Mercurial > audlegacy-plugins
annotate src/psf2/peops2/reverb.c @ 3145:76f5bcf547d5
crossfade: remove unused config variable
author | Michal Lipski <tallica@o2.pl> |
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date | Sat, 09 May 2009 18:51:03 +0200 |
parents | 62cc6d667119 |
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rev | line source |
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1 /*************************************************************************** |
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2 reverb.c - description |
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3 ------------------- |
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4 begin : Wed May 15 2002 |
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5 copyright : (C) 2002 by Pete Bernert |
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6 email : BlackDove@addcom.de |
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7 ***************************************************************************/ |
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8 |
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9 /*************************************************************************** |
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10 * * |
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11 * This program is free software; you can redistribute it and/or modify * |
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12 * it under the terms of the GNU General Public License as published by * |
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13 * the Free Software Foundation; either version 2 of the License, or * |
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14 * (at your option) any later version. See also the license.txt file for * |
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15 * additional informations. * |
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16 * * |
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17 ***************************************************************************/ |
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18 |
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19 //*************************************************************************// |
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20 // History of changes: |
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21 // |
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22 // 2004/04/04 - Pete |
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23 // - changed to SPU2 functionality |
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24 // |
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25 // 2003/01/19 - Pete |
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26 // - added Neill's reverb (see at the end of file) |
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27 // |
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28 // 2002/12/26 - Pete |
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29 // - adjusted reverb handling |
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30 // |
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31 // 2002/08/14 - Pete |
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32 // - added extra reverb |
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33 // |
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34 // 2002/05/15 - Pete |
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35 // - generic cleanup for the Peops release |
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36 // |
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37 //*************************************************************************// |
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38 |
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39 #include "stdafx.h" |
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40 |
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41 #define _IN_REVERB |
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42 |
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43 // will be included from spu.c |
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44 #ifdef _IN_SPU |
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45 |
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46 //////////////////////////////////////////////////////////////////////// |
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47 // globals |
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48 //////////////////////////////////////////////////////////////////////// |
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49 |
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50 // REVERB info and timing vars... |
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51 |
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52 int * sRVBPlay[2]; |
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53 int * sRVBEnd[2]; |
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54 int * sRVBStart[2]; |
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55 |
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56 //////////////////////////////////////////////////////////////////////// |
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57 // START REVERB |
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58 //////////////////////////////////////////////////////////////////////// |
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59 |
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60 INLINE void StartREVERB(int ch) |
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61 { |
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62 int core=ch/24; |
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63 |
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64 if((s_chan[ch].bReverbL || s_chan[ch].bReverbR) && (spuCtrl2[core]&0x80)) // reverb possible? |
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65 { |
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66 if(iUseReverb==1) s_chan[ch].bRVBActive=1; |
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67 } |
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68 else s_chan[ch].bRVBActive=0; // else -> no reverb |
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69 } |
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70 |
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71 //////////////////////////////////////////////////////////////////////// |
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72 // HELPER FOR NEILL'S REVERB: re-inits our reverb mixing buf |
