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Posts posted by ivop
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1 hour ago, emkay said:Not really.
Well, REALLY : https://en.wikipedia.org/wiki/Nyquist–Shannon_sampling_theorem
1 hour ago, emkay said:If using 400Hz sampling frequency, one could set
1,2,3,4
or
1,2,1,2
or
1,2,3,1 2,3,1,2 3,1,2,3 ... to get waves inbetween.
Yes, but NEVER a tone above 200Hz if your sampling/replay frequency is 400Hz.
1 hour ago, emkay said:And, btw. There seems to be an adaption mistake somehow.
What do you mean by that?
1 hour ago, emkay said:Sampling rate at 16kHz means a sampling frequency at 8kHz. And, if there is a special waveform to get, it's half of the sampling frequency.
No, a sampling frequency of 16kHz means a sampling rate of 16khz. It's the same. Half the sampling frequency, or rate, is the maximum (analog) frequency that can be captured at that rate. Or frequency
1 hour ago, emkay said:Things get technically completely weird, if a "Frequency" means to have a "cycle completed" while a "rate" means to get a "sample" frequently at some point on a specific time.
So where did the frequently getting of a sample the name of "Sampling Frequency"?
I see it also in German named "Frequency" instead of "Rate" .
See above. You are confusing sampling frequency/rate and the note-that-you-are-hearing frequency. The latter can never be higher than half of the former.
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13 minutes ago, Levas said:Source code and compiled binaries for onboard USB-serial adapter. Still not working with flow control. I don't know if it is firmware or windows generic serial driver problems.
Not direct link, but link to my web pages. Scroll down for download.
http://www.vabolis.lt/2021/04/17/arm21-stm32-usb-cdc-ir-hardware-handshake/
(lots of old man's growling in the post)
I love Google translate from Lithuanian to English.
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Shit those authors.
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Bishop I added fog?
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Sunny Minnesota. Thanks for the laugh. I remember they asked Prince back in the day why he still lived in Chanhassen, MN. He said: “It's so cold, it keeps the bad people out.”
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16 hours ago, rensoup said:I believe I might have already asked but is 400hz enough for interesting bass sound ?
The Nyquist–Shannon sampling theorem states that the sampling frequency must be at least twice the frequency you want to capture. So if your capture and replay frequency is 400Hz, the maximum you can capture is a 200Hz square wave with a 50/50 duty cycle. If you want sawtooths for example, your maximum decreases, or it'll turn into a square wave.
Square wave: $1C, $ 10, $1C, $10, .... repeat and replay at 400Hz, note will be 200Hz
Sawtooth: $1C, $18, $14, $10, .... repeat and replay at 400Hz, note will be 100Hz.
And so on...
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16 hours ago, rensoup said:Despite its name (Softsynth), I thought the software was purely a sample based soundtracker, the demo disk posted earlier has a bunch of songs which I guessed all used the same sample set. But if it's all generated on the fly, that's even cooler.
It's actually a mix. It uses small snippets (preferably exactly one wave period), and loops those at different speeds (i.e. jumping through it with a different delta and loop). AtariSid6 also has different snippets for different volumes, but that is not strictly necessary.
QuoteOne thing I forgot is that for volume mode only, AUDF would not be required and would compress very nicely.
Indeed, you only need AUDC, but with full samples you can forget about decent compression with LZSS, or any lossless format for that matter. That's why FFT/DCT based lossy compression schemes were invented
BUT! The small windows of 256 or 512 bytes can be compressed pretty effecient, for example with delta compression and RLE, which is what AtariSid6 already does.
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At first, I was thinking Digi Drum 2, too. They sound like the same resampled fixed pot Linn LM-1 samples.
@SenorRossie and I used to create a lot of patterns BITD. And my "little" brother and I used to rap to beats
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IMHO WSYNC wastes way too much cycles. You only have 114-9 = 105 cycles per scanline. To waste 4 on the STA and several more on the wait time is not economic. AtariSid6 would not have been possible that way. I know MTP is only 7.6kHz. That might help if the player code is interleaved with the sample playing, but that is very specific to this player. AtariSid1 and 2 were done this way, when I still played the SAP-R equivalent of SID, i.e. register dumps. Once you start calling generic 6502 player routines, you have to invert what you would normally do. Now you call the player in the main loop, synced to VCOUNT, and play the (generated or full) samples driven by an interrupt.
