 93db S/N - honest). BUT with 113db S/N of the Behringers DA section you can attenuate 20db in the digital domain before you "loose bits" as it is so charmingly called.
OF course. But IF you print a 150 dpi Picture on a 1200dpi printer you can reduce the size of the picture dramatically (to 1/8) without loosing any of the detail.
Of course. And if I feed it 14 Bit attenuated by 60db (assuming youd DAC is a true 24 Bit device with -141db S/N) it will still be equivalent in resolution to 14 Bit, despite being attenyated by "10 Bit".
Actually, feed a 24 Bit system with 19bit equivalent analogue resolution (-111db S/N) a 16 Bit signal and attenuate the the signal dititally by 2 Bit (12db) and your signal still has full 16-bit equivalent resolution, it is just 12db less "loud". In fact, you can attenuate by 3 Bit (18db) and still have your full 16 Bit "resolution" retained.
Luckily enough most digital attenuation systems operate by something similar to "window shifting" the data in a much longer (32 - 64 Bit) length "word" (I know this is way too much a simlification but must suffice here) and not by truncation. The limit tends to be the analogue noisefloor, though narrowband analysis shows some converters remaining linear considerably BELOW the noisefloor....
... msg 2 lng....
Thunau said:
I "hold up" the Mike and look at my PC Screen. My PC being especially moved to the living/listening room for that. Software is freeware. PC I need to surf web. PC is Double hight Pizza Box size, screen 15" LCD. Smart, small & powerfull at 2.4GHz P4, 80gb HD, DVD/CD-RW Drive, 512MB RAM. Gawd, I remember thinking a 420MB HARDDISK was an AWFULLY BIG piece of storage - now my PC has more male sheep than that, hell, my grapics card has more RAM than my first Windoze NT Workstation which was a pretty penny back in the summe rof 94....
Thunau said:
Hardly freeware, is it?
Thunau said:
Actually, my current PC is a LOT more expensive than the Behringer hardware box but more importantly, even with a good Soundcard (which pushes the cost of the Behringer Box) it sounds worse in loop through mode than my (modified) behringer 8024 Digital EQ. Moreover, despite being from the manufacturer designed to be for a computer REALLY quiet it is a lot noisier (mechanically) than the background noise in my listening room.
So - cheaper? MAYBE but unlikely (unless quality really plays no role). Better or even equal on sonics? Hardly AND you have to keep the box in a different room and run bleeding long cables. And of course what happens if I want to surf the web and listen to music at the same time?
I don't dispute that a fully PC based solution is more flexible. But how flexible do you really need?
Thunau said:
I'm not trying to. I run 30++ Year old 15" Tannoy Coaxials with 300B SE Amp's. Oh, and a digital EQ for room correction. That system, for monitoring or just playbak takes some beating, both from the High End and Pro-Audio side of things.... But (other than the size of the corner horn/reflex enclosures the system is not at all impressive. The music usually is though.
Thunau said:
Hmm. In the REAL world a lot of people bolt a lot of drivers onto a plan Baffle like to drive them active and get good sound. Doing this analogue is no small job. I did stuff like that in the 1980's, full feedback loop around the woofers, current feed on midrange & tweeter, subtractive X-Overs and all that Jazz. In the end of the day my passive EV based "Pseudo Klipsch Cornwall" (build from an EV Kit) where a lot better and the 12" East German made Studio Coaxials where better in a lot of ways except dynamics and level handling, both driven from old V69 Style push-pull Valve Amps.
What does that tell us? Just because it is "newer" it ain't neccesarily better. When we got an early Yamaha Digital X-O for out large PA system it knocked the stuffing out of my own design LR active X-Over and that was really not bad at all - we took it on sound only over anything else we could have afforded and put the spare cash into more drivers...
Thunau said:
Ahhm. I don't use DOS anymore either. But I have Calsod and it is jolly good at what it does. The user interface is non existent, but still better than a lot of early Unix or Mainframe stuff, so no sweat here. Of course, you cannot expect a modern WIMP (WIMP - Windoze Icon Menu Pointer) guy to use real mans software, but those stripped down things work surprisingly well. And I for one already have invested the money for the SW AND the time to learn it.... ;-)
BTW, under every GOS (Graphic Operating System) lurks a COS (Charater based Operating System). And no matter what platform, there are still loads of things that are much quicker, more reliable and controllable on a system prompt....
