[Gadget67]

I understand this sentiment completely. If it wasn't for Covid lockdown still in my area I would probably revert to my DAP, iem and a few good tunes.

I can totally relate to that! Mine is covered with dust.

 

[jarnopp]

Hello Rob ,But what about 24/96 and other,not bandwidth limited signals,hi res formats?

I am still a bit puzzled why for example a cymbal crash or a triangle in a real live situation contains overtones up to 100khz , but bandwidth limiting cuts those out around 20khz as I guess it also does with aliazing artifacts.
But what if even if we ,can´t hear those supersonic harmonics ,overtones , they may still be an integral part of an instrument´s orginal timbre?

If PGGB does not use apodizing filters with hi res and apply many millions more taps than even Mscaler with no such slow roll off filter would that not in theory make it possible to get even closer to the real sound of acoustic instruments?

A violin reaches above 30khz in natural harmonics does it not?

During your lecture in Singapore a few years ago you mentioned that to perfectly recover a 24/96 signal as Mscaler recovers a bandwidth limited 16/44.1 one would need 256M taps?

What if this new software has actually achieved that goal with 24 bit signals?

And if it hasn´t, but you can achieve that goal with your new computer and software code instead of FPGA based I guess that would be very welcome.

I suspect you would personally make even more income from offering such a product as download,to music lovers directly at a similar price PGGB sells for than from the relatively few who can afford hardware based upscaling as in Mscaler.

I think millions more music lovers would be interested to get as good as ,or better than current mscaling ,for 500USD instead of paying 5000-10000USD.


Sorry if my questions are amateurish again,but I am as always, genuinely interested in both the tech facts involved and in getting as close as possible to what I have heard from direct mic-feed at classical recording sessions.
PS. My Mscaler sounds fantastic with both rbcd and hi res, but I would much rather travel with two or three 4tb portable harddrives and pre-upscaled music,than having to worry about leaving my expensive Mscaler in a hotel room again.

Cheers CC

I think we are confusing music recording with music reproduction. Rob always states that his goal is to reproduce the original bandwidth limited signal. That is what the recording is. If there are supersonic or subsonic characteristics that influence the human audible frequency, then those effects should be captured in the bandwidth limited signal and already be there to reproduce. I guess his might be similar to X-rays and film. We can’t see them and there is no point in trying to reproduce them, unless they have had an effect on the (human) visible portion.

Of course, none of this answers the question of when PGGB improves the sound. I think we are all interested in exploring that.

 

[Christer]

I understand this sentiment completely. If it wasn't for Covid lockdown still in my area I would probably revert to my DAP, iem and a few good tunes.

And I would be at a live symphonic concert asap.
Cheers CC

 

[524744]

Dear fellow audiophiles, after enjoying all the posts on this forum I humbly wish to share with you how I achieved my present end-game Chord system with Wireworld, Lieder & Oyaide solid silver cables and Dynaudio Special 40 speakers & Sennheiser HD800S headphones. As this is my first post, I beg your indulgence.

I live in a small apartment so the Chord system comprising the TT2, MScaler, TToby with Dynaudio speakers is optimum in terms of SQ. Source equipment comprises Moon 260T transport, Denon DCD1520AE CD player & Naim Uniti Core.

The music emitting from my system is warm, extremely detailed and the timbre is truly magnificent. I opine that this fabulous SQ is mainly due to the TT2/MScaler combination with the rather transparent Wireworld solid silver cables. I find the vocals full, focused and detailed. Instruments sound natural and full of life. I especially enjoy the piano notes derived from Reference Recording's Nojima Plays Liszt - can literally hear every single note played and when Nojima steps on the pedal.

Would appreciate any advice from anyone how I can improve the system further without spending too much as I have used up my allotted hi-fi funds. Thanks to all for making life more enjoyable.

 

[andrewd01]

Dear fellow audiophiles, after enjoying all the posts on this forum I humbly wish to share with you how I achieved my present end-game Chord system with Wireworld, Lieder & Oyaide solid silver cables and Dynaudio Special 40 speakers & Sennheiser HD800S headphones. As this is my first post, I beg your indulgence.

