4 ms·

If the only thing you do with your music is listen, then yes, 24/192 delivers questionable value compared to the more popular 16/44.1 or 16/48 formats. However

If the only thing you do with your music is listen, then yes, 24/192 delivers questionable value compared to the more popular 16/44.1 or 16/48 formats. However, all musicians I know use these high-rez formats internally. The reason for that, when you apply audio effects, especially complex VST ones, these discretization artifacts noticeably decrease the result quality. Maybe, the musicians who distribute their music in 24/192 format expect their music to be mixed and otherwise processed.

zaroth 9y ago
TFA specifically addresses this and agrees 100% with you on it. Develop in 24/192, but ship in 16/48. Good point about supporting better downstream remixing!
- daurnimator 9y ago
  I feel like there is some parallel to open source and GPL type licenses: you can never know which of your customers may wish to remix or further work on your materials, so you should ship the "source" material.
kazinator 9y ago
I do not believe it. 24 bits is definitely needed for processing (better yet, use floating-point). Not 192 kHz; no friggin' way. Repeated processing through multiple blocks at a given sample rate does not produce cumulative discretization problems in the time domain; it's just not how the math works.
- WalterSear 9y ago
  Unless you are altering the time or pitch, which, these days, you are more than you are not. But no, musicians aren't releasing things at ultra-resolutions because they expect others to reuse their work. The ones that are, are providing multitracks.
- mbell 9y ago
  > Repeated processing through multiple blocks at a given sample rate does not produce cumulative discretization problems in the time domain; it's just not how the math works. That isn't entirely true. e.g. It's common for an audio DSP to use fixed point 24bit coefficients for an FIR filter. If you're trying to implement a filter at low frequency then there can be significant error due to coefficient precision, that error is reduced by increasing the sampling rate.
- Const-me 9y ago
  > better yet, use floating-point Both your inputs (ADC) and outputs (DAC) are fixed-point. Why would you want to use a floating point in between? Technically, 64-bit floating point format would be enough for precision. But that would inflate both bandwidth and CPU requirements for no value. 32-bit floating point ain’t enough. Many people in the industry already use 32-bit integers for these samples. > Not 192 kHz; no friggin' way. I think you’re underestimating the complexity of modern musician-targeted VST effects. Take a look: https://www.youtube.com/watch?v=-AGGl5R1vtY https://www.youtube.com/watch?v=-AGGl5R1vtY I’m not an expert, i.e. I’ve never programmed that kind of software. But I’m positive such effects are overwhelmingly more complex than just multiply+add these sample values. Therefore, extra temporal resolution helps. BTW, professionals use 24bit/192kHz audio interfaces for decades already. E.g. ESI Juli@ was released in 2004, and that was very affordable device back then.
  - deleted 9y ago
    [deleted]
  - euyyn 9y ago
    > Therefore, extra temporal resolution helps. But it's not resolution, right? It's extra frequencies outside the audible range. Is there any natural process that would make those affect the audible components, if I were listening to the music live instead of a recording?
    - sitharus 9y ago
      Yes, you can get a beat effect (https://en.wikipedia.org/wiki/Beat_(acoustics) https://en.wikipedia.org/wiki/Beat_(acoustics)) as the ultrasonic frequencies interfere with the audible ones. It's generally not considered desirable.
      - kazinator 9y ago
        Only if the two are mixed in a nonlinear way (e.g. "heterodyned"). If a sonic and ultrasonic frequency are combined together, but a low pass filter doesn't pass the ultrasonic one, the ultrasonic one doesn't exist on the other end. Hence, there can be no beat.
- jancsika 9y ago
  > Not 192 kHz; no friggin' way. That's an anthropocentric statement. I want to leave it at that for now just to see if I get dinged for low effort.
  - jancsika 9y ago
    Only one ding? I'm disappointed. Anyway, check out the frequency range for other animals: https://en.wikipedia.org/wiki/Hearing_range https://en.wikipedia.org/wiki/Hearing_range Notice that there are a number of species whose range extends well past 20kHz. Even with 192kHz you're still chopping off the upper end of what dolphins and porpoises can hear and produce. So please convince Apple and friends that you need 200+kHz to truly capture the "warmth" of Neil Young's live albums. Then we'll be able to crowdsource recording all the animals and eventually synthesize the sounds to communicate with them all. Maybe then we can synthesize a compelling, "We're really sorry, we promise we're going to fix all the things now," for the dolphins and convince them not to leave. :)