Graphical Shells are for looking good, not for getting work done.
Thunau said:
Well, I have been using for 7 and yes, I know it inside out and it has no ******** "wizzards", "helpers", "graphic shells" and other superfluous junk that gets in the way of actually getting results.
Thunau said:
Pull out one of the trusty parametric EQ section in the Behringer, loop it in and set it suitably. I will agree that it is possible to do better, but there is the question between very good sonically and "technically perfect". Also, if Calsod tells me I need a 2nd order filter with a Q of 1.8 I know it is really telling me that I have made wrong assumptions to start with.
Thunau said:
BUT you trade applying a high Q "shelving" parametric for a reduced slope in the main X-Over and so you get most of your processing power back. It's not an issue from where I stand. Just set the damn thing right and get on with it.
Thunau said:
Okay. THAT makes a lot of stuff and yes, their VCA's are a little better than those they functionally emulate. Subjectively ist much of a likeness though.
Thunau said:
Of course (BTW, CD is limited to 93db S/N - honest). BUT with 113db S/N of the Behringers DA section you can attenuate 20db in the digital domain before you "loose bits" as it is so charmingly called.
Thunau said:
OF course. But IF you print a 150 dpi Picture on a 1200dpi printer you can reduce the size of the picture dramatically (to 1/8) without loosing any of the detail.
Thunau said:
Of course. And if I feed it 14 Bit attenuated by 60db (assuming youd DAC is a true 24 Bit device with -141db S/N) it will still be equivalent in resolution to 14 Bit, despite being attenyated by "10 Bit".
Thunau said:
Actually, feed a 24 Bit system with 19bit equivalent analogue resolution (-111db S/N) a 16 Bit signal and attenuate the the signal dititally by 2 Bit (12db) and your signal still has full 16-bit equivalent resolution, it is just 12db less "loud". In fact, you can attenuate by 3 Bit (18db) and still have your full 16 Bit "resolution" retained.
Luckily enough most digital attenuation systems operate by something similar to "window shifting" the data in a much longer (32 - 64 Bit) length "word" (I know this is way too much a simlification but must suffice here) and not by truncation. The limit tends to be the analogue noisefloor, though narrowband analysis shows some converters remaining linear considerably BELOW the noisefloor....
... msg 2 lng....
...msg cont....
Actually - with all due respect - the poor workman blames his tools. It is not the DSP systems that are too stupid.... ;-)
I argued a fully digital "processor" system at the heart of an active Loudspeaker System which would account for all driver non-linearities (dynamic and static as well as thermal) the room interactions and so on by using a suitable measurement signal and processing in the mid 90's with Ken Kantor (NHT) and some other heavy guns on the Bass List. We are still nowhere close.
I for one will not hold my breath. But keep that youthfull enthusiam.
No processing whatsoever is "lossless". I find however that doing things in software are a lot easier to get right than doing them in hardware. I have done both in a variety of areas.
EVEN IF I COULD BE BOTHERED, the concept of making an active equalised X-Over for my tannoys using Op-Amp's and all and making it "universal" as 3-Way 1st to 4th order, 3 X Parametric EQ per way and all the rest when I can get a reasonable Digital Unit that does all of that plus timealignment in software for under 1500 Bucks (I'm not neccesarily talking Behringer here) tells me it's to cut my losses.
Well, I'll find out soon. The ones in their oder digital EQ are good enough that the slight if notable loss they cause (or whatever in the box does cause it) is easily offset by the gains from fixing the more insidious problems of room/speaker interaction to a reasonable degree.
Maybe, after I have actually spend month working out the best sounding possible curves, EQ's etc while using as little processing as possible, I'll sit down and turn the compount X-Over curves into a passive LCR line level X-Over, but I suspect I will not be able to be arsed....
Sayonara
Thunau said:
Actually - with all due respect - the poor workman blames his tools. It is not the DSP systems that are too stupid.... ;-)
Thunau said:
I argued a fully digital "processor" system at the heart of an active Loudspeaker System which would account for all driver non-linearities (dynamic and static as well as thermal) the room interactions and so on by using a suitable measurement signal and processing in the mid 90's with Ken Kantor (NHT) and some other heavy guns on the Bass List. We are still nowhere close.
I for one will not hold my breath. But keep that youthfull enthusiam.
