T+A DAC 200

[Jud]

2 hours ago, Nkam said:

As a musician of 40 years?   No 

sorry I didn’t mean to mention that as a snob.  
merely stating that I have trained ears.  Not just a casual listener. 
 

the attack is softer.   
 

the sound isn’t smoother.  You can hear texture fine.  Maybe a bit more than PCM.  

 

Filter choice may also matter here.

[Nkam]

1 hour ago, Jud said:

 

Filter choice may also matter here.

 

I’ve used many many filters.    
 

We all hear differently and have different tastes. 
 

im not saying anything ‘ bad’.  
 

I’ve talked with others who think it sounds soft as well, and/or prefer PCM. 
 

it’s not a big deal.  
 

no one is trying to change anyones mind.     
 

know what I mean?    It’s all good. 
 

i mean many engineers also say R2R isn’t the best for audio. They also have opinions that DSD is inferior.  
then there are those who say DSD is the superior way to listen. 
 

I still would like to hear it on a powerful system to have the settings Miska advised to.  
 

and the DAC has to be very good as well.    That’s a considerable investment for myself.  
 

I will someday.  

[Allan F]

3 hours ago, Nkam said:

sorry I didn’t mean to mention that as a snob.  

 

"No worries" as the Aussies say. 🙂 I didn't mean "smoother" in a negative sense, i.e. just less "harsh". In any case, we all don't hear the same things.

[Nkam]

40 minutes ago, Allan F said:

 

"No worries" as the Aussies say. 🙂 I didn't mean "smoother" in a negative sense, i.e. just less "harsh". In any case, we all don't hear the same things.

Ha.   True too 

 

yeah, fortunately or unfortunately we have different tastes. 
 

but I’d still love to hear it on a great setup.    
why rob myself of that ? 
It might actually change my mind. 

[Nkam]

Any measurements of the T+A?

 

Much obliged 

[OE333]

On 4/11/2023 at 7:47 AM, Nkam said:

Does the DAC 200 have extra headroom for intersample overs? 
To avoid clipping internally ?

 

Yes. There is no clipping no matter of which upsampling method used.

[OE333]

2 hours ago, Nkam said:

Any measurements of the T+A?

 

... anything special you are interested in ?

 

[Nkam]

1 hour ago, OE333 said:

 

... anything special you are interested in ?

 

 

yeah 

 

any FFT to show which harmonics are dominant?

I saw the specs and the THD is low enough.  
just which harmonics are there.  
 

thanks in advance! 

[OE333]

4 hours ago, Nkam said:

any FFT to show which harmonics are dominant?

 

... no problem: I will post a few FFTs when I'm back in office.

[pavi]

7 hours ago, OE333 said:

 

... anything special you are interested in ?

 

 

does the t+a dac200 perform equally well at dsd512 & dsd1024? 

at what bitrate is it most linear? the holo may is optimal at 20 bits, as is well attested. is there a similar data point for the dac200?
 

thank you!

[Miska]

3 hours ago, pavi said:

does the t+a dac200 perform equally well at dsd512 & dsd1024? 

 

This depends on how you want to weight things, which aspects are most important for your case. I use it at DSD256 or DSD512.

 

3 hours ago, pavi said:

at what bitrate is it most linear? the holo may is optimal at 20 bits, as is well attested. is there a similar data point for the dac200?

 

Since DAC 200 is not R2R NOS DAC, it doesn't have such point. If you want to send PCM there, use 32-bit output which is best food for it's internal DSP.

 

[Nkam]

5 hours ago, OE333 said:

 

... no problem: I will post a few FFTs when I'm back in office.

 

thank you so much.  
 

im just curious as to how much the 2nd and 3rd harmonic affect the sound to make it sound ‘ more real’. 
 

Just from personal experience I have found DACs with very low THD to not sound as natural as DACs with a bit higher THD.  But 2nd and 3rd dominant. 
 

and when I say too little I mean anything below 0.001%.  
 

im probably wrong as this is just an assumption of mine. 
 

