28 ms·

I agree with the silliness of 192kHz, but not 24-bits. Here is why: In typical PCM recordings, like CDs, mid-range frequencies (e.g. 1kHz to 4kHz) are recorde

by diydsp 12y ago

I agree with the silliness of 192kHz, but not 24-bits. Here is why: In typical PCM recordings, like CDs, mid-range frequencies (e.g. 1kHz to 4kHz) are recorded with lower amplitudes because our ears are more sensitive to them. Sampling theory is correct and 16-bits can reproduce any waveform with ~100dB of range, however, in a complex waveform consisting of low, mid and high frequencies, the mid- and hi-range frequencies quite simply get shortchanged. Imagine a recording of a bass sinusoid and a mid-range sinusoid of equal volume. It might use e.g. 10 bits to store the bass and only 6 to store the high frequencies. (2^10sin(200wt)+2^6sin(4000wt)). That means the resolution of the high frequencies is less than the lower frequencies. When the volume of those frequencies changes dynamically, the high frequencies' amplitudes are more quantized. That is quite simply why 16-bits are not enough. This is similar to the problem with storing waveforms unprecompensated on vinyl. The precompensation makes up for the non-uniformity of the medium. It could be done with 16-bit digital as well. Or alternatively, larger sample sizes like 24 can be used. I haven't A/B tested this. The A/B test in the article compares CD with SACD. SACD isn't PCM, so its artifacts are going to be totally different from 24-bit PCM.

