8 ms·

While the extra quality is lost on listeners, from experience I've found that super-HQ source material can change your results (for the better) when fed through

by tomc1985 9y ago

While the extra quality is lost on listeners, from experience I've found that super-HQ source material can change your results (for the better) when fed through distortion/compression/warming effects processors

mrob 9y ago
Is that because of the HQ audio, or because the processing software introduces less artifacts at those sample rates/bit depths? I.e. could you get the same results by up-sampling CD quality audio?
- trevyn 9y ago
  I imagine it's because lossy codecs are tuned for human perceptual limitations, and when you process audio, it can "pull" areas of the sound that are otherwise hidden from your perception into perceptual ranges, analogous to how fiddling with the brightness/contrast of a highly compressed JPEG image can accentuate the artifacts.
  - anyfoo 9y ago
    We are not really talking about lossy codecs here, though, are we?
    - trevyn 9y ago
      Ah, my mistake. But perhaps you could consider 24/192 -> 16/48 a lossy codec in this context; the same argument does apply, for the same reasons.
      - anyfoo 9y ago
        That way it does make sense, editing might require higher fidelity than we can perceive in the final project to avoid artifacts.
    - kuschku 9y ago
      Actually, the article is not talking about it, but the topic very much is lossy codecs. Specifically, if 24/192 AAC is worth it compared to 16/44.1 AAC (and the answer to that is yes, although the answer to 24/192 WAV is no)
  - microcolonel 9y ago
    But what about lossless 16/44.1KHz audio? It's not a "compressed" version of 24/192Khz. Chances are that your 192KHz music will be low pass filtered anyway, 192KHz is a trick that sound chipset vendors added so that integrators wouldn't have to make high quality analog filters for their ADCs. There's likely nothing above 22.05KHz except shaped noise, production artifacts, and very, very quiet ultrasound noises which have nothing to do with the music.
    - TheOtherHobbes 9y ago
      There's no such thing as lossless 16/44.1 because it's practically impossible to make an artefact-free antialiasing/reconstruction filter pair clocked at 44.1k. The point of higher sample rates isn't to capture bat squeaks, but to relax the engineering constraints on pre-post filtering. Nyquist is fine in theory, and if you've never actually tried to implement a clean filter you'll likely watch the xiph video and think "Well that makes sense." If actually know something about practical DSP and the constraints and challenges of real filter design you're not going to be quite so easily impressed. Likewise with higher bit depths. "Common sense" suggests that no one should be able to hear a noise signal at -90dB. Common sense is dead wrong, because the effects of a single bit of dither are absolutely audible. And if you can hear the effects of noise added at -90dB, you can certainly hear the effects of quantisation noise artefacts on reverb tails and long decaying notes at -30 to -40dB, added by recording at 16 bits instead of 24 bits. Whether or not that level of detail is present in a typical pop or classical recording is a different issue. Realistically most music is heavily compressed and limited, so the answer is usually "no." And not all sources have 24-bits of detail [1]. (Recordings made on the typical digital multitrack machines used in the 80s and 90s certainly don't.) That doesn't mean that a clean unprocessed recording of music with a wide dynamic range made on no-compromise equipment won't show the difference clearly. Speaking from experience, it certainly does. [1] Technically no sources have 24-bits of detail. The best you'll get from a real world converter is around 22-bits.
      - microcolonel 9y ago
        > you'll likely watch the xiph video and think "Well that makes sense." What video? This thread is about an article. > The point of higher sample rates isn't to capture bat squeaks, but to relax the engineering constraints on pre-post filtering. I just said that, I think.
- tomc1985 9y ago
  I think it's because processes that affect things at the individual sample level (which distortion/compression often do) have a lot more detail to work with. The author mentions this a bit in the article
- TazeTSchnitzel 9y ago
  More bits means: • Less quantisation noise, so your noise floor is a bit lower and therefore, when you're mixing stuff together, you accumulate less noise • More numerical room to play with, you can scale or shift the volume up and down more without clipping and with less loss of accuracy With 16-bit CD audio, you can't just convert to 24-bit and suddenly have less noise. You might get more room, though. As for higher sampling rates (more kHz, if you will), I think Monty mentioned some benefit regarding less sharp aliasing filters (can have a larger transition band from 20–48kHz, say, rather than from 20—22.1kHz), but it's not something I understand the benefit of well.
  - dspig 9y ago
    filters with a sharper cutoff will 'ring' more, smearing sharp changes in the signal out over time. It's a pretty subtle effect when it's all happening above 20 kHz though
- squeaky-clean 9y ago
