Thanks Scott and Jacob. Yes, as an anti aliasing filter. I will crudely sim it.
They knew each other (Saltzburg festival). They serviced different aspects of symphonic music.
Deal with Mitropoulos recordings if you are interested in symphonic composers from late 1800 and onwards. And his passion was operatic music.
Dimitri Mitropoulos | Discography | Discogs
Very interesting (to read about, explains a lot) personality. I suggest Trotter’s book https://www.amazon.com/Priest-Music-Life-Dimitri-Mitropoulos/dp/0931340810
George
Bill
What Ed has commented on effects of music on cats is applicable to humans too. So, consider not to expose your babies and kids to music that will induce on them intense psychological stress (e.g. almost anything from 20th century classic music).
George
What Ed has commented on effects of music on cats is applicable to humans too. So, consider not to expose your babies and kids to music that will induce on them intense psychological stress (e.g. almost anything from 20th century classic music).
George
Oh they do it the other way around. Endless Raffi. But they have worked out how to mix 'wheels on the bus' and 'baby shark' together whilst singing in the car!
But no Shoenberg till they are older 🙂
But no Shoenberg till they are older 🙂
The Nyquist Sampling Theorem states that: A bandlimited continuous-time signal can be sampled and perfectly reconstructed from its samples if the waveform is sampled over twice as fast as it's highest frequency component.
The Nyquist criterion states that a repetitive waveform can be correctly reconstructed provided that the sampling frequency is greater than double the highest frequency to be sampled.
I have asked this before in various ways .... what is the technique used to accurately capture a NON-repetitive, NON-continuous waveform?
Not sure we adequately addressed it. Was it? Or was I asleep and missed it?
THx-RNMarsh
The Nyquist criterion states that a repetitive waveform can be correctly reconstructed provided that the sampling frequency is greater than double the highest frequency to be sampled.
I have asked this before in various ways .... what is the technique used to accurately capture a NON-repetitive, NON-continuous waveform?
Not sure we adequately addressed it. Was it? Or was I asleep and missed it?
THx-RNMarsh
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Who me? You want MY opinion?
Well, if I had to capture a transient once and reconstruct it, I would use a lot more samples per waveform or capture and save it and run it 10 times to make it/simulate a continuous signal and reconstruct it if my processor was fast enough to do it near real time.
THx-RNMarsh
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@1Audio
I have connected the transformers to the HV part.
There are a numbers of things to mention.
1) The Cap size has been multiplied by 1.5 to get the peak at 23Khz.
Explanation for this is because the traces on the stator PCB's are on the outside.
This means that the glass fibre material with a rel. permittivity of 4.8 also makes part of the dielectricum between membrane and stator, thereby increasing the cap value.
The ratios between the 8 sections are probably O.K., so you only have to measure one single bass panel.
This should be ca. 520pF.
2) The 2.7H coil is extremely sensitive to the slightest change in value.
Just 1 or 2% percent changes the whole HF FR .
3) The image below is the best I could get for the moment, but around 20Khz FR is still quite messy and the impedance peak that should be around 16 Ohm, is way too high.
So to conclude, I think we need a better definition of the 2.7H coil against frequency, and also the unknown audio transformer is still playing a substantial role.
Maybe you still have some ideas to give it a try.
Hans
I have connected the transformers to the HV part.
There are a numbers of things to mention.
1) The Cap size has been multiplied by 1.5 to get the peak at 23Khz.
Explanation for this is because the traces on the stator PCB's are on the outside.
This means that the glass fibre material with a rel. permittivity of 4.8 also makes part of the dielectricum between membrane and stator, thereby increasing the cap value.
The ratios between the 8 sections are probably O.K., so you only have to measure one single bass panel.
This should be ca. 520pF.
2) The 2.7H coil is extremely sensitive to the slightest change in value.
Just 1 or 2% percent changes the whole HF FR .
3) The image below is the best I could get for the moment, but around 20Khz FR is still quite messy and the impedance peak that should be around 16 Ohm, is way too high.
So to conclude, I think we need a better definition of the 2.7H coil against frequency, and also the unknown audio transformer is still playing a substantial role.
Maybe you still have some ideas to give it a try.
Hans
Attachments
This is another myth that is popping up every now and then.
Non repetitive and non continuous waveforms are no exception to the Nyquist criterium.
Hans
You were asleep..😉
For a non continuous waveform, you have to sample fast enough to capture the sidebands. Nothing more, nothing less.
To know If you are capturing it all, look for the gibbs in the difference between input and output. Using the Fs of the filter, you can determine by trial and error where the upper sideband limit is.
