I recently met a musician who claims he can quite easily hear the difference between 44.1KHz and 96KHz.
This shocked me a little because I'd always been told that the human ear cannot hear any higher quality than CD (44.1KHz) quality.
So... was this guy just lying (or fooled by his senses), or was I being lied to when I was told the human ear cannot hear any higher quality than CD?
The most profound differences are not higher frequency response but the effect of the anti-aliasing filters on the frequencies that can be heard. At 96kHz sampling rate the filter effects at 20kHz are somewhat different than when sampling at 44.1kHz. This is a measurable result but whether or not many, or any, people can truly hear it does not seem to have been established by impartial studies. Or maybe it has but the results don't please the people who promote higher sampling frequencies.
The results of those filters are not exactly a given anyway. There are a variety of ways, and many graduations of these, to accomplish the filtering, with different final effects. There are those who claim to prefer digital recording and playback with no such filters, regardless of the images. There are some who have developed special (non-conventual) processes to produce CD standard files from high sample rate masters without anti-aliasing filters, but also without the stronger images that would result from the more normal downsampling minus the filters. Most or all of this is too subtle for anyone to really hear it if they don't know to which form they are listening at the moment.
There are some limited studies that indicate people may be able to respond to frequencies well above 20kHz (maybe as high as 30-35kHz) under very limited conditions. The mechanisms seem to be something other than hearing, such as direct conduction of vibrations through parts of the body. Such high frequencies are very rarely strong enough to accomplish this.
call him/her on it... make em prove it to you
it is possible that they can hear a difference, but i think it is highly unlikely
record something at 96khz... and make a copy at 44.1khz (resample with something good like SSRC) keep the bitdepth the same
Andy, I didn't realise antialiasing filters were applied to music. Do all audio players use antialiasing?
If no one can hear higher than 44.1KHz, what is antialiasing needed for? Surely it would be too fast to tell the difference between the original aliased sound and the smoother antialiased sound?
MOST audio players (as part of the DAC) use anti-alaising filters. The image is reflected back down from the Nyquist limit. That means it gets mixed into the music. You generally can't detect it as something separate, on its own, it just adds stuff that should not be there.
It comes in reverse order. The lower the signal frequency, above the Nyquist limit, the higher the frequency of its image. Which also means, the higher the frequency above the Nyquist limit, the lower the frequency of its image.
Nyquist limit = 22,050 Hz at 44.1KHz sampling rate
image of audio at 24kHz appears at
(22,050 Hz - (24,000 - 22,050) = 1950Hz) = 20,100 Hz
image of audio at 28kHz appears at
(22,050 Hz - (28,000 - 22,050) = 5950Hz) = 16,100 Hz
Duhh... sorry, that all went completely over my head!
I know that oversampling in A/D and D/A converters are employed to both spread the same amount of noise over a wider spectrum (noise shaping) and to avoid aliasing, even with 44.1 khz.
Digital Finite Impulse Responce filters can be used to filter the signal.
Enrico
Duhh... sorry, that all went completely over my head!
[a href="index.php?act=findpost&pid=362285"][{POST_SNAPBACK}][/a]
It's not half as hard to understand as it sounds.
the Nyquist-thing:you need double the frequency (samples per second here) to record a certain frequency (frequency of the signal wave)
If you want to record a Signal that's 800 Hz you need 1600 Hz (practicaly a bit more afaik, but this is theoretical) to record it
If you'd record a Signal of 800 Hz with a samlingrate of 1500 Hz (1500/2 = 750) there would be 50 Hz too much, those "come back down" and appear at 750 Hz in this case. This is a not really recorded signal and unwanted, the so-called Alias-signal.
So if I'm right a Anti-alias Filter is a low-pass filter (lets signals below a certain frequency go through) that cuts off every frequenzy above half the sample frequency to avoid alias-signals.
The Problem is that those filters can't be "perfect" but if you apply them at a higher sampling-frequency the result will be closer to perfect and might sound a tiny bit better, but that's unaudible for nearly everyone.
hope I didn't tell something wrong and also hope that it's a bit clearer now,
and sorry for my bad english
Oh I think I get it.
It's a lot like picture resolution. In order to resolve a certain level of detail, you need ideally more than double the number of lines than the intended resolution. Ideally as many as possible. If your resolution is too low, the detail blurs together to form new colours that were never part of the original image.
right, it's principialy the same
I found a nice explaination somewhere a few days ago, but I think there's no need for searching since you know how it works.
The main reasons for the higher rate are quite good explained (I think) in AndyH-has post below your original question.
Duhh... sorry, that all went completely over my head!
[a href="index.php?act=findpost&pid=362285"][{POST_SNAPBACK}][/a]
simply put:
frequencies so high that you can't hear them, produce digital conversion artifacts in the range you *can* hear. This phenomenon is called 'aliasing'.
Antialiasing filters block out those artifacts.
Duhh... sorry, that all went completely over my head!
[a href="index.php?act=findpost&pid=362285"][{POST_SNAPBACK}][/a]
simply put:
frequencies so high that you can't hear them, produce digital conversion artifacts in the range you *can* hear. This phenomenon is called 'aliasing'.
Antialiasing filters block out those artifacts.
[a href="index.php?act=findpost&pid=362539"][{POST_SNAPBACK}][/a]
Greetings !
I was just Googling about this right now. There's a site that says that anti-aliasing filters are already good enough at 44 KHz. You shouldn't probably hear the difference because of filter problems. The matter seems to be simpler than that. For example, take a violin and a cello. They may produce faint harmonics at ~ 30 KHz (let's say they the former has one higher peak at 30 KHz and the latter, at 32 KHz). If you listen to them (that is, nothing to do with recording), you may hear a 2KHz beating. However, when you record them, you would do that separately. Record the violin at 44 KHz and you'll lose that important peak. Then record the cello and you'll lose that other important peak. Now mix them together: there's no 2 KHz beating ! If you recorded them together you could sample at 44 KHz and still get the beating. Since you don't, then you'll have to record at least at ~65 KHz. 96 KHz would then be a convenience sampling rate (ie 2x48 KHz).
That's what I read. I'm no speciallist on that. You may google for it also.
Henrique Dante de Almeida
[...] For example, take a violin and a cello. They may produce faint harmonics at ~ 30 KHz (let's say they the former has one higher peak at 30 KHz and the latter, at 32 KHz). If you listen to them (that is, nothing to do with recording), you may hear a 2KHz beating. [...]
[a href="index.php?act=findpost&pid=362544"][{POST_SNAPBACK}][/a]
Why may I hear something like that ?
Sebi
I recently met a musician who claims he can quite easily hear the difference between 44.1KHz and 96KHz.
This shocked me a little because I'd always been told that the human ear cannot hear any higher quality than CD (44.1KHz) quality.
So... was this guy just lying (or fooled by his senses), or was I being lied to when I was told the human ear cannot hear any higher quality than CD?
[a href="index.php?act=findpost&pid=361950"][{POST_SNAPBACK}][/a]
Hi,
this is a 2Khz (stereo) square wave, represented in 16/44.1 PCM
(http://i1.tinypic.com/nd61b4.jpg)
A square wave is actually composed of a sine-wave fundamental (of 2KHz in this case) with an infinite number of it's odd order harmonics folded back into it (3rd, 5th, 7th etc). In fact a'perfect' squarewave doesn't exist, it would have an infintely short rise and decay for each cycle, requiring an infinite number of harmonics, but the more (higher-frequency) of those odd-orders you add, the closer you get to one. This is how the 'edges' needed for digital data transmission are created on such things as analogue phone lines.
This waveform obviously doesn't exist in 'nature', there's no way of producing it acoustically, transmitting it through the air and capturing it with a microphone, it has to be synthesized.
So, this sythesized 2KHz sqaurewave actually has harmonic components extending to 100's of KHz and beyond. Strange but true. You can't 'hear' them, but they're there, they create theis waveform by reinforcing or attenuating the original 2KHz sine.
To actually reproduce this wave 'perfectly' in the analogue domain as the output of a DAC (that is, downstream of it's anti-aliasing filter) is as 'impossible' as the waveform itself is. Filter ringing and phase-shifting between frequencies will produce various effects such as rippling which can be seen graphically if the output is re-captured digitally or monitored in real-time on an oscilloscope.
Now as it happens almost *all* musical instruments produce sound swith harmonic components extending to 40KHz, 50KHz and beyond. Some, such as muted brass produce very substantial pressure levels indeed at these frequencies.
Can we hear them, or sense them in any way? Doubtful (even if you go with the putative non-aural mechanisms some suggest).
BUT they are nonetheless intrinsic to the waveform which results when they are captured - it is *irrelevent* that we cannot 'hear' them, or that the recording hardware or digital protocol is 'band-limited'.
On playback of a recording, the same digital-filtering effects which can be seen graphically in the output of the simple, mathematical square-wave will affect the ultrasonic components of musical instruments and *will* at the very least have an effect on timbre, from innocuous to possibly ear-shredding.
Please don't anybody tell me they *havn't* at some heard point heard a recording of violin or trumpet playing on a CD-based system that didn't make them want to clap their hands over their ears!
I'm not at all surprised to hear that a musician says he/she can hear their instrument reproduced more faithfully with higher sampling rate PCM.
Higher sampling rate = much more benign filtering and more realistic music.
R.
>>edits - yptos as usual.
Now as it happens almost *all* musical instruments produce sound swith harmonic components extending to 40KHz, 50KHz and beyond. Some, such as muted brass produce very substantial pressure levels indeed at these frequencies.
Can we hear them, or sense them in any way? Doubtful (even if you go with the putative non-aural mechanisms some suggest).
OK., so far so good, though whenever someone brings up square waves in a discussion of digital, I expect the worst.
BUT they are nonetheless intrinsic to the waveform which results when they are captured - it is *irrelevent* that we cannot 'hear' them, or that the recording hardware or digital protocol is 'band-limited'.
Wrong. If we can't hear them -- or their effects in the audible range -- then they are indeed irrelevant to our audio experience.
On playback of a recording, the same digital-filtering effects which can be seen graphically in the output of the simple, mathematical square-wave will affect the ultrasonic components of musical instruments and *will* at the very least have an effect on timbre, from innocuous to possibly ear-shredding.
