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R. Belmont: I haven't done anything with the AICA since my last patch, so feel free to work on it.

I do have a (preliminary) Gunbird 2 rip, though, in case anyone is interested. I'm not sure if the rip is correct, however, since IMXL/ISEL/DISDL/DIPAN are constant (0/0/F/0) for all key-ons which would indicate the game's music is mono (can this actually be the case?). I went ahead and ripped all sequences (MSB) and one-shot banks (OSB). Interestingly, Highly Theoretical fails to play any of the OSB tracks, but AOSDK handles them just fine.

http://www.sendspace.com/file/evlrsa

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The arcade Gunbird 2 appears to be mono too, even though it's sequenced on a chip that's quite capable of good stereo (YMF278B, which is essentially the SCSP's father).

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Mmm, nice. Doing type 2 as DK suggests does improve things quite a bit. Gonna try and get interpolation going with ADPCM now.

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New SDK:

http://rbelmont.mameworld.info/aosdk_140a5.zip

- Fixed ADPCM looping as per Deunan. Sakura Taisen 4 is distortion-free now, but ST3 still has some (possibly due to other causes?)
- I tried to get ADPCM interpolation going. And failed smirk
- Fixed some Skies of Arcadia songs from crashing - they were setting an active slot playing 16-bit samples' control register so that PCMS = 2, and thus adbase was NULL. Not sure exactly what real h/w would do here since neither KEYONEX or KEYONB are set in the new CTRL value.
- Ideas welcome on whatever we're still missing, notably the filter.

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Originally Posted By R. Belmont

- I tried to get ADPCM interpolation going. And failed smirk


Code:
diff -Nru aosdk_base/eng_dsf/aica.c aosdk/eng_dsf/aica.c
--- aosdk_base/eng_dsf/aica.c	2008-02-14 00:03:50.000000000 -0800
+++ aosdk/eng_dsf/aica.c	2008-02-13 21:30:06.000000000 -0800
@@ -1037,15 +1037,14 @@
 			slot->nxtbase = base;
 			slot->nxtstep = curstep;
 
-			s=(slot->cur_sample);
-	//		s=(int) slot->cur_sample*((1<<SHIFT)-fpart)+(int) slot->nxt_sample*fpart;
+			s=(int) slot->cur_sample*((1<<SHIFT)-fpart)+(int) slot->nxt_sample*fpart;
 		}
 		else
 		{
 			s = 0;
 		}
 
-		sample=s;
+		sample=s>>SHIFT;
 	}
 	
 	slot->prv_addr=slot->cur_addr;


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And this is why I shouldn't work on code at midnight. Thanks, ks smile

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Hmm... so basically you keep two ADPCM states per channel, one for main sample and one for interpolation? Interesting.
There's a limit on pitch changes (vs sound's own) when it comes to ADPCM, but what happens if you're standing on LEA-1 and steps_to_go is greater than 1, say 2 for example? Won't it read LEA-1 (correct) and then LEA sample/garbage (that's past loop end now) and contaminate decoder state? This is a very rare case, but still...

I don't know what happens when you change sample type on active channel - I could try to test that. I'm kinda busy now, away from home/town much, so it will have to wait a bit on the TO-DO list.
My code will happily switch but ADPCM state will be undefined as it's preset on KEY ON. What AEG phase/level was that channel when the switch happened?

If there are any channels playing ADPCM wrong, try to catch what sample type it is (2/3) and what are LSA and LEA. I can't find good test cases where LSA > 0.

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a5 with kingshriek's fix:

http://rbelmont.mameworld.info/aosdk_140a5a.zip

Sounds very nice. I could listen to Skies of Arcadia all day on it now :-)

BTW, KS, how are we doing vs. HE on the problems you were having with some of your rips?

PS: I'll post new AO builds with DSF support later today. Richard, if you want to get a head start, just use the eng_dsf from the aosdk.

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Hmmm... sounds a little screwy on my machine. Has this been de-endianed yet?

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The 16-bit samples may need to be de-endianed. I can't think of anywhere else that should be a problem though.

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