lukifer 12y ago
There is another reason 24-bit music is desirable: it's good for remix culture (silly IP laws notwithstanding). If I pay for music, I want to truly own it, including the possibility of someday making a mashup, a music video, a hip-hop beat, et cetera. A 24-bit source gives casual creatives the same quality material as the original masters, for a relatively paltry 1.5x increase in file size.
nnnnni 12y ago
There's also the fact that you can FEEL some sounds that you can't hear. Chris Randall of Sister Machine Gun (at least used to?) use a low-frequency generator at live shows to produce a sound that the audience could feel but not hear in order to make the music more intense. I suspect that you'd gain some of that effect with a larger bit size. ...or I could be completely wrong. Whatever.
- deleted 12y ago
  [deleted]
- murki 12y ago
  The entire sub-bass genre is based around that concept. Case in point James Blake https://www.youtube.com/watch?v=oOT2-OTebx0&hd=1 https://www.youtube.com/watch?v=oOT2-OTebx0&hd=1
- badsock 12y ago
  The increase in sample rate to 192kHz only allows frequencies above 22KHz to be represented (i.e. no effect whatsoever on the low frequencies that you mention). Pushing the bits to 24 only lowers the noise levels (which at 16 are already demonstrably imperceptible). What you mention, though, points directly to what /will/ improve the quality of sound reproduction: speakers. It gets harder and harder to move that much air with precision as you get lower and lower in frequency. It's a definite technical limitation, but it's to do with very high-power amps and giant speakers, not the recording format. We have (to the degree that humans can prove that they can perceive), perfect reproduction from digital recordings, perfect amplifiers for reasonable prices (at lower-than-concert-power-levels at least), but we haven't yet developed good enough speakers to cover the whole perceptible range of frequencies to anywhere near the same degree. Audiophiles love to try and improve the whole chain, but really the only place it matters is at the very end.
  - leoc 12y ago
    According to this interesting-looking article here http://www.theregister.co.uk/2014/07/02/feature_the_future_loudspeaker_design/ http://www.theregister.co.uk/2014/07/02/feature_the_future_l... the problem isn't frequency response but things like time delay, and it's not due simply to the very end but also to systems leading up to and around it like inaccurate crossovers and speaker cabinets that introduce time delay.
    - badsock 12y ago
      Agreed - my definition of "the end" being anything past the far end of the speaker cable.
sillysaurus3 12y ago
The correct way to attack this isn't by attacking the theory. It's to gather a lot of people and ask them to press a button indicating whether the audio they hear is 16-bit or 24-bit. If the results are no better than chance, then 24-bit doesn't matter, regardless of how sound the underlying argument is. EDIT: The experiment would also be extremely difficult to design. For example, you'd need to run this test with music, not simple sounds. So the question is, which music? I think whatever is most popular at the time would be a good candidate, because if people are listening to music they hate, they won't care about the fine details of the audio. But that introduces an element of uncertainty and noise into the results which is hard to control for. Some people might deliver accurate results with https://www.youtube.com/watch?v=2zNSgSzhBfM https://www.youtube.com/watch?v=2zNSgSzhBfM but not with https://www.youtube.com/watch?v=4Tr0otuiQuU https://www.youtube.com/watch?v=4Tr0otuiQuU whereas for others it's the opposite. Or, it could be the exact opposite: Maybe you can only detect whether a sound is 24-bit when it's a simple tone, and not music. Age is also a factor. My hearing is worse than a decade ago. The headphones used by the test are another factor. If you feed 24-bit input to headphones, there's no guarantee that the speakers are performing with 24-bit resolution. In fact, this may be the source of most of the confusion in the debate. I'm not sure how you'd even check whether speakers are physically moving back and forth "at 24-bit resolution" rather than a 16-bit resolution.
- xkcd-sucks 12y ago
  If the results are no better than chance, it means the study methodology is flawed OR there's no effect.
  - capnrefsmmat 12y ago
    No, it means the study methodology is flawed OR the effect is too small to be detected with the sample size. So you'd have to decide in advance what difference is meaningful and choose your sample size to ensure you can detect it.
- justatdotin 12y ago
  If the results are no better than chance, then it remains possible that a small subset of the test group actually can appreciate an improvement. Content providers may like to cater for that small subset. disclaimer: I am not in that hypothetical subset.
hamiltonkibbe 12y ago
The resolution of both sinusoids in the example you suggest is exactly 16bits Edit: While it would be possible to represent each of those individual sinusoids in 6 and 10 bits individually, the signal you describe has the high frequency signal "riding on " the low frequency signal. you need 16 bits to represent the amplitude of that signal at e.g. t = pi/4.
- deleted 12y ago
  [deleted]
- TD-Linux 12y ago
  2^10 + 2^6 = 1088. You only need 11 bits to not clip with that signal.
  - hamiltonkibbe 12y ago
    you'd need 11 bits to encode the signal at 0dBFS. Theres amplitude information stored in those other 5 zeros...
xorcist 12y ago
I don't understand what you mean, so I can't say you're wrong, but higher frequencies are not more quantized than low. In fact, you should be very careful about to think about quantization in digital sound reproduction, because it can easily lead you astray. Think of bit depth as a measure of dynamic range. Do look at that digital music primer for geeks posted elsewhere in this thread, it makes for awesome reading.