At the very most, that is *possible*, but not *certain* to happen, nor is it at all certain that whatever effect you hear on timbre you hear, is due to the sampling rate. You'd have to rule out lots of other causes. Generally the biggest 'hit' the accuracy of a digital recording takes is when the signal passes through the mic and the speakers -- the electromechanical parts of the chain. These are by far the least linear.
Please don't anybody tell me they *havn't* at some heard point heard a recording of violin or trumpet playing on a CD-based system that didn't make them want to clap their hands over their ears!
Please tell me that that you don't consider this proof that Redbook standard *necessarily* affects the timbre of a recording. (I've heard all-analog recordings that make me want to cover my ears, btw.)
I'm not at all surprised to hear that a musician says he/she can hear their instrument reproduced more faithfully with higher sampling rate PCM.
I'm not surprised that pepoel claim all sorts of things. I'm far more surprised when they've actually tested those claims properly. Because that;s so very rare.
Higher sampling rate = much more benign filtering and more realistic music.
It can mean that. Doesn't necessarily mean that. It's down to how well the filtering is implemented.
.
[...] For example, take a violin and a cello. They may produce faint harmonics at ~ 30 KHz (let's say they the former has one higher peak at 30 KHz and the latter, at 32 KHz). If you listen to them (that is, nothing to do with recording), you may hear a 2KHz beating. [...]
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=362544")
Why may I hear something like that ?
Sebi
[a href="index.php?act=findpost&pid=362564"][{POST_SNAPBACK}][/a]
It was just an example. If you were talking that the 2 KHz was a mistake, I'm sorry, it should be 1 KHz. If not, it's because of the following. I supposed that there would be an instrument which would produce a significant harmonic at 30 KHz (actually this is true, for example, for violins and flutes), and another that would produce it at 32 KHz. Since those frequencies are actually a pressure in the same medium (that is the air and then your ear), the expansions and compressions generated by the instruments will add to each other some times and cancel each other some other times at a rate of 1 KHz. Mathematically, cos(32KHz)+cos(30KHz) = 2*cos(31KHz)*cos(1KHz). Add the time in equation and you have a 1 KHz harmonic with a variable intensity of 2*cos(31KHz*t). In practice, you should have a few harmonics for every instrument in this region. For most of them, they will be so faint, that you won't ever notice it. The already cited instruments, however, are known to cause audible beating which enrich the listening experience. Unfortunately I have no link to that, except one that also claims this is true, but doesn't cite sources either :-/.
[a href="http://www.dvdsoftwareguide.com/all-about-dvd-4-guide.html]http://www.dvdsoftwareguide.com/all-about-dvd-4-guide.html[/url]
One should hope, thus, that every recording is made at very high sampling rates. After they are mixed and filtered with high quality equipment, they may be safely downsampled to human listening limits again.
On playback of a recording, the same digital-filtering effects which can be seen graphically in the output of the simple, mathematical square-wave will affect the ultrasonic components of musical instruments and *will* at the very least have an effect on timbre, from innocuous to possibly ear-shredding.
Please don't anybody tell me they *havn't* at some heard point heard a recording of violin or trumpet playing on a CD-based system that didn't make them want to clap their hands over their ears!
R.
>>edits - yptos as usual.
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=362582")
Again, I don't think that filtering artifacts are relevant. The issues only happen with bugged filtering. Recent equipment shouldn't cause audible artifacts. Concerning the bad violin or trumpet, see the other discussion.
[a href="http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm]http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm[/url]
Henrique Dante de Almeida
@gangran dizzy
i hear alot of audiophiles around me claiming alot.
They do have a hard time proving it... atctually they have failed all there claims when doing probally scientific correct testing.
Wrong. If we can't hear them -- or their effects in the audible range -- then they are indeed irrelevant to our audio experience.
[a href="index.php?act=findpost&pid=362595"][{POST_SNAPBACK}][/a]
I don't want to get into yet another nit-picking session on this.
You simply havn't grasped the point I'm making.
If the intrinsic ultrasonic content of our squarewave (of whatever frequency) cannot be maintained with 'time-domain coherency' on playback (within a band-limited playback-system), then QED - neither can any other *captured* sound/waveform which is defined by it's ultrasonic components.
The timbre of many, if not most, musical instruments *is* defined by this ultrasonic content. I'm amazed how few people haven woken up to this.
As I said - the ability to perceive/hear ultrasonic frequencies or to capture them discreetly with recording equipment is completely *irrelevent*.
Time-domain coherency is the key to realistic reproduction of music - instruments, voices, whatever.
Of course, if we're lsitening to a Stratocaster and an overdriven Marshall stack, this might all be moot.
R.
Please don't anybody tell me they *havn't* at some heard point heard a recording of violin or trumpet playing on a CD-based system that didn't make them want to clap their hands over their ears!
I'm not at all surprised to hear that a musician says he/she can hear their instrument reproduced more faithfully with higher sampling rate PCM.
Higher sampling rate = much more benign filtering and more realistic music.
R.
>>edits - yptos as usual.
[a href="index.php?act=findpost&pid=362582"][{POST_SNAPBACK}][/a]
By reading your statement(s) here, one would think you have not paid any attention to the last several discussions on hi-rez audio on hydrogenaudio.org. Please go back and reference these prior discussions.
-Chris
Again, I don't think that filtering artifacts are relevant. The issues only happen with bugged filtering. Recent equipment shouldn't cause audible artifacts. Concerning the bad violin or trumpet, see the other discussion.
http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm (http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm)
Henrique Dante de Almeida
[a href="index.php?act=findpost&pid=362600"][{POST_SNAPBACK}][/a]
It really isn't a matter of opinion.
Oversampling filters serve as a panacea for the limited resolution of RB CD (16/44 PCM).
But many people are building completely filterless/non-oversampling DACS. Why, one should be bound to ask?
Here's the rub; OS DACs do sine waves pretty well up to (insert freq; 10KHz?) but are utterly incapable of resolving a squarewave at anything close to this freq.
Non-OS DACs make an unholy mess of sines above 10KHz, but (at least some of them) can do squares at this frequency and beyond.
In a previous discsussion here at HA, someone siad that the the Non-OS DAC's inability to reproduce HF sines meant they were "broken".
Why, then, does the OS DAC's inability to reproduce HF squares not mean they are *edit >* NOT "broken"?
R.
By reading your statement(s) here, one would think you have not paid any attention to the last several discussions on hi-rez audio on hydrogenaudio.org. Please go back and reference these prior discussions.
-Chris
[a href="index.php?act=findpost&pid=362621"][{POST_SNAPBACK}][/a]
I do actually pay attention, and not just to HA.
Do you have a point?
R.
You simply havn't grasped the point I'm making.
I have grasped your point, as I'm sure the other post has as well: You think that ultrasonic information has some relevance to the audible waveform, though such has not ever been shown in a credible peer-reviewed study. But your premise as presented here is in error. 1. Assuming that analog filters are used for the anti-alias filters, no one has shown the phase distortion that occurs as a result to be audible. 2. Most systems today should be using linear phase digital filtes, in which pase distortion is not even an issue. 3. Since you can only hear frequencies <X, the waveform content >X is not relevant to audibility. You can not hear square waves, for example. You can only hear the sine waves composing the square waves which are <X frequency. The content >X is only relevant to a pretty looking graphical respresentation of the waveform, not to audible parameters. How you think supersonic content has to do with time domain coherancy of the audible waveforms, I have yet to figure out.
-Chris
Of course, if we're lsitening to a Stratocaster and an overdriven Marshall stack, this might all be moot.
[a href="index.php?act=findpost&pid=362616"][{POST_SNAPBACK}][/a]
I take that back. It might well be just as important.
[...] For example, take a violin and a cello. They may produce faint harmonics at ~ 30 KHz (let's say they the former has one higher peak at 30 KHz and the latter, at 32 KHz). If you listen to them (that is, nothing to do with recording), you may hear a 2KHz beating. [...]
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=362544")
Why may I hear something like that ?
Sebi
[a href="index.php?act=findpost&pid=362564"][{POST_SNAPBACK}][/a]
It was just an example. If you were talking that the 2 KHz was a mistake, I'm sorry, it should be 1 KHz. If not, it's because of the following. I supposed that there would be an instrument which would produce a significant harmonic at 30 KHz (actually this is true, for example, for violins and flutes), and another that would produce it at 32 KHz. Since those frequencies are actually a pressure in the same medium (that is the air and then your ear), the expansions and compressions generated by the instruments will add to each other some times and cancel each other some other times at a rate of 1 KHz. Mathematically, cos(32KHz)+cos(30KHz) = 2*cos(31KHz)*cos(1KHz). Add the time in equation and you have a 1 KHz harmonic with a variable intensity of 2*cos(31KHz*t). In practice, you should have a few harmonics for every instrument in this region. For most of them, they will be so faint, that you won't ever notice it. The already cited instruments, however, are known to cause audible beating which enrich the listening experience. Unfortunately I have no link to that, except one that also claims this is true, but doesn't cite sources either :-/.
[a href="http://www.dvdsoftwareguide.com/all-about-dvd-4-guide.html]http://www.dvdsoftwareguide.com/all-about-dvd-4-guide.html[/url]
One should hope, thus, that every recording is made at very high sampling rates. After they are mixed and filtered with high quality equipment, they may be safely downsampled to human listening limits again.
[a href="index.php?act=findpost&pid=362598"][{POST_SNAPBACK}][/a]
That's really interesting actually. I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. Though at 2khz not 1khz
Why do you say the hi-res mix may be safely downsampled to 'human hearing limits'? If I resampled the above wav file to 44.1khz I ended up with silence!
Again, I don't think that filtering artifacts are relevant. The issues only happen with bugged filtering. Recent equipment shouldn't cause audible artifacts. Concerning the bad violin or trumpet, see the other discussion.
http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm (http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm)
Henrique Dante de Almeida
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=362600")
It really isn't a matter of opinion.
Oversampling filters serve as a panacea for the limited resolution of RB CD (16/44 PCM).
Wrong. Resolution is a function of word length (bit depth), not sampling rate.
Oversampling filters were a solution to the difficulty (not impossibility) in implementing excellent filtering at 44.1.
But many people are building completely filterless/non-oversampling DACS. Why, one should be bound to ask?
