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Reference DAC Module - Discrete R-2R Sign Magnitude 24 bit 384 KHz

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I had once a paper some decades ago, who claimed ringing (pre & post overshoots) below 1/100 of the peak will not be hear able...

May that was the point why Wadia with the spline oversampling is so nice listen to..

Hp

-80dB is the figure cited in the AES papers I've seen.
Not sure it's an entirely positive to chase this with 44.1kHz as the trade off is imaging above nyquist.
 
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I share your excitement about this DAC. With one of the filters I was trapped in the illusion I would listen to open reel tape. Never before I had this experience from a digital source. But for studio use it would need two more things: the filters shouldn't change the phase at all or each filter should tell in which way the phase has been shifted. The same is true for the amplitude at certain frequencies. Otherwise one has to fine tune with each filter change the sweet spots for making decisions for the mix.

I appreciate the hard work of the filter-brewing-crew but would wish that each filter could be tested with at least one example of classical or jazz music (3-4 instruments) which has been recorded in a natural acoustic environment.That would give a lot more insight regarding the phase-frequency relation. Different recording should sound different regarding the space in which they have been recorded.

After testing all output options (except transformer coupled) this DAC gets my highest recommendations only with a tube output stage.

What tube output stage did you use?
 
I have to stress that I've been developing the filters on what is essentially an "out of the box" DAM setup. I use a 7.5VAC transformer in a external case hooked to the DAM by an umbilical to reduce the possible influence of radiated mains interference, and run direct from the XLR outs to a Hypex UcD180 amp.

Hi Paul,
Have you tried the stock SE output of the DAM ?
Many report that the SE output sounds better than the buffered balanced output.
Regards,
Danny
 
What tube output stage did you use?

I use a cathode follower with 6H6p tubes (cathode resistor 200 ohms). I recommend to try different values for the L-pad . The final sound depends a lot on this especially with the option to use different filters and 2 fixed volume settings in the case of the dam DAC. Normally one would use values like 30k in series/10k to ground or variations of the in series resistor’(cap is 1uF/400V). I got the best results with 10k in series and 4.3k to ground. I experienced exactly the same general effect with the Buffalo III DAC (really nice but nothing special) which I use together with the same output stage. I cannot tell if these L-pad values are the only reason for the extremely liquid, powerful, balanced, analog-like sound in the case of the R2R DAC as I use both DACs together with a miniSHARC for the volume control which allows me to bypass the DAC’s volume control. This sound character is less obvious if the DAC is set to 0dB instead of +15dB and the mini SHARC volume control is close to the max. So it is a combination of different factors as certain filters also sound quite different with 0dB and +15dB settings. I didn’t try how it sounds with the DAC volume control ON and without the miniSHARC.
 
It definitely sounds better. A lot better.

This. There is no contest. Upon first trying the buffered outputs (always used unbuffered from the beginning) I ended up removing all the components for this section as they were just excess baggage I knew I would never use again.
I plan on placing a board directly above this now unused area for my relay shunt circuit I will be working on.
 
Disabled Account
Joined 2005
Hi Paul,
Have you tried the stock SE output of the DAM ?
Many report that the SE output sounds better than the buffered balanced output.
Regards,
Danny

I have an aversion to SE.
Its fine for hooking up a TV, but that is about it imo.

Once Søren releases a firmware update that will allow dual mono I switch across but until then...
 
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I have an aversion to SE.
Its fine for hooking up a TV, but that is about it imo.

Once Søren releases a firmware update that will allow dual mono I switch across but until then...

But since the signal is originally SE and is "in a bad way" converted to balanced, you should at least try it - or put a transformer on something else like that on the output to convert to balanced.
 
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Now why would I spend $300 on transformers as a stopgap measure?

How many have listened to the buffer after a decent burn in? I had the DAM running for about 220 hours on a burn in track while I was away on holidays. It's in no way scientific, but It seemed to be smoother sounding after that treatment. Someone actually recommended giving the DAC 400 hours on pink noise,.

The DAM has also been powered 24/7 for the last couple of months,

Perhaps it makes no difference but tbh I don't hear the buffered output as being particular bad. The op amps used are widely regarded as being some of the best available..
 
I have an aversion to SE.
Its fine for hooking up a TV, but that is about it imo.

With due respect that is just silly.

Balanced connections almost inevitably involve additional and often compromised circuitry. Their popularity is due in a large part to the marketing noise in some corners of the audiophile market. In a domestic audio system they add nothing and usually take a lot away.
 
Disabled Account
Joined 2005
With due respect that is just silly.

