96Khz vs 192Khz

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I'll take a shot.

The "load" would be on DACs sample rate filter eq at SD sample rates. ie. The fact that said eq is in use at all.

In SD sample rates (especially 44.1k) the sampling frequency is RIGHT next to the audio band. So close that the sampling frequency needs to be eq filtered out with a steep lo pass or it will roll into the audio band. This is in the analog domain after the AD chip. This kind of steep eq is a hard circuit to build! It turns out that the factor you are comparing between different DACs at SD sampling rates is that analog eq circuit.

Meanwhile HD sample rates put the sampling frequency miles away from the audio band. The margin is so wide that eq filtering is not needed. Like tape bias whistle.

If your DAC sounds better at 96k because of this, upsampling SD program is an excellent workaround. The music signal is in there fully. The analog filter eq is your problem. We remove it from the equation. And yes, it's gross compared to any generational loss from upsampling. So much so that it's not fair to mention.

Right, so 96k puts the sampling frequency miles above the audio band. Not sure what anyone thinks they need 192k or above to do. I mentioned my tests. There may be some processing that benefits from extreme HD sample rates (I doubt it). When you're done with that, the full program can be put into 96k with no loss.

This goes to the basic question of, what creates “better“ in audio? And is the new different thing necessarily better?

In many cases 192K sounds different than 96K, but is it necessarily better? Too many people in audio, I think, find something that sounds different then what they had before, and automatically assume it is “better“. I’ve been a violinist for over 50 years and played in live orchestras for about that long, and I can guarantee you than sitting in a live orchestra the bass is nowhere as big as when people add a subwoofer, but they swear that a subwoofer is “better“. Or that they must have a perfectly flat 20 Hz to 20,000 Hz system and room response, when the real world is nothing like that. I have never heard a recording reproduced on any system truly sound like a live orchestra that you’re sitting in the middle of, or even sitting in “Row 16 center”. Sound reproduction no matter how good the system, is never the real thing, but to me feels more like something an artist painted that is kind of like the real thing, so that may br about the best you can do. Who doesn’t like a beautiful painting?
And of course the other side of that “better” issue is the status of having “better” gear. Don’t even get me started with that. I DIY power cords and interconnects that are light years beyond some of the expensive stuff you can get. If you have a little imagination and don’t mind rolling your sleeves up, you can have a great result for relatively very little.
Bottom line: if you like it, great. If you can, be happy with that. But if it isn’t quite the real thing, there’s always something that could sound better and you could continually be in that loop of dissatisfaction with the sound, tilting at audio-demon windmills continually trying to find the ultimate sound, which does not exist in reproduced audio once you’ve heard live music or even played it. If you can resist being a junkie audiophile needing the next better “hit“, be happy with being happy with what you have.
You might even try the “audio think method“ and imagine your system is the greatest thing ever and could never be better than that. Yeah, right. Ha ha

Truth be told, a musician in an orchestra rarely gets caught up in the music, and is mostly thinking about how to play better. Sounds familiar, doesn’t it?
 
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Anyone talking about the smaller steps being better does not understand sampling theorem and needs to read Shannon and Nyquist. You can draw pretty pictures that make it look like smaller steps should be better, but that's rubbish it isn't how it works.
I would have to add that the comment about steps was just me wondering out loud. I don't know the specific reasons that they were talking about doubling the 192k standard, but Kelly is one smart guy (and incredibly talented). I should look and see if that webcast is still up online.
 
This goes to the basic question of, what creates “better“ in audio?

For me, "better" is a question that can only be addressed behind the mixing board during the process of creation. After mixing, mastering, and distribution to consumer formats, the interest is now in accuracy to the master mix. I could say the most accurate copy of the master (a clone if it's a digital master) is better. I try to use the word "accurate" to be clear.

