add blox binary codec to codecs.conf
[mplayer/glamo.git] / libaf / af_volnorm.c
blob9bfaef62204f694b512c6848db431fe4a412afde
1 /*
2 * Copyright (C) 2004 Alex Beregszaszi & Pierre Lombard
4 * This file is part of MPlayer.
6 * MPlayer is free software; you can redistribute it and/or modify
7 * it under the terms of the GNU General Public License as published by
8 * the Free Software Foundation; either version 2 of the License, or
9 * (at your option) any later version.
11 * MPlayer is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14 * GNU General Public License for more details.
16 * You should have received a copy of the GNU General Public License along
17 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
18 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
21 #include <stdio.h>
22 #include <stdlib.h>
23 #include <string.h>
25 #include <inttypes.h>
26 #include <math.h>
27 #include <limits.h>
29 #include "af.h"
31 // Methods:
32 // 1: uses a 1 value memory and coefficients new=a*old+b*cur (with a+b=1)
33 // 2: uses several samples to smooth the variations (standard weighted mean
34 // on past samples)
36 // Size of the memory array
37 // FIXME: should depend on the frequency of the data (should be a few seconds)
38 #define NSAMPLES 128
40 // If summing all the mem[].len is lower than MIN_SAMPLE_SIZE bytes, then we
41 // choose to ignore the computed value as it's not significant enough
42 // FIXME: should depend on the frequency of the data (0.5s maybe)
43 #define MIN_SAMPLE_SIZE 32000
45 // mul is the value by which the samples are scaled
46 // and has to be in [MUL_MIN, MUL_MAX]
47 #define MUL_INIT 1.0
48 #define MUL_MIN 0.1
49 #define MUL_MAX 5.0
51 // Silence level
52 // FIXME: should be relative to the level of the samples
53 #define SIL_S16 (SHRT_MAX * 0.01)
54 #define SIL_FLOAT (INT_MAX * 0.01) // FIXME
56 // smooth must be in ]0.0, 1.0[
57 #define SMOOTH_MUL 0.06
58 #define SMOOTH_LASTAVG 0.06
60 #define DEFAULT_TARGET 0.25
62 // Data for specific instances of this filter
63 typedef struct af_volume_s
65 int method; // method used
66 float mul;
67 // method 1
68 float lastavg; // history value of the filter
69 // method 2
70 int idx;
71 struct {
72 float avg; // average level of the sample
73 int len; // sample size (weight)
74 } mem[NSAMPLES];
75 // "Ideal" level
76 float mid_s16;
77 float mid_float;
78 }af_volnorm_t;
80 // Initialization and runtime control
81 static int control(struct af_instance_s* af, int cmd, void* arg)
83 af_volnorm_t* s = (af_volnorm_t*)af->setup;
85 switch(cmd){
86 case AF_CONTROL_REINIT:
87 // Sanity check
88 if(!arg) return AF_ERROR;
90 af->data->rate = ((af_data_t*)arg)->rate;
91 af->data->nch = ((af_data_t*)arg)->nch;
93 if(((af_data_t*)arg)->format == (AF_FORMAT_S16_NE)){
94 af->data->format = AF_FORMAT_S16_NE;
95 af->data->bps = 2;
96 }else{
97 af->data->format = AF_FORMAT_FLOAT_NE;
98 af->data->bps = 4;
100 return af_test_output(af,(af_data_t*)arg);
101 case AF_CONTROL_COMMAND_LINE:{
102 int i = 0;
103 float target = DEFAULT_TARGET;
104 sscanf((char*)arg,"%d:%f", &i, &target);
105 if (i != 1 && i != 2)
106 return AF_ERROR;
107 s->method = i-1;
108 s->mid_s16 = ((float)SHRT_MAX) * target;
109 s->mid_float = ((float)INT_MAX) * target;
110 return AF_OK;
113 return AF_UNKNOWN;
116 // Deallocate memory
117 static void uninit(struct af_instance_s* af)
119 if(af->data)
120 free(af->data);
121 if(af->setup)
122 free(af->setup);
125 static void method1_int16(af_volnorm_t *s, af_data_t *c)
127 register int i = 0;
128 int16_t *data = (int16_t*)c->audio; // Audio data
129 int len = c->len/2; // Number of samples
130 float curavg = 0.0, newavg, neededmul;
131 int tmp;
133 for (i = 0; i < len; i++)
135 tmp = data[i];
136 curavg += tmp * tmp;
138 curavg = sqrt(curavg / (float) len);
140 // Evaluate an adequate 'mul' coefficient based on previous state, current
141 // samples level, etc
143 if (curavg > SIL_S16)
145 neededmul = s->mid_s16 / (curavg * s->mul);
146 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
148 // clamp the mul coefficient
149 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
152 // Scale & clamp the samples
153 for (i = 0; i < len; i++)
155 tmp = s->mul * data[i];
156 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
157 data[i] = tmp;
160 // Evaulation of newavg (not 100% accurate because of values clamping)
161 newavg = s->mul * curavg;
163 // Stores computed values for future smoothing
164 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
167 static void method1_float(af_volnorm_t *s, af_data_t *c)
169 register int i = 0;