 

[bogi]

Although I am not T+A DAC 200 owner I would like to add my subjective opinion to the DSD softness debate. It's just an opinion so I don't ask anybody to agree or fully agree with me.

IME instrument attacks (transients) usually sound 'softer' but at the same time more detailed with direct DSD than with oversampled delta sigma PCM. IMO the reason is additional distortion which PCM output contains and which is not present in the case of direct DSD. If that wouldn't be distortion, then with DSD input I couldn't hear more fine detail. That's the only reason why I like to upsample PCM content to DSD with HQPlayer.

I think that the mentioned distortion is result of intermodulation artifacts caused by unfiltered second oversampling stage of delta sigma DACs - that's what one can bypass with direct DSD (as well as with DSD signal fed to ESS chip based DACs). I perceive that additional distortion as some additional hardness on transients. My opinion is that most of people are already used to 'PCM sound' containing such a distortion and consider it to be correct (since we are listening to it for years everywhere). IMO that's the reason why sound coming from direct DSD may be perceived as softer, even if it provides more fine detail and is more accurate. If I am right then it is not by coincidence that such a 'PCM hardness' is not present when playing from vinyl on pure analog chain.

If 'soft' would mean less dynamics and 'hard' more dynamics, then IMO DSD256 or DSD512 is not softer.

[Jud]

2 hours ago, bogi said:

Although I am not T+A DAC 200 owner I would like to add my subjective opinion to the DSD softness debate. It's just an opinion so I don't ask anybody to agree or fully agree with me.

IME instrument attacks (transients) usually sound 'softer' but at the same time more detailed with direct DSD than with oversampled delta sigma PCM. IMO the reason is additional distortion which PCM output contains and which is not present in the case of direct DSD. If that wouldn't be distortion, then with DSD input I couldn't hear more fine detail. That's the only reason why I like to upsample PCM content to DSD with HQPlayer.

I think that the mentioned distortion is result of intermodulation artifacts caused by unfiltered second oversampling stage of delta sigma DACs - that's what one can bypass with direct DSD (as well as with DSD signal fed to ESS chip based DACs). I perceive that additional distortion as some additional hardness on transients. My opinion is that most of people are already used to 'PCM sound' containing such a distortion and consider it to be correct (since we are listening to it for years everywhere). IMO that's the reason why sound coming from direct DSD may be perceived as softer, even if it provides more fine detail and is more accurate. If I am right then it is not by coincidence that such a 'PCM hardness' is not present when playing from vinyl on pure analog chain.

If 'soft' would mean less dynamics and 'hard' more dynamics, then IMO DSD256 or DSD512 is not softer.

OK, now I'm confused. I'm not a T+A 200 owner, so if my questions are too OT or impertinent, please just ignore them.

 

- What is 'delta sigma PCM'?

 

- I thought all oversampling stages had to be filtered?

 

Thanks.

[1125FPS]

3 hours ago, bogi said:

Although I am not T+A DAC 200 owner I would like to add my subjective opinion to the DSD softness debate. It's just an opinion so I don't ask anybody to agree or fully agree with me.

IME instrument attacks (transients) usually sound 'softer' but at the same time more detailed with direct DSD than with oversampled delta sigma PCM. IMO the reason is additional distortion which PCM output contains and which is not present in the case of direct DSD. If that wouldn't be distortion, then with DSD input I couldn't hear more fine detail. That's the only reason why I like to upsample PCM content to DSD with HQPlayer.

I think that the mentioned distortion is result of intermodulation artifacts caused by unfiltered second oversampling stage of delta sigma DACs - that's what one can bypass with direct DSD (as well as with DSD signal fed to ESS chip based DACs). I perceive that additional distortion as some additional hardness on transients. My opinion is that most of people are already used to 'PCM sound' containing such a distortion and consider it to be correct (since we are listening to it for years everywhere). IMO that's the reason why sound coming from direct DSD may be perceived as softer, even if it provides more fine detail and is more accurate. If I am right then it is not by coincidence that such a 'PCM hardness' is not present when playing from vinyl on pure analog chain.

