XLR
I use XLR all over the signal path, so I do not use adapters, but I can offer this information:
1-Converting "XLR OUTs to RCA" is easyly done, if you are dealing with short interconects.(short cables).
You just need to use only one of the differenctial outs (pin 2)and Pin 1 and then lower the level by around 11 something db. This is easyly done with a pair of resistors. Ther's a lot of schematics in the net that shows you cleary how to do it.And is not expensive at all! Do a Google search.
2-Converting "RCA to XLR ins" is no that easy. You need to apply gain by the same amount.Either electronically with an active gain stage or using good quality transformers. That's going to be more expensive.
You can live without increasing the gain, but your signal to noise ratio will suffer almost 12 db!
3-Many DCX2496 owners use the digital in and then siimply atenuate the output to go to the amps and that works reasonable ok.
cheers
Ric
I use XLR all over the signal path, so I do not use adapters, but I can offer this information:
1-Converting "XLR OUTs to RCA" is easyly done, if you are dealing with short interconects.(short cables).
You just need to use only one of the differenctial outs (pin 2)and Pin 1 and then lower the level by around 11 something db. This is easyly done with a pair of resistors. Ther's a lot of schematics in the net that shows you cleary how to do it.And is not expensive at all! Do a Google search.
2-Converting "RCA to XLR ins" is no that easy. You need to apply gain by the same amount.Either electronically with an active gain stage or using good quality transformers. That's going to be more expensive.
You can live without increasing the gain, but your signal to noise ratio will suffer almost 12 db!
3-Many DCX2496 owners use the digital in and then siimply atenuate the output to go to the amps and that works reasonable ok.
cheers
Ric
br,
For digital input you can use the input sliders to reduce the level if required. It can be a bit confusing however because the level indicators don't reflect this change and might still be bouncing up into the red even though the unit is not clipping internally.
If you have any equalization programmed that is boosting levels above 0db then you risk clipping. I would suggest set the input sliders no higher than -3db, and even lower if there is some EQ boost programmed. The clipping indicators may still be flashing, but it's probably okay. You need an oscilloscope and test CD with some 0dFS recorded tracks to really confirm this.
RCA/XLR adaptors are available from many outlets for cheap. Check MCMelectronics.com.
Some of the Behringer units adjust the gain automatically if using unbalanced interfaces so there really isn't any penalty for doing this. The 12db reduction in SN ratio mentioned by Ric is actually a different subject. Pro units generally operate at a level 12db higher than consumer units. It's good to take advantage of this extra headroom if you can, but if it means interfacing with transformers and/or outboard active devices it's probably not worth the trouble.
The most important thing to consider with the DCX is volume control implementation. If you're using a digital input then you need a good quality four/six channel attenuator or VCA or something similar.
Cheers,
Davey.
For digital input you can use the input sliders to reduce the level if required. It can be a bit confusing however because the level indicators don't reflect this change and might still be bouncing up into the red even though the unit is not clipping internally.
If you have any equalization programmed that is boosting levels above 0db then you risk clipping. I would suggest set the input sliders no higher than -3db, and even lower if there is some EQ boost programmed. The clipping indicators may still be flashing, but it's probably okay. You need an oscilloscope and test CD with some 0dFS recorded tracks to really confirm this.
RCA/XLR adaptors are available from many outlets for cheap. Check MCMelectronics.com.
Some of the Behringer units adjust the gain automatically if using unbalanced interfaces so there really isn't any penalty for doing this. The 12db reduction in SN ratio mentioned by Ric is actually a different subject. Pro units generally operate at a level 12db higher than consumer units. It's good to take advantage of this extra headroom if you can, but if it means interfacing with transformers and/or outboard active devices it's probably not worth the trouble.
The most important thing to consider with the DCX is volume control implementation. If you're using a digital input then you need a good quality four/six channel attenuator or VCA or something similar.
Cheers,
Davey.
