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Behringer DCX2496 digital X-over
Behringer DCX2496 digital X-over
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Old 27th June 2008, 03:09 PM   #1741
revintage is offline revintage
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Ulli:

I am totally aware of how the DCX2496 works.

It was the volume control by squarewave I was wondering of.

But as you say it must be the same whatever digital volume you use before the DCX.

I am also temped to use Jans active, just to get the remote volume control, but with my Lundahls both in and out in place of the OP-amps.

But for me the ultimate solution is to use Silk TVCs on the output. I have a custom made remote-controlled Shallco 24-pole switch with 4 decks. Maybe one could add two more decks.
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Old 28th June 2008, 06:23 AM   #1742
squarewaves is offline squarewaves  United States
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Hello again, thanks for the single ended output information on the Selectronic piece. I've been leaning towards the pieces from Pilgham's site simply out of familiarity, I've seen the schematics in an audioxpress magazine article. I haven't seen exactly how the Selectronic's I/O board works. Also the information on the active I/O kit from you, Jan is helpful. I like the way your passive I/O board uses the set of relays for mute and that it removes many op-amps from the signal path. Lars, I admire your use of transformers on the outputs, but I'm afraid it would be too big of a leap for me. I need to try an intermediate step before I get to that point. (hope that makes sense) :/

Ulli, As I understand I only need one Vreg that I would use to regulate the voltage to the DACs? And you have two because you use one on the ADCs? That's the best I can figure at the moment. Before I order, my question is since I'll most likely never use analog input, Do I get by with just one Vreg kit? Thanks.

About me: I am most likely not as experienced with the highest of fidelity sound (in terms of owning the equipment to produce such sound and the experience of listening to said equipment for long periods of time) as many of you here are. I am probably more tolerant of sound that is less than high end "audiophile" quality, as most of the people here but I'd say I do fall into the audiophile category as a person simply because I concern myself with musical playback accuracy to a greater extent than does the general population. I do appreciate accuracy and clarity but the little kid in me still likes to play it really loud every now and again. And on a final note, I'm no engineer by any stretch. I'm just a lowly tech. I have a two year degree in electronics from about 20 years ago.

The floating point hardware volume control, kmixer.sys, and kernel streaming isn't as glorified as it sounds. It simply means that I bypassed windows low resolution method for mixing system sounds and media sounds together. When the music is playing on my setup nothing else in the system can access the sound-card. In fact the volume control sliders in Windows Volume Control panel have no effect. Nor does the volume slider control on the media player. I must use the M-Audio software sliders to control the hardware (sound-card) that adjusts the volume of the sound. It adds a little complexity to how I listen because I have to use two applications. A volume control application to control sound level and the media player application to select what song I want to listen to. Kernel streaming is a plug-in for the media player. It sends the audio stream directly to the sound-card for the most direct path possible. The volume is done in the digital domain at 36 bit resolution via hardware DSP on the sound-card. There are no DA to AD conversions along the signal path. There is only one analog conversion and you know where that is. The DCX DACs.

Do I lose resolution at attenuated levels? Most likely, but there's a small chance that I don't. If I listen to 16/44, there probably isn't because the DSP uses a greater bit depth and so does the DCX; thereby carrying the higher resolution through to the end. (this might not be true in reality, I'll explain later) If I play 24/96 source material then I would say I do lose resolution at attenuated levels. But I can't hear it. You guys might but I can't. Most of my recordings are of CD quality. I might have one or two samples of HDCD or a clip from DVD-A but most of my source material is recorded 16/44. I set this up about 4 years ago and been listening to it without any further modification since. Now upon discovery of the mods that are available for the DCX, the curiosity has gotten the best of me. I want to venture into the next level of the sound listening experience.

