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1 | /***************************************************************************\r |
2 | reverb.c - description\r | |
3 | -------------------\r | |
4 | begin : Wed May 15 2002\r | |
5 | copyright : (C) 2002 by Pete Bernert\r | |
6 | email : BlackDove@addcom.de\r | |
7 | ***************************************************************************/\r | |
8 | /***************************************************************************\r | |
9 | * *\r | |
10 | * This program is free software; you can redistribute it and/or modify *\r | |
11 | * it under the terms of the GNU General Public License as published by *\r | |
12 | * the Free Software Foundation; either version 2 of the License, or *\r | |
13 | * (at your option) any later version. See also the license.txt file for *\r | |
14 | * additional informations. *\r | |
15 | * *\r | |
16 | ***************************************************************************/\r | |
17 | \r | |
18 | #include "stdafx.h"\r | |
19 | \r | |
20 | #define _IN_REVERB\r | |
21 | \r | |
22 | // will be included from spu.c\r | |
23 | #ifdef _IN_SPU\r | |
24 | \r | |
25 | ////////////////////////////////////////////////////////////////////////\r | |
26 | // globals\r | |
27 | ////////////////////////////////////////////////////////////////////////\r | |
28 | \r | |
29 | // REVERB info and timing vars...\r | |
30 | \r | |
31 | int * sRVBPlay = 0;\r | |
32 | int * sRVBEnd = 0;\r | |
33 | int * sRVBStart = 0;\r | |
34 | int iReverbOff = -1; // some delay factor for reverb\r | |
35 | int iReverbRepeat = 0;\r | |
36 | int iReverbNum = 1; \r | |
37 | \r | |
38 | ////////////////////////////////////////////////////////////////////////\r | |
39 | // SET REVERB\r | |
40 | ////////////////////////////////////////////////////////////////////////\r | |
41 | \r | |
42 | void SetREVERB(unsigned short val)\r | |
43 | {\r | |
44 | switch(val)\r | |
45 | {\r | |
46 | case 0x0000: iReverbOff=-1; break; // off\r | |
47 | case 0x007D: iReverbOff=32; iReverbNum=2; iReverbRepeat=128; break; // ok room\r | |
48 | \r | |
49 | case 0x0033: iReverbOff=32; iReverbNum=2; iReverbRepeat=64; break; // studio small\r | |
50 | case 0x00B1: iReverbOff=48; iReverbNum=2; iReverbRepeat=96; break; // ok studio medium\r | |
51 | case 0x00E3: iReverbOff=64; iReverbNum=2; iReverbRepeat=128; break; // ok studio large ok\r | |
52 | \r | |
53 | case 0x01A5: iReverbOff=128; iReverbNum=4; iReverbRepeat=32; break; // ok hall\r | |
54 | case 0x033D: iReverbOff=256; iReverbNum=4; iReverbRepeat=64; break; // space echo\r | |
55 | case 0x0001: iReverbOff=184; iReverbNum=3; iReverbRepeat=128; break; // echo/delay\r | |
56 | case 0x0017: iReverbOff=128; iReverbNum=2; iReverbRepeat=128; break; // half echo\r | |
57 | default: iReverbOff=32; iReverbNum=1; iReverbRepeat=0; break;\r | |
58 | }\r | |
59 | }\r | |
60 | \r | |
61 | ////////////////////////////////////////////////////////////////////////\r | |
62 | // START REVERB\r | |
63 | ////////////////////////////////////////////////////////////////////////\r | |