The mastering step can get into the weeds...
This can be anything from literally not touching the audio beyond song sequencing and segues to full on restoration work with a troubled live mix that has no multitrack to remix from. Or it can get into the pop market volume war hyped damaging treatment with no respect for the music or mix whatsoever. There are examples of teamwork where an accomplished mastering engineer makes a few tweaks that improve a mix. (Even though it's working from the mix and not the source multitracks.) So one master might be subjectively better sounding than another or even the original raw mix.

The delivery formats are so often attributed to the sound. It's the mix and mastering that account for 99% of what you hear. Unless it's a 192k or lower mp3 or a damaged analog copy or a malfunctioning analog system. That stuff can be very altering.
 
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I am sure you are correct, but that’s not a useful comment. Rather than refer people to somebody else, why don’t you simply explain in a few words how it really does work, so that everybody can easily understand why you are saying that smaller steps are not necessarily better?
It takes more than a few words to explain how sampling theorem works. Far better to read original sources than read me trying to explain it as well as the original.
 
Haha. I like Hawkwind a lot. Always one of my favorites. Most of their albums are truly some of the worst fidelity I've ever heard. It has no charm and only hurts the music.
 
I DIY power cords and interconnects that are light years beyond some of the expensive stuff you can get.
Yep! 16awg or bigger extension cord is premium speaker cable. I make most of my cables. I used Gepco star quad cable and Neutric connectors for balanced xlr cables and I have cables going strong for 30 years that have been on hundreds of live stages. Under $10/cable.

No special plutonium nyborg alloy cables here! Or cermac speaker wire stands. (That magazine back pages ad still stands out to me!) Or stones that give off energy and align the azimuth of your listening shockra. Don't know what my sign is either...
 
If the audio is recorded and mastered well and not over compressed as far as I am concerned CD quality is fine. However I have several 96/24 kHz Remastered files from ZZTop and the Grateful Dead that sound spectacular.
My original 1980s CD pressings of Dire Straits and Fleetwood Mac Rumours still are some of the best ever digital music I have in my collection.
 
192/24 has been around since the advent of DVD~Audio when a lot of the included Stereo layers were indeed at that sampling rate. I don't download or load my discs onto a hard drive so I, personally, welcome 192/24. Most of the audiophile companies, domestic and Japanese, have also included 192 on their DVD~A stereo discs as well as BD~A discs and to my ears they are some of the best sounding discs in my collection. And who can quibble with the recent spate of Rhino QUADIO BD~A releases [The Best of the Doors, Chicago and the Doobies] at that higher sampling rate?

Whether it's overkill as compared to 96/24, I have NO qualms with 192 ..... IMO, anything beats that atrocious 44.1 sampling rate which continues to plague the RBCD.

And while I'm at it, I wish ALL RBCDs would be MQA encoded .... as unfolded, they include 24 bit resolution with high sampling rates and really sound magnificent. DON'T KNOCK IT UNLESS YOU'VE ACTUALLY TRIED IT! Reading about it and experiencing it FIRSTHAND are two different animals altogether!

On this basis it seems that all you've 'tried' are FIRSTHAND comparisons that don't control for mastering differences, playback level differences, or perceptual biases. Methods that are atrocious for actually determining how things sound different due to sampling rate.
 
I think the answer to this is actually quite simple.

TECHNICALLY higher bit rates are better.

IN REALITY it guarantees nothing.

There are simply too many variables in the mix. Discussions like this often occur, but the number 1 variable most everyone forget is the quality of our hearing. Audiophiles often seem to claim to have the best hearing not only in the world today, but ever. In fact, they were blessed with super hearing at birth.

Of course, the reality is different.

At some point I believe we have to recognize that the mastering is the be-all and end-all. Yes, sure, we could take extreme stances to suggest this isn't the case, but in the real world, the mastering trumps the media, bit rates, and everything else. When it comes to quality, that's where the conversation starts and ends the majority of the time.

The rest is mostly down to the equipment - amps, speakers, decks, etc.

But yes - technically, higher is better. The issue is, it's a gross simplification when it comes to what sounds better.

