OOT What an effort on the ergonomics !A few years ago I had to help friends putting together a display of about 120 photos for a show. I had to learn photoshop very quickly, and it was actually fun seeing the algorithms I learned in the late 70's being used in the software of today.
May I suggest you to find the last installable version of Lightroom from the same company ? (they want to hire its use, remotely, on a proprietary server version to increase their giants profits).
Very intuitive and powerful enough.
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...phase relative to an exact 1kHz...
A 1kHz test tone on a recording?
OOT What an effort on the ergonomics !
May I suggest you to find the last installable version of Lightroom from the same company ? (they want to hire its use, remotely, on a proprietary server version to increase their profits).
Very intuitive and powerful enough.
Alas, I only do it on an as need basis. Once I taught my friends how to use the more advanced features, subsequent shows they were able to handle easily.
I use this principle at work. It's easy enough to just play McGyver by coming in as the hero, it is more difficult to teach the up and comings critical thinking skills so they don't need me. I do not worry about ego, after all, I had kids...😀
Jn
People can only agree on auditory perception when they listen to the same thing, a lot of testing has been done BTW, people do largely agree on thresholds and mechanisms of perception that we all share, ie, psychoacoustics.The limit is easier to determine, here, because it seems easier for people to agree on visual perception than auditory: The useful limit is that after which an increase in the power of separation of the sensor brings no more improvement to the one, limited, of the lens (in constant progress).
A little more than twice this one, thank you Mr. Nyquist.
And anti aliasing filters are now useless. Winner play again, twice the bet.
In audio, good microphones go up to 40,000 Hz and super tweeters too.
On the level point of view, by the signal/noise ratio of the best mics preamps.
The acoustic noise is not to be considered, as it is part of what we want (or not) to record.
Everything else is a matter of personal perception.
@jneutron,
Nordmark's test signal stream was jittered not dithered...... 🙂
@Hans Polak,
Not meant offensive, but acting as "Mr. Teflon" is obviously not that easy in real forum life......
Even around 1985 Ericsson/Rifa's hybrid filter modules designed for use as antialiasing filters in digital recorders or other recording equipment and as reconstruction filter in CD-players only offered ~30 dB attenuation at 22.05 kHz.
At least a newer modell in 1987 was around 70 dB.
If you mean "cannot convince against...." in the sense that it could not get corrobation for delivering better sound than the PCM version, then you're right, but the "Detmold test" was an ABX only looking for difference.
4 out of 110 participants got were able to get results with very low probabilities, which means were not compatible to the hypothesis of random guessing.
But AFAIR they did not offer any preference decision in the questionaires.
Although all other results were below the significance level critierion choosen by the authors, looking at the data overall seems to bring further evidence for a difference under these test conditions.
Of course, you're right, but Shannon already pointed out in this paper from 1949, that it is "difficult" to fullfill both conditions (i.e. all frequencies in the band and all signals restricted to finite time span).
Nordmark's test signal stream was jittered not dithered...... 🙂
@Hans Polak,
Not meant offensive, but acting as "Mr. Teflon" is obviously not that easy in real forum life......
One very important thing before digitizing is the BW restriction to avoid aliasing, so as a reaction to the article “What Nyquist did’n’t say” the following:
When sampling at 44.1kHz it is almost impossible to create an analogue filter that goes from 0dB at 20Khz to ca. -100dB at 22.05Khz.
That is why some are fighting against 44.1/16 as a decent CD format.
Even around 1985 Ericsson/Rifa's hybrid filter modules designed for use as antialiasing filters in digital recorders or other recording equipment and as reconstruction filter in CD-players only offered ~30 dB attenuation at 22.05 kHz.
At least a newer modell in 1987 was around 70 dB.
But DSD once called "the best invention since sliced bread" making all other formats obsolete, cannot convince in a large test against 176.4/24.
See “Perceptual Discrimination of Digital Audio Coding Formats”, performed by the University of Music of Detmold, Germany, also published as AES paper 6086.
If you mean "cannot convince against...." in the sense that it could not get corrobation for delivering better sound than the PCM version, then you're right, but the "Detmold test" was an ABX only looking for difference.
4 out of 110 participants got were able to get results with very low probabilities, which means were not compatible to the hypothesis of random guessing.
But AFAIR they did not offer any preference decision in the questionaires.
Although all other results were below the significance level critierion choosen by the authors, looking at the data overall seems to bring further evidence for a difference under these test conditions.
The other point mentioned a few times in this thread is that Nyquist/Shanon is only true for continuous signals and not for short events.
I have tried to debunk this mythe in LA vol 8 page 19 as absolutely not true.
Of course, you're right, but Shannon already pointed out in this paper from 1949, that it is "difficult" to fullfill both conditions (i.e. all frequencies in the band and all signals restricted to finite time span).
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No. Sampling exactly at zero crossing can happen with other integer multiply of signal frequency, 3 X and greater do not violate Nyquist. Somehow you are also spreading nonsense.Sampling at zero crossings means sampling at exactly twice the frequency. This violates Nyquist, since the theorem says "greater" not "greater or equal"...
Yep, nothing is more satisfying than seeing our children and assistants succeed better than we did. It helps to forgive our character flaws and, who knows, leave a trace?It's easy enough to just play McGyver by coming in as the hero, it is more difficult to teach the up and comings critical thinking skills so they don't need me. I do not worry about ego, after all, I had kids...😀
About photography (thanks to your kind words) I wonder the miracle of the lithium niobate.