- nullc 9y ago
  > Not 192 kHz; no friggin' way. It can be useful to run your signal processing chain at a higher rate because many digital effects are not properly bandlimited internally (and it would be pretty CPU hungry to do so). But that doesn't mean you need to store data even that you'll process later at 192KHz though it might be easier to do so.
- dharma1 9y ago
  for DAW use - this has been mentioned before on this thread, but 192khz has two beneficial effects - lower latency for realtime playing, and it "forces" some synths and effects to oversample, thus reducing aliasing. All this comes at a high computational and storage cost though. I personally use 44khz 24bit settings for DAW use.
- Applejinx 9y ago
  Lots of my plugin users are resorting to massive oversampling, even when it's not appropriate, simply because in their experience so many of the plugins they use are substantially better when oversampled. 192K is an extremely low rate for oversampling.
  - abpavel 9y ago
    I wonder how you measure the quality difference for higher rates? 384K, 512K and beyond? I hear from audiophiles that there is a very distinct difference, but there is absolutely no basis for it in science.
    - Applejinx 9y ago
      Not so: this oversampling is mostly about generating distortion products without aliasing, so in the context I mean, the difference is obvious. But, it's a tradeoff- purer behavior under distortion, versus a dead quality that comes from extreme over-processing. I've never seen an audiophile talk about 384K+ sampling rate unless they mean DSD, and with that, it's for getting the known unstable HF behavior up and out of the audible range. Oversampling in studio recording is mostly about eliminating aliasing in software that's producing distortion, and it's only relevant in that context: I don't think it's nearly so relevant on, say, an EQ.
  - kazinator 9y ago
    That could be simply that some of these plugins are either written to specifically work correctly only at 192 kHz; i.e. it's a matter of matching the format they expect.
- leni536 9y ago
  What if I simply want to slow down the signal 2 times (without pitch correction)? Then 44.1 kHz is obviously not enough. Maybe 192 kHz is way overkill though, but I would argue that that's the point. You don't want it to become a bottleneck ever for any effects.
  - kazinator 9y ago
    It is only insufficient if there are components beyond 20 kHz that you want to bring down into the audio range as you're slowing down.
mwcampbell 9y ago
To truly support downstream remixing, musicians would need to distribute the original, separate tracks. Occasionally an independent musician will do this, but it's not at all common AFAIK.
- Const-me 9y ago
  For software, you have a source code, you compile it with a standard-complying compiler, and you’ll more or less reproduce the result. Music is different. If you have the original multi-track composition, you can’t reproduce the result unless you also have the original DAW software (of the specific version), and all the original VST plugins (again, of the specific version each). All that software is non-free, and typically it’s quite expensive (esp. VSTi). Some software/plugins are only available on Windows or Mac. There’s little to no compatibility even across different versions of the same software. Musicians don’t support nor maintain their music. Therefore, DAW software vendors don’t care about open formats, interoperability, or standardization.
  - ibn_ibid 9y ago
    You could bounce each track as an individual audio file
  - mwcampbell 9y ago
    All true. One independent musician that I like, Kevin Reeves, made the tracks from his first album available for free download for a while. But he published them as WAV files. So, of course, one could only create one's own mixes, not easily reproduce his. But if he had just posted the files saved by his DAW (or his mix engineer's DAW), that would have been useless to just about everyone. Aside: Though the album was completed in mid-2006, it was recorded with DAW software that was obsolete even then, specifically an old version of Pro Tools for Mac OS 9, because both Kevin and his mix engineer (for that album) are blind, and Pro Tools for OS X was inaccessible to blind users at the time. BTW, I'm merely a dilettante when it comes to recording and especially mixing.
  - golergka 9y ago
    Traktor stems and remix decks though.