Because there's no silly idea that some audiophile won't embrace. There are belt-driven CD players out there too. And no, it's not *many* people doing this. It's a relatively tiny cult, as with most audiophile tweaks. One question to ask is *who* is doing it.
I looked up your posts here and I see you were arguing for cryogenic treatment of wires elsewhere.
I suggest interested parties check out Nika Aldrich's comments on quare waves,
before they disappear from the the Web (they're now cache-only). Or buy his book on 'Digital Audio Explained' -
[a href="http://72.14.207.104/search?q=cache:stHyib1uSZcJ:www.musicgearnetwork.com/cgi-bin/ultimatebb.cgi%3Fubb%3Dget_topic%3Bf%3D3%3Bt%3D000822+nika+%22square+wave%22&hl=en&gl=us&ct=clnk&cd=1]http://72.14.207.104/search?q=cache:stHyib...us&ct=clnk&cd=1[/url]
The bottom line with the point above is that, depending on the quality and design of the filter, 44.1 or 48k are perfectly adequate to COMPLETELY represent any signals under 20k. Thus, for the sake of what is commonly accepted as our ears' hearing ability (from 20Hz to 20kHz) 96kHz recording is totally unnecessary. For further information on the theories of the potential validity of 96kHz, see the topic that I referred to above in my original post. There are indeed some theories that are worth exploring, only one of which is the "psychoacoustic" theory that we can percieve information that our ears are not attributed to being able to hear. I must tell you that this is, I believe, the weakest of all of the theories.
As for the notion that things that we can't hear can affect the things that we can hear, the answer is "no". Defiantly "no". Your ear acts as the same type of filter that we discussed above. If we take a 1kHz sine wave and then add all kinds of processing to it of very high frequencies (50k and such) so that in the end it doesn't really look like a 1kHz sine wave at all, and then put a filter on it that filters out everything over 2kHz, all you'll be left with is your 1kHz sine wave. I don't care how much junk you added. Once you add that 2kHz filter, it's right back to a 1kHz sine wave.
The thing is that your ears work this way also. If you take a clarinet note and add all kinds of garbage at ultra high frequencies to it for the sake of who-knows-what and whatnot, by the time you listen to it, all you'll hear is the clarinet.
If, however, you take the same clarinet and add some eq to it at 2.5k which also induces some wacky 50kHz stuff to happen, it would be incorrect to say that the 50kHz signal is CREATING differences that you can hear. What would be more correct is to say that you are processing the signal at 2.5k and some side effects at 50k, but that your ear won't hear the results of what happened at 50k. All that you'll end up hearing is the clarinet and the change of it at 2.5k. The 50k didn't CAUSE the change. It is a BIPRODUCT of the change, and an inaudible one that will be filtered away.
That's really interesting actually. I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. Though at 2khz not 1khz
[a href="index.php?act=findpost&pid=362629"][{POST_SNAPBACK}][/a]
Listen to the re-mastered release of Steely Dan's "Can't Buy A Thrill". I gather it was transferred from 50KHZ (!) open-reel digital transfers of the analogue 2-tracks that Roger Nichols and the band saw as imperetive back in the early 80's.
From the get-go, there is an obvious 'beat' or pumping effect happening on the intro to "Do It Again" which renders it practically unlistenable. Probably a result of artifacts resulting from sample-rate conversion, reinforcing inate problems with PCM.
I looked up your posts .....
[a href="index.php?act=findpost&pid=362634"][{POST_SNAPBACK}][/a]
I'm flattered.
R.
You bore me.
The last word is yours, please do savour it.
R.
But many people are building completely filterless/non-oversampling DACS. Why, one should be bound to ask?
Here's the rub; OS DACs do sine waves pretty well up to (insert freq; 10KHz?) but are utterly incapable of resolving a squarewave at anything close to this freq.
R.
[a href="index.php?act=findpost&pid=362622"][{POST_SNAPBACK}][/a]
Hello,
For the few things that I've read here/know about SP, not using filters/aliasing reduction would create a violin++, instead of a normal violin. That may or may not be what the composer intended. Is this correct ?
For the square waves, human beings can't hear them. Our ears would filter them anyway. If the sound device already does this, then great, less work for my psychoacoustic organs.
Why do you say the hi-res mix may be safely downsampled to 'human hearing limits'? If I resampled the above wav file to 44.1khz I ended up with silence!
Try it with two separate tracks (remember, that was the original proposal). Mix them together to produce the beat frequency. Now you have captured it in a file. Before you were only producing it as you listened to it. Now when you downsample, it can't go away.
That's really interesting actually. I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. Though at 2khz not 1khz
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=362629")
Listen to the re-mastered release of Steely Dan's "Can't Buy A Thrill". I gather it was transferred from 50KHZ (!) open-reel digital transfers of the analogue 2-tracks that Roger Nichols and the band saw as imperetive back in the early 80's.
From the get-go, there is an obvious 'beat' or pumping effect happening on the intro to "Do It Again" which renders it practically unlistenable. Probably a result of artifacts resulting from sample-rate conversion, reinforcing inate problems with PCM.
[a href="index.php?act=findpost&pid=362641"][{POST_SNAPBACK}][/a]
Wow. You 'gather' that? Roger Nichols has a forum online. Maybe he could evaluate your theory authoritatively.
[a href="http://www.ioforums.net/forums/]http://www.ioforums.net/forums/[/url]
I believe his current forum name is 'ioforums33'
Why do you say the hi-res mix may be safely downsampled to 'human hearing limits'? If I resampled the above wav file to 44.1khz I ended up with silence!
Try it with two separate tracks (remember, that was the original proposal). Mix them together to produce the beat frequency. Now you have captured it in a file. Before you were only producing it as you listened to it. Now when you downsample, it can't go away.
[a href="index.php?act=findpost&pid=362653"][{POST_SNAPBACK}][/a]
No, I did mix it down to a file. As a quick way to do it I put a 30khz tone in the left channel of a stereo wav file and a 32khz tone in the right then mixed it down to mono. Play this back and I can hear a quiet 2khz tone. The frequency analyser only shows peaks at 30khz and 32khz despite me clearly being able to hear a sound.
I downsample to 44.1khz and the resulting file is totally silent. I can put flac files on the web to demonstrate if you want (obviously, you need a soundcard that supports 96khz playback).
That's really interesting actually. I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. Though at 2khz not 1khz
Why do you say the hi-res mix may be safely downsampled to 'human hearing limits'? If I resampled the above wav file to 44.1khz I ended up with silence!
[a href="index.php?act=findpost&pid=362629"][{POST_SNAPBACK}][/a]
Could it be that it's not really safe to downsample the sound ? :-D. Try downsampling to integer frequencies (ex: 48 KHz) to see if it works :-/
Henrique Dante de Almeida
http://www.ioforums.net/forums/view_topic....rum_id=1&page=1 (http://www.ioforums.net/forums/view_topic.php?id=1&forum_id=1&page=1)
Let's get the ball rolling here.
1) Most of the 96k stuff you buy on DVD-A is up-sampled from 44.1k or 48k material. I know because they have done it to some of my stuff.
2) In a double blind test (two blind guys), nobody was able to tell the difference between the same mix printed to 96k and 44.1k at the same time.
3) If you take a 44.1k mix and up-sample it to 96k and send someone the two files to listen to, they will always pick the 96k file as the better sounding of the two. But they are identical.
4) I don't even want to talk about 192k.
Roger
Has anyone read Bob Katz' 'Mastering Audio'? He claims to have proven that with the higher sampling rates, it's not the extended frequency response, that we're hearing as better. The higher sampling rates, just move the low pass filter out of our hearing range. He says that with better filters in the converters, 44.1/48 can sound just as good!
Why do you say the hi-res mix may be safely downsampled to 'human hearing limits'? If I resampled the above wav file to 44.1khz I ended up with silence!
[a href="index.php?act=findpost&pid=362629"][{POST_SNAPBACK}][/a]
If the beat frequency is an actual artifact that existed before recording, or as an audible artifact in the recording, it would be recorded within a 44.1kHz sample rate, since it would be an artifact in the audible range. However, if it is an artifact that exists after the fact(as some hardware will distort with intermodulation effects when presented with multiple high amplitude ultrasonic signals), then of course it would disappear when downsampled, since you removed the ultrasonic compoents causing the distortion(s). This was a sythetically created file, remember, not a recording of an actual sonic event that contains the lower band artifact distortions originally. It sounds like you may have hardware that is not well suited for ultrasonic content, unless you are just cranking the volume up to insane levels, exaggerating the problem that would not normally exist. However, it would not normally exist anyways, since the ultrasonic content would be tens of dBs under the main bandwidth midrange and bass levels.
-Chris
That's really interesting actually. I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. Though at 2khz not 1khz
Why do you say the hi-res mix may be safely downsampled to 'human hearing limits'? If I resampled the above wav file to 44.1khz I ended up with silence!
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=362629")
Could it be that it's not really safe to downsample the sound ? :-D. Try downsampling to integer frequencies (ex: 48 KHz) to see if it works :-/
Henrique Dante de Almeida
[a href="index.php?act=findpost&pid=362660"][{POST_SNAPBACK}][/a]
Same result if I downsample to 48khz. Though there is a beat-frequency at 2khz it is made up of two component continual sinewaves at 30khz and 32khz, PCM only considers the component elements not the result, in the same way as a square-wave is approximated through a large number of sinewaves. Sample below 64khz and the 2khz baby gets thrown out with the 30 and 32khz bathwater.
This is my (badly worded) explaination anyway, from my understanding of the Nyquist theorem...
Here is the 96khz file containing the two tones, I've doubled the mono up to stereo, please be careful with it.
Play this loud and you may very well DESTROY your speakers, headphones or amplifier[a href="http://www.arafel.org.uk/~mandel/ha/ultratones.flac]http://www.arafel.org.uk/~mandel/ha/ultratones.flac[/url]
Edit: Added a bigger warning
Here is the 96khz file containing the two tones, I've doubled the mono up to stereo, please be careful with it, lots of ultrasonic noise has destructive potential...
http://www.arafel.org.uk/~mandel/ha/ultratones.flac (http://www.arafel.org.uk/~mandel/ha/ultratones.flac)
[a href="index.php?act=findpost&pid=362669"][{POST_SNAPBACK}][/a]
I did not use your samples, but I did create a file exactly as you described. The 2kHz byproduct tone is present in any loop back recording, and will survive a 44.1Khz sample rate file, since the 2kHz is a byproduct of intermodulation of the signals that must occur in the hardware analog stage. The synthetic file you created has nothing to do with a natural occurance. Being a digital creation program, it can create aritifical circumstances(such as seen here) where you can have two discrete tones exist without intermodulation behaviour(that must happen in the analog realm). But when this happend in the real world(analog realm), the intermodulation artifacts will be present, and will be recordable directly.
