Stripping frequencies above 20kHz negates the effect on the lower frequencies since those lower frequencies are not "modified" by the higher ones. The human hear can actually hear the very high harmonics when they're combined with a lower fundamental frequency.
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz. If it's played at 44.1kHz, the 30kHz frequency is gone and all you'll hear is 10kHz, not a "different sounding" 10kHz.
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz
You are going to have to provide me with a citation to back that up because that goes against everything I've learned and experience in 17 years of working in acoustics.
Basically, if you produce two ultrasonic frequencies, they will create an interference pattern at a much lower frequency than either of the individual frequencies. Modulate a signal on the difference between two signals, and you can create a directional speaker, since ultrasonic sounds tend to be highly directional (so long as the diameter of the transducer is greater than 1/2 wavelength, which is almost guaranteed with ultrasonic signals). This is how the "sound cannons" that are being deployed for crowd control work.
That article describes hetrodyning which happens because ultrasonic frequencies at high amplitudes interacts nonlinearly with air. You are not going to see that effect with sound waves generated near the audible spectrum, and normal loudspeakers are going to generate ultrasonic sound waves.
Yes, but the effects of interference patterns between multiple ultrasonic frequencies is the same, and definitely does affect the audible spectrum. This is why we must filter the square wave that comes out of a DAC. And the limitations of filters (phase shifts and roll-off) are why modern CD players oversample the signal--so that the filtering can be performed well beyond the audible spectrum.
Yes, but the effects of interference patterns between multiple ultrasonic frequencies is the same, and definitely does affect the audible spectrum
has nothing to do with this:
This is why we must filter the square wave that comes out of a DAC
The only reason that square waves "must" be filtered is to reduce the potential of damaging tweeters. If you want to record a square wave with the purpose of later reproducing the square wave, than you don't want to filter it - once you filter it, it's no longer a square wave.
OK, if you say so. I think you're misunderstanding a fundamental concept of digital to analog converters. But if you think it's just to prevent blowing your speakers, that's OK.
The reason that square wave sucks is because it introduces tons of high frequency content (your amp probably won't reproduce the high frequency content anyway, so I don't think most Japanese consumer amps will damage your speakers--that is, the amp will act like a filter anyway). That high frequency content then creates alias effects (think of moire patterns when looking at super high-res photos that are scaled down without anti-aliasing). Those alias effects sound like shit to the human ear.
The point of filtering is to anti-alias the resulting analog signal after conversion from digital to analog. The point of upsampling is to move that filter well beyond the audible range, so you can use a 1st-order filter (gentle slope, but it introduces no phase effects). The fact that a square wave hurts your speakers is inconsequential--the amp will effectively filter the signal anyway. Unfortunately, it will filter the signal without anti-aliasing, which introduces those nasty interference patterns within the audible spectrum (that is, if you feed a straight 44.1KHz sampled square wave to your speakers without upsampling/filtering).
Recording music is supposed to be a snapshot (with room for interpretation) of the composition at play.
Trying to record an edge case like this is the same as recording in a room with bad acoustics. So you end up with some weird (but not faithful) representation of the sound which is a snapshot of the microphone's characteristics and directionality of the ultrasonic tones. It's not reasonable to assume any microphone will behave exactly like a human ear. Even if you could, you're going to have to mimic the tiny random movements a normal person would make listening to a sound, movements which would definitely impact the perception of the sound, because microphones are much more stationary than any human would be.
The "different sounding" argument two posts above is silly, because sound is almost never that monochromatic, and if it is, it's usually boring. Also I don't understand how missing out on an odd order harmonic would be a bad thing :) The reality is none of these arguments are based in a reality of what people would hear, and because of that, the arguments aren't practical.
In reality, 20 bits at 48kHz (or 64kHz) would be more than acceptable for even the most discerning of ears and probably the most practical in terms of space and fidelity, but it'd be a weird format to distribute in.
> Basically, if you produce two ultrasonic frequencies, they will create an interference pattern at a much lower frequency than either of the individual frequencies.
So the interference pattern will be made up of one low frequency sound and higher frequency harmonics. Once again the higher frequency harmonics are redundant, because you only need to record the lower frequency sound.
