Trinity's claims vs. what actually happens in digital

esldude said:

I think I have read the same thing Peter St. You can look over this page:

http://www.trinity-ed.de/typo/index.php?id=19&L=1

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Hey, that's the one ! Super.

I think it got confusing in the translation to English. He refers to using his 8 DACs in a time delay as analog oversampling (the LIANOTEC as he refers to it). It really is just staggering the time each DAC is fed the digital signal and combining the outputs to create 8x interpolation.

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Not being native English myself I don't have too many problems with that. Point is that it depends somewhat on the difference where the analog part of this *happens* versus where it is *initiated* which should be in digital. For this we need to know a little bit more about electronics (and Dietmar will correct me where I go wrong) and in this case this is about the sheer impossibility to delay in analog. So, if that could happen (or happens with the Trinity) it would be a pure analog thing. Btw, because the times between samples are ultra short I think I would manage to do it in pure analog after all - just use certain lengths of wires. Or various types of resistences and varying current drivers. And only with a fixed sampling rate.
But no. This will just happen in the digital domain (1000 times more easy) and all it further needs is the summing of the currents which go automatically. And since the summing happens in the analog domain one could call it an "analog oversampling" but not all the way fair IMO. Still, digital it certainly also is not.
And so it is initialized in digital while it has its working (or workout) in analog.

How this is for the better for jitter is quite something else, or let's say that I don't see that; this is a totally different subject where now not samples are to be released from some best low jitter (phase noise !) oscillator, but something has to be arranged for the delay-timing as well. So mind you, one sample arrives at the input (behind the interface) and it has to be split into 8 samples with the exact same subsequent timing (the output samples are all 8 the same one). Now, if that were to let loose on 352.8 material it would need an ~88MHz oscillator (maybe I'm off one factor of 2) which needs to be 4 times better on the jitter because of the 4 times more close spacing of the output samples. Well, something like that. Do notice that I compare with my own situation of 768 as max which compares to 3072 mentioned for the Trinity. So as mentioned in that other thread, Phasure's oscillators could be 4x "worse" but the 4x better in the Trinity is just a necessity to be "on par" so to speak.
Another thing is that we don't need low ppm etc. (so no low ppb as well) but low phase noise. Didn't see anything about that.
In the end the story is way longer, because the 3072 sounds nice but the reasons for it do not much (to me, and this is what my large post is a.o. going to be about).
That I for myself all combine it with an isolated interface of which I exactly know what it all brings in the design department to not let that imply higher jitter - and that I can expect this isolation not working for the better at all in the Trinity, is again another very long story. And just saying, for me at this moment all this needs is a phase noise plot of the oscillator, no matter Dietmar would not understand why this is sufficient (at this moment !).

Down the page he explains that without a digital filter lower sample rates were not produced well enough. So for 32-96 khz sample rates they employ 8x digital oversampling which is then fed through the LIANOTEC system to get a further 8x interpolation. A total of 64x samples as I understand it. So if I understand this correctly at the sample rates higher than 96 khz there is no digital oversampling, but merely the 8x interpolation from the time delayed output of the 8 DAC's per channel.

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In my view this is all totally correct (I mean the reasons for it etc.). This is why I needed to find that text again because it is crucial for good design.
It could also be crucial for better understanding in general because I don't think I saw much about this from the nay sayers.
That this is just a ringing filter is something quite else, which *also* should be taken into account, but now from the yah sayers.

I still feel as though something is confused somewhere in that description or something left out. That would lead to a bit rate output higher at the lower sample rates. For instance 48 khz would become 3.072 mhz and 192 khz would become 1.536 mhz. Maybe he is using 2x oversampling at the higher bit rates, and I missed that in the description. Either way it appears he is not using a filter at the output at any sample rates.

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Hmm ... I didn't see anything wrong with it while going through it all, but let's see;
First off, LIANOTEC is not about bit rates as such. I mean, not as in "bandwidth". There's no additional bandwidth needed anywhere, except for something which clocks into the buffers before the samples are output (but never mind, because there are a few ways to set this up).
So start at the end : 8x LIANOTEC implies an output rate of 3072 (for 48KHz based material). But watch out, this only happens when the input sampling rate is 384. Thus, when the input rate is 192, the output rate is only 1536. 96 would lead to 768. In the mean time that page you found back for me, states that up to 96 an 8x oversampling filter is utilized. So, 8x 96 = 768. This now *is* bandwidth related, because the limit of the 1704. But no worries, because what comes behind it is LIANOTEC and this is just that "dumn" thing. All what happens again is that 8 1704's are time delayed again and the output will be 8x 768 = 6144. Nothing will sweat because as said, this is not bandwitdh related (up to some extend, but alas). So you are correct. And I myself *was* correct in not seeing it because I couldn't find that page anymore.
One notice now I myself am as far as this : The phase noise (jitter) has to be two times better again and the oscillator should be in the 192MHz range (the 48KHz based).

