Mercurial > mplayer.hg
annotate libaf/af_surround.c @ 33263:5f527a9a9521
Add an exit function.
This function will allow performing clean-up operations.
(MPlayer calls guiDone() before exiting, but only if the GUI has been
initialized, i.e. if guiInit() has been called successfully. Any
exit_player()/exit_player_with_rc() after GUI's cfg_read() until
guiInit(), or any exit_player() during guiInit() itself will end the GUI
without calling guiDone(). This exit function will at least handle
abortions during guiInit() itself. It will be called twice in case of an
guiExit() after GUI initialization - first directly, next by guiDone()
via MPlayer's exit_player_with_rc().)
author | ib |
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date | Tue, 03 May 2011 12:19:22 +0000 |
parents | 8fa2f43cb760 |
children | a93891202051 |
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 free(s->dl); |
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113 free(s->dr); |
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114 // Allocate new delay queues |
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115 s->dl = calloc(LD,af->data->bps); |
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116 s->dr = calloc(LD,af->data->bps); |
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117 if((NULL == s->dl) || (NULL == s->dr)) |
29049 | 118 mp_msg(MSGT_AFILTER, MSGL_FATAL, "[delay] Out of memory\n"); |
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119 |
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120 // Initialize delay queue index |
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121 if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0)) |
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122 return AF_ERROR; |
12569 | 123 // printf("%i\n",s->wi); |
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124 s->ri = 0; |
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125 |
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126 if((af->data->format != ((af_data_t*)arg)->format) || |
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127 (af->data->bps != ((af_data_t*)arg)->bps)){ |
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128 ((af_data_t*)arg)->format = af->data->format; |
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129 ((af_data_t*)arg)->bps = af->data->bps; |
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130 return AF_FALSE; |
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131 } |
8678 | 132 return AF_OK; |
133 } | |
134 case AF_CONTROL_COMMAND_LINE:{ | |
135 float d = 0; | |
136 sscanf((char*)arg,"%f",&d); | |
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137 if ((d < 0) || (d > 1000)){ |
29049 | 138 mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Invalid delay time, valid time values" |
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139 " are 0ms to 1000ms current value is %0.3f ms\n",d); |
8678 | 140 return AF_ERROR; |
141 } | |
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142 s->d = d; |
8678 | 143 return AF_OK; |
144 } | |
145 } | |
146 return AF_UNKNOWN; | |
147 } | |
148 | |
149 // Deallocate memory | |
150 static void uninit(struct af_instance_s* af) | |
151 { | |
22176 | 152 if(af->data) |
8678 | 153 free(af->data->audio); |
22176 | 154 free(af->data); |
155 free(af->setup); | |
8678 | 156 } |
157 | |
158 // The beginnings of an active matrix... | |
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159 static float steering_matrix[][12] = { |
8678 | 160 // LL RL LR RR LS RS |
161 // LLs RLs LRs RRs LC RC | |
162 {.707, .0, .0, .707, .5, -.5, | |
163 .5878, -.3928, .3928, -.5878, .5, .5}, | |
164 }; | |
165 | |
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166 // Experimental moving average dominance |
8678 | 167 //static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0; |
168 | |
169 // Filter data through filter | |
170 static af_data_t* play(struct af_instance_s* af, af_data_t* data){ | |
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171 af_surround_t* s = (af_surround_t*)af->setup; |
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172 float* m = steering_matrix[0]; |
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173 float* in = data->audio; // Input audio data |
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174 float* out = NULL; // Output audio data |
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175 float* end = in + data->len / sizeof(float); // Loop end |
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176 int i = s->i; // Filter queue index |
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177 int ri = s->ri; // Read index for delay queue |
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178 int wi = s->wi; // Write index for delay queue |
8678 | 179 |
180 if (AF_OK != RESIZE_LOCAL_BUFFER(af, data)) | |
181 return NULL; | |
182 | |
183 out = af->data->audio; | |
184 | |
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185 while(in < end){ |
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186 /* Dominance: |
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187 abs(in[0]) abs(in[1]); |
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188 abs(in[0]+in[1]) abs(in[0]-in[1]); |
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189 10 * log( abs(in[0]) / (abs(in[1])|1) ); |
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190 10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */ |
8678 | 191 |
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192 /* About volume balancing... |
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193 Surround encoding does the following: |
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194 Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S |
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195 So S should be extracted as: |
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196 (Lt-Rt) |
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197 But we are splitting the S to two output channels, so we |
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198 must take 3dB off as we split it: |
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199 Ls=Rs=.707*(Lt-Rt) |
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200 Trouble is, Lt could be +1, Rt -1, so possibility that S will |
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201 overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by |
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202 6dB (/2). This keeps the overall balance, but guarantees no |
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203 overflow. */ |
8678 | 204 |
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205 // Output front left and right |
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206 out[0] = m[0]*in[0] + m[1]*in[1]; |
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207 out[1] = m[2]*in[0] + m[3]*in[1]; |
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208 |
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209 // Low-pass output @ 7kHz |
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210 FIR((&s->lq[i]), s->w, s->dl[wi]); |
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211 |
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212 // Delay output by d ms |
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213 out[2] = s->dl[ri]; |
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214 |
8678 | 215 #ifdef SPLITREAR |
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216 // Low-pass output @ 7kHz |
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217 FIR((&s->rq[i]), s->w, s->dr[wi]); |
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218 |
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219 // Delay output by d ms |
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220 out[3] = s->dr[ri]; |
8678 | 221 #else |
222 out[3] = -out[2]; | |
223 #endif | |
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224 |
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225 // Update delay queues indexes |
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226 UPDATEQI(ri); |
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227 UPDATEQI(wi); |
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228 |
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229 // Calculate and save surround in circular queue |
8678 | 230 #ifdef SPLITREAR |
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231 ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]); |
8678 | 232 #else |
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233 ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]); |
8678 | 234 #endif |
235 | |
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236 // Next sample... |
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237 in = &in[data->nch]; |
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238 out = &out[af->data->nch]; |
8678 | 239 } |
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240 |
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241 // Save indexes |
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242 s->i = i; s->ri = ri; s->wi = wi; |
8678 | 243 |
244 // Set output data | |
245 data->audio = af->data->audio; | |
24888 | 246 data->len *= 2; |
8678 | 247 data->nch = af->data->nch; |
248 | |
249 return data; | |
250 } | |
251 | |
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252 static int af_open(af_instance_t* af){ |
8678 | 253 af->control=control; |
254 af->uninit=uninit; | |
255 af->play=play; | |
24888 | 256 af->mul=2; |
8678 | 257 af->data=calloc(1,sizeof(af_data_t)); |
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258 af->setup=calloc(1,sizeof(af_surround_t)); |
8678 | 259 if(af->data == NULL || af->setup == NULL) |
260 return AF_ERROR; | |
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261 ((af_surround_t*)af->setup)->d = 20; |
8678 | 262 return AF_OK; |
263 } | |
264 | |
265 af_info_t af_info_surround = | |
266 { | |
267 "Surround decoder filter", | |
268 "surround", | |
269 "Steve Davies <steve@daviesfam.org>", | |
270 "", | |
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271 AF_FLAGS_NOT_REENTRANT, |
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272 af_open |
8678 | 273 }; |