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Nebula3 cutting edge? Yes and no!

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Re: Nebula3 cutting edge? Yes and no!

Postby Gstell » Sat Oct 15, 2011 11:54 pm

Darren,
This post was not supposed to be about me and my capabilities but an incentive for others to listen to the material themselves in their environment with their equipment and their set of ears. The ultimate goal is to convince Giancarlo to implement these formats (because of their sound and not their specifications) into NAT and Nebula so we do not have to sample everything in 5-10 years again.

Said that. Every day I start of with specific boosts and then cuts at 32, 64, 125, 250, 500, 1k,2k,4k,8k,and 16k using pink noise and some of the reference records I listed. I do use different headphones and speakers as well as converters I own.

This routine allows me quickly to learn the differences in resolution (graininess) and frequency response issues (most prominent in 64hz region) of a particular device.
I have learned this technique via a teacher at the conservatory of Music who encouraged us not to take our "perfect-pitch" hearing for granted but improve upon it by hearing into different sounds.


monitors: Adam P11, A7, Samson Rubicon, Velodyne HGS 10, Sennheiser HD600, HD280, other cheap in ear.
Converters: Mytek 182X8, Mytek96, RME DA, Creamware ADDA,MP3player.




Listening Room specifications: (Concrete floor ceiling and walls)
Dimensions; 250"x687"x9-10'

Treatments:
1 box 10'X 339" depth 170.5", 5-7" fiber wool cover =79.202
1 box 10'X 173" depth 393", 5-7" fiber wool cover =34.3612

4 corner traps at 10'long 26"deep = 519.3822
4 corner traps at 140" long 13" deep =1038.7644
1 corner traps at 104" long 13" deep =1038.7644

2 boxes 32"x82" 27" deep covered with 5" fiber-wool =500.1458
1 centered box 2 boxes 32"x82" 27" deep to be used as reflector

rest of walls and ceiling fiber-wool 5-7" =2700.7874-1929.1339
carpet covered Plywood floor 4"-1' above Concrete.

Still under construction!
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Re: Nebula3 cutting edge? Yes and no!

Postby giancarlo » Sun Oct 16, 2011 6:16 pm

ok starting from 1.3.500c we support 352800 and 384000. I fixed also conversion from 48000 and 88200 (it was not working before).
It was fixed both in nebula and nat.

About other points, nat already manages 32 and 64 bits wav recording.
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Re: Nebula3 cutting edge? Yes and no!

Postby Gstell » Wed Oct 19, 2011 2:24 am

Thank you Giancarlo.
That is great news.
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Re: Nebula3 cutting edge? Yes and no!

Postby Gstell » Thu Oct 20, 2011 11:50 am

ngarjuna
If one is conservative DSD64 (which is what SACD uses) seems to have a resolution of 40-60khz
DSD128 doubles these specifications.
I am guessing that DAD is onto something when claiming DSD is very beautiful (maybe an effect that is similar to what some people contribute to tape or tubes???) and 32bit/384khz PCM being very transparent.
I am hopeful that in the Nebula world we can mix and match those two attributes in our recordings.

I did some pink noise recordings from a MiniMoog into Mytek192x8 and discovered that there is a drastic increase of high frequencies between 44.1 and 96khz and what seems to be a drastic step of increased depth/definition (or what one wants to call it) between 96 and 192khz. I will do an additional recording in DSD and then post all the files here.

Understanding this loss in clearly audible high frequency content makes me see how prominent the low pass filter really influences PCM recordings. I am guessing that pre/post-ringing as well as other filter artifacts (artifacts above 20khz are well known to fold down into the audible frequency range!) in combination with timing insensitivity are the primary culprits of the particular sound of lower resolutions.
I do not quiet understand the effect of a converters analog/digital hardware on sample-rate and bit depth but one may hear the result of different hardware and software implementations listening to a variety of converter models next to each other.
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Re: Nebula3 cutting edge? Yes and no!