I live in a small apartment so the Chord system comprising the TT2, MScaler, TToby with Dynaudio speakers is optimum in terms of SQ. Source equipment comprises Moon 260T transport, Denon DCD1520AE CD player & Naim Uniti Core.

The music emitting from my system is warm, extremely detailed and the timbre is truly magnificent. I opine that this fabulous SQ is mainly due to the TT2/MScaler combination with the rather transparent Wireworld solid silver cables. I find the vocals full, focused and detailed. Instruments sound natural and full of life. I especially enjoy the piano notes derived from Reference Recording's Nojima Plays Liszt - can literally hear every single note played and when Nojima steps on the pedal.

Would appreciate any advice from anyone how I can improve the system further without spending too much as I have used up my allotted hi-fi funds. Thanks to all for making life more enjoyable.

I think you have a great system. I am a huge fan of Dynaudio speakers and have been using them for the last 14 years (Contour S3.4).
To be honest I don’t think I would throw any more money at what you have - it ends up being an expensive rabbit hole.

 

[524744]

I think you have a great system. I am a huge fan of Dynaudio speakers and have been using them for the last 14 years (Contour S3.4).
To be honest I don’t think I would throw any more money at what you have - it ends up being an expensive rabbit hole.

thanks andrewd01 for your kind advice.

 

[Spawn300Z]

Dear fellow audiophiles, after enjoying all the posts on this forum I humbly wish to share with you how I achieved my present end-game Chord system with Wireworld, Lieder & Oyaide solid silver cables and Dynaudio Special 40 speakers & Sennheiser HD800S headphones. As this is my first post, I beg your indulgence.

I live in a small apartment so the Chord system comprising the TT2, MScaler, TToby with Dynaudio speakers is optimum in terms of SQ. Source equipment comprises Moon 260T transport, Denon DCD1520AE CD player & Naim Uniti Core.

The music emitting from my system is warm, extremely detailed and the timbre is truly magnificent. I opine that this fabulous SQ is mainly due to the TT2/MScaler combination with the rather transparent Wireworld solid silver cables. I find the vocals full, focused and detailed. Instruments sound natural and full of life. I especially enjoy the piano notes derived from Reference Recording's Nojima Plays Liszt - can literally hear every single note played and when Nojima steps on the pedal.

Would appreciate any advice from anyone how I can improve the system further without spending too much as I have used up my allotted hi-fi funds. Thanks to all for making life more enjoyable.

Hello,

I have a very similar setup to yours but I went with Mark & Daniels Maximus Mini+s. I also have dual TTobys. It’s a very good setup. You don’t have do do anything really.

I would only suggest may something like the Hifiman HE-1000 SE. This is a great headphone off your setup without breaking the bank. You can find them used on here.

However, I’m thinking about adding a tube amp to my setup. Then I think I’m done. But I love the form factor of the Chord Stack. Enjoy your setup man.

 

[Hooster]

Would appreciate any advice from anyone how I can improve the system further without spending too much as I have used up my allotted hi-fi funds. Thanks to all for making life more enjoyable.

Your system is pretty much as good as it gets. Improve it by purchasing better snacks and drinks. Also, take your time to explore music. There is so much available on streaming services such as Qobuz.

 

[524744]

Hello,

I have a very similar setup to yours but I went with Mark & Daniels Maximus Mini+s. I also have dual TTobys. It’s a very good setup. You don’t have do do anything really.

I would only suggest may something like the Hifiman HE-1000 SE. This is a great headphone off your setup without breaking the bank. You can find them used on here.

However, I’m thinking about adding a tube amp to my setup. Then I think I’m done. But I love the form factor of the Chord Stack. Enjoy your setup man.

Thanks for your suggestions. May I enquirer why do you have twin TTobys and how do you connect them.
I have also thought about adding tube amp but In my experience with it I found hum to be an obstacle to the SQ.
Please advise in what way the Hifi Man HE-1000SE is better than my HD800S for my present setup.