-Chris
Here is the 96khz file containing the two tones, I've doubled the mono up to stereo, please be careful with it, lots of ultrasonic noise has destructive potential...
http://www.arafel.org.uk/~mandel/ha/ultratones.flac (http://www.arafel.org.uk/~mandel/ha/ultratones.flac)
[a href="index.php?act=findpost&pid=362669"][{POST_SNAPBACK}][/a]
I did not use your samples, but I did create a file exactly as you described. The 2kHz byproduct tone is present in any loop back recording, and will survive a 44.1Khz sample rate file, since the 2kHz is a byproduct of intermodulation of the signals that must occur in the hardware analog stage. The synthetic file you created has nothing to do with a natural occurance. Being a digital creation program, it can create aritifical circumstances(such as seen here) where you can have two discrete tones exist without intermodulation behaviour(that must happen in the analog realm). But when this happend in the real world(analog realm), the intermodulation artifacts will be present, and will be recordable directly.
-Chris
[a href="index.php?act=findpost&pid=362672"][{POST_SNAPBACK}][/a]
I agree with all your points. However, to quote an article linked to previously in this thread (http://www.digitalprosound.com/Htm/SoapBox/soap2_Apogee.htm):
Why Record Ultrasonics?
As is widely recognized, most of us can ’t hear much above 18 kHz, but that does not mean that there isn’t anything up there that we need to record – and here's another reason for higher sampling rates. Plenty of acoustic instruments produce usable output up to around the 30 kHz mark – something that would be picked up in some form by a decent 30 in/s half-inch analog recording. A string section, for example, could well produce some significant ultrasonic energy.
Arguably, the ultrasonic content of all those instruments blends together to produce audible beat frequencies which contribute to the overall timbre of the sound. If you record your string section at a distance with a stereo pair, for example, all those interactions will have taken place in the air before your microphones ever capture the sound.You can record such a signal with 44.1 kHz sampling and never worry about losing anything –as long as your filters are of good quality and you have enough bits.
If, however, you recorded a string section with a couple of 48-track digital machines, mic on each instrument feeding its own track so that you can mix it all later, your close-mic technique does not pick up any interactions.The only time they can happen is when you mix – by which time the ultrasonic stuff has all been knocked off by your 48 kHz multitrack recorders, so that will never happen. It would thus seem that high sampling rates allow the flexibility of using different mic techniques with better results.
Here when mixing the 48 track recording down to stereo all in the digital domain there is a similar situation to the artificial one I created. Suppose that the first violins are playing a note with an overtone at 30khz and the second violins a note with a tone at 32khz, the close miking prevents the intermodulation distortion in the air from being recorded, however if all mixing is done at 96khz the distortion can reappear.
While as you say a loop-back recording at 44.1khz will capture the sound fine a straight downsample in the digital domain won't. The latter is what occurs when a CD is produced from a hi-res master.
Here is the 96khz file containing the two tones, I've doubled the mono up to stereo, please be careful with it, lots of ultrasonic noise has destructive potential...
http://www.arafel.org.uk/~mandel/ha/ultratones.flac (http://www.arafel.org.uk/~mandel/ha/ultratones.flac)
[a href="index.php?act=findpost&pid=362669"][{POST_SNAPBACK}][/a]
Its a very interesting sound. My own analysis indicates there is no virtualy apparent 2khz sinusoidal content at all. But the beating of the high frequencies occurs in 2 khz non sinusoidal pulses.
The content of the beating is still ultrasonic though. I have a notion that this beating wouldn't be detected by the human ear because it depends on an ability of the ear to be affected (some sensory part to resonate) with 30Khz tone.
A pcm recording device could be affected by the tones at lower sampling frequencies because of the aliasing characterist of taking discreet level measurements rather than level integrals .
Then the representation of them is further complicated by their beating.
What would really reach the microphone, would be silence alternating with a 30khz (31?) tone, 2000 times a second, what is recorded is a variation in instantaneous level, heard in playback as an inaccurate reconstruction of that variations most likely cause.
I hazard audio should not be sampled at rates below the capability of mics to capture instantaneous level, or there is no way to do the antialiasing. The audability of the two high frequency tones should be a technical artifact of the sampling process, corrected by high quality downsampling. The experiment may highlight a problem with playback of high sample rate files.
regards'
Same result if I downsample to 48khz. Though there is a beat-frequency at 2khz it is made up of two component continual sinewaves at 30khz and 32khz, PCM only considers the component elements not the result, in the same way as a square-wave is approximated through a large number of sinewaves. Sample below 64khz and the 2khz baby gets thrown out with the 30 and 32khz bathwater.
This is my (badly worded) explaination anyway, from my understanding of the Nyquist theorem...
Here is the 96khz file containing the two tones, I've doubled the mono up to stereo, please be careful with it, lots of ultrasonic noise has destructive potential...
http://www.arafel.org.uk/~mandel/ha/ultratones.flac (http://www.arafel.org.uk/~mandel/ha/ultratones.flac)
[a href="index.php?act=findpost&pid=362669"][{POST_SNAPBACK}][/a]
Hey,
Could you try again, but with 30 KHz and 16 KHz tones ? Record them at 96 KHz, then downsample them to 48 KHz. (I am wondering if we cannot really listen to the 30+32 tones)
Same result if I downsample to 48khz. Though there is a beat-frequency at 2khz it is made up of two component continual sinewaves at 30khz and 32khz, PCM only considers the component elements not the result, in the same way as a square-wave is approximated through a large number of sinewaves. Sample below 64khz and the 2khz baby gets thrown out with the 30 and 32khz bathwater.
This is my (badly worded) explaination anyway, from my understanding of the Nyquist theorem...
Here is the 96khz file containing the two tones, I've doubled the mono up to stereo, please be careful with it, lots of ultrasonic noise has destructive potential...
http://www.arafel.org.uk/~mandel/ha/ultratones.flac (http://www.arafel.org.uk/~mandel/ha/ultratones.flac)
[a href="index.php?act=findpost&pid=362669"][{POST_SNAPBACK}][/a]
Hey,
Could you try again, but with 30 KHz and 16 KHz tones ? Record them at 96 KHz, then downsample them to 48 KHz. (I am wondering if we cannot really listen to the 30+32 tones)
[a href="index.php?act=findpost&pid=362767"][{POST_SNAPBACK}][/a]
If I do that then all I can hear at 96 or 48khz is the 16khz tone making my ears bleed
Oh except that it sounds more of a clear tone when downsampled to 48khz. There is something removed by the downsample
This thread is fascinating! But most of it is too techy for me unfortunately.
Could anyone try again to explain how these byproduct tones/beats come about? And what is the low pass filter's involvement in this? (What is a low pass filter anyway?)
Well, here is what little I can remember from RF theory class (applies here as well). When you have a signal, say at 1kHz, and you send it through a non-linear element, such as the real world, you start producing harmonics at integer multiples, example at 2kHz, 3kHz, etc. Each of these is weaker than the one before though.
When you have 2 signals (sines, whatever) you still get the harmonics, and you start getting the cross-products, which is the beat frequencies you guys hear. Thing is, you only hear them cause they are getting sent through a whole bunch of non-linear elements, such as mixers, amps, speakers, the air, etc. BTW, sometimes you want very linear components in the signal path, for example power amps, etc. Sometimes you want very non-linear components, such as mixers, which are used in every single RF device you own, to bring in the stratospheric 2GHz or whatever frequencies down to more manageable ranges.
So, in theory, if you keep the 30k and 32k signals completly in the "digital/mathematical" domain, and you resample down to below their nyquist freq, and the resampling is "linear", then the signals should just disappear. If the resampling is non-linear, which could be on purpose to get a better sounding sound or whatever, then you might still get the 2k cross-product. In a way, the beat frequency is an illusion created by the real world.
This thread is fascinating! But most of it is too techy for me unfortunately.
Could anyone try again to explain how these byproduct tones/beats come about? And what is the low pass filter's involvement in this? (What is a low pass filter anyway?)
Imagine two people walking side by side but with different length of steps. They can start pacing together on the same foot, but soon they will be out of step, and on different foots, some more steps later theyll get into pace again and some steps more -out of pace...the cycle will continue.
That cycle is the beating of the two different paces.
The PCM record is like noting down the gross foot position every click of a timer, so on clicks where both walkers feet are agreeing the gross foot position will be complimentry , when the walkers are directly out of step, each foot's position will counter the other and if directly out of step (and the feet are equal weight) their positions impression in the gross feet record will be zero.
In the record it looks like there is one foot moving at a similar rate but twice as much as each of the real feet, but coming in and out of existence in time with the beat period.
The question of how this beat period is generating an actual tone (other than the two tones related to the walkers paces) on playback of the record, is in contention in this thread.
Some believe that the beat period will be heard as a tone (walkers pace), or that the period will produced a tone in the air ~maybe a fluid dynamic phenomenon or something. It should be remembered that the beat isnt a tone though, its the periodic rising and falling in loudness of a tone.
The frequency of the tones being examined here is ultrasonic but playing their PCM record is producing an audible tone with the frequency of the beat period. This is suggesting to some people that beat periods are hearable as tones.
A lowpass filter cleans a pcm record of all mathematicaly apparent suggestions of sinusoidal oscillations above a certain frequency, (allows all the lower frequencies to pass).
When the ultrasonic test signal is lowpassed to remove theoreticaly inaudible frequencies from the record, the 'sound' of the beat frequency disappears as well.
The question is, is the 'sound' of the beat frequency something we would hear in real life or is it an error of the digital recording and playback process?
I guess the best way to proof that it has influence on the realworld-sound is to produce such a signal in the realworld. If we can't it's not a proof that it's impossible, but if we can...