Last note for now : I personally have learned that as it seems each EE from Münich are amongst the best. Dietmar will be another one and if I were you guys I wouldn't start with distrust. That a couple of things are beyond our imagination or knowledge is something else, and that no EE can even start to explain to us laymans how these things are (e.g.) "allowed" to work, is not his fault. That in the mean time such a same EE can bash over email to customers from me saying things like I don't know sh*t on several subjects apparently is part of the job, but not much my style.

Dietmar, I realize that I am the xth person who charismatically tells others through a stupid internet forum how anoher one's (yours) product work. Please feel free to correct me anywhere necessary on the above. At least I myself don't like it when others do about my product, especially when it is plain wrong of course.

More to come ...
Peter

Hi Peter,
just a quick answer, even if it does not answers all of you open Points, but I have to finish the next 3 DACs.
Reading the WBF I get the feeling a few do not understand the excellent jitter behaviour of the LIANOTEC architecture.
The 8 x “LIANOTEC” architecture reduce the sensitivity to jitter by a factor of 3.
Enclosed the calculation for DACs with a ladder architecture. How to read the plot.
At 20kHz measurement bandwidth and a sampling frequency of 352.8kHz you need for a Signal to Noise Ratio of 127.5dB for a 1kHz signal a clock with a jitter of less than 200ps.
If you use LIANOTEC architecture you will get with the same 200ps clock source a SNR of 136.5dB.
Or if you want to reach the same SNR the clock can jitter with 600ps, means this architecture is 3x less sensitive to jitter.
With the VCOCXO and a jitter of a few fs (thousand times better than the theoretical value) I am much, much better than necessary. The jitter is specified in the Manual, which contains a lot of usefull informations.


I want to mention only another Point about the accuracy. All XRCD and K2HD are mastered with a Rhubidium clock http://www.xrcd.com/tech/xrcd24a_e.html
My Task was it to bring master tape Sound in your livingroom and that means for me I have to be as accurate as the recording Studio.
I use a Rhubidium master clock Generator in my lab to synchronize also all of my Equipment.

About a analog delay realized via PCB lenght or coax cable. The delay is depending of the "Epsilon relative" and for a 50Ohm coax line roughly in a range of 4,8ns/m!!! Not usefull for my application.
I used such technique to deskew differnt current sources I used for the blue laser Diode Driver, I developed for the blue ray stabdardisation.
By the way that was the first blue laser Driver in the world with rise times in a range of 350ps!!!
You see I have designed almost everything in my life.

The highest clock rate in the TRINITY DAC is the master clock. The the remaining digital artefacts are at 6.144MHz for 48, 96 and 192kHz recordings and therefore outside of the bandwidht of normal Audio preamplifier.
For pure Lianotec (Default setting) they are at 1.536MHz.
I call it analog oversampling, since LIANOTEC does not add any digital artefacts in to Output Signal, like pre-ringing or ringing at all.
KR Dietmar

Jitter reduced by averaging the 8 DACs?

I chalk some of the claims up to Marketing....try reading PSAudio's press release on how they just reinvented digital. Up to the Buyer's ears to decide. What I have been more surprised about is how much hype has been generated on at least 5 of the forums I follow by a small handful of the same people. A very few own the product, a very few more have heard it and some other just reference "reviews" like Jeff's on wizard's blog like they are fact As for the few of those who do own the product, I am glad they enjoy the product but the claims include so much hyperbole that they are tough to take seriously. I respect Adam's perspective based on his posts I have read but with regard to even some of his claims...I can only say wow. We'll know this hobby is subjective and very personal. If you are excited about a product tell us why and what elements get you closer to the music. Claims that it beats X or Y don't help as many have not heard those either. And the claims come across the wrong way To be clear I am not jealous....I do own the Vivaldi stack....I can afford the Trinity if I decide I like it and since I love the hobby I will try to listen to the Trinity DAC next time I am in NYC....but please keep it real guys....you guys are acting like you are in some exclusive club....I would rather you share what you are hearing

I am fortunate to know Audiocrack personally who owns Metronome Kalista Ref/Kondo DAC, Zanden 4-box, DCS Scarlatti...and he has done the shoot out at home with the Trinity. As posted here, he also bought one and has elected to sell his Metronome setup.

