5 ms·

44100hz sample rate is enough to reproduce well above the highest frequency even young humans can hear, and 16 bits of depth is enough for 96db of dynamic range

by jrajav 4y ago

44100hz sample rate is enough to reproduce well above the highest frequency even young humans can hear, and 16 bits of depth is enough for 96db of dynamic range, enough to make the noise floor of any consumer system totally inaudible. Standard CDs might not be good enough for archival and further production work (where you also need wiggle room for further processing or format changes), but they have plenty of headroom to reproduce any kind of sound for playback purposes. Any more is just placebo.

doix 4y ago
So it's been many years since I looked at this stuff, but the reason for higher sample rate isn't to do with humans being able to hear over 20khz. If you sample at 44khz you need a high pass filter at 22khz to avoid aliasing. Since that's pretty close to 20khz, it needs a pretty sharp drop off. Having a filter with a sharp drop off can introduce artifacts. With 96khz sampling rate, you don't need as a sharp drop off in your filter. The above is mostly accurate, hopefully someone else will comment and tell me how I'm wrong and correct the inaccuracies. Edit: to anyone reading this, read the replies if you want the correct explanation. To everyone that replied, thank you :).
- kazinator 4y ago
  You are correct; with a 96 kHz sampling rate, you don't need such a brick wall filter. But then you dohn't have to actually keep the data at 96 kHz; it can be reduced to 44, this time using a digital filter. A sharply cutting off analog filter is expensive to produce. It has multiple stages to create the multiple poles. High precision resistors and capacitors have to be used to get all those circuit stages to line up. The filter will have phase distortion. That's the basis of "supersampling": sampling at a higher rate with a simpler filter with less of a cutoff, then completing the job with a digital filter to get to the target sample rate. This can be done in reverse, in reproducion. Take, say, a 48 KHz signal, and digitally interpolate it to a higher sample rate like 96 KHz. That is fed to the DAC. The filter after the DAC then doesn't need such a steep cutoff after 20 KHz. A greater bit depth can be used; like 16 bit samples interpolated to 24 bit at a higher rate, fed to a 24 bit DAC. The digital filter or interpolator doesn't care about accurate resistors, capacitors or drift in component values over time or due to heat; it does the same thing with the same data every time. Digital filters can look at future values also. The state of an analog filter is determined by only the current and past values of the signal; but digital signal processing can delay the signal a little bit and look at a "box" around the current value. I think that is key to preserving phase relationships.
  - PaulDavisThe1st 4y ago
    > with a 96 kHz sampling rate, you don't need such a brick wall filter. This is incorrect. See my comment above/below.
- TylerE 4y ago
  Yes, i hate when people talk about Nyquist as if its some Panacea. Its vwry “spherical cow”. It assumes all your filters, DACs, are perfect, and that time is infinite.
  - masklinn 4y ago
    The exact same spherical cow issue applies, with worse effects, when you increase your sampling rate unnecessarily. And you can use oversampling to correct for the filtering issues. That's pretty standard in modern DACs and ADCs.
  - PaulDavisThe1st 4y ago
    Nyquist isn't about such assumptions at all. It is an information theoretic value, based on the provable claim that there is zero information present in the original analog waveform that is not represented in the digital version. Yes, your filters, DAC and speaker wire could still create a less than ideal listening situation, but the fundamental aspect of the Nyquist frequency is not concerned with any of that.
    - kazinator 4y ago
      There is zero information present in a waveform that is not contained in the digital version, if that original waveform is confined below the Nyquist limit. Either it is that way already, or else is derived from an original-original waveform that isn't, by low-pass filtering.
      - PaulDavisThe1st 4y ago
        That's an excellent coda/qualification of what I said. Put into a more listener centric context: there's no audible information in a waveform that is not captured by sampling it at 44.1khz.
  - Dylan16807 4y ago
    You only need so much margin, though.