  For distortion and compression, the only possible difference I can think of is aliasing, and yeah you could just oversample your input, then downsample again on the way out. This is really really common in most professional plugins, often with an adjustable amount to balance quality and CPU usage. I also have some gear that aliases at a volume you can't hear, but when you plug it into an analog distortion pedal, the aliasing in the high frequencies becomes apparent. This would be avoided if it had a higher sample rate so the aliasing was far out of the audible range. For other sorts of effects like spectral processing, pitch shifting, the extra detail above 22khz really does make a difference, especially if pitching down.
- davidgerard 9y ago
  Because modern production can involve hundreds or even thousands of digital effects. So 24-bit, or even 32-bit float just because you can, is standard. As Monty demonstrates, it's a fraudulent waste to try to sell the result as a product to the end listener.
acchow 9y ago
> Its playback fidelity is slightly inferior to 16/44.1 or 16/48, and it takes up 6 times the space. The article is highly technical. Does anyone have a way to describe this phenomenon intuitively?
- mikewhy 9y ago
  "Diminishing returns"?
- hamandcheese 9y ago
  24/192 can capture audio well beyond what we can hear. These sounds can, though, lead to distortion that is within our hearing range, depending on the equipment used. Therefore, rather than just being useless extra data, it can be actively harmful to the listening experience.
  - tomc1985 9y ago
    It's not that much different to think of than 480p vs 4K/2160p -- dramatically higher resolution, increasing the width and length of the 'grid' that sound is recorded into -- except that in the case of audio fidelity, quality after a certain point causes problems with playback systems, resulting in audible artifacts and distortion.
    - pzone 9y ago
      That analogy works if you modify it: - 480p vs 2160p is measuring the resolution of your cell phone propped up on a pillow at the other end of your living room - experimental evidence shows that your eyesight is not good enough to pick up on the increased resolution, you've maxed out your sensory perception - Your phone stutters trying to stream at 4k so the playback might actually be worse.
      - tomc1985 9y ago
        All good points! Indeed all that extra data makes the decode stage that much more difficult. Same happens with audio too, but high-res audio isn't nearly as bulky as video or nearly as resource-intensive to decode
sp332 9y ago
Sure, when you're editing music it can be helpful to have super high sample rates and lots of bits to work with. But for listening, it's just a waste of space.
- kobeya 9y ago
  He was talking about listening.
  - sp332 9y ago
    tomc1985 said "fed through distortion/compression/warming effects processors" which is editing, not listening.
    - tomc1985 9y ago
      Yeah, editing
      - ioquatix 9y ago
        Lots of pre-amps can do this in real time, and some installation audio systems do it as part of pre-processing (e.g. compression, limiter, etc).
jotux 9y ago
This is specifically addressed in the link under the section titled "When does 24 bit matter?"
InclinedPlane 9y ago
Well, yeah. For properly mixed audio, you don't need anything beyond 44-khz/16-bit. If you need to do extensive processing, or you're working with raw audio and want the extra margin to be able to recover a sound when it may be too quiet (or too loud) or otherwise compromised then more samples and greater bit-depth can be highly useful.
jchanimal 9y ago
I've heard (and it passes the smell test) that even if your ears can't resonate at frequencies higher than human hearing, the space you are in can, and your ears are sensitive to the differences in arrival time that are smaller than the pitches we can't hear. So at least in high sample rate music the imaging could potentially be cleaner. Assuming super great everything else.
- stephen_g 9y ago
  What you're sort of describing - having more of the higher frequencies resonate together - can actually cause intermodulation distortion (which itself can be in the audible range), which can make it sound worse... I haven't had time to check the linked article but if it's the one I'm thinking of I'm pretty sure it goes into this. In the end though, double blind testing (again, I think mentioned in the article) shows that there's a threshold (a bit below 48kHz, probably why 44.1kHz was chosen for CD audio) after which people cant distinguish between higher sample rates at any more accuracy than randomly guessing.
  - jchanimal 9y ago
    I think it's the intermodalation distortion or as I call it "the room sound" that your non-conscious mind perceives. The best example of this I've heard is when listening to identical masters on SACD/DSD vs PCM audio. Notably different sound when reproduced via loudspeakers.
- __david__ 9y ago
  That's one of those things that sounds true without putting thought into it but is ultimately false. A 44.1kHz sampled sounds can encode phases with offsets smaller than 22µs (1/44.1k). How much smaller? I'm not exactly sure, but I believe it's dependent on your sample size. My guess is 1/2^16 (assuming sample resolution of 16 bits), but I might be off by a factor of 2 (or indeed 100% wrong) there. Caveat emptor. If I'm right then that's 346 pico-seconds, which I'm betting is smaller than your brain can differentiate.
  - deleted 9y ago
dangayle 9y ago
It's also great if you you're doing a bunch of time/pitch mangling.
kazinator 9y ago
In the amplitude domain, you definitely need more bits for that type of processing, and many other types. Digital distortion basically simulates high gain with soft clipping, which basically takes a narrow slice of the amplitude domain and magnifies it. The extra resolution has to be there for that not to turn into ugly sounding shit with artifacts.