Iam using a 6 channel NLE attenuator in front of my Poweramp. The Behringer gets the digital signal from a Teac vrds 10 but now I have regornize some new strange sounds and it is not when the input clips - has set it to -4dB. The sounds comes and goes on some CDs - Its some "Clicks" sound.
The Firmware is ver. 1.14 I will today try to upgrade it to 1.15 dont no the difference.
If I use the analog input it works fine but that was not the setup I wanted.
The Firmware is ver. 1.14 I will today try to upgrade it to 1.15 dont no the difference.
If I use the analog input it works fine but that was not the setup I wanted.
Davey said:
Yes, in a 24 bit unit the penalty for lowering the gain of a 16 bit signal is somewhat lessened, but you should know where in the signal path the gain change is least harmful. I think that lowering of the output is the best way to do it. Trimming the input results in processing of a lower resolution source.
As to clipping, the SHARK processor is a 32 bit floating point device. You can hardly clip it internally. The internal dynamic range is huge - about 1500dB. As long as you lower your signal in the digital domain before it hits the DA converters to compensate for any processing related signal boost you are fine.
I think that the best way to adapt the Behringer to a ha-fi setup would be to lower the output gain of the buffer amplifiers. I know, it probably would be hard due to use of surface mount components, but the resulting headroom would be most welcome.
Jan,
Yeah, I understand your suggestion and the internal dynamic range of the processor, but in this case the clipping problem is not related to the SHARC unit....at least I don't think so. More likely the input is maxed somehow because level reduction applied in the digital domain will not fix the problem. I'm unsure how the architecture allows this, but it does. Hmmmm.
Lowering the gain of the output amplifiers also won't correct the problem since the waveform they see is already clipped. It will just result in lower amplitude clipped waveforms.
On my unit the output waveform is clipped with 0dbFS recorded information on the digital input if the input sliders are set anywhere higher than -1db. This holds true regardless of EQ, level shifting, crossover effects, etc, that are programmed downstream.
I've investigated other setup configurations to try and work around this "problem," but have been unsuccessful so far.
Cheers,
Davey.
Yeah, I understand your suggestion and the internal dynamic range of the processor, but in this case the clipping problem is not related to the SHARC unit....at least I don't think so. More likely the input is maxed somehow because level reduction applied in the digital domain will not fix the problem. I'm unsure how the architecture allows this, but it does. Hmmmm.
Lowering the gain of the output amplifiers also won't correct the problem since the waveform they see is already clipped. It will just result in lower amplitude clipped waveforms.
On my unit the output waveform is clipped with 0dbFS recorded information on the digital input if the input sliders are set anywhere higher than -1db. This holds true regardless of EQ, level shifting, crossover effects, etc, that are programmed downstream.
I've investigated other setup configurations to try and work around this "problem," but have been unsuccessful so far.
Cheers,
Davey.
Pretty much all of the modern popular music is "maximized" to the point of clipping in the mastering process, but a lot of it comes to the mastering house already clipped in the recording process. It's is very hard to remove clipping especially if the client won't have it 1 dB lower than another popular CD used as reference during mastering. I have been guilty of doing such damage in the past, but since I got out of the recording world I learned to hate it. Now to me it's just grotesque, and is one of the reasons why I don't listen much to modern rock and pop.
What music are you listening to?
Your input meters are showing the digital input level before the digital attenuator. Most likely they are preset to trigger the red LED a couple of dB's below 0dBFS. After all this is a pro audio unit, meant to be fed output of a mixing console. If you put in clipped material the clipping will be preserved through the signal path. Not much you can do about it, other than use analog inputs and feed it the signal at "safe" levels.
Maybe the digital output stages of your CD player have some kind of interpolating circuit that reduces the audible effects of clipping. Some do. If you insist on using the digital input, find CD's that are not clipped, or just accept the fact that your favorite music is recorded poorly.