- masklinn 4y ago
  Anything above 44 is problematic because lots of systems have non-linear responses and will create audible intermodulations of the ultraharmonics. Monty (of xiph) has an article on the subject (well on 24/192 but it applies all the same) which goes into the gory details: https://people.xiph.org/~xiphmont/demo/neil-young.html https://people.xiph.org/~xiphmont/demo/neil-young.html
  - pnut 4y ago
    Interesting video on sample rate effects during digital music production https://youtu.be/-jCwIsT0X8M https://youtu.be/-jCwIsT0X8M
- ajross 4y ago
  That's the "further processing" notion the grandparent comment talked about. A correctly processed signal at 44.1kHz is more than sufficient to reproduce all audio. But if you're going to do something with the sample series (like low-pass it for aliasing protection in your example, or to resample it for conversion, or for any effects or mixing step really) it's good to have extra data in there to prevent headaches. Likewise sampling at 24+ bits prevents accumulated error in repeated processing, etc... There's a space in the audio world for more bits of data. But the final output format isn't where it belongs.
- NovemberWhiskey 4y ago
  That's fine - but, to the extent that's true, it's true only about the recording and mastering process. CD, as a delivery mechanism, is totally fine and perceptually flawless.
- PaulDavisThe1st 4y ago
  [ EDIT: contents of this post were factually incorrect. What Kazinator wrote in response is correct, and I don't want to leave incorrect ideas floating around. ]
  - mturmon 4y ago
    This explanation loses me at "infinity per dB" ... and I'm an EE. I think you're trying to cover too much ground here, it's really confusing to try to understand what you mean. I believe the comment you're responding to is talking about the analog filter that is needed to avoid aliasing -- as the first words of your comment correctly note/explain. And in particular, the original comment seems to be noting the phase distortion (in frequencies near the cutoff) that analog brick-wall filters will cause. This has been a design contention for decades, really, ever since the CD format was introduced. It's a big design space, with options for gentler analog filters, followed by very fast digital sampling, and further tricks with filtering in the digital domain, where you don't have to worry about getting great capacitors, etc. It may be out-of-scope to lay all that out in one paragraph!
    - PaulDavisThe1st 4y ago
      That's why I had some blank lines in there :) You're right that its a big design space. The key takeaway is that "yes, higher sample rates can actually make a difference, but almost entirely down to the filter design, not because Nyquist moves ... and you probably cannot hear the difference."
  - kazinator 4y ago
    If the goal is audio capture and reproduction, the filter used when sampling at 96 kHz still starts rolling off at around 20 kHz. The wrap-around aliasing artifacts do not begin until around half of 96 kHz, or around 48 kHz. So the filter only has to be steep enough to hit a large attenuation at 48 kHz: full signal to virtually nothing over the space of 28 kHz. If you're sampling at 48 kHz, the signal has to be severely attenuated already; it has to go from 20 kHz to deep cutoff in just the space of a few kHz. At 96 kHz can achieve the effect as if you were sampling at 48 kHz, with a steep filter. You sample at 96 kHz with a milder filter, and then purely in the digital realm, you down-sample to 48 kHz. There is an overall filter consisting of the original analog one plus the digital processing. That is cheaper and more reliable than doing it all in analog. An analog filter with a steep cut off will be challenging in mass production because of the strict component tolerances. Sure, you could use a steep filter with 96 kHz also. Say, a steep filter that starts cutting off at 30 kHz. It would still be a less demanding filtering application because of the margin that you have in the frequency domain. The multiple poles of the filter don't have to be lined up as well. E.g. if the first pole starts rolling off at around 30 kHz, and then next ones at 31, and the third one at 28, ... it doesn't matter because you're still hitting the absolute target of there being next to nothing at 48 kHz, and nearly the full signal at 20 kHz.
kazinator 4y ago
