
Joined: Nov 2003
Posts: 459
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Joined: Nov 2003
Posts: 459 
One more for today, from Saturn this time:
Conveni, The  Ano Machi wo Dokusen Seyo (1997)(Masterpiece, Access, Human):
http://2sf.hcs64.com/rip/Conveni,%20The%20%20Ano%20Machi%20wo%20Dokusen%20Seyo%20(1997)(Masterpiece,%20Access,%20Human).zip
This game uses 2.04 version of the driver (the one that's very quiet). Did use the original driver and opted for volume=100 tag. Seems to help, don't hear anything bad with the rip (well, it's still kinda quite, but at least you can listen to it).
//Edit
Crap, noticed one of the tracks did overflow the tone bank. Proper rerip uploaded, correct BGM07.SSF CRC32: E1BF4CC1
Last edited by Knurek; 01/06/09 08:37 PM. Reason: rerip info




Joined: Mar 2001
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Nifty. Treasure Strike was crappy as advertised, but more rips is more rips. I'll check out Conveni a bit later.




Joined: Nov 2003
Posts: 459
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Conveni is nice if you like muzak. 's to be expected given that it's a convenience store sim.




Joined: Mar 2001
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Heh. You weren't kidding about Conveni being quiet. Nice songs though. BGM09 sounds like it's threatening to break out into reggae before it just stops.




Joined: Feb 2008
Posts: 107
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I can't get that damned filter to work. I've been trying to match the output of kingshriek's script to what (I think) I should be getting and it doesn't correspond.
Given f=1.0 and different q values I end up with cutoff F at 915kHz. You'd think with f that high it should easily go up to 22kHz (and above). On the other hand, if I'm reading the figure right, I should expect F as low as 4Hz for f=0.25 and about 50Hz for f=0.5
The Q has range of 3 to 20.25dB: Q [dB] = 0.75 * (reg_val  3) where reg_val is 0 to 31.
Is there a clever way to figure out the filter equation given some points with f and resulting cutoff F known? Assuming Q is neutral, that is 0dB.




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Q is 0.75 * (reg_val  4). Or, 0.75 * reg_val  3. It's quite funny how I used bold on the part I made mistake in, and can't even edit it now Anyway, BIOS uses filtering as well  not to mention envelopes. Shenmue music seems to prefer single values (infinite time between FEG points). The problem is that if I move the cutoff point (for a given filter value) lower it makes Shenmue sound better  but also breaks BIOS sounds. And vice versa. The only immediate difference that I see is that on the most broken sound effects BIOS uses Q (register) value of zero. I'm beginning to wonder if the filter equation proposed by Corlett is indeed correct  hence the question, how do I come up with something like that given the cutoff vs filter value curve. Q unknown but assumed to be 0dB.




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Posts: 56
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Posts: 56 
Given f (derived from the filter value) and q (given), you need to solve the following equation to get the cutoff frequency: abs(H(f, q, w)) = 3.0 dB where H is the filter transfer function and w is the normalized digital frequency (with 1.0 corresponding to the Nyquist frequency of 22050 Hz). In other words, the cutoff frequency will be 22050 * w. For this particular filter, the transfer function is: H(f, q, w) = f / (1  (1  f + q)*z + q*z*z) where z is the point on the unit circle in the complex plane at angle pi*w: z = exp(j*pi*w) Quick python script to do the above: from cmath import *
from bisect import bisect
import sys
def transfer_function(f, q, w):
zi = exp(1.0j*pi*w)
return f / (1.0  (1.0  f + q)*zi + q*zi*zi)
def bisection(func, x0, x1, TOL=1.0e6):
while abs(x1  x0) > 2*TOL:
x2 = (x0 + x1) / 2.0
if func(x0) * func(x2) > 0:
x0 = x2
else:
x1 = x2
return (x0 + x1) / 2.0
def calculate_f(filtervalue):
return ((((filtervalue & 0xFF)  0x100) << 4) >> ((filtervalue>>8)^0x1F))/8192.0
def lpfcutoff(filtervalue, q):
f = calculate_f(filtervalue)
func = lambda w: 20.0*log10(abs(transfer_function(f, q, w))).real + 3.0
return 22050.0*bisection(func, 0.0, 1.0)
if __name__ == '__main__':
argv = sys.argv
argc = len(argv)
if argc == 1:
sys.stdout.write('Usage: python %s <filtervalue> [<q>]\n' % argv[0])
sys.exit(0)
if argv[1][:2].lower() == '0x':
filtervalue = int(argv[1], 16)
else:
filtervalue = int(argv[1])
if argc == 2:
q = 0.0
else:
q = float(sys.argv[2])
sys.stdout.write('%f\n' % lpfcutoff(filtervalue, q))




