7 ms·

Are you, per chance, a dog posting on the internet? Since 44.1khz sample rate is already past the range of the human ear, regardless of training.

clawlor 3mo ago
Max representable frequency is half the sampling rate (nyquist-shannon theorem), which is still a bit above normal but IIRC the extra headroom has something to do with eliminating aliasing
- Blackthorn 3mo ago
  Indeed. And what is the max frequency that a human can hear?
  - Rotundo 3mo ago
    Depends on age of the listener, on average, 30 to 50 year olds hear a maximum frequency of 14 to 16 kHz.
    - Blackthorn 3mo ago
      Right. Which are quite below 1/2 of 44.1k!
      - OkayPhysicist 3mo ago
        Sure, but those are averages. I'm 30-ish, and my hearing doesn't cut out until somewhere in the 21kHz range. When I was younger, it was even higher. One of my roommates in college had one of those anti-rodent high-frequency noise generators, we almost came to blows over it.
  - speak_on 3mo ago
    The artifacts produced by pure 44.1 kHz convertion are aliased back down to lower frequencies. It's not about a theoretical human ear, it's about the actual physics of AD/DA conversion.
    - PaulDavisThe1st 3mo ago
      But the energies of the signal present above the Nyquist frequency (22050Hz in this case) are almost always incredibly weak, and double blind testing rarely shows any indication that humans can actually hear the aliasing.
      - speak_on 3mo ago
        Mixing process often involves hundreds of tracks, and if each introduces aliasing, this can become a problem. Some engineers do swear by "the final mix is 16/44.1 so why mix at a different resolution?" mantra - that's fine too.
        PaulDavisThe1st 3mo ago
        This is false. Aliasing is not additive in any meaningful way.
        speak_on 3mo ago
        Ok dude, you obviously never recorded anything. Twelve mics on a drum kit, 60 tracks of rhythm guitars, several bass guitar layers, vocals, backing vocals, electric organ, percussions, saxophone solo. Do you think recording them at 44.1 somehow creates a shared "cloud-based" aliasing artifact that I store in S3?
        PaulDavisThe1st 3mo ago
        > Ok dude, you obviously never recorded anything. https://ardour.org/ https://ardour.org/ is my website.
        speak_on 3mo ago
        Ha! OK I take that back. Firstly, it's an amazing experience to randomly interact with people like you - I love and use your software. Hats off and thanks for what you offered to the industry! But secondly, your statement makes even less sense to me: obviously artifacts do add up. Yes, not linearly, like any complex audio in general. But the more tracks with artifacts I have, the more artifacts I have overall. It's not like they cancel each other (outside of normal frequency cancellation).
move-on-by 3mo ago
I don’t have great hearing, so I’m not sure I can really weigh in here (thanks punk concerts in my teens). I remember similar arguments around screens and 60Hz vs ‘the human eye’. I think a lot of people, myself included, can easily perceive the difference between 60Hz and something higher- given the right conditions. I would not be so quick to disregard claims of more sensitive hearing.
- labcomputer 3mo ago
  I would. It’s really simple. The human threshold-of-hearing curve intersects the threshold-of-pain curve at about 20 kHz. Above that frequency (or thereabouts) the sound has to be so loud that it will literally instantly damage your hearing before you can hear it. This has been replicated across many studies for more than 100 years. Flicker threshold is completely different. You can’t damage your vision by increasing the FPS, and it has always been commercially desirable to use a lower frequency because that is cheaper.
  - speak_on 3mo ago
    Would you agree that a trained human could identify artifacts produced by an imperect conversion process? If you lean "yes", then that's your answer: AD/DA is not a Rust function perfectly implementing the Nyquist theorem, it's a collection of physical components many of which introduce artifacts into the audio path. This thread is not about the theory of human hearing, the electronic components are literally imperfect.
    - PaulDavisThe1st 3mo ago
      They're no more imperfect than the pickups on an electric guitar, the assembly inside the microphone, the circuit in the compressor and everything else in the analog signal chain that exists long before AD happens.
      - speak_on 3mo ago
        Absolutely! All these examples have imperfect audio paths - that is the point.
MertsA 3mo ago
You need at least twice the frequency range for sample rate in order to represent the original signal. That's slightly misleading though, that's from the Nyquist-Shannon sampling theory and it's a mathematical fact but that is true for exact numerical samples, once you add in quantization that muddies the water a bit. Taken at the extreme, it's straightforward to see why a 1 bit quantization per sample at 44.1 kHz would not capture a perfect representation of some analog signal even if there's only a 1 kHz frequency component to the signal. If we instead decide to sample at 10 MHz but still one bit quantization, now that 1 kHz frequency component can be much more accurately represented even though we're still using the worst quantization possible. Don't think of quantization like a square wave or a step pattern, think of it as "the signal is closer to here than any other discrete value". Now in terms of realistic audio encoding, 16 bit at 44.1 kHz is designed to be a faithful representation as far as human hearing is concerned. Can someone with a trained ear potentially tell the difference between that and 24 bit at 192 kHz? In a studio environment it's possible. Most audiophile claims are dubious and a blind A/B test catches them out on most of it but the Nyquist-Shannon sampling theorem does not directly apply to quantized samples, it's about exact samples and with quantization, sampling rate is intertwined somewhat with the quantization depth.
speak_on 3mo ago
As I responded below, you are confusing math with physical reality. A true 44.1 kHz converter can't realistically capture frequencies ~18-20 kHz due to the limitations of filters used in the process. A perfect lowpass brick-wall filter just does not exist - they all introduce artifacts, which a trained ear can identify. You don't need to be a dog to hear the difference, just someone who does not assume that Nyquist theorem can be magically applied in the real world (and, ideally, someone who utilizes high quality converters with oversampling).
- vor_ 3mo ago
  Is there evidence that a trained ear can reliably perceive these artifacts in a blind test of converters? I'd be interested in reading those links since converters typically oversample into the mHz range. At 11.29 mHz (256x 44.1 mHz), Nyquist will be at 5.64 mHz. Even the cheapest consumer converters are performing this type of oversampling.
  - speak_on 3mo ago
    If you are looking for studies, this one comes to mind: https://www.researchgate.net/publication/289039184_The_audibility_of_typical_digital_audio_filters_in_a_high-fidelity_playback_system https://www.researchgate.net/publication/289039184_The_audib... A quick search returned this PDF with a nice diagram of what aliasing looks like: https://download.tek.com/document/76W_30631_0_HR_Letter.pdf https://download.tek.com/document/76W_30631_0_HR_Letter.pdf To draw a design parallel: pixel-perfect design isn't something we are born with, noticing tiny details is a developed skill. And yes, you are on point: oversampling is used extensively, but this just points at the exact issue: Nyquist theorem gave us a math algorithm, we still need to account for the electronic component imperfections. And then we are entering a different space of quality/precision/psychoacoustics/perception/etc. Meaning, not all converters, not all pre-amps, not all mics "sound" the same, even when they use same types of components on paper.
    - vor_ 3mo ago
      Oh, dear, that AES 2014 paper from Meridian (which was trying to push its controversial proprietary MQA audiophile system the same year) was widely criticized on audio forums when it came out, ranging from the rectangular dithering method to the use of a hard metal tweeter that could cause IM. Do you have more convincing sources?