He is keeping Scarlatti primarily for SACD...and while he loves the Zanden and felt initially that it 'held its own' on redbook cd playback (he owns the Trinity transport so its was a like for like shootout)...I think over time now, he finds he prefers the Trinity.

So I feel like I know the voice behind the strong comments here on Trinity...I trust his ears.

I am a big fan of the Zanden. I am perfectly happy with redbook cds, and have done shootouts with many digital players and found none to tempt me. I too would like to hear the Trinity, but I am instinctively not concerned it is hype. That does not mean I go out and buy one, but having spoken to a few other people I know who either own one or have listened carefully...I suspect it is the real deal. The question is whether 'the real deal' makes it so much further ahead.

I suppose when digital combines pure tonality, deep colors, and extraordinary detail/exactitude of instrument separation...that IS a big deal in the DIGITAL world (whereas it might not be for tape guys for example)...because historically digital has 'stereotypically' fallen on one side of that fence or the other. That started to change about 5-7 years ago (Wadia 9 series, Stahl-Tek), and the last 2+ years has started to see really marked steps forward in that regard (TotalDAC, Vivaldi, Light Harmonics, Trinity, for example).

In fact, I also hear similarly supreme positives about Vivaldi, which equally has enjoyed a big amount of press and favourable commentary by owners/members and auditioners (including me).

Guess I will have to find out with respect to Trinity when I hear it for myself.

TRINITY said:

...Reading the WBF I get the feeling a few do not understand the excellent jitter behaviour of the LIANOTEC architecture.
The 8 x “LIANOTEC” architecture reduce the sensitivity to jitter by a factor of 3.
Enclosed the calculation for DACs with a ladder architecture. How to read the plot.
At 20kHz measurement bandwidth and a sampling frequency of 352.8kHz you need for a Signal to Noise Ratio of 127.5dB for a 1kHz signal a clock with a jitter of less than 200ps.
If you use LIANOTEC architecture you will get with the same 200ps clock source a SNR of 136.5dB...

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Hi, Trinity,

Assuming that I correctly understand what you are suggesting about LIANOTEC and jitter, I don't believe that your analysis is correct. I read your post as suggesting that a higher overall DAC Johnson noise floor could mask the benefits of having a lower jitter noise floor. Therefore, a given DAC system may require a lower Johnson noise floor in order to take advantage of a low clock jitter noise floor. Which may be true for random jitter, but is probably less true for deterministic jitter. Interestingly, if I were to invert this argument, then we could make the subjective effects clock jitter a non-factor simply by sufficiently increasing the DAC system's random noise floor. Digitally generated dither may or may not accomplish such a thing, as digitally generated dither is still subject to clock jitter. This might requirer the injection of analog dither or noise. A thought topic deserving of it's own thread, perhaps. Anyhow, I suggest that this matter doesn't have to do with a reduced jitter sensitivity (or, more clearly stated, as an increased jitter rejection) of the DAC system.

Your 9dB SNR improvement figure is fully commensurate with reduced Johnson noise due to paralleling 8 D/A units, relative to the noise produced by a single such D/A unit. This is the same noise floor improvement one would get from paralleling 8 digitally oversampled D/A units, without the use of LIANOTEC. Therefore, I think that you might make the argument that paralleled D/A units could help a DAC system to take advantage of a reduced clock jitter floor, but not that LIANOTEC itself plays a determining role.

Being another DAC manufacturer, I read this with much interest. My experience since the advent of digital audio has been that jitter was initially the main deficiency and then later with the advent of oversampling Sigma-Delta D/A chips and lower jitter solutions, digital filtering became the main deficiency of digital audio. Thus the recent interest in older NOS D/A chips.

My own strategy is to use minimal or very high frequency digital filtering with a modern D/A chip and do more filtering in the analog domain. Its a kind of Sigma-Delta/NOS hybrid approach. This strategy delivers a much more lifelike analog sound. If I back-off only a little on my analog filtering I can start to see stair-steps. I don't care about this distortion because downstream preamps, amps and speakers filter it out nicely. Adding only a little more analog filtering impacts audio quality. I have found that the best way to determine the right amount of filtering is by ear, after all that is the final instrument that we use.