Thunau said:
No processing whatsoever is "lossless". I find however that doing things in software are a lot easier to get right than doing them in hardware. I have done both in a variety of areas.
EVEN IF I COULD BE BOTHERED, the concept of making an active equalised X-Over for my tannoys using Op-Amp's and all and making it "universal" as 3-Way 1st to 4th order, 3 X Parametric EQ per way and all the rest when I can get a reasonable Digital Unit that does all of that plus timealignment in software for under 1500 Bucks (I'm not neccesarily talking Behringer here) tells me it's to cut my losses.
Thunau said:
Well, I'll find out soon. The ones in their oder digital EQ are good enough that the slight if notable loss they cause (or whatever in the box does cause it) is easily offset by the gains from fixing the more insidious problems of room/speaker interaction to a reasonable degree.
Maybe, after I have actually spend month working out the best sounding possible curves, EQ's etc while using as little processing as possible, I'll sit down and turn the compount X-Over curves into a passive LCR line level X-Over, but I suspect I will not be able to be arsed....
Sayonara
"I don't dispute that a fully PC based solution is more flexible. But how flexible do you really need?"
This is what this whole discussion boils down to. A couple of years ago thre was a commercial on TV ( I forget for whom, I think it was a car company) in which a kid asks a question: "If things are left "Good enough" will they ever be good enough?".
I would like to keep making each of my loudspeakers objectively better than the last one. If you feel that a 30 year old EV corner-honker can't be topped, good for you. One less thing to stress about.
A while ago I saved a post to the Madisound board by Siegfried Linkwitz.
I can't find it now, but the message was, "you as a DIY'er are not restricted by commercial restraints, do what ever you want, be as creative as you can, pursue new things in the name of progress" etc. I feel that getting a generic crossover/processor is against that spirit.
My collegues at work developed new ways of building line arrays, new types of EQ, etc. The Clair iO processor has transient-perfect, infinite slope crossovers. It takes a lot of DSP horsepower (it turns a 2x6 box into a 1x3 box) to do something that is disputed as to being even audible. Why did they do it? Because it progresses the state of the art, because they can, because they are not satisfied with what has been done. It's in their blood.
I'm the same way when it comes to my speakers. I like to pursue new things. Getting the Behringer is just not "good enough".
BTW, Behringer is coming out with an 8 ch. mic pre/AD/DA for about $129.00 if I remember correctly. It has ADAT optical interface and presumably the same converters as the DCX2496. An ADAT card for a PC is cheap. Optical cable can be quite long. Get Mathlab to spit out some filters and you can do as well as anything out there. Takes a bit more work but thats why we post at diyaudio.com, right?
This is what this whole discussion boils down to. A couple of years ago thre was a commercial on TV ( I forget for whom, I think it was a car company) in which a kid asks a question: "If things are left "Good enough" will they ever be good enough?".
I would like to keep making each of my loudspeakers objectively better than the last one. If you feel that a 30 year old EV corner-honker can't be topped, good for you. One less thing to stress about.
A while ago I saved a post to the Madisound board by Siegfried Linkwitz.
I can't find it now, but the message was, "you as a DIY'er are not restricted by commercial restraints, do what ever you want, be as creative as you can, pursue new things in the name of progress" etc. I feel that getting a generic crossover/processor is against that spirit.
My collegues at work developed new ways of building line arrays, new types of EQ, etc. The Clair iO processor has transient-perfect, infinite slope crossovers. It takes a lot of DSP horsepower (it turns a 2x6 box into a 1x3 box) to do something that is disputed as to being even audible. Why did they do it? Because it progresses the state of the art, because they can, because they are not satisfied with what has been done. It's in their blood.
I'm the same way when it comes to my speakers. I like to pursue new things. Getting the Behringer is just not "good enough".
BTW, Behringer is coming out with an 8 ch. mic pre/AD/DA for about $129.00 if I remember correctly. It has ADAT optical interface and presumably the same converters as the DCX2496. An ADAT card for a PC is cheap. Optical cable can be quite long. Get Mathlab to spit out some filters and you can do as well as anything out there. Takes a bit more work but thats why we post at diyaudio.com, right?
I have been playing with 2 of these units for couple of weeks now. Although the potential of these are enormous I still find some serious problems with it.
