Mercurial > mplayer.hg
annotate libaf/af_surround.c @ 31538:09e7a988ab93
Fix resizing with -wid after previous commit.
author | reimar |
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date | Thu, 01 Jul 2010 20:00:19 +0000 |
parents | 0f1b5b68af32 |
children | 8fa2f43cb760 |
rev | line source |
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1 /* |
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2 * Filter to do simple decoding of matrixed surround sound. |
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3 * 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 * original author: Steve Davies <steve@daviesfam.org> |
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7 * |
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8 * This file is part of MPlayer. |
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9 * |
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10 * MPlayer is free software; you can redistribute it and/or modify |
8678 | 11 * it under the terms of the GNU General Public License as published by |
12 * the Free Software Foundation; either version 2 of the License, or | |
13 * (at your option) any later version. | |
14 * | |
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15 * MPlayer is distributed in the hope that it will be useful, |
8678 | 16 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the | |
18 * GNU General Public License for more details. | |
19 * | |
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20 * You should have received a copy of the GNU General Public License along |
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21 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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22 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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23 */ |
8678 | 24 |
25 /* The principle: Make rear channels by extracting anti-phase data | |
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26 from the front channels, delay by 20ms and feed to rear in anti-phase |
8678 | 27 */ |
28 | |
29 | |
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30 /* SPLITREAR: Define to decode two distinct rear channels - this |
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31 doesn't work so well in practice because separation in a passive |
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32 matrix is not high. C (dialogue) to Ls and Rs 14dB or so - so |
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33 dialogue leaks to the rear. Still - give it a try and send |
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34 feedback. Comment this define for old behavior of a single |
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35 surround sent to rear in anti-phase */ |
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36 #define SPLITREAR 1 |
8678 | 37 |
38 #include <stdio.h> | |
39 #include <stdlib.h> | |
40 #include <string.h> | |
41 | |
42 #include "af.h" | |
43 #include "dsp.h" | |
44 | |
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45 #define L 32 // Length of fir filter |
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46 #define LD 65536 // Length of delay buffer |
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47 |
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48 // 32 Tap fir filter loop unrolled |
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49 #define FIR(x,w,y) \ |
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50 y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \ |
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51 + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \ |
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52 + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \ |
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53 + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \ |
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54 + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \ |
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55 + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \ |
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56 + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \ |
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57 + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31]) |
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58 |
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59 // Add to circular queue macro + update index |
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60 #ifdef SPLITREAR |
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61 #define ADDQUE(qi,rq,lq,r,l)\ |
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62 lq[qi]=lq[qi+L]=(l);\ |
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63 rq[qi]=rq[qi+L]=(r);\ |
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64 qi=(qi-1)&(L-1); |
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65 #else |
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66 #define ADDQUE(qi,lq,l)\ |
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67 lq[qi]=lq[qi+L]=(l);\ |
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68 qi=(qi-1)&(L-1); |
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69 #endif |
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70 |
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71 // Macro for updating queue index in delay queues |
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72 #define UPDATEQI(qi) qi=(qi+1)&(LD-1) |
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73 |
8678 | 74 // instance data |
75 typedef struct af_surround_s | |
76 { | |
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77 float lq[2*L]; // Circular queue for filtering left rear channel |
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78 float rq[2*L]; // Circular queue for filtering right rear channel |
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79 float w[L]; // FIR filter coefficients for surround sound 7kHz low-pass |
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80 float* dr; // Delay queue right rear channel |
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81 float* dl; // Delay queue left rear channel |
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82 float d; // Delay time |
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83 int i; // Position in circular buffer |
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84 int wi; // Write index for delay queue |
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85 int ri; // Read index for delay queue |
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86 }af_surround_t; |
8678 | 87 |
88 // Initialization and runtime control | |
89 static int control(struct af_instance_s* af, int cmd, void* arg) | |
90 { | |
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91 af_surround_t *s = af->setup; |
8678 | 92 switch(cmd){ |
93 case AF_CONTROL_REINIT:{ | |
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94 float fc; |
8678 | 95 af->data->rate = ((af_data_t*)arg)->rate; |
96 af->data->nch = ((af_data_t*)arg)->nch*2; | |
14245 | 97 af->data->format = AF_FORMAT_FLOAT_NE; |
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98 af->data->bps = 4; |
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99 |
8678 | 100 if (af->data->nch != 4){ |
29049 | 101 mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Only stereo input is supported.\n"); |
8678 | 102 return AF_DETACH; |
103 } | |
104 // Surround filer coefficients | |
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105 fc = 2.0 * 7000.0/(float)af->data->rate; |
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106 if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){ |
29049 | 107 mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Unable to design low-pass filter.\n"); |
8678 | 108 return AF_ERROR; |
109 } | |
110 | |
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111 // Free previous delay queues |
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112 if(s->dl) |
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113 free(s->dl); |
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114 if(s->dr) |
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115 free(s->dr); |
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116 // Allocate new delay queues |
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117 s->dl = calloc(LD,af->data->bps); |
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118 s->dr = calloc(LD,af->data->bps); |
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119 if((NULL == s->dl) || (NULL == s->dr)) |
29049 | 120 mp_msg(MSGT_AFILTER, MSGL_FATAL, "[delay] Out of memory\n"); |
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121 |
