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[mplayer/greg.git] / libaf / af_resample.c
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1 /*
2 * This audio filter changes the sample rate.
4 * Copyright (C) 2002 Anders Johansson ajh@atri.curtin.edu.au
6 * This file is part of MPlayer.
8 * MPlayer is free software; you can redistribute it and/or modify
9 * it under the terms of the GNU General Public License as published by
10 * the Free Software Foundation; either version 2 of the License, or
11 * (at your option) any later version.
13 * MPlayer is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
16 * GNU General Public License for more details.
18 * You should have received a copy of the GNU General Public License along
19 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
23 #include <stdio.h>
24 #include <stdlib.h>
25 #include <inttypes.h>
27 #include "libavutil/common.h"
28 #include "libavutil/mathematics.h"
29 #include "af.h"
30 #include "dsp.h"
32 /* Below definition selects the length of each poly phase component.
33 Valid definitions are L8 and L16, where the number denotes the
34 length of the filter. This definition affects the computational
35 complexity (see play()), the performance (see filter.h) and the
36 memory usage. The filterlength is choosen to 8 if the machine is
37 slow and to 16 if the machine is fast and has MMX.
40 #if !HAVE_MMX // This machine is slow
41 #define L8
42 #else
43 #define L16
44 #endif
46 #include "af_resample_template.c"
48 // Filtering types
49 #define RSMP_LIN (0<<0) // Linear interpolation
50 #define RSMP_INT (1<<0) // 16 bit integer
51 #define RSMP_FLOAT (2<<0) // 32 bit floating point
52 #define RSMP_MASK (3<<0)
54 // Defines for sloppy or exact resampling
55 #define FREQ_SLOPPY (0<<2)
56 #define FREQ_EXACT (1<<2)
57 #define FREQ_MASK (1<<2)
59 // Accuracy for linear interpolation
60 #define STEPACCURACY 32
62 // local data
63 typedef struct af_resample_s
65 void* w; // Current filter weights
66 void** xq; // Circular buffers
67 uint32_t xi; // Index for circular buffers
68 uint32_t wi; // Index for w
69 uint32_t i; // Number of new samples to put in x queue
70 uint32_t dn; // Down sampling factor
71 uint32_t up; // Up sampling factor
72 uint64_t step; // Step size for linear interpolation
73 uint64_t pt; // Pointer remainder for linear interpolation
74 int setup; // Setup parameters cmdline or through postcreate
75 } af_resample_t;
77 // Fast linear interpolation resample with modest audio quality
78 static int linint(af_data_t* c,af_data_t* l, af_resample_t* s)
80 uint32_t len = 0; // Number of input samples
81 uint32_t nch = l->nch; // Words pre transfer
82 uint64_t step = s->step;
83 int16_t* in16 = ((int16_t*)c->audio);
84 int16_t* out16 = ((int16_t*)l->audio);
85 int32_t* in32 = ((int32_t*)c->audio);
86 int32_t* out32 = ((int32_t*)l->audio);
87 uint64_t end = ((((uint64_t)c->len)/2LL)<<STEPACCURACY);
88 uint64_t pt = s->pt;
89 uint16_t tmp;
91 switch (nch){
92 case 1:
93 while(pt < end){
94 out16[len++]=in16[pt>>STEPACCURACY];
95 pt+=step;
97 s->pt=pt & ((1LL<<STEPACCURACY)-1);
98 break;
99 case 2:
100 end/=2;
101 while(pt < end){
102 out32[len++]=in32[pt>>STEPACCURACY];
103 pt+=step;
105 len=(len<<1);
106 s->pt=pt & ((1LL<<STEPACCURACY)-1);
107 break;
108 default:
109 end /=nch;
110 while(pt < end){
111 tmp=nch;
112 do {
113 tmp--;
114 out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch];
115 } while (tmp);
116 len+=nch;
117 pt+=step;
119 s->pt=pt & ((1LL<<STEPACCURACY)-1);
121 return len;
124 /* Determine resampling type and format */
125 static int set_types(struct af_instance_s* af, af_data_t* data)
127 af_resample_t* s = af->setup;
128 int rv = AF_OK;