64 | \r | |
65 | INLINE void StartREVERB(int ch)\r | |
66 | {\r | |
67 | if(s_chan[ch].bReverb && (spuCtrl&0x80)) // reverb possible?\r | |
68 | {\r | |
69 | if(iUseReverb==2) s_chan[ch].bRVBActive=1;\r | |
70 | else\r | |
71 | if(iUseReverb==1 && iReverbOff>0) // -> fake reverb used?\r | |
72 | {\r | |
73 | s_chan[ch].bRVBActive=1; // -> activate it\r | |
74 | s_chan[ch].iRVBOffset=iReverbOff*45;\r | |
75 | s_chan[ch].iRVBRepeat=iReverbRepeat*45;\r | |
76 | s_chan[ch].iRVBNum =iReverbNum;\r | |
77 | }\r | |
78 | }\r | |
79 | else s_chan[ch].bRVBActive=0; // else -> no reverb\r | |
80 | }\r | |
81 | \r | |
82 | ////////////////////////////////////////////////////////////////////////\r | |
83 | // HELPER FOR NEILL'S REVERB: re-inits our reverb mixing buf\r | |
84 | ////////////////////////////////////////////////////////////////////////\r | |
85 | \r | |
86 | INLINE void InitREVERB(void)\r | |
87 | {\r | |
88 | if(iUseReverb==2)\r | |
89 | {memset(sRVBStart,0,NSSIZE*2*4);}\r | |
90 | }\r | |
91 | \r | |
92 | ////////////////////////////////////////////////////////////////////////\r | |
93 | // STORE REVERB\r | |
94 | ////////////////////////////////////////////////////////////////////////\r | |
95 | \r | |
96 | INLINE void StoreREVERB(int ch,int ns)\r | |
97 | {\r | |
98 | if(iUseReverb==0) return;\r | |
99 | else\r | |
100 | if(iUseReverb==2) // -------------------------------- // Neil's reverb\r | |
101 | {\r | |
102 | const int iRxl=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x4000;\r | |
103 | const int iRxr=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x4000;\r | |
104 | \r | |
105 | ns<<=1;\r | |
106 | \r | |
107 | *(sRVBStart+ns) +=iRxl; // -> we mix all active reverb channels into an extra buffer\r | |
108 | *(sRVBStart+ns+1)+=iRxr;\r | |
109 | }\r | |
110 | else // --------------------------------------------- // Pete's easy fake reverb\r | |
111 | {\r | |
112 | int * pN;int iRn,iRr=0;\r | |
113 | \r | |
114 | // we use the half channel volume (/0x8000) for the first reverb effects, quarter for next and so on\r | |
115 | \r | |
116 | int iRxl=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x8000;\r | |
117 | int iRxr=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x8000;\r | |
118 | \r | |
119 | for(iRn=1;iRn<=s_chan[ch].iRVBNum;iRn++,iRr+=s_chan[ch].iRVBRepeat,iRxl/=2,iRxr/=2)\r | |
120 | {\r | |
121 | pN=sRVBPlay+((s_chan[ch].iRVBOffset+iRr+ns)<<1);\r | |
122 | if(pN>=sRVBEnd) pN=sRVBStart+(pN-sRVBEnd);\r | |
123 | \r | |
124 | (*pN)+=iRxl;\r | |
125 | pN++;\r | |
126 | (*pN)+=iRxr;\r | |
127 | }\r | |
128 | }\r | |
129 | }\r | |
130 | \r | |
131 | ////////////////////////////////////////////////////////////////////////\r | |
132 | \r | |
133 | INLINE int g_buffer(int iOff) // get_buffer content helper: takes care about wraps\r | |
134 | {\r | |
135 | short * p=(short *)spuMem;\r | |
136 | iOff=(iOff*4)+rvb.CurrAddr;\r | |
137 | while(iOff>0x3FFFF) iOff=rvb.StartAddr+(iOff-0x40000);\r | |
138 | while(iOff<rvb.StartAddr) iOff=0x3ffff-(rvb.StartAddr-iOff);\r | |