IMO
 
Okay what I'm getting from all this is, record in whatever makes you happy, as long as your master is done right (provided there are no recording discrepancies).
 
This goes to the basic question of, what creates “better“ in audio? And is the new different thing necessarily better?

A few points about this post:

sitting in a live orchestra the bass is nowhere as big as when people add a subwoofer, but they swear that a subwoofer is "'better“.

That actually depends on how loud the subwoofer is. Too many people crank it up too much. When correctly set up, the listener should not notice it. The purpose of the subwoofer is to cover the frequencies the speakers don't cover.

a perfectly flat 20 Hz to 20,000 Hz system and room response,

This is also unattainable in practice in the listening room.

I have never heard a recording reproduced on any system truly sound like a live orchestra that you’re sitting in the middle of, or even sitting in “Row 16 center”.

I don't want to hear it as though I am sitting in the orchestra. As for sitting in the audience, I have created such recordings.

I DIY power cords and interconnects that are light years beyond some of the expensive stuff you can get

There is nothing special about interconnecting cords. As long as the cord impedance is nowhere as large as the load impedance and the cord has any necessary shielding, the cord will stay out of the sound.

"Better" is in the ear of the beholder.
 
The first time I listened to a CD through a tube amp, I thought it didn't sound right. It was with a Sony walkman CD player. I got some other CD players after that which were better quality, but there was always something about it that didn't sound right. It sounds like it lacks dimension in some ways. I tried listening to 24/96 audio DVDs and SACDs and they sound better as in more natural or closer to analog. I also tried recording things on a DAT 16/44.1 and 16/48 and thought 16/48 was superior.

See this thread about Si vs. Ge and tubes vs. transistors.
http://www.rickresource.com/forum/viewtopic.php?f=44&t=415236
 
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64 bits or Floating Point arithmetic in the mixing would be better
I think most DAWs have 64-bit audio engines. I know Pro Tools does. All processing is done in that domain.
PS. No one knows what the sound of ultrasonic noise is because no audio speakers reproduce that high up. Or human hearing. If someone things they're "feeling" something... Well it's not coming out of your speakers.
Not true. Many amplifiers/speakers have bandwidth extending above the 20kHz threshold. My ADAM Audio A7X powered monitor speakers with air motion transformer tweeters go up to 50kHz. And while we can't hear that high, intermodulation distortion means that those ultrasonic frequencies can affect the frequencies we do hear. Hypothetically, assuming high-bandwidth gear was used in every step of the recording/mixing/mastering/playback chain, this could lead to a more true-to-life experience. Unfortunately, there's no good objective way to test that hypothesis.

Regardless, the chief benefits of higher sample rates are:
  • Ability to use gentler slopes for the anti-aliasing (recording) and reconstruction (playback) filters in the ADC/DAC, preserving greater phase linearity in the audible spectrum
  • Smoother, less audibly lossy temporal manipulation in the digital realm (analogous to using high frame rates for slow motion video)
 
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The first time I listened to a CD through a tube amp, I thought it didn't sound right. It was with a Sony walkman CD player. I got some other CD players after that which were better quality, but there was always something about it that didn't sound right. It sounds like it lacks dimension in some ways. I tried listening to 24/96 audio DVDs and SACDs and they sound better as in more natural or closer to analog. I also tried recording things on a DAT 16/44.1 and 16/48 and thought 16/48 was superior.

How did you 'try' them?

See this thread about Si vs. Ge and tubes vs. transistors.
http://www.rickresource.com/forum/viewtopic.php?f=44&t=415236


(after seeing it). Stop reading audiophile silliness and start reading about how to control for common variables when determining audible difference.
 
Okay what I'm getting from all this is, record in whatever makes you happy, as long as your master is done right (provided there are no recording discrepancies).
Always record at 24 bit or higher.

I personally record at higher sample rates than 44 or 48kHz. But whatever differences there might be, they are often (if not always) negligible compared to bit depth differences.
 
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