If enough people complain, maybe something will change in the world of music production. Quite a few mastering engineers are already outspoken about it. Most notably Bob Katz of Digital Domain.
What music are you listening to?
Your input meters are showing the digital input level before the digital attenuator. Most likely they are preset to trigger the red LED a couple of dB's below 0dBFS. After all this is a pro audio unit, meant to be fed output of a mixing console. If you put in clipped material the clipping will be preserved through the signal path. Not much you can do about it, other than use analog inputs and feed it the signal at "safe" levels.
Maybe the digital output stages of your CD player have some kind of interpolating circuit that reduces the audible effects of clipping. Some do. If you insist on using the digital input, find CD's that are not clipped, or just accept the fact that your favorite music is recorded poorly.
If enough people complain, maybe something will change in the world of music production. Quite a few mastering engineers are already outspoken about it. Most notably Bob Katz of Digital Domain.
Jan,
I'm using a test CD with 0dbFS recorded sine waves and monitoring the outputs with an oscilloscope. I've analyzed this test disk with other means and there are no clipped samples on any of the tracks. I've also generated some of my own CD's using CoolEdit.
I may have overstated the "problem" here....it's not really a problem as long as the input sliders are set to -1db. Everything comes through clean as a whistle in that case. However, as you mentioned, the triggering point for the red LED's appears to be significantly below 0dbFS since they are illuminated constantly with this level input even though the waveforms look good.
My concern is for less experienced users who don't have the testing capabilities I have to identify this effect. The level indicators do not reliably indicate a clipping condition so it's possible to have a significant effect added to "hot" CD's and not realize it. It would only be severe if the input sliders were bumped up considerably...not a typical situation. However, it's something to be aware of.
Cheers,
Davey.
I'm using a test CD with 0dbFS recorded sine waves and monitoring the outputs with an oscilloscope. I've analyzed this test disk with other means and there are no clipped samples on any of the tracks. I've also generated some of my own CD's using CoolEdit.
I may have overstated the "problem" here....it's not really a problem as long as the input sliders are set to -1db. Everything comes through clean as a whistle in that case. However, as you mentioned, the triggering point for the red LED's appears to be significantly below 0dbFS since they are illuminated constantly with this level input even though the waveforms look good.
My concern is for less experienced users who don't have the testing capabilities I have to identify this effect. The level indicators do not reliably indicate a clipping condition so it's possible to have a significant effect added to "hot" CD's and not realize it. It would only be severe if the input sliders were bumped up considerably...not a typical situation. However, it's something to be aware of.
Cheers,
Davey.
I have now upgraded it to 1.15 and the problem are still there.
The problem is very stable when I play the old Dire Strasits CD from 1978 - Vertigo 800 051-2.
Track 4 - "Six Blade knife"
The level is only comming up to -5dB
The click sounds and phase distortions is very clear on that track.
Iam a little confused could it be my ZapPulse Amp that makes the problem instead ???
But to day its new yearsday so Ill wait to next year.
Happy new year out there
- Dion
The problem is very stable when I play the old Dire Strasits CD from 1978 - Vertigo 800 051-2.
Track 4 - "Six Blade knife"
The level is only comming up to -5dB
The click sounds and phase distortions is very clear on that track.
Iam a little confused could it be my ZapPulse Amp that makes the problem instead ???
But to day its new yearsday so Ill wait to next year.
Happy new year out there
- Dion
dkxdn said:
The clicks are most likely due to scratches on the surface of the CD. CD Red Book spec calls for a certain amount of data redundancy and the error correction is built in the data format. But once the number of errors reach certain level they become non-recoverable. There is just not enough good data to figure out what the missing bits are. Some CD players will interpolate the data, some will output silence in those moments. In your case it sounds like the digital stream has moments of digital silence where the data is missing. The next sample might be at fairly high level and the instantanious jump from 0V to maybe 1V results in a audible click.
Few suggestions.
