John Curl's Blowtorch preamplifier part III

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They are free to spew whatever they dreamed last night about.

Well excuse me, I did "dream" this up last night.:)

Excuse me if this has been shown before. I was trying to figure out a way to use the frequency domain to do down sampling with FFT's. This is a trick that does not work with power of 2 FFT's. If you take a file sampled at 192K and want to convert it to one sampled at 44.1k using power of two FFT's there is no way to have a common frequency per bin. Using an N X 44.1 FFT on one file and N X 192 FFT on the other all the bins have exactly the same spacing.

I tried the cymbal file truncated to 10sec taking the 10 X 192k file and using a real FFT I placed the bins from 0 Hz to 22050 Hz in a blank 10 X 44.1 sized file. The inverse FFT gives the resampled result which is almost exactly what the Audition resampler outputs. I have noticed that doing SRC back and forth in Audition or Cooledit builds up a very slight delay which I have not figured out. Creating MP3's seems to have the same problem which makes exact comparison harder.
 
With Audition, no, but I could use Rephase to build a 250-tap brickwall kernel/IR for 44.1... haven't used Rephase for quite a while but it should be doable without major effort, I might give it try...

Then again, if @Hans could share his kernel in a format that I can use I could analyse and use it in Audition directly. Such formats are:
- WAV with 32 or 64 bit floats
- binary file (plain array of double), sample-rate
- text file with coefficients (could convert that with a C quickie)

The filter I used is from a LTSpice library and supplied as a black box.
I entered a stopband of below -100dB and 250 taps.
But then again, what can be the purpose of trying to replicate this simulation.
What do we have to prove and why ?

Hans
 
The filter I used is from a LTSpice library and supplied as a black box.
I entered a stopband of below -100dB and 250 taps.
But then again, what can be the purpose of trying to replicate this simulation.
What do we have to prove and why ?

Hans
Your post filter shows a different frequency from the pre.
It is not known if it is a result of clipping out the upper sideband, or a result of your software, or some other unknown.

Is there a reason you are purposely ignoring this? I mean, it was your data.

Jn
 
Well excuse me, I did "dream" this up last night.:)

Excuse me if this has been shown before. I was trying to figure out a way to use the frequency domain to do down sampling with FFT's. This is a trick that does not work with power of 2 FFT's. If you take a file sampled at 192K and want to convert it to one sampled at 44.1k using power of two FFT's there is no way to have a common frequency per bin. Using an N X 44.1 FFT on one file and N X 192 FFT on the other all the bins have exactly the same spacing.

I tried the cymbal file truncated to 10sec taking the 10 X 192k file and using a real FFT I placed the bins from 0 Hz to 22050 Hz in a blank 10 X 44.1 sized file. The inverse FFT gives the resampled result which is almost exactly what the Audition resampler outputs. I have noticed that doing SRC back and forth in Audition or Cooledit builds up a very slight delay which I have not figured out. Creating MP3's seems to have the same problem which makes exact comparison harder.

Since the conversion ratio is 44100/192000=441/1920 you could insert M=441-1=440 zeroes between each original sample. Filter this sample at 44.1KHz/2=22.05KHz and then pick every N=1920th sample out of the output. Voila, decimated.

The interpolation method (more difficult to implement, IMO) is mathematically equivalent, since choosing an interpolation function is equivalent with choosing the above filter impulse response.
 
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Thanks, interesting. Do they ever provide details of the setup? There are always going to be compromises as you are aware. Have you ever been tempted to do what SL did here? AS_creation
Tempted yes as have I been with getting a dummy head using molds of my own ears for testing. But like many things right now with two chaos butterflies in the house free time whilst my brain is still functioning at some sort of human level is limited.


Perhaps this would also give a better rendering of accurate than Richard's test?


It 'might' allow a better AB comparison for sure.
 
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. A pity. They promised us eternal life.


I've had plenty of CDR self destruct over the years. But other than scratched to death by the kids (and even then some polish cleaned most those up enough to rip) I've not had a CD die on me. Whatever Howie did with his massive plasma dodads he did well.



Now I have some thousands of albums on the server. I just haven't found the ideal way to search within those for what I want, but that is what they refer to as a happy problem.
 
I've had plenty of CDR self destruct over the years. But other than scratched to death by the kids (and even then some polish cleaned most those up enough to rip) I've not had a CD die on me. Whatever Howie did with his massive plasma dodads he did well.



Now I have some thousands of albums on the server. I just haven't found the ideal way to search within those for what I want, but that is what they refer to as a happy problem.

I found a copy of Microsoft Musical Instruments (1992) on CD going through some old stuff at my parent's house last month. It was in perfect condition and read just fine in my LG BD-RW drive.

CD-Rs on the other hand... it depends on the type of dye, but most don't last that long.
 
Now I have some thousands of albums on the server. I just haven't found the ideal way to search within those for what I want, but that is what they refer to as a happy problem.

Every piece of music I own is also ripped on a server as MP3 (yes, audiophiles are entitled to being horrified about, poor me can't tell the CD original from an 192k and up MP3), a total of about 400Gb, about 3000 albums.

The answer to your search quest was, for me, DLNA.
 
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Best I have found for me is command line locate x | grep y . The music player I use in the kitchen does have a fancy tree search but the old and pathetic tablet running it barfs trying to load all the results.



My biggest problem is that I am rubbish with stuffing music into genre pigeon holes. To me there are two sorts of music. The stuff I like, and the other stuff. the rest is dreamed up by music journos to make them sound hip and with it.



The day I work out where to file Alabama 3 I think I will have worked out a tagging system I can use.
 
Since the conversion ratio is 44100/192000=441/1920 you could insert M=441-1=440 zeroes between each original sample. Filter this sample at 44.1KHz/2=22.05KHz and then pick every N=1920th sample out of the output. Voila, decimated.

The interpolation method (more difficult to implement, IMO) is mathematically equivalent, since choosing an interpolation function is equivalent with choosing the above filter impulse response.

I asked why the 192k and not the easier 176.4k 20 pages ago. No response.
 
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