Lars, and anyone else interested. I have a theory that I do lose resolution by using digital volume control but it could be possible to implement a non degrading method. If the source audio (for example 16 bits) is converted and then attenuated (digitally) at a bit resolution higher (36 bits for example) than that of the original and maintains that resolution to the end (use a 36 bit DAC), I do not believe it would lose resolution to a point to where it got worse than the source because the divisions in the scale have become finer. (36 bits vs 16) However, this is not what happens in the real world device. I would guess even though I have hardware that resolves at 36 bit detail it probably reverts back to 16 bits for transmission to the DCX. In this case I would conclude the system causes a degraded sound. If the sound-card were to revert only to 24/96 and the DCX maintain it then I would guess I am not losing much since the output resolution would remain higher than most of my source material. (16/44) I can conclude that a digital system with an output signal that is the same resolution as its input signal and where attenuation is applied digitally, you end up with degradation based on this simple truth. The "step size" or "divisions of the scale" of any particular digitally quantified system is fixed. The smaller the signal the fewer steps there are to represent that signal. By step size I mean the smallest difference possible in amplitude change at the output of a DAC when the digital input is changed by the smallest possible digital value in decimal notation: one) I'm not at all an expert and there may be sides to it that are unbeknownst to me or that I'm overlooking, thereby not enabling me to reach a full logical conclusion. But based on what I think I know... that's it. And like I said I've listened to it this way for years and have been quite content. I cannot hear any degradation. It sounds clear at low volume as well as high. But then again, maybe, my ears could be made of tin rather than gold. Or that my speakers, amplifiers, and wiring isn't the greatest in the world. There is a possibility that a floating point process can change the amplitude in a non degrading manner but I don't know. To me if the output is converted back to the same resolution as the input, regardless of how the digital attenuation is achieved, there will be some form of degradation, but I simply don't really know. Discussions can be found by searching the net for "floating point vs integer sound quality". I would consider RaneNote Fixed-point vs. Floating-point dsp reliable.

I'll post back on my progress with the kits and try not to write a book next time. Cheers.
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Old 28th June 2008, 08:34 AM   #1743
jan.didden is offline jan.didden  Europe
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Quote:
Originally posted by squarewaves
[snip]Lars, and anyone else interested. I have a theory that I do lose resolution by using digital volume control but it could be possible to implement a non degrading method. If the source audio (for example 16 bits) is converted and then attenuated (digitally) at a bit resolution higher (36 bits for example) than that of the original and maintains that resolution to the end (use a 36 bit DAC), I do not believe it would lose resolution to a point to where it got worse than the source because the divisions in the scale have become finer. (36 bits vs 16) However, this is not what happens in the real world device. I would guess even though I have hardware that resolves at 36 bit detail it probably reverts back to 16 bits for transmission to the DCX. In this case I would conclude the system causes a degraded sound. If the sound-card were to revert only to 24/96 and the DCX maintain it then I would guess I am not losing much since the output resolution would remain higher than most of my source material. (16/44) I can conclude that a digital system with an output signal that is the same resolution as its input signal and where attenuation is applied digitally, you end up with degradation based on this simple truth. [snip] Cheers.

Squarewaves,

This is a very clear explanation, and I can find no fault in your reasoning. I agree.

For the sake of discussion, lets say you go to 20 bits. That means you would end up with, say, the values:

1010101010101010 (16 bit) -> 10101010101010100000 (20 bit)
1010101010101011-> 10101010101010110000

Now you do the volume control, you go to:

10101010101010100000-> 10101010101010100100
10101010101010110000-> 10101010101010110100

Now you can convert this to analog in a 20 bit DAC.
Have you lost resolution? I don't think so. Is this how it would work?

Jan Didden
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Old 28th June 2008, 09:09 AM   #1744
ttan98 is offline ttan98  Australia
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Default Intermittent Fault

I have been using the DCX 2496 over the last few months, it has been playing faultlessly.

Only yesterday and also today, after playing for a 1-2 hrs it starts to play up. It stops playing, no display signal/led display on the front panel. After I power off and on again, the units returns to normal operation.

I have not opened up the unit yet, I suspect the cable connections could be the main suspect. Rejiggling the cable connections to the main board may solve the problem. If it does not solve it there could be a "near" dry joint somewhere and it becomes apparent after it the unit heats up.

Anyone experiences similar problem? Any suggestions to fix this problem?

Thanks in advance.
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Old 28th June 2008, 09:14 AM   #1745
jan.didden is offline jan.didden  Europe
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Well, if it stops completely, it's probably not just a single contact or cable. Changes are it's the power supply.

Jan Didden
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Old 28th June 2008, 10:36 AM   #1746
modmix is offline modmix  Germany
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Behringer DCX2496 digital X-over
Quote:
Ulli, As I understand I only need one Vreg that I would use to regulate the voltage to the DACs? And you have two because you use one on the ADCs? That's the best I can figure at the moment. Before I order, my question is since I'll most likely never use analog input, Do I get by with just one Vreg kit? Thanks.
I still listen to FM radio. That's why I still use analog input.