If the Trinity DAC designer can deliver a more natural sounding digital filtering technique, he is on the right track and understands the true limitations of current DAC technology. There are very few digital filters, both embedded and custom that sound natural.

And BTW, the video is BS. It totally ignores that the D/A device used in the measurements has both digital and analog filtering in it and the fact that the problem does not lie in steady-state sinewave signals, its all about how transients are reproduced. If the music system is only asked to reproduce a 1kHz sinewave this will be sufficient, but not for real music. The measurement system noise floor and B/W are also grossly insufficient to make meaningful measurements IMO. And this claim that 20kHz is sufficient B/W. I don't think so. This is like saying I want to measure a signal with a scope that has 1MHz components in it and buying a scope with 1MHz B/W. Never work. You need headroom with the scope and you need headroom for the human ear too.

And arguing ad-nauseum about Johnson noise is a waste of time IMO. Most DACs have several active stages for I/V, filtering and gain. The noise, compression and distortion of these active stages far outweighs the effect of Johnson noise. The noise from the series resistors in the signal path is of more concern. The noise from the ground-loop formed with the preamp or amp is also much higher than either of these. Solve these like I have and then think about how to reduce Johnson noise.

Most DAC designers are chipping around at the edges trying to improve performance rather that going deep to solve the significant roadblocks like jitter, digital filtering, analog stage noise/volume distortion and ground-loop noise. At least these guys are doing something about the #2 problem.

Steve N.

Fascinating post...thanks for posting Steve N.

Enatai252 said:

I chalk some of the claims up to Marketing....try reading PSAudio's press release on how they just reinvented digital. Up to the Buyer's ears to decide. What I have been more surprised about is how much hype has been generated on at least 5 of the forums I follow by a small handful of the same people. A very few own the product, a very few more have heard it and some other just reference "reviews" like Jeff's on wizard's blog like they are fact As for the few of those who do own the product, I am glad they enjoy the product but the claims include so much hyperbole that they are tough to take seriously. I respect Adam's perspective based on his posts I have read but with regard to even some of his claims...I can only say wow. We'll know this hobby is subjective and very personal. If you are excited about a product tell us why and what elements get you closer to the music. Claims that it beats X or Y don't help as many have not heard those either. And the claims come across the wrong way To be clear I am not jealous....I do own the Vivaldi stack....I can afford the Trinity if I decide I like it and since I love the hobby I will try to listen to the Trinity DAC next time I am in NYC....but please keep it real guys....you guys are acting like you are in some exclusive club....I would rather you share what you are hearing

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As an owner of the Trinity dac and Trinity cd-drive and one of the persons who wrote about his experiences with only one purpose - to inform other members on this forum - I feel somewhat puzzled about this reply: always trying to be nuanced and never using stupid phrases like this combo is the best in the world or more or less similar wording I am wondering what the heck are you actually referring to? That said the Trinity combo is mighty impressive and I am not using these words lightly.

Audiocrack said:

As an owner of the Trinity dac and Trinity cd-drive and one of the persons who wrote about his experiences with only one purpose - to inform other members on this forum - I feel somewhat puzzled about this reply: always trying to be nuanced and never using stupid phrases like this combo is the best in the world or more or less similar wording I am wondering what the heck are you actually referring to? That said the Trinity combo is mighty impressive and I am not using these words lightly.

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Audiocrack....reread the posts... I don't think any explanation for my comment is necessary. Your posts and Adam's are very informative on this site and I enjoy reading....you do refer to the Jeff post on AudioExoctics as a data point to consider "The proud owner of the Dcs Vivaldi stack compared his stack to the Trinity dac. See for his findings the Audio Exotics website. Although upfront very skeptical about the qualities of the Trinity dac compared to his Vivaldi stack he clearly preferred the sound created by the Trinity dac. An interesting read I believe! which by your own admission is a commercial site and known for a little sensationalism. To be clear, I am glad you like your new DAC, I would not have bought something at this price point without listening first, but it seems to have worked out very well for you. I also mentioned that I would try to get to NYC to listen especially with RBCDs as it seems to be unclear how the Trinity DAC performs on 16/44. I know you said you like it but since you also did not prefer the Vivaldi over the Scarlatti which I have also listened to both in my system... I am pretty sure our systems, tastes and ears vary widely. I enjoy the hobby and I enjoy auditioning all of the different options out there....keep it real was all I was saying. And again I am truly glad that you are enjoying the DAC....its your money.