 

[524744]

Your system is pretty much as good as it gets. Improve it by purchasing better snacks and drinks. Also, take your time to explore music. There is so much available on streaming services such as Qobuz.

Thank you for your kind words. Over the last 3 decades I have amassed 3000 CDs which I have ripped into the Uniti Core. For casual listening I play music from the Uniti but for serious listening I still prefer listening to CD played on Moon transport. I have wondered about streaming but my some friends advised that SQ is better from CD. Please give me your thoughts on CD vs Quobuz. Would very much appreciate your input. BTW I am one of those who are into 60s music!!!

 

[Spawn300Z]

Thanks for your suggestions. May I enquirer why do you have twin TTobys and how do you connect them.
I have also thought about adding tube amp but In my experience with it I found hum to be an obstacle to the SQ.
Please advise in what way the Hifi Man HE-1000SE is better than my HD800S for my present setup.

I run them from XLR with a custom adapter. I plan to get the Elac Vela BS-403s or the B&W 8O5 D3s. So I plan run the TTobys Bi-Amped one for each speaker. A lot like how they did here. But until I get the speakers I’m using one to drive my Susvaras and the other my Mark & Daniel speakers.

I feel the HE-1000SE are so much more musical. They scale amazingly well. It is very transparent and clean sounding, but faster and with a better body then the Sennheiser HD800S. They are a perfect pairing with your system. I really enjoy them with my setup.

 

[524744]

I run them from XLR with a custom adapter. I plan to get the Elac Vela BS-403s or the B&W 8O5 D3s. So I plan run the TTobys Bi-Amped one for each speaker. A lot like how they did here. But until I get the speakers I’m using one to drive my Susvaras and the other my Mark & Daniel speakers.

I feel the HE-1000SE are so much more musical. They scale amazingly well. It is very transparent and clean sounding, but faster and with a better body then the Sennheiser HD800S. They are a perfect pairing with your system. I really enjoy them with my setup.

wow!! Appreciate the prompt reply. Shall try to audition the HE-1000SE asap.

 

[Hooster]

Thank you for your kind words. Over the last 3 decades I have amassed 3000 CDs which I have ripped into the Uniti Core. For casual listening I play music from the Uniti but for serious listening I still prefer listening to CD played on Moon transport. I have wondered about streaming but my some friends advised that SQ is better from CD. Please give me your thoughts on CD vs Quobuz. Would very much appreciate your input. BTW I am one of those who are into 60s music!!!

I mostly listen via an Oppo UDP-205 which has an in built CD transport and it also accepts usb from my computer. To me there is nothing to be gained by listening to a CD vs streaming from my PC (which according to many purists is "bad".) In fact I actually prefer the sound of streamed music from Quobuz vs the same music off a CD. I have no idea why that is, but it is what it is.
The advice of friends is fine but I suggest you try streaming from a PC and see what you think. Just connect it to your mscaler. Quobuz has a free trial period so you have nothing to lose.

 

[524744]

I mostly listen via an Oppo UDP-205 which has an in built CD transport and it also accepts usb from my computer. To me there is nothing to be gained by listening to a CD vs streaming from my PC (which according to many purists is "bad".) In fact I actually prefer the sound of streamed music from Quobuz vs the same music off a CD. I have no idea why that is, but it is what it is.
The advice of friends is fine but I suggest you try streaming from a PC and see what you think. Just connect it to your mscaler and see what you think. Quobuz has a free trial period so you have nothing to lose.

Thanks for advice.

 

[Rob Watts]

Part of me feels a bit weary to have this further potential thing, I’m not sure why, why aren’t I more excited to try this pggb, maybe I just feel like I’m time poor and maybe not a great audiophile lol.

I think your feelings would be transformed if this thing actually delivered on reproducing music with emotion; but it singularly fails to do this. I would go to any lengths to genuinely improve my ability to enjoy music.

Has anyone discovered a readily available case for the Hugo TT 2?