What would be needed to do so?
The question is, is the 'sound' of the beat frequency something we would hear in real life or is it an error of the digital recording and playback process?
The sound of the beat freq has nothing to do with the digital process. You could say it is a completly analog phenomenon. It will happen with a digital source as well as with some high-bandwidth analog tape or a vinyl deck or whatever. It is just the mix of two different signals.
-
Here when mixing the 48 track recording down to stereo all in the digital domain there is a similar situation to the artificial one I created. Suppose that the first violins are playing a note with an overtone at 30khz and the second violins a note with a tone at 32khz, the close miking prevents the intermodulation distortion in the air from being recorded, however if all mixing is done at 96khz the distortion can reappear.
While as you say a loop-back recording at 44.1khz will capture the sound fine a straight downsample in the digital domain won't. The latter is what occurs when a CD is produced from a hi-res master.
[a href="index.php?act=findpost&pid=362673"][{POST_SNAPBACK}][/a]
Did you note the relative level of the intermodulated product? It was only a few dBs above the noisefloor, when I generated the 30kHz and 32kHz tones at *-3dBFs! Even then, it was only audible when I cranked the volume high enough that would never be usable for music listening. This is totally unrealistic circumstance. First of all, the difference tone would be masked in real music due to the other signals. Second, the relative levels of harmonics in ultrasonic range in real music is tens of dBs under 0 dBFs, not -3dBFs. The effect as seen in your example, for the most part in real circumstances, would be buried under the noisefloor, and if it did manage to occur over the noisefloor, it would likely be masked by anything in the lower bands.
-Chris
* Edit notice: The final amplitude of the waveform after generating and mixing these individual waveforms was -3dBFs. It appears that my poorly worded statement could be read as if I mean that each individual waveform was -3dBFs before mixing.
Did you note the relative level of the intermodulated product? It was only a few dBs above the noisefloor, when I generated the 30kHz and 32kHz tones at -3dBFs! [a href="index.php?act=findpost&pid=362911"][{POST_SNAPBACK}][/a]
So you added two sine waves with an amplitude of -3dBFs? In that case the result will clip and that may explain the tone you heard..
The sound of the beat freq has nothing to do with the digital process. You could say it is a completly analog phenomenon. It will happen with a digital source as well as with some high-bandwidth analog tape or a vinyl deck or whatever. It is just the mix of two different signals.[a href="index.php?act=findpost&pid=362903"][{POST_SNAPBACK}][/a]
I know a beat will happen whatever, but a sound is generated in some playback configurations and not others.
So you added two sine waves with an amplitude of -3dBFs? In that case the result will clip and that may explain the tone you heard..
No ones been listening to clipped test signals
It has been mentioned that in the real world very high frequencies like this may produce audible products, but with these test signals no post processing has been done to the sine waves to simulate possible real world phenomenon, the impression of two ultrasonic sines should be all that is present in the PCM record -no subtle air dynamic phenomon in this signal. But an unexplained noise is present on playback. Since, high quality lowpassing leaves no audible noise, it is consistent with the virtual sines not having generated any extra audible product.
So threre really appears to me to be a problem in the playback. Maybe this is lax internal resampling in the soundcard, or patterned rounding error, i can only guess.
Maybe try same experiment at 24 bits, or try playback through a ~XiFi
Did you note the relative level of the intermodulated product? It was only a few dBs above the noisefloor, when I generated the 30kHz and 32kHz tones at -3dBFs! [a href="index.php?act=findpost&pid=362911"][{POST_SNAPBACK}][/a]
So you added two sine waves with an amplitude of -3dBFs? In that case the result will clip and that may explain the tone you heard..
[a href="index.php?act=findpost&pid=362936"][{POST_SNAPBACK}][/a]
I am sorry, I made a typo earlier. The final waveform peak amplitude as viewed after mixing the two signals was -3dBFs, not the two individual signals before mixing. Please note that the 2kHz *tone* was not audible unless very specific conditions were present, as I previously specified.
-Chris
more on the 'beating' issue, from James Johnston:
A Graphical Explanation involving Sampling
http://www.audioasylum.com/audio/general/messages/43134.html (http://www.audioasylum.com/audio/general/messages/43134.html)
Opener:
"The questions about modulation of sine waves near the Nyquist limit seem to be common and repeated. The image URL here has a plot that, I think, explains how the "beating" comes about due to adding in frequencies that are solely above the Nyquist limit, and thus how filtering them out removes any "beating" one observes in raw, unfiltered data from a DAC (before the anti-imaging filter)."
This thread is fascinating! But most of it is too techy for me unfortunately.
Could anyone try again to explain how these byproduct tones/beats come about? And what is the low pass filter's involvement in this? (What is a low pass filter anyway?)
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=362848")
Fascinating? Hmm.... May seem so. But I think that some posts here are actually a bit misleading/misinformed.
Why may I hear something like that ?
[a href="index.php?act=findpost&pid=362564"][{POST_SNAPBACK}][/a]
[...] Mathematically, cos(32KHz)+cos(30KHz) = 2*cos(31KHz)*cos(1KHz). Add the time in equation and you have a 1 KHz harmonic with a variable intensity of 2*cos(31KHz*t). [...]
[a href="index.php?act=findpost&pid=362598"][{POST_SNAPBACK}][/a]
So what ? Just because there's suddenly a cos(1kHz*t) in a product you think you should be able to hear this one? Do you realize that you can also interpret your formula as a 31 kHz sinusoid with an amplitude modulation at a rate of 1 kHz ? Truth is: It does not matter whether a product representation contains usually audible signals. The first form (cos(32kHz)+cos(30kHz)) is the really interesting one. Why ? Because we've a bunch of filters in our inner ear (hair + hair cell) seperating frequencies -- every hair (depending on its location/length) has a certain resonance frequency and acts like a bandpass filter. From signal theory we know such filters are linear and time-invariant ([a href="http://en.wikipedia.org/wiki/LTI_system_theory]LTI system theory[/url], please check the linearity property on this site).
The linearity suggests that if you can't hear a single tone at 30 or 32 kHz, you also cannot hear anything when cos(30kHz*t)+cos(32kHz*t) is "in the air".
That's really interesting actually. I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. Though at 2khz not 1khz
[a href="index.php?act=findpost&pid=362629"][{POST_SNAPBACK}][/a]
This is likely due to one of these things:
- improper anti alias filtering of your sound reproduction system
- non-linear distortions of your sound reproduction system (including clipping)
edit: typos
HTH,
Sebi
There's too much interpreting here. I've found a "third-party" argument which mirrors my original statement and I hope it is pretty understandable. However, I insist that the 2 KHz was wrong and 1 KHz would be the correct result (there's the same error in the quote that follows).
(...)
3. The third theory of recording at 96kS/s has to do with high frequency information during the mixing process.
Whenever two audio signals are combined, the result is a harmonic structure that produces additional frequencies as a combination of the summing of the two. If a note is played at 100Hz and another is played at 150Hz, there will be two additional frequencies produced as a result. One at 50 Hz (B-A) and the other at 250Hz (B+A). It is the first - the subtractive - that we shall discuss here. If two frequencies are produced at 30k and 45k, the effect will produce a harmonic overtone at 15k - well within the human hearing spectrum (at least for those of us in our 20's).
While this 15k signal would be picked up on a microphone in the room during a live classical performance, the issue has to do with multitrack performances where the music was tracked on separate microphones in isolated facilities. There would be no opportunity for this high frequency information to be mixed and reproduce these lower harmonic overtones because at 44.1k, the highest frequency recordable is at about 22k.
Theoretically, if we put two oscillators in a room - one at 30k and one at 45k, a microphone would pick up a 15k harmonic that we could hear. If we had the same oscillators in different rooms and recorded them both at 44.1k and mixed them we would not hear this phenomena. If, however, we recorded them at 96k (assuming the filters on the converters rolled off at 48k and not earlier) and mixed them we would once again hear this 15k overtone. This is supposedly an example of where recording and mixing at 96k can achieve different sonic characteristics even if the recording was to be reproduced on a 48k medium. The overtones discussed above would be produced in the mixing process and would endure the downsampling process.
Upon trying to put together this test I had a difficult time finding scientific enough equipment to produce and measure these frequencies. The theory is plausable, but the effect would be fairly minimal as these overtones are at very low amplitude. Regardless, the theory has a chance of holding water if a valid test can be done. I have heard of no such tests. Perhaps we could Roger to oblige us.
(...)
The text is at http://www.musicplayer.com/ubb/Forum2/HTML/001473.html (http://www.musicplayer.com/ubb/Forum2/HTML/001473.html)
There's a link there with info about some instruments harmonics. (There's life above 20 kilohertz (http://www.cco.caltech.edu/~boyk/spectra/spectra.htm)). By visual inspection, the article doesn't talk about the cited argument. However, it seems to have info on the amount of energy many instruments have above 20 KHz, which may be interesting for those who were considering that the effect (which could be a feature, not a bug) is unlistenable.
JJ again , at even greater length and detail, on high frequencies in audio and other matters generally. I see he even engaged 'RockFan' (who was on this thread earlier)
on pretty much the same issues he's brought up. Whihc makes me believe there's llittle point arguing them further.
http://www.skepticforum.com/viewtopic.php?...6385681595377ed (http://www.skepticforum.com/viewtopic.php?t=85&postdays=0&postorder=asc&start=40&sid=628df939781a1b4f96385681595377ed)
(For those unaware, Mr. Johnston, retired from Bell/ATT labs, was a co-inventor of MP3)
So what ? Just because there's suddenly a cos(1kHz*t) in a product you think you should be able to
I'm not thinking anything, I'm just repeating what I've just read.
hear this one? Do you realize that you can also interpret your formula as a 31 kHz sinusoid with an amplitude modulation at a rate of 1 kHz ? Truth is: It does not matter whether a product
representation contains usually audible signals. The first form (cos(32kHz)+cos(30kHz)) is the really interesting one. Why ? Because we've a bunch of filters in our inner ear (hair + hair cell) seperating
Remember that when we want to tune a guitar, if we have a reference (tuned) string, we tune the others by listening to the beating. And I believe that music professionals may also tune it by listening to the deviating mean frequency. Both things are at the right side of the equation, not the linear one. I can surely tune a guitar by listening to the beatings. If the beating is non-linear so what ? I listen to it.
