x11_common: Get window dimensions also for -wid 0
[mplayer/glamo.git] / libaf / af_surround.c
blob098234b364851cd6d81314d6c012566a9fb92969
1 /*
2 * Filter to do simple decoding of matrixed surround sound.
3 * This will provide a (basic) surround-sound effect from
4 * audio encoded for Dolby Surround, Pro Logic etc.
6 * original author: Steve Davies <steve@daviesfam.org>
8 * This file is part of MPlayer.
10 * MPlayer is free software; you can redistribute it and/or modify
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.
15 * MPlayer is distributed in the hope that it will be useful,
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.
20 * You should have received a copy of the GNU General Public License along
21 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
22 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
25 /* The principle: Make rear channels by extracting anti-phase data
26 from the front channels, delay by 20ms and feed to rear in anti-phase
30 /* SPLITREAR: Define to decode two distinct rear channels - this
31 doesn't work so well in practice because separation in a passive
32 matrix is not high. C (dialogue) to Ls and Rs 14dB or so - so
33 dialogue leaks to the rear. Still - give it a try and send
34 feedback. Comment this define for old behavior of a single
35 surround sent to rear in anti-phase */
36 #define SPLITREAR 1
38 #include <stdio.h>
39 #include <stdlib.h>
40 #include <string.h>
42 #include "af.h"
43 #include "dsp.h"
45 #define L 32 // Length of fir filter
46 #define LD 65536 // Length of delay buffer
48 // 32 Tap fir filter loop unrolled
49 #define FIR(x,w,y) \
50 y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
51 + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
52 + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
53 + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \
54 + w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \
55 + w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \
56 + w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \
57 + w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31])
59 // Add to circular queue macro + update index
60 #ifdef SPLITREAR
61 #define ADDQUE(qi,rq,lq,r,l)\
62 lq[qi]=lq[qi+L]=(l);\
63 rq[qi]=rq[qi+L]=(r);\
64 qi=(qi-1)&(L-1);
65 #else
66 #define ADDQUE(qi,lq,l)\
67 lq[qi]=lq[qi+L]=(l);\
68 qi=(qi-1)&(L-1);
69 #endif
71 // Macro for updating queue index in delay queues
72 #define UPDATEQI(qi) qi=(qi+1)&(LD-1)
74 // instance data
75 typedef struct af_surround_s
77 float lq[2*L]; // Circular queue for filtering left rear channel
78 float rq[2*L]; // Circular queue for filtering right rear channel
79 float w[L]; // FIR filter coefficients for surround sound 7kHz low-pass
80 float* dr; // Delay queue right rear channel
81 float* dl; // Delay queue left rear channel
82 float d; // Delay time
83 int i; // Position in circular buffer
84 int wi; // Write index for delay queue
85 int ri; // Read index for delay queue
86 }af_surround_t;
88 // Initialization and runtime control
89 static int control(struct af_instance_s* af, int cmd, void* arg)
91 af_surround_t *s = af->setup;
92 switch(cmd){
93 case AF_CONTROL_REINIT:{
94 float fc;
95 af->data->rate = ((af_data_t*)arg)->rate;
96 af->data->nch = ((af_data_t*)arg)->nch*2;
97 af->data->format = AF_FORMAT_FLOAT_NE;
98 af->data->bps = 4;
100 if (af->data->nch != 4){
101 mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Only stereo input is supported.\n");
102 return AF_DETACH;
104 // Surround filer coefficients
105 fc = 2.0 * 7000.0/(float)af->data->rate;
106 if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
107 mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Unable to design low-pass filter.\n");
108 return AF_ERROR;
111 // Free previous delay queues
112 if(s->dl)
113 free(s->dl);
114 if(s->dr)
115 free(s->dr);
116 // Allocate new delay queues
117 s->dl = calloc(LD,af->data->bps);
118 s->dr = calloc(LD,af->data->bps);
119 if((NULL == s->dl) || (NULL == s->dr))
120 mp_msg(MSGT_AFILTER, MSGL_FATAL, "[delay] Out of memory\n");
122 // Initialize delay queue index
123 if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
124 return AF_ERROR;
125 // printf("%i\n",s->wi);
126 s->ri = 0;
128 if((af->data->format != ((af_data_t*)arg)->format) ||
129 (af->data->bps != ((af_data_t*)arg)->bps)){
130 ((af_data_t*)arg)->format = af->data->format;
131 ((af_data_t*)arg)->bps = af->data->bps;
132 return AF_FALSE;
134 return AF_OK;
136 case AF_CONTROL_COMMAND_LINE:{
137 float d = 0;
138 sscanf((char*)arg,"%f",&d);
139 if ((d < 0) || (d > 1000)){
140 mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Invalid delay time, valid time values"
141 " are 0ms to 1000ms current value is %0.3f ms\n",d);
142 return AF_ERROR;
144 s->d = d;
145 return AF_OK;
148 return AF_UNKNOWN;
151 // Deallocate memory
152 static void uninit(struct af_instance_s* af)
154 if(af->data)
155 free(af->data->audio);
156 free(af->data);
157 free(af->setup);
160 // The beginnings of an active matrix...