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73 //////////////////////////////////////////////////////////////////////// |
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74 |
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75 INLINE void InitREVERB(void) |
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76 { |
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77 if(iUseReverb==1) |
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78 { |
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79 memset(sRVBStart[0],0,NSSIZE*2*4); |
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80 memset(sRVBStart[1],0,NSSIZE*2*4); |
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81 } |
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82 } |
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83 |
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84 //////////////////////////////////////////////////////////////////////// |
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85 // STORE REVERB |
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86 //////////////////////////////////////////////////////////////////////// |
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87 |
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88 INLINE void StoreREVERB(int ch,int ns) |
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89 { |
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90 int core=ch/24; |
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91 |
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92 if(iUseReverb==0) return; |
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93 else |
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94 if(iUseReverb==1) // -------------------------------- // Neil's reverb |
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95 { |
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96 const int iRxl=(s_chan[ch].sval*s_chan[ch].iLeftVolume*s_chan[ch].bReverbL)/0x4000; |
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97 const int iRxr=(s_chan[ch].sval*s_chan[ch].iRightVolume*s_chan[ch].bReverbR)/0x4000; |
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98 |
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99 ns<<=1; |
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100 |
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101 *(sRVBStart[core]+ns) +=iRxl; // -> we mix all active reverb channels into an extra buffer |
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102 *(sRVBStart[core]+ns+1)+=iRxr; |
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103 } |
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104 } |
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105 |
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106 //////////////////////////////////////////////////////////////////////// |
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107 |
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108 INLINE int g_buffer(int iOff,int core) // get_buffer content helper: takes care about wraps |
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109 { |
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110 short * p=(short *)spuMem; |
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111 iOff=(iOff)+rvb[core].CurrAddr; |
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112 while(iOff>rvb[core].EndAddr) iOff=rvb[core].StartAddr+(iOff-(rvb[core].EndAddr+1)); |
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113 while(iOff<rvb[core].StartAddr) iOff=rvb[core].EndAddr-(rvb[core].StartAddr-iOff); |
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114 return (int)*(p+iOff); |
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115 } |
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116 |
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117 //////////////////////////////////////////////////////////////////////// |
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118 |
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119 INLINE void s_buffer(int iOff,int iVal,int core) // set_buffer content helper: takes care about wraps and clipping |
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120 { |
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121 short * p=(short *)spuMem; |
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122 iOff=(iOff)+rvb[core].CurrAddr; |
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123 while(iOff>rvb[core].EndAddr) iOff=rvb[core].StartAddr+(iOff-(rvb[core].EndAddr+1)); |
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124 while(iOff<rvb[core].StartAddr) iOff=rvb[core].EndAddr-(rvb[core].StartAddr-iOff); |
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125 if(iVal<-32768L) iVal=-32768L;if(iVal>32767L) iVal=32767L; |
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126 *(p+iOff)=(short)iVal; |
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127 } |
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128 |
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129 //////////////////////////////////////////////////////////////////////// |
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130 |
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131 INLINE void s_buffer1(int iOff,int iVal,int core) // set_buffer (+1 sample) content helper: takes care about wraps and clipping |
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132 { |
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133 short * p=(short *)spuMem; |
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134 iOff=(iOff)+rvb[core].CurrAddr+1; |
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135 while(iOff>rvb[core].EndAddr) iOff=rvb[core].StartAddr+(iOff-(rvb[core].EndAddr+1)); |
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136 while(iOff<rvb[core].StartAddr) iOff=rvb[core].EndAddr-(rvb[core].StartAddr-iOff); |
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137 if(iVal<-32768L) iVal=-32768L;if(iVal>32767L) iVal=32767L; |
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138 *(p+iOff)=(short)iVal; |
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139 } |
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140 |
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141 //////////////////////////////////////////////////////////////////////// |
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142 |
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143 INLINE int MixREVERBLeft(int ns,int core) |
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144 { |
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145 if(iUseReverb==1) |
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146 { |