Remember that when we want to tune a guitar, if we have a reference (tuned) string, we tune the others by listening to the beating. And I believe that music professionals may also tune it by listening to the deviating mean frequency. Both things are at the right side of the equation, not the linear one. I can surely tune a guitar by listening to the beatings. If the beating is non-linear so what ? I listen to it.
[a href="index.php?act=findpost&pid=363022"][{POST_SNAPBACK}][/a]
Yes that's correct. BUT the beating we hear is not a "tone" it is the actual tone going up and down in
amplitude. So if we can't hear the original tone, we also cannot hear it changing in amplitude. So you cannot hear a 32 kHz tone "beating".
In other words you can hear the following (the 1 kHz tone, at least):
(tone of 32 kHz) + (tone of 1 kHz)
but you can NOT hear the following (which is the case here):
(tone of 32 kHz) * (tone of 1 kHz)
(...)
Whenever two audio signals are combined, the result is a harmonic structure that produces additional frequencies as a combination of the summing of the two. If a note is played at 100Hz and another is played at 150Hz, there will be two additional frequencies produced as a result. One at 50 Hz (B-A) and the other at 250Hz (B+A). It is the first - the subtractive - that we shall discuss here. If two frequencies are produced at 30k and 45k, the effect will produce a harmonic overtone at 15k - well within the human hearing spectrum (at least for those of us in our 20's).
That is quite nonsense Im afraid hdante. The poster is roughly calculating beat frequencies between tones and confusing them with new tones that have the frequency of the beat. Mind a beat in a tone (or between tones) is definitely not a new tone and doesnt 'look' like a tone in a PCM record or sound like tone in RL. SebG put it like this:
you can also interpret your formula as a 31 kHz sinusoid with an amplitude modulation at a rate of 1 kHz
(the amplitude modulation can be seen in the pcm record as a 2kHz pulsing of the tone.)
Now sebG is one of those who doesnt like people talking about this stuff or attempting to enquire outside of contemporary academic language and references, hence this statement:
Fascinating? Hmm.... May seem so. But I think that some posts here are actually a bit misleading/misinformed.
Unfortunately hes not been inspired to actualy point out any particular misleads or misinforms -which are par for the course in open forum whatever the Tos. And I fear if him or his like do choose to elaborate it will be in a frustrated manner, so watch out for that
good luck'
Here is a little Matlab code for anyone who likes to try the experiment of adding two ultra sonic tones. Make sure your hardware supports a sampling rate of 96 kHz.
I heard a tone, by the way,
which is probably due to imperfections (non-linearities) in my hardware (the speakers for example).
Warning: be carefull with cranking up the volume, this is generally not good for hi-fi equipment.
% Ulta-sonic experiment
Fs = 96000; % Sampling frequency
f1 = 25000; % Frequency of 1st tone
f2 = 26000; % Frequency of 2nd tone
T = 5; % Length of signal
A = -6; % Amplitude (dB)
% Construct tones
N = round(T * Fs); % # samples
t = (0:N-1)' / Fs; % Time axis
B = 0.5 * 10^(A/20); % Amplitude per tone
p1 = B * sin(2*pi*f1*t);
p2 = B * sin(2*pi*f2*t);
p = p1 + p2; % Total tone
sound(p,Fs)
Here is a little Matlab code for anyone who likes to try the experiment of adding two ultra sonic tones. Make sure your hardware supports a sampling rate of 96 kHz.
I heard a tone, by the way, which is probably due to imperfections in my hardware (the speakers for example).
If its handy, could you try it at 24bits? If not ill try later.
Here is a little Matlab code for anyone who likes to try the experiment of adding two ultra sonic tones. Make sure your hardware supports a sampling rate of 96 kHz.
I heard a tone, by the way, which is probably due to imperfections in my hardware (the speakers for example).
If its handy, could you try it at 24bits? If not ill try later.
[a href="index.php?act=findpost&pid=363028"][{POST_SNAPBACK}][/a]
My Matlab version (6.1) doesn't support 24 bits for the 'sound' command. I could try to use the DAQ toolbox, though, but I don't really feel like it because I know we cannot hear the beating as a "tone" anyway
Here is a little Matlab code for anyone who likes to try the experiment of adding two ultra sonic tones. Make sure your hardware supports a sampling rate of 96 kHz.
I heard a tone, by the way, which is probably due to imperfections in my hardware (the speakers for example).
If its handy, could you try it at 24bits? If not ill try later.
[a href="index.php?act=findpost&pid=363028"][{POST_SNAPBACK}][/a]
My Matlab version (6.1) doesn't support 24 bits for the 'sound' command. I could try to use the DAQ toolbox, though, but I don't really feel like it because I know we cannot hear the beating as a "tone" anyway
[a href="index.php?act=findpost&pid=363030"][{POST_SNAPBACK}][/a]
Ok, ill try with audacity. I think it might be coming from rounding error...
Here is a little Matlab code for anyone who likes to try the experiment of adding two ultra sonic tones. Make sure your hardware supports a sampling rate of 96 kHz.
I heard a tone, by the way, which is probably due to imperfections in my hardware (the speakers for example).
If its handy, could you try it at 24bits? If not ill try later.
[a href="index.php?act=findpost&pid=363028"][{POST_SNAPBACK}][/a]
My Matlab version (6.1) doesn't support 24 bits for the 'sound' command. I could try to use the DAQ toolbox, though, but I don't really feel like it because I know we cannot hear the beating as a "tone" anyway
[a href="index.php?act=findpost&pid=363030"][{POST_SNAPBACK}][/a]
Ok, ill try with audacity. I think it might be coming from rounding error...
[a href="index.php?act=findpost&pid=363031"][{POST_SNAPBACK}][/a]
In that case I would expect a noise-like sound instead of something that sounds like a composition of several pure tones. But still you can try to do it in Audacity.
Unfortunately hes not been inspired to actualy point out any particular misleads or misinforms -which are par for the course in open forum whatever the Tos. And I fear if him or his like do choose to elaborate it will be in a frustrated manner, so watch out for that
[a href="index.php?act=findpost&pid=363025"][{POST_SNAPBACK}][/a]
I consider "if you play cos(30kHz*t)+cos(32kHz*t) you also may hear cos(1kHz) because cos(30kHz*t)+cos(32kHz*t) = 2*cos(31kHz)*cos(1kHz)" to be misinformation.
I consider "That's really interesting actually. I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. Though at 2khz not 1khz" to be misleading. Although that may be the case it may lead ppl to think that what hdante said is true whereas the reasons for mandel hearing something is likely related to non-linear distortions or improper D/A conversion.
I might have exaggerated things though.
Sebi
That is quite nonsense Im afraid hdante. The poster is roughly calculating beat frequencies between tones and confusing them with new tones that have the frequency of the beat. Mind a beat in
Ok, I think we reached the point. What the two links I posted here (about which I've realised my unfortunate belief that they were authoritative) are probably sugesting is that the beat is what is outside the hearing limits, while the tone is what is inside the hearing limits.
The beat, in the guitar tuning sense, is what is off the low hearing limit (eg 5Hz). The beat, in the ultra sonic sense is what is above the high hearing limit, (eg 31 KHz).
About the thing with Sebastian, I've just confirmed. At 440 Hz + 450 Hz I do listen both the 445 Hz tone and the beating.
Now you, ear engineers, say something about those.
Henrique Dante de Almeida
Points well made sebG - seems you are cool
-Tried 24 bits in audacity, the noise was even louder than with mandels 16bit file. Could be audacity, my laptop sound card or anything.
Seems the exact cause of this noise is intractable to me, but the well learned ones I think would agree that it is an artifact and shouldnt be heard in ideal circumstances.
many regards'
I consider "if you play cos(30kHz*t)+cos(32kHz*t) you also may hear cos(1kHz) because cos(30kHz*t)+cos(32kHz*t) = 2*cos(31kHz)*cos(1kHz)" to be misinformation.
I consider "That's really interesting actually. I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. Though at 2khz not 1khz" to be misleading. Although that may be the case it may lead ppl to think that what hdante said is true whereas the reasons for mandel hearing something is likely related to non-linear distortions or improper D/A conversion.
I might have exaggerated things though.
Sebi
[a href="index.php?act=findpost&pid=363033"][{POST_SNAPBACK}][/a]
If you read more carefully, I have already stated that I don't believe that the 2 KHz is correct. I understand that you are able to realise that I think it's a bug in his headphones or something.
Ok, ill try with audacity. I think it might be coming from rounding error...
[a href="index.php?act=findpost&pid=363031"][{POST_SNAPBACK}][/a]
As long as anyone here doesn't own a 96 KHz ready studio, we'll have a hard time doing this.
Ok, I think we reached the point. What the two links I posted here (about which I've realised my unfortunate belief that they were authoritative) are probably sugesting is that the beat is what is outside the hearing limits, while the tone is what is inside the hearing limits.
Sorry, I do not think those articles are very authoritative.
About the thing with Sebastian, I've just confirmed. At 440 Hz + 450 Hz I do listen both the 445 Hz tone and the beating.
And how exactly did you do that? If you add two tones of 440 Hz and 450 Hz you should not end up with a 445 Hz tone, since addition is
linear. You should hear a superposition of both tones (you should hear them both at once), plus a beating effect, which is again just a temporal change in
amplitude, not in frequency.
Ok, ill try with audacity. I think it might be coming from rounding error...
[a href="index.php?act=findpost&pid=363031"][{POST_SNAPBACK}][/a]
As long as anyone here doesn't own a 96 KHz ready studio, we'll have a hard time doing this.
[a href="index.php?act=findpost&pid=363039"][{POST_SNAPBACK}][/a]
Not really, high quality equipment shouldnt theoreticaly make the noise.
Its made in by laptop and desktops sound chipset alright.
Seems a good advertisement for not using high samplerate on standard equipment.
Just reminded' maybe take the samples offline, could burn out a few tweeters with passing noobs playing them louder and louder to see if they have the noise ?
And how exactly did you do that? If you add two tones of 440 Hz and 450 Hz you should not end up with a 445 Hz tone, since addition is linear. You should hear a superposition of both tones (you should hear them both at once), plus a beating effect, which is again just a temporal change in amplitude, not in frequency.
