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How many bits, how fast, just how much resolution is enough?


BlueSkyy

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44.1/16, maybe 44.1/24 or 48/24. In extreme cases with exemplary gear and young terrific hearing 48/24 might be needed.

 

Some of the more knowledgeable people for the most extreme possibilities estimate you might need 65 khz/20 bit. As that isn't an available rate, 96/24 or 88/24 does it for you with room to spare. If one just wishes to absolutely end any possibility of any person in the world hearing something in less than max fidelity due to recording limitations then 96/24 is enough and that.....is.......simply......that.

 

Long winded version of jjjjusta's reply.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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Absolutelly. Base of audio design is 16-20 kHz.

 

16/44.1 was taken as minimal available that time (invention of 16/44).

 

All further increasing of sample rates was based on pragmatical technical reasons (real analog filtration abilities in DAC) on newer technologies.

 

Inceasing of bit depth give expanding of signal/noise ratio in listened dynamic range.

 

DSD's sample rate is matter of signal/noise ratio in audible frequency range in complex with analog filters of DAC.

 

Some revisionist history here I think. Before the CD became a standard various digital recording formats were around being experimented with for music. They ranged from 13 bits to 16 bits. Sample rates ran from 32 khz to 50 khz. Some used various pre and deemphasis schemes. When Philips and then Sony decided to back a new standard Philips had decided 14 bits was enough while Sony insisted it be 16 bits. Both agreed sample rate needed to exceed 40 khz by some small amount. As is commonly known we got 44.1 khz because it fit the Sony pro video recorders that were adapted to hold digital data. More than that would not fit the data into video frame efficiently. 48 khz was later adapted for video use due to how it fit with scan and frame rates of the existing broadcast video at the time. For one period of time pro video standards considered using 60 khz because it fit all the various frame rates nicely regardless of country. It was deemed wasteful and unnecessary.

 

So no, 44.1 wasn't chosen as some minimum available at the time. Basically anything above 40 khz would work and the other factors caused it to be 44.1 for CD.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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In the post you quoted the image on the right has the same sample rate but more "amplitude" values.

The increased resolution is obvious.

 

R

 

Usually when someone is explaining digital to you and use the staircase waveform, STOP. Stop listening to anything else they have to say.

 

The higher sample rate allows more bandwidth. You can as the video demonstrates with purely analog gear at each end fully reconstruct a waveform up to the limited bandwidth with only two samples per wave. The "extra" samples of a higher sample rate aren't constructing a more resolved or more accurate waveform. Or looked at another way, a high sample rate 1 khz tone is not more accurately reconstructed than a redbook 1 khz tone even though it has more sample points per wave. You couldn't tell the difference between the two. The staircase idea would mislead you into thinking the high rate 1 khz tone is more accurately reconstructed.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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#2 filtering may be easier/less intrusive as I alluded to above - this is mentioned in that article by "monty" tho I downrate things without a Methods section, or by some dude on the internet - to be taken seriously, your name and affiliation will appear at the top of your peer-reviewed article

 

#3 - um... ah... I thought I had a 3rd one last night...

 

I don't think his video was meant as a peer review thing. If you look at the website it is hosted from then you know his affiliation. Among other things he is partly responsible for the Ogg Vorbis format. His methods were clear to me in the video. There is or was a comments area on the host site. If you have a question not answered you probably could ask. He shows a good quality analog source feeding a good quality analog analyzer. Then puts a low budget AD/DA in the middle. Is there something about this you question or doubt or wonder about?

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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So, 16 bits are enough to accurately capture the wave, but often not enough to capture the whole wave?

 

Not sure I understand your question. Could you elaborate? Bit depth and sample rate are separate things. The bit depth won't alter how often samples are taken.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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I didn't use a staircase waveform, there's no waveform in this example only, as far as I know, quantisation error:

 

 

 

And I don't know why people started talking about sample rate when I was referring to bit depth...

