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annotate libaf/af_surround.c @ 15130:ba68f417b416

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author gpoirier
date Mon, 11 Apr 2005 21:40:05 +0000 (2005-04-11)
parents de13fd557440
children cea0eb833758
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8678
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1 /*
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2 This is an libaf filter to do simple decoding of matrixed surround
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3 sound. This will provide a (basic) surround-sound effect from
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4 audio encoded for Dolby Surround, Pro Logic etc.
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5
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6 * This program is free software; you can redistribute it and/or modify
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7 * it under the terms of the GNU General Public License as published by
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8 * the Free Software Foundation; either version 2 of the License, or
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9 * (at your option) any later version.
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10 *
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11 * This program is distributed in the hope that it will be useful,
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12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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14 * GNU General Public License for more details.
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15 *
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16 * You should have received a copy of the GNU General Public License
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17 * along with this program; if not, write to the Free Software
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18 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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19
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20 Original author: Steve Davies <steve@daviesfam.org>
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21 */
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22
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23 /* The principle: Make rear channels by extracting anti-phase data
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24 from the front channels, delay by 20ms and feed to rear in anti-phase
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25 */
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26
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27
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28 /* SPLITREAR: Define to decode two distinct rear channels - this
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29 doesn't work so well in practice because separation in a passive
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30 matrix is not high. C (dialogue) to Ls and Rs 14dB or so - so
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31 dialogue leaks to the rear. Still - give it a try and send
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32 feedback. Comment this define for old behavior of a single
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33 surround sent to rear in anti-phase */
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34 #define SPLITREAR 1
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35
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36 #include <stdio.h>
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37 #include <stdlib.h>
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38 #include <string.h>
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39 #include <unistd.h>
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40
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41 #include "af.h"
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42 #include "dsp.h"
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43
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44 #define L 32 // Length of fir filter
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45 #define LD 65536 // Length of delay buffer
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46
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47 // 32 Tap fir filter loop unrolled
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48 #define FIR(x,w,y) \
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49 y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
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50 + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
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51 + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
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52 + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \
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53 + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \
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54 + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \
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55 + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \
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56 + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31])
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57
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58 // Add to circular queue macro + update index
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59 #ifdef SPLITREAR
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60 #define ADDQUE(qi,rq,lq,r,l)\
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61 lq[qi]=lq[qi+L]=(l);\
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62 rq[qi]=rq[qi+L]=(r);\
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63 qi=(qi-1)&(L-1);
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64 #else
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65 #define ADDQUE(qi,lq,l)\
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66 lq[qi]=lq[qi+L]=(l);\
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67 qi=(qi-1)&(L-1);
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68 #endif
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69
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70 // Macro for updating queue index in delay queues
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71 #define UPDATEQI(qi) qi=(qi+1)&(LD-1)
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72
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73 // instance data
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74 typedef struct af_surround_s
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75 {
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76 float lq[2*L]; // Circular queue for filtering left rear channel
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77 float rq[2*L]; // Circular queue for filtering right rear channel
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78 float w[L]; // FIR filter coefficients for surround sound 7kHz low-pass
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79 float* dr; // Delay queue right rear channel
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80 float* dl; // Delay queue left rear channel
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81 float d; // Delay time
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82 int i; // Position in circular buffer
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83 int wi; // Write index for delay queue
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84 int ri; // Read index for delay queue
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85 }af_surround_t;
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86
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87 // Initialization and runtime control
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88 static int control(struct af_instance_s* af, int cmd, void* arg)
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89 {
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90 af_surround_t *s = af->setup;
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91 switch(cmd){
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92 case AF_CONTROL_REINIT:{
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93 float fc;
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94 af->data->rate = ((af_data_t*)arg)->rate;
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95 af->data->nch = ((af_data_t*)arg)->nch*2;
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96 af->data->format = AF_FORMAT_FLOAT_NE;
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97 af->data->bps = 4;
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98
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99 if (af->data->nch != 4){
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100 af_msg(AF_MSG_ERROR,"[surround] Only stereo input is supported.\n");
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101 return AF_DETACH;
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102 }
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103 // Surround filer coefficients
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104 fc = 2.0 * 7000.0/(float)af->data->rate;
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105 if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
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106 af_msg(AF_MSG_ERROR,"[surround] Unable to design low-pass filter.\n");
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107 return AF_ERROR;
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108 }
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109
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110 // Free previous delay queues
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111 if(s->dl)
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112 free(s->dl);
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113 if(s->dr)
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114 free(s->dr);
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115 // Allocate new delay queues
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116 s->dl = calloc(LD,af->data->bps);
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117 s->dr = calloc(LD,af->data->bps);
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118 if((NULL == s->dl) || (NULL == s->dr))
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119 af_msg(AF_MSG_FATAL,"[delay] Out of memory\n");
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120
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121 // Initialize delay queue index
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122 if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
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123 return AF_ERROR;
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124 // printf("%i\n",s->wi);
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125 s->ri = 0;
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126
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127 if((af->data->format != ((af_data_t*)arg)->format) ||
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128 (af->data->bps != ((af_data_t*)arg)->bps)){
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129 ((af_data_t*)arg)->format = af->data->format;
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130 ((af_data_t*)arg)->bps = af->data->bps;
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131 return AF_FALSE;
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132 }
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133 return AF_OK;
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134 }
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135 case AF_CONTROL_COMMAND_LINE:{
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136 float d = 0;
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137 sscanf((char*)arg,"%f",&d);
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138 if ((d < 0) || (d > 1000)){
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139 af_msg(AF_MSG_ERROR,"[surround] Invalid delay time, valid time values"
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caec33353477 1000l and I imho this would win 'The Bug/Typo of the Year' award
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140 " are 0ms to 1000ms current value is %0.3f ms\n",d);
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141 return AF_ERROR;
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142 }
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143 s->d = d;
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144 return AF_OK;
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145 }
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146 }
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147 return AF_UNKNOWN;
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148 }
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149
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150 // Deallocate memory
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151 static void uninit(struct af_instance_s* af)
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152 {
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153 if(af->data->audio)
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154 free(af->data->audio);
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155 if(af->data)
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156 free(af->data);
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157 if(af->setup)
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158 free(af->setup);
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159 }
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160
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161 // The beginnings of an active matrix...