- hamiltonkibbe 12y ago
  He was referring to the specific example he gave in which the higher-frequency had an amplitude of 2^6 and the lower frequency sinusoid had an amplitude of 2^10
  - RogerL 12y ago
    And that is not how digital sampling works - you don't allocate bits to different frequencies.
    - cozzyd 12y ago
      On the other hand, with perceptual audio coding (e.g. mp3, m4a, ogg) you do.
    - hamiltonkibbe 12y ago
      I don't think anyone is arguing that you do, the high-frequency sinusoid he described has a peak-to-peak amplitude of 2 * UINT6_MAX (although it appears as though the intention was a sinusoid with a p-p amplitude of UINT6_MAX) which on its own (in terms of the integer numbers given in his example) be represented in a 6-bit system. This isn't really relevant though, because that signal would be 0dBFS in a 6-bit system. The higher 10 bits are far from unused by the sinusoid in a 16 bit system, as they allow the amplitude of the sinusoid relative to the other sinusoid to remain unchanged. The little stairstep digitally-sampled sinusoid picture might look a little more rough for the "6-bit" sinusoid than the "10-bit" sinusoid but that's A)kind a pathological case and B)not at all representative of what gets sent to the amplifier after the DAC. (Think about the spectrum of all those little "stairsteps" and what happens to them once they are sent across a shunt capacitor...) anyways
hackinthebochs 12y ago
192kHz may not be as silly as you think: http://www.ncbi.nlm.nih.gov/pubmed/10848570 http://www.ncbi.nlm.nih.gov/pubmed/10848570
- bsder 12y ago
  Except that high frequencies attenuate fairly quickly in air. Sound attenuates proportional to f^2 in air. So, a 10-fold increase in frequency causes a 100-fold increase in attenuation. So, a 100KHz signal has 10,000 times the attenuation for a 1KHz signal over the same range. In addition, attenuation due to water vapor is particularly bad above about 40KHz. Much of that study cannot be replicated, and people have tried: http://en.wikipedia.org/wiki/Hypersonic_effect http://en.wikipedia.org/wiki/Hypersonic_effect In addition, the effect went away when using headphones. However, I can certainly believe that if you can pump enough energy to rattle things at ultrasonic frequencies you are going to get a result. Especially since ultrasonic frequencies rattle in water particularly effectively. As an example, if I pump enough energy into an ultraviolet or infrared signal at your eye I will eventually get a detection result in your brain. However, pain and a burned retina are not what we think about when we consider a brain response.
- badsock 12y ago
  Thanks for posting that! However, it's worth point out that that study has its detractors: http://en.wikipedia.org/wiki/Hypersonic_effect http://en.wikipedia.org/wiki/Hypersonic_effect I find the intermodulation argument a convincing one - it's hard to not have it affect the test, and if they haven't taken specific steps to avoid it then it would be easy to join a large amount of tests that have fallen prey to it. I note, however, that Wikipedia doesn't mention any specifically brain-scan-based studies that counter it. If you know of any more I'd be very interested in hearing about it.
  - retrogradeorbit 12y ago
    What gets me about the intermodulation effect is that some people want to have it both ways. They are so stuck in the 'there is no difference' camp they fail to see the self contradiction in the argument. On one hand, higher frequencies above 20KHz can't be heard at all, so there's no point having them! You can't hear them! Then on the other hand, higher frequencies above 20kHz affect the audible region of the sound (intermodulation distortion), so you can hear them, so make sure they aren't there! What if the presence of the higher frequencies in a spectrum that shares a harmonic relation to the audible region causes intermodulation distortion that is pleasing and musical to the ear. What if the complete absence of this high frequency information, or alternatively a non-harmonic higher frequency signal (say some kind of switching or power supply noise) causes the audible region to be perceived in a less pleasant manner?
badsock 12y ago
If you're playing a 16 bit PCM at a reasonable listening level of 85dB SPL, then your 6 bit sinusoid is at 25dB SPL, which is quieter than a whisper at 6 feet away in a library. The quantization noise floor of a 6 bit recording is a further ~30dB quieter. So, the noise of that signal is -5dB SPL. 0dB SPL was set to be the lowest possible perceivable level of a single sound in an aechoic chamber. And that's not even considering other sounds in the recording, or ambient noise levels in a typical living room, etc. In your example, moving to 24 bit would been a long way from having any effect (other than a 50% increase in file size). And if you use, say, an 8 bit signal as an example, then things are even less noisy. Note that the noise is the only consideration here: any fidelity loss is represented in that figure. The audio engineers of yore who (among other things) decided that 16 bits was more than enough for final mixdown were much more competent than they get credit for (many were downright amazing at what they did, in fact). They thought of stuff like this.
ArkyBeagle 12y ago
Different frequencies aren't stored in different bits. It's all mixed together. If you mix -10dB of 100 Hz and -20db of 1000Hz, the composite will be the same at 8 bits, just much, much noisier. There is, curiously enough, nothing at all like "vertical resolution" in digital audio - resolution in the amplitude dimension. 24 bits is great for recording, but the end distribution medium only needs to be 16 bits, after you normalize the thing you tracked at 24 bits. You can get libsndfile and FFTW and do the tests yourself.