Enatai252 said:

Audiocrack....reread the posts... I don't think any explanation for my comment is necessary. Your posts and Adam's are very informative on this site and I enjoy reading....you do refer to the Jeff post on AudioExoctics as a data point to consider "The proud owner of the Dcs Vivaldi stack compared his stack to the Trinity dac. See for his findings the Audio Exotics website. Although upfront very skeptical about the qualities of the Trinity dac compared to his Vivaldi stack he clearly preferred the sound created by the Trinity dac. An interesting read I believe! which by your own admission is a commercial site and known for a little sensationalism. To be clear, I am glad you like your new DAC, I would not have bought something at this price point without listening first, but it seems to have worked out very well for you. I also mentioned that I would try to get to NYC to listen especially with RBCDs as it seems to be unclear how the Trinity DAC performs on 16/44. I know you said you like it but since you also did not prefer the Vivaldi over the Scarlatti which I have also listened to both in my system... I am pretty sure our systems, tastes and ears vary widely. I enjoy the hobby and I enjoy auditioning all of the different options out there....keep it real was all I was saying. And again I am truly glad that you are enjoying the DAC....its your money.

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Thanks for your reply. Btw, in the part you made deep black I was not referring to Jeff but a person 'called' Longius who was new on the AE website. But you are right, everything that is written on this very commercial website should be considered very carefully because they seem to want (very likely for commercial reasons) to create a hype on a regular basis. Futhermore I do not like the arrogant tone of some folks - BTW sometimes I am wondering are they all really different persons or are maybe only a few people contributing under different names? - because they present themselves as their opinion is the only one audio truth which of course is nonsens. Tastes differ and that is part of the fun I would say. So I am very critical regarding the AE website myself but that said some of their products are to my ears excellent indeed such as Tripoint and Trinity. Anyway, I try to be open and honest about what I experience on my 'audio front' without hyping audio components. Are you willing to give a description of your system? In what way do you think are audio tastes differ?

Enatai252 said:

I also mentioned that I would try to get to NYC to listen especially with RBCDs as it seems to be unclear how the Trinity DAC performs on 16/44.

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Enatai252 - just to make things clear: I listen almost exclusively (99%) to 16/44 files. A close friend of mine, who also bought the Trinity after auditioning one at home (among other top-end DACSs) listens exclusively to 16/44 files - he has not a single HD file.

Elberoth said:

Enatai252 - just to make things clear: I listen almost exclusively (99%) to 16/44 files. A close friend of mine, who also bought the Trinity after auditioning one at home (among other top-end DACSs) listens exclusively to 16/44 files - he has not a single HD file.

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Adam

If I understand your set up, you use a CAPS server controlled by JRiver which for your system converts 16/44 to 24/176 and digitally attenuates the signal prior to feeding the DAC ,via USB, since you don't use a preamp My system uses Ethernet into the Upsampler at native rate using Naim HDX server to control NAS and the upsampler than feeds 24/176 or 24/352 to the DAC. I run my DAC at 6V which I prefer to the 2v option

My understanding is also that Trinity DAC has to oversample anything less than 176 prior to feeding its DACs which seems to circumvent all the discussion about no filter. You also lose all the benefits of the internal clocks unless you use USB input. So I will have to listen myself before convinced that RBCD through a transport is truly what this product is designed for. This DAC does not have an Ethernet connection so I would have to compare with either the Vivaldi Upsampler or the Trinity drive to match as closely as possible to how my system is setup. I believe the T drive will then use USB into the DAC. Does the Trinity drive upsample 16/44 prior to sending out via usb

There are so many variables with how CA is delivered that I think comparisons are hard to appreciate unless you really understand the setups of each system and some DACs will likely differ in performance depending on all the ancillary components and input method. Not as easy as comparing a box to a box

I am trying to get to NYC and will report any thoughts here. Getting a direct comparison will be hard as is sometimes the case with more esoteric brands

I am glad you are happy with your system. Based on your room description I think you have a great environment for listening.

I do like the 1704 chip and still will listen to study system which has a CD555 with DR power supply. Bad recordings still sound bad but good recordings sound great

I actually think I would prefer the old Trinity system..,.not sure why it never was a success. Same technology. It was discussed back in 2012 on this forum. Wizard announced a new version in the works but that was expected to be priced at 10-15k euros. Wonder if I can find one of the V1 models to listen to

Enatai252 said:

I actually think I would prefer the old Trinity system..,.not sure why it never was a success. Same technology. It was discussed back in 2012 on this forum. Wizard announced a new version in the works but that was expected to be priced at 10-15k euros. Wonder if I can find one of the V1 models to listen to

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There are 4 features that make the Trinity DAC so special:

- lack of the analog reconstruction filter and output stage + the option to run the DAC with no oversampling
- 1704 R2R DACs is a proprietary, time staggered arrangement
- USB input (not uncommon nowadays, but necessary for the next feature)
- ultra expensive, ultra low phase noise clock, which is 1000x more accurate than the clocks dCS uses for example

The old model lacked the last two features, which are a key elements for the performance of the new Trinity DAC. I'm not sure why anyone would like to get the old one over the new one - unless you want to use it exclusively with SPDIF sources (and throw away the available performance that comes with their USB solution).