mrob 9y ago

Is that because of the HQ audio, or because the processing software introduces less artifacts at those sample rates/bit depths? I.e. could you get the same results by up-sampling CD quality audio?

trevyn 9y ago
I imagine it's because lossy codecs are tuned for human perceptual limitations, and when you process audio, it can "pull" areas of the sound that are otherwise hidden from your perception into perceptual ranges, analogous to how fiddling with the brightness/contrast of a highly compressed JPEG image can accentuate the artifacts.
- anyfoo 9y ago
  We are not really talking about lossy codecs here, though, are we?
  - trevyn 9y ago
    Ah, my mistake. But perhaps you could consider 24/192 -> 16/48 a lossy codec in this context; the same argument does apply, for the same reasons.
    - anyfoo 9y ago
      That way it does make sense, editing might require higher fidelity than we can perceive in the final project to avoid artifacts.
  - kuschku 9y ago
    Actually, the article is not talking about it, but the topic very much is lossy codecs. Specifically, if 24/192 AAC is worth it compared to 16/44.1 AAC (and the answer to that is yes, although the answer to 24/192 WAV is no)
- microcolonel 9y ago
  But what about lossless 16/44.1KHz audio? It's not a "compressed" version of 24/192Khz. Chances are that your 192KHz music will be low pass filtered anyway, 192KHz is a trick that sound chipset vendors added so that integrators wouldn't have to make high quality analog filters for their ADCs. There's likely nothing above 22.05KHz except shaped noise, production artifacts, and very, very quiet ultrasound noises which have nothing to do with the music.
  - TheOtherHobbes 9y ago
    There's no such thing as lossless 16/44.1 because it's practically impossible to make an artefact-free antialiasing/reconstruction filter pair clocked at 44.1k. The point of higher sample rates isn't to capture bat squeaks, but to relax the engineering constraints on pre-post filtering. Nyquist is fine in theory, and if you've never actually tried to implement a clean filter you'll likely watch the xiph video and think "Well that makes sense." If actually know something about practical DSP and the constraints and challenges of real filter design you're not going to be quite so easily impressed. Likewise with higher bit depths. "Common sense" suggests that no one should be able to hear a noise signal at -90dB. Common sense is dead wrong, because the effects of a single bit of dither are absolutely audible. And if you can hear the effects of noise added at -90dB, you can certainly hear the effects of quantisation noise artefacts on reverb tails and long decaying notes at -30 to -40dB, added by recording at 16 bits instead of 24 bits. Whether or not that level of detail is present in a typical pop or classical recording is a different issue. Realistically most music is heavily compressed and limited, so the answer is usually "no." And not all sources have 24-bits of detail [1]. (Recordings made on the typical digital multitrack machines used in the 80s and 90s certainly don't.) That doesn't mean that a clean unprocessed recording of music with a wide dynamic range made on no-compromise equipment won't show the difference clearly. Speaking from experience, it certainly does. [1] Technically no sources have 24-bits of detail. The best you'll get from a real world converter is around 22-bits.
    - microcolonel 9y ago
      > you'll likely watch the xiph video and think "Well that makes sense." What video? This thread is about an article. > The point of higher sample rates isn't to capture bat squeaks, but to relax the engineering constraints on pre-post filtering. I just said that, I think.
tomc1985 9y ago
I think it's because processes that affect things at the individual sample level (which distortion/compression often do) have a lot more detail to work with. The author mentions this a bit in the article
TazeTSchnitzel 9y ago
More bits means: • Less quantisation noise, so your noise floor is a bit lower and therefore, when you're mixing stuff together, you accumulate less noise • More numerical room to play with, you can scale or shift the volume up and down more without clipping and with less loss of accuracy With 16-bit CD audio, you can't just convert to 24-bit and suddenly have less noise. You might get more room, though. As for higher sampling rates (more kHz, if you will), I think Monty mentioned some benefit regarding less sharp aliasing filters (can have a larger transition band from 20–48kHz, say, rather than from 20—22.1kHz), but it's not something I understand the benefit of well.
- dspig 9y ago
  filters with a sharper cutoff will 'ring' more, smearing sharp changes in the signal out over time. It's a pretty subtle effect when it's all happening above 20 kHz though
squeaky-clean 9y ago
For distortion and compression, the only possible difference I can think of is aliasing, and yeah you could just oversample your input, then downsample again on the way out. This is really really common in most professional plugins, often with an adjustable amount to balance quality and CPU usage. I also have some gear that aliases at a volume you can't hear, but when you plug it into an analog distortion pedal, the aliasing in the high frequencies becomes apparent. This would be avoided if it had a higher sample rate so the aliasing was far out of the audible range. For other sorts of effects like spectral processing, pitch shifting, the extra detail above 22khz really does make a difference, especially if pitching down.
davidgerard 9y ago
Because modern production can involve hundreds or even thousands of digital effects. So 24-bit, or even 32-bit float just because you can, is standard. As Monty demonstrates, it's a fraudulent waste to try to sell the result as a product to the end listener.

