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Is 192kHz hi-res audio recording in studio worth it?


Øyvind Skald

What do you record in?  

27 members have voted

  1. 1. What resolution?

    • 44.1 kHz
      9
    • 48 kHz
      4
    • 44.1/48 kHz
      8
    • 88.2 kHz
      2
    • 96 kHz
      4
    • 192 kHz
      0
  2. 2. Do you think there is a point of recording in more than 44.1KHz

    • Yes
      16
    • No
      11


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1 hour ago, Wibbles said:

At this point, 6 out of 15 voters record at higher than 44.1kHz, but only 4 out of 15 think there's any point at recording at higher rate.

I am one of those 4,  so at least 3 of the voters recording at higher than 44.1kHz think there's no point.  9_9

I did put the  44.1/48 kHz up there because YouTube only uses 48 kHz audio. So for making music/sounds to YouTube i guess using 48 kHz is best.  I cant hear any loss in sound when resampling from 44.1 kHz to 48 kHz.  But  i  do record in it when making something specific for YouTube/video, just just to get the same sound trough to the video.

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Here we go again. It can be argued that the math show exactly that there's nothing to be gained within the spectrum of audio that humans can actually perceive by going higher than 44.1. My buddy who is a speaker designer, worked for Gibson, JBL now Samsung and does TONS of measurements AND is a musician too agrees.

Old me? I have recorded in many, many places, from Capitol Studios and Ocean Way to a closet; have used endless amounts of gear, from $5000 preamps to $2 ones. Have engineered., mixed, mastered for myself and others. Recorded a track or two for "Modern Art" for which we were nominated for a Grammy at home through a ART DPS2, a $150 preamp into my 2408 and everybody thought the sound killed. I've experimented with higher sample rates and have come to the conclusion that there is no difference, other than some synths or programs that simply behave differently when running at 96K; for the audio itself there is NO discernible difference in my opinion.

 

R

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22 hours ago, pwalpwal said:

jut to be clear, i'm not talking about recording audio at higher res, nor playback at higher resolutions, i'm talking about in the box mixing/summing of 1s and 0s, dither be damned :D

It doesn't matter. The stair-steps are one way to interpret a series of values on a graph. But the lines between the points are imaginary and don't exist in reality. We are talking about pulses with a certain amplitude, taken at a constant time interval. Think a series of lollipops with equal distance to each other, where only the length of the stick varies.

This has implications, because it means that as long as a signal contains for example no frequencies above 20 kHz, it doesn't matter if it's captured at a pulse rate of 44.1 kHz, or 192 kHz. The reconstructed signal looks identical. There are no stairs that get smaller at higher sample rates. It has nothing to do with dither, which would only be relevant to the lengths of the lollipop sticks, not the (imaginary) information between them.

The zero-order-hold (stair stepped) representation of digital voltage sampling is maybe one of the biggest reasons why people think analog is somehow intrinsically more clean/continuous than digital audio, when in reality even the cheapest AD/DA chipsets beat practically any analog recording medium in terms of noise and lack of distortion.

This is why when you right-click a WAV or AIFF file in Windows, you'll be told that the file contains PCM data, where PCM stand for Puls-Code-Modulation.

By the way, I voted 96 kHz because I made a few tests and came to the conclusion that I can substantially mitigate aliasing artifacts when using heavy distortion. As the final delivery medium I'm completely content with properly band-passed 44.1 kHz.

Edited by Michael Anderwald
Typo.
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21 hours ago, synkrotron said:

recording/mixing/mastering my "music" is not going to help it at all... 

Of course that's just me being negative so that you all go check out my shit and make a purchase.

 

 

have a nice day

I already have, thanks. Great stuff.  :D

 

Wibbles loves a drone. B|

 

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21 hours ago, synkrotron said:

recording/mixing/mastering my "music" is not going to help it at all... 

Of course that's just me being negative so that you all go check out my shit and make a purchase.

 

 

have a nice day

I dont know if you have any Inspiration from Christopher Franke, but I can  hear something there.

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Pwalpal, I went through that npr thing a while back and I think I got about 66% which I suppose is good.  I actually ended up there after a video from Rick Beato on the topic using his friend as the young, perfect pitch hearing music major.  She got 66%.  I felt good matching that, but in all honesty I could only discern the difference when hearing them right after each other.  If I heard them 10 minutes apart there is no way I could do anything other than guess. 

It's not exactly the same measure as this thread but similar I think at the end of it. 

I use 24/44.1 here. I am happy with the results and as mentioned above, my room, my equipment and my talent at playing, mixing, and mastering will fail me before the differences in these settings will. 

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For me, it's not even necessarily for the benefit of those that may notice the difference. I work hard to make my songs the best that I can make them. I record at 24/96 because I want them to represent my efforts as accurately as I possibly can. I want them to sound as good as I'm able to for MY benefit, more than anyone else's.

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Too friggin drunk to read all the hard core advice here, so sorry if this is already covered. My opinion lay below layman, so may not be worth a hill of compost in the bigger, more informed picture.

