Comment by ChuckMcM | Hacker News Reader

Comment by ChuckMcM | Hacker News Reader

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Interesting take, for folks who have played around with digital signal processing (DSP) and low bandwidth / noise tolerant digital signals can probably come up with a half dozen different ways to do this and have it both survive all encoding methodologies and all compression/decompression schemes (at least ones that strive to keep the audio "nearly identical" to the human ear)

It would not survive if you had an analysis package that could back compute the terms used by the model to generate the track and you re-generated the track with your own post processing, but that is a lot of work that most people would avoid.

It would be hilarious if someone built a model that you could query for it to tell you if there was subaudible information in the track and if so remove it (automating the above step to an AI task).

There have been projects that put subaudible tones in audio to trigger toys. So when you're watching a cartoon your toy would respond to the cartoon. Some people may, or may not, have considered doing something like that for muzak at businesses so that your phone OS could report back you had entered the business within a certain window of looking at an ad for that business.

As soon as ML generated voice becomes really good, call scammers will use it against elderly people. That's going to be the #1 application far and away from anything. And as soon as it's out, a whole bunch of extremely privileged ML people will throw their hands up and say, "oh well, cats out of the bag."

nothing new, generated voices are already used for scam. so far they are recordings. like "daughter" screaming for help, or "boss" ordering money transfer. but it's possible there will be real time generators soon. which means scammer will be talking and adjusting depending on the reaction of the victim. there still will be some delay as converter will have some lag. unless the 'scammer' is actually AI. then it may look naturally, especially for not prepared victims. all attacker needs to mimic the voice today is just 3 seconds of recording.

I don't disagree with this sentiment. But I also think it will really change help desk / call center work as well. If you can make the person on the phone have the same accent and language as the caller it makes it easier to understand and more 'comfortable' psychologically. Speech to text + translate could turn the caller's question into a text query in the native language of the agent, the agent's typed response going translate + text to Speech could respond.

That makes the call center easier for both customer and agent and opens up the number of people who could be agents thus increasing competition and cutting costs to deliver call center service.

I wouldn't consider multinational corporations defrauding me into thinking I'm dealing with my fellow countrymen as positive. In fact, it's literally the plot of the movie They Live.

Imagine if Chinese companies masqueraded as American when they sell you phones and gadgets, or American companies masqueraded as Indonesian as they buy up their farmland.

Probably true, but only because the phone system is such a mess. We need to change the rules and ban any attempt to forward a call with changed callerid. If the phone number you forward isn’t the one that will make the phone on the desk that just called ring, it’s a minimum million dollar fine.

Next we need to look at fixing the everyone can call everyone else with no pre approval. Smartphones could easily check the number against a list of people you have called before or phones vetted to be at hospitals or schools.

Businesses would want to disable it, but it would massively cut down on fraud.

Ai is not the problem.

Voice cloning of grandkids is a very targeted attack. You can achieve the same results today at a similar cost if you already have specific info like that

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.

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.

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

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.

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)

If you generate the music yourself you can tune it any way you like. Using 440 hz as the basis lets you still do any kind of music, people wouldn't notice the difference since you mostly hear how it changes and not absolute pitch.

> 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.

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.

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?

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

Even an audio analysis tool won't reveal much if the watermark is sophisticated enough. I believe, a basic watermark adds a few spots on the spectrogram in a known arrangement, a bit like a constellation in the skies, but a watermark can be based on the bloom filter idea: hundreds of barely visible spots arranged with a hash function will look like noise.

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