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162 static float steering_matrix[][12] = {
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163 // LL RL LR RR LS RS
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164 // LLs RLs LRs RRs LC RC
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165 {.707, .0, .0, .707, .5, -.5,
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166 .5878, -.3928, .3928, -.5878, .5, .5},
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167 };
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168
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169 // Experimental moving average dominance
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170 //static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
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171
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172 // Filter data through filter
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173 static af_data_t* play(struct af_instance_s* af, af_data_t* data){
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174 af_surround_t* s = (af_surround_t*)af->setup;
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175 float* m = steering_matrix[0];
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176 float* in = data->audio; // Input audio data
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177 float* out = NULL; // Output audio data
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178 float* end = in + data->len / sizeof(float); // Loop end
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179 int i = s->i; // Filter queue index
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180 int ri = s->ri; // Read index for delay queue
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181 int wi = s->wi; // Write index for delay queue
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182
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183 if (AF_OK != RESIZE_LOCAL_BUFFER(af, data))
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184 return NULL;
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185
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186 out = af->data->audio;
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187
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188 while(in < end){
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189 /* Dominance:
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190 abs(in[0]) abs(in[1]);
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191 abs(in[0]+in[1]) abs(in[0]-in[1]);
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192 10 * log( abs(in[0]) / (abs(in[1])|1) );
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193 10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */
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194
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195 /* About volume balancing...
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196 Surround encoding does the following:
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197 Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
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198 So S should be extracted as:
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199 (Lt-Rt)
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200 But we are splitting the S to two output channels, so we
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201 must take 3dB off as we split it:
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202 Ls=Rs=.707*(Lt-Rt)
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203 Trouble is, Lt could be +1, Rt -1, so possibility that S will
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204 overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by
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205 6dB (/2). This keeps the overall balance, but guarantees no
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206 overflow. */
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207
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208 // Output front left and right
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209 out[0] = m[0]*in[0] + m[1]*in[1];
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210 out[1] = m[2]*in[0] + m[3]*in[1];
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211
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212 // Low-pass output @ 7kHz
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213 FIR((&s->lq[i]), s->w, s->dl[wi]);
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214
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215 // Delay output by d ms
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216 out[2] = s->dl[ri];
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217
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218 #ifdef SPLITREAR
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219 // Low-pass output @ 7kHz
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220 FIR((&s->rq[i]), s->w, s->dr[wi]);
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221
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222 // Delay output by d ms
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diff changeset
223 out[3] = s->dr[ri];
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224 #else
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225 out[3] = -out[2];
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226 #endif
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227
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228 // Update delay queues indexes
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229 UPDATEQI(ri);
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230 UPDATEQI(wi);
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231
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232 // Calculate and save surround in circular queue
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233 #ifdef SPLITREAR
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234 ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
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235 #else
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236 ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
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237 #endif
db0810dcab33 Port of pl_surround.c to af-layer.
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238
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239 // Next sample...
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240 in = &in[data->nch];
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241 out = &out[af->data->nch];
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242 }
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243
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244 // Save indexes
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245 s->i = i; s->ri = ri; s->wi = wi;
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246
db0810dcab33 Port of pl_surround.c to af-layer.
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247 // Set output data
db0810dcab33 Port of pl_surround.c to af-layer.
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248 data->audio = af->data->audio;
db0810dcab33 Port of pl_surround.c to af-layer.
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249 data->len = (data->len*af->mul.n)/af->mul.d;
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250 data->nch = af->data->nch;
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251
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252 return data;
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253 }
db0810dcab33 Port of pl_surround.c to af-layer.
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254
db0810dcab33 Port of pl_surround.c to af-layer.
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255 static int open(af_instance_t* af){
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256 af->control=control;
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257 af->uninit=uninit;
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258 af->play=play;
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259 af->mul.n=2;
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260 af->mul.d=1;
db0810dcab33 Port of pl_surround.c to af-layer.
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261 af->data=calloc(1,sizeof(af_data_t));
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262 af->setup=calloc(1,sizeof(af_surround_t));
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263 if(af->data == NULL || af->setup == NULL)
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264 return AF_ERROR;
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diff changeset
265 ((af_surround_t*)af->setup)->d = 20;
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266 return AF_OK;
db0810dcab33 Port of pl_surround.c to af-layer.
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267 }
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268
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269 af_info_t af_info_surround =
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270 {
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271 "Surround decoder filter",
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272 "surround",
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273 "Steve Davies <steve@daviesfam.org>",
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274 "",
8763
19e96e60a3d0 Speed optimizations (runs twise as fast) and bugfix (wrong cutoff frequency buffer over run noise and garbeled output when wrong input format)
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275 AF_FLAGS_NOT_REENTRANT,
8678
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276 open
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277 };