If 'soft' would mean less dynamics and 'hard' more dynamics, then IMO DSD256 or DSD512 is not softer.

I am going to have to agree 100% with this post. 

 

In my teens and early 20s I spent many hours in several then-fine studios, including Different Fur Trading Company (SF, engineer John Viera upgraded their Studer 15ips half track master deck to 30ips,) my friend Kelly Quan's studio and The Record Plant in Sausalito, etc.  My pure analog reference is not cost no object vinyl but rather some of the best magnetic tape.  I hope and presume no one with similar experience debates that the best magnetic tape simply incinerates any vinyl rig Michael Fremer has heard or imagined.  I respect Fremer's accomplishments, but this point prohibits me from sharing his vinyl obsession.  (Software cost and availability obviously favors vinyl over tape.)

 

Miska's HQ Player DSD settings > DAC 200 provides more musical detail and musically pertinent information than any prior digital experience.  I heard the $85k MSB DAC summer of 2019 and dCS's best 3-box $100k rig in the mid-00s.  For unknown reason I am unable to hear PCM on my current system and shall post after I compare DSD vs. PCM later.  I doubt PCM favorably compares to DSD in this system but I'm happy to discover I'm wrong.

 

3 years ago, my wife and I independently and confidentially wrote notes comparing DSD vs. PCM Re. 12 songs, HQ Player > Holo Audio May KTE.  When required we repeated the format switch.  We agreed on all or most songs, preferring PCM about 2/3rds of the songs and DSD on the balance. 

 

Using HQ Player, IMO the May KTE is only about 10% better than Spring 3 KTE.  

 

Used with HQ Player both May Audio DACs are best values at their respective SRPs of $3100 (Spring 3 KTE) and $5500 (May KTE,) the former being the better value.  But at $7200 SRP DAC 200 incinerates both May DACs, not in the same zip code, exceeding the SRP differences as a ratio or absolute amount.  

 

Suppose the AB test comprises HQ Player's advanced DSD settings with:

A: DAC 200 > well chosen $3k power amp and well chosen $10k speakers

B: Holo Audio DAC > virtually cost no object power amp and speakers

 

IMO it would be difficult to impossible to assemble system B with better performance.  I suspect, same as Ivor Tiefenbrun predicted 50 year ago in his "Audio Hierarchy," B has a possibly insurmountable hurdle being that the better the resolution of the amp and speakers, the more it reveals the source's weaknesses.  A has the advantage of supplying among the world's best musical sources to the highest value amp and speakers.   

 

I can still enjoy and appreciate vinyl.  But I suspect if someone compared superb vinyl to my current rig and still preferred the former, the reason is simply because they are more accustomed to the vinyl rig's distortion.  If it's not obvious, all bets off if the digital conversion employed an inferior master tape or inferior ADC.

[Nkam]

14 minutes ago, 1125FPS said:

I am going to have to agree 100% with this post. 

 

In my teens and early 20s I spent many hours in several then-fine studios, including Different Fur Trading Company (SF, engineer John Viera upgraded their Studer 15ips half track master deck to 30ips,) my friend Kelly Quan's studio and The Record Plant in Sausalito, etc.  My pure analog reference is not cost no object vinyl but rather some of the best magnetic tape.  I hope and presume no one with similar experience debates that the best magnetic tape simply incinerates any vinyl rig Michael Fremer has heard or imagined.  I respect Fremer's accomplishments, but this point prohibits me from sharing his vinyl obsession.  (Software cost and availability obviously favors vinyl over tape.)

 

Miska's HQ Player DSD settings > DAC 200 provides more musical detail and musically pertinent information than any prior digital experience.  I heard the $85k MSB DAC summer of 2019 and dCS's best 3-box $100k rig in the mid-00s.  For unknown reason I am unable to hear PCM on my current system and shall post after I compare DSD vs. PCM later.  I doubt PCM favorably compares to DSD in this system but I'm happy to discover I'm wrong.