 

R

 

 

Well in my case because I took my own advice, and stopped paying attention when I saw the stair case looking graph. Also because you mentioned the more points you have taken the more accurate. That describes a higher vs lower sample rate.

 

Bit depth relates to quantisation error which creates quantisation distortion. Using 24 bits lowers the error and distortion vs 16 bits. Good dithering also mitigates quantisation error to the point with 16 bit it lies below the noise floor of most analog electronics. You can look at tests by Stereophile of DACs. They will show the spectrum for 16 bit signals and 24 bit signals. If the analog part of the gear is good switching to 24 bit lowers the noise floor. Maybe 20 db in the better gear. Maybe not much at all in some gear. So yes 24 bit depth can be more accurate than 16 bit. The difference is likely too small to notice, but it is there. Not to mention finding microphones and recording venues with noise floors lower than dithered 16 bit levels.

 

MSB Technology Analog DAC D/A converter and Analog Power Base power supply Measurements | Stereophile.com

 

Here you can see how the switch between 16 and 24 bit results in a noise floor drop of 28 db in this excellent MSB DAC. Look at figure 5. Even at dithered 16 bit nothing sticks up above -130 db besides the 1 khz tone.

 

Here look at figure 4 for a much less expensive and similarly good DAC. The noise floor drops about 30 db with 24 bit.

Benchmark DAC2 HGC D/A processor/headphone amplifier Measurements | Stereophile.com

 

OOPS, mansr answered in between and gave a much shorter answer.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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Okay, how about at 90 db?

 

http://www.computeraudiophile.com/f8-general-forum/how-audible-noise-your-system-30596/

 

Try these signals with embedded noise and see where it disappears for you.

 

I agree in either case you aren't going to hear that conversation. Also, dithered 16 bit would capture much of the conversation even at -101 db. Not as cleanly as 24 bit if the other electronics are very, very quiet.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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https://dl.orangedox.com/uY0pgsmWy5cGHg8L9g

 

Here you can download a pair of 30 second clips of the John Mayall clip. They are in a single zip file of 2.8 mb.

 

I reduced it to peaks of -101 db. RMS level is -115 db. One version was saved as 24 bit and another as 16 bit. Both dithered. I then applied 70 db of digital gain. So you can hear what is left. The 24 bit is pretty clean, the 16 bit is noisy, yet you can hear the music is still there encoded in the file. Also you could never do this in the analog world as cleanly as the 24 bit because analog electronics have far more noise. Still it will give you the idea how low a signal is still encoded in dithered 16 bit.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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This was an interesting listen. The music is indeed still there, under all the noise.

 

24 bit makes things a lot easier though, no?

 

24 bit does, but had I run this through a DAC and its analog output, or via a power amp before applying that digital gain, it would pick up enough noise the result is virtually identical to the dithered 16 bit. You might gain a little less noise if your gear is really quiet.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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That's what I meant when I said it's "cheating". I.E. it doesn't solve the problem, it merely masks it, like a bandaid on a cut finger.

 

It's interesting how dithering was discovered. Digital mastering engineers kept wondering, in the early days of CD, why CDs mastered from analog tape sounded better than the latest DDD recordings, since both went through the same digital equipment. Turned out that the tape hiss from the analog recoding was acting like an inadvertent dithering signal. The engineers, once having discovered this, experimented with reducing the noise level to find the right balance between adding enough noise to act as a dither, but not so much as to be blatantly audible. Once they found that happy medium, the rest was recording history. It is interesting to note that as long as recording engineers use enough record level to keep at least 4-bits below the level of the quietest passages in one's recording, and 4-bits above the loudest passage (in other words use 16-bits within a 24-bit "envelope") 24 and 32-bit recording don't need dither, until down-converting the file to Red Book for CD mastering.