Joined: Feb 2008
Posts: 107
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Still, this is just another bruteforce method Don't get me wrong, I know that IIR filter design is not exactly elementary school math. I just hoped there would be a way to figure out the whole thing  without having to try out one conversion equation after another... I've already modified the previous script to generate f to cutoff frequency table, it's way off from that in AICA docs. So it seems that at least we can't use the filter value directly as the f coefficient (I've tried this too by the way). This script is more accurate because I was to lazy to actualy solve the transfer function, rather I just made the loop do 1000 points and waited for it to dive below 3dB. That does not address the main issue I have now  is the filter itself correct or not. This is what little I understand: Basic 1pole filter: out = f * in  (1  f) * prev_outBasic 1pole filter with negative feedback: out = f * in  (1  f) * prev_out  q * prev_outIn the end q * (prev_out  prev_prev_out) is just a slightly more convoluted way of introducing feedback  or am I wrong? This feedback will cause resonance, though (if I get this right) q in AICA can also have negative values, causing a dip first  very dangerous with f close to 1 as it will tend to amplify and overload. The problem is I never found anyone doing actual filtering with this kind of feedback  though I've stumbled upon statement that resonance is not really used much except in synthesiers (which AICA is). I'm going to run more tests on Dreamcast this weekend. Perhaps FEG is much more closely tied to AEG than I initially thought... With my luck it's going to turn out this is not a filter problem at all, but something entierly else  just like it was with the DSP




Joined: Feb 2004
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Posts: 2,383 Likes: 98 
/me goes into signal processing lecturer mode: All IIR filters introduce feedback  after all, that's the way they generate an infinite impulse response (hence the name). The output from an IIR filter is only bounded for a subset of possible inputs  it is possible to give input that will cause an IIR filter to go into selfsustaining oscillations. That's why FIR filters are much more common in control systems. Unconditional stability is a big plus. FIR filters also have linear phase response. However, the transport delays introduced by an FIR filter can be problematic, too. FIR filters are easy to design, too  just do an inverse Fourier transform of the impulse response to get the frequency response, and vice versa. Why use an IIR filter? An IIR filter with a given number of taps can be made to have much steeper rolloff than an FIR filter with the same number of taps. An IIR filter can be made to have lower transport delay, which could make or break a controller. Finally, if you really do want to produce oscillations under certain conditions, an FIR filter will never do that for you. If you want to play with digital filter design a bit, you can download DFDP (an old Fortran program that runs under DOS). The next step up would be to use MATLAB Simulink's linear system analysis features to play around. If you're more serious about learning digital control system theory, the text you can't go past is Katsuhiko Ogata's DiscreteTime Control Systems (2nd Edition).




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Posts: 107 
Words "MATLAB" and "Simulink" scare me. See, back at Uni they've managed to convince me that I in fact do not like math. Or know it, for that matter I have just enough knowledge to understand what "zi = exp(1.0j*pi*w)" does in kingshriek's script and why, or how to build my own function to compute frequency cutoff for a given FIR/IIR filter. But that's it. There is always a chance I will figure this out, probably by sheer luck trying every possible filter equation I can come up with, but that will take massive amounts of time and work. I'm hoping someone with more experience in that matter could, with little effort, just pull something nice out of his/her magic hat



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