1. Clean the CD . Sometimes it's just dust or dirt that causes errors.
2. Try a different transport
3. If you don't have a different transport try ripping the CD using EAC. It will reduce the number of errors dramatically, then make a CDR and see if the clicks are still there.
4. Double check your S/PDIF connection. Are you using 75Ohm cable, is your wiring correct?
Hope this helps.
I have now check the cable:
75ohm coax cable normaly used for TV and radio tranmission, one end with RCA plugs and the other end with XLR gold from Neutrik.
Then I changed it with another one:
Kimber PBJ cable one end with RCA plugs and the other end with XLR not gold from Neutrik.
And what happens ? All the click and the funny phase sound was gone ! and I believed that the coax cable was the right way.
I have checked the coax cable again and again but it seems to be ok. The connection on both cables on the XLR is the signal on pin 2 and GND on pin 3.
Mhn ???
75ohm coax cable normaly used for TV and radio tranmission, one end with RCA plugs and the other end with XLR gold from Neutrik.
Then I changed it with another one:
Kimber PBJ cable one end with RCA plugs and the other end with XLR not gold from Neutrik.
And what happens ? All the click and the funny phase sound was gone ! and I believed that the coax cable was the right way.
I have checked the coax cable again and again but it seems to be ok. The connection on both cables on the XLR is the signal on pin 2 and GND on pin 3.
Mhn ???
An externally hosted image should be here but it was not working when we last tested it.
in case you haven't already done this, check if connection pin 1 to 3 will help? (if the dig. input is servobalanced, maybe problems occur because of floating from earth etc.)
What I don't understand:
Everyone wants to do hardware-upgrades, but isn't most of the sound of the processor due to dsp code? The converters etc are normal quality 24/96, good enough to start with. A lot of the audio plug-ins (Rtas Vst etc) also improved huge soundquality the last years, all due to change in code. I think there are probably a lot ways to program filters in dsp's.....
And I agree completely that you'll need a six way accurate volume control after the processor, to control main volume.
Make sure that the processor is always running with max. bitdepth.
With analog/digi inputs used: lets say leave 4 dB headroom on the input (calibrated to 0dBU/0dBFS). If you only play cd's/dvd's/sacd's from a preamp orso it is very easy to calibrate, because the input voltage can impossible become higher. If you play records you need probably more margin.
In this way you should get good results for hifi, otherwise the unit is no good.
Has anyone checked the pricy XTA 226 soundwise? there is a installation version (with no controls on the front configure with notebook through rs-485, or with a short datacable), sells for around 2k. 2 ins 6 out.
In Prosound/PA world this is considered as the best sounding commercial digi processor so far....
There is also the processor used by clair brothers, wich they buy somewhere else I believe (I have found it somewhere on the net in the past, think it was a german company), with a touchscreen, but it is more expensive, looks quite serious.
cheers
What I don't understand:
Everyone wants to do hardware-upgrades, but isn't most of the sound of the processor due to dsp code? The converters etc are normal quality 24/96, good enough to start with. A lot of the audio plug-ins (Rtas Vst etc) also improved huge soundquality the last years, all due to change in code. I think there are probably a lot ways to program filters in dsp's.....
And I agree completely that you'll need a six way accurate volume control after the processor, to control main volume.
Make sure that the processor is always running with max. bitdepth.
With analog/digi inputs used: lets say leave 4 dB headroom on the input (calibrated to 0dBU/0dBFS). If you only play cd's/dvd's/sacd's from a preamp orso it is very easy to calibrate, because the input voltage can impossible become higher. If you play records you need probably more margin.
In this way you should get good results for hifi, otherwise the unit is no good.
Has anyone checked the pricy XTA 226 soundwise? there is a installation version (with no controls on the front configure with notebook through rs-485, or with a short datacable), sells for around 2k. 2 ins 6 out.
In Prosound/PA world this is considered as the best sounding commercial digi processor so far....