 

3 years ago, my wife and I independently and confidentially wrote notes comparing DSD vs. PCM Re. 12 songs, HQ Player > Holo Audio May KTE.  When required we repeated the format switch.  We agreed on all or most songs, preferring PCM about 2/3rds of the songs and DSD on the balance. 

 

Using HQ Player, IMO the May KTE is only about 10% better than Spring 3 KTE.  

 

Used with HQ Player both May Audio DACs are best values at their respective SRPs of $3100 (Spring 3 KTE) and $5500 (May KTE,) the former being the better value.  But at $7200 SRP DAC 200 incinerates both May DACs, not in the same zip code, exceeding the SRP differences as a ratio or absolute amount.  

 

Suppose the AB test comprises HQ Player's advanced DSD settings with:

A: DAC 200 > well chosen $3k power amp and well chosen $10k speakers

B: Holo Audio DAC > virtually cost no object power amp and speakers

 

IMO it would be difficult to impossible to assemble system B with better performance.  I suspect, same as Ivor Tiefenbrun predicted 50 year ago in his "Audio Hierarchy," B has a possibly insurmountable hurdle being that the better the resolution of the amp and speakers, the more it reveals the source's weaknesses.  A has the advantage of supplying among the world's best musical sources to the highest value amp and speakers.   

 

 

 

 

i disagree 100%.  
lol

sorry I’m just Messin.    
my humor. 
 

i honestly don’t know which is ‘ correct’.   I would be lying if I said I did. 
 

but even PCM sounds more correct to me in the time domain region.   
 

again, I may be wrong.  
 

40 years of being a musician tho.  Gotta account for something.  
 

but honestly I’m not trying to impose my opinion. 
It’s exactly that.  Just an opinion. 
and good discussion too gents 

 

cheers 

[austinpop]

22 minutes ago, 1125FPS said:

I doubt PCM favorably compares to DSD in this system but I'm happy to discover I'm wrong.

 

If you're interested in exploring just how good the PCM pipeline of the T+A DAC-200 really is, as I did in Chris's house (see https://audiophilestyle.com/ca/bits-and-bytes/my-visit-to-audiophile-style-hq-—-another-take-on-immersive-vs-2ch-audio-r1186/),

 

try it with a few tracks pre-upsampled to 32/16FS with PGGB. The trial license allows you to process some number of tracks for free.

 

Use the settings as shown:

 

For these tracks, you can now compare:

1. the original native PCM files, with DAC-200 set to BEZ2 or the filter that most appeals to you
2. the real-time upsampled DSD256 or DSD512 from HQPlayer, with the DAC-200 set to NOS2
3. the PGGB-upsampled 32/705.6 or 32/768 WAV files, with the DAC-200 set to NOS2.

Feel free to share your findings.

[luisma]

54 minutes ago, 1125FPS said:

I am going to have to agree 100% with this post. 

You really know how to stir this pot :), but you are an owner of this DAC (and many others) and your empirical opinion counts and it matters, at least to me.

 

56 minutes ago, 1125FPS said:

For unknown reason I am unable to hear PCM on my current system and shall post after I compare DSD

My apologies and I am responsible partially for this, I will call you in the next few days to get this fixed and working.

 

57 minutes ago, 1125FPS said:

I doubt PCM favorably compares to DSD in this system but I'm happy to discover I'm wrong.