 

History of Dither

 

The concept of dithering to reduce quantization patterns was first applied by Lawrence G. Roberts in his 1961 MIT master's thesis and 1962 article though he did not use the term dither. By 1964, dither was being used in the modern sense. To ameliorate the negative effects of quantization, early workers in the field simply combined white noise to the original ADC input signal. Dither is often used in digital audio and video processing, where it is applied to sample-rate conversions and to bit-depth transitions; it is utilized in many different fields where digital processing and analysis are used — especially waveform analysis. These uses include systems using digital signal processing such as digital audio, video, photography, seismology, radar weather forecasting systems and more. Considering the premise is that quantization and re-quantization of digital data yields error, if that error is repeating and correlated to the signal, the error that results is repeating, cyclical, and mathematically determinable.

 

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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But your system does. From a review of the Vega:

 

 

 

The Auralic uses an ESS SABRE DAC chip to do this, foregoing ESS's internal upsampling to ~44.1MHz.

 

That's why these discussions always strike me as such a waste of time. Just about no one in these forums is actually listening to analog that's been reconstructed straight from RedBook. The vast majority are listening to analog reconstructed from a bitstream running at somewhere between 2.8MHz and 11.2MHz. A substantial minority of folks whose DACs use the ESS chip's internal upsampling are listening to analog reconstructed from a bitstream running at about 44.1MHz.

 

The engineers who built your DACs decided long ago (about 25 years - that's when delta-sigma DACs began to take over the market) that MHz sample rates were the least expensive most practical solution. If you want to talk about theory rather than reality, that was also resolved very long ago (implied by Nyquist's work in 1928, with Shannon's mathematical proof published in 1948). A sample rate just over twice the highest "frequency of interest" is all you need to satisfy the conditions of the mathematical proof.

 

And those sigma-delta DACs give us the closest approximation to theoretical performance of pure ladder DACs for PCM.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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snip.....

 

Then the first pro/audiophile asynchronous sample rate conversion (ASRC) DACs hit the market, most notably from Benchmark and Lavry. These DACs resampled to non-even multiples of the input rate (in Benchmark's case, 110 KHz; I don't know what Lavry used, though I recall the input rate was limited to 96KHz) as a means of jitter reduction, prior to putting the resulting bitstream through sigma-delta modulation to MHz sample rates. I recall Lavry in particular publishing white papers saying one certainly didn't need higher sample rates than 96KHz and suggesting higher rates were actually deleterious, while not mentioning their own DACs internally used the same MHz sample rates as everyone else.

 

But I'm sure these papers, which did as much as anything to spur discussion of how 44.1 or surely 96KHz rates were perfectly adequate (when even inexpensive non-audiophile commodity players used chips designed by non-audiophile engineers that upsampled internally to MHz rates) had nothing to do with Lavry's commercial objectives. :)

 

If you read Lavry's papers he isn't hiding anything. He talks about how his own converters are delta sigma and run in mhz range. Even then what he talks about as getting less accurate sampling at too high a speed and fewer effective bits is still in effect. He warns the reader to keep both sampling rates clearly separate in their minds when trying to understand how this works. Doesn't sound like someone pulling a fast one if you read it rather than what others say about the paper.

 

http://lavryengineering.com/pdfs/lavry-sampling-theory.pdf

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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The problem may be the relative timing of tones. You can distinguish e.g. multiple HF tones based on their relative timing or relative phase. The smallest audible timing differences are measured in microseconds and simply cannot be encoded using a 44.1KHz sample rate. Some analog recordings might do better..

 

Please..not this bull carp again. Redbook can encode timing to within about 55 picoseconds accuracy. It is not limited by the time between samples.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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Perhaps it's just the way Lavry's writing strikes me, but on re-reading I once again feel his arguments are far more slippery than clear.

 

He implies the decimation from the original SDM processed bitstream at MHz rates to RedBook at 16 bit 44.1KHz is needed for "accuracy," when in fact it is marketing-driven by the need to convert to CDs for sale. In doing this he completely bypasses the issue of whether the distinction he repeatedly makes between "audio sample rates" (the rate of the audio data) and "other sample rates" (such as the rates used by DACs internally) needs to exist at all.

