Sampling rates higher than 44.1Khz? Options

[Grand Dizzy]

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?

This post has been edited by Grand Dizzy: Feb 5 2006, 01:11

[AndyH-ha]

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

[Grand Dizzy]

Duhh... sorry, that all went completely over my head!

[krabapple]

Duhh... sorry, that all went completely over my head!

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.

[hdante]

Duhh... sorry, that all went completely over my head!

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.

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

[SebastianG]

[...] 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. [...]

Why may I hear something like that ?

Sebi

[hdante]

[...] 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. [...]

Why may I hear something like that ?

Sebi

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 :-/.

http://www.dvdsoftwareguide.com/all-about-dvd-4-guide.html

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.

[mandel]

[...] 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. [...]

Why may I hear something like that ?

Sebi

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 :-/.

http://www.dvdsoftwareguide.com/all-about-dvd-4-guide.html

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.

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!

This post has been edited by mandel: Feb 7 2006, 23:56

[hdante]

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!

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

[mandel]

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!

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

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
http://www.arafel.org.uk/~mandel/ha/ultratones.flac

Edit: Added a bigger warning

This post has been edited by mandel: Feb 9 2006, 22:30

[WmAx]

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

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

This post has been edited by WmAx: Feb 8 2006, 01:18

[mandel]

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

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

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.

This post has been edited by mandel: Feb 8 2006, 01:48

[WmAx]

-
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.

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.

This post has been edited by WmAx: Feb 9 2006, 15:38

[bug80]

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!

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..

This post has been edited by bug80: Feb 9 2006, 10:48