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.
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?
> Sitar for example, sounds very different and I believe its because they don't follow that practice
They still follow the same practice, maybe they don't center it on 440hz but the concept of tones and how far the steps between them are is the same all over the world, they just use different scales.
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