Postby ngarjuna » Thu Oct 20, 2011 2:30 pm

gstell wrote:ngarjuna
...
I did some pink noise recordings from a MiniMoog into Mytek192x8 and discovered that there is a drastic increase of high frequencies between 44.1 and 96khz and what seems to be a drastic step of increased depth/definition (or what one wants to call it) between 96 and 192khz. I will do an additional recording in DSD and then post all the files here.

I'm not really sure what you mean by drastic here but in normal comparisons (music as opposed to broadband noise) 96k nulls with 44.1 down below -70dB. I think drastic is overstating it.

We come from a technological landscape with far lower fidelity than the tools we have now. And, despite having these fancy tools, the ultra-clear super high sample rates have yet to produce a Dark Side of the Moon or Sgt. Pepper's or Thriller. The fact is we've had sufficient technology to record professional sounding records since the late 50s. That has never changed or ceased to be the case. This quest to always get more and more fidelity, while understandable (since it is exactly the kind of hobby engineers would be prone to), is a fruitless pursuit. You can chase the dragon endlessly but when it's all said and done it's all about what actually got produced (not what could have been with .00001% greater fidelity). And that just doesn't have much to do with ultrasonic sample rates. If we lacked the technology to make accurate recordings it would be one thing; but these infinitesimally small increments are simply not adding anything significant to the production landscape despite the fact that in some careful, well orchestrated listening tests they occasionally demonstrate an audible difference (keeping in mind that sometimes these tests do not demonstrate an audible difference).

Understanding this loss in clearly audible high frequency content makes me see how prominent the low pass filter really influences PCM recordings. I am guessing that pre/post-ringing as well as other filter artifacts (artifacts above 20khz are well known to fold down into the audible frequency range!) in combination with timing insensitivity are the primary culprits of the particular sound of lower resolutions.
I do not quiet understand the effect of a converters analog/digital hardware on sample-rate and bit depth but one may hear the result of different hardware and software implementations listening to a variety of converter models next to each other.

The low pass filter is the main difference from one sample rate to another (and in many cases from one converter to another). But the attenuation at 44.1 is usually pretty close to 20k, not exactly a frequency area rich with melodic content. It can be easily compensated with a very gentle EQ boost at 20k or, as is probably more common, gets compensated for seamlessly when you are mixing at 44.1 (you just use a bit less low pass).

I have nothing against people working at higher sample rates; I don't think that miniscule attenuation is worth cutting my resources in half but, hey, your resources are not my resources. If people find the difference that major, whether it be psychological or not, they should have at it. But let's not overstate the case.
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Re: Nebula3 cutting edge? Yes and no!

Postby Gstell » Fri Oct 21, 2011 5:35 am

'm not really sure what you mean by drastic here but in normal comparisons (music as opposed to broadband noise) 96k nulls with 44.1 down below -70dB. I think drastic is overstating it.


Pink noise (MiniMoog analog out,into Mytek192x8, into Wavelab V7.xx).

41khz:
https://rapidshare.com/files/3207252979/pink4402.wav
48khz:
https://rapidshare.com/files/978187989/pink4802.wav
88khz:
https://rapidshare.com/files/1485030372/pink8802.wav
96khz:
https://rapidshare.com/files/1344441614/pink9602.wav
192khz:
https://rapidshare.com/files/3660724063/pinkmoog192.wav
DSD: not available yet!

I must ask you did you listen to the examples I posted? I would rather we compare apples to apples (that is why I opened an account with Rapidshare for nebula users to have access to the same material I have. Which versions of Dark Side of the Moon, Sgt. Pepper's or Thriller do you own? Could you upload it for comparison to my files?
I own various recordings of Dark Side of the Moon even the DSD (which I like the most) version. None of them made it into my selection (missing high frequency content being one of the reasons).

2. It would be useful if we knew your signal chain (equipment you are listening to)?

3. OK, another more accurate statement (although all words are subjective) would be I was shocked to hear the differences between 44.1khz and 96khz, 96khz and 192khz as well as the beauty of Linn sampler disk version 3.
Again my point is that "I" did not now differences existed until I spent hours investigating thousands of recordings.
I started this research for a program I am creating to train my hearing more efficiently. As a result higher resolutions have become a major part of this ear-training course.
I highly recommend Dave Moulton's Golden Ears as a point of reference to learn the value of pink noise in one's ear-training.
To be practical many people can hear the difference between a C and an D. The same is true for a frequency boost at 1K or 250hz.
Pink noise is and has been an important tool to investigate and calibrate audio systems. I found the same to be true for my ears.