Balanced connections almost inevitably involve additional and often compromised circuitry. Their popularity is due in a large part to the marketing noise in some corners of the audiophile market. In a domestic audio system they add nothing and usually take a lot away.

That is your opinion, which your are entitled to. Based on my listening experiences I disagree, but that is my opinion. After all IMO = In MY Opinion and means exactly what it says.

FWIW This is what Nelson Pass wrote in one of the BOZ article.

What the circuit doesn't do is as important as what it does; it does not amplify any portion of the signal which is the same at both inputs. Ideally it completely rejects the common input signal, and the quality of this rejection is referred to as the Common Mode Rejection Ratio (CMRR), which tells how much of the common input signal gets through.

Being that the noise picked up from the environment is usually common to both input lines, it is rejected at the input of the balanced circuit, and thus is much less of a problem. Actual home audio systems using balanced interconnects typically have about 1/10 the background noise and hum.

Another reason to use balanced preamplifying gain stages is that many high end DAC designs offer balanced outputs in which separate DAC circuits are used for each of the two phases of output. Using separate balanced DAC circuits reduces the random noise by 3 dB, the same as if they were in parallel, and reduces common noise by a larger figure. There is also the potential for reduction of distortion with such an approach, but to realize the full performance of these circuits, the gain stage following must have a balanced input.

My previous DAC was fully balanced with two pcm63p per channel, and it always performed far better running balanced than SE.

As I posted, I'm waiting on a dual mono update to the firmware at which point I'll run unbuffered balanced outputs. I've got little interest in applying special seasoning and sauces in the form of valve buffers, trafo, or whatever...

The only thing I've seriously considered doing and may still depending on how long we have to wait for the firmware, is that I'll possibly build up a THAT1646 unbalanced to balanced converter and see how that goes. I have a couple of chips sitting a box somewhere just a matter of rigging them up and building a power supply.
 
Spzzzzkt you have said that you don't want to start modifying the dac until you're done with the filter experiments. You think the filters are the most significant factor. I would agree. Yet many have stated that there is a significant improvement in quality of sound via the direct out connection, avoiding the inbuilt conversion to balanced. You have a cost free possibility there to better hear your filter experiments, and yet you refuse to try it. I'm sure I'm not the only one who doesn't understand that.
 
Looking at Douglas Self's list of advantages and disadvantages of balanced interconnects in his book on Small Signal Audio Desin
I see for the usual short wired and well controlled situation at home only the typically more noisy input as item of importance.

Moreover if you look at the output of the DAM of an single 1/(64*44100) second pulse you see that it looks different.
SE:
SO0.jpg
Balanced:
Balanced.jpg

There will probably be no impact at the audio frequency range, but, as I interpret the images, there is some smoothing of (unwanted) very high frequencies at the balanced output. You might see that as an asset or an fault of the balanced output, your choice.
 
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Joined 2005
I'm sure I'm not the only one who doesn't understand that.

I don't know why.

If you were doing any component comparison how would you do it?

Basically,

a) Listen to filter "a" to establish a baseline.
b) Switch to filter "b" and listen to determine comparative differences and assess whether the change is positive, negative, or a combination of the two.
c) Switch back to filter "a" and listen again and verify you are actually here the difference you thought you were hearing.
d) Switch between "a" and "b" until you have a good feel for the differences then decide if "b" brings an overall gain in quality.

The important thing is to limit the variables, and focus on differences between filters. The differences between two variants are usually fairly apparent, especially when you are using a small selection of reference tracks.

I've done comparisons with three other DAC's at various points, but it's not especially useful. The DAM with stock filters was pretty close in tonal balance to a NAD M51 a friend owns, and the other two DAC's I own are a lot darker sounding, but they also sound very dark against the M51.

More recently I've also been doing reality checks vs Audirvana+'s Izotope SRC upsampling to 352.8kHz (which bypasses FIR1) to make sure things are not going too far astray. Given that the Izotope SRC is probably the best that is readily available and I can switch between the internal DAM filters and Izotope fairly rapidly it makes a good reference for direct comparison.

So the focus is on comparative changes in a specific setup. Providing the setup is constant and only a single element is altered at a time - the filters in this case - I believe the results are valid. It does mean that I can't use feedback that I can't verify on my system.

Even if the method was completely flawed, I upload any filters I think are of interest and anyone who is even remotely interested can download and listen for themselves. I trust that those who do so are sufficiently intelligent to be able to make up their own minds what works in their system and what doesn't.

There is enough information in the filter threads for anyone build their own filters. It is a Do It Yourself Audio forum after all.