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Also, don't forget that if you use true samples, that'll cost a lot of memory. Playing a softsynth in the back (either HARDBass, or something else, like a stripped AtariSid6), uses a lot less memory, because it repeats small waves 256 or 512 bytes. And as Sandor says, those waves can be anything. Not just a sawtooth bass, but can contain thirds and fifths to simulate a chord or organ sound. Think the original "Softsynth" program.
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12 minutes ago, x=usr(1536) said:I'm curious as to how a SysCheck would compare to a 1064 in that situation with regard to the on-board RAM.
With the jumpers in the right position, a SysCheck2 definitely replaces the on-board RAM, and even ROM if you want to.
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It's the power-up pattern of the memory. "Cleared" works, DRAM1 (for example) does not work. It assumes all memory is zeroed out. And 128K plus BASIC off
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3 hours ago, x=usr(1536) said:It also occurred to me that removing all on-board RAM and connecting the 1064 (which is known-working) wasn't tried, so I'll add that to the troubleshooting list as well.
If this doesn't work, the new RAM chips have probably damaged one or more of the chips on your motherboard 😕
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13 hours ago, Rybags said:If it's so far unmodded and not working with the original 16K back then something is definitely wrong.
Yes, that implies the machine was damaged with the new RAM chips plugged in. If, let's say the RAM was damaged in the first place, which ICs could it take with it? Same for when it was aligned wrong by mistake?
Even you think you do it right, sometimes it is not. I remember demonstrating the MIDIMuse interface with an S2 module in Herten, Germany (ABBUC meeting). Showing the S2 separately to somebody, then plugged it back in. Just before I fired up my Atari, a spectator said I misaligned the connector. Pfew
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15 hours ago, VinsCool said:Awesome! I might borrow several things from it in the sheet I pretty much started yesterday from your initial work, all this stuff is going to be incredibly useful!
I really like where all of this is going now :3Yes, this is all very cool
And borrow all you want! That's why I share it, for the bettering of Pokey sounds
I'm also pretty fond of the Diatonic intervals. You'll optimally need 24 tables (one for each major key, and one for each minor key), but one could create a simple C/C++/C# program to generate them. The infill is there only for the jazz chords
Oh, and you'll need to know in which key your song is
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Added some LEDs for +12V and +5V:
You know, otherwise you are bound to plug in or unplug a card with the board still powered
I'm fairly sure this is the board I'm going to order soon. Any last minute comments?
It's 20x20cm. 5 boards would set me back $79 + shipping ($20+) + tax ($20+) + custom duties (~$15), which is somewhere between $130-$150 probably. That's for a four layer 2.0mm board, 2oz front and back copper, and 1oz sandwiched copper for the inner layers. Oldskool 4lyr 1.6mm 1oz/0.5z is on $13 cheaper, so a sturdy board it is! It would come out to circa $30 per board. Not too bad IMHO.
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7 hours ago, emkay said:The analog strings build their own waves at different times. So every time the resulting tone will get different.
That's why the detuning needs to be done until the waves won't logically add together into loud interferences.
You were right up to here. The detuning of the B-string has nothing to do with loudness or when the wave starts in regard to when it was plucked or picked!! It is all about frequency or pitch or whatever you want to call it
This was about a 12-TET 3rd note retuned to a diatonic 3rd.
Here a new version of my spreadsheet, already!
I added Diatonic Major and Minor, and both with sharps/flats infill by Pythagorean tuning, and finally one that combines Diatonic major and minor intervals, and fills the two remaining gaps by Pythagorean tuning.
On the Diatonic Major sheet:
The Major 3rd is indeed circa 14 cents lower, as was mentioned and explained in the Scar Tissue video.
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Here's a spreadsheet to calculate Pythagorean tuning, or any other RA:TIO for that matter.
To keeps things manageable, I decided to base everything on a single clock. You can change that from NTSC to PAL, and vice versa. That's half the amount of columns
I also skipped the HEX columns. And the delta columns are now in cents instead of hertz.
In this example, note '1' is tuned to 27.5 Hz, but you can change that to any frequency you want.
Just for fun, I created a second sheet with the exact 12-TET ratios.
The third sheet is an alternate Pythagorean tuning. Gb (1024:729) and F# (729:512) are different ratios. It's the devil's interval!!
You can copy the sheet as many times as you want and try out different intervals.
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10 hours ago, emkay said:I'm still not sure, want you want to say.
Of course the 8 bit resolution is rather limited.
QuoteI posted the two videos above.
Why is no one else blaming this detuning ?