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How many points are used in a typical cd playback system using red book?
Jn
Not sure what you mean. We used the same strategy, 10ms time windows, to plot frequency response for a 1kHz to (48,50)kHz sweep. In this case the sampling frequency was 96k or 192k.
No. Sampling exactly at zero crossing can happen with other integer multiply of signal frequency, 3 X and greater do not violate Nyquist. Somehow you are also spreading nonsense.
I mean, two samples per period, at zero crossing. Sorry for the misunderstanding, you ought to follow the entire discussion some 20 pages back.
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A 1kHz test tone on a recording?
The FFT is done in double precision math bin 10 on a 480 point FFT (48k sampling) is exactly 1kHz. The point is the phase error is continuous and not quantized.
The plot shows the slow phase drift due to the TT not being exactly 33 1/3 rpm, the degrees/sec gives the instantaneous frequency with very high resolution. The three humps are the LP eccentricity, you can do the geometry to see how small these effects are.
No. Sampling exactly at zero crossing can happen with other integer multiply of signal frequency, 3 X and greater do not violate Nyquist. Somehow you are also spreading nonsense.
Think it over, please. Where will be the samples that do not catch zeros? If signal frequency is below half Fs? Like 12kHz signal sampled at 48kHz?
Why are people trying to prove Nyquist doesn't work when it's been shown by measurements that it does, even in ways that are difficult to understand? Perhaps that is the real issue, people can't believe it 🙂
Why are people trying to prove Nyquist doesn't work when it's been shown by measurements that it does, even in ways that are difficult to understand? Perhaps that is the real issue, people can't believe it 🙂
Nyquist theorem is not the most intuitive thing in signal processing. There are some great threads on the the HA forum (otherwise much more technical than this one) that clearly shows Nyquist becoming quickly a game of bruised egos.
The FFT is done in double precision math bin 10 on a 480 point FFT (48k sampling) is exactly 1kHz. The point is the phase error is continuous and not quantized.
Okay, thank you.
What I was trying to get out of JN is what he is thinking in terms of and why he keeps talking about what the sample points look like on graph paper.
One example he might be thinking of is if one wants a 20kHz bandwidth FFT with .1Hz bins, then it takes a lot of points.
On the other hand if one knows there is only a single fixed sine and wants to determine what it is, that seems like the simplest case since the equation for a sine is defined in terms of three variables (A, omega, theta) and with perfect data three samples could suffice.
What you did with record was clever, but don't know if you are saying it is essentially the same type of mathematical problem JN is trying to solve?
That we seem to start from point zero again, each time when a topic is reemerging, is a bit disturbing given the fact that we are more ore less always the same bunch of guys discussing it......
What else? Anyone left who isn't incompetent and/or doesn't understand the basics of everything?
Maybe then we don't have to repeat it in every post in this thread, just once a week for starters and later advancing to once a month, mhm?
I remember having read in novel a couple of years back, where the first-person narrator talked about a boy in his school who desperately wanted to be a real good teacher.
Meeting him again 20 years later, the narrator says: "Unfortunately he managed only to become a veritable ******* instead" 🙂
Could make a good intention for the new year (and beyond of course); let's behave less often like an *******.
What else? Anyone left who isn't incompetent and/or doesn't understand the basics of everything?
Maybe then we don't have to repeat it in every post in this thread, just once a week for starters and later advancing to once a month, mhm?
I remember having read in novel a couple of years back, where the first-person narrator talked about a boy in his school who desperately wanted to be a real good teacher.
Meeting him again 20 years later, the narrator says: "Unfortunately he managed only to become a veritable ******* instead" 🙂
Could make a good intention for the new year (and beyond of course); let's behave less often like an *******.
That was the original point the theory involves a limit. On another issue unless you eliminate the possibility of high frequency IM in your choice of speakers or headphones any perceptual tests of these effects are a waste of time.
Did I miss something wrt IMD?
Of course you're right but I really can't remember when I've disputed that.....
Sorry, you are right.... you ought to follow the entire discussion some 20 pages back.
Not sure what you mean. We used the same strategy, 10ms time windows, to plot frequency response for a 1kHz to (48,50)kHz sweep. In this case the sampling frequency was 96k or 192k.
What I mean is, my CD player does not take 10 ms worth of the data stream and calculate what the output should be.
Jn
Jakob2, you are correct, sorry for the error...jitter.
Using jitter in that way is new to me too. I can sort-of see an analogy here, but there are several differences, too many to make it useful in any way. One is that jitter is the result of noise (see phase noise in a sine-wave oscillator, the noise converts to jitter when made into a square wave).Is there any other aspect of jitter really? Clock jitter of course in the end effect data... Never heard any aspect of cone movement being referred to as "jitter" - that was new to me.
//
This form of speaker distortion is the result of signals, not noise.
I've heard this called doppler distortion or FM distortion. It's just like the horn on a train as it goes past, it's coming toward you resulting in a higher than standing-still frequency, then going away it has a lower frequency. It would be impractical to move a train back and forth at 20Hz, but you could mount a horn on one of those vibration testers they use to test spacecraft for launch survival and get the same effect as a full-range (or moderately widerange) speaker playing two tones.
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