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122 // Initialize delay queue index |
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123 if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0)) |
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124 return AF_ERROR; |
12569 | 125 // printf("%i\n",s->wi); |
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126 s->ri = 0; |
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127 |
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128 if((af->data->format != ((af_data_t*)arg)->format) || |
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129 (af->data->bps != ((af_data_t*)arg)->bps)){ |
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130 ((af_data_t*)arg)->format = af->data->format; |
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131 ((af_data_t*)arg)->bps = af->data->bps; |
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132 return AF_FALSE; |
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133 } |
8678 | 134 return AF_OK; |
135 } | |
136 case AF_CONTROL_COMMAND_LINE:{ | |
137 float d = 0; | |
138 sscanf((char*)arg,"%f",&d); | |
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139 if ((d < 0) || (d > 1000)){ |
29049 | 140 mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Invalid delay time, valid time values" |
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141 " are 0ms to 1000ms current value is %0.3f ms\n",d); |
8678 | 142 return AF_ERROR; |
143 } | |
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144 s->d = d; |
8678 | 145 return AF_OK; |
146 } | |
147 } | |
148 return AF_UNKNOWN; | |
149 } | |
150 | |
151 // Deallocate memory | |
152 static void uninit(struct af_instance_s* af) | |
153 { | |
22176 | 154 if(af->data) |
8678 | 155 free(af->data->audio); |
22176 | 156 free(af->data); |
157 free(af->setup); | |
8678 | 158 } |
159 | |
160 // The beginnings of an active matrix... | |
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161 static float steering_matrix[][12] = { |
8678 | 162 // LL RL LR RR LS RS |
163 // LLs RLs LRs RRs LC RC | |
164 {.707, .0, .0, .707, .5, -.5, | |
165 .5878, -.3928, .3928, -.5878, .5, .5}, | |
166 }; | |
167 | |
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168 // Experimental moving average dominance |
8678 | 169 //static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0; |
170 | |
171 // Filter data through filter | |
172 static af_data_t* play(struct af_instance_s* af, af_data_t* data){ | |
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173 af_surround_t* s = (af_surround_t*)af->setup; |
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174 float* m = steering_matrix[0]; |
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175 float* in = data->audio; // Input audio data |
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176 float* out = NULL; // Output audio data |
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177 float* end = in + data->len / sizeof(float); // Loop end |
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178 int i = s->i; // Filter queue index |
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179 int ri = s->ri; // Read index for delay queue |
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180 int wi = s->wi; // Write index for delay queue |
8678 | 181 |
182 if (AF_OK != RESIZE_LOCAL_BUFFER(af, data)) | |
183 return NULL; | |
184 | |
185 out = af->data->audio; | |
186 | |
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187 while(in < end){ |
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188 /* Dominance: |
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189 abs(in[0]) abs(in[1]); |
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190 abs(in[0]+in[1]) abs(in[0]-in[1]); |
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191 10 * log( abs(in[0]) / (abs(in[1])|1) ); |
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192 10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */ |
8678 | 193 |
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194 /* About volume balancing... |
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195 Surround encoding does the following: |
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196 Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S |
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197 So S should be extracted as: |
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198 (Lt-Rt) |
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199 But we are splitting the S to two output channels, so we |
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200 must take 3dB off as we split it: |
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201 Ls=Rs=.707*(Lt-Rt) |
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202 Trouble is, Lt could be +1, Rt -1, so possibility that S will |
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203 overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by |
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204 6dB (/2). This keeps the overall balance, but guarantees no |
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205 overflow. */ |
8678 | 206 |
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207 // Output front left and right |
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208 out[0] = m[0]*in[0] + m[1]*in[1]; |
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209 out[1] = m[2]*in[0] + m[3]*in[1]; |
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210 |
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211 // Low-pass output @ 7kHz |
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212 FIR((&s->lq[i]), s->w, s->dl[wi]); |
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213 |
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214 // Delay output by d ms |
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215 out[2] = s->dl[ri]; |
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216 |
8678 | 217 #ifdef SPLITREAR |
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218 // Low-pass output @ 7kHz |
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219 FIR((&s->rq[i]), s->w, s->dr[wi]); |
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220 |
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221 // Delay output by d ms |
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222 out[3] = s->dr[ri]; |
8678 | 223 #else |
224 out[3] = -out[2]; | |
225 #endif | |
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226 |
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227 // Update delay queues indexes |
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228 UPDATEQI(ri); |
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229 UPDATEQI(wi); |
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230 |
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231 // Calculate and save surround in circular queue |
8678 | 232 #ifdef SPLITREAR |
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233 ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]); |
8678 | 234 #else |
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235 ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]); |
8678 | 236 #endif |
237 | |
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238 // Next sample... |
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239 in = &in[data->nch]; |
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240 out = &out[af->data->nch]; |
8678 | 241 } |
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242 |
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243 // Save indexes |
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244 s->i = i; s->ri = ri; s->wi = wi; |
8678 | 245 |
246 // Set output data | |
247 data->audio = af->data->audio; | |
24888 | 248 data->len *= 2; |
8678 | 249 data->nch = af->data->nch; |
250 | |
251 return data; | |
252 } | |
253 | |
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254 static int af_open(af_instance_t* af){ |
8678 | 255 af->control=control; |
256 af->uninit=uninit; | |
257 af->play=play; | |
24888 | 258 af->mul=2; |
8678 | 259 af->data=calloc(1,sizeof(af_data_t)); |
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260 af->setup=calloc(1,sizeof(af_surround_t)); |
8678 | 261 if(af->data == NULL || af->setup == NULL) |
262 return AF_ERROR; | |
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263 ((af_surround_t*)af->setup)->d = 20; |
8678 | 264 return AF_OK; |
265 } | |
266 | |
267 af_info_t af_info_surround = | |
268 { | |
269 "Surround decoder filter", | |
270 "surround", | |
271 "Steve Davies <steve@daviesfam.org>", | |
272 "", | |
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273 AF_FLAGS_NOT_REENTRANT, |
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274 af_open |
8678 | 275 }; |