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24/192 Downloads ... and why they make no sense?


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"A 10us transient signal completely between the samples will not be recorded. While looking for a nice graph to show this, I stumbled across this paper by Tim Westcott. Amazing, I don't seem to disagree with anything in this paper. It is, indeed, the same thing I have been saying."

 

I thought we agreed to drop this debate...

 

 

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Maybe you could help me by explaining the connection?

 

Nope- as I said, I just thought they might be interesting. Oh that, and what it is doing is separating out transient information from steady state information.

 

 

Paul

 

 

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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How many people seem to think that engineering and art are somehow separated, and performed by two groups of mentally different people. You see it sometimes here.

 

The origin of the word 'art' covers both bases. Look it up. And is often forgotten that the supposedly narrow minded 'engineers' do in fact go to concerts, art galleries, read books other than engineering manuals, and so on. But how many of the 'arty' people ever read a technical book? They are the ones who's knowledge is only half complete and narrow.

 

Two examples of people with a broad knowledge:

 

Leonardo da Vinci.

 

GT in OZ, on the 'What's your occupation?' thread earlier today. Have a look.

 

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"Look it up. And is often forgotten that the supposedly narrow minded 'engineers' do in fact go to concerts, art galleries, read books other than engineering manuals, and so on. But how many of the 'arty' people ever read a technical book? They are the ones who's knowledge is only half complete and narrow."

 

A pretty narrow minded paradigm if you ask me... I know many technically adept artists.

 

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I'm sure there are plenty of examples on all sides of that.

 

I know one person who plays in a symphony orchestra for a living, and spends his evenings studying plate tectonics. (I think I spelled that right.) He's also writing a composition based upon how Hawaii was formed. You should hear how he renders the fury of a "hot spot."

 

-Paul

 

 

 

 

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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Do you think as Leonardo got older, he added more resolution/paint strokes to his paintings, to make them more realistic?:)

 

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"...Do you think as Leonardo got older, he added more resolution/paint strokes to his paintings, to make them more realistic?:)..."

 

You don't ned more paint or strokes to add more resolution in a good paintings as Leonardo's ones are. More creativity, yes, and even, sometimes with less paint (or strokes).

 

If you study on depth Leonardo's paintings, you can see, maybe, the after meaning of his paints. But I guess than only Leonardo knows exactly those meanings...

 

But I can tell you another history, and is my humble opinion on hi-rez digital audio: Since I get less noise, I can see the 'musical paint' more clearly, with lees paint (or strokes) added from the noise.

 

Mark,

 

The most evident divorce between art and engineering that comes to my mind right now are architects and engineers. For example, in my country, where it rains a lot, the engineers wants a good roof with no leaks (me also), but the architect wants it to be beauty and doesn't matter about the leaks.

 

That's why I'm changing my home roof (after get fired the architect).

 

Roch

 

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Gentlemen, let me remind you about an important matter: The recording and playback chain is full of errors, small and big. Every single component introduces distortion along the signal chain, starting with the microphone and ending with your loudspeakers/room acoustics.

 

Active electronics distort more than passive ones (cables). And transducers (mics/loudspeakers) distort more than electronics.

 

But, and this is my point, the errors adds up. It accumulates through the chain. Even if your loudspeakers have 2% distortion, you are still able to distinguish between 2 different op-amps inside your DAC with distortion figures as low as 0,001%.

 

Same goes with risetime and bandwidth. Although your loudspeakers are heavily limited in high frequency extension, you still can distinguish between CD and hi-rez.

 

Because of the accumulative effect, it is always worth optimising the individual links in the chain of a hi-fi system.

 

 

 

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"same goes with risetime and bandwidth. Although your loudspeakers are heavily limited in high frequency extension, "I" still can distinguish between CD and hi-rez."

 

Fixed.

 

 

 

 

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Does anyone think Leonardo would prefer a 500kb jpeg of one of his paintings to that painting? That's the proper comparison, so you can say I fixed this analogy for you.

 

The proper analogy to music is 1411kbps to captured soundwave. I don't think more brush strokes, here, can really get us to that original, but I certainly prefer a 2mb jpeg to 500kb.

 

I can hear the difference between low and high res (same recording, both downsampled if needed) in two seconds. I personally know what direction that would tend me as regards theoretical investigations and speculation.