4. I understand if one is being cautious about statements people leave in forums, After all you do not know me.
That is why I make material that express my statements available for all to investigate.

6. What nulls with what? examples/illustrations please! Also what does a null test under this scenario prove? I have yet to find some technical explanations that illustrate a persons "Perfect Pitch" (the ability to identify musical notes by ear). In audio there are still some things we can hear that have not been explained in a satisfactory manner.

The low pass filter is the main difference from one sample rate to another (and in many cases from one converter to another). But the attenuation at 44.1 is usually pretty close to 20k, not exactly a frequency area rich with melodic content. It can be easily compensated with a very gentle EQ boost at 20k or, as is probably more common, gets compensated for seamlessly when you are mixing at 44.1 (you just use a bit less low pass).


And yet the bulk of CD's lack this high frequency content! Not to mention the use of crude and misguided pairwise panning still implemented in today's consoles. The state of the art in commercial recording has completely missed-interpreted Blumlein's 1929 patent on audio display as well as missed Michael Gerzon's advances in the 70's. Lets not overstate the sense and quality of commercial recording industry.
Do we again want to record to VHS if we could have Betatape?



I have nothing against people working at higher sample rates; I don't think that miniscule attenuation is worth cutting my resources in half but, hey, your resources are not my resources. If people find the difference that major, whether it be psychological or not, they should have at it. But let's not overstate the case.


I agree that allocating precious resources are a deciding factor in music production. I for one rather have low-pass filter artifacts not invade hours of my work especially in my source material.
I have worked in Computers for years and one common denominator of all PC users is that one's data is infinitely more important that disk-space or processor resources. Just as important as data backup is data integrity. If I have resource issues I rather down-sample my source material for that project than compromise the source material.
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Re: Nebula3 cutting edge? Yes and no!

Postby ngarjuna » Fri Oct 21, 2011 6:28 am

gstell wrote:Pink noise (MiniMoog analog out,into Mytek192x8, into Wavelab V7.xx).

41khz:
https://rapidshare.com/files/3207252979/pink4402.wav
48khz:
https://rapidshare.com/files/978187989/pink4802.wav
88khz:
https://rapidshare.com/files/1485030372/pink8802.wav
96khz:
https://rapidshare.com/files/1344441614/pink9602.wav
192khz:
https://rapidshare.com/files/3660724063/pinkmoog192.wav
DSD: not available yet!

I must ask you did you listen to the examples I posted? I would rather we compare apples to apples (that is why I opened an account with Rapidshare for nebula users to have access to the same material I have. Which versions of Dark Side of the Moon, Sgt. Pepper's or Thriller do you own? Could you upload it for comparison to my files?
I own various recordings of Dark Side of the Moon even the DSD (which I like the most) version. None of them made it into my selection (missing high frequency content being one of the reasons).

2. It would be useful if we knew your signal chain (equipment you are listening to)?

3. OK, another more accurate statement (although all words are subjective) would be I was shocked to hear the differences between 44.1khz and 96khz, 96khz and 192khz as well as the beauty of Linn sampler disk version 3.
Again my point is that "I" did not now differences existed until I spent hours investigating thousands of recordings.
I started this research for a program I am creating to train my hearing more efficiently. As a result higher resolutions have become a major part of this ear-training course.
I highly recommend Dave Moulton's Golden Ears as a point of reference to learn the value of pink noise in one's ear-training.
To be practical many people can hear the difference between a C and an D. The same is true for a frequency boost at 1K or 250hz.
Pink noise is and has been an important tool to investigate and calibrate audio systems. I found the same to be true for my ears.

4. I understand if one is being cautious about statements people leave in forums, After all you do not know me.
That is why I make material that express my statements available for all to investigate.