Anyhow, I do not believe the old Trinity was ever meant to be a best selling product - with the price of 60k euro, it would be simply impossible (to put things at a proper perspective: in 2005 when the Trinity DAC V1 was launched, the top of the range dCS Elgar Plus, which was considered state of the art at that time, was 15k euro; Trinity DAC was … 60k euro; only a lucky few were able to afford one). Just like the Bugatti Veyron or a MacLaren P1 will never become popular cars you see everyday.

Enatai252 said:

If I understand your set up, you use a CAPS server controlled by JRiver which for your system converts 16/44 to 24/176 and digitally attenuates the signal prior to feeding the DAC ,via USB, since you don't use a preamp.

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That is not correct. I do not upsample in Jriver from 44 to 176kHz (although I could - it may sound better vs using the oversampling in the DAC, depending which algorithm is more transparent; I need to try this one day). The only manipulation of the digital signal I do is changing the bit depth of the output signal from 16 to 24 bit. It is done to make greater room for the digital attenuation. Digital attenuation performed on 16-bit signals is not very transparent.

Enatai252 said:

My understanding is also that Trinity DAC has to oversample anything less than 176 prior to feeding its DACs which seems to circumvent all the discussion about no filter.

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There are TWO filters in every DAC: analog reconstruction filter and digital oversampling filter. The Trinity has no analog reconstruction filter and no output stage apart from a simple I/V stage. The digital oversampling filter is switchable. It is a convenience feature more than anything else, which allows, among other things, the DAC to be used with regular SPDIF sources. When you feed the DAC with 44-96kHz signals, the digital oversampling filter is engaged – so it is just like with 99% DACs available on the market today (including EMM, dCS, Metronome etc). When using a computer source, you have an option to perform upsampling in the computer and run the DAC completely filterless with no digital filtering. Depending on the quality of the oversampling algorithm used it may sound better then the oversampling filter or … it may not.

Enatai252 said:

You also lose all the benefits of the internal clocks unless you use USB input.

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That is not the Trinity’s fault. That is the limitation of SPDIF interface. Thank Sony and Philips for that. With the SPIDF interface, the DAC has to lock to the incoming clock from transport. No way to get around this, unless you send the clock signal back from the DAC to the transport on a separate line (but then it is no longer a SPDIF interface). USB interface allows you to use a free running clock inside the DAC, with no need to lock to the jittery transport clock, no jitter inducing PLLs. Only in such an arrangement, the potential of this ultra low nose clock can be relised.

Elberoth said:

(...) That is not correct. I do not upsample in Jriver from 44 to 176kHz (although I could - it may sound better vs using the oversampling in the DAC, depending which algorithm is more transparent; I need to try this one day). The only manipulation of the digital signal I do is changing the bit depth of the output signal from 16 to 24 bit. It is done to make greater room for the digital attenuation. Digital attenuation performed on 16-bit signals is not very transparent. (...)

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Adam,
What is the typical attenuation you must use in Jriver?

Only 7-9 dB. The MSB monos have very low gain - 20dB.

Elberoth said:

...There are TWO filters in every DAC: analog reconstruction filter and digital oversampling filter. The Trinity has no analog reconstruction filter and no output stage apart from a simple I/V stage. The digital oversampling filter is switchable...

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It sounds like you are familiar with the inner workings of the Trinity DAC. Do you know what the functioning of alternate oversampling filter option is? Regarding what makes this DAC sound the way it does, I must wonder how much of it's sound is due simply to the massive paralleling of 8 D/A chips. One other comment, if the I/V stage is active, then I think it qualifies as an output stage.