 

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I'm sorry Leonardo, but I didn't get you in this forum.

 

I guess you, Leonardo, are not a digital artist (but I'm not absolutely right about this), but somebody correct me, just in case you are (now).

 

Thanks Thomas!

 

But, my analogy was on the 'reverse thinking' I'm always on, not your fault, Thomas.

 

In the way I can see low-rez is, spaces filled with noise (listenable noise). The same case with 'hi-rez' upsampled digital tracks, from low-rez originals.

 

Please don't ask me for more explanations, my English is very limited.

 

Thanks,

 

Roch

 

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"There's Life Above 20 Kilohertz!

A Survey of Musical Instrument Spectra to 102.4 KHz" http://www.cco.caltech.edu/~boyk/spectra/spectra.htm

 

Keys jangling

71 SPL (dB)

10 dB above background to 60kHz

68% of Power Above 20 kHz

 

PeterSt said "This is nothing about the "fineness" or refinement or "nice silk" sound, most people tend to think about, when talking HiRes. It is almost the other way around."

 

The smoothness I hear in high resolution formats I attribute to sounding less digital. I think it is a combination of bit rate and sampling frequency, since BIS's 24/44.1kHz downloads sound considerably smoother than any 16/44.1kHz, however 24/96kHz usually sounds smoother still.

 

I find it amazing that music can be reduced to only two parameters in digital, sampling frequency and bit rate. To me that seems like magic and something I would never believe possible if I didn't hear it with my own ears. Music is much more than frequency response and dynamic range so the other two dozen parameters of music reproduction are contained in the bit rate and sampling frequency including smoothness or stridency of the recovered waveform once converted back to analog.

 

The 24 bit rate affords a longer word length and with 256 times the resolution of 16 bit this coupled with a higher sampling frequency that moves the actual sampling of the waveform further from the audible frequencies is what I feel accounts for the smoother sound of high resolution digital. In addition that longer word length offers increased "micro" dynamics as well as increased "macro" dynamics otherwise know as dynamic range. Micro dynamics offer increased intricate sonic details which I believe is the "fineness" you are referring to. I really don't worry about WHY, I just enjoy music in high resolution and don't enjoy music on CDs or in 16 bit PCM.

 

Also I agree with those who say that short transient events will not be recorded at 44.1kHz as it is not fast enough and very short events can occur in between samples. Transient response I feel is a bigger advantage to higher sampling rates than ultrasonic frequency extension. However ultrasonic overtones of real instruments are said affect their tonal qualities and realism and that is why live acoustic music sounds and feels so real, as those "natural" ultrasonics are not removed.

 

 

I have dementia. I save all my posts in a text file I call Forums.  I do a search in that file to find out what I said or did in the past.

 

I still love music.

 

Teresa

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- Though distortion, or errors, or loss of information, or whatever you'd like to call it, is generally cumulative, I can think of particular situations where it might not be. For example, suppose the rise time of an inharmonic attack is faster than a particular speaker cone can move. If a lower sample rate "spreads out" that rise time slightly, but it is still faster than the speaker cone can move, one might not hear a difference in the attack vs a higher sample rate.

 

- A complete transient event that occurs so fast it's missed at a 44.1kHz sample rate seems to me uncharacteristic of music. As PeterSt says, (inharmonic) attacks characteristic of music have a tremendously fast rise time, but the decay of the sound is not nearly as fast. Think rimshot, cymbals, pluck of string followed by guitar note, etc. Many of us have heard digital "dropouts" from material sampled at 44.1kHz, and they are so lightning fast that it takes an accumulation of them to sound as "slow" as the fastest tick you can imagine. It's not that puny little tick we're missing from instrumental attacks, it's an extremely fast rise to high amplitude followed by a much slower decay. A lower sample rate may cause the rise time to be longer, but it won't altogether miss the amplitude (it may miss the highest peak, but it will get close) as some may be thinking.

 

At least, the foregoing is as it seems to me things might work (rank amateur that I am).

 

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

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I am getting the idea from reading here, the less expensive the system, and the lower the quality of the recordings, the more hi-res is important. I guess with mid-fi, one needs to max. out those dollars not in better equipment, but more bits and resolution. I am all for high-res, but please, on $200 DACS and $500 speakers? Now, there is nothing wrong with mid-fi, but I see where hi-res tries to make it up, for some.

