| 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 |