I'm with you on this, I used to preferred PCM until ASDM7EC came out, never looked back. My digital audio knowledge is very limited and mathematically I would love to know what the modulator does that makes it so good, the reconstruction accuracy and DSD format which provides extra is a big part of it (at least I think), when I was going through Calculus IV at college I remember the Fourier Series and I did some programming for CNC using numerical methods Fourier libraries but my brain is having trouble understanding how 10^x accuracy converting can make the audio sound that good. Anyways this is not important as long as I can use it (I have to deal with other complex issues on my daily life)

 

1 hour ago, 1125FPS said:

I can still enjoy and appreciate vinyl.  But I suspect if someone compared superb vinyl to my current rig and still preferred the former, the reason is simply because they are more accustomed to the vinyl rig's distortion

My friend, I suspect that I appreciate the imperfections, I truly love clear, no background noise, impeccable sound, but at the same time I don't get bothered by imperfections in "presentation", even phase, timbre is something that throws' me off if it's not exact. I'm a fan of live venues, and live venues are many times full of imperfections, reflections, noise (are you kidding me :), still enjoyable. Ultimately I believe (and others can digress of course) that musical enjoyment could be "flexible" in many aspects, we spend hours, days, weeks here on AS (thank you Chris) trying to make sense of it all, and we do enjoy this, but at the end it's a more simple journey if you do like audio / music.

 

1 hour ago, 1125FPS said:

if someone compared superb vinyl to my current rig and still preferred the former, the reason is simply because they are more accustomed to the vinyl rig's distortion.

It's just different, not necessarily better measuring. (but we are not in ASR here)

 

49 minutes ago, austinpop said:

try it with a few tracks pre-upsampled to 32/16FS with PGGB.

Thanks Rajiv.

 

1 hour ago, Nkam said:

i honestly don’t know which is ‘ correct’.   I would be lying if I said I did. 
 

but even PCM sounds more correct to me in the time domain region

My DAC200 was supposed to be delivered today but Fedex messed up again, I could be able by next week to provide a "subjective" answer to this.

 

 

[bogi]

6 hours ago, Jud said:

- What is 'delta sigma PCM'?

I meant output of delta sigma DAC when it is fed by PCM signal. My post is not related to output of R2R DACs, since I never owned such one.

 

6 hours ago, Jud said:

- I thought all oversampling stages had to be filtered?

Every resampling creates images of audio band at multiples of target fs. Miska mentioned numerous times, that hardware oversampling of delta sigma DACs is realized in 2 stages. For the 1st one it may be possible to choose filter type. For T+A DAC200 they are FIR1, FIR2, Bezier1, Bezier2 and it is realized outside of DAC chip. High frequency content containing images of audio band is filtered in this stage. This is only up to some fs. Then the 2nd stage follows within delta sigma DAC chip which is simplified (usually sample and hold). See this Miska's post. High frequency content remains unfiltered and therefore images of audio band at multiples of last filtered target fs are coming to input of delta sigma modulator and should be filtered by analog filter behind the modulator. But that analog filter is usually not so steep that it could fully filter the nearest images, as shown many times in Miska's measurements.

For demonstration see for example 2th, 3th and 4th graph of old iFi Micro measuremens. https://audiophilestyle.com/blogs/entry/428-ifi-idsd-micro-measurements/. Images of audio band (a pair mirrored - non mirrored) appear around every multiple of 352.8k, which is the border fs between oversampling stages.

With DSD these images of audio band don't appear on measurements. The reason is that upsampling in HQPlayer filters these images digitally up to the target fs.

[Jud]

34 minutes ago, bogi said:

I meant output of delta sigma DAC when it is fed by PCM signal. My post is not related to output of R2R DACs, since I never owned such one.

 

Every resampling creates images of audio band at multiples of target fs. Miska mentioned numerous times, that hardware oversampling of delta sigma DACs is realized in 2 stages. For the 1st one it may be possible to choose filter type. For T+A DAC200 they are FIR1, FIR2, Bezier1, Bezier2 and it is realized outside of DAC chip. High frequency content containing images of audio band is filtered in this stage. This is only up to some fs. Then the 2nd stage follows within delta sigma DAC chip which is simplified (usually sample and hold). See this Miska's post. High frequency content remains unfiltered and therefore images of audio band at multiples of last filtered target fs are coming to input of delta sigma modulator and should be filtered by analog filter behind the modulator. But that analog filter is usually not so steep that it could fully filter the nearest images, as shown many times in Miska's measurements.

