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T+A DAC 200


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On 3/13/2023 at 5:45 PM, OE333 said:

 

 

This is indeed an intersting question.

 

First: the facts:

The DAC200 uses 4 PCM1795 converter chips for converting PCM signals to analog. The PCM1795 consists of a 8x oversampling filter followed by a ΣΔ DAC.

In case of the T+A devices the 8x oversampling inside the PCM1795 is completely bypassed and the digital signal is directly fed to the ΣΔ  modulator.

If one of the DAC200's PCM oversampling filters (FIR1, FIR2, Bezier1, Bezier2) is selected, the oversampling is done in an external DSP (using T+A's own oversampling algorithms) , the internal filters of the PCM1795 are never used.

 

If the oversampling is switched OFF and NOS1 or NOS2 mode is selected, the external DSP is also bypassed and the incoming PCM stream is directly routed to the ΣΔ  modulator stage of the DACs without any signal processing before the modulator stage.

 

So, as a first result we can state that in the DAC200 all PCM oversampling filters are bypassed in case of NOS mode.

 

The question now is: does the ΣΔ  modulator stage perform oversampling or not ?

 

This question is a bit philosophical and the answer depends on weather you regard the output signal of the ΣΔ  modulator as a highly oversampled 1 bit digital signal (in this case it performs a kind of oversampling), or if you regard the modulators output signal as an analog signal having an average value representing the analog output value - in this case the modulator is DAC delivering an analog output voltage which only needs some averaging (analog low-pass filtering) to get rid of the unwanted high frequency noise and to deliver the wanted analog signal average.

 

Some more details about this topic are given in this paper: https://www.beis.de/Elektronik/DeltaSigma/DeltaSigma.html

 

 

More than this consideration it might be interesting to look at the output signal of the DAC200 and find out if it behaves as would be expected from a NOS DAC.

 

 

1.) Outut signal from the I/V stage in NOS2 mode when the DAC200 is fed with a digital step signal @fs=44.1 kHz (blue trace):

 

 

Step_44_1_IV_stage_out.thumb.png.853bb223b7b2086121687f66e49e90b9.png

 

The output signal after the I/V stage of the DAC. It rises instantly at the digital signal step, without any trace of interpolation. A NOS R2R converter would behave exactly the same way.

 

Please note: This signal is the signal before the final analog reconstruction filter. It is an internal signal and can not be measured at the output jacks of the DAC200.

 

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2.) The same signal after the analog reconstruction filter, as it appears at the DAC200 output:

 

Step_44_1_DAC200_out.thumb.png.d65b3aae62d416f18b2c6c999aed608c.png

 

The output signal after the analog output filter slews a bit slower than the signal in the first measurement. This slower slew corersponds to the 120 kHz cut-off frequency of the DAC200's analog output filter (in "WIDE" mode) - it is NOT a consequence of digital interpolation.

 

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For comparison:

This is the DAC200 output signal with the same digital step input signal with oversampling filter "FIR1":

 

Step_44_1_FIR.png

 

 

This output waveform shows the much slower response due to the interpolation performed by the digital FIR oversampling filter.

 

 

 

The presented step measurements clearly show the Non Oversampling behaviour of the DAC200.

The output is exactly what would be expected from a NOS DAC and the presented output signals are indistinguishable from the output signals of a NOS  R2R DAC fed with the same digital input signal.

 

Imho the DAC200 is very well suited to operate with high quality external oversamplers such as HQ player, because inside of the DAC200 no oversampling/interpolation takes place and the quality of the externally oversampled signal is preserved in all detail.

 

 

From the discussion above, is the reconstruction filter also part of the burr-brown 1795 chip?

 

Also from the discussion, is it advisable to always oversample at least x8 (on a high quality external oversampler) in order for the reconstruction filter to properly remove high frequency aliased artifacts?

 

I assume you average multiple 1795 signals after the reconstruction filter to improve Signal to noise? Is this the case?

 

Any comments on this theory 
 

http://www.mlssa.com/pdf/Upsampling-theory-rev-2.pdf
 

it hypothesizes a technical reason as to why oversampling may sound better (but it is quite unorthodox)? 

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On 7/3/2023 at 12:56 AM, 1125FPS said:

 

I respectfully request Miska or anyone else explain the unique nature of DAC 200's DSD NOS modes.

 

 

The NOS mode in DAC200 is available for PCM signals only.