[a href="index.php?act=findpost&pid=363041"][{POST_SNAPBACK}][/a]
You may try it also.
And how exactly did you do that? If you add two tones of 440 Hz and 450 Hz you should not end up with a 445 Hz tone, since addition is linear. You should hear a superposition of both tones (you should hear them both at once), plus a beating effect, which is again just a temporal change in amplitude, not in frequency.
[a href="index.php?act=findpost&pid=363041"][{POST_SNAPBACK}][/a]
You may try it also.
[a href="index.php?act=findpost&pid=363043"][{POST_SNAPBACK}][/a]
I did and I absolutely don't have a clue what you're talking about. I can even give you the spectral results in a figure. If you add two tones with different frequencies, you end up with a superposition of those tones. That's all there is to it.
I did and I absolutely don't have a clue what you're talking about. I can even give you the spectral results in a figure. If you add two tones with different frequencies, you end up with a superposition of those tones. That's all there is to it.
[a href="index.php?act=findpost&pid=363045"][{POST_SNAPBACK}][/a]
I meant listen to it. Listen to a 440 Hz tone, listen to a 445 tone, listen to a 450 tone, an then listen to a 440+450 tone.
I did and I absolutely don't have a clue what you're talking about. I can even give you the spectral results in a figure. If you add two tones with different frequencies, you end up with a superposition of those tones. That's all there is to it.
[a href="index.php?act=findpost&pid=363045"][{POST_SNAPBACK}][/a]
I meant listen to it. Listen to a 440 Hz tone, listen to a 445 tone, listen to a 450 tone, an then listen to a 440+450 tone.
[a href="index.php?act=findpost&pid=363046"][{POST_SNAPBACK}][/a]
... yes? I still hear a 440 and a 450 tone together? It is hard to discriminate what the frequency of the highest note is, however, because of an effect called
masking in the frequency domain which is a psychoacoustic effect, quite different from the "beating" effect what you were talking about earlier.
... yes? I still hear a 440 and a 450 tone together? It is hard to discriminate what the frequency of the highest note is, however, because of an effect called masking in the frequency domain which is a psychoacoustic effect, quite different from the "beating" effect what you were talking about earlier.
[a href="index.php?act=findpost&pid=363047"][{POST_SNAPBACK}][/a]
I'm sorry, I hear the base frequency above 440 Hz. I may take that back, however. I may keep with the beat, only.
... yes? I still hear a 440 and a 450 tone together? It is hard to discriminate what the frequency of the highest note is, however, because of an effect called masking in the frequency domain which is a psychoacoustic effect, quite different from the "beating" effect what you were talking about earlier.
[a href="index.php?act=findpost&pid=363047"][{POST_SNAPBACK}][/a]
I'm sorry, I hear the base frequency above 440 Hz.
[a href="index.php?act=findpost&pid=363050"][{POST_SNAPBACK}][/a]
Ok. It is important to notice that all changes in frequency in this case
happen in your ears and brains not in the recording or playback process.
I may take that back, however. I may keep with the beat, only.
By now, do you believe me if I say that the "beat" is just a change in amplitude? And that we can only hear a tone "beat" if we can also hear the tone without the beating, i.e. if the frequency of the original tone is in the hearable range?
I meant listen to it. Listen to a 440 Hz tone, listen to a 445 tone, listen to a 450 tone, an then listen to a 440+450 tone.
[a href="index.php?act=findpost&pid=363046"][{POST_SNAPBACK}][/a]
... yes? I still hear a 440 and a 450 tone together?
I dont, I hear a single wowowowowowing tone. SebG formulated this should be 445 Hz, amplitude modulated sinusoidaly at 5Hz, which is ten wobbles -or semicycles, a second.
I think if you listen to progressively further apart tones, the wobbling becomes faster until its not perceived any more, that should be the change from hearing one beating tone to hearing two non interfering tones. As you move them closer - especialy while tuning guitar eg, you mostly hear one tone (main note +guitar overtones) that is increasing and decreasing in loudness - once a second or longer, keep tuning closer and closer until finaly the notes are close enough for them to properly resonate with each other through the body, and your in tune (the strings then sustain for much longer).
I meant listen to it. Listen to a 440 Hz tone, listen to a 445 tone, listen to a 450 tone, an then listen to a 440+450 tone.
[a href="index.php?act=findpost&pid=363046"][{POST_SNAPBACK}][/a]
... yes? I still hear a 440 and a 450 tone together?
I dont, I hear a single wowowowowowing tone. SebG formulated this should be 445 Hz, amplitude modulated sinusoidaly at 5Hz, which is ten wobbles -or semicycles, a second.
I only hear one tone if the difference is lower, say 445 and 450 Hz. Again, this is all perception, we
hear two tones but we
perceive them as one wobbling tone.
I meant listen to it. Listen to a 440 Hz tone, listen to a 445 tone, listen to a 450 tone, an then listen to a 440+450 tone.
[a href="index.php?act=findpost&pid=363046"][{POST_SNAPBACK}][/a]
... yes? I still hear a 440 and a 450 tone together?
I dont, I hear a single wowowowowowing tone. SebG formulated this should be 445 Hz, amplitude modulated sinusoidaly at 5Hz, which is ten wobbles -or semicycles, a second.
I only hear one tone if the difference is lower, say 445 and 450 Hz. Again, this is all perception, we hear two tones but we perceive them as one wobbling tone.
[a href="index.php?act=findpost&pid=363058"][{POST_SNAPBACK}][/a]
Does it follow we could be subjected to one wobbling tone and percieve it as two?
How could we know the difference if there were two beatles singing mating tones to each other, or one strange wobbling beatle all on its own
It's a matter of time/frequency resolution. If you do a short-window FFT on two closely spaced (in frequency domain) sines you'll see one wobbeling sine. With a long analysis window, you'll see 2 constant sines.
edit: you may try cool edit spectrum analyzer with varying FFT lengths
Sebi
By now, do you believe me if I say that the "beat" is just a change in amplitude? And that we can only hear a tone "beat" if we can also hear the tone without the beating, i.e. if the frequency of the original tone is in the hearable range?
[a href="index.php?act=findpost&pid=363051"][{POST_SNAPBACK}][/a]
I surely believe you when you say the beat is just a change in amplitude. And the requirement of the original tone is reasonable. But, as I say, I couldn't hear the original 440 Hz tone. It would be really nice if there's research on that. I think there's no more of this to be discussed (it's getting boring, ain't it ?).
By now, do you believe me if I say that the "beat" is just a change in amplitude? And that we can only hear a tone "beat" if we can also hear the tone without the beating, i.e. if the frequency of the original tone is in the hearable range?
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=363051")
I surely believe you when you say the beat is just a change in amplitude. And the requirement of the original tone is reasonable. But, as I say, I couldn't hear the original 440 Hz tone. It would be really nice if there's research on that.
[a href="index.php?act=findpost&pid=363072"][{POST_SNAPBACK}][/a]
There's lots of research on that. Just google on 'psychoacoustics' and you will find lots of information. For example:
[a href="http://en.wikipedia.org/wiki/Psychoacoustics]http://en.wikipedia.org/wiki/Psychoacoustics[/url]
I think there's no more of this to be discussed (it's getting boring, ain't it ?).
Ok.
There's too much interpreting here. I've found a "third-party" argument which mirrors my original statement and I hope it is pretty understandable. However, I insist that the 2 KHz was wrong and 1 KHz would be the correct result (there's the same error in the quote that follows).
(...)
[...]
Theoretically, if we put two oscillators in a room - one at 30k and one at 45k, a microphone would pick up a 15k harmonic that we could hear. If we had the same oscillators in different rooms and recorded them both at 44.1k and mixed them we would not hear this phenomena. If, however, we recorded them at 96k (assuming the filters on the converters rolled off at 48k and not earlier) and mixed them we would once again hear this 15k overtone. This is supposedly an example of where recording and mixing at 96k can achieve different sonic characteristics even if the recording was to be reproduced on a 48k medium. The overtones discussed above would be produced in the mixing process and would endure the downsampling process.
[...]
[a href="index.php?act=findpost&pid=363020"][{POST_SNAPBACK}][/a]
This guy seems to be agreeing with my opinion on this, except for the last point about reproducing on a 48k medium. From my experiments when you downsample to 48k or 44.1k the 'ultrasonic' tones disappear and the beat frequency disappears with it (as you'd expect)! So unless mastering engineers start making 44.1khz masters by going 96k > analog > 44.1k it is a moot point. Has anybody managed to resample and the beating remain?
Oh no !
(http://perso.numericable.fr/laguill2/smileys/delth.jpg)
At 96kHz sampling rate the filter effects at 20kHz are somewhat different than when sampling at 44.1kHz. This is a measurable result but whether or not many, or any, people can truly hear it does not seem to have been established by impartial studies. [{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=361980")
You can try it yourself : [a href="http://www.hydrogenaudio.org/forums/index.php?showtopic=17118]http://www.hydrogenaudio.org/forums/index....showtopic=17118[/url] . However, if the files sound different, it doesn't mean that the high resolution one is better. It can be worse because of the ultrasonic content introducing distortions in the audible range.
MOST audio players (as part of the DAC) use anti-alaising filters. The image is reflected back down from the Nyquist limit. That means it gets mixed into the music. [{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=362224")
No. This is the case in an ADC. In a DAC, it is the opposite, it's the content below the Nyquist frequency that is reflected above the Nyquist limit. They are the ultrasonic harmonics that create the rectangle steps if the output is unfiltered.
Anyway, since CD are recorded with anti-alias filters, there is nothing left above 22050 Hz that could be "reflected back" in a DAC.
So if I'm right a Anti-alias Filter is a low-pass filter (lets signals below a certain frequency go through) that cuts off every frequenzy above half the sample frequency to avoid alias-signals.
The Problem is that those filters can't be "perfect" but if you apply them at a higher sampling-frequency the result will be closer to perfect [a href="index.php?act=findpost&pid=362307"][{POST_SNAPBACK}][/a]
They can be perfect in practice, but we don't want them to be perfect. It would cause a lot of oscillations at 22050 Hz. It could cause distortions and even endager the tweeters.