139 | return (int)*(p+iOff);\r | |
140 | }\r | |
141 | \r | |
142 | ////////////////////////////////////////////////////////////////////////\r | |
143 | \r | |
144 | INLINE void s_buffer(int iOff,int iVal) // set_buffer content helper: takes care about wraps and clipping\r | |
145 | {\r | |
146 | short * p=(short *)spuMem;\r | |
147 | iOff=(iOff*4)+rvb.CurrAddr;\r | |
148 | while(iOff>0x3FFFF) iOff=rvb.StartAddr+(iOff-0x40000);\r | |
149 | while(iOff<rvb.StartAddr) iOff=0x3ffff-(rvb.StartAddr-iOff);\r | |
150 | if(iVal<-32768L) iVal=-32768L;if(iVal>32767L) iVal=32767L;\r | |
151 | *(p+iOff)=(short)iVal;\r | |
152 | }\r | |
153 | \r | |
154 | ////////////////////////////////////////////////////////////////////////\r | |
155 | \r | |
156 | INLINE void s_buffer1(int iOff,int iVal) // set_buffer (+1 sample) content helper: takes care about wraps and clipping\r | |
157 | {\r | |
158 | short * p=(short *)spuMem;\r | |
159 | iOff=(iOff*4)+rvb.CurrAddr+1;\r | |
160 | while(iOff>0x3FFFF) iOff=rvb.StartAddr+(iOff-0x40000);\r | |
161 | while(iOff<rvb.StartAddr) iOff=0x3ffff-(rvb.StartAddr-iOff);\r | |
162 | if(iVal<-32768L) iVal=-32768L;if(iVal>32767L) iVal=32767L;\r | |
163 | *(p+iOff)=(short)iVal;\r | |
164 | }\r | |
165 | \r | |
166 | ////////////////////////////////////////////////////////////////////////\r | |
167 | \r | |
168 | INLINE int MixREVERBLeft(int ns)\r | |
169 | {\r | |
170 | if(iUseReverb==0) return 0;\r | |
171 | else\r | |
172 | if(iUseReverb==2)\r | |
173 | {\r | |
174 | static int iCnt=0; // this func will be called with 44.1 khz\r | |
175 | \r | |
176 | if(!rvb.StartAddr) // reverb is off\r | |
177 | {\r | |
178 | rvb.iLastRVBLeft=rvb.iLastRVBRight=rvb.iRVBLeft=rvb.iRVBRight=0;\r | |
179 | return 0;\r | |
180 | }\r | |
181 | \r | |
182 | iCnt++; \r | |
183 | \r | |
184 | if(iCnt&1) // we work on every second left value: downsample to 22 khz\r | |
185 | {\r | |
186 | if(spuCtrl&0x80) // -> reverb on? oki\r | |
187 | {\r | |
188 | int ACC0,ACC1,FB_A0,FB_A1,FB_B0,FB_B1;\r | |
189 | \r | |
190 | const int INPUT_SAMPLE_L=*(sRVBStart+(ns<<1)); \r | |
191 | const int INPUT_SAMPLE_R=*(sRVBStart+(ns<<1)+1); \r | |
192 | \r | |
193 | const int IIR_INPUT_A0 = (g_buffer(rvb.IIR_SRC_A0) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_L * rvb.IN_COEF_L)/32768L;\r | |
194 | const int IIR_INPUT_A1 = (g_buffer(rvb.IIR_SRC_A1) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_R * rvb.IN_COEF_R)/32768L;\r | |
195 | const int IIR_INPUT_B0 = (g_buffer(rvb.IIR_SRC_B0) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_L * rvb.IN_COEF_L)/32768L;\r | |
196 | const int IIR_INPUT_B1 = (g_buffer(rvb.IIR_SRC_B1) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_R * rvb.IN_COEF_R)/32768L;\r | |
197 | \r | |
198 | const int IIR_A0 = (IIR_INPUT_A0 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_A0) * (32768L - rvb.IIR_ALPHA))/32768L;\r | |
199 | const int IIR_A1 = (IIR_INPUT_A1 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_A1) * (32768L - rvb.IIR_ALPHA))/32768L;\r | |