 

 

T+A Fellow   (Head of R&D @ T+A 1989-2021)

(*) My postings represent my private and personal opinion and hopefully are helpful to the members of this forum

 

T+A MP200 | T+A DAC200 | T+A A200 | T+A Talis S300 | DAW: Core i7 8700K - Linux 5.4.0 - Roonserver + HQP | NAA on RockPiE (RK3328)

 

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On 7/2/2023 at 10:02 PM, Shadorne said:

From the discussion above, is the reconstruction filter also part of the burr-brown 1795 chip?

 

Also from the discussion, is it advisable to always oversample at least x8 (on a high quality external oversampler) in order for the reconstruction filter to properly remove high frequency aliased artifacts?

 

I assume you average multiple 1795 signals after the reconstruction filter to improve Signal to noise? Is this the case?

 

 

The reconstruction filter in DAC200 is not part of the BB 1795. It is a discrete analog filter circuit following the D/A converters in analog domain.

 

If using an external upsampler for PCM my recommendation is to upsample 8x externally and use the NOS mode of DAC200.

 

The averaging of the multiple D/A converter outputs is done in the (discrete analog)  I/V stage following the converters. Purpose of this averaging is to minimize noise and converter distortion.

 

 

T+A Fellow   (Head of R&D @ T+A 1989-2021)

(*) My postings represent my private and personal opinion and hopefully are helpful to the members of this forum

 

T+A MP200 | T+A DAC200 | T+A A200 | T+A Talis S300 | DAW: Core i7 8700K - Linux 5.4.0 - Roonserver + HQP | NAA on RockPiE (RK3328)

 

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On 6/7/2023 at 9:21 PM, CharlieR said:

I hope you are patient, I just heard that they have been very busy with Munich and that nothing is leaving the factory until at least next Friday. I had placed my order on the 16th but I have no idea what place I might be in the backlog, they say patience is a virtue...

Finally, received my T+A DAC 200 (with HDMI) in silver. I must say it is very musical right from the first moment. Hope with more hours of burn-in it will be even better.

 

Currently trying HQP embedded trial which is running on my Antipodes S40 Server and using iFi Zen Stream as NAA.

But, the zen stream supports DSD only over PCM (DoP format). And also upto DSD128 only. I find this as a limitation.

 

can the experts here suggest a few streaming transports which support HQP NAA or any DSD upsampling streamers?

IMG_0735.jpeg

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Thanks for such a good post, learned a lot. Let me ask you a question, I use lumin u1mini, connect dac200 through AES, there is noise when playing dsd64 files, does the dop method not work? At the same time, I used another naa usb to connect to the dac200 usb interface and it was normal. Is it impossible for dac200 to connect usb and AES at the same time, and then use one of the interfaces to play dsd files? Please excuse my poor English.

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11 hours ago, Shadorne said:


I have had PS Audio DSD for a few years - nice smooth but rather veiled (sort of analog sound if you will) - I understand why folks love this DAC because typical entry level sigma delta DACs tend towards harshness or glare. The T+A D200 is a significant upgrade and will probably really get you excited again about your music collection. Redbook upsampled linear phase to DSD64 or higher is where the magic is at with T+A D200. I have eclectic taste and never cared for audiophile recordings so anything that makes the majority of the vast Redbook catalog sound great is well worth it, and T+A does just that, more than any DAC I have heard so far.

Hi Shadorne, I see that you have the DAC3, how does that compare to the TA DAC200?

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17 hours ago, SPAZ said:

Hi Shadorne, I see that you have the DAC3, how does that compare to the TA DAC200?


They are very close in terms of detail.
 

The DAC 3 images more between the speakers and in the plane of the speakers. The DAC 3 has a blackness (silence) which suggests a lower noise floor. I remember noticing this extra blackness over the DAC 2 and DAC 1 which I compared. This is most likely the ESS chip methodology to reduce noise through barrel rotator selection of any of the 64 x 1 bit sigma deltas built into the chip. The DAC 3 is slightly harsher/more dynamic in the 2 - 4K range on transients - plucking of strings is more noticeable. DAC 3 is slightly more mid bass heavy.

 

The T+A is thinner in mid bass and less harsh/dynamic in 2-4K range but without any loss of detail. Overall it is slightly more polite or analog/vinyl sounding but no loss of detail. The sound is most similar to DAC 3 using PCM Redbook input and the FIR1 filter (as it should be). The overall sound has a higher noise floor but I have more to say about this. Honestly with PCM Redbook the difference is not much at all - both sound great as they should. 
 