Oh, I am. It seems we are not constantly on the same page
Vinscool's tables try to minimize the difference of dist A notes to dist C notes, more or less.
What I'm talking about it, leave dist C out of the mix, and be freed of the limitation of dist C. Now you can use just intonation for each musical key, and sound clearer, more open. I hope you watch the video on Scar Tissue.
QuoteI'm talking of some 0.xxHz , while the tunes above shift the tuning in more than 10Hz difference from the high to the low tones.
If an Orchestra is "tuned to 432Hz" , all notes at all instruments were adjusted to that.
Not the high tones to 443Hz and the low tones to 432Hz.
Exactly. Here we agree. That's what Vinscool's tables fix.
Trying Pythagorean tuning is something completely different, and won't work properly with dist C because these notes are fixed.
I'll post a Calc sheet tonight. Almost finished.
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9 minutes ago, VinsCool said:I imagine it should be easy enough om the same sheet to add with a simple formula?
Not really, because the first note of a tuning is not always C, or A, or whichever note. But I will add a 12-TET column where every frequency is 100 cents apart. And a few columns that calculate back to pokey timer values.
Quotei was also wondering how could it work for tuning based on existing known frequencies, that will be especially useful for distortions and exotic combinations
The idea is that you fill in note '1' equals 55Hz, or something else, and it calculates all octaves according to different ratios.
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24 minutes ago, emkay said:It's correct for 99% of Synthesizer music... so what?
If we had 16-bit resolution for each channel. But we have not. With our limitted 8-bit resolution, I'm sure just tuning will improve songs that don't use dist C basses, but just dist A at 15kHz, 64kHz or in 16-bit 1.77MHz. Especially the 64kHz notes will sound more in tune. But you'll need a table for each key.
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11 minutes ago, emkay said:Fact is simply that NTSC needs the given calculation, while PAL needs the "440Hz" adjustment.
That's correct. I was thinking you were thinking 440Hz was something special, but it appears that's not what you meant
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The 432Hz emkay references is nothing but a conspiracy theory. He's correct that it is not something better. 432Hz tuning is also arbitrary. Every tuning is arbitrary. Even during a two hour concert, the tuning of an orchestra might change due to changing circumstances.
6 minutes ago, VinsCool said:I must say thanks to Ivo's spreadsheet, I was able to play with it properly now and I was amazed to find out a majority of my Alternate Tuning was in reality quite good for the most part, and the interval problems mentioned earlier did indeed show up for a couple notes, the same ones I tried too hard to tune, it simply caused more issues on the long run, haha.
Thanks. I really should do a diatonic or even Pythagorean tuning spreadsheet. Which means you select a frequency for the base note (1), and then 1, 1#, 2, 2#, 3, 4, 4#, 5, 5#, 6, 6#, and 7 are calculated for every octave with said ratios.
6 minutes ago, VinsCool said:Sooo I guess it's just to do a few changes here and there, adjust the other tables if necessary, and that should be good enough, hopefully.
16-bit definitely gives a much smaller error margin, thankfully.Definitely. Pokey's 16-bit value is exactly what you want for music. High resolution in the lows, less resolution in the highs. SID is the opposite, which is why they struggle with in tune basses.
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20 minutes ago, emkay said:And the "440Hz" solution is way correct there.
Again, 440Hz is arbitrary. It all depends on the circumstances. Vinscool proved you wrong by tuning his table to A=443.9Hz, because in the musical keys he used, it sounds more in tune with the Dist C basses, which sadly are at fixed frequencies.
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3 hours ago, mytek said:I think I know why you are so interested in this circuit
.
I think you think right
If space permits, I might implement both this circuit and the original charge pump, and switch with a jumper, to leave the option open to use +5V only.
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3 minutes ago, mytek said:BTW, the LM4040AIZ-10.0 limits the available voltage to 10V prior to the 4.7K series resistor.
Yes, I was aware of the Zener diode. Hence I calculated ~9.9V relative to 10V.
So the charge pump is out of spec, too
It seems that GTIA can handle 20-25% overvoltage on CADJ, at least at a low current. Or as you say, nobody leaves it at that, because you will have some strange colors.
But what if the pot fails while you are not looking, and CADJ is constantly overvoltaged for a long time? Would that be bad?

XL keyboard technical drawings
in Atari 8-Bit Computers
Posted
Progress.
Here's a generic schematic of the XL keyboard.
No PCB, yet.
Project Helena.pdf