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when you say (paraphrasal quotes) "the inaudible will be preserved through lossy compression designed around knowledge of human hearing" you're essentially sayi

when you say (paraphrasal quotes) "the inaudible will be preserved through lossy compression designed around knowledge of human hearing" you're essentially saying that "undefined behavior will be preserved through optimization", and that is a boldly optimistic claim.

ChuckMcM 3y ago
I think you and I may be talking about different things. If you know what phase noise is and understand the term -10 dBc @ 1kHz, then we might be on the same page. If you don't consider the following; There are tones, things you can hear typically in the range of 15 Hz to 22 kHz. When those tones vary in frequency over time by a few Hertz, it is unlikely that you can perceive that variation. Your ear is both logarithmic in its sensitivity to volume and not particularly sensitive to absolute frequency. If you shift an entire spectrum "up" a couple of Hz your ear won't notice, and similarly if you shift it "down" a couple of Hz you won't notice. This is especially true if you do it slowly (over the course of 100 milliseconds or so). Tape decks and turntables do this, they try to minimize it though. The specification that tells you how closely they track rotational speed is "Wow and Flutter" and a good number is < 0.3%, a studio recorder might be less than 0.1%. With DSP you can easily pull out a spectrum shift of < 0.05% if it is 'regular'. That shift can be a "frequency shift keying" (FSK) signal with forward error correction in it. (think shifted 'up' as one (1) bits and shifted 'down' as zero (0) bits). Running the clip through an FFT and monitoring phase shift in the bins would recover this string of bits. And even if you didn't get enough to error correct the original message, their presence would be unmistakable. All existing audio compression schemes do not affect the phase relationship of the spectrum they are compressing. That's a design feature. To be fair, I wouldn't understand any of this if I hadn't been delving into software defined radio and learning DSP techniques for modulation and demodulation with data recovery in the presence of signal distortions and interference. And I completely understand that there is a language challenge when the terms 'audibly imperceptible' and 'inaudible' are treated as synonyms.
- fsckboy 3y ago
  very good explanation! thank you (i've studied DSP and built things out of wires, I didn't need to look any of it up... but great explanation) if there is information in the signal that is inaudible (tiny freq modulation), that information can be removed and greater compression will be achieved without loss of sound quality. So, perhaps mp3 doesn't do that, and perhaps it doesn't occur with enough frequency (hahaha) to make it worthwhile, but still, preservation is not guaranteed. but TIL what you are talking about, useful, thanks
- icy_deadposts 3y ago
  I get that you can monitor phase shift in a signal, if you have knowledge of what the reference is. If I expect a 1kHz sine wave, and you transmit a signal that sounds imperceptibly no different than a pure 1khz sine wave (to a human ear), you could be encoding data in a signal that actually varies over time between 998hz and 1002 hz which could be extracted with DSP. But, how would I extract a phase shift from an arbitrary novel music, for example [0], where I don't know what the expected reference is? [0]https://www.youtube.com/watch?v=yrJYo9d12o8 https://www.youtube.com/watch?v=yrJYo9d12o8
  - ChuckMcM 3y ago
    Off the top of my head, I'd likely try to embed a reference tone. If I couldn't do that imperceptibly I'd then look at what a discontinuous but small phase shift would look like. That would had harmonics and again, it would be a question of how perceptible were the harmonics.
    - icy_deadposts 3y ago
      I'm honestly not familiar with audio compression. But if its very low amplitude reference tone, isn't that the sort of thing that would get removed during compression? and if its large amplitude wouldn't it become a matter of "AI music always has a [low|high pitch] hum in the background?
  - Jensson 3y ago
    Music tones are based on the tuning standard 440 hz and factors from there, that is your reference point. The next tone above and below is so far away that you can make slight alterations and detect how far away from the standard it is by just measuring a single tone.
    - icy_deadposts 3y ago
      I believe (but could be wrong) that's only true for western music, tuned to 440hz as an "A" with 12 steps between octaves. Eastern music for example, with stringed instruments without frets, like Sitar for example, sounds very different and I believe its because they don't follow that practice. Highlighting that was my intent between linking a particular piece of music thats well outside the mainstream top 40. (alternatively, consider my tone-deaf self in middle school trying to tune my euphonium by adjusting the slide randomly and pretending i could hear a difference)