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147 if(!rvb[core].StartAddr || !rvb[core].EndAddr || |
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148 rvb[core].StartAddr>=rvb[core].EndAddr) // reverb is off |
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149 { |
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150 rvb[core].iLastRVBLeft=rvb[core].iLastRVBRight=rvb[core].iRVBLeft=rvb[core].iRVBRight=0; |
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151 return 0; |
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152 } |
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153 |
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154 rvb[core].iCnt++; |
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155 |
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156 if(rvb[core].iCnt&1) // we work on every second left value: downsample to 22 khz |
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157 { |
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158 if((spuCtrl2[core]&0x80)) // -> reverb on? oki |
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159 { |
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160 int ACC0,ACC1,FB_A0,FB_A1,FB_B0,FB_B1; |
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161 |
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162 const int INPUT_SAMPLE_L=*(sRVBStart[core]+(ns<<1)); |
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163 const int INPUT_SAMPLE_R=*(sRVBStart[core]+(ns<<1)+1); |
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164 |
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165 const int IIR_INPUT_A0 = (g_buffer(rvb[core].IIR_SRC_A0,core) * rvb[core].IIR_COEF)/32768L + (INPUT_SAMPLE_L * rvb[core].IN_COEF_L)/32768L; |
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166 const int IIR_INPUT_A1 = (g_buffer(rvb[core].IIR_SRC_A1,core) * rvb[core].IIR_COEF)/32768L + (INPUT_SAMPLE_R * rvb[core].IN_COEF_R)/32768L; |
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167 const int IIR_INPUT_B0 = (g_buffer(rvb[core].IIR_SRC_B0,core) * rvb[core].IIR_COEF)/32768L + (INPUT_SAMPLE_L * rvb[core].IN_COEF_L)/32768L; |
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168 const int IIR_INPUT_B1 = (g_buffer(rvb[core].IIR_SRC_B1,core) * rvb[core].IIR_COEF)/32768L + (INPUT_SAMPLE_R * rvb[core].IN_COEF_R)/32768L; |
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169 |
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170 const int IIR_A0 = (IIR_INPUT_A0 * rvb[core].IIR_ALPHA)/32768L + (g_buffer(rvb[core].IIR_DEST_A0,core) * (32768L - rvb[core].IIR_ALPHA))/32768L; |
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171 const int IIR_A1 = (IIR_INPUT_A1 * rvb[core].IIR_ALPHA)/32768L + (g_buffer(rvb[core].IIR_DEST_A1,core) * (32768L - rvb[core].IIR_ALPHA))/32768L; |
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172 const int IIR_B0 = (IIR_INPUT_B0 * rvb[core].IIR_ALPHA)/32768L + (g_buffer(rvb[core].IIR_DEST_B0,core) * (32768L - rvb[core].IIR_ALPHA))/32768L; |
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173 const int IIR_B1 = (IIR_INPUT_B1 * rvb[core].IIR_ALPHA)/32768L + (g_buffer(rvb[core].IIR_DEST_B1,core) * (32768L - rvb[core].IIR_ALPHA))/32768L; |
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174 |
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175 s_buffer1(rvb[core].IIR_DEST_A0, IIR_A0,core); |
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176 s_buffer1(rvb[core].IIR_DEST_A1, IIR_A1,core); |
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177 s_buffer1(rvb[core].IIR_DEST_B0, IIR_B0,core); |
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178 s_buffer1(rvb[core].IIR_DEST_B1, IIR_B1,core); |
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179 |
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180 ACC0 = (g_buffer(rvb[core].ACC_SRC_A0,core) * rvb[core].ACC_COEF_A)/32768L + |
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181 (g_buffer(rvb[core].ACC_SRC_B0,core) * rvb[core].ACC_COEF_B)/32768L + |
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182 (g_buffer(rvb[core].ACC_SRC_C0,core) * rvb[core].ACC_COEF_C)/32768L + |
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183 (g_buffer(rvb[core].ACC_SRC_D0,core) * rvb[core].ACC_COEF_D)/32768L; |
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184 ACC1 = (g_buffer(rvb[core].ACC_SRC_A1,core) * rvb[core].ACC_COEF_A)/32768L + |
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185 (g_buffer(rvb[core].ACC_SRC_B1,core) * rvb[core].ACC_COEF_B)/32768L + |
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186 (g_buffer(rvb[core].ACC_SRC_C1,core) * rvb[core].ACC_COEF_C)/32768L + |
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187 (g_buffer(rvb[core].ACC_SRC_D1,core) * rvb[core].ACC_COEF_D)/32768L; |
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188 |
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189 FB_A0 = g_buffer(rvb[core].MIX_DEST_A0 - rvb[core].FB_SRC_A,core); |
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190 FB_A1 = g_buffer(rvb[core].MIX_DEST_A1 - rvb[core].FB_SRC_A,core); |
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191 FB_B0 = g_buffer(rvb[core].MIX_DEST_B0 - rvb[core].FB_SRC_B,core); |
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192 FB_B1 = g_buffer(rvb[core].MIX_DEST_B1 - rvb[core].FB_SRC_B,core); |
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193 |
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194 s_buffer(rvb[core].MIX_DEST_A0, ACC0 - (FB_A0 * rvb[core].FB_ALPHA)/32768L,core); |
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195 s_buffer(rvb[core].MIX_DEST_A1, ACC1 - (FB_A1 * rvb[core].FB_ALPHA)/32768L,core); |
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196 |
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197 s_buffer(rvb[core].MIX_DEST_B0, (rvb[core].FB_ALPHA * ACC0)/32768L - (FB_A0 * (int)(rvb[core].FB_ALPHA^0xFFFF8000))/32768L - (FB_B0 * rvb[core].FB_X)/32768L,core); |
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198 s_buffer(rvb[core].MIX_DEST_B1, (rvb[core].FB_ALPHA * ACC1)/32768L - (FB_A1 * (int)(rvb[core].FB_ALPHA^0xFFFF8000))/32768L - (FB_B1 * rvb[core].FB_X)/32768L,core); |
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199 |
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200 rvb[core].iLastRVBLeft = rvb[core].iRVBLeft; |
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201 rvb[core].iLastRVBRight = rvb[core].iRVBRight; |
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202 |
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203 rvb[core].iRVBLeft = (g_buffer(rvb[core].MIX_DEST_A0,core)+g_buffer(rvb[core].MIX_DEST_B0,core))/3; |