clawlor 3mo ago

Max representable frequency is half the sampling rate (nyquist-shannon theorem), which is still a bit above normal but IIRC the extra headroom has something to do with eliminating aliasing

Blackthorn 3mo ago
Indeed. And what is the max frequency that a human can hear?
- Rotundo 3mo ago
  Depends on age of the listener, on average, 30 to 50 year olds hear a maximum frequency of 14 to 16 kHz.
  - Blackthorn 3mo ago
    Right. Which are quite below 1/2 of 44.1k!
    - OkayPhysicist 3mo ago
      Sure, but those are averages. I'm 30-ish, and my hearing doesn't cut out until somewhere in the 21kHz range. When I was younger, it was even higher. One of my roommates in college had one of those anti-rodent high-frequency noise generators, we almost came to blows over it.
- speak_on 3mo ago
  The artifacts produced by pure 44.1 kHz convertion are aliased back down to lower frequencies. It's not about a theoretical human ear, it's about the actual physics of AD/DA conversion.
  - PaulDavisThe1st 3mo ago
    But the energies of the signal present above the Nyquist frequency (22050Hz in this case) are almost always incredibly weak, and double blind testing rarely shows any indication that humans can actually hear the aliasing.
    - speak_on 3mo ago
      Mixing process often involves hundreds of tracks, and if each introduces aliasing, this can become a problem. Some engineers do swear by "the final mix is 16/44.1 so why mix at a different resolution?" mantra - that's fine too.
      - PaulDavisThe1st 3mo ago
        This is false. Aliasing is not additive in any meaningful way.
        speak_on 3mo ago
        Ok dude, you obviously never recorded anything. Twelve mics on a drum kit, 60 tracks of rhythm guitars, several bass guitar layers, vocals, backing vocals, electric organ, percussions, saxophone solo. Do you think recording them at 44.1 somehow creates a shared "cloud-based" aliasing artifact that I store in S3?
        PaulDavisThe1st 3mo ago
        > Ok dude, you obviously never recorded anything. https://ardour.org/ https://ardour.org/ is my website.
        speak_on 3mo ago
        Ha! OK I take that back. Firstly, it's an amazing experience to randomly interact with people like you - I love and use your software. Hats off and thanks for what you offered to the industry! But secondly, your statement makes even less sense to me: obviously artifacts do add up. Yes, not linearly, like any complex audio in general. But the more tracks with artifacts I have, the more artifacts I have overall. It's not like they cancel each other (outside of normal frequency cancellation).