by diydsp 12y ago

lukifer 12y ago

There is another reason 24-bit music is desirable: it's good for remix culture (silly IP laws notwithstanding). If I pay for music, I want to truly own it, including the possibility of someday making a mashup, a music video, a hip-hop beat, et cetera. A 24-bit source gives casual creatives the same quality material as the original masters, for a relatively paltry 1.5x increase in file size.

nnnnni 12y ago

There's also the fact that you can FEEL some sounds that you can't hear. Chris Randall of Sister Machine Gun (at least used to?) use a low-frequency generator at live shows to produce a sound that the audience could feel but not hear in order to make the music more intense. I suspect that you'd gain some of that effect with a larger bit size. ...or I could be completely wrong. Whatever.

deleted 12y ago
[deleted]
murki 12y ago
The entire sub-bass genre is based around that concept. Case in point James Blake https://www.youtube.com/watch?v=oOT2-OTebx0&hd=1 https://www.youtube.com/watch?v=oOT2-OTebx0&hd=1
badsock 12y ago
The increase in sample rate to 192kHz only allows frequencies above 22KHz to be represented (i.e. no effect whatsoever on the low frequencies that you mention). Pushing the bits to 24 only lowers the noise levels (which at 16 are already demonstrably imperceptible). What you mention, though, points directly to what /will/ improve the quality of sound reproduction: speakers. It gets harder and harder to move that much air with precision as you get lower and lower in frequency. It's a definite technical limitation, but it's to do with very high-power amps and giant speakers, not the recording format. We have (to the degree that humans can prove that they can perceive), perfect reproduction from digital recordings, perfect amplifiers for reasonable prices (at lower-than-concert-power-levels at least), but we haven't yet developed good enough speakers to cover the whole perceptible range of frequencies to anywhere near the same degree. Audiophiles love to try and improve the whole chain, but really the only place it matters is at the very end.
- leoc 12y ago
  According to this interesting-looking article here http://www.theregister.co.uk/2014/07/02/feature_the_future_loudspeaker_design/ http://www.theregister.co.uk/2014/07/02/feature_the_future_l... the problem isn't frequency response but things like time delay, and it's not due simply to the very end but also to systems leading up to and around it like inaccurate crossovers and speaker cabinets that introduce time delay.
  - badsock 12y ago
    Agreed - my definition of "the end" being anything past the far end of the speaker cable.

sillysaurus3 12y ago

The correct way to attack this isn't by attacking the theory. It's to gather a lot of people and ask them to press a button indicating whether the audio they hear is 16-bit or 24-bit. If the results are no better than chance, then 24-bit doesn't matter, regardless of how sound the underlying argument is. EDIT: The experiment would also be extremely difficult to design. For example, you'd need to run this test with music, not simple sounds. So the question is, which music? I think whatever is most popular at the time would be a good candidate, because if people are listening to music they hate, they won't care about the fine details of the audio. But that introduces an element of uncertainty and noise into the results which is hard to control for. Some people might deliver accurate results with https://www.youtube.com/watch?v=2zNSgSzhBfM https://www.youtube.com/watch?v=2zNSgSzhBfM but not with https://www.youtube.com/watch?v=4Tr0otuiQuU https://www.youtube.com/watch?v=4Tr0otuiQuU whereas for others it's the opposite. Or, it could be the exact opposite: Maybe you can only detect whether a sound is 24-bit when it's a simple tone, and not music. Age is also a factor. My hearing is worse than a decade ago. The headphones used by the test are another factor. If you feed 24-bit input to headphones, there's no guarantee that the speakers are performing with 24-bit resolution. In fact, this may be the source of most of the confusion in the debate. I'm not sure how you'd even check whether speakers are physically moving back and forth "at 24-bit resolution" rather than a 16-bit resolution.

xkcd-sucks 12y ago
If the results are no better than chance, it means the study methodology is flawed OR there's no effect.
- capnrefsmmat 12y ago
  No, it means the study methodology is flawed OR the effect is too small to be detected with the sample size. So you'd have to decide in advance what difference is meaningful and choose your sample size to ensure you can detect it.
justatdotin 12y ago
If the results are no better than chance, then it remains possible that a small subset of the test group actually can appreciate an improvement. Content providers may like to cater for that small subset. disclaimer: I am not in that hypothetical subset.

hamiltonkibbe 12y ago

The resolution of both sinusoids in the example you suggest is exactly 16bits Edit: While it would be possible to represent each of those individual sinusoids in 6 and 10 bits individually, the signal you describe has the high frequency signal "riding on " the low frequency signal. you need 16 bits to represent the amplitude of that signal at e.g. t = pi/4.

deleted 12y ago
[deleted]
TD-Linux 12y ago
2^10 + 2^6 = 1088. You only need 11 bits to not clip with that signal.
- hamiltonkibbe 12y ago
  you'd need 11 bits to encode the signal at 0dBFS. Theres amplitude information stored in those other 5 zeros...

xorcist 12y ago

I don't understand what you mean, so I can't say you're wrong, but higher frequencies are not more quantized than low. In fact, you should be very careful about to think about quantization in digital sound reproduction, because it can easily lead you astray. Think of bit depth as a measure of dynamic range. Do look at that digital music primer for geeks posted elsewhere in this thread, it makes for awesome reading.

hamiltonkibbe 12y ago
He was referring to the specific example he gave in which the higher-frequency had an amplitude of 2^6 and the lower frequency sinusoid had an amplitude of 2^10
- RogerL 12y ago
  And that is not how digital sampling works - you don't allocate bits to different frequencies.
  - cozzyd 12y ago
    On the other hand, with perceptual audio coding (e.g. mp3, m4a, ogg) you do.
  - hamiltonkibbe 12y ago
    I don't think anyone is arguing that you do, the high-frequency sinusoid he described has a peak-to-peak amplitude of 2 * UINT6_MAX (although it appears as though the intention was a sinusoid with a p-p amplitude of UINT6_MAX) which on its own (in terms of the integer numbers given in his example) be represented in a 6-bit system. This isn't really relevant though, because that signal would be 0dBFS in a 6-bit system. The higher 10 bits are far from unused by the sinusoid in a 16 bit system, as they allow the amplitude of the sinusoid relative to the other sinusoid to remain unchanged. The little stairstep digitally-sampled sinusoid picture might look a little more rough for the "6-bit" sinusoid than the "10-bit" sinusoid but that's A)kind a pathological case and B)not at all representative of what gets sent to the amplifier after the DAC. (Think about the spectrum of all those little "stairsteps" and what happens to them once they are sent across a shunt capacitor...) anyways