Maybe depends on what the target of the end product is.

If you are translating to another system with thousand dollar monitors that need every nuance of a 192/24 or floating 32bit representation, then you got to go with the big guns.

Some claim that 24 bit depth can record trails of faint reverbs or the like that merit the argument to record at higher bit rates.. Reference the depth of the noise floor. Twice, three or way more better than what the ancient rock and roll masters had struggling with an Ampex tape machine. 56 db s/n ratio was all they had, but they did magic.

If the target is a CD boombox, you are stuck with the 44.1k/16 bit protocol. Everything you have oversampled is lost. Noise floor comes up to 16 bit obscuring the silent subtleties. All connect the dots points on the digital timeline are thrown out down sampling to 44.1 k.

Anything oversampled to get pristine quality has to be degraded to make the CD. Anything undersampled has to be bloated up to an empty space with no added data.

This why in my ignorance, since my target is eventually a CD,  record at 44.1k and 16 bit depth, just do it the way it ultimately has to be restructured to make the CD and not having to worry about artifacts in the conversion.

Something I found. My old PCI Delta 1010-LT allows recording at 22.5 k or whatever close. Gets beyond a curse of pristine digital unforgiveness and seems to throw in a bit of old analog goodness.

My opinion, if you are going to publish a CD, do 44.1 k and 16 bit depth. All else upper and lower is lost until they update the CD standard, and you are at the mercy of the integrity of whatever you have chosen to up sample or downsample your final mix.

My two cents.

John

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Here's my two cents (possibly more, possibly less!).

1) Personally, I did a test on a $4,000 stereo system (15 years ago) and "I" couldn't tell the difference between a 192 kbps .mp3 and anything higher so I've imported my huge collection of music (about 350,000 songs currently) into my computer at that resolution to save disk space.

 

2) That said, I also have done PhD work into brain functioning and other woo-woo experimental areas and there's some strong evidence that we can experience frequencies far beyond the typical 20-20,000 Hz that we can hear.  Frequencies as high as 100 kHz appear to be absorbed by our bodies allowing for expanded abilities like accelerated learning (I happen to have a device called an Echofone that was originally used in experiments with dolphins as well as a Flanigan Neurophone which utilizes bone conduction to experience higher frequencies and can actually allow deaf people to hear).

 

3) CD's record at 44.1k, however many sound processors (especially the earlier ones) end up rounding any math applied to incoming signals which reduces the overall sampling rate into the 30's (e.g., 32k).  This accounts for the lo-fi sound of many synths from the 80's.  This is why I used to (and will again when I have another studio) record at 96k and then reduce to 44.1k at the very end of the process.  Now, any rounding will happen above 44.1k so the end result sounds full.

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Thanks Craig.

Can understand lessening artifacts by downsampling from a higher resolution to get the CD out.

Still don't understand why if the CD is a target, why not just do 44.1/16 and do away with any struggle trying to minimize artifacts in an extra conversion process that has to happen carved in stone.

My understanding is that any advantage is dumped in the down conversion, and the conversion itself adds garbage even if possibly too subliminal for us to hear depending on what we used to convert with.

Somebody set me straight so I can make sense of this.

John

 

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8 hours ago, John K said:

Still don't understand why if the CD is a target, why not just do 44.1/16 and do away with any struggle trying to minimize artifacts in an extra conversion process that has to happen carved in stone.

When done right, there aren't any artifacts due to conversion that wouldn't be much worse when working in 44.1 kHz and 16 bits all through the rendering pipeline. The hard part about sampling rate conversion is finding the best sounding low-pass filter for the material (which is solved very simply by using one of the popular sample rate conversion algorithms like SoX , or Audacity, or r8brain, or any other good one as can be found on http://src.infinitewave.ca/).

Converting from 24 bit to 16 bit has no artifacts either. You either simply truncate numbers which raises the noise floor (which in 16 bit resolution can result e.g. in audible distortion of long reverb tails at high listening levels), or you use dithering to utilize the statistic distribution of signals below the 16 bit noise floor, which adds a tiny bit of noise, but that noise contains audible information thus lowering the noise floor even further. Just like gradients are dithered in digital imaging to avoid banding with hard edges in gradients.

8 hours ago, John K said:

My understanding is that any advantage is dumped in the down conversion, and the conversion itself adds garbage even if possibly too subliminal for us to hear depending on what we used to convert with.

Not all advantage is being dumped, and the conversion doesn't add garbage. Unless you consider dithering garbage, in which case you just leave it off without losing anything that would have been there in the first place.

People used to compare this to saying "It doesn't make sense to shoot a movie on 35mm film stock, when the movie is going to be released straight to VHS." So, you're trying to keep the production quality as high as you can, as long as you can from the beginning of the production process in order to end up with the best possible end product. This way you capture the most information at data acquisition and lose the least of it during the steps further down the line.

Best,
Michael

 

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