 

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I really don't know. I'm on ELS speakers, and since the mass of the membrane is about 1 gram, this speakers are very, very fast on rise time, and on decay.

 

Then is very easy to notice the improvement over 16/44, on 24/96, of course 24/192, but mostly on pure DSD. As I have posted here a lot of times, music come easy to my ears/brain, effortless, like on life unamplified concerts.

 

Thanks to my English, I'm very bad making analogies, but maybe I can compare hi-rez to more torque in a car engine?

 

Again, I don't want to completely blame the redbook CD, I have a lot that love from outstanding recordings, bettered (from a couple of years ago listening sessions) thanks to Audirvana Plus and Playback Designs.

 

Talk2Me,

 

I am getting the idea from reading here, the less expensive the system, and the lower the quality of the recordings, the more hi-res is important.

 

I could think the contrary, since you get the 'lower quality recordings' amplified, you could listen with more detail the 'bad'. I wouldn't buy a hi-rez of a bad recording. Why to pay more?

 

Roch

 

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"Just listened again, yep, 16 bit sux. "

I'm in total agreement.

 

"I don't know how anyone can say 16 bit is hi-res in any sense of the word. Except maybe by comparison with MP3 - maybe that's the issue here. Even though I'm now getting the very best reproduction of ripped CD now I've ever heard, including that of a 2K player (thanks to the new V-Link 192 doing the bit arranging for a high end DAC), it is still easy to hear the problems of ALL 16 bit material.

To my ears 24 bit has a wider, deeper soundstage, more ambiance and ease of presentation. If I put on a lossless 16 bit even from an audiophile label right after playing a 24 bit recording, I notice the music close-in, become congested with an added touch of stridency. Also 24 bit has more realistic bass and a warmer midrange. I just don't see a reason for 16 bit PCM to exist anymore.

 

"as Teresa has mentioned so often. Just listen..."

Personally I am not interested in technical reasons why something sounds better or technical mumbo-jumbo used in an attempt to prove that something that is worse sounds as good as something that is better. I just use only one tool, my ears. That's it. And that is what I recommend to every single person, trust your ears. Oh, and make sure to practice good ear heath, and keep excess earwax cleaned out.

 

I have dementia. I save all my posts in a text file I call Forums.  I do a search in that file to find out what I said or did in the past.

 

I still love music.

 

Teresa

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""Personally I am not interested in technical reasons why something sounds better or technical mumbo-jumbo used in an attempt to prove that something that is worse sounds as good as something that is better. I just use only one tool, my ears. That's it. And that is what I recommend to every single person, trust your ears. Oh, and make sure to practice good ear heath, and keep excess earwax cleaned out.""

 

Teresa well said/written

 

The Truth Is Out There

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"I just use only one tool, my ears. That's it. And that is what I recommend to every single person, trust your ears."

 

I wish I could use only my ears, but we must also choose a DAC, and a computer and a software player for computer audio playback.

 

 

"...since BIS's 24/44.1kHz downloads sound considerably smoother than any 16/44.1kHz..."

 

What system do you use to hear a difference between 24/44.1 and 16/44.1 recordings?

 

 

 

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How are inharmonics approximated (or are you saying they are not)? With harmonics, or in some other way? Please realize I'm asking from a position of great ignorance and would sincerely like to know.

 

Let me try to explain.

 

First: forget about harmonics. Or inharmonics. These do not apply to music.

 

Ooph, now I have to explain that.

 

If you describe a signal as a fundamental and a number of harmonics, you are in fact discussing the signal's Fourier series (series, not transform).

 

A Fourier series only exists for signals that are periodic and steady state.

 

Music is neither. Music is infinitely more complex and interesting than that. Harmonic series do not apply to music signals in general.

 

Then why do we keep rambling on about harmonics?

 

Because the concept of fundamental and overtones is roughly valid during the sustained part of a note played or sung. The note has a fundamental pitch, and during the sustain it has a harmonic composition that remains more or less stable in time.

And these determine mostly the timbre of what we hear, including the schoolbook stalwart of why a piano sounds different from a violin.

So there is some merit in harmonics. Or inharmonics, for the matter.