On another note, I just want to thank Rob Watts for the amazing engineering that went into the Hugo TT 2. I am listening to the TT 2 with the Oriolus Trailii for the first time tonight having just registered my new unit. And it is truly one of those in love smile moments

Thank-you.

Rob,
The section of the FAQ quoted below adds more information that seems to contradict the statement above that it is a slow roll-off filter. They state that it is a windowed sinc function. Obviously the only way to know for sure would be to process the file with a known windowed sinc filter and compare the results to a PGGB processed file at the sample level.

“Does PGGB use Apodizing filters, and why?​

Apodization in Greek means cutting off the foot. It has different technical meaningsdepending on the application. In Signal processing, it just means using a windowing function to reduce ringing artifacts due to the abrupt truncation at the beginning and end of a sample window. In digital-audio, the term has been used and also misused. In digital-audio, more often than not 'apodizing' is used to mean it is a non-brick-wall filter which has reduced or no 'pre-ringing'. It is implemented as a slow-roll off filter. PGGB's filters do not fall under this category.

The problem is getting hung-up on the terminology and forgetting what we are trying to achieve. With apodizing filters used in PGGB, you should forget everything else you may find online about apodizing filters. We used the term 'apodizing' because it is immediately understood to mean it has something to do with improving CD audio. It is not a slow roll off filter, it is not minium-phase. It uses a windowed sinc filter. We could call it something else, but we do not want to add to the list of myriad jargons. It is apodizing in the frequency domain in the sense that it 'cut-off' aliasing artifacts introduced during CD creation and in our humble opinion, it is the only meaningful thing one can do to alleviate if not reverse the damage done to CD audio.”

If it is a windowed sinc function it is no longer true sinc following Whittaker-Shannon interpolation filter - the only filter that will reconstruct the original bandwidth signal perfectly. But what I don't understand is why they are using windowed sinc at all for HD recordings as they claim this is not being apodized. Doing it off-line with unlimited time, and doing the sinc function appropriately (you must pre-process and post process the file correctly) means you can in practice do a pretty much good approximation to true Whittaker-Shannon interpolation (given the constraints of 64 bit floating point which is another big problem). Windowed sinc is categorically not true Whittaker-Shannon interpolation.

The limitations of 64 bit floating point is actually a big problem. Floating point creates noise floor modulation, as the resolution is being modulated by the exponent. The level of noise floor modulation this produces is audible - and every time I hear processed audio I hear the same thing - a bright, slightly hard sound quality. This is entirely consistent with noise floor modulation. The WTA filters actually work to a higher internal accuracy than 64b FP, and the coefficients and outputs are all noise-shaped. This guarantees zero noise floor modulation.

Hello Rob ,But what about 24/96 and other,not bandwidth limited signals,hi res formats?

I am still a bit puzzled why for example a cymbal crash or a triangle in a real live situation contains overtones up to 100khz , but bandwidth limiting cuts those out around 20khz as I guess it also does with aliazing artifacts.
But what if even if we ,can´t hear those supersonic harmonics ,overtones , they may still be an integral part of an instrument´s orginal timbre?

If PGGB does not use apodizing filters with hi res and apply many millions more taps than even Mscaler with no such slow roll off filter would that not in theory make it possible to get even closer to the real sound of acoustic instruments?

A violin reaches above 30khz in natural harmonics does it not?

During your lecture in Singapore a few years ago you mentioned that to perfectly recover a 24/96 signal as Mscaler recovers a bandwidth limited 16/44.1 one would need 256M taps?

What if this new software has actually achieved that goal with 24 bit signals?

And if it hasn´t, but you can achieve that goal with your new computer and software code instead of FPGA based I guess that would be very welcome.

I suspect you would personally make even more income from offering such a product as download,to music lovers directly at a similar price PGGB sells for than from the relatively few who can afford hardware based upscaling as in Mscaler.

I think millions more music lovers would be interested to get as good as ,or better than current mscaling ,for 500USD instead of paying 5000-10000USD.


