Digital primer

Gregadd

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While I am not planing to design a digital device I think it is about time to learn the basics. I doubt if the mebers are as green as I am but if you like I can postt some of the videos I am watching here, I hope this is the right place to put it. Moderators please feel free to move it if you desire.


Please feel free to commit.
 

Gregadd

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Gregadd

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What is an fft?

 

Gregadd

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What is dither?

 

Gregadd

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Maybe someone can help me The CD sampling rate is listed as 44.1khz. It seems to me it ought to be 44.1k samples/hz. 44.1khz is a frequency not a sampong rate. 22khz is a frequency not a sampling rate. Does that make any sense?
 
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Gregadd

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We now have a rudimentary understanding of sampling rate, bit depth jitterr and ,dither. We also know what an FFT is and what it basically represents. We definitely are not ready to take AMIRM on in a debate. We can however go over to the Science thread and examine an FFT without our eyes glazing over. Once I have completely digested these videos I'll try to find somethng a little more in depth. Suggestions are welcome.
 

Kal Rubinson

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Maybe someone can help me The CD sampling rate is listed as 44.1khz. It seems to me it ought to be 44.1k samples/hz. 44.1khz is a frequency not a sampong rate. 22khz is a frequency not a sampling rate. Does that make any sense?
The sampling rate is the number of samples per second. So, it might be expressed as 44.1k/sec but that is equivalent to 44.1kHz since 1Hz=1/sec.
 

Orb

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Maybe someone can help me The CD sampling rate is listed as 44.1khz. It seems to me it ought to be 44.1k samples/hz. 44.1khz is a frequency not a sampong rate. 22khz is a frequency not a sampling rate. Does that make any sense?

It is because from an engineering perspective it comes from cycles per second and the use of say a sampling gate as historically used in digital telephony with a frequency value of 8khz (8000 samples a second also giving the 125us interval-frame); this is where you get the original digital circuits sizes from 8khz x bits; so 64kbps for a single slot for original PCM digital circuit.
E1 and T1 then expanded this to either 30 channels or 24 channels, note I mention 30 channels instead of 32 as the extra 2 were for signalling and timing-framing related.
These would give us then either 2048k (2meg circuit Europe) or 1536kbps (1.5meg circuit North America).

So CD is a 44.1khz sampling rate x bit depth (16) to give the kbps and also indirectly interval points.
Cheers
Orb
 

Gregadd

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I suppose there is no confusion as long as you prefix with sampling rate or frequency. Thanks
 

Orb

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Gregadd it may help to think of sampling requiring an accurate sampling gate-clock, which ties into a frequency.
In other words the rate at which samples can be taken matches the frequency of the sampling clock.
Cheers
Orb
 

Gregadd

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That pretty much sums it up. God has a sense of humor. You end with noise that resembles tape hiss.
 

Gregadd

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Gregadd

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Gregadd

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amirm.png

Lets start with something simple .What can we tell form this? We know it is an FFT courtesy of AMRM.
 
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amirm

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Lets start with something simple .What can we tell form this W know it is an FFT courtesy of AMURM.
The most important thing we can tell is that you spelled my name wrong. :D

BTW, I will be explaining all of this in a series of videos I plan to produce "one of these days!" :)

Here is the text version. The setup is a computer playing a .wav file. The computer is playing a 12 Khz tone (it has a bit more but that is not important here). A perfect system produces that 12 Khz tone with nothing else. Now, if you looked at that waveform, it would look like a sine wave that it is. We call that "time domain." Unfortunately it is hard to analyze time domain signals visually. What if that sine wave is squashed by .01%? You could not tell that visually although we have clearly distorted the signal.

What we do to make the analysis simpler, or even possible, is to convert that "time domain" signal into "frequency domain." A guy by the name of Fourier proved that we can do that for infinitely running signals. We don't have infinite signal but for the purposes of this analysis, let's say we have plenty of it. FFT is a computer optimization of that transform that makes it run a lot faster.

In this situation, I have an audio analyzer that has an analog to digital converter. It captures whatever my DAC is producing. Once there, it has a digital representation in time. It applies the FFT to it and produces the graph that you see.

The graph faithfully shows that 12 Khz that we were playing. If my DAC and the ADC in the analyzer were ideal/perfect, that is all we would see. We don't see that. What we see are:

1. Additional spikes at certain frequencies. We have one at 5 Khz for example in purple. This says that if we play a pure 12 Khz tone at this level, the system will produce the 12 Khz but also add to it this 5 Khz tone. Clearly the 5 Khz is distortion. This distortion may be related to the frequency of the tone we are playing in which case we call it "correlated." Or it could always be there like 60 Hz hum from mains in which case it is "uncorrelated."

Now, we don't play single tones but music. Music can be decomposed into a sequence of single tones. If anytime we play a tone, we get another little sister of it 7 Khz lower (i.e. 12 Khz - 5 Khz), then the same thing will happen to every tone in your music. There will then be a spray of such extra tones corresponding to the rich spectrum of your music.

2. The second thing we see is noise. These are the random looking little ripples at the bottom. This information is not to be used in this analysis. The reason is that we averaged the 8 separate captures. Random noise then adds and subtracts from itself and its *measured* level lowers. We further can analyze the signal at higher resolution than its sampling rate, which again lowers the measured noise. In other words, we have optimized the system to have least amount of measurement noise to show up those little spikes. In reality, if you just listened to the DAC, its noise floor would be higher than these exceedingly small levels.

All of this tells us the meaning of measurements. It doesn't say any of this is necessarily bad from audibility point of view. That is a much more complicated topic.
 

amirm

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To help with the understanding of the previous explanation, here is the FFT of the file as it sits on my computer:



This is what it looks like *before* being played. As you see, there is a single spike in the middle and no other ones. There is also no wiggliness at the bottom like the noise we get out of a real system.

The spike is not narrow because of some of the details I skipped over in my last post due to it not being an infinitely long file and the fact that there are some modifications to make it different than a pure sine wave.
 

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