161 static float steering_matrix[][12] = {
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},
168 // Experimental moving average dominance
169 //static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
171 // Filter data through filter
172 static af_data_t* play(struct af_instance_s* af, af_data_t* data){
173 af_surround_t* s = (af_surround_t*)af->setup;
174 float* m = steering_matrix[0];
175 float* in = data->audio; // Input audio data
176 float* out = NULL; // Output audio data
177 float* end = in + data->len / sizeof(float); // Loop end
178 int i = s->i; // Filter queue index
179 int ri = s->ri; // Read index for delay queue
180 int wi = s->wi; // Write index for delay queue
182 if (AF_OK != RESIZE_LOCAL_BUFFER(af, data))
183 return NULL;
185 out = af->data->audio;
187 while(in < end){
188 /* Dominance:
189 abs(in[0]) abs(in[1]);
190 abs(in[0]+in[1]) abs(in[0]-in[1]);
191 10 * log( abs(in[0]) / (abs(in[1])|1) );
192 10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */
194 /* About volume balancing...
195 Surround encoding does the following:
196 Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
197 So S should be extracted as:
198 (Lt-Rt)
199 But we are splitting the S to two output channels, so we
200 must take 3dB off as we split it:
201 Ls=Rs=.707*(Lt-Rt)
202 Trouble is, Lt could be +1, Rt -1, so possibility that S will
203 overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by
204 6dB (/2). This keeps the overall balance, but guarantees no
205 overflow. */
207 // Output front left and right
208 out[0] = m[0]*in[0] + m[1]*in[1];
209 out[1] = m[2]*in[0] + m[3]*in[1];
211 // Low-pass output @ 7kHz
212 FIR((&s->lq[i]), s->w, s->dl[wi]);
214 // Delay output by d ms
215 out[2] = s->dl[ri];
217 #ifdef SPLITREAR
218 // Low-pass output @ 7kHz
219 FIR((&s->rq[i]), s->w, s->dr[wi]);
221 // Delay output by d ms
222 out[3] = s->dr[ri];
223 #else
224 out[3] = -out[2];
225 #endif
227 // Update delay queues indexes
228 UPDATEQI(ri);
229 UPDATEQI(wi);
231 // Calculate and save surround in circular queue
232 #ifdef SPLITREAR
233 ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
234 #else
235 ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
236 #endif
238 // Next sample...
239 in = &in[data->nch];
240 out = &out[af->data->nch];
243 // Save indexes
244 s->i = i; s->ri = ri; s->wi = wi;
246 // Set output data
247 data->audio = af->data->audio;
248 data->len *= 2;
249 data->nch = af->data->nch;
251 return data;
254 static int af_open(af_instance_t* af){
255 af->control=control;
256 af->uninit=uninit;
257 af->play=play;
258 af->mul=2;
259 af->data=calloc(1,sizeof(af_data_t));
260 af->setup=calloc(1,sizeof(af_surround_t));
261 if(af->data == NULL || af->setup == NULL)
262 return AF_ERROR;
263 ((af_surround_t*)af->setup)->d = 20;
264 return AF_OK;
267 af_info_t af_info_surround =
269 "Surround decoder filter",
270 "surround",
271 "Steve Davies <steve@daviesfam.org>",
273 AF_FLAGS_NOT_REENTRANT,
274 af_open