Sorry if my questions are amateurish again,but I am as always, genuinely interested in both the tech facts involved and in getting as close as possible to what I have heard from direct mic-feed at classical recording sessions.
PS. My Mscaler sounds fantastic with both rbcd and hi res, but I would much rather travel with two or three 4tb portable harddrives and pre-upscaled music,than having to worry about leaving my expensive Mscaler in a hotel room again.

Cheers CC

I will handle your question re natural harmonics later.

But you say "...you mentioned that to perfectly recover a 24/96 signal as Mscaler recovers a bandwidth limited 16/44.1 one would need 256M taps?"

Yes I did, but I actually said to recover - guaranteed to better than 24 bits - you would need coefficients that were identical to the coefficients in an ideal Whittaker-Shannon interpolation filter - and in the case of WTA where half of the coefficients are Whittaker-Shannon (that is ideal sinc) and to do this you then need about 256M taps.

But - the key here is the number of taps that are Whittaker-Shannon - NOT the number of taps. If you were to use a filter to filter above 20kHz - call it apodizing or slow roll off or whatever, like the PGGB for 44.1kHz then NO taps would be Whittaker-Shannon - and that would mean a filter using billions of the wrong coefficients will simply return the wrong result, so the long taps are completely useless.

I have been working on the theory of interpolation since 1981. And back then I realised that to properly reconstruct the timing of transients (which is absolutely essential for transparency and musicality) you needed to use a Whittaker-Shannon type filter. Then in 1998 I started actually designing and listening to these filters built on those principles - and hence the WTA filter. Since then, I have done thousands of listening tests fine tuning the WTA, and trying to understand what is going on. The most important thing I learned (there is a lot more to this subject than I can talk about) is that the majority of the interpolation coefficients must be Whittaker-Shannon - and if you do not do this, then transient timing is corrupted, meaning that instruments start modulating one another's timing. When that happens all hell breaks loose perceptually, as the timing of transients is the most important perceptual cue that the brain uses to reconstruct the audio illusion. There is truly something very unique about Whittaker-Shannon coefficients and the moment you design a filter that is not Whittaker-Shannon then musicality is destroyed. Of course it is very much more complex than this, as the change from ideal Whittaker-Shannon coefficients to zero (where the filter isn't responding to past or future data) is vitally important too.

A final point - Whittaker-Shannon interpolation will perfectly reconstruct the timing of transients. But we don't know what level of transient uncertainty the brain needs (I think it's close to zero) and the only way of telling is to keep on making it closer to Whittaker-Shannon as possible - or rather keep the total uncertainty to be as small as possible, which means Whittaker-Shannon. So I don't know how many ideal coefficients are needed to give us complete transparency - but this is something I am very much working on.

I think we are confusing music recording with music reproduction. Rob always states that his goal is to reproduce the original bandwidth limited signal. That is what the recording is. If there are supersonic or subsonic characteristics that influence the human audible frequency, then those effects should be captured in the bandwidth limited signal and already be there to reproduce. I guess his might be similar to X-rays and film. We can’t see them and there is no point in trying to reproduce them, unless they have had an effect on the (human) visible portion.

Of course, none of this answers the question of when PGGB improves the sound. I think we are all interested in exploring that.

Agreed. I have recently been doing initial listening tests on my bandwidth limiting filters for the ADC project, and they sound very, very good indeed - no evidence of any loss of transparency. So at this stage there is zero evidence that we need signals above 20 kHz - but what we do need are transients that have the same timing as the original, with zero uncertainty. HD recordings help that process, but are not innately essential to do this.

Having said that, my conclusions are tentative (actually all of my thinking/theories/conclusions are tentative, subject to continuous re-appraisal via carefully controlled and characterised listening tests and objective analysis). Once the ADC project is up and running, then I will know a great deal more, and should be able to know for certain whether higher order harmonics above the audio band is important.

Anyway - it's important to remember the only thing that is really important is getting emotional whilst listening to music. So happy listening!