• These commercial threads are for private transactions. diyAudio.com provides these forums for the convenience of our members, but makes no warranty nor assumes any responsibility. We do not vet any members, use of this facility is at your own risk. Customers can post any issues in those threads as long as it is done in a civil manner. All diyAudio rules about conduct apply and will be enforced.

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

When I set the volume over V+01, I can hear some noisy sound such as crushed and squashed.
If the volume is set to below V+00, I can’t hear any noise.

While I don't understand how it could be related to volume adjustment on the digital side. This makes me think about problems with power supplies : ground loops or transformers pollution. Could it be a modulable loop via the potentiometer ?

Looking at your nice little box, I'm surprised to see the transformer and PSU that close to components on the audio path.
 
Hi all.

Have been listening my unit for 1.5days now. I made simple regulated lps for it and feeding JLsounds i2s in. Housed in a sloppy temporary metal box. I will post pictures once i have better psu-s and nice case.

I was hesitant initially, but this dac at the price and effort is just god send.
I was expecting something much, much worse. This is no toy.

The Dam with its stock filters utterly shamed my BMC Puredac. The Puredac was the best dac i had heard thus far out of many fairly expensive units. My experience is still limited as i have not heard Totaldac or MSB stuff.

With the latest 1021filtNQ_C128x100 filter its performance got even better. For my tastes and in my system it outperforms any source i have heard by a hefty margin. With this filter the sound is effortless, but dense and hefty at the same time, very nice.

It is not sounding like digital source, it is sounding like a good vinyl rig with perhaps better clarity and maybe some minor flaws, i have to listen more to tell. With stock filters it sounded very reminiscent of vinyl in good and bad.

I have no way to test the unbuffered outputs with Genelec monitors, but even through the buffer it is sounding really good. I got worried by some earlier comments.
Still, unbuffered to headphone amp (Sansui integrated) and HD800 feels even more natural.

Thanks Soren and all you filter developer guys,
I'm a happy camper after so many years.
 

Attachments

  • rear_panel.jpg
    rear_panel.jpg
    83.8 KB · Views: 987
While I don't understand how it could be related to volume adjustment on the digital side. This makes me think about problems with power supplies : ground loops or transformers pollution. Could it be a modulable loop via the potentiometer ?

Looking at your nice little box, I'm surprised to see the transformer and PSU that close to components on the audio path.

Thanks for your comment.
I'll check any problems with ground loops.
 
Just popped back from the Filter Brewing thread to see what may have developed re firmware or other ideas for modifying and improving. Seems to lots of chat about measurements and whether this dac is "better" then that dac... define "better"?! I tried measuring 'our dac' with my Tek TDS 2040 which until now has proven itself to be a decent piece of kit. I was unable, unfortunately, to measure anything I thought meaningful as the noise floor on the FFT plots fell about 70-80dB down regardless of how I measured, even with the probes to ground. Thank goodness, he says. I have since been measuring with my ears!!

So, I've no idea how our dac measures, how each filter measures, how much it rings, etc. etc. What I do know however, is it sounds and how the filters impact sound, and which I most prefer. Surely, that's what this whole project is about, our working together to make a dac which *sounds* 'as good as it gets' rather than beats the pants off every other dac out there on the scope? Keep up the good work one and all, I'm heading back over to the dark side of filter design where the talk is mostly of sound quality and how the measurements relate to sound.
 
I found out that i could send SE to my monitor speakers, wow.

The on-board buffer really limits what this magnificent device can do. The texture and low level information is spooky, i never knew my Genelec-s could do those things.
I now believe ;)

This dac could be really useful in recording business. I have never heard so much differentiation in bass.
 
I found out that i could send SE to my monitor speakers, wow.

The on-board buffer really limits what this magnificent device can do. The texture and low level information is spooky, i never knew my Genelec-s could do those things.
I now believe ;)

This dac could be really useful in recording business. I have never heard so much differentiation in bass.

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.
 
Disabled Account
Joined 2005
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.

The filters are always tested against two reference tracks.

First is "Bonita" from Morelenbaum^2 Sakamoto's "Casa".
If you've never heard of this disc you are really missing something...
Morelenbaum2/Sakamoto Casa: (Tribute To Jobim) Review By Jim Luce
It's beautifully recorded, vocals cello and grand piano. The intro to Bonita also has a field recording made at Impenema beach of waves gently breaking on the sand in the background.

The other is "So What" from Miles Davis' Kind of Blue.

Another reference I use to double check imaging if I have any doubts is a field recording of a dawn chorus made in northern NSW back in 2009. I have a fairly good idea of what was going on as I was standing about 4-5 metres behind the mic rig.

https://soundcloud.com/offtracksound/dawn-on-bennets-gorge-track

The download link for the 16/44.1 wav should be enabled on this.
It's not perfect by any means but I find it a good test of accurate reproduction of ambience, detail and "sound staging".