zaroth 9y ago

TFA specifically addresses this and agrees 100% with you on it. Develop in 24/192, but ship in 16/48. Good point about supporting better downstream remixing!

daurnimator 9y ago
I feel like there is some parallel to open source and GPL type licenses: you can never know which of your customers may wish to remix or further work on your materials, so you should ship the "source" material.

kazinator 9y ago

I do not believe it. 24 bits is definitely needed for processing (better yet, use floating-point). Not 192 kHz; no friggin' way. Repeated processing through multiple blocks at a given sample rate does not produce cumulative discretization problems in the time domain; it's just not how the math works.

WalterSear 9y ago
Unless you are altering the time or pitch, which, these days, you are more than you are not. But no, musicians aren't releasing things at ultra-resolutions because they expect others to reuse their work. The ones that are, are providing multitracks.
mbell 9y ago
> Repeated processing through multiple blocks at a given sample rate does not produce cumulative discretization problems in the time domain; it's just not how the math works. That isn't entirely true. e.g. It's common for an audio DSP to use fixed point 24bit coefficients for an FIR filter. If you're trying to implement a filter at low frequency then there can be significant error due to coefficient precision, that error is reduced by increasing the sampling rate.
Const-me 9y ago
> better yet, use floating-point Both your inputs (ADC) and outputs (DAC) are fixed-point. Why would you want to use a floating point in between? Technically, 64-bit floating point format would be enough for precision. But that would inflate both bandwidth and CPU requirements for no value. 32-bit floating point ain’t enough. Many people in the industry already use 32-bit integers for these samples. > Not 192 kHz; no friggin' way. I think you’re underestimating the complexity of modern musician-targeted VST effects. Take a look: https://www.youtube.com/watch?v=-AGGl5R1vtY https://www.youtube.com/watch?v=-AGGl5R1vtY I’m not an expert, i.e. I’ve never programmed that kind of software. But I’m positive such effects are overwhelmingly more complex than just multiply+add these sample values. Therefore, extra temporal resolution helps. BTW, professionals use 24bit/192kHz audio interfaces for decades already. E.g. ESI Juli@ was released in 2004, and that was very affordable device back then.
- deleted 9y ago
  [deleted]
- euyyn 9y ago
  > Therefore, extra temporal resolution helps. But it's not resolution, right? It's extra frequencies outside the audible range. Is there any natural process that would make those affect the audible components, if I were listening to the music live instead of a recording?
  - sitharus 9y ago
    Yes, you can get a beat effect (https://en.wikipedia.org/wiki/Beat_(acoustics) https://en.wikipedia.org/wiki/Beat_(acoustics)) as the ultrasonic frequencies interfere with the audible ones. It's generally not considered desirable.
    - kazinator 9y ago
      Only if the two are mixed in a nonlinear way (e.g. "heterodyned"). If a sonic and ultrasonic frequency are combined together, but a low pass filter doesn't pass the ultrasonic one, the ultrasonic one doesn't exist on the other end. Hence, there can be no beat.
jancsika 9y ago
> Not 192 kHz; no friggin' way. That's an anthropocentric statement. I want to leave it at that for now just to see if I get dinged for low effort.
- jancsika 9y ago
  Only one ding? I'm disappointed. Anyway, check out the frequency range for other animals: https://en.wikipedia.org/wiki/Hearing_range https://en.wikipedia.org/wiki/Hearing_range Notice that there are a number of species whose range extends well past 20kHz. Even with 192kHz you're still chopping off the upper end of what dolphins and porpoises can hear and produce. So please convince Apple and friends that you need 200+kHz to truly capture the "warmth" of Neil Young's live albums. Then we'll be able to crowdsource recording all the animals and eventually synthesize the sounds to communicate with them all. Maybe then we can synthesize a compelling, "We're really sorry, we promise we're going to fix all the things now," for the dolphins and convince them not to leave. :)
nullc 9y ago
> Not 192 kHz; no friggin' way. It can be useful to run your signal processing chain at a higher rate because many digital effects are not properly bandlimited internally (and it would be pretty CPU hungry to do so). But that doesn't mean you need to store data even that you'll process later at 192KHz though it might be easier to do so.
dharma1 9y ago
for DAW use - this has been mentioned before on this thread, but 192khz has two beneficial effects - lower latency for realtime playing, and it "forces" some synths and effects to oversample, thus reducing aliasing. All this comes at a high computational and storage cost though. I personally use 44khz 24bit settings for DAW use.
Applejinx 9y ago
Lots of my plugin users are resorting to massive oversampling, even when it's not appropriate, simply because in their experience so many of the plugins they use are substantially better when oversampled. 192K is an extremely low rate for oversampling.
- abpavel 9y ago
  I wonder how you measure the quality difference for higher rates? 384K, 512K and beyond? I hear from audiophiles that there is a very distinct difference, but there is absolutely no basis for it in science.
  - Applejinx 9y ago
    Not so: this oversampling is mostly about generating distortion products without aliasing, so in the context I mean, the difference is obvious. But, it's a tradeoff- purer behavior under distortion, versus a dead quality that comes from extreme over-processing. I've never seen an audiophile talk about 384K+ sampling rate unless they mean DSD, and with that, it's for getting the known unstable HF behavior up and out of the audible range. Oversampling in studio recording is mostly about eliminating aliasing in software that's producing distortion, and it's only relevant in that context: I don't think it's nearly so relevant on, say, an EQ.
- kazinator 9y ago
  That could be simply that some of these plugins are either written to specifically work correctly only at 192 kHz; i.e. it's a matter of matching the format they expect.
leni536 9y ago
What if I simply want to slow down the signal 2 times (without pitch correction)? Then 44.1 kHz is obviously not enough. Maybe 192 kHz is way overkill though, but I would argue that that's the point. You don't want it to become a bottleneck ever for any effects.
- kazinator 9y ago
  It is only insufficient if there are components beyond 20 kHz that you want to bring down into the audio range as you're slowing down.