When upsampled to DSD64 or DSD256 there is a much bigger difference in the sound of the T+A D200 Vs the DAC3 or itself in PCM mode. (The DSD DAC in the T+A D200 is a completely separate beast). The sound is still thinner in the mid bass and slightly softer on string transients - so the sound character is still consistent but in DSD it gains more air, possibly higher noise floor and slightly more “horn” like in character for vocals but the surprise is a much bigger soundstage (without loss of precise imaging) and the location of everything jumps out at you with ease and there is a much stronger sense of space around everything (just as you would be aware of space at a concert due to reflections and high frequency cues). This definitely has to do with higher frequencies (above 3K) as the ear is very sensitive to the left right direction of higher frequencies and this is the kind of cues that tell us most about a space. Of course in audio the space can range from holographic artificial to very convincing “you are there” at a live event depending on the recording quality.

 

Initial conclusion, for PCM only users the DAC3 may be the better choice as it is much cheaper. However if you can feed your T+A D200 with DSD64 or above then definitely a worthwhile upgrade.

 

Hypotheses: I have no idea what is going on but I now have a suspicion that chip manufacturer attempts to achieve a lower noise floor through random sampling/averaging of multiple 1 bit sigma deltas is somehow also removing ultra low level transient detail. The DSD in the T+A D200 is a single converter and therefore must be extremely ultra linear compared to the chips with multiple converters - as a result it is both noisier and better at preserving more low level detail (the kind of detail that makes a soundstage jump out and come alive). I can better hear obvious mixing effects when listening to the DSD converter Vs the chip DAC, such as the change in noise floor when more channels are added to a mix. 

 

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13 hours ago, Miska said:

 

For that difference, you could buy a pretty powerful computer to run upsampling! And you can still use it simultaneously for reading and posting this forum...

 

 

For a senior contributor on this forum that feels like a pretty cheap shot.

 

I'm aware of this as an option but I've zero interest in doing so. And honestly from all I've read your software sounds like more hassle to get to grips with than I'm willing to invest. I want off the shelf plug and play excellence so that's what I've bought.

 

And my phone and laptop are doing just fine for reading and posting on the forums.

 

Cheers,

Alan

Synergistic Research Powercell UEF SE > Sonore OpticalModule (LPS-1.2 & DXP-1A5DSC) > EtherRegen (SR4T & DXP-1A5DSC) > (Sablon 2020 LAN) Innuos PhoenixNet > Muon Streaming System > Grimm MU1 > (Sablon 2020 AES) > Mola Mola Tambaqui DAC > PS Audio M1200 monoblocks > Focal Sopra No2 speakers

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18 hours ago, BigAlMc said:

 

Thanks @Shadorne,

 

The reply is much appreciated and I agree with your characterisation.

 

So best of luck to yourself and the other DAC 200 owners. I'm sure it's an amazing bit of kit. But in the end it didn't fit my use-case so I went on a different path.

 

Cheers,

Alan

 

You won’t be disappointed by all accounts the Mola Mola is fantastic. Would be interested to hear your evaluation of this against the DAC 3. The nice thing about DAC 3 is it serves as a reference - to me it always sounds the same on any input and it has no configurable options - so any evaluation or comparison to it is likely to translate better (that can also be its downside). I have had DACs come and go but for twenty years I have always owned a Benchmark DAC - it serves as a reference and allows me to confirm what I hear is real and NOT imagined. My sonic memory is not perfect and switching level matched inputs Vs a reference allows me to rule out confirmation bias. 

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2 hours ago, BigAlMc said:

 

For a senior contributor on this forum that feels like a pretty cheap shot.

 

I'm aware of this as an option but I've zero interest in doing so. And honestly from all I've read your software sounds like more hassle to get to grips with than I'm willing to invest. I want off the shelf plug and play excellence so that's what I've bought.

 

And my phone and laptop are doing just fine for reading and posting on the forums.

 

Cheers,

Alan

Alan, HQPlayer is an amazing bit of software, and Jussi is to be commended for his continuing efforts to keep making it even better.  Here I use HQPlayer and oversampling to DSD 256 or 512, with EC modulators, to fantastic effect with both my DIY DSC-2 DAC and Bricasti M3, and the results are fantastic.