Yes, one is for the DACs and the other one is for the ADC.
From the circuit diagram I read:
IC7 feeds +5VA-1 into both ADC chips (IC2 5393 / IC3 5983)
IC8 feeds +5VA-2 into three DAC chips (IC4-6, all 4393)

The picture on page 1 of http://www.linearaudio.nl/Documents/...inSup%20CG.pdf shows just one Vreg - the one for the DACs.

Ulli
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Old 29th June 2008, 10:39 AM   #1747
modmix is offline modmix  Germany
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Behringer DCX2496 digital X-over
Quote:
Originally posted by janneman
For the sake of discussion, lets say you go to 20 bits. That means you would end up with, say, the values:

1010101010101010 (16 bit) -> 10101010101010100000 (20 bit)
1010101010101011-> 10101010101010110000

---snip--- Is this how it would work?
Jan,

I agree in general. But, is that the way conversion is done?
Isn't it, that the analog signal (let's say 1V max) is represented by bits (16 or 20 ones, respectively). The analog range is the same in both cases (here: 1V).

I added values in decimal (which I'm more used to):
1010101010101010 43690
1111111111111111 65535
=> signal is at 2/3 of full range

2/3 of 11111111111111111111 1048575
would be 1048575 * 2/3 = 699050
10101010101010101010 699050
10101010101010100000 699040

This a delta of 0,0014 %.

Of course, it fully depends on how conversion is done.

Ulli
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Old 29th June 2008, 11:05 AM   #1748
Will is offline Will  Malaysia
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Quote:
Originally posted by AR2
And here re the Lundahls during the assembly in separate case that houses volume control and preamps.

PS I am just out of the door for the roadtrip, so I will not be able to answer phone calls for a week.


Hi AR2,
Could you please elaborate further your 6 channel preamp configuration; I'm assuming that you have the input section configured using balanced to unbalanced lundahl. What's after the lundahl? do you have an Alps 6 channel pot, and what's after the attenuator, an active output stage or is it just passively coupled directly to your power amp?


cheers.
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Old 29th June 2008, 11:09 AM   #1749
jan.didden is offline jan.didden  Europe
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Ulli,

If I understand your example correctly, you effectively multiply each sample by (2^20-1)/(2^16-1) (for 20 bits), right.

In my example, adding a bunch of zeroes at the end, is effectively multiplying each sample by 2^4.

I don't see that this is fundamentally different, except for some rounding of the '1' factor?

Jan Didden
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Old 29th June 2008, 05:26 PM   #1750
AR2 is offline AR2  United States
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Hello Will,
I wrote several time here about my set up, and feel little bit uncomfortable being redundant. I guess when the tread becomes as big as this one than we have to do it again for the newer readers. My apologies to people that are reading this tread from the beginning.

Here is the tread on volume control/preamps:
http://www.diyaudio.com/forums/showt...threadid=76095

Here is the tread about my speakers in order to connect the dots:
http://www.diyaudio.com/forums/showt...threadid=21523

I made some changes in the meantime regarding the amps. On the bottom I have my new DIY Pass Lab X600:
http://home.comcast.net/~burningamp/index.html
and on the top I have Aleph 30 that is closer in output to Aleph 5 (higher rail voltage)

So with all this different amps and speakers I had to have different gains in preamp stage. My whole original output section from DCX is gone. Lundahls are directly connected to AKM chips, balanced. Lundahls are balanced and than signal goes to 6 ch. relay volume control - balanced. After that I have XBosoz balanced preamps for bass channels and for high channels. Mid - tube amp do not need any preamp, so there is a direct connection between my 2A3 amp and volume control. For that I use switch to convert from balanced to single ended.

When I use DCX just as two channel test set up on different speakers I use direct output from Lundahls and volume control. That sounds better than any preamp when added, and that is what I prefer as long as your amp/speaker combination could work with that - if you have enough gain.

As you could see Lundahls are outside of DCX box. They could fit inside obviously without the original in/output board that I am not using anyway. My plan is to make PC board that will hold Lundahls in the DCX box and that will pass digital signal to the processor board. If anyone is interested we should combine the efforts.

I hope this answers your question.
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