The problem with a cymbal is, there is so much U/S in the high bw signal, that in essence there will be no threshold. That is why I mentioned 40Khz as a reasonable guideline, as I cannot see any envelope causing sidebands above that.
jn
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I got that. I was awke for that part 🙂 The key for me is the BW obviously needed to be at least 40Khz which i said before you concluded same.
Its the number I used in analog as min but also from your tests would take care of the problem also. And had in mind waiting for your analysis.
In the Traditional terms........ what is that sample rate? NUmber?
The repetitive-signal bandwidth represents the highest-frequency sinewave signal that the scope's input circuits can accept with three decibel maximum attenuation (the point where distortion becomes unacceptable). It's important to note that this frequency limit applies to repetitive waveforms (signals that repeat in a regular and reliable fashion).
The real-time bandwidth, in contrast, defines the highest frequency sinewave that a DSO can capture by sampling in a single pass, using a single trigger. This is also sometimes called the single-shot bandwidth.
Every 5 u/sec sample or more? Thats what I would use.
THx-RNMarsh
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Richard,
A DSO doesn't have a brick wall filter in front of it to limit the input to frequencies below Fs/2, and it doesn't have reconstruction filter to correct the display of a waveform limited to below Fs/2. Among other reasons, that makes it a totally different thing from digital audio.
A DSO doesn't have a brick wall filter in front of it to limit the input to frequencies below Fs/2, and it doesn't have reconstruction filter to correct the display of a waveform limited to below Fs/2. Among other reasons, that makes it a totally different thing from digital audio.
From what I can recall, the 40 Khz bandwidth recommendation was made by John Curl (perhaps also among others) 40 plus years ago.
I do not recall any analysis of modulation sidebands back then, just some anecdotal recollection that it made a difference. Unsupported IIRC by analysis or measurement.
And he was either ignored or castigated.
As far as I am concerned, I think he hit the nail on the head. If you were touting that back then, kudos to you as well.
jn
I know. Lets not use one for my question. A hypothetical product... maybe one which should exist.
-Richard
Well, yes, it was suggested by others a long time ago. Not from JC to me. It was proposed in an article based on phase changes etal causing detectable sound change. but I am talking about what is said here and now. I had concluded from what you were saying that a wider BW would take care of it. Suggested 40Khz based on your work shown here. You agreed that would be sufficient.
40Khz coincided with what we used in analog world for a long time. A coincidence here that I used same number. But the number which should have been used in creating the CD standard.. as I assume they too knew this. Maybe other factors not related to audio quality was at play.
Now with a clean sheet of paper, what would we spec now for sampling rate, BW etal. If it didnt need to be compatible with anything related to CD?
THx-RNMarsh
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X2
It's much easier to say a lot of nonsense than go relearn (or learn) the basics.
(u(t) - u(t-t_0))f(t) is something every sophomore/junior in EE should understand*. The resultant Fourier transform gets uglier when we replace the unit step functions with smoother ramp functions but the concept remains the same. And it better all sit below fs/2 or there will be aliasing.
In 1983/84 Philips had implemented a new listening facility to examine the audibility of certain effects and while still using a 16 Bit system they choose a sampling rate of 100 kHz.
@RNMarsh,
as said before, the people at Philips and Sony both preferred a compact format, Sony had presumably already a portable device in mind and on Philips side it was based on the compact cassette format and the reasoning that it has to be something very different in size than the vinyl LP to gain enough consumer interest.
And the quite cost preserving combination of a PCM adapter and a videotape recorder was already in use, so using the Fs of these systems made sense.
The Telarc/Soundstream experience which led to the 50 kHz sampling rate could have pointed to something different, but as usual there was some pressure to get the new system out.
@RNMarsh,
as said before, the people at Philips and Sony both preferred a compact format, Sony had presumably already a portable device in mind and on Philips side it was based on the compact cassette format and the reasoning that it has to be something very different in size than the vinyl LP to gain enough consumer interest.
And the quite cost preserving combination of a PCM adapter and a videotape recorder was already in use, so using the Fs of these systems made sense.
The Telarc/Soundstream experience which led to the 50 kHz sampling rate could have pointed to something different, but as usual there was some pressure to get the new system out.
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NOW, we are making progress here. Again, the same number I had in mind also as a minimum. Maybe even 200Khz. I am not interested in some arb BW limit based on marketing or even tech limits. Consider what ever max BW is needed and then nyq. sampling from that.
The max BW and sampling however should really be based upon the single-shot capture for music signals. Not CW. Very different numbers.
THx-RNMarsh
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