acchow 9y ago

> Its playback fidelity is slightly inferior to 16/44.1 or 16/48, and it takes up 6 times the space. The article is highly technical. Does anyone have a way to describe this phenomenon intuitively?

mikewhy 9y ago
"Diminishing returns"?
hamandcheese 9y ago
24/192 can capture audio well beyond what we can hear. These sounds can, though, lead to distortion that is within our hearing range, depending on the equipment used. Therefore, rather than just being useless extra data, it can be actively harmful to the listening experience.
- tomc1985 9y ago
  It's not that much different to think of than 480p vs 4K/2160p -- dramatically higher resolution, increasing the width and length of the 'grid' that sound is recorded into -- except that in the case of audio fidelity, quality after a certain point causes problems with playback systems, resulting in audible artifacts and distortion.
  - pzone 9y ago
    That analogy works if you modify it: - 480p vs 2160p is measuring the resolution of your cell phone propped up on a pillow at the other end of your living room - experimental evidence shows that your eyesight is not good enough to pick up on the increased resolution, you've maxed out your sensory perception - Your phone stutters trying to stream at 4k so the playback might actually be worse.
    - tomc1985 9y ago
      All good points! Indeed all that extra data makes the decode stage that much more difficult. Same happens with audio too, but high-res audio isn't nearly as bulky as video or nearly as resource-intensive to decode

sp332 9y ago

Sure, when you're editing music it can be helpful to have super high sample rates and lots of bits to work with. But for listening, it's just a waste of space.

kobeya 9y ago
He was talking about listening.
- sp332 9y ago
  tomc1985 said "fed through distortion/compression/warming effects processors" which is editing, not listening.
  - tomc1985 9y ago
    Yeah, editing
    - ioquatix 9y ago
      Lots of pre-amps can do this in real time, and some installation audio systems do it as part of pre-processing (e.g. compression, limiter, etc).

jotux 9y ago

This is specifically addressed in the link under the section titled "When does 24 bit matter?"

InclinedPlane 9y ago

Well, yeah. For properly mixed audio, you don't need anything beyond 44-khz/16-bit. If you need to do extensive processing, or you're working with raw audio and want the extra margin to be able to recover a sound when it may be too quiet (or too loud) or otherwise compromised then more samples and greater bit-depth can be highly useful.

jchanimal 9y ago

I've heard (and it passes the smell test) that even if your ears can't resonate at frequencies higher than human hearing, the space you are in can, and your ears are sensitive to the differences in arrival time that are smaller than the pitches we can't hear. So at least in high sample rate music the imaging could potentially be cleaner. Assuming super great everything else.

stephen_g 9y ago
What you're sort of describing - having more of the higher frequencies resonate together - can actually cause intermodulation distortion (which itself can be in the audible range), which can make it sound worse... I haven't had time to check the linked article but if it's the one I'm thinking of I'm pretty sure it goes into this. In the end though, double blind testing (again, I think mentioned in the article) shows that there's a threshold (a bit below 48kHz, probably why 44.1kHz was chosen for CD audio) after which people cant distinguish between higher sample rates at any more accuracy than randomly guessing.
- jchanimal 9y ago
  I think it's the intermodalation distortion or as I call it "the room sound" that your non-conscious mind perceives. The best example of this I've heard is when listening to identical masters on SACD/DSD vs PCM audio. Notably different sound when reproduced via loudspeakers.
__david__ 9y ago
That's one of those things that sounds true without putting thought into it but is ultimately false. A 44.1kHz sampled sounds can encode phases with offsets smaller than 22µs (1/44.1k). How much smaller? I'm not exactly sure, but I believe it's dependent on your sample size. My guess is 1/2^16 (assuming sample resolution of 16 bits), but I might be off by a factor of 2 (or indeed 100% wrong) there. Caveat emptor. If I'm right then that's 346 pico-seconds, which I'm betting is smaller than your brain can differentiate.
- deleted 9y ago

dangayle 9y ago

It's also great if you you're doing a bunch of time/pitch mangling.

kazinator 9y ago

In the amplitude domain, you definitely need more bits for that type of processing, and many other types. Digital distortion basically simulates high gain with soft clipping, which basically takes a narrow slice of the amplitude domain and magnifies it. The extra resolution has to be there for that not to turn into ugly sounding shit with artifacts.