I listen to a variety of other tracks because in the real world music isn't just about making audiophiles feel impressed about the abilities of their systems. Some old pressings are particularly problematic - ABC's "Lexicon of Love" original pressing has typical 80's low peak levels and has a fair bit of sibilance in the recording - it's difficult to get everything balanced so it sounds right. I also listen to a fair bit of electronic music, and again that is something that presents a different challenge in that the music tends have far heavier bass emphasis than acoustic recordings. The danger is that just listening to and adjusting to bring out the bottom end, and ambient detail in acoustic audiophile grade recordings will result in overblown bass and screeching treble on recordings that are not designed specifically to flatter your system.

If I remember right you are using miniShark DRC prior to the DAM. You should have a read of the DIRAC paper on room correction from the MiniDSP website in particular the problems caused by minimum and intermediate phase DAC filters when using DRC. I suspect most of your issues with phase arise from this.

cheers
Paul
 
Last edited:
If I remember right you are using miniShark DRC prior to the DAM. You should have a read of the DIRAC paper on room correction from the MiniDSP website in particular the problems caused by minimum and intermediate phase DAC filters when using DRC. I suspect most of your issues with phase arise from this.
I think it’s one of those things that cannot be explained easily. It’s really no big deal and I probably never would have found out about it if I hadn’t had a chance to compare so many different filters while using the same DAC. I expected a different overall ‘character’ from each filter but somehow no effect on the subtle information about the venue . Maybe it’s the neg. effect of the filters of the DRC or it’s because of the positive effect of more clarity with the DRC that I can hear these things? If I bypass all DRC filter settings there is no change regarding the subtle information about the acoustic character of the venue in combination with DAC filters except that everything sounds less present and clear, impulses sound a bit more ‘smeared’ etc. exactly according to the acoustic character my listening room adds to everything I play without correction. I lose all positive effects of DRC but don’t gain anything .
I use the miniSHARC only for minimal filtering (PEQ-IIR filters) in combination with acoustic room treatment. I don't use DIRAC live and FIR filters and I am aware of phase issues with filters. Moreover I don’t have x-over phase issues as I use single driver speakers.
It could be possible that it’s simply the changes in the frequency/amplitude of the filters that superimpose its effect on the more subtle information and appear only acoustically as changes of certain ambient information of the recording while it is nothing more than the natural effect of the amplitude/frequency settings of different filter and in this way it would be ok and should be expected from filter changes.
Logically and empirically I have based my conclusions on the fact that if I use analog as source and use ADC/DAC with DRC (I used Behringer for this test) the character of the analog recording changes with every recording according to the different character of the venue of the recording (very obvious with classical music). If the ADC would cause this effect - which I cannot avoid with digital sources as that has been done already in the recording studio - it would add a similar effect also to my analog sources. While my speakers position and all other settings are the same, the illusion of a 3D room where the recording took place changes with the recording. With the R2R DAC/different filters this information also changes from recording to recording but much more subtle and always embedded in the overall character of the used filter. And maybe that’s just how it is and nobody comments on it as normally a DAC doesn’t come with the option to change filters back and forth. I can live with it.
 
Disabled Account
Joined 2005
Logically and empirically I have based my conclusions on the fact that if I use analog as source and use ADC/DAC with DRC (I used Behringer for this test) the character of the analog recording changes with every recording according to the different character of the venue of the recording (very obvious with classical music).

The filters that are currently being worked on are nowhere near neutral. The C128 series are prioritising very short filter length and very slow roll-off to minimise ringing, and uses a Nyquist filter algorithm which is the only one I've come across that preserves the original sample data.

The main reason this is being pursued is that these filters seem have a NOS like sound while eliminating some of the potential issues from unrestricted imaging at higher frequencies, especially with switching amps.

That said I hear differences in the recovery of the acoustic space in recordings between filters even in the relatively minor variations of the C128 series. My classical collection is pretty limited, so the recordings I have tend to have fairly gross differences in acoustic ambience. The acoustic space of Tavener's Ikon of Eros is clearly different to that of Academy of Ancient Music recording of Mozart Mass in C Minor K427, and to Kronos Quartets White Man Sleeps or Steve Reich's Different Trains.

Things like Laswell's Means of Deliverance also have a real sense of the acoustic of the recording space. It might not be to everyones taste, nor particularly well know outside his immediate fan base, but it is very well recorded.

There is a great article on the studio and the recording session up here:
Bill Laswell’s “Means of Deliverance” — Making a Groundbreaking Solo Bass Record | SonicScoop - Creative, Technical & Business Connections For NYC?s Music & Sound Community

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.

The position I'm taking is that the filters need to sound good on an unmodified DAM first and foremost. There is just too much variability in the tweaks and mods that people make for me to chase down anomalies based on any one report, especially when I'm not hearing the issue in my system.

cheers
Paul
 
The filters that are currently being worked on are nowhere near neutral. The C128 series are prioritising very short filter length and very slow roll-off to minimise ringing, and uses a Nyquist filter algorithm which is the only one I've come across that preserves the original sample data.l

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