Sampling

Started by June 24, 2013
Somebody just told me something I cannot accept. That the usual sampling theorem
doesn't work too well when there are fast moving transients in a signal. To me this
means that you are just not sampling fast enough ie the higher harmonics are not
being captured. Is this one of those Hi-Fi myths that when you sample an audio
signal at say 48kHz that it won't catch the cymbals in an orchestra or band?
On Mon, 24 Jun 2013 13:47:58 -0700 (PDT), gyansorova@gmail.com wrote:

>Somebody just told me something I cannot accept. That the usual sampling th= >eorem doesn't work too well when there are fast moving transients in a sign= >al. To me this means that you are just not sampling fast enough ie the high= >er harmonics are not being captured. Is this one of those Hi-Fi myths that = >when you sample an audio signal at say 48kHz that it won't catch the cymbal= >s in an orchestra or band?
A "fast moving transient", if I understand correctly, implies high frequencies. The usual sampling theorem takes into account the spectral content of the signal, so if there are "high" frequencies due to transients they either need to be removed by the anti-alias filter or the sampling rate adjusted to include them. There's no problem with the sampling theorem, just some people's understanding of what "fast moving transient" means in the context of the bandlimiting requirements of the theorem. Eric Jacobsen Anchor Hill Communications http://www.anchorhill.com
On 6/24/13 1:47 PM, gyansorova@gmail.com wrote:
> Somebody just told me something I cannot accept. That the usual sampling theorem
doesn't work too well when there are fast moving transients in a signal. To me this means that you are just not sampling fast enough ie the higher harmonics are not being captured. Is this one of those Hi-Fi myths that when you sample an audio signal at say 48kHz that it won't catch the cymbals in an orchestra or band? probably. there is content above 24 kHz coming outa a cymbal. the question is whether or not we can hear that content above 24 kHz (or even 20 kHz). an experiment that someone with a really high quality audio recording environment (like a modern version of Pro Tools) and something to manipulate data analytically (say MATLAB) would be to *record* the cymbal or music with a cymbal and all sorts of other percussion (like an orchestra) directly with very expensive B&K microphones and mic preamps ( http://www.bksv.com/products/transducers/acoustic/microphones.aspx ) sample at 192 kHz (4 times the normal), save the data as a WAV file and open it into a MATLAB program (with audioread or wavread). you can confirm, by windowing long segments and FFTing, that there is content in there above 24 kHz (perhaps as high as 95.9 kHz). then, using a nice long FIR optimally designed using firpm or firls, obliterate all of the energy above 22 kHz (say, down by 120 dB) while leaving the content below 20 kHz unchanged (say the passband ripple is 0.01 dB for most, nearly all of it below 20 kHz). then play that sound back at the same 192 kHz and compare to the original 192 kHz recording. if someone suggests they can hear the difference between the brickwall LPFed music and the original, it's time to do some blind or double-blind testing where they have to identify changed recordings among true pairs (where one is LPFed and the other not) and false pairs (where both are identical). run this test for several pairs of audio recordings. for each subject (with golden ears), subtract the number of false positive from true positives and do the same with the false negatives and true negatives. add the two differences together which would be the net number of trials or "guesses" that the listener got right (minus the number he/she got wrong). let there be an equal number of pairs with true differences as with pairs with no difference (and the listener doesn't know which is which and, in fact, must tell us which is which), if the listener gets a net score well above zero, it means that he/she can really hear the difference (or is a very lucky guesser). if it's around zero, it means he/she is guessing (and any claims that he/she can hear the difference is disproven) and if it's significantly below zero, either the listener is very unlucky in his/her guessing or perhaps the listener *can* hear a difference and is deliberately answering the question exactly wrong. i really don't think there are many people, if any at all, that can hear the difference between this full bandwidth audio (with bandwidth up to 96 kHz) and the same audio where everything above 22 kHz is fully removed (and *only* that audio above 22 kHz is removed). i would be a skeptic, but what i described above is how i would decide. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge."
>Somebody just told me something I cannot accept. That the usual sampling
th=
>eorem doesn't work too well when there are fast moving transients in a
sign=
>al. To me this means that you are just not sampling fast enough ie the
high=
>er harmonics are not being captured. Is this one of those Hi-Fi myths that
=
>when you sample an audio signal at say 48kHz that it won't catch the
cymbal=
>s in an orchestra or band? >
I'm sure he/she meant to say that 'sampling' doesn't work too well when there are fast moving transients, and that the sampling theorem does a good job of explaining why. _____________________________ Posted through www.DSPRelated.com
On Mon, 24 Jun 2013 17:39:18 -0500, "dszabo" <62466@dsprelated> wrote:

>>Somebody just told me something I cannot accept. That the usual sampling >th= >>eorem doesn't work too well when there are fast moving transients in a >sign= >>al. To me this means that you are just not sampling fast enough ie the >high= >>er harmonics are not being captured. Is this one of those Hi-Fi myths that >= >>when you sample an audio signal at say 48kHz that it won't catch the >cymbal= >>s in an orchestra or band? >> > >I'm sure he/she meant to say that 'sampling' doesn't work too well when >there are fast moving transients, and that the sampling theorem does a good >job of explaining why. > >_____________________________ >Posted through www.DSPRelated.com
Nicely put. Eric Jacobsen Anchor Hill Communications http://www.anchorhill.com
On Mon, 24 Jun 2013 14:24:10 -0700, robert bristow-johnson wrote:

> On 6/24/13 1:47 PM, gyansorova@gmail.com wrote: >> Somebody just told me something I cannot accept. That the usual >> sampling theorem doesn't work too well when there are fast moving >> transients in a signal. To me this means that you are just not sampling >> fast enough ie the higher harmonics are not being captured. Is this one >> of those Hi-Fi myths that when you sample an audio signal at say 48kHz >> that it won't catch the cymbals in an orchestra or band? > > probably. there is content above 24 kHz coming outa a cymbal. the > question is whether or not we can hear that content above 24 kHz (or > even 20 kHz). > > an experiment that someone with a really high quality audio recording > environment (like a modern version of Pro Tools) and something to > manipulate data analytically (say MATLAB) would be to *record* the > cymbal or music with a cymbal and all sorts of other percussion (like an > orchestra) directly with very expensive B&K microphones and mic preamps > ( http://www.bksv.com/products/transducers/acoustic/microphones.aspx ) > sample at 192 kHz (4 times the normal), save the data as a WAV file and > open it into a MATLAB program (with audioread or wavread). > > you can confirm, by windowing long segments and FFTing, that there is > content in there above 24 kHz (perhaps as high as 95.9 kHz). then, > using a nice long FIR optimally designed using firpm or firls, > obliterate all of the energy above 22 kHz (say, down by 120 dB) while > leaving the content below 20 kHz unchanged (say the passband ripple is > 0.01 dB for most, nearly all of it below 20 kHz). > > then play that sound back at the same 192 kHz and compare to the > original 192 kHz recording. if someone suggests they can hear the > difference between the brickwall LPFed music and the original, it's time > to do some blind or double-blind testing where they have to identify > changed recordings among true pairs (where one is LPFed and the other > not) and false pairs (where both are identical). > > run this test for several pairs of audio recordings. for each subject > (with golden ears), subtract the number of false positive from true > positives and do the same with the false negatives and true negatives. > add the two differences together which would be the net number of trials > or "guesses" that the listener got right (minus the number he/she got > wrong). let there be an equal number of pairs with true differences as > with pairs with no difference (and the listener doesn't know which is > which and, in fact, must tell us which is which), if the listener gets a > net score well above zero, it means that he/she can really hear the > difference (or is a very lucky guesser). if it's around zero, it means > he/she is guessing (and any claims that he/she can hear the difference > is disproven) and if it's significantly below zero, either the listener > is very unlucky in his/her guessing or perhaps the listener *can* hear a > difference and is deliberately answering the question exactly wrong. > > i really don't think there are many people, if any at all, that can hear > the difference between this full bandwidth audio (with bandwidth up to > 96 kHz) and the same audio where everything above 22 kHz is fully > removed (and *only* that audio above 22 kHz is removed). i would be a > skeptic, but what i described above is how i would decide.
Double blind testing of audio equipment is bunk: http://www.avguide.com/forums/blind-listening-tests-are-flawed-editorial And be sure to dip your power cords in LN2: http://www.jenalabs.com/ac-products/powercords.html -- Tim Wescott Wescott Design Services http://www.wescottdesign.com
Tim Wescott <tim@seemywebsite.really> writes:
> [...] > And be sure to dip your power cords in LN2: > http://www.jenalabs.com/ac-products/powercords.html
Only $500! -- Randy Yates Digital Signal Labs http://www.digitalsignallabs.com
On 6/24/13 4:46 PM, Tim Wescott wrote:
> > > And be sure to dip your power cords in LN2: > http://www.jenalabs.com/ac-products/powercords.html >
how do my power chords (like a gratifying slam on E major) make use of the natural logarithm of 2? it's gotta be something related to octaves. -- r b-j rbj@audioimagination.com "Imagination is more important than knowledge."
I've noticed that when I walk into a building where music is playing, I can always
tell instantly when it's live music, and the sound of real cymbals is what gives it
away. But I don't think this is due to wider bandwidth. My theory is that the
spatial radiation pattern of the cymbals is different from all the other sound
sources. If the cymbal were played back over speakers it would get the same
radiation pattern as everything else, and you would hear the difference.
Am 25.06.13 06:33, schrieb radams2000@gmail.com:
> I've noticed that when I walk into a building where music is playing, > I can always tell instantly when it's live music
I have this impression, too, that I can distinguish live music from playback. , and the sound of
> real cymbals is what gives it away. But I don't think this is due to > wider bandwidth. My theory is that the spatial radiation pattern of > the cymbals is different from all the other sound sources. If the > cymbal were played back over speakers it would get the same radiation > pattern as everything else, and you would hear the difference. >
My explanation is the dynamic range compression that every commercial recording does. When you ever tried recording something yourself and then played it back, you wonder why it sounds soooo thin in comparison to a studio recording. The biggest answer is dynamic compression, nicely explained here: http://www.youtube.com/watch?v=3Gmex_4hreQ I must admit that I have also applied (moderate) dynamic compression to my own home-made recording, because otherwise it sounded too quiet. I think that the usual cheap playback equipment simply doesn't have enough power to realistically reproduce the dynamic range of real world signals. It's like tone mapping HDR images: without it, we can't see anything, but if done improperly, it sticks out like a sore thumb (and people call this "the HDR look"), and we are *always* able to distinguish the real sun from an HDR image. Christian