 

The paper states (at page 23 of 27) "In the case of DA converters, the data is interpolated to higher rates which help filtering and response. Such oversampling and up sampling are local processes and tradeoff aimed at optimizing the conversion hardware." [Emphasis added.]

If the original "audio sample rate" doesn't go through the decimation step but stays at the original higher rate, then one gets to "help filtering and response" and "optimiz[e] the conversion hardware" without requiring the "local processes" of oversampling/upsampling. And if these higher rates do indeed "help filtering and response," then where exactly is the tradeoff between speed and accuracy he's trying to sell?

 

Well this was written in 2004 by someone who mades ADCs and DACs for pro audio use. At that time CD was the only pertinent market to orient toward. SACD was not big enough to worry about. Downloads of hirez material weren't available, and the market for such recordings on DVD based discs never took off either.

 

Even if now there is no limitation on the accuracy at the high rates it is Dan Lavry's opinion anything over 96 khz is unnecessary. At one time 96 khz was less capable even over sigma delta chips. I have a Focusrite recording interface which uses a balanced DAC chip able to work up to 192 khz. The device is only setup to work to 96 khz rates. Focusrite said at the time this was because the DACs didn't work as well at 192 khz and measured results suffered. They recently released mark II version that do work at 192 khz. Same DACs. Same specs. No specs specific to 192 khz other than latency. So it is possible nothing was changed other than turning on 192 khz operation. And 192 khz may still not work as well. I don't know for certain of course.

 

The idea to keep pushing sample rates up both in the operation of delta-sigma designs and in the PCM or DSD available rates is an easy selling point. Just like megapixels for cameras. I would be more convinced if someone had rational explanations for what point it sounds so good no improvement can be detected with higher sample rates. So far it is just higher is better as if the improvement can be extended forever with more more more.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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Try it..

 

Think about those numbers you picked for a moment please. 10,000 hz tones offset from each other by 100 microseconds.

 

It makes no difference really, but are you talking one tone in each channel or both in the same channel?

 

Hint:___this isn't going to work out the way you think it will.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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this timing argument is interfering with the other arguments here

 

you guys have 55 picoseconds to wrap it up or start your own delay phase thread

 

Come on now. Resolution in time is what higher sample rates give along with wider bandwidth. I don't think it is important, but that is the difference.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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I have posted quite a number of measurement results to show what are the differences... What else is needed?

I seem to have missed where you have the specs for a sample rate filter combination so good no further audible improvement is possible by increasing sample rate or improving the filter. One might make a better measured result beyond this, but hear not the improvement. Care to tell us what it is?

 

That way there is a target instead of this mindless idea more is better ad infinitum.

 

Sent from my Nexus 6P using Computer Audiophile mobile app

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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If you think you don't hear the measurable improvements, including improvements in the audio band, it's your opinion. However, you cannot claim that nobody else in this world is able to hear it.

 

 

 

As long as measurable (and/or audible) signal fidelity keeps improving by improving implementation, I will keep doing it ad infinitum. I have no reason not to. So far I've been also hearing improvements, but not going to claim anything about anybody else's hearing or non hearing. That's something they need to decide on their own.

 

If improvements become unmeasurable, there just needs to be better measurements. Luckily measurement equipment also keeps improving at steady pace. ;)

 

My target is beyond what is possible, so the target will be never reached. That's how I like it.

 

So you don't have an answer to my question. ad infinitum is your approach.

 

Thank you for your response.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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I'm not sure this from Miska -

 

[emphasis added]

 

- really deserved the rather negative responses to it. Seems rather reasonable to me.

