Mixing Question - Levels

[jdhogg]

Casey

As for nyquist.....back to digital theory school for you I'm afraid, even though your description is quite typical.

What it should state is "2 x max frequency is the MINIMUM sampling rate to accurately record a signal"

......I'll leave it there and keep the worms in the can.

As for noise.......Noise is any unwanted signal and occurs at various frequencies.......so that includes the snare mic picking up the drummer breaking wind when hes pushing the beat a little too hard.

[mazz]

Unless his wind breaking is in tune and adds a nice midrange texture to the piece.

[Casey H]

Nyquist's theorem says that A/D sampling is best done at at least 2x the highest frequency component of the signal you are sampling.

JD:
Isn't "at least" the same as MINIMUM?

Casey

PS Since folks can't seem to have a discussion without it taking on a nasty tone... I'm out.

[gtrmann]

PS Since folks can't seem to have a discussion without it taking on a nasty tone... I'm out.

I am sorry you feel that way, If I offended you, it was not intentional... I APOLOGIZE...!!!!!!

[gtrmann]

Nyquist's theorem says that A/D sampling is best done at at least 2x the highest frequency component of the signal you are sampling.

Completely true...... component is in reference to the harmonics of the complex waveform.....

[gtrmann]

Harry Nyquist..... A. amazing electrical engineer. He Worked at AT&T - Bell labs from 1917 to 1954..... He developed the Nyquist Theorem in 1924 while working on telegraph transmission lines. He determined the analog bandwith of the telegraph line ( a pair of twisted copper wires ) had to be twice the frequency of the telegraph data pulse frequency. dot dot dash dash dot dot............digital transmission.......pulse width modulation......Telegraph

[ernstinen]

Alright, I thought a new thread would be started on this analog vs. digital debate, but since not, here goes!

I'm not going to talk about bit rates --- that's been covered well enough!

But in an article by Bob Moog (not exact quotes): The 44.1 sample rate goes up to 22.05 khz. That is unacceptable, since a quality analog deck goes up to 40-50 khz. Analog is superior for that reason. Maybe most people can not hear above 22 khz, but think of all the musical overtones and information there are between 22k and 40k. An enormous amount. That is where all the "air" is in recordings....

I personally have stated this before, but the "average" person's ears are not a musician's ears. When I used to master analog, I would rent a Massenbug analog EQ, which would go up to 26k. *I* can hear that high, even after years of playing in loud bands. It may not be actually hearing those frequencies, but it is more "feeling" the frequencies. We've talked about this in length before, so don't flame me on this!

I'd like to hear other's opinions, especially those who agree with me!

Best Regards,

Ern

[jdhogg]

Nyquist's theorem says that A/D sampling is best done at at least 2x the highest frequency component of the signal you are sampling.

JD:
Isn't "at least" the same as MINIMUM?

Casey

PS Since folks can't seem to have a discussion without it taking on a nasty tone... I'm out.

Casey are referring to me?....I was writing that with a smile on my face and have no animosity to you whatsoever....you seem a very decent person. That is the problem with forums, if I had said that to you over a beer you would know I was only having a laugh, unfortunately you could not see my smile....sorry if you took it the wrong way.

Not wishing to upset you but your statement is not "completely true" because you use the words "best done". Maybe I am being picky on your statement and you do indeed understand the concept 100%. Sorry if that is the case.

For any that are interested........maybe none

Nyquist was describing how a sine wave could be captured if its peaks and troughs were in phase with the sampling points....

(smiling )
Heres what some people dont realize.........nyquist is basicaly saying for a sinewave, sample at x2 the frequency AND assumes that you sample the peak and trough of the sine wave........BUT what people forget is that assumes that the peaks and troughs are always perfectly lined up at the sampling frequency points....which of course they are not in an analogue system....what happens eg to a 20kHz wave component being sampled at 40kHz(x2) that appears delayed 1/4 cycle/90 phase from the sampling points?...........at the sampling points the wave is now at the crossing point ie 0.......therefore the sample sees nothing and records nothing........ so as eg 20k component signals appear effectively at random phase there is an error for any signal that is not captured at its peak..... .This loss of detail is why cymbals imho never sound fantastic.

point being x2 is the multiple to correctly sample an in sync signal and in sync only happens 1 in 90 times in the real world.


.....ok im out

[gtrmann]

Harry Nyquest passed away in 1976. The math function that is used is the mathematical duality of original Nyquest Theroum of 1926 for telegraph lines .... kinda like saying ........if this holds to be true, than this must also......

A square wave is a repetitive analog signal that has the maximum harmonic content possible. In order to reproduce a square wave you have to have 10 times the bandwidth in relation to the fundamental, or 20 samples...

44.1K ( sampling rate ) / 20 samples = 2.205 Khz highest frequency you can accurately reproduce..

A sample rate of 44.1 K cannot faithfully reproduce a square wave with a fundamental above 2.205 Khz. If you increase the frequency of the square wave above 2.205 K, the square wave starts to distort ( distortion = not accurate representation of the original ). The higher you increase frequency of the square wave, the less accurate the reproduction. By the time you get to 11 K the reproduction accuracy is really poor.

Of course the fundamental frequency of most instruments doesn't go far above 4K, most don't even get that high.... So anything above that is just harmonics anyway.........

The inaccurate reproduction of the harmonic content, ........ that digital sound.......... people have gotten used to it much as they got used to the clicks and pops of old vinyl records.....

The amount of memory available was the limiting factor when this technology was developed in the early 80s.....

We will be saying soon, remember when we used to play 44.1 16 bit music.....YUCK......LOL

If you want more accurate reproduction of your harmonic content go to 192K 24 bit...

Until then write better songs......

Long Live Harry Nyquest.........

[Casey H]

If Harry knew you were misspelling his name, he'd be rolling over in his grave.... Probably at 2x the highest.... well, whatever...

It's "Nyquist", not "Nyquest"....

Casey

(PS Note the shiny, happy, laughing smilies)