200 | const int IIR_B0 = (IIR_INPUT_B0 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_B0) * (32768L - rvb.IIR_ALPHA))/32768L;\r | |
201 | const int IIR_B1 = (IIR_INPUT_B1 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_B1) * (32768L - rvb.IIR_ALPHA))/32768L;\r | |
202 | \r | |
203 | s_buffer1(rvb.IIR_DEST_A0, IIR_A0);\r | |
204 | s_buffer1(rvb.IIR_DEST_A1, IIR_A1);\r | |
205 | s_buffer1(rvb.IIR_DEST_B0, IIR_B0);\r | |
206 | s_buffer1(rvb.IIR_DEST_B1, IIR_B1);\r | |
207 | \r | |
208 | ACC0 = (g_buffer(rvb.ACC_SRC_A0) * rvb.ACC_COEF_A)/32768L +\r | |
209 | (g_buffer(rvb.ACC_SRC_B0) * rvb.ACC_COEF_B)/32768L +\r | |
210 | (g_buffer(rvb.ACC_SRC_C0) * rvb.ACC_COEF_C)/32768L +\r | |
211 | (g_buffer(rvb.ACC_SRC_D0) * rvb.ACC_COEF_D)/32768L;\r | |
212 | ACC1 = (g_buffer(rvb.ACC_SRC_A1) * rvb.ACC_COEF_A)/32768L +\r | |
213 | (g_buffer(rvb.ACC_SRC_B1) * rvb.ACC_COEF_B)/32768L +\r | |
214 | (g_buffer(rvb.ACC_SRC_C1) * rvb.ACC_COEF_C)/32768L +\r | |
215 | (g_buffer(rvb.ACC_SRC_D1) * rvb.ACC_COEF_D)/32768L;\r | |
216 | \r | |
217 | FB_A0 = g_buffer(rvb.MIX_DEST_A0 - rvb.FB_SRC_A);\r | |
218 | FB_A1 = g_buffer(rvb.MIX_DEST_A1 - rvb.FB_SRC_A);\r | |
219 | FB_B0 = g_buffer(rvb.MIX_DEST_B0 - rvb.FB_SRC_B);\r | |
220 | FB_B1 = g_buffer(rvb.MIX_DEST_B1 - rvb.FB_SRC_B);\r | |
221 | \r | |
222 | s_buffer(rvb.MIX_DEST_A0, ACC0 - (FB_A0 * rvb.FB_ALPHA)/32768L);\r | |
223 | s_buffer(rvb.MIX_DEST_A1, ACC1 - (FB_A1 * rvb.FB_ALPHA)/32768L);\r | |
224 | \r | |
225 | s_buffer(rvb.MIX_DEST_B0, (rvb.FB_ALPHA * ACC0)/32768L - (FB_A0 * (int)(rvb.FB_ALPHA^0xFFFF8000))/32768L - (FB_B0 * rvb.FB_X)/32768L);\r | |
226 | s_buffer(rvb.MIX_DEST_B1, (rvb.FB_ALPHA * ACC1)/32768L - (FB_A1 * (int)(rvb.FB_ALPHA^0xFFFF8000))/32768L - (FB_B1 * rvb.FB_X)/32768L);\r | |
227 | \r | |
228 | rvb.iLastRVBLeft = rvb.iRVBLeft;\r | |
229 | rvb.iLastRVBRight = rvb.iRVBRight;\r | |
230 | \r | |
231 | rvb.iRVBLeft = (g_buffer(rvb.MIX_DEST_A0)+g_buffer(rvb.MIX_DEST_B0))/3;\r | |
232 | rvb.iRVBRight = (g_buffer(rvb.MIX_DEST_A1)+g_buffer(rvb.MIX_DEST_B1))/3;\r | |
233 | \r | |
234 | rvb.iRVBLeft = (rvb.iRVBLeft * rvb.VolLeft) / 0x4000;\r | |
235 | rvb.iRVBRight = (rvb.iRVBRight * rvb.VolRight) / 0x4000;\r | |
236 | \r | |
237 | rvb.CurrAddr++;\r | |
238 | if(rvb.CurrAddr>0x3ffff) rvb.CurrAddr=rvb.StartAddr;\r | |
239 | \r | |
240 | return rvb.iLastRVBLeft+(rvb.iRVBLeft-rvb.iLastRVBLeft)/2;\r | |
241 | }\r | |
242 | else // -> reverb off\r | |
243 | {\r | |
244 | rvb.iLastRVBLeft=rvb.iLastRVBRight=rvb.iRVBLeft=rvb.iRVBRight=0;\r | |
245 | }\r | |
246 | \r | |
247 | rvb.CurrAddr++;\r | |
248 | if(rvb.CurrAddr>0x3ffff) rvb.CurrAddr=rvb.StartAddr;\r | |
249 | }\r | |
250 | \r | |
251 | return rvb.iLastRVBLeft;\r | |
252 | }\r | |
253 | else // easy fake reverb:\r | |
254 | {\r | |
255 | const int iRV=*sRVBPlay; // -> simply take the reverb mix buf value\r | |
256 | *sRVBPlay++=0; // -> init it after\r | |
257 | if(sRVBPlay>=sRVBEnd) sRVBPlay=sRVBStart; // -> and take care about wrap arounds\r | |
258 | return iRV; // -> return reverb mix buf val\r | |
259 | }\r | |
260 | }\r | |