All my comments are in regards to “digital in” setup. I find the sound quality going in analog marginal. First the input volume is way too high for home use. You have to turn the input volume control way done which causes bit lose and tremendous increases in S/N (that is after turning down the output volume). One more, perhaps unmanageable, problem is a clear collapse of the sound stage. Every thing is right in the middle, to a point that I had to ck if I am not playing L-R in mono (I wasn’t). Anyone with some thought on the matter?
All my comments are in regards to “digital in” setup. I find the sound quality going in analog marginal. First the input volume is way too high for home use. You have to turn the input volume control way done which causes bit lose and tremendous increases in S/N (that is after turning down the output volume). One more, perhaps unmanageable, problem is a clear collapse of the sound stage. Every thing is right in the middle, to a point that I had to ck if I am not playing L-R in mono (I wasn’t). Anyone with some thought on the matter?
awolf said:
mmm...I've been using it for a solid week and I think it sounds better than I expected.
Your "too high volume for home use" problem can be fixed easyly atenuating the signal going from the analog outs of the crosover to the amps. Simple resistive attenuators -a couple of resistors -will do.
I'll report back when I have time to do a more "in deep" listen.
cheers
Ric
awolf,
You need to ensure that a fairly large signal is presented at the analog inputs of the DCX2496 in order to maximize its S/N. If you look at the specs, it says there that 22 dBu is a max input level. It would seem that around 10 V is what you need to aim for. However, since DCX2496 is a unity gain device, such a huge input would create all kinds of problems, like the once you are talking about.
When it comes to the digital input, I would imagine you are using the SPDI/F, the level does not really matter as long as it is within specs. However, that still may appear to be too loud.
You need a multi-channel volume control placed after the DCX2496 and you also need a line level amplifier to boost the input signal to the required levels. If you do that, then expected S/N of the Behringer is 113 dB and that is better then the 16 bits you are getting, and that on a good day, from your CD or SACD. So DCX2496 should be fine. I am waiting for mine along with the DEQ2496. By the way I think that 113 dB is an A-weighted value, so in the real world that number must be lower, but still most likely better then 16 bits.
Multi-channel volume is a tricky project, but I just read the Apox thread and am quite impressed. I just might make one for my HT system here. I will need to control two 3-way channels (the fronts), two 2-way channels (the rears), and two single channels (center and the sub). So a total of 12 lines. I’ll need 6 boards and it is possible with Apox design. Each board can be individually controlled, - nice.
Vadim
You need to ensure that a fairly large signal is presented at the analog inputs of the DCX2496 in order to maximize its S/N. If you look at the specs, it says there that 22 dBu is a max input level. It would seem that around 10 V is what you need to aim for. However, since DCX2496 is a unity gain device, such a huge input would create all kinds of problems, like the once you are talking about.
When it comes to the digital input, I would imagine you are using the SPDI/F, the level does not really matter as long as it is within specs. However, that still may appear to be too loud.
You need a multi-channel volume control placed after the DCX2496 and you also need a line level amplifier to boost the input signal to the required levels. If you do that, then expected S/N of the Behringer is 113 dB and that is better then the 16 bits you are getting, and that on a good day, from your CD or SACD. So DCX2496 should be fine. I am waiting for mine along with the DEQ2496. By the way I think that 113 dB is an A-weighted value, so in the real world that number must be lower, but still most likely better then 16 bits.
Multi-channel volume is a tricky project, but I just read the Apox thread and am quite impressed. I just might make one for my HT system here. I will need to control two 3-way channels (the fronts), two 2-way channels (the rears), and two single channels (center and the sub). So a total of 12 lines. I’ll need 6 boards and it is possible with Apox design. Each board can be individually controlled, - nice.
Vadim
I use one of these to attenuate the signal before my power amps:
http://www.decibelhifi.com.au/prod18.htm
As for digital crossovers collapsing the soundstage, before I built my dipoles I compared the sound with the dbx Driverack PA in and out of the circuit with my Wilson WATTs, which image very well.
There was no effect on soundstage that I could hear.
Steve
http://www.decibelhifi.com.au/prod18.htm
As for digital crossovers collapsing the soundstage, before I built my dipoles I compared the sound with the dbx Driverack PA in and out of the circuit with my Wilson WATTs, which image very well.
There was no effect on soundstage that I could hear.
Steve
Behringer DCX2496
Thanks for the reply. Vadim seems to grasp the problem. Simply putting an L pad on the output is not as simple as one think. Also, these are all balance output.