There is also the processor used by clair brothers, wich they buy somewhere else I believe (I have found it somewhere on the net in the past, think it was a german company), with a touchscreen, but it is more expensive, looks quite serious.
cheers
dokter dB said:
That processor is sold now as the Lake Contour. Lake Technologies is from Australia. They originally developed it for Clair Brothers (who I work for) as the Clair iO. It is the most sophisticated audio DSP for loudspeaker control available. It costs about $3400.00. For home use it might be a bit overkill though.
I have modded my DCX2496 after listening to it stock for about 10 minutes.. to verify it "....was working before he took it apart".
I am getting a Rane unit in on evaluation to play with as well. Except I can't do more than look inside the Rane as it is a demo, straight from the distibutor...
I am also running the unit right now via digital coax fom a NAD 5000 CD player. The pins 1 and 3 are shorted at the RCA jack and run to their prospective places on the XLR end. The opposite for the output cables, where they are open at the XLR end as well as shorted at the RCA end. (the ground and the neg signal wire are shorted to ground at the XLR end and both are ground at the RCA. Pin three at the output XLR remains open and unloaded)
I am getting a Rane unit in on evaluation to play with as well. Except I can't do more than look inside the Rane as it is a demo, straight from the distibutor...
I am also running the unit right now via digital coax fom a NAD 5000 CD player. The pins 1 and 3 are shorted at the RCA jack and run to their prospective places on the XLR end. The opposite for the output cables, where they are open at the XLR end as well as shorted at the RCA end. (the ground and the neg signal wire are shorted to ground at the XLR end and both are ground at the RCA. Pin three at the output XLR remains open and unloaded)
Thunau said:
yeah thats the one, i've seen/used it last summer in italy/swiss (with clair i4, as guest eng.).
Nice pocessor, controlsurface with dedicated dsp units... i believe an answer on the bss varicurve surface/host system....
But why is it too much for hifi? some hifi lunatics spent their money on expensive silver-left-turned-moonray pre eldered capacitors
, or pre 70's technolgy tube-amps (however, they do sound nice), this is just as crazy to me! And probably, if you talk to your boss you'll get it for a dealer price....
All off topic, sorry guys....
But I will follow this topic very close, maybe i'll buy the behranger (pronounced in chique french
) too, it costs nothing, nice to experiment with filters, maybe before building the final filters analog.pooge?
Isn't anyone else had their unit completely apart 6-7 times in the first few days, cuz they's rebuilding it, like me? anyone?
Heck, I'm going to be usng mine with tube gear in terms of analog in, so I have to re-do the input impedence to 47k or higher from the stock 20k. My MFA only has 6DJ8's on the output and can't drive that 20k input impedence without noticable limits and distortion.. The unit is very likely to never see any pro gear thrown at it - so swapping up to 47k isn't going to be a problem. (it's easier to swap the resistors than it is to fix my high output-high current SS preamp....)
Poogeing is good for you, it'll put solder burns on your chest!
Isn't anyone else had their unit completely apart 6-7 times in the first few days, cuz they's rebuilding it, like me? anyone?
Heck, I'm going to be usng mine with tube gear in terms of analog in, so I have to re-do the input impedence to 47k or higher from the stock 20k. My MFA only has 6DJ8's on the output and can't drive that 20k input impedence without noticable limits and distortion.. The unit is very likely to never see any pro gear thrown at it - so swapping up to 47k isn't going to be a problem. (it's easier to swap the resistors than it is to fix my high output-high current SS preamp....)
Poogeing is good for you, it'll put solder burns on your chest!
To Jan, Rickren, Awolf and other DCX2496 owners
May I repeat Morten question?
---------------------------------------
Measurements done by this guy:
http://users.pandora.be/airborne/dcx.htm
suggest that the high pass filter levels out at - 40 Db in the mid and bass range.
Can anybody confirm/contradict these measurements?
---------------------------------------
Thank you.
May I repeat Morten question?