All that being said, I have listened to the Tambaqui in quite a few systems, and it is the one single DAC I feel I could be happy with, without using HQPlayer oversampling.  Now I have not heard every DAC, but the Tambaqui has stood apart from others in what it is capable of for me.  I consider Bruno Putzeys (Tambaqui designer) one of the audio designers who knows his maths, and I also see that the Tambaqui has three powerful DSP processing chips on board to accomplish its own oversampling and filtering operations (OK, nowhere near the computing power required by HQPlayer, but more than any other DAC I am aware of).  If anyone can design digital filters and DS modulators as well as Jussi can, it is likely to be Bruno Putzeys (and perhaps Andreas Koch of Playback Designs).  The Tambaqui sounds great to me, and I understand that HQPlayer is not for everyone-I suspect you will be very, very happy with it!  Be sure to give it the best source feed possible to allow it to be at its best!  And please post about your results with it in the Tambaqui thread here.

SO/ROON/HQPe: DSD 512-Sonore opticalModuleDeluxe-Signature Rendu optical with Well Tempered Clock--DIY DSC-2 DAC with SC Pure Clock--DIY Purifi Amplifier-Focus Audio FS888 speakers-JL E 112 sub-Nordost Tyr USB, DIY EventHorizon AC cables, Iconoclast XLR & speaker cables, Synergistic Purple Fuses, Spacetime system clarifiers.  ISOAcoustics Oreas footers.                                                       

                                                                                           SONORE computer audio

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1 hour ago, barrows said:

Alan, HQPlayer is an amazing bit of software, and Jussi is to be commended for his continuing efforts to keep making it even better.  Here I use HQPlayer and oversampling to DSD 256 or 512, with EC modulators, to fantastic effect with both my DIY DSC-2 DAC and Bricasti M3, and the results are fantastic.

All that being said, I have listened to the Tambaqui in quite a few systems, and it is the one single DAC I feel I could be happy with, without using HQPlayer oversampling.  Now I have not heard every DAC, but the Tambaqui has stood apart from others in what it is capable of for me.  I consider Bruno Putzeys (Tambaqui designer) one of the audio designers who knows his maths, and I also see that the Tambaqui has three powerful DSP processing chips on board to accomplish its own oversampling and filtering operations (OK, nowhere near the computing power required by HQPlayer, but more than any other DAC I am aware of).  If anyone can design digital filters and DS modulators as well as Jussi can, it is likely to be Bruno Putzeys (and perhaps Andreas Koch of Playback Designs).  The Tambaqui sounds great to me, and I understand that HQPlayer is not for everyone-I suspect you will be very, very happy with it!  Be sure to give it the best source feed possible to allow it to be at its best!  And please post about your results with it in the Tambaqui thread here.

 

Hey @barrows,

 

Long time. Hope you're well mate.

 

I don't doubt for a second that HQPlayer is fantastic and I have a lot of respect for what @Miska is doing with it. The coverage and comments on here leave me in no doubt he's doing something very right and understandably has a very loyal following.

 

I just felt the sarcastic response was unwarranted when I explained I'd chosen a different path to audio nirvana.

 

I'm sure HQPlayer and the DAC 200 is an outstanding combo. I just wanted something simpler.

 

Will post on the Tambaqui thread once I've got anything to share. The Mola Mola arrived a couple hours ago so defo too early to comment.

 

I know I'm off topic here so back to DAC 200 stuff for the lucky T+A owners.

 

Cheers,

Alan

Synergistic Research Powercell UEF SE > Sonore OpticalModule (LPS-1.2 & DXP-1A5DSC) > EtherRegen (SR4T & DXP-1A5DSC) > (Sablon 2020 LAN) Innuos PhoenixNet > Muon Streaming System > Grimm MU1 > (Sablon 2020 AES) > Mola Mola Tambaqui DAC > PS Audio M1200 monoblocks > Focal Sopra No2 speakers

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Not to stray too far from topic but I too was between the DAC200 and the Mola Mola. I'm sure the Mola Mola is great but when doing research I found out it has over a 30 ms delay in audio (most likely due to its processing) and this is fine for most people that would use the DAC strictly for music but since I game on this PC and such that was a deal breaker for me. As much as I loved the Yggdrasil (still do) having the ability to use DSD is a game changer for audiophile sound quality. 