6. What nulls with what? examples/illustrations please! Also what does a null test under this scenario prove? I have yet to find some technical explanations that illustrate a persons "Perfect Pitch" (the ability to identify musical notes by ear). In audio there are still some things we can hear that have not been explained in a satisfactory manner.

And yet the bulk of CD's lack this high frequency content! Not to mention the use of crude and misguided pairwise panning still implemented in today's consoles. The state of the art in commercial recording has completely missed-interpreted Blumlein's 1929 patent on audio display as well as missed Michael Gerzon's advances in the 70's. Lets not overstate the sense and quality of commercial recording industry.
Do we again want to record to VHS if we could have Betatape?

I agree that allocating precious resources are a deciding factor in music production. I for one rather have low-pass filter artifacts not invade hours of my work especially in my source material.
I have worked in Computers for years and one common denominator of all PC users is that one's data is infinitely more important that disk-space or processor resources. Just as important as data backup is data integrity. If I have resource issues I rather down-sample my source material for that project than compromise the source material.

Honestly, I think you're missing my point: I indeed did not listen to the samples provided; no disrespect intended but I do not believe splitting the hair for "more fidelity" is a worthwhile pursuit at all. I don't think there is any problem with the delivery quality of audio today. Most non-engineers and plenty of engineers agree. And the reason is that nobody even cares about the unbelievably minute differences that only golden eared pros with great listening skills can even identify as different. The reason nobody cares is because they're listening to something else entirely: the material. Rocky McRoll can have the most sonically impressive album ever produced but it still isn't a classic. As I said before, if the current media had an actual problem, it would be worth addressing. But it doesn't. The manufacturers after all these years would love to sell you all new boom boxes and stereo components and car decks; there's almost no interest from the mass market in such a thing. Because there's no money to be made there, everyone is perfectly happy with CDs except cranky audiophiles; and for them, there's a plethora of overpriced equipment to play endless comparison games with. Don't get me wrong, it's a fine hobby if that's what you're into. But that has nothing to do with mainstream commercial production.

And the differences are not that staggering between 96k 44.1k whether on the Events in my private studio or on custom far fields in the bigger rooms I work in; and the shittiest monitors I use are still 1000% better than the average speaker in deployment today anyway. I hear a lot of hype about this magical difference, much more than the difference itself merits. The impact of halving your resources is in no way insignificant, either. It's tolerable, maybe, depending on what kind of system you use and what that actually leaves you with but there should be a damned good reason for a willingness to operate at 50% capacity. I don't hear a good reason.

I'd rather have a HP filter built in for compressor programs (soon to be released, as I understand it, yay!), for example.
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Re: Nebula3 cutting edge? Yes and no!

Postby Gstell » Fri Oct 21, 2011 12:07 pm

We are getting side tracked here. You seemed to have forgotten that this is a nebula thread!

Lets say the difference between 44.1khz 192khz and 384khz is small. So are the effects of many Nebula programs. Consoles come to mind!

A nebula program entails not only the intended devices function and harmonics but also some unwanted artifacts as the artifacts and character of the converters used during the sampling process. For every time a nebula program is applied to an audio mix these unwanted artifacts accumulate in the processed material as well, Yes?
Giancarlo correct me if I am wrong here!
So if one multiplies a potential small problem????

At the very least higher samplerates will reduce impact of the the converters filters inside the Nebula program.
Even a "hobby engineer" would recognize this!


"cheerio old chap"
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Re: Nebula3 cutting edge? Yes and no!

Postby Gstell » Sat Feb 18, 2012 7:47 pm

I just wanted to mention since I had endless encounters with folks citing "Nyquist Frequency Theorem" to disprove or ignore sample-rate quality differences.

-----Beeeeeeep------

The theorem assumes ideal real-world conditions,it only applies to signals that are sampled for infinite time; time-limited conditions cannot be perfectly bandlimited (filtered). Perfect mathematical reconstruction is not possible in the real world but only an approximation!!

From Wikipedia: http://en.wikipedia.org/wiki/Nyquist-Sh ... iderations

"A few consequences can be drawn from the theorem:

If the highest frequency B in the original signal is known, the theorem gives the lower bound on the sampling frequency for which perfect reconstruction can be assured. This lower bound to the sampling frequency, 2B, is called the Nyquist rate.