---------------------------------------
Measurements done by this guy:
http://users.pandora.be/airborne/dcx.htm
suggest that the high pass filter levels out at - 40 Db in the mid and bass range.
Can anybody confirm/contradict these measurements?
---------------------------------------
Thank you.
Re: pooge?
KBK,
I am also planning to rebuild mine; do you have anything like a schematic for it? Even handdrawn?
Jan Didden
KBK said:Isn't anyone else had their unit completely apart 6-7 times in the first few days, cuz they's rebuilding it, like me? anyone?
Heck, I'm going to be usng mine with tube gear in terms of analog in, so I have to re-do the input impedence to 47k or higher from the stock 20k. My MFA only has 6DJ8's on the output and can't drive that 20k input impedence without noticable limits and distortion.. The unit is very likely to never see any pro gear thrown at it - so swapping up to 47k isn't going to be a problem. (it's easier to swap the resistors than it is to fix my high output-high current SS preamp....)
Poogeing is good for you, it'll put solder burns on your chest!
KBK,
I am also planning to rebuild mine; do you have anything like a schematic for it? Even handdrawn?
Jan Didden
Bare feet runnin'
I don't have a schematic, I do what I have always done.... by the time I've figured out how to keep the unit operational, while being fully modified, is the time I have finally figured out how it is constructed. All the chips and their suggested layouts are on the net. I just went and downloaded the tech books for them and started wire tracing via my multimeter. The rest of the circuit is standard buffered-balanced I/O and PS. Nothing special. 99.99% of all manufacturers of equiupment tend to stick to the stock implementation of chips suggested by the manufacturers, which is a total shame. Most will perform much better if handled/supported better, as all of us here know. The CS chip is a XO/frequency divider and is extensively used, it appears, so the PS of it is critical, one would suspect.
I had to put os-cons to decouple the A/D and the CS chip on the chip pins directly, as there was no other place to put them. I did that at 2 am this past morning. Warning! Don't have a few beers first......
I now have the RANE 26X multi-processor/computer controlled unit in for eval from the distributor. I will do a stripping of the unit to see what is inside, and then compare the two. A stock Rane against the modded Behringer. Rane makes quite a bit about it's in-house digital audio experts (at least the way things were voiced to me) and algorithms being critical, not the 96khz vs the Rane's 48khz. Due to the 96khz I/O, I believe the modded Behringer will show as being better overall, but we shall see. I have to make cables for the Rane, as it is bare wire connections. It is 8 in, 8 or 12 out, I believe. Some crazy number like that.
I don't have a schematic, I do what I have always done.... by the time I've figured out how to keep the unit operational, while being fully modified, is the time I have finally figured out how it is constructed. All the chips and their suggested layouts are on the net. I just went and downloaded the tech books for them and started wire tracing via my multimeter. The rest of the circuit is standard buffered-balanced I/O and PS. Nothing special. 99.99% of all manufacturers of equiupment tend to stick to the stock implementation of chips suggested by the manufacturers, which is a total shame. Most will perform much better if handled/supported better, as all of us here know. The CS chip is a XO/frequency divider and is extensively used, it appears, so the PS of it is critical, one would suspect.
I had to put os-cons to decouple the A/D and the CS chip on the chip pins directly, as there was no other place to put them. I did that at 2 am this past morning. Warning! Don't have a few beers first......
I now have the RANE 26X multi-processor/computer controlled unit in for eval from the distributor. I will do a stripping of the unit to see what is inside, and then compare the two. A stock Rane against the modded Behringer. Rane makes quite a bit about it's in-house digital audio experts (at least the way things were voiced to me) and algorithms being critical, not the 96khz vs the Rane's 48khz. Due to the 96khz I/O, I believe the modded Behringer will show as being better overall, but we shall see. I have to make cables for the Rane, as it is bare wire connections. It is 8 in, 8 or 12 out, I believe. Some crazy number like that.