Musings on soekris Reference Dac Module

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Thanks for the great explanation! Explorations like these are made possible by people like you who bring in lots of outside knowledge. 44.1Khz/96Khz benefits from software upsampling too. Maybe try upsampling to 192khz in software, there could still be a noticeable benefit. And if you have a USB interface lying around I would urge you to try it for yourself. I believe there is a huge difference and it's probably due to the more correct and perfect filter.


If TotalDac is supposed to be the most natural and musical, this would be a huge step in that direction. Don't take my word for it, look at the theoretical improvements. I don't believe there is more distortion added by using software oversampling.

I recall Soren mentioning that the 6moon reviewers really loved the DSD mode in the lampanizer. The removal of oversampling artifacts might be the common cause here.

I think up/and downsampling is definitely a significant factor WRT to degrading sound quality. I deactivate real time oversampling in all music production / mixing etc. software because it changes the signal in a bad way. But I don't think it has anything to do with harmonics or spuria that clock in at - 160 db below signal level. I think what happens in the time domain because of the anti-aliasing filters is mostly responsible for the effect.

You can up/downsample a file in software and try to null it with the (inverted) original, there will something left that is actually audible, way above -160 db...
 
And how many taps did it calculate on?

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16383.


I replied with questions about possibility in increasing filter length and whether it might theoretically improve performance.
 

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I think up/and downsampling is definitely a significant factor WRT to degrading sound quality. I deactivate real time oversampling in all music production / mixing etc. software because it changes the signal in a bad way. But I don't think it has anything to do with harmonics or spuria that clock in at - 160 db below signal level. I think what happens in the time domain because of the anti-aliasing filters is mostly responsible for the effect.

You can up/downsample a file in software and try to null it with the (inverted) original, there will something left that is actually audible, way above -160 db...

Got it. I think my earlier fft comparator screenshots caused confusion. They were meant to show that the Izotope which had >1M taps do indeed perform better theoretically than SoX with maybe 500k taps max.

What's interesting is the extreme improvement in sound going from Paul's excellent (both theoretically and subjectively) EQHQv5 Apo with 2k taps to external sampling via SSRC with 16k taps. Increasing filter length/taps is probably the easiest way to get better performance. I would hope for 1M+, mostly because we can and it does matter. Afterwards, we can maybe do some filter optimization, but it requires quite a lot of expertise in signal processing. I remain convinced that the longer filters provide superior performance and bring this project much closer to what one would hope TotalDAC could achieve.
 
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I just don't have time to update the vref posts at the moment, so I've taken them down.

There are some issues with the sims of the vref, to the point I think it's safe to say any sim that hasn't been verified by real world testing should be treated with a healthy amount of scepticism.

What I've found is that the .noise sims tally up in terms of broad trends with the white noise testing of the vref. This is because the white noise is basically a steady state load, and the vref buffer is not subject to the same kind of load variation as with sine or square wave tests.

Once you start looking at more dynamic test signals the sims become significantly less accurate. It has been pointed out by zfe and others that the load presented to the vref by the shift registers and R2R ladder is very complex, so it's not so surprising that it's difficult to simulate dynamic loading accurately.

I'm finding with the low-res mod and 2700uF per vref that there is a loss of dynamics, which is one of the reasons I was looking at possibility of using staggered cap values, rather than 4 or 5 x single value. I want to take a look at the bc560/550 mod at some point to see if this helps, but I want to do before//after measurements to see what effect this mod actual has.

The other niggle is the "super reg" supply I'm using is still dumping crap into the ground of the DAM at low level despite having replaced virtually everything except the transformer. The noise is high enough that it just breaks through above the level of ripple on the vref when playing back white noise. For the square and sine tests, the noise is actually a dominant factor in the vref posts so makes it hard to see what is going on. I'm still clinging to the faint glimmer of hope it's my build not the design, but I'm running out of part replacement options.

Not sure if anyone remembers this post. Soren, you could be right that simple simulations may not reflect reality in this case.
 
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Thanks Sören! I hear great differences between caps in this position. I need to try without then also. No playing: RS PRO Polystyrene Capacitor 470pF 63V dc ±1% Tolerance Through Hole 7.5mm diameter | RS Components
with MoreDAMFilters latest -3,5dB version.

Playing: Mahler: Symphony #4 In G - 2. In Gemächlicher Bewungen, Ohne Hast

Exciting, involving, real, detailed, dark as in reality, resolved, clean. Wonderful.

Best yet.

//

You need to try the software oversampling.... and then tell us if the caps still matter....:eek:

I suspect you're not hearing a problem with the capacitor - I've noticed that removing (or reducing) caps around the feedback loop of an opamp driven from a DAC (so subject to fast edges) also improves the SQ, reducing harshness. Its more than likely due to the cap's loading on the output stage of the opamp and hence the opamp's power supplies, not that the cap itself isn't transparent enough - NP0 caps are about as close to perfect caps as its possible to buy.

Tbh idk if there's much harshness left to be reduced. At this point I would take attempts at reducing harshness very carefully and only implement a change if there's theoretical justification. The need for smoothness and sweetness can overcompensate and produce dullness instead. If you want perfect smoothness, try Soren's botched 1.20 beta filters. :)


That is an interesting observation.