Not only that, but what you hear at those upper frequencies isn't tone anyway. the highest harmonics of tonal sounds live in that region, plus aperiodic signals (the hiss from sibilants, cymbals crashing, and so on). The content can be faked, and this is used in low-bandwidth codecs and such. https://en.wikipedia.org/wiki/Bandwidth_extension https://en.wikipedia.org/wiki/Bandwidth_extension
bayindirh 4y ago
> Any more is just placebo. This is a very simplistic view of things, on many fronts, but I'll try to be as concise as possible. In a classical wind/bow orchestra, there are more harmonics at play than the 20Hz-20KHz band, and these harmonics affects our perception of the sound and soundstage. However, recreation of this is very hard, because you need both recording and playback chains which can handle these harmonics as well. To faithfully reproduce such orchestra, you need a speaker for every instrument, ideally with the exact air movement capacity of the instrument you're mirroring. This is not practical. Instead we mic all of them, mix all of them, and add room echo to the mix to capture harmonics as best as we can. However, this can't replicate some instruments anywhere around its real sound. Examples are Turkish qanun [0] and Chinese guzheng [1]. These instruments sound bland, flat and shallow on all recordings, but listening them live, directly with your ears is a goose pimlples inducing experience. Higher end stringed instruments have a similar vibe to them. So, a 16bit/44KHz signal at 20Hz-20KHz band cannot reproduce these instruments with any faith. Standard CDs, high quality vinyls, lossless files carry a lot of information, but not all information we can process with our ears. Like how even the best digital cameras cannot reproduce the colors we can see with our eyes. Reaching these resolution levels are neither cheap, nor practical, hence we use what's practical. As a result, your master-pressed-vinyl possibly can't carry this information either, because the recording chain was not able to capture that amount of information, even if you went all the way to install your own power pole to feed clean power to your impossibly expensive audio equipment with all that capacity to reproduce that sound. IOW, you can't extract the sound which is not there to begin with. Your friendly ex-orchestra player reported from its AKAI-AM2850. [0]: https://en.wikipedia.org/wiki/Qanun_(instrument) https://en.wikipedia.org/wiki/Qanun_(instrument) [1]: https://en.wikipedia.org/wiki/Guzheng https://en.wikipedia.org/wiki/Guzheng
- tzs 4y ago
  > To faithfully reproduce such orchestra [...] It has never been clear to me what the goal of home playback is supposed to be. Let's simplify from a full orchestra to just a solo piano. Is the goal 1. to sound like it would if that piano was being played in my living room, or 2. to sound like what I'd have heard if I was there in the concert hall sitting in a good seat when the recording was made, or 3. to sound like what it would sound like if a replica of my living room was built inside the concert hall but with walls that do not transmit sound, with my speakers replaced by speaker-sized holes in the wall behind where the speakers normally sit, and I was sitting in that replica living room during the concert?
tzs 4y ago
> 44100hz sample rate is enough to reproduce well above the highest frequency even young humans can hear It's enough to reproduce well above what we can hear by the normal mechanism that we use for most of our hearing (sound vibrates the eardrum, which vibrates some tiny bones in the middle ear, which cause fluid in the cochlea to ripple which disturb sensory hairs which we perceive as sound). Ultrasonic sounds can be conducted to the cochlea via the surrounding bone bypassing the eardrum and middle ear. Those bone conducted ultrasonics can be perceived. See [1]. This is probably not important in music except in maybe a few rare cases. I'd still like to see better frequency handling though, so that our music wouldn't sound terrible to the dogs and cats that are forced to listen with their owners. We can't normally hear much past 20 KHz (more like 12 KHz for many of us), but they can. There are several musical instruments that have significant ultrasonic output. It would be an interesting experiment to see if dogs and cats like those instruments live better than they do recorded (or dislike live less than they dislike recorded). Then compare to the same experiment but with instruments that don't have ultrasonic output. [1] https://pubmed.ncbi.nlm.nih.gov/23384569/ https://pubmed.ncbi.nlm.nih.gov/23384569/