The only possible way ultrasound can be picked up by the ear is if the ear has a non-linear response to the input sound. Going by the information in the article linked, it is highly unlikely that any significant non-linearity exists in the ear.
It's definitely possible for two sounds to be indistinguishable when played separately, but when played together it is revealed that they are in fact different (see link below). Whether this applies for sounds with frequencies above 20kHz I don't know.
I'd like to see a citation as well. Doesn't seem like it would be the hardest experiment to set up either.
Me and my brother would sing at each other in certain tones such that we created harmonics in both our ears. It wasn't pleasant, but it was interesting. Regardless, I'd smash my equipment if it made harmonics like that.
If that is true, surely in your up thread example of recording a triangle, the "impact on lower then 20kHz frequencies" would already have happened during the recording process in between the triangle and the microphone, and would have been captured perfectly on recording equipment that's proven capable of capturing everything below 20kHz? So we'd "hear" the effect as part of the recording instead of requiring it to happen in our listening room…
Yes, but if you sample the frequency to create a step wave, then neglect to filter the results, you will end up reproducing tons of high frequencies. That is why we need to filter the output for signals >20KHz...to remove these harmonics that result from reproducing the square wave.
Of course, filters aren't perfect, and result in phase shift and roll-off. So we over-sample the signal to create a signal with a much higher frequency than 20KHz, so that the filtering occurs well outside the audible band, allowing us to filter out all of these harmonics without affecting the desired signal.
Basically, the end result is that by sampling the signal, you are introducing high frequency content that must be removed prior to playback. This high frequency content is one of the reasons old CD players from the 80s and 90s cause "listener fatigue", although I have no sources to back up that last statement.
Yup... people need to get very clear in their heads the difference between the recording/sampling/mixing/mastering stages, where high bitrate/width/gear/knowledge is helpful, and playback, which is a completely different thing.
(not for eatmyshorts -you get this I gather) - everyone gets that "upsampling" can't add detail to a recording right? You can't get more than you've got.... no matter what you do. There is no magic. You upsample so you drive harmonics generated in the digital-to-analog process during playback further up in the spectrum so when you get to the analog stage you can use a nice gentle analog filter to filter them out. Without the upsampling, you need a nasty steep analog filter to filter them out, and that can have audible side-effects (or at least measurable) in the audible spectrum.
eatmyshorts - correct me if I mis-stated any of that please....
You got it 100% correct. You upsample simply to move the frequency of the analog filter higher, with a gentle rolloff (and ideally a 1st order filter, so you introduce no phase effects) to get your final signal.
Well, the superposition principle only holds in linear media. Sound waves can propagate in linear media, but they can also propagate in nonlinear media, and any medium that can carry sound will go nonlinear at sufficiently high amplitudes.
I don't know about the physics of the speaker itself generating the overtone (in cabinet), but it could certainly resonate a wine glass in the room, for example.
Yes, overtones exist, and yes, overtones affect the sound, and yes, if you filtered the sound to remove overtones in the audible range then it would sound different. However, if you remove overtones outside the audible range then it will not make an audible difference (this is what xiphmont was saying in TFA).
So no, your wikipedia link is not a citation for the claim that cmer made.
Yes. A 40KHz tone and a 41KHz tone will interfere with each other and can create 1KHz tones that are audible. Edited to correct error, thanks anechoic.
No, Holosonics is not creating sound from beating, they are using heterodyning, which takes advantage of how high-amplitude ultrasonic sound waves interact with the atmosphere, that's different from beating.
They don't— the air is linear (except at insane sound pressures) so there is no interference. While the ear is not linear, it doesn't respond at those frequencies.
If it really worked that way it would be trivial to demonstrate. Alas, it doesn't.
I know where you're going with this I think, and I'm not disagreeing outright, but wouldn't this be captured during the high-bitrate (or good analog?) recording and mixing phase if the recording/mixing/mastering engineer were doing things right? At least, as well as possible?
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz. If it's played at 44.1kHz, the 30kHz frequency is gone and all you'll hear is 10kHz, not a "different sounding" 10kHz.