For demonstration see for example 2th, 3th and 4th graph of old iFi Micro measuremens. https://audiophilestyle.com/blogs/entry/428-ifi-idsd-micro-measurements/. Images of audio band (a pair mirrored - non mirrored) appear around every multiple of 352.8k, which is the border fs between oversampling stages.

With DSD these images of audio band don't appear on measurements. The reason is that upsampling in HQPlayer filters these images digitally up to the target fs.

Yep, I had seen this and now understand exactly what you were referring to. It is of course possible to do something other than bad filtering at the 2nd or 3rd oversampling (8x oversampling is usual, so 3 doublings) though as you say the measurements appear to show it is not typical at least in those DACs for which Miska has released measurements.

[bogi]

33 minutes ago, Jud said:

Yep, I had seen this and now understand exactly what you were referring to. It is of course possible to do something other than bad filtering at the 2nd or 3rd oversampling (8x oversampling is usual, so 3 doublings) though as you say the measurements appear to show it is not typical at least in those DACs for which Miska has released measurements.

No, I still meant it otherwise. Taking the iFi Micro as an example, I did not mean 2th and 3th step of 8x oversampling (3 steps of fs doubling, 1->2, 2->4, 4->8). One or more (or all three) of these steps may be skipped if DAC is fed by higher than 44.1k / 48k PCM input (up to 352.8 / 384k).

As 1st oversampling stage I meant those max. 3 steps as a whole. That stage is filtered, so it does not leave images of audio band in frequency spectrum. This 1st stage ends at some fs, for iFi it was 352.8/384k, but it may be higher with other chips like ESS. But still it is not at delta sigma stage operating frequency. Therefore 2nd oversampling stage is needed and that one is unfiltered.

Although it may be possible to substitute the on chip (I mean DAC chip) 1st stage by external solution (T+A DAC200 uses this approach), it is not possible to substitute or skip the 2nd stage. The target fs of the 2nd stage is ~ 10MHz (like 128x44.1/128*48, or 256x44.1/256*48), exact value depends on DAC chip.

Every time when Miska mentioned ZOH (Zero Order Hold) or SaH (S&H) (Sample And Hold) he meant the on chip unfiltered 2nd oversampling stage. That means simple repeating of previous sample n times - it's the easiest way to increase sample rate up to that ~10MHz range. DAC chips don't have computational power to do more on such a high sample rates.

[Miska]

5 hours ago, austinpop said:

For these tracks, you can now compare:

1. the original native PCM files, with DAC-200 set to BEZ2 or the filter that most appeals to you
2. the real-time upsampled DSD256 or DSD512 from HQPlayer, with the DAC-200 set to NOS2
3. the PGGB-upsampled 32/705.6 or 32/768 WAV files, with the DAC-200 set to NOS2.

Feel free to share your findings.

 

When you use DSD to DAC 200, it doesn't matter if it is set to NOS2 or something else, that applies only to PCM. For DSD output only setting that applies is the analog LPF setting of wide or 60 kHz. You get about 20 dB lower distortion in DSD mode compared to PCM mode...

 

Issue with those super long filters is that once hit, they create constant ringing around the original Nyquist.

 

[Zaphod Beeblebrox]

6 hours ago, Miska said:

Issue with those super long filters is that once hit, they create constant ringing around the original Nyquist.

Sorry if this is OT:

 

This argument is no different than what has been used by the proponents of minimum phase and apodizing filters on the 'dreaded, terrible' pre-ringing of linear phase filters which cannot really be seen nor heard with true music signals but can be easily simulated with artificial test signals (impulse, step, 22.05kHz signal with exactly two samples/cycle)  and does not have much empirical evidence to support it.

 

Yes, in theory any abrupt change in the frequency domain will lead to ringing in the time domain and vice-versa, some of which can be mitigated with windowing. However, ringing at the Nyquist is a phenomenon that is overblown for multiple reasons.