261 | \r | |
262 | ////////////////////////////////////////////////////////////////////////\r | |
263 | \r | |
264 | INLINE int MixREVERBRight(void)\r | |
265 | {\r | |
266 | if(iUseReverb==0) return 0;\r | |
267 | else\r | |
268 | if(iUseReverb==2) // Neill's reverb:\r | |
269 | {\r | |
270 | int i=rvb.iLastRVBRight+(rvb.iRVBRight-rvb.iLastRVBRight)/2;\r | |
271 | rvb.iLastRVBRight=rvb.iRVBRight;\r | |
272 | return i; // -> just return the last right reverb val (little bit scaled by the previous right val)\r | |
273 | }\r | |
274 | else // easy fake reverb:\r | |
275 | {\r | |
276 | const int iRV=*sRVBPlay; // -> simply take the reverb mix buf value\r | |
277 | *sRVBPlay++=0; // -> init it after\r | |
278 | if(sRVBPlay>=sRVBEnd) sRVBPlay=sRVBStart; // -> and take care about wrap arounds\r | |
279 | return iRV; // -> return reverb mix buf val\r | |
280 | }\r | |
281 | }\r | |
282 | \r | |
283 | ////////////////////////////////////////////////////////////////////////\r | |
284 | \r | |
285 | #endif\r | |
286 | \r | |
287 | /*\r | |
288 | -----------------------------------------------------------------------------\r | |
289 | PSX reverb hardware notes\r | |
290 | by Neill Corlett\r | |
291 | -----------------------------------------------------------------------------\r | |
292 | \r | |
293 | Yadda yadda disclaimer yadda probably not perfect yadda well it's okay anyway\r | |
294 | yadda yadda.\r | |
295 | \r | |
296 | -----------------------------------------------------------------------------\r | |
297 | \r | |
298 | Basics\r | |
299 | ------\r | |
300 | \r | |
301 | - The reverb buffer is 22khz 16-bit mono PCM.\r | |
302 | - It starts at the reverb address given by 1DA2, extends to\r | |
303 | the end of sound RAM, and wraps back to the 1DA2 address.\r | |
304 | \r | |
305 | Setting the address at 1DA2 resets the current reverb work address.\r | |
306 | \r | |
307 | This work address ALWAYS increments every 1/22050 sec., regardless of\r | |
308 | whether reverb is enabled (bit 7 of 1DAA set).\r | |
309 | \r | |
310 | And the contents of the reverb buffer ALWAYS play, scaled by the\r | |
311 | "reverberation depth left/right" volumes (1D84/1D86).\r | |
312 | (which, by the way, appear to be scaled so 3FFF=approx. 1.0, 4000=-1.0)\r | |
313 | \r | |
314 | -----------------------------------------------------------------------------\r | |
315 | \r | |
316 | Register names\r | |
317 | --------------\r | |
318 | \r | |
319 | These are probably not their real names.\r | |
320 | These are probably not even correct names.\r | |
321 | We will use them anyway, because we can.\r | |
322 | \r | |
323 | 1DC0: FB_SRC_A (offset)\r | |
324 | 1DC2: FB_SRC_B (offset)\r | |
325 | 1DC4: IIR_ALPHA (coef.)\r | |
326 | 1DC6: ACC_COEF_A (coef.)\r | |
327 | 1DC8: ACC_COEF_B (coef.)\r | |
328 | 1DCA: ACC_COEF_C (coef.)\r | |
329 | 1DCC: ACC_COEF_D (coef.)\r | |
330 | 1DCE: IIR_COEF (coef.)\r | |
331 | 1DD0: FB_ALPHA (coef.)\r | |
332 | 1DD2: FB_X (coef.)\r | |
333 | 1DD4: IIR_DEST_A0 (offset)\r | |
334 | 1DD6: IIR_DEST_A1 (offset)\r | |