If instead the sampling frequency is known, the theorem gives us an upper bound for frequency components, B<fs/2, of the signal to allow for perfect reconstruction. This upper bound is the Nyquist frequency, denoted fN.

Both of these cases imply that the signal to be sampled must be bandlimited; that is, any component of this signal which has a frequency above a certain bound should be zero, or at least sufficiently close to zero to allow us to neglect its influence on the resulting reconstruction. In the first case, the condition of bandlimitation of the sampled signal can be accomplished by assuming a model of the signal which can be analysed in terms of the frequency components it contains; for example, sounds that are made by a speaking human normally contain very small frequency components at or above 10 kHz and it is then sufficient to sample such an audio signal with a sampling frequency of at least 20 kHz. For the second case, we have to assure that the sampled signal is bandlimited such that frequency components at or above half of the sampling frequency can be neglected. This is usually accomplished by means of a suitable low-pass filter; for example, if it is desired to sample speech waveforms at 8 kHz, the signals should first be lowpass filtered to below 4 kHz.

In practice, neither of the two statements of the sampling theorem described above can be completely satisfied, and neither can the reconstruction formula be precisely implemented. The reconstruction process that involves scaled and delayed sinc functions can be described as ideal. It cannot be realized in practice since it implies that each sample contributes to the reconstructed signal at almost all time points, requiring summing an infinite number of terms. Instead, some type of approximation of the sinc functions, finite in length, has to be used. The error that corresponds to the sinc-function approximation is referred to as interpolation error. Practical digital-to-analog converters produce neither scaled and delayed sinc functions nor ideal impulses (that if ideally low-pass filtered would yield the original signal), but a sequence of scaled and delayed rectangular pulses. This practical piecewise-constant output can be modeled as a zero-order hold filter driven by the sequence of scaled and delayed dirac impulses referred to in the mathematical basis section below. A shaping filter is sometimes used after the DAC with zero-order hold to make a better overall approximation.

Furthermore, in practice, a signal can never be perfectly bandlimited, since ideal "brick-wall" filters cannot be realized. All practical filters can only attenuate frequencies outside a certain range, not remove them entirely. In addition to this, a "time-limited" signal can never be bandlimited. This means that even if an ideal reconstruction could be made, the reconstructed signal would not be exactly the original signal. The error that corresponds to the failure of bandlimitation is referred to as aliasing.

The sampling theorem does not say what happens when the conditions and procedures are not exactly met, but its proof suggests an analytical framework in which the non-ideality can be studied. A designer of a system that deals with sampling and reconstruction processes needs a thorough understanding of the signal to be sampled, in particular its frequency content, the sampling frequency, how the signal is reconstructed in terms of interpolation, and the requirement for the total reconstruction error, including aliasing, sampling, interpolation and other errors. These properties and parameters may need to be carefully tuned in order to obtain a useful system."

------Beeeeep--------
Last edited by Gstell on Sun Feb 19, 2012 11:03 pm, edited 1 time in total.
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Re: Nebula3 cutting edge? Yes and no!

Postby mathias » Sun Feb 19, 2012 12:38 am

yes, you are right gstell.
our world is limited and unperfect.
BUT not only converters and filter-implementations, also our senses vary from day to day, people to people,...
maybe it is not that bad, as it is ... (already mentioned)

everybody can understand your passion for going further in hearing and trying to bring things forward.

if you could temper your tone, i would gladly listen longer to this thread and maybe add my opinion, but i don't like rants and teacher-like mentions in this kind of discussion. you yourself mentioned "science", whatever that means for you.

at the moment i feel like you want to press your issues forward (maybe i am wrong).

peace,
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system 1: windows 8 32 bit - samplitude prox, tracktion6, reaper
system 2: mac osx yosemite - reaper(32+64bit), tracktion6(32+64bit)

both systems on: macbook pro (late 2009), core 2 duo 3,06 ghz, 4 gb ram, graphic: nvidia geforce 9600M GT 512 MB
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