by jrajav 4y ago

doix 4y ago

So it's been many years since I looked at this stuff, but the reason for higher sample rate isn't to do with humans being able to hear over 20khz. If you sample at 44khz you need a high pass filter at 22khz to avoid aliasing. Since that's pretty close to 20khz, it needs a pretty sharp drop off. Having a filter with a sharp drop off can introduce artifacts. With 96khz sampling rate, you don't need as a sharp drop off in your filter. The above is mostly accurate, hopefully someone else will comment and tell me how I'm wrong and correct the inaccuracies. Edit: to anyone reading this, read the replies if you want the correct explanation. To everyone that replied, thank you :).

kazinator 4y ago
You are correct; with a 96 kHz sampling rate, you don't need such a brick wall filter. But then you dohn't have to actually keep the data at 96 kHz; it can be reduced to 44, this time using a digital filter. A sharply cutting off analog filter is expensive to produce. It has multiple stages to create the multiple poles. High precision resistors and capacitors have to be used to get all those circuit stages to line up. The filter will have phase distortion. That's the basis of "supersampling": sampling at a higher rate with a simpler filter with less of a cutoff, then completing the job with a digital filter to get to the target sample rate. This can be done in reverse, in reproducion. Take, say, a 48 KHz signal, and digitally interpolate it to a higher sample rate like 96 KHz. That is fed to the DAC. The filter after the DAC then doesn't need such a steep cutoff after 20 KHz. A greater bit depth can be used; like 16 bit samples interpolated to 24 bit at a higher rate, fed to a 24 bit DAC. The digital filter or interpolator doesn't care about accurate resistors, capacitors or drift in component values over time or due to heat; it does the same thing with the same data every time. Digital filters can look at future values also. The state of an analog filter is determined by only the current and past values of the signal; but digital signal processing can delay the signal a little bit and look at a "box" around the current value. I think that is key to preserving phase relationships.
- PaulDavisThe1st 4y ago
  > with a 96 kHz sampling rate, you don't need such a brick wall filter. This is incorrect. See my comment above/below.
TylerE 4y ago
Yes, i hate when people talk about Nyquist as if its some Panacea. Its vwry “spherical cow”. It assumes all your filters, DACs, are perfect, and that time is infinite.
- masklinn 4y ago
  The exact same spherical cow issue applies, with worse effects, when you increase your sampling rate unnecessarily. And you can use oversampling to correct for the filtering issues. That's pretty standard in modern DACs and ADCs.
- PaulDavisThe1st 4y ago
  Nyquist isn't about such assumptions at all. It is an information theoretic value, based on the provable claim that there is zero information present in the original analog waveform that is not represented in the digital version. Yes, your filters, DAC and speaker wire could still create a less than ideal listening situation, but the fundamental aspect of the Nyquist frequency is not concerned with any of that.
  - kazinator 4y ago
    There is zero information present in a waveform that is not contained in the digital version, if that original waveform is confined below the Nyquist limit. Either it is that way already, or else is derived from an original-original waveform that isn't, by low-pass filtering.
    - PaulDavisThe1st 4y ago
      That's an excellent coda/qualification of what I said. Put into a more listener centric context: there's no audible information in a waveform that is not captured by sampling it at 44.1khz.
- Dylan16807 4y ago
  You only need so much margin, though.
masklinn 4y ago
Anything above 44 is problematic because lots of systems have non-linear responses and will create audible intermodulations of the ultraharmonics. Monty (of xiph) has an article on the subject (well on 24/192 but it applies all the same) which goes into the gory details: https://people.xiph.org/~xiphmont/demo/neil-young.html https://people.xiph.org/~xiphmont/demo/neil-young.html
- pnut 4y ago
  Interesting video on sample rate effects during digital music production https://youtu.be/-jCwIsT0X8M https://youtu.be/-jCwIsT0X8M
ajross 4y ago
That's the "further processing" notion the grandparent comment talked about. A correctly processed signal at 44.1kHz is more than sufficient to reproduce all audio. But if you're going to do something with the sample series (like low-pass it for aliasing protection in your example, or to resample it for conversion, or for any effects or mixing step really) it's good to have extra data in there to prevent headaches. Likewise sampling at 24+ bits prevents accumulated error in repeated processing, etc... There's a space in the audio world for more bits of data. But the final output format isn't where it belongs.
NovemberWhiskey 4y ago
That's fine - but, to the extent that's true, it's true only about the recording and mastering process. CD, as a delivery mechanism, is totally fine and perceptually flawless.
PaulDavisThe1st 4y ago
[ EDIT: contents of this post were factually incorrect. What Kazinator wrote in response is correct, and I don't want to leave incorrect ideas floating around. ]
- mturmon 4y ago
  This explanation loses me at "infinity per dB" ... and I'm an EE. I think you're trying to cover too much ground here, it's really confusing to try to understand what you mean. I believe the comment you're responding to is talking about the analog filter that is needed to avoid aliasing -- as the first words of your comment correctly note/explain. And in particular, the original comment seems to be noting the phase distortion (in frequencies near the cutoff) that analog brick-wall filters will cause. This has been a design contention for decades, really, ever since the CD format was introduced. It's a big design space, with options for gentler analog filters, followed by very fast digital sampling, and further tricks with filtering in the digital domain, where you don't have to worry about getting great capacitors, etc. It may be out-of-scope to lay all that out in one paragraph!
  - PaulDavisThe1st 4y ago
    That's why I had some blank lines in there :) You're right that its a big design space. The key takeaway is that "yes, higher sample rates can actually make a difference, but almost entirely down to the filter design, not because Nyquist moves ... and you probably cannot hear the difference."
- kazinator 4y ago
  If the goal is audio capture and reproduction, the filter used when sampling at 96 kHz still starts rolling off at around 20 kHz. The wrap-around aliasing artifacts do not begin until around half of 96 kHz, or around 48 kHz. So the filter only has to be steep enough to hit a large attenuation at 48 kHz: full signal to virtually nothing over the space of 28 kHz. If you're sampling at 48 kHz, the signal has to be severely attenuated already; it has to go from 20 kHz to deep cutoff in just the space of a few kHz. At 96 kHz can achieve the effect as if you were sampling at 48 kHz, with a steep filter. You sample at 96 kHz with a milder filter, and then purely in the digital realm, you down-sample to 48 kHz. There is an overall filter consisting of the original analog one plus the digital processing. That is cheaper and more reliable than doing it all in analog. An analog filter with a steep cut off will be challenging in mass production because of the strict component tolerances. Sure, you could use a steep filter with 96 kHz also. Say, a steep filter that starts cutting off at 30 kHz. It would still be a less demanding filtering application because of the margin that you have in the frequency domain. The multiple poles of the filter don't have to be lined up as well. E.g. if the first pole starts rolling off at around 30 kHz, and then next ones at 31, and the third one at 28, ... it doesn't matter because you're still hitting the absolute target of there being next to nothing at 48 kHz, and nearly the full signal at 20 kHz.