Mathematically, cos(32KHz)+cos(30KHz) = 2*cos(31KHz)*cos(1KHz). Add the time in equation and you have a 1 KHz harmonic with a variable intensity of 2*cos(31KHz*t). [a href="index.php?act=findpost&pid=362598"][{POST_SNAPBACK}][/a]
Mathematically, a 32 kHz frequency is a trigonometric function of time, not of frequency !
We have
Cos(32000 x 2 x Pi x t) + Cos(30000 x 2 x Pi x t) = 2 x Cos(31000 x 2 x Pi x t) x Cos (1000 x 2 x Pi x t)
For any t real.
Thus the sum of a 30 kHz and a 32 kHz sine is a 31 kHz sine that has a variable amplitude. This amplitude is given by the absolute value of Cos(1000 x 2 x Pi x t), that reach zero 2000 times per second, and 1 2000 times per second too.
I just created a 96khz wav file with a 30k and a 32k tone and could hear a beat frequency as you say. [a href="index.php?act=findpost&pid=362629"][{POST_SNAPBACK}][/a]
You cannot count 2000 beats in your head in one second ! What you hear is not a beat frequency, but a 2 kHz sinewave. It is one of the two intermodulation products of the 30 kHz and 32 kHz sinewaves, that are 2 kHz and 62 kHz (f2 - f1 and f2 + f1). This is called intermodulation distortion, or IMD. It occurs in amplifiers and speakers, as well as harmonic distortion, whose sum, Total Harmonic Distortion, is noted THD.
there's no silly idea that some audiophile won't embrace. There are belt-driven CD players out there too. [a href="index.php?act=findpost&pid=362634"][{POST_SNAPBACK}][/a]
Don't forget the anti-gravitational foils, that cancel the space-time distortions caused by mechanic parts of the CD players, and prevent the sound to move from the speakers to your ears : [a href="http://www.belt.demon.co.uk/product/ref/ref.html]http://www.belt.demon.co.uk/product/ref/ref.html[/url] , or the interconnect cables shielded against gamma rays : http://www.hifi-cables.com/prod.html (http://www.hifi-cables.com/prod.html) !
But my favourite one is the golden screwdriver : http://www.wbtusa.com/wbt0481.html (http://www.wbtusa.com/wbt0481.html)
Here when mixing the 48 track recording down to stereo all in the digital domain there is a similar situation to the artificial one I created. Suppose that the first violins are playing a note with an overtone at 30khz and the second violins a note with a tone at 32khz, the close miking prevents the intermodulation distortion in the air from being recorded, however if all mixing is done at 96khz the distortion can reappear.[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=362673")
But who says that intermodulation occurs in the air ?
It occurs in amplifiers, for sure ( [a href="http://world.std.com/~griesngr/intermod.ppt]http://world.std.com/~griesngr/intermod.ppt[/url] ). Nika Aldrich and I both ran an experiment where two tones are played in one speaker, 12 kHz and 18 khz, for example (18 kHz in inaudible for me). I could hear the 4 kHz intermodulation at loud volumes (play this more than half a second long and your tweeters are fried ! ). Then I played the 12 kHz in the left speaker, and the 18 kHz in the right one. I moved in several locations in the room in order to check that I was not standing at some anti-resonant nodes, but I could not hear the 4 kHz intermodulation anymore. Just the 12 kHz tone. Thus the intermodulation is stronger in the amplifier than in the air.
If you play a high resolution recording, your amplifier will likely introduce more intermodulation than the air of the concert hall, and the result will be farther from the truth than a 44.1 kHz recording.
The question of how this beat period is generating an actual tone (...) on playback of the record, is in contention in this thread.[a href="index.php?act=findpost&pid=362893"][{POST_SNAPBACK}][/a]
Let's consider the example of an amplifier. Say that its voltage gain is 3.00 when no signal passes though it. Input 1.00 milliVolt, it outputs 3.00 milliVolts.
This won't be true when the amplifier is driven at high powers. When the output voltage rises, it slowly goes towards the clipping limit. The ascention is not perfectly linear. For example, if you input 1 Volt, it might output 2.99 Volts instead of 3 Volts. The Voltage gain is no more 3.00, but 2.99.
This will cause a periodic deformation of the additional signals fed into the amplifier.
When you feed 30 kHz into this amplifier, its gain will be 3.00 60000 times each second, but it will fall to 2.99 60000 times per second. The 32 kHz sine that you then add won't benefit from a constant gain, but from a variable gain that oscillates between 2.99 and 3.00 60000 times per second.
This is the cause of the apparition of a real 2 kHz sinewave, that should not exist in a perfect amplifier.Some believe that the beat period will be heard as a tone (walkers pace), or that the period will produced a tone in the air ~maybe a fluid dynamic phenomenon or something. [a href="index.php?act=findpost&pid=362893"][{POST_SNAPBACK}][/a]
According to David Griesinger's study, the amplifier is the main culprit. He did not measure significant amounts of intermodulation distortion in the tested tweeters.
We should however expect a lot of IMD in woofers, since their response will be drastically reduced when their diaphragm moves far from its resting position.
But David Griesinger's measurments were made 15 cm away from the tweeter. Thus we can't tell from these data if the intermodulation in significant through, say, 15 meters of air. All we can say is that it will depend on the sound pressure level. Intermodulation will appear when the sound compresses the air enough for its acoustic properties to be affected. Like it appears in the amplifier when the voltage is high enough to affect its gain.
When the ultrasonic test signal is lowpassed to remove theoreticaly inaudible frequencies from the record, the 'sound' of the beat frequency disappears as well.[a href="index.php?act=findpost&pid=362893"][{POST_SNAPBACK}][/a]
...unless it has suffered from intermodulation before the lowpass. Then a real 2 kHz sinewave have appeared, and is still there after the lowpass.
I guess the best way to proof that it has influence on the realworld-sound is to produce such a signal in the realworld. If we can't it's not a proof that it's impossible, but if we can...
What would be needed to do so?
[a href="index.php?act=findpost&pid=362902"][{POST_SNAPBACK}][/a]
A microphone and a mic amplifier more linear than air
And how exactly did you do that? If you add two tones of 440 Hz and 450 Hz you should not end up with a 445 Hz tone, since addition is linear. You should hear a superposition of both tones (you should hear them both at once), plus a beating effect, which is again just a temporal change in amplitude, not in frequency.
[a href="index.php?act=findpost&pid=363041"][{POST_SNAPBACK}][/a]
Wrong.
Cos(440 x 2 x Pi x t) + Cos(450 x 2 x Pi x t) = 2 x Cos(445 x 2 x Pi x t) x Cos (5 x 2 x Pi x t)
For any t real.
In english : the sum of a 440 Hz sine and 450 sine is a 445 Hz sine whose amplitude reaches zero 10 times per second.
This is the same deal as joint stereo versus stereo. You can write it Left + Right, or Mid + Side, this is exactly the same signal. Just two ways of writing it.
Here, you can write 440 Hz and 450 Hz, or 445 Hz variating, this is exactly the same signal. Two ways of writing it.
You can hear the frequencies together or separately, like you can hear an instrument note as a whole, or listen to its harmonics separately.
So you
really get a 445 Hz oscillation. But you cannot say that its frequency is 445 Hz, because by definition, a
frequency is a signal whose amplitude never changes. Its spectrum is made from 440 Hz and 450 Hz components.
Filter it at 445 Hz, and only the 440 Hz part remains, without the beating.
Another way of seeing it would be a 10 Hz signal whose fundamental's amplitude is completely null. The second harmonic is silent too. So are the next harmonics... Only the 44th and 45th harmonics are not silent.
I think i can see that happening in the amp - that the wavering current demand for beating 'hybrid' tones is not perfectly satisfied - producing the new tones.
True to form sir, if there are any left believing "mika@music player"'s ~midlevel acoustic misunderstanding in this thread.. theyve only got themselves to blame. I can see you earn your money
But my favourite one is the golden screwdriver http://www.wbtusa.com/wbt0481.html (http://www.wbtusa.com/wbt0481.html)
Id give it a try though, good for mashing into difficult slots maybe
For an example of ultrasonic sound producing audible sounds, try http://www.atcsd.com/pdf/HSSWHTPAPERRevE.pdf (http://www.atcsd.com/pdf/HSSWHTPAPERRevE.pdf) and http://www.atcsd.com/hss.html (http://www.atcsd.com/hss.html)
Personally don't like the idea of high amplitude ultrasonic sound being shot at me without plenty of medical research showing that high SPL ultrasound really is harmless and that it won't cause someone's hearing aid to burst into flame or explode.
As for sampling rates, 44.1 kHz's fine with me since a decent microphone should record anything that my ear would if it were in the same general location and I most definately can't hear beyond 20 kHz (and that's being generous. )
And how exactly did you do that? If you add two tones of 440 Hz and 450 Hz you should not end up with a 445 Hz tone, since addition is linear. You should hear a superposition of both tones (you should hear them both at once), plus a beating effect, which is again just a temporal change in amplitude, not in frequency.
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Wrong.
So you really get a 445 Hz oscillation. But you cannot say that its frequency is 445 Hz, because by definition, a frequency is a signal whose amplitude never changes. Its spectrum is made from 440 Hz and 450 Hz components.
How can you first say I'm wrong and then say exactly the same thing as I did in other words
Anyway, it really depends on how far the tones are apart in the frequency domain. If you add two tones of 200 and 2000 Hz you don't really hear a 1100 Hz tone beating. For 440 Hz and 450 Hz I really hear two seperate tones and for 445 Hz and 450 Hz I don't anymore.
Anyway, it really depends on how far the tones are apart in the frequency domain. If you add two tones of 200 and 2000 Hz you don't really hear a 1100 Hz tone beating. For 440 Hz and 450 Hz I really hear two seperate tones and for 445 Hz and 450 Hz I don't anymore.
Just to clarify something real quick here...
The beat frequency is the difference or the addition of the other two frequencies. For 2000 and 200, the beat freqs will be 1800 and 2200. In the other example that has been presented, with 30k and 32k, 2k is indeed the correct beat frequency (not 1k as someone expected) with another beat frequency at 62k.
If a signal mixer is being used, then you also get those real freqs, plus all of their sums and differences, plus all of their harmonics. These complications are why radio devices interfere with each other, even if they are in completly different frequency ranges.