I decided to remove the 470pf NPO's from the output buffer feedback loop just to see if it made any difference. I think there might a subtle improvement in terms of detail and air, but it's pretty minor. It's possibly a bit smoother too but as vuki observes, without an unmodified DAC to compare with I could be just imagining things.

Sounds like no need to worry about those anymore...



I expected a diagram of the DAC to better understand the operation but Soekris is very busy and has been explaining the DAC in numerous interventions. I have made a selection of this to understand. I made the diagram that I could understand about the initial design. The current limits without changing architecture are:

Up to 1016 tabs at 44.1K/48K input sample rate.
Up to 508 tabs max at 88.2K/96K input sample rate.
Up to 252 tabs max at 176.4K/192K input sample rate.
124 tabs at 352.8K/384K input sample rate, but normally bypassed.
FIR2 is operating at 2.822M/3.072M and can have up to 120 tabs.

To better understand the level of DAC DAM1021 I have to meet other products on the market that have been mentioned in this thread..

MSB has one pass filter at 16x (705.6 Khz.) with 3200 taps. Or 32x (1.4112 Mhz.) With 6000 taps. The filter have noise shaping. The accumulators have 80-bit. Not round the result before adding next sample, intermediate data is stored at 80 bit resolution. Finaly convert to 24 or 26 bits including ultrasonic dither and noise shaping to have in sonic frecuences more than 24 o 26 bit resolution.

Ayre has one pass filter at 26x (1,1466 Mhz.) with 64-bit accumulators.

Dam 1021 have 2 or 3 cascade filters 2 FIR and 1 IIR. (64x 2,822400 Mhz.) with 67 bit MAC accumulator. Unlike the rest of DAC allows to manufacture their own filters, perform low pass, bandpass, highpass for speakers, Room Correction. Very interesting for diy builders and experiments in the forum community.

In case it's useful. I found it interesting that Soren chose 353/384khz as the intermediate frequency, any higher and we wouldn't be able to fully bypass FIR1 with our current I2S max sampling frequency.
 
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I have no further worries about the hardware at the moment. Unless someone discovers an easily correctable problem with the dam1021, the design (with added vref caps, otherwise bass appeared way too thin and punchy) seems quite perfect to me now. Of course, adding more caps and improving various power supply might still help, but probably not by a long shot. I think further evidence of the efficacy of various advanced mods is needed, with the software OS implemented (or whatever improved version we might discover of it), for us to know its effectiveness. Maybe 1121's reclock design can make a difference, but there's no way to change it on the 1021 now. Is better optimized and more powerful software oversampling the new frontier?


Is there an easy way to test objectively and subjectively the difference between, say 16k tap and 500k tap in SoX? Also, it's sad, marginally, that my DTS tracks now sound unbearably harsh... Dolby Atmos only supports up to 16/48 - there probably isn't a way to edit the windows audio graph pipeline to add an upsampler at the end. Though Soren's ultra-soft 1.20 beta filters did sound quite good. For the movie-lovers :)
 
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The DAM is internally also upsampling by default,
by using an upsampling plugin like Resampler-V or Resampler(Sox mod2) in Foobar
we just move the upsampling from the DAM FPGA to the PC that has much more memory and processing power to do this.
 
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Total Dac is a no over sampling dac...did you try that?

NOS by itself is no better than EQHQv5, at least not by much if any. This is completely different. Also total DAC does use oversampling to compensate for high frequencies. Regardless Soren doesn’t believe in NOS and it does make sense objectively.

Yes. Only some people interpret this as a significant improvement.

Add to this 10M taps and the attraction becomes irresistible :)

How do we get 10M? That would be an awesome next step.

Also does your graph show that we’re bypassing fir2 as well? Would that be advisable?
 
I installed iZotope RX7 advanced but the SRC doesn't seem to work as a VST2 plugin... any ideas?... Also no one publishes their internal filter length figure...

And since Foobar2000 only supports 32bit VST2 there's no way it'll work with 64bit iZotope SRC even if the latter works as a plugin form somehow


Sox can only set one target output frequency it seems, not so convenient. Hopefully they implement a x8 in their next version. It's probably safe to assume that resampler-v uses highest quality setting in sox...
 
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The signal being compared to has to be simulated in order to be measured, I guess that's what I meant. I use Linux full time so not entirely familiar with RMAA and the variety of windows based tools. The tests have to be well controlled and repeatable. I've seen others here using ltspice and whatnot to make/measure test .wav files.

I reposted a statement from Paul just last night in which he said the simulations are likely flawed. If we want real measurements it should be on the vref ripple?.. not easy to get to uV precision. Soren said he tested transistor mod but he didn’t like something about having to feed larger currents to keep impedance low. For the caps, I still think larger and lower ESR is better, seems that there is no evidence against it (in terms of absolute performance, frequency balance is a different thing). The best we can do is maybe find out how much location could matter and what capacitance is enough.

Have you tried software oversampling? It’s not on the same level with NOS or Paul’s filters.
 
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