335 | 1DD8: ACC_SRC_A0 (offset)\r | |
336 | 1DDA: ACC_SRC_A1 (offset)\r | |
337 | 1DDC: ACC_SRC_B0 (offset)\r | |
338 | 1DDE: ACC_SRC_B1 (offset)\r | |
339 | 1DE0: IIR_SRC_A0 (offset)\r | |
340 | 1DE2: IIR_SRC_A1 (offset)\r | |
341 | 1DE4: IIR_DEST_B0 (offset)\r | |
342 | 1DE6: IIR_DEST_B1 (offset)\r | |
343 | 1DE8: ACC_SRC_C0 (offset)\r | |
344 | 1DEA: ACC_SRC_C1 (offset)\r | |
345 | 1DEC: ACC_SRC_D0 (offset)\r | |
346 | 1DEE: ACC_SRC_D1 (offset)\r | |
347 | 1DF0: IIR_SRC_B1 (offset)\r | |
348 | 1DF2: IIR_SRC_B0 (offset)\r | |
349 | 1DF4: MIX_DEST_A0 (offset)\r | |
350 | 1DF6: MIX_DEST_A1 (offset)\r | |
351 | 1DF8: MIX_DEST_B0 (offset)\r | |
352 | 1DFA: MIX_DEST_B1 (offset)\r | |
353 | 1DFC: IN_COEF_L (coef.)\r | |
354 | 1DFE: IN_COEF_R (coef.)\r | |
355 | \r | |
356 | The coefficients are signed fractional values.\r | |
357 | -32768 would be -1.0\r | |
358 | 32768 would be 1.0 (if it were possible... the highest is of course 32767)\r | |
359 | \r | |
360 | The offsets are (byte/8) offsets into the reverb buffer.\r | |
361 | i.e. you multiply them by 8, you get byte offsets.\r | |
362 | You can also think of them as (samples/4) offsets.\r | |
363 | They appear to be signed. They can be negative.\r | |
364 | None of the documented presets make them negative, though.\r | |
365 | \r | |
366 | Yes, 1DF0 and 1DF2 appear to be backwards. Not a typo.\r | |
367 | \r | |
368 | -----------------------------------------------------------------------------\r | |
369 | \r | |
370 | What it does\r | |
371 | ------------\r | |
372 | \r | |
373 | We take all reverb sources:\r | |
374 | - regular channels that have the reverb bit on\r | |
375 | - cd and external sources, if their reverb bits are on\r | |
376 | and mix them into one stereo 44100hz signal.\r | |
377 | \r | |
378 | Lowpass/downsample that to 22050hz. The PSX uses a proper bandlimiting\r | |
379 | algorithm here, but I haven't figured out the hysterically exact specifics.\r | |
380 | I use an 8-tap filter with these coefficients, which are nice but probably\r | |
381 | not the real ones:\r | |
382 | \r | |
383 | 0.037828187894\r | |
384 | 0.157538631280\r | |
385 | 0.321159685278\r | |
386 | 0.449322115345\r | |
387 | 0.449322115345\r | |
388 | 0.321159685278\r | |
389 | 0.157538631280\r | |
390 | 0.037828187894\r | |
391 | \r | |
392 | So we have two input samples (INPUT_SAMPLE_L, INPUT_SAMPLE_R) every 22050hz.\r | |
393 | \r | |
394 | * IN MY EMULATION, I divide these by 2 to make it clip less.\r | |
395 | (and of course the L/R output coefficients are adjusted to compensate)\r | |
396 | The real thing appears to not do this.\r | |
397 | \r | |
398 | At every 22050hz tick:\r | |
399 | - If the reverb bit is enabled (bit 7 of 1DAA), execute the reverb\r | |
400 | steady-state algorithm described below\r | |
401 | - AFTERWARDS, retrieve the "wet out" L and R samples from the reverb buffer\r | |
402 | (This part may not be exactly right and I guessed at the coefs. TODO: check later.)\r | |
403 | L is: 0.333 * (buffer[MIX_DEST_A0] + buffer[MIX_DEST_B0])\r | |