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204 rvb[core].iRVBRight = (g_buffer(rvb[core].MIX_DEST_A1,core)+g_buffer(rvb[core].MIX_DEST_B1,core))/3; |
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205 |
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206 rvb[core].iRVBLeft = (rvb[core].iRVBLeft * rvb[core].VolLeft) / 0x4000; |
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207 rvb[core].iRVBRight = (rvb[core].iRVBRight * rvb[core].VolRight) / 0x4000; |
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208 |
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209 rvb[core].CurrAddr++; |
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210 if(rvb[core].CurrAddr>rvb[core].EndAddr) rvb[core].CurrAddr=rvb[core].StartAddr; |
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211 |
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212 return rvb[core].iLastRVBLeft+(rvb[core].iRVBLeft-rvb[core].iLastRVBLeft)/2; |
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213 } |
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214 else // -> reverb off |
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215 { |
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216 rvb[core].iLastRVBLeft=rvb[core].iLastRVBRight=rvb[core].iRVBLeft=rvb[core].iRVBRight=0; |
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217 } |
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218 |
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219 rvb[core].CurrAddr++; |
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220 if(rvb[core].CurrAddr>rvb[core].EndAddr) rvb[core].CurrAddr=rvb[core].StartAddr; |
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221 } |
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222 |
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223 return rvb[core].iLastRVBLeft; |
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224 } |
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225 return 0; |
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226 } |
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227 |
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228 //////////////////////////////////////////////////////////////////////// |
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229 |
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230 INLINE int MixREVERBRight(int core) |
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231 { |
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232 if(iUseReverb==1) // Neill's reverb: |
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233 { |
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234 int i=rvb[core].iLastRVBRight+(rvb[core].iRVBRight-rvb[core].iLastRVBRight)/2; |
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235 rvb[core].iLastRVBRight=rvb[core].iRVBRight; |
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236 return i; // -> just return the last right reverb val (little bit scaled by the previous right val) |
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237 } |
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238 return 0; |
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239 } |
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240 |
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241 //////////////////////////////////////////////////////////////////////// |
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242 |
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243 #endif |
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244 |
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245 /* |
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246 ----------------------------------------------------------------------------- |
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247 PSX reverb hardware notes |
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248 by Neill Corlett |
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249 ----------------------------------------------------------------------------- |
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250 |
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251 Yadda yadda disclaimer yadda probably not perfect yadda well it's okay anyway |
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252 yadda yadda. |
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253 |
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254 ----------------------------------------------------------------------------- |
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255 |
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256 Basics |
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257 ------ |
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258 |
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259 - The reverb buffer is 22khz 16-bit mono PCM. |
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260 - It starts at the reverb address given by 1DA2, extends to |
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261 the end of sound RAM, and wraps back to the 1DA2 address. |
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262 |
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263 Setting the address at 1DA2 resets the current reverb work address. |
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264 |
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265 This work address ALWAYS increments every 1/22050 sec., regardless of |
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266 whether reverb is enabled (bit 7 of 1DAA set). |
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267 |
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268 And the contents of the reverb buffer ALWAYS play, scaled by the |
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269 "reverberation depth left/right" volumes (1D84/1D86). |
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270 (which, by the way, appear to be scaled so 3FFF=approx. 1.0, 4000=-1.0) |
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271 |
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272 ----------------------------------------------------------------------------- |
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273 |
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274 Register names |
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275 -------------- |
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276 |
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277 These are probably not their real names. |