move-on-by 3mo ago

I don’t have great hearing, so I’m not sure I can really weigh in here (thanks punk concerts in my teens). I remember similar arguments around screens and 60Hz vs ‘the human eye’. I think a lot of people, myself included, can easily perceive the difference between 60Hz and something higher- given the right conditions. I would not be so quick to disregard claims of more sensitive hearing.

labcomputer 3mo ago
I would. It’s really simple. The human threshold-of-hearing curve intersects the threshold-of-pain curve at about 20 kHz. Above that frequency (or thereabouts) the sound has to be so loud that it will literally instantly damage your hearing before you can hear it. This has been replicated across many studies for more than 100 years. Flicker threshold is completely different. You can’t damage your vision by increasing the FPS, and it has always been commercially desirable to use a lower frequency because that is cheaper.
- speak_on 3mo ago
  Would you agree that a trained human could identify artifacts produced by an imperect conversion process? If you lean "yes", then that's your answer: AD/DA is not a Rust function perfectly implementing the Nyquist theorem, it's a collection of physical components many of which introduce artifacts into the audio path. This thread is not about the theory of human hearing, the electronic components are literally imperfect.
  - PaulDavisThe1st 3mo ago
    They're no more imperfect than the pickups on an electric guitar, the assembly inside the microphone, the circuit in the compressor and everything else in the analog signal chain that exists long before AD happens.
    - speak_on 3mo ago
      Absolutely! All these examples have imperfect audio paths - that is the point.

MertsA 3mo ago

You need at least twice the frequency range for sample rate in order to represent the original signal. That's slightly misleading though, that's from the Nyquist-Shannon sampling theory and it's a mathematical fact but that is true for exact numerical samples, once you add in quantization that muddies the water a bit. Taken at the extreme, it's straightforward to see why a 1 bit quantization per sample at 44.1 kHz would not capture a perfect representation of some analog signal even if there's only a 1 kHz frequency component to the signal. If we instead decide to sample at 10 MHz but still one bit quantization, now that 1 kHz frequency component can be much more accurately represented even though we're still using the worst quantization possible. Don't think of quantization like a square wave or a step pattern, think of it as "the signal is closer to here than any other discrete value". Now in terms of realistic audio encoding, 16 bit at 44.1 kHz is designed to be a faithful representation as far as human hearing is concerned. Can someone with a trained ear potentially tell the difference between that and 24 bit at 192 kHz? In a studio environment it's possible. Most audiophile claims are dubious and a blind A/B test catches them out on most of it but the Nyquist-Shannon sampling theorem does not directly apply to quantized samples, it's about exact samples and with quantization, sampling rate is intertwined somewhat with the quantization depth.

speak_on 3mo ago

As I responded below, you are confusing math with physical reality. A true 44.1 kHz converter can't realistically capture frequencies ~18-20 kHz due to the limitations of filters used in the process. A perfect lowpass brick-wall filter just does not exist - they all introduce artifacts, which a trained ear can identify. You don't need to be a dog to hear the difference, just someone who does not assume that Nyquist theorem can be magically applied in the real world (and, ideally, someone who utilizes high quality converters with oversampling).

vor_ 3mo ago
Is there evidence that a trained ear can reliably perceive these artifacts in a blind test of converters? I'd be interested in reading those links since converters typically oversample into the mHz range. At 11.29 mHz (256x 44.1 mHz), Nyquist will be at 5.64 mHz. Even the cheapest consumer converters are performing this type of oversampling.
- speak_on 3mo ago
  If you are looking for studies, this one comes to mind: https://www.researchgate.net/publication/289039184_The_audibility_of_typical_digital_audio_filters_in_a_high-fidelity_playback_system https://www.researchgate.net/publication/289039184_The_audib... A quick search returned this PDF with a nice diagram of what aliasing looks like: https://download.tek.com/document/76W_30631_0_HR_Letter.pdf https://download.tek.com/document/76W_30631_0_HR_Letter.pdf To draw a design parallel: pixel-perfect design isn't something we are born with, noticing tiny details is a developed skill. And yes, you are on point: oversampling is used extensively, but this just points at the exact issue: Nyquist theorem gave us a math algorithm, we still need to account for the electronic component imperfections. And then we are entering a different space of quality/precision/psychoacoustics/perception/etc. Meaning, not all converters, not all pre-amps, not all mics "sound" the same, even when they use same types of components on paper.
  - vor_ 3mo ago
    Oh, dear, that AES 2014 paper from Meridian (which was trying to push its controversial proprietary MQA audiophile system the same year) was widely criticized on audio forums when it came out, ranging from the rectangular dithering method to the use of a hard metal tweeter that could cause IM. Do you have more convincing sources?