I have tried it with a multichannel controller and it was a disaster. I think the trick may be to change the gain on the output opamp. But that will be tricky to. And then, we still have the issue of sound stage. I am comparing the Behringer to other active XO (all analog). It has potential but also problems
Thanks for the reply. Vadim seems to grasp the problem. Simply putting an L pad on the output is not as simple as one think. Also, these are all balance output.
I have tried it with a multichannel controller and it was a disaster. I think the trick may be to change the gain on the output opamp. But that will be tricky to. And then, we still have the issue of sound stage. I am comparing the Behringer to other active XO (all analog). It has potential but also problems
Cool
Should not be that hard
I saw some chips/circuits for "converting" ttl to s/pdif! I belive the d/a uses ttl/cmos signas for input!? So you could just bypass the d/a by the input to your own pcb with 6 of these circuits, then connect it to the xlr outputs... That would be great! then you don't have to care for other modifications on the behringer
Should not be that hard
I saw some chips/circuits for "converting" ttl to s/pdif! I belive the d/a uses ttl/cmos signas for input!? So you could just bypass the d/a by the input to your own pcb with 6 of these circuits, then connect it to the xlr outputs... That would be great! then you don't have to care for other modifications on the behringer
sfdoddsy said:
Steve, I was looking at the pics early in this thread with just this idea - since I (think) I've seen you active on some of the Panny XR threads, I suspect that you're thinking the same thing I am - running digital straight into one of the little Panasonic XR units.
It actually looks like it would be 'pretty easy' to tap into the I2S lines going into the DAC chips. The pitch on the SMD devices doesnt' look _that_ bad, but I haven't seen one in person. Assuming you could do this, then it wouldn't be much problem to wire up either some spdif transmitters or some line drivers/buffers to send the I2S lines out directly.
The XR25/XR45 might be a bit tough to mod for direct I2S input, but it's really tough to say without the service manual.
Unfortunately I am not competent enough to do something like this myself. I can drill holes in MDF, but electronics, no. I would have to pay someone to do it.
I talked to a place here in Australia (www.soundlabs.com) and they mentioned something in the order of $800 US to install a BDE kit.
That's a little more than I want to pay.
If anyone can do such a thing for me over there for much less, since that's where I'm getting the 2496, I'd be interested to hear. In other words, I'll supply a DCX2496, and pay for parts etc, you get to see if it works. As long as it doesn't destroy the current functionality.
The XR45 reference is to a thread at AVS about using the Panasonic SA-XR45 as a digital amp for a fully digital system all the way to the speakers.
John Meyer at Newform Speakers is currently recommending using the DCX 2496 and the XR45 to biamp his very good speakers. It struck me (and others), that it if you could modify the DCX2496 for digital output, then this would be the missing link in the chain.
Add a DEQ2496 and you get digital room EQ as well.
Cheers
Steve
I talked to a place here in Australia (www.soundlabs.com) and they mentioned something in the order of $800 US to install a BDE kit.
That's a little more than I want to pay.
If anyone can do such a thing for me over there for much less, since that's where I'm getting the 2496, I'd be interested to hear. In other words, I'll supply a DCX2496, and pay for parts etc, you get to see if it works. As long as it doesn't destroy the current functionality.
The XR45 reference is to a thread at AVS about using the Panasonic SA-XR45 as a digital amp for a fully digital system all the way to the speakers.
John Meyer at Newform Speakers is currently recommending using the DCX 2496 and the XR45 to biamp his very good speakers. It struck me (and others), that it if you could modify the DCX2496 for digital output, then this would be the missing link in the chain.
Add a DEQ2496 and you get digital room EQ as well.
Cheers
Steve
Transformers for DCX2496
I am sure the owners of the DCX2496 are having difficulty coming to terms with its lack-lustre sound in its stock form.
What do you guys think of the idea of using a line-level transformer directly driven by the AK4393 dac chip's differential output?
Will Sowter's 3575 be a good choice?http://www.sowter.co.uk/specs/3575.htm
What would be your recommendation?
I am sure the owners of the DCX2496 are having difficulty coming to terms with its lack-lustre sound in its stock form.
What do you guys think of the idea of using a line-level transformer directly driven by the AK4393 dac chip's differential output?
Will Sowter's 3575 be a good choice?http://www.sowter.co.uk/specs/3575.htm
What would be your recommendation?