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278 These are probably not even correct names. |
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279 We will use them anyway, because we can. |
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280 |
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281 1DC0: FB_SRC_A (offset) |
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282 1DC2: FB_SRC_B (offset) |
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283 1DC4: IIR_ALPHA (coef.) |
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284 1DC6: ACC_COEF_A (coef.) |
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285 1DC8: ACC_COEF_B (coef.) |
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286 1DCA: ACC_COEF_C (coef.) |
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287 1DCC: ACC_COEF_D (coef.) |
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288 1DCE: IIR_COEF (coef.) |
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289 1DD0: FB_ALPHA (coef.) |
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290 1DD2: FB_X (coef.) |
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291 1DD4: IIR_DEST_A0 (offset) |
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292 1DD6: IIR_DEST_A1 (offset) |
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293 1DD8: ACC_SRC_A0 (offset) |
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294 1DDA: ACC_SRC_A1 (offset) |
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295 1DDC: ACC_SRC_B0 (offset) |
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296 1DDE: ACC_SRC_B1 (offset) |
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297 1DE0: IIR_SRC_A0 (offset) |
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298 1DE2: IIR_SRC_A1 (offset) |
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299 1DE4: IIR_DEST_B0 (offset) |
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300 1DE6: IIR_DEST_B1 (offset) |
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301 1DE8: ACC_SRC_C0 (offset) |
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302 1DEA: ACC_SRC_C1 (offset) |
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303 1DEC: ACC_SRC_D0 (offset) |
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304 1DEE: ACC_SRC_D1 (offset) |
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305 1DF0: IIR_SRC_B1 (offset) |
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306 1DF2: IIR_SRC_B0 (offset) |
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307 1DF4: MIX_DEST_A0 (offset) |
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308 1DF6: MIX_DEST_A1 (offset) |
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309 1DF8: MIX_DEST_B0 (offset) |
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310 1DFA: MIX_DEST_B1 (offset) |
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311 1DFC: IN_COEF_L (coef.) |
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312 1DFE: IN_COEF_R (coef.) |
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313 |
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314 The coefficients are signed fractional values. |
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315 -32768 would be -1.0 |
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316 32768 would be 1.0 (if it were possible... the highest is of course 32767) |
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317 |
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318 The offsets are (byte/8) offsets into the reverb buffer. |
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319 i.e. you multiply them by 8, you get byte offsets. |
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320 You can also think of them as (samples/4) offsets. |
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321 They appear to be signed. They can be negative. |
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322 None of the documented presets make them negative, though. |
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323 |
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324 Yes, 1DF0 and 1DF2 appear to be backwards. Not a typo. |
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325 |
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326 ----------------------------------------------------------------------------- |
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327 |
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328 What it does |
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329 ------------ |
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330 |
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331 We take all reverb sources: |
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332 - regular channels that have the reverb bit on |
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333 - cd and external sources, if their reverb bits are on |
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334 and mix them into one stereo 44100hz signal. |
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335 |
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336 Lowpass/downsample that to 22050hz. The PSX uses a proper bandlimiting |
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337 algorithm here, but I haven't figured out the hysterically exact specifics. |
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338 I use an 8-tap filter with these coefficients, which are nice but probably |
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339 not the real ones: |
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340 |
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341 0.037828187894 |
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342 0.157538631280 |
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343 0.321159685278 |
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344 0.449322115345 |
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345 0.449322115345 |
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346 0.321159685278 |
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347 0.157538631280 |
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348 0.037828187894 |
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349 |
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350 So we have two input samples (INPUT_SAMPLE_L, INPUT_SAMPLE_R) every 22050hz. |
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351 |
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352 * IN MY EMULATION, I divide these by 2 to make it clip less. |
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353 (and of course the L/R output coefficients are adjusted to compensate) |