Thanks for the clarifications Adam

I have found that in my experience the Ethernet connection directly into my Upsampler/DAC is more transparent and more resolving than USB (and less complicated to be honest) which I believe is consistent with some of the points you have made about the challenges with isolation and noise when using USB... so it does not appear that the Trinity DAC would suit my current setup but would be more of an alternative to it which would require a change in how I currently deliver ripped music to my system. I have found that there is a difference in SQ depending on whether I use the QNAP NAS music server or the Naim HDX server and I imagine there would be differences in a USB setup based on the server, software and USB cable. I think the best way for me to compare would be use the Trinity Drive/DAC with a computer USB solution as compared to the Ethernet source into Upsampler/DAC for the Vivaldi which I find to be the best combo in my system. Not sure others compare that way as not many details given. It also sounds like the V1 version of the Trinity DAC would be something I would be more interested in....16 DACs per channel must be better than 8 per channel right? as 100x more accurate clocks are better right? maybe a V3 on the horizon soon. BTW, the dcs clock spec is .1ppm once stabilized so I think the correct statement is 100x...although I doubt there is any audible difference after a certain point of stability.

I don't believe that the price of the V1 was any deterrent to success any more so than today's price as the subset of those willing/able to pay the price of the DAC/Tdrive is very small in both cases. (many more able than willing)

Finally, it sounds like you have a system you really enjoy which is great, I also was not fond of the MSB but you seem to have found the sweet spot for your system....I will have to try to listen for myself soon. If I filter all of the AE forum sensationalism and the claims of badly recorded 80's CDs or internet radio sounding better than current hi-rez recordings....which I discount heavily. I do believe that both you and Audiocrack have provided insightful comments on the DAC performance and that the DAC should be on any list, along with several others that have been mentioned here, for those wanting a high quality digital playback system. Cheers

Reading the description of the Trinity DAC, it seems the advantages are the elimination of the analog reconstruction filter and a better impulse response.

Now, I wonder if the same results can't be accomplished by using only digital oversampling instead of digital+analog oversampling like in the Trinity for sample rates lower than 176.4kHz.

To be more specific, I think at the Chord QBD76 way of doing it (delta sigma DAC with 2608 times oversampling for 44.1kHz sample rates), though they are not clear if they eliminate the analog filter.

From its description:

"
WTA Filter
The WTA filter algorithm has taken nearly thirty years of research to develop. It solves the question as to why higher sampling rates sound better. It is well known that 96 kHz (DVD Audio) recordings sound better than 44.1 kHz (CD) recordings. Most people believe that this is due to the presence of ultrasonic information being audible. What is not well known is that 768 kHz recordings sound better than 384 kHz and that the sound quality limit for sampling lies in the MHz region. 768 kHz recordings cannot sound better because of information above 200 kHz being important – simply because musical instruments, microphones, amplifiers and loudspeakers do not work at these frequencies. So if it is not the extra bandwidth that is important, why do higher sampling rates sound better?

The answer is not being able to hear inaudible supersonic information, but the timing of transients. It has long been known that the ear/brain can detect differences in the phase of sound between the ears to the order of microseconds. This timing difference between the ears is used for localising high frequency sound. Since transients can be detected down to microseconds, then the recording system needs to resolve timing of one microsecond implying sampling of 1 MHz.

However, 44.1 kHz sampling is capable of accurately resolving transients – if the FIR filters have infinite long tap lengths. Sampling theory is proved by having brick wall filters that span from the birth to the end of the universe – not practical in a real world filter. All reconstruction filters have relatively short tap lengths – the largest commercial device is only about 256 taps. It is due to this short tap length and the filter algorithm employed that generates the transient timing errors. These errors turned out to be very audible. Going from 256 taps to 2048 taps gave a massive improvement in sound quality – much smoother, more focused sound quality, with an incredibly deep and precise sound stage.

The initial experiments used variations on existing filter algorithms. Unfortunately, going from 1024 taps to 2048 taps gave a very big improvement in sound quality, and it was implying that almost infinite tap length filters were needed for the ultimate sound quality. At this stage, a new type of algorithm was developed – the WTA filter. This was designed to minimise transient timing errors from the outset, thereby reducing the need for extremely long tap lengths. The WTA algorithm was a success – a 256 tap WTA filter sounds better than all other conventional filters, even with 2048 taps. WTA filters still benefit from long tap lengths; there is a large difference going from 256 taps to 1024 taps.