 

Well, I wrote that I would like for someone to specify for rational reasons a target beyond which no improvement is possible. Miska responded. Then instead of having a target, he informs us his approach to make it better no matter what without end. That is perfectly okay. Such people generate some of the wonderful things in the world. It instead indicates he doesn't have an answer to my question however.

 

I have used his software on a trial basis and can confirm it gives better measured results than other software for playback. I couldn't hear an improvement. As he says that doesn't mean no one will. Nevertheless, the idea there is no end to improvements in that area which will be audible does not make much sense to me. It markets really well.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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Again.......I've figured after a few years of this some of you would have stopped trying to separate time from frequency in an effort to incorrectly attribute suspected audible differences to this mystical place. Can't have one without the other fellas! If timing changes, so does the response at the listening position.

 

Wow, you were very mistaken. ;)

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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But as you point out in a part of your post I snipped, he said the same in 2012, and as far as I know continues to maintain the same position. See, e.g., http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf

 

 

Yes, we will always have specsmanship in the marketing of these products. On the other hand, I don't see *current* support for Lavry's position that anything over a 96KHz sample rate leads to worse results.

 

I don't know if that would still be the case with the better devices available. I would think he reached the opinion (which I happen to share) that you get no benefit for more than 96 khz. 96 khz is enough to push every reasonable complaint about filtering and everything else completely out of consideration. And while many would say why not do 192/384 etc, in the pro recording area which is his business, as cheap as storage is those higher rates for recording, mixing and mastering become very burdensome. Not just in cost of using up hard drives, but in terms of how long processing such streams take.

 

BTW, most 96 khz and above devices don't use the opportunity to use a less steep filter. They just extend the bandwidth and keep steep filters. Lavry does use a gentler filter starting to roll off above 30 khz when doing 96 khz sampling in his gear. Maybe filtering that starts at 30 khz on DXD would be great. I take it Lavry thinks you gain nothing.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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snip..

 

You know, your filters have a very good time-frequency response if they are at most 1st order. But you just need to use very high sampling rate in such case...

snip

 

Say a sampling rate around 20 ghz.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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Because this is argument that is never going to end, just like any argument that depends on human senses. Be it whiskey, wines, music whatever.

 

One colleague at Nokia (when I was still working there) once claimed to me that digital filter rejection, or audio electronics distortion doesn't have to be less than -40 dB because nobody is able to hear anything beyond that. And lot of people have claimed that nobody is able to hear difference between 128 kbps MP3 and lossless. Local technical magazine has stopped reviewing audio equipment since 80's when CD was introduced because they claimed that every CD player has sounded the same since and nobody is able to hear any differences.

 

It is not difficult to demonstrate these are not true.

 

Nothing in world is going to change opinion of these people and trying to argue with them is just like trying to argue which one is better, Shiraz or Cabernet Sauvignon. Or whether Islay or Speyside is better. It is just futile and waste of time.

 

Well preferences are another matter. Mixing the old apples and oranges there.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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Certainly it is not. And for 96 kHz PCM you end up doing over 10x decimation factors from the actual AD conversion stage. And again over 10x interpolation factors for the DA conversion stage. Just wasted effort.

 

 

 

Processing 384k, even tens of channels in realtime is completely non-issue these days. Really.

 

 

 

Going down >100 dB in 30 kHz to 48 kHz band is practically brickwall. It is more than 100 dB/oct, that is not "gentle".

 

 

Gentler doesn't mean gentle. Yes it is practically a brick wall, but most DACs don't take advantage of that.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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Just to clarify - is this about HQplayer? And if so, what CPU does your computer use?

 

 

Yes HQplayer. Excellent playback software. Probably the best I know of. I don't use it due to other considerations, but it is excellent.

 

I did the trial on a machine with ample RAM and an I5 processor. A Lenovo home server actually.

And always keep in mind: Cognitive biases, like seeing optical illusions are a sign of a normally functioning brain. We all have them, it’s nothing to be ashamed about, but it is something that affects our objective evaluation of reality. 

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