hackinthebochs 12y ago

192kHz may not be as silly as you think: http://www.ncbi.nlm.nih.gov/pubmed/10848570 http://www.ncbi.nlm.nih.gov/pubmed/10848570

bsder 12y ago
Except that high frequencies attenuate fairly quickly in air. Sound attenuates proportional to f^2 in air. So, a 10-fold increase in frequency causes a 100-fold increase in attenuation. So, a 100KHz signal has 10,000 times the attenuation for a 1KHz signal over the same range. In addition, attenuation due to water vapor is particularly bad above about 40KHz. Much of that study cannot be replicated, and people have tried: http://en.wikipedia.org/wiki/Hypersonic_effect http://en.wikipedia.org/wiki/Hypersonic_effect In addition, the effect went away when using headphones. However, I can certainly believe that if you can pump enough energy to rattle things at ultrasonic frequencies you are going to get a result. Especially since ultrasonic frequencies rattle in water particularly effectively. As an example, if I pump enough energy into an ultraviolet or infrared signal at your eye I will eventually get a detection result in your brain. However, pain and a burned retina are not what we think about when we consider a brain response.
badsock 12y ago
Thanks for posting that! However, it's worth point out that that study has its detractors: http://en.wikipedia.org/wiki/Hypersonic_effect http://en.wikipedia.org/wiki/Hypersonic_effect I find the intermodulation argument a convincing one - it's hard to not have it affect the test, and if they haven't taken specific steps to avoid it then it would be easy to join a large amount of tests that have fallen prey to it. I note, however, that Wikipedia doesn't mention any specifically brain-scan-based studies that counter it. If you know of any more I'd be very interested in hearing about it.
- retrogradeorbit 12y ago
  What gets me about the intermodulation effect is that some people want to have it both ways. They are so stuck in the 'there is no difference' camp they fail to see the self contradiction in the argument. On one hand, higher frequencies above 20KHz can't be heard at all, so there's no point having them! You can't hear them! Then on the other hand, higher frequencies above 20kHz affect the audible region of the sound (intermodulation distortion), so you can hear them, so make sure they aren't there! What if the presence of the higher frequencies in a spectrum that shares a harmonic relation to the audible region causes intermodulation distortion that is pleasing and musical to the ear. What if the complete absence of this high frequency information, or alternatively a non-harmonic higher frequency signal (say some kind of switching or power supply noise) causes the audible region to be perceived in a less pleasant manner?

badsock 12y ago

If you're playing a 16 bit PCM at a reasonable listening level of 85dB SPL, then your 6 bit sinusoid is at 25dB SPL, which is quieter than a whisper at 6 feet away in a library. The quantization noise floor of a 6 bit recording is a further ~30dB quieter. So, the noise of that signal is -5dB SPL. 0dB SPL was set to be the lowest possible perceivable level of a single sound in an aechoic chamber. And that's not even considering other sounds in the recording, or ambient noise levels in a typical living room, etc. In your example, moving to 24 bit would been a long way from having any effect (other than a 50% increase in file size). And if you use, say, an 8 bit signal as an example, then things are even less noisy. Note that the noise is the only consideration here: any fidelity loss is represented in that figure. The audio engineers of yore who (among other things) decided that 16 bits was more than enough for final mixdown were much more competent than they get credit for (many were downright amazing at what they did, in fact). They thought of stuff like this.

ArkyBeagle 12y ago

Different frequencies aren't stored in different bits. It's all mixed together. If you mix -10dB of 100 Hz and -20db of 1000Hz, the composite will be the same at 8 bits, just much, much noisier. There is, curiously enough, nothing at all like "vertical resolution" in digital audio - resolution in the amplitude dimension. 24 bits is great for recording, but the end distribution medium only needs to be 16 bits, after you normalize the thing you tracked at 24 bits. You can get libsndfile and FFTW and do the tests yourself.