ChuckMcM 3y ago

I think you and I may be talking about different things. If you know what phase noise is and understand the term -10 dBc @ 1kHz, then we might be on the same page. If you don't consider the following; There are tones, things you can hear typically in the range of 15 Hz to 22 kHz. When those tones vary in frequency over time by a few Hertz, it is unlikely that you can perceive that variation. Your ear is both logarithmic in its sensitivity to volume and not particularly sensitive to absolute frequency. If you shift an entire spectrum "up" a couple of Hz your ear won't notice, and similarly if you shift it "down" a couple of Hz you won't notice. This is especially true if you do it slowly (over the course of 100 milliseconds or so). Tape decks and turntables do this, they try to minimize it though. The specification that tells you how closely they track rotational speed is "Wow and Flutter" and a good number is < 0.3%, a studio recorder might be less than 0.1%. With DSP you can easily pull out a spectrum shift of < 0.05% if it is 'regular'. That shift can be a "frequency shift keying" (FSK) signal with forward error correction in it. (think shifted 'up' as one (1) bits and shifted 'down' as zero (0) bits). Running the clip through an FFT and monitoring phase shift in the bins would recover this string of bits. And even if you didn't get enough to error correct the original message, their presence would be unmistakable. All existing audio compression schemes do not affect the phase relationship of the spectrum they are compressing. That's a design feature. To be fair, I wouldn't understand any of this if I hadn't been delving into software defined radio and learning DSP techniques for modulation and demodulation with data recovery in the presence of signal distortions and interference. And I completely understand that there is a language challenge when the terms 'audibly imperceptible' and 'inaudible' are treated as synonyms.

fsckboy 3y ago
very good explanation! thank you (i've studied DSP and built things out of wires, I didn't need to look any of it up... but great explanation) if there is information in the signal that is inaudible (tiny freq modulation), that information can be removed and greater compression will be achieved without loss of sound quality. So, perhaps mp3 doesn't do that, and perhaps it doesn't occur with enough frequency (hahaha) to make it worthwhile, but still, preservation is not guaranteed. but TIL what you are talking about, useful, thanks
icy_deadposts 3y ago
I get that you can monitor phase shift in a signal, if you have knowledge of what the reference is. If I expect a 1kHz sine wave, and you transmit a signal that sounds imperceptibly no different than a pure 1khz sine wave (to a human ear), you could be encoding data in a signal that actually varies over time between 998hz and 1002 hz which could be extracted with DSP. But, how would I extract a phase shift from an arbitrary novel music, for example [0], where I don't know what the expected reference is? [0]https://www.youtube.com/watch?v=yrJYo9d12o8 https://www.youtube.com/watch?v=yrJYo9d12o8
- ChuckMcM 3y ago
  Off the top of my head, I'd likely try to embed a reference tone. If I couldn't do that imperceptibly I'd then look at what a discontinuous but small phase shift would look like. That would had harmonics and again, it would be a question of how perceptible were the harmonics.
  - icy_deadposts 3y ago
    I'm honestly not familiar with audio compression. But if its very low amplitude reference tone, isn't that the sort of thing that would get removed during compression? and if its large amplitude wouldn't it become a matter of "AI music always has a [low|high pitch] hum in the background?
- Jensson 3y ago
  Music tones are based on the tuning standard 440 hz and factors from there, that is your reference point. The next tone above and below is so far away that you can make slight alterations and detect how far away from the standard it is by just measuring a single tone.
  - icy_deadposts 3y ago
    I believe (but could be wrong) that's only true for western music, tuned to 440hz as an "A" with 12 steps between octaves. Eastern music for example, with stringed instruments without frets, like Sitar for example, sounds very different and I believe its because they don't follow that practice. Highlighting that was my intent between linking a particular piece of music thats well outside the mainstream top 40. (alternatively, consider my tone-deaf self in middle school trying to tune my euphonium by adjusting the slide randomly and pretending i could hear a difference)