404 | R is: 0.333 * (buffer[MIX_DEST_A1] + buffer[MIX_DEST_B1])\r | |
405 | - Advance the current buffer position by 1 sample\r | |
406 | \r | |
407 | The wet out L and R are then upsampled to 44100hz and played at the\r | |
408 | "reverberation depth left/right" (1D84/1D86) volume, independent of the main\r | |
409 | volume.\r | |
410 | \r | |
411 | -----------------------------------------------------------------------------\r | |
412 | \r | |
413 | Reverb steady-state\r | |
414 | -------------------\r | |
415 | \r | |
416 | The reverb steady-state algorithm is fairly clever, and of course by\r | |
417 | "clever" I mean "batshit insane".\r | |
418 | \r | |
419 | buffer[x] is relative to the current buffer position, not the beginning of\r | |
420 | the buffer. Note that all buffer offsets must wrap around so they're\r | |
421 | contained within the reverb work area.\r | |
422 | \r | |
423 | Clipping is performed at the end... maybe also sooner, but definitely at\r | |
424 | the end.\r | |
425 | \r | |
426 | IIR_INPUT_A0 = buffer[IIR_SRC_A0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L;\r | |
427 | IIR_INPUT_A1 = buffer[IIR_SRC_A1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R;\r | |
428 | IIR_INPUT_B0 = buffer[IIR_SRC_B0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L;\r | |
429 | IIR_INPUT_B1 = buffer[IIR_SRC_B1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R;\r | |
430 | \r | |
431 | IIR_A0 = IIR_INPUT_A0 * IIR_ALPHA + buffer[IIR_DEST_A0] * (1.0 - IIR_ALPHA);\r | |
432 | IIR_A1 = IIR_INPUT_A1 * IIR_ALPHA + buffer[IIR_DEST_A1] * (1.0 - IIR_ALPHA);\r | |
433 | IIR_B0 = IIR_INPUT_B0 * IIR_ALPHA + buffer[IIR_DEST_B0] * (1.0 - IIR_ALPHA);\r | |
434 | IIR_B1 = IIR_INPUT_B1 * IIR_ALPHA + buffer[IIR_DEST_B1] * (1.0 - IIR_ALPHA);\r | |
435 | \r | |
436 | buffer[IIR_DEST_A0 + 1sample] = IIR_A0;\r | |
437 | buffer[IIR_DEST_A1 + 1sample] = IIR_A1;\r | |
438 | buffer[IIR_DEST_B0 + 1sample] = IIR_B0;\r | |
439 | buffer[IIR_DEST_B1 + 1sample] = IIR_B1;\r | |
440 | \r | |
441 | ACC0 = buffer[ACC_SRC_A0] * ACC_COEF_A +\r | |
442 | buffer[ACC_SRC_B0] * ACC_COEF_B +\r | |
443 | buffer[ACC_SRC_C0] * ACC_COEF_C +\r | |
444 | buffer[ACC_SRC_D0] * ACC_COEF_D;\r | |
445 | ACC1 = buffer[ACC_SRC_A1] * ACC_COEF_A +\r | |
446 | buffer[ACC_SRC_B1] * ACC_COEF_B +\r | |
447 | buffer[ACC_SRC_C1] * ACC_COEF_C +\r | |
448 | buffer[ACC_SRC_D1] * ACC_COEF_D;\r | |
449 | \r | |
450 | FB_A0 = buffer[MIX_DEST_A0 - FB_SRC_A];\r | |
451 | FB_A1 = buffer[MIX_DEST_A1 - FB_SRC_A];\r | |
452 | FB_B0 = buffer[MIX_DEST_B0 - FB_SRC_B];\r | |
453 | FB_B1 = buffer[MIX_DEST_B1 - FB_SRC_B];\r | |
454 | \r | |
455 | buffer[MIX_DEST_A0] = ACC0 - FB_A0 * FB_ALPHA;\r | |
456 | buffer[MIX_DEST_A1] = ACC1 - FB_A1 * FB_ALPHA;\r | |
457 | buffer[MIX_DEST_B0] = (FB_ALPHA * ACC0) - FB_A0 * (FB_ALPHA^0x8000) - FB_B0 * FB_X;\r | |
458 | buffer[MIX_DEST_B1] = (FB_ALPHA * ACC1) - FB_A1 * (FB_ALPHA^0x8000) - FB_B1 * FB_X;\r | |
459 | \r | |
460 | -----------------------------------------------------------------------------\r | |
461 | */\r | |
462 | \r |