 

But these do not describe the whole musical signal. Not by far.

 

 

Thankfully there is another mathematical tool for describing and discussing signals like music: the Fourier transform. The Fourier transform applies to signals that do not do acrobatics like moving from absolute zero to max in zero time. So there may be some astronomical events that are not covered by the transform, but music surely is.

 

The Fourier transform maps a progression of magnitude versus time onto a progression of (complex/imaginary - i.e. amplitude and phase) magnitude versus frequency. Or time domain to frequency domain. This map is 1-to-1 and complete, i.e. nothing is lost.

 

The frequency domain is a decomposition into sines, much like the Fourier series, but the salient point is that it is not a sum of discrete frequencies but rather an integral over a continuum of frequencies. In other words we describe any time domain signal now as an infinite amount of frequencies, and this even within a limited bandwidth. It is a very far cry from fundamentals and harmonics.

 

In the sampling theorem we use the Fourier transform to discuss properties of the original signal. We do not use the transform (i.e. the decomposition) of the signal in the act of sampling itself, so there really are no machines trying to approximate the transform. There is no need for that.

 

Are you following so far?

 

 

 

 

 

 

 

 

 

 

 

 

But what I don't see is what you try to prove with the spacing of your pulses

 

That temporal features of 1us make it past 44.1kHz unscathed. Or, IOW, that the temporal granularity is much smaller than 1/44100 second.

 

But now run the same on top of two different frequencies. 5Khz and 12 Khz will be fine. Through that, run the same pulses with same time shift.

I think it will be easy to find spots where the same shift is not there.

 

I can't comprehend what 'running the same on the top of two frequencies' could mean, but my all means do this test yourself and provide evidence of the problems you think you will find.

 

Run the test with the pulses so close that they run into eachother's sweep-up and -down. See what's left of the pulses and the level of the peaks.

 

The pulses will merge. They have to, due to the severe bandlimiting to 22kHz. Assuming that the auditory system is limited to some 20kHz as well this is not a problem. (Disagreeing with the latter is fine, but not the subject of this discussion.)

 

 

 

I find it slightly amusing that we talk about sampling, and everyone jumps immediately to filters and reconstruction of the signal.

 

Worrying that you find this amusing. The filters, i.e. anti-aliasing and anti-imaging aka reconstruction, are mandatory.

 

Oh, many textbooks ignore this. And ADC and DAC components targetted at metrology lack these filters. Hence the rules of thumb for sampling N x higher than the highest frequency of interest (including the article you linked to). But all of these actually violate the sampling theorem. Whereas in audio we don't violate it (*). We do have proper AI filtering, and we do have proper reconstruction (bar the NOS brigade, who relegate the reconstruction to the tweeters and ultimately the ear, which is fine - sort-of).

 

 

(* At least: not that blatantly.)

 

The actual samples themselves contain no information about what happens between the samples. A 10us transient signal completely between the samples will not be recorded.

 

Put your transient through the - mandatory - AI filter. See how the output gets sampled.

 

 

I find it amazing that music can be reduced to only two parameters in digital, sampling frequency and bit rate.

 

Three: you forgot linearity, but that is often taken for granted.

 

But it is not the music that is reduced to these parameters, but rather the channel through which the music flows. Bandwidth, signal to noise ratio, linearity.

 

A fust holds a fine vintage wine. Relevant parameters are volume and type and ago of wood (I think). Are we reducing the wine here?

(Chemists, please no witty jokes here!)

 

 

 

 

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Digital depicts inifitely short time. Say 0.00000000µs.

 

Oh, it definitely doesn't, it represents a smooth transition from value to another of period 1/fs so that it doesn't exceed fs/2 bandwidth limit.

 

The transients how I define them, sure do exist. Look in the files !!

 

Please first declare your transient as a function so that we have some substance to discuss about. Now I'm not sure what you think being a transient vs the signal seen by ADC.

 

To correctly look at the files in waveform, you have to sinc-interpolate the values to pixels. Drawing straight lines between samples is just plain wrong and you'll get to completely wrong conclusions.

 

 

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A lower sample rate may cause the rise time to be longer

 

Ehm, *shorter/faster* !

(without me saying that this is more real or something)

 

Funny eh ?

 

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