kazinator 4y ago

Not only that, but what you hear at those upper frequencies isn't tone anyway. the highest harmonics of tonal sounds live in that region, plus aperiodic signals (the hiss from sibilants, cymbals crashing, and so on). The content can be faked, and this is used in low-bandwidth codecs and such. https://en.wikipedia.org/wiki/Bandwidth_extension https://en.wikipedia.org/wiki/Bandwidth_extension

bayindirh 4y ago

> Any more is just placebo. This is a very simplistic view of things, on many fronts, but I'll try to be as concise as possible. In a classical wind/bow orchestra, there are more harmonics at play than the 20Hz-20KHz band, and these harmonics affects our perception of the sound and soundstage. However, recreation of this is very hard, because you need both recording and playback chains which can handle these harmonics as well. To faithfully reproduce such orchestra, you need a speaker for every instrument, ideally with the exact air movement capacity of the instrument you're mirroring. This is not practical. Instead we mic all of them, mix all of them, and add room echo to the mix to capture harmonics as best as we can. However, this can't replicate some instruments anywhere around its real sound. Examples are Turkish qanun [0] and Chinese guzheng [1]. These instruments sound bland, flat and shallow on all recordings, but listening them live, directly with your ears is a goose pimlples inducing experience. Higher end stringed instruments have a similar vibe to them. So, a 16bit/44KHz signal at 20Hz-20KHz band cannot reproduce these instruments with any faith. Standard CDs, high quality vinyls, lossless files carry a lot of information, but not all information we can process with our ears. Like how even the best digital cameras cannot reproduce the colors we can see with our eyes. Reaching these resolution levels are neither cheap, nor practical, hence we use what's practical. As a result, your master-pressed-vinyl possibly can't carry this information either, because the recording chain was not able to capture that amount of information, even if you went all the way to install your own power pole to feed clean power to your impossibly expensive audio equipment with all that capacity to reproduce that sound. IOW, you can't extract the sound which is not there to begin with. Your friendly ex-orchestra player reported from its AKAI-AM2850. [0]: https://en.wikipedia.org/wiki/Qanun_(instrument) https://en.wikipedia.org/wiki/Qanun_(instrument) [1]: https://en.wikipedia.org/wiki/Guzheng https://en.wikipedia.org/wiki/Guzheng

tzs 4y ago
> To faithfully reproduce such orchestra [...] It has never been clear to me what the goal of home playback is supposed to be. Let's simplify from a full orchestra to just a solo piano. Is the goal 1. to sound like it would if that piano was being played in my living room, or 2. to sound like what I'd have heard if I was there in the concert hall sitting in a good seat when the recording was made, or 3. to sound like what it would sound like if a replica of my living room was built inside the concert hall but with walls that do not transmit sound, with my speakers replaced by speaker-sized holes in the wall behind where the speakers normally sit, and I was sitting in that replica living room during the concert?

tzs 4y ago

> 44100hz sample rate is enough to reproduce well above the highest frequency even young humans can hear It's enough to reproduce well above what we can hear by the normal mechanism that we use for most of our hearing (sound vibrates the eardrum, which vibrates some tiny bones in the middle ear, which cause fluid in the cochlea to ripple which disturb sensory hairs which we perceive as sound). Ultrasonic sounds can be conducted to the cochlea via the surrounding bone bypassing the eardrum and middle ear. Those bone conducted ultrasonics can be perceived. See [1]. This is probably not important in music except in maybe a few rare cases. I'd still like to see better frequency handling though, so that our music wouldn't sound terrible to the dogs and cats that are forced to listen with their owners. We can't normally hear much past 20 KHz (more like 12 KHz for many of us), but they can. There are several musical instruments that have significant ultrasonic output. It would be an interesting experiment to see if dogs and cats like those instruments live better than they do recorded (or dislike live less than they dislike recorded). Then compare to the same experiment but with instruments that don't have ultrasonic output. [1] https://pubmed.ncbi.nlm.nih.gov/23384569/ https://pubmed.ncbi.nlm.nih.gov/23384569/