170 float *data = (float*)c->audio; // Audio data
171 int len = c->len/4; // Number of samples
172 float curavg = 0.0, newavg, neededmul, tmp;
174 for (i = 0; i < len; i++)
176 tmp = data[i];
177 curavg += tmp * tmp;
179 curavg = sqrt(curavg / (float) len);
181 // Evaluate an adequate 'mul' coefficient based on previous state, current
182 // samples level, etc
184 if (curavg > SIL_FLOAT) // FIXME
186 neededmul = s->mid_float / (curavg * s->mul);
187 s->mul = (1.0 - SMOOTH_MUL) * s->mul + SMOOTH_MUL * neededmul;
189 // clamp the mul coefficient
190 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
193 // Scale & clamp the samples
194 for (i = 0; i < len; i++)
195 data[i] *= s->mul;
197 // Evaulation of newavg (not 100% accurate because of values clamping)
198 newavg = s->mul * curavg;
200 // Stores computed values for future smoothing
201 s->lastavg = (1.0 - SMOOTH_LASTAVG) * s->lastavg + SMOOTH_LASTAVG * newavg;
204 static void method2_int16(af_volnorm_t *s, af_data_t *c)
206 register int i = 0;
207 int16_t *data = (int16_t*)c->audio; // Audio data
208 int len = c->len/2; // Number of samples
209 float curavg = 0.0, newavg, avg = 0.0;
210 int tmp, totallen = 0;
212 for (i = 0; i < len; i++)
214 tmp = data[i];
215 curavg += tmp * tmp;
217 curavg = sqrt(curavg / (float) len);
219 // Evaluate an adequate 'mul' coefficient based on previous state, current
220 // samples level, etc
221 for (i = 0; i < NSAMPLES; i++)
223 avg += s->mem[i].avg * (float)s->mem[i].len;
224 totallen += s->mem[i].len;
227 if (totallen > MIN_SAMPLE_SIZE)
229 avg /= (float)totallen;
230 if (avg >= SIL_S16)
232 s->mul = s->mid_s16 / avg;
233 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
237 // Scale & clamp the samples
238 for (i = 0; i < len; i++)
240 tmp = s->mul * data[i];
241 tmp = clamp(tmp, SHRT_MIN, SHRT_MAX);
242 data[i] = tmp;
245 // Evaulation of newavg (not 100% accurate because of values clamping)
246 newavg = s->mul * curavg;
248 // Stores computed values for future smoothing
249 s->mem[s->idx].len = len;
250 s->mem[s->idx].avg = newavg;
251 s->idx = (s->idx + 1) % NSAMPLES;
254 static void method2_float(af_volnorm_t *s, af_data_t *c)
256 register int i = 0;
257 float *data = (float*)c->audio; // Audio data
258 int len = c->len/4; // Number of samples
259 float curavg = 0.0, newavg, avg = 0.0, tmp;
260 int totallen = 0;
262 for (i = 0; i < len; i++)
264 tmp = data[i];
265 curavg += tmp * tmp;
267 curavg = sqrt(curavg / (float) len);
269 // Evaluate an adequate 'mul' coefficient based on previous state, current
270 // samples level, etc
271 for (i = 0; i < NSAMPLES; i++)
273 avg += s->mem[i].avg * (float)s->mem[i].len;
274 totallen += s->mem[i].len;
277 if (totallen > MIN_SAMPLE_SIZE)
279 avg /= (float)totallen;
280 if (avg >= SIL_FLOAT)
282 s->mul = s->mid_float / avg;
283 s->mul = clamp(s->mul, MUL_MIN, MUL_MAX);
287 // Scale & clamp the samples
288 for (i = 0; i < len; i++)
289 data[i] *= s->mul;
291 // Evaulation of newavg (not 100% accurate because of values clamping)
292 newavg = s->mul * curavg;
294 // Stores computed values for future smoothing
295 s->mem[s->idx].len = len;
296 s->mem[s->idx].avg = newavg;
297 s->idx = (s->idx + 1) % NSAMPLES;
300 // Filter data through filter
301 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
303 af_volnorm_t *s = af->setup;
305 if(af->data->format == (AF_FORMAT_S16_NE))
307 if (s->method)
308 method2_int16(s, data);
309 else
310 method1_int16(s, data);
312 else if(af->data->format == (AF_FORMAT_FLOAT_NE))
314 if (s->method)
315 method2_float(s, data);
316 else
317 method1_float(s, data);
319 return data;
322 // Allocate memory and set function pointers
323 static int af_open(af_instance_t* af){
324 int i = 0;
325 af->control=control;
326 af->uninit=uninit;
327 af->play=play;
328 af->mul=1;
329 af->data=calloc(1,sizeof(af_data_t));
330 af->setup=calloc(1,sizeof(af_volnorm_t));
331 if(af->data == NULL || af->setup == NULL)
332 return AF_ERROR;
334 ((af_volnorm_t*)af->setup)->mul = MUL_INIT;
335 ((af_volnorm_t*)af->setup)->lastavg = ((float)SHRT_MAX) * DEFAULT_TARGET;
336 ((af_volnorm_t*)af->setup)->idx = 0;
337 ((af_volnorm_t*)af->setup)->mid_s16 = ((float)SHRT_MAX) * DEFAULT_TARGET;
338 ((af_volnorm_t*)af->setup)->mid_float = ((float)INT_MAX) * DEFAULT_TARGET;
339 for (i = 0; i < NSAMPLES; i++)
341 ((af_volnorm_t*)af->setup)->mem[i].len = 0;
342 ((af_volnorm_t*)af->setup)->mem[i].avg = 0;
344 return AF_OK;
347 // Description of this filter
348 af_info_t af_info_volnorm = {
349 "Volume normalizer filter",
350 "volnorm",
351 "Alex Beregszaszi & Pierre Lombard",
353 AF_FLAGS_NOT_REENTRANT,
354 af_open