• First the Nyquist is 22.05kHz for CD rates far beyond human audible range.
• There is rarely any significant energy near the Nyquist with real music.
• With true music signals where it has already gone through a bunch of processing and filtering, it is very hard to demonstrate/show  ringing near Nyquist after oversampling with a good quality brick wall filter.

There is also a nice blog post here (it talks about other things too but the main focus is the ringing):  Archimago's Musings: MUSINGS: Digital Interpolation Filters and Ringing (plus other Nyquist discussions and "proof" of High-Resolution Audio audibility)

 

ps: PGGB 256 does away with the concept of ultra long filters and taps, but it is still equivalent to using a near ideal sinc filter.

 

 

[Miska]

2 hours ago, Zaphod Beeblebrox said:

This argument is no different than what has been used by the proponents of minimum phase and apodizing filters on the 'dreaded, terrible' pre-ringing of linear phase filters which cannot really be seen nor heard with true music signals but can be easily simulated with artificial test signals (impulse, step, 22.05kHz signal with exactly two samples/cycle)  and does not have much empirical evidence to support it.

 

I have done quite a bit of analysis with different types of content. And it really exists also with music signals. HQPlayer as also runtime analysis for these kind of behaviours, you can see how high the counter can get during a typical track.

 

Quote

• First the Nyquist is 22.05kHz for CD rates far beyond human audible range.

 

It is much closer to the human audible range than the images you would get by using a very leaky oversampling filter. So does the filter matter? If it does, why the whine wouldn't? So what you are saying is contradictory with your reasoning that your filters matter.

 

In addition, it is right where many metal dome tweeters have +20 dB resonance peak.

 

I'm not fond of having constant 22.05 kHz whine, especially at +20 dB levels.

 

2 hours ago, Zaphod Beeblebrox said:

 

• There is rarely any significant energy near the Nyquist with real music.
• With true music signals where it has already gone through a bunch of processing and filtering, it is very hard to demonstrate/show  ringing near Nyquist after oversampling with a good quality brick wall filter.

 

No so much, especially with typical processed loudness wars RedBook. A bit of hard digital clipping commonly prominent on modern recordings is enough to trigger it.

 

2 hours ago, Zaphod Beeblebrox said:

ps: PGGB 256 does away with the concept of ultra long filters and taps, but it is still equivalent to using a near ideal sinc filter.

 

"Ideal" sinc filter is ideal only for infinitely long constant signals in frequency domain. But it is totally non-ideal for signals that change all the time, in time domain.

 

For frequency domain ideal filter is infinitely long sinc, and for time domain ideal filter is Dirac delta pulse. You like to go closer to the former, and MQA/Ayre/etc like to go closer to the latter. Both extremes are both right and wrong at the same time.

 

[dsyzling]

15 hours ago, austinpop said:

 

If you're interested in exploring just how good the PCM pipeline of the T+A DAC-200 really is, as I did in Chris's house (see https://audiophilestyle.com/ca/bits-and-bytes/my-visit-to-audiophile-style-hq-—-another-take-on-immersive-vs-2ch-audio-r1186/),

 

try it with a few tracks pre-upsampled to 32/16FS with PGGB. The trial license allows you to process some number of tracks for free.

 

Use the settings as shown:

 

For these tracks, you can now compare:

1. the original native PCM files, with DAC-200 set to BEZ2 or the filter that most appeals to you
2. the real-time upsampled DSD256 or DSD512 from HQPlayer, with the DAC-200 set to NOS2
3. the PGGB-upsampled 32/705.6 or 32/768 WAV files, with the DAC-200 set to NOS2.

Feel free to share your findings.

 

@austinpop thanks for the write up on your experiences with the T+A DAC-200. You've lived with Chord Dave for sometime now and I think you still have it in a second system. How would you say the T+A compares to the Chord with HQPlayer and PGGB upsampled content? Do you think it comes close to the timing of the Dave on transients and drive? Also wondered if you have any reflections on accuracy of timbre?