The DAC64 used 1024 taps, with a WTA algorithm optimised for good stop-band, inband ripple and minimizing transient errors. With the superior capacity of the Spartan 3 FPGA, four 70 bit DSP cores have been designed, which allowed tap length to be increased from 1024 taps to 4096 taps. But this has not been the sole improvement; 4096 taps allowed substantial changes to the WTA algorithm, as longer tap lengths gives greater flexibility in the algorithm. Optimising the algorithm for the longer tap lengths took considerable effort, with hundreds of listening tests. The improvements to the algorithm and the increased tap lengths gave improved accuracy in reconstructing the transient edges. This gave huge subjective improvements – better sense of timing, more soundstage depth and better imagery, and improved pitch reproduction, with particular improvements in bass definition.

5th Generation Pulse Array DAC.
The QBD76 also features considerable improvements to the Pulse Array DAC, in particular the noise shaper architecture. All high-end DAC’s today, like Pulse Array, are delta-sigma DAC’s that employ noise shapers and an output truncator. The output truncator takes the high resolution signal and truncates it to suit the DAC’s output resolution. Now the action of truncation increases noise, and generates distortion. The function of the noise shaper is to reduce the noise within the audio bandwidth, and it does this with the use of integrators. The number of integrators determines the order of the noise shaper. However, the noise shaper does an excellent job of reducing inband distortion and noise (that is audio bandwidth distortion and noise), but is less good at high frequencies. High frequency noise and distortion needs to be filtered out with analogue circuitry, otherwise this noise will generate in-band distortion in the pre and power amplifiers. Indeed, one can hear the effects of very small amounts of out of band noise, as a hardness or grain in the treble, and it is very important to reduce this out of band noise as much as possible.

Considerable work has been done to improve this aspect of Pulse Array, and new noise shaper architecture has been developed, with the specific aim of reducing this noise. Amongst other improvements, 5th generation Pulse Array now has 8th order noise shaping (the highest order of noise shaping of any known DAC), and 2608 times oversampling and digital filtering with 44.1 kHz inputs. The effects of these improvements in out of band noise are shown in the chart below:



This chart shows the distortion and noise from the noise shaper and truncator, and compares the new 5th generation Pulse Array, to the 4th generation Pulse Array, and also illustrates the performance of a commercial “state of the art” high-end DAC. Note that this is just the digital performance of the noise shaper, it assumes perfect analogue performance!

At 20 kHz, distortion and noise for the high end DAC is only -145 dB. Since the distortion and noise is a measure of the resolution of the noise shaper, in practice one requires that this resolution to be several orders of magnitude greater than the required performance. The high end DAC is not able to resolve 24 bit data correctly. 4th generation Pulse Array is much better, at 218 dB, but 5th generation is superb at -298 dB. This means that 5th generation Pulse Array’s noise shaper’s resolution is effectively perfect within the audio bandwidth.

At 200 kHz, the high end DAC is at -87 dB resolution, so large analogue filtering will be required. These extra analogue components will degrade sound quality. 4th generation Pulse Array produces a creditable -126 dB, so little filtering is needed. But 5th generation is at -152 dB – that is better than 24 bit performance at 200 kHz!"

Anatta said:

Reading the description of the Trinity DAC, it seems the advantages are the elimination of the analog reconstruction filter and a better impulse response.

Now, I wonder if the same results can't be accomplished by using only digital oversampling instead of digital+analog oversampling like in the Trinity for sample rates lower than 176.4kHz...

Click to expand...

I'm uncertain as to what you refer to by "digital+analog oversampling" with respect to Trinity's reconstruction filter implementation. Are you saying that Trinity also utilize some digital reconstruction filter? Not disagreeing with you here, I simply don't know exactly what they have implemented.

Setting the above question aside for the moment, with oversampled interpolation the ultrasonic image bands will still be present, but shifted up in frequency by a factor equal to the oversampling ratio. So, while the typical analog filter can be designed to have a lessor subjective impact on the audio band frequencies, it should still be present in some form. That assumes that the first ultrasonic image alias band has been strongly filtered away. Which, in a simple first-order analog interpolation scheme without any other strong anti-image filter, such as that apparently employed by Trinity, it will not have been. The improved impulse response comes at the cost of incorrect signal reconstruction due to the ultrasonic image frequencies not having been cut-off. This is not too dissimilar to how a non-oversampling DAC performs, except that there will be musch less severe audio band treble response roll-off with oversampling due to it's necessarily shorter D/A converter output hold period.

Linear interpolation can indeed be obtained using strictly digital means by implementing a oversampling moving-average digital filter with a single D/A converter, and so, at much lower hardware cost too. However, linear interpolation via multiple time-staggered parallel D/A converter analog output summation will eliminate the potential for certain digital processing error artifacts, as well as lower the thermal noise-floor.