mwcampbell 9y ago

To truly support downstream remixing, musicians would need to distribute the original, separate tracks. Occasionally an independent musician will do this, but it's not at all common AFAIK.

Const-me 9y ago
For software, you have a source code, you compile it with a standard-complying compiler, and you’ll more or less reproduce the result. Music is different. If you have the original multi-track composition, you can’t reproduce the result unless you also have the original DAW software (of the specific version), and all the original VST plugins (again, of the specific version each). All that software is non-free, and typically it’s quite expensive (esp. VSTi). Some software/plugins are only available on Windows or Mac. There’s little to no compatibility even across different versions of the same software. Musicians don’t support nor maintain their music. Therefore, DAW software vendors don’t care about open formats, interoperability, or standardization.
- ibn_ibid 9y ago
  You could bounce each track as an individual audio file
- mwcampbell 9y ago
  All true. One independent musician that I like, Kevin Reeves, made the tracks from his first album available for free download for a while. But he published them as WAV files. So, of course, one could only create one's own mixes, not easily reproduce his. But if he had just posted the files saved by his DAW (or his mix engineer's DAW), that would have been useless to just about everyone. Aside: Though the album was completed in mid-2006, it was recorded with DAW software that was obsolete even then, specifically an old version of Pro Tools for Mac OS 9, because both Kevin and his mix engineer (for that album) are blind, and Pro Tools for OS X was inaccessible to blind users at the time. BTW, I'm merely a dilettante when it comes to recording and especially mixing.
- golergka 9y ago
  Traktor stems and remix decks though.