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Topic: Sampling rates higher than 44.1Khz? (Read 69025 times) previous topic - next topic
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Sampling rates higher than 44.1Khz?

Reply #25
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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

Henrique Dante de Almeida
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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. 


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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]

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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.

Sampling rates higher than 44.1Khz?

Reply #26
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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.

Sampling rates higher than 44.1Khz?

Reply #27
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I looked up your posts .....   
[a href="index.php?act=findpost&pid=362634"][{POST_SNAPBACK}][/a]


I'm flattered.

R.

Sampling rates higher than 44.1Khz?

Reply #28
You bore me.

The last word is yours, please do savour it.

R.

Sampling rates higher than 44.1Khz?

Reply #29
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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.

Sampling rates higher than 44.1Khz?

Reply #30
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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.

Sampling rates higher than 44.1Khz?

Reply #31
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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]


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'

Sampling rates higher than 44.1Khz?

Reply #32
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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).

Sampling rates higher than 44.1Khz?

Reply #33
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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

Sampling rates higher than 44.1Khz?

Reply #34
http://www.ioforums.net/forums/view_topic....rum_id=1&page=1



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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


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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!




Sampling rates higher than 44.1Khz?

Reply #35
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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

Sampling rates higher than 44.1Khz?

Reply #36
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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]


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

Sampling rates higher than 44.1Khz?

Reply #37
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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
[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

Sampling rates higher than 44.1Khz?

Reply #38
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Quote

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
[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):

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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.

Sampling rates higher than 44.1Khz?

Reply #39
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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
[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'
no conscience > no custom

Sampling rates higher than 44.1Khz?

Reply #40
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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
[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)

Sampling rates higher than 44.1Khz?

Reply #41
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Quote


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
[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

Sampling rates higher than 44.1Khz?

Reply #42
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?)

Sampling rates higher than 44.1Khz?

Reply #43
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.

 

Sampling rates higher than 44.1Khz?

Reply #44
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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?
no conscience > no custom

Sampling rates higher than 44.1Khz?

Reply #45
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?

Sampling rates higher than 44.1Khz?

Reply #46
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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.

Sampling rates higher than 44.1Khz?

Reply #47
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-
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.

Sampling rates higher than 44.1Khz?

Reply #48
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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..

Sampling rates higher than 44.1Khz?

Reply #49
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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.

Quote
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
no conscience > no custom