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Hi, I just got my DAC 200 and have some questions to the experts here in this forum:
 

1.) I am using two digital transports as sources together with the DAC 200, an Auralic G1.1 (with CD-rips on the internal SSD, connected via USB) and an Ayon CD-T II (connected via Digital Coax). Both sources have also the possibility for up/re-sampling/filtering the signal.

Now the question, what is best for SQ:
a.) raw data without up/re-sampling/filtering all the way to the end (AVM A6.2 ME), according to the "fact" they can´t add anything that hasn´t been already there?

b.) up/re-sampling/filtering already at the digital transports and then send this without additional treatment, except DA convertion through the DAC to the Amplifier?

c.) up/re-sampling/filtering in both stages, digital transport and after this also in the DAC200?

 

Forgive me if these are stupid questions but I am quite new to this.

 

Thanks for Your support.

Greetings

Wolfram

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On 7/18/2023 at 8:09 PM, Tihon said:

 

 

In my opinion people seriously underestimate the power of upsampling done right. And most often because they do not try what they need to try, but extrapolate their experience gained with other software, inappropriate DACs, suboptimal settings, etc.

It is fairly well established and accepted that upsampling is the way to go in order to have analog filters that remove aliased/ghost images effectively with minimal impact to pass band audio. 

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2 hours ago, Woolf said:

Hi, I just got my DAC 200 and have some questions to the experts here in this forum:
 

1.) I am using two digital transports as sources together with the DAC 200, an Auralic G1.1 (with CD-rips on the internal SSD, connected via USB) and an Ayon CD-T II (connected via Digital Coax). Both sources have also the possibility for up/re-sampling/filtering the signal.

Now the question, what is best for SQ:
a.) raw data without up/re-sampling/filtering all the way to the end (AVM A6.2 ME), according to the "fact" they can´t add anything that hasn´t been already there?

b.) up/re-sampling/filtering already at the digital transports and then send this without additional treatment, except DA convertion through the DAC to the Amplifier?

c.) up/re-sampling/filtering in both stages, digital transport and after this also in the DAC200?

 

Forgive me if these are stupid questions but I am quite new to this.

 

Thanks for Your support.

Greetings

Wolfram

It seems that general forum consensus is to upsample to DSD256 or higher and feed that to the true 1 bit DSD converter in the T+A D200 is best for many folks  - for me, in my setup, the soundstage really expands and deepens with no loss of detail (matching other observations).

 

That said there is a lot of praise for the PCM converter too which I can attest - it also sounds great.

 

In my experience you lose nothing on any setting with the T+A D200 operating in any mode - it all sounds absolutely excellent. However, everyone may have a personal preference according to what works best in their setup.

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Hi everyone - When playing DSD files, does the DAC 200's filters come into play? or are they bypassed?

 

Similarly, when playing PCM 44.1K/48K redbook, if we select NOS 1/2 filters, what will happen? Will it bypass the PCM upsampling in DAC 200?

 

The reason why I have asked about PCM is, I was listening this album from Roon(Qobuz) with NOS 1 filter. It is just magic. Such an analog sound I can feel.

 

image.thumb.png.2ccb21f4845b9dd994520e9034cfca54.png

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2 hours ago, kishoreburidi said:

Hi everyone - When playing DSD files, does the DAC 200's filters come into play? or are they bypassed?

 

Similarly, when playing PCM 44.1K/48K redbook, if we select NOS 1/2 filters, what will happen? Will it bypass the PCM upsampling in DAC 200?

 

The reason why I have asked about PCM is, I was listening this album from Roon(Qobuz) with NOS 1 filter. It is just magic. Such an analog sound I can feel.

 

image.thumb.png.2ccb21f4845b9dd994520e9034cfca54.png

Correct. For PCM data the 6 filters are in effect. If you use NOS 1 then there is no upsampling, no digital filter and the analog filter is at 60KHz. The sound of this will greatly depend on your system as there will be a lot of high frequency ghost data. That said if you look at another post of mine there is a theory/hypothesis for why this might sound better - here is the link again http://www.mlssa.com/pdf/Upsampling-theory-rev-2.pdf

 

The whole thing is linked to noise specs and DAC chip methodologies and the fact that high frequencies if fairly random might help to linearize the response while remaining inaudible.

 

The concern from a traditional viewpoint is that all this high frequency noise coming through on your output is undoubtedly going to cause increased IMD distortion within the in band audible range.

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