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354 The real thing appears to not do this. |
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355 |
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356 At every 22050hz tick: |
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357 - If the reverb bit is enabled (bit 7 of 1DAA), execute the reverb |
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358 steady-state algorithm described below |
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359 - AFTERWARDS, retrieve the "wet out" L and R samples from the reverb buffer |
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360 (This part may not be exactly right and I guessed at the coefs. TODO: check later.) |
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361 L is: 0.333 * (buffer[MIX_DEST_A0] + buffer[MIX_DEST_B0]) |
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362 R is: 0.333 * (buffer[MIX_DEST_A1] + buffer[MIX_DEST_B1]) |
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363 - Advance the current buffer position by 1 sample |
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364 |
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365 The wet out L and R are then upsampled to 44100hz and played at the |
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366 "reverberation depth left/right" (1D84/1D86) volume, independent of the main |
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367 volume. |
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368 |
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369 ----------------------------------------------------------------------------- |
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370 |
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371 Reverb steady-state |
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372 ------------------- |
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373 |
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374 The reverb steady-state algorithm is fairly clever, and of course by |
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375 "clever" I mean "batshit insane". |
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376 |
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377 buffer[x] is relative to the current buffer position, not the beginning of |
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378 the buffer. Note that all buffer offsets must wrap around so they're |
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379 contained within the reverb work area. |
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380 |
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381 Clipping is performed at the end... maybe also sooner, but definitely at |
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382 the end. |
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383 |
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384 IIR_INPUT_A0 = buffer[IIR_SRC_A0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L; |
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385 IIR_INPUT_A1 = buffer[IIR_SRC_A1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R; |
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386 IIR_INPUT_B0 = buffer[IIR_SRC_B0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L; |
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387 IIR_INPUT_B1 = buffer[IIR_SRC_B1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R; |
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388 |
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389 IIR_A0 = IIR_INPUT_A0 * IIR_ALPHA + buffer[IIR_DEST_A0] * (1.0 - IIR_ALPHA); |
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390 IIR_A1 = IIR_INPUT_A1 * IIR_ALPHA + buffer[IIR_DEST_A1] * (1.0 - IIR_ALPHA); |
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391 IIR_B0 = IIR_INPUT_B0 * IIR_ALPHA + buffer[IIR_DEST_B0] * (1.0 - IIR_ALPHA); |
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392 IIR_B1 = IIR_INPUT_B1 * IIR_ALPHA + buffer[IIR_DEST_B1] * (1.0 - IIR_ALPHA); |
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393 |
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394 buffer[IIR_DEST_A0 + 1sample] = IIR_A0; |
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395 buffer[IIR_DEST_A1 + 1sample] = IIR_A1; |
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396 buffer[IIR_DEST_B0 + 1sample] = IIR_B0; |
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397 buffer[IIR_DEST_B1 + 1sample] = IIR_B1; |
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398 |
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399 ACC0 = buffer[ACC_SRC_A0] * ACC_COEF_A + |
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400 buffer[ACC_SRC_B0] * ACC_COEF_B + |
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401 buffer[ACC_SRC_C0] * ACC_COEF_C + |
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402 buffer[ACC_SRC_D0] * ACC_COEF_D; |
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403 ACC1 = buffer[ACC_SRC_A1] * ACC_COEF_A + |
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404 buffer[ACC_SRC_B1] * ACC_COEF_B + |
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405 buffer[ACC_SRC_C1] * ACC_COEF_C + |
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406 buffer[ACC_SRC_D1] * ACC_COEF_D; |
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407 |
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408 FB_A0 = buffer[MIX_DEST_A0 - FB_SRC_A]; |
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409 FB_A1 = buffer[MIX_DEST_A1 - FB_SRC_A]; |
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410 FB_B0 = buffer[MIX_DEST_B0 - FB_SRC_B]; |
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411 FB_B1 = buffer[MIX_DEST_B1 - FB_SRC_B]; |
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412 |
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413 buffer[MIX_DEST_A0] = ACC0 - FB_A0 * FB_ALPHA; |
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414 buffer[MIX_DEST_A1] = ACC1 - FB_A1 * FB_ALPHA; |
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415 buffer[MIX_DEST_B0] = (FB_ALPHA * ACC0) - FB_A0 * (FB_ALPHA^0x8000) - FB_B0 * FB_X; |
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416 buffer[MIX_DEST_B1] = (FB_ALPHA * ACC1) - FB_A1 * (FB_ALPHA^0x8000) - FB_B1 * FB_X; |
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417 |
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418 ----------------------------------------------------------------------------- |
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419 */ |
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420 |