129 float rd = 0;
131 // Make sure this filter isn't redundant
132 if((af->data->rate == data->rate) || (af->data->rate == 0))
133 return AF_DETACH;
134 /* If sloppy and small resampling difference (2%) */
135 rd = abs((float)af->data->rate - (float)data->rate)/(float)data->rate;
136 if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) &&
137 (data->format != (AF_FORMAT_FLOAT_NE))) ||
138 ((s->setup & RSMP_MASK) == RSMP_LIN)){
139 s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN;
140 af->data->format = AF_FORMAT_S16_NE;
141 af->data->bps = 2;
142 af_msg(AF_MSG_VERBOSE,"[resample] Using linear interpolation. \n");
144 else{
145 /* If the input format is float or if float is explicitly selected
146 use float, otherwise use int */
147 if((data->format == (AF_FORMAT_FLOAT_NE)) ||
148 ((s->setup & RSMP_MASK) == RSMP_FLOAT)){
149 s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT;
150 af->data->format = AF_FORMAT_FLOAT_NE;
151 af->data->bps = 4;
153 else{
154 s->setup = (s->setup & ~RSMP_MASK) | RSMP_INT;
155 af->data->format = AF_FORMAT_S16_NE;
156 af->data->bps = 2;
158 af_msg(AF_MSG_VERBOSE,"[resample] Using %s processing and %s frequecy"
159 " conversion.\n",
160 ((s->setup & RSMP_MASK) == RSMP_FLOAT)?"floating point":"integer",
161 ((s->setup & FREQ_MASK) == FREQ_SLOPPY)?"inexact":"exact");
164 if(af->data->format != data->format || af->data->bps != data->bps)
165 rv = AF_FALSE;
166 data->format = af->data->format;
167 data->bps = af->data->bps;
168 af->data->nch = data->nch;
169 return rv;
172 // Initialization and runtime control
173 static int control(struct af_instance_s* af, int cmd, void* arg)
175 switch(cmd){
176 case AF_CONTROL_REINIT:{
177 af_resample_t* s = (af_resample_t*)af->setup;
178 af_data_t* n = (af_data_t*)arg; // New configureation
179 int i,d = 0;
180 int rv = AF_OK;
182 // Free space for circular bufers
183 if(s->xq){
184 for(i=1;i<af->data->nch;i++)
185 if(s->xq[i])
186 free(s->xq[i]);
187 free(s->xq);
188 s->xq = NULL;
191 if(AF_DETACH == (rv = set_types(af,n)))
192 return AF_DETACH;
194 // If linear interpolation
195 if((s->setup & RSMP_MASK) == RSMP_LIN){
196 s->pt=0LL;
197 s->step=((uint64_t)n->rate<<STEPACCURACY)/(uint64_t)af->data->rate+1LL;
198 af_msg(AF_MSG_DEBUG0,"[resample] Linear interpolation step: 0x%016"PRIX64".\n",
199 s->step);
200 af->mul = (double)af->data->rate / n->rate;
201 return rv;
204 // Calculate up and down sampling factors
205 d=av_gcd(af->data->rate,n->rate);
207 // If sloppy resampling is enabled limit the upsampling factor
208 if(((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (af->data->rate/d > 5000)){
209 int up=af->data->rate/2;
210 int dn=n->rate/2;
211 int m=2;
212 while(af->data->rate/(d*m) > 5000){
213 d=av_gcd(up,dn);
214 up/=2; dn/=2; m*=2;
216 d*=m;
219 // Create space for circular bufers
220 s->xq = malloc(n->nch*sizeof(void*));
221 for(i=0;i<n->nch;i++)
222 s->xq[i] = malloc(2*L*af->data->bps);
223 s->xi = 0;
225 // Check if the the design needs to be redone
226 if(s->up != af->data->rate/d || s->dn != n->rate/d){
227 float* w;
228 float* wt;
229 float fc;
230 int j;
231 s->up = af->data->rate/d;
232 s->dn = n->rate/d;
233 s->wi = 0;
234 s->i = 0;
236 // Calculate cuttof frequency for filter
237 fc = 1/(float)(max(s->up,s->dn));
238 // Allocate space for polyphase filter bank and protptype filter
239 w = malloc(sizeof(float) * s->up *L);
240 if(NULL != s->w)
241 free(s->w);
242 s->w = malloc(L*s->up*af->data->bps);
244 // Design prototype filter type using Kaiser window with beta = 10
245 if(NULL == w || NULL == s->w ||
246 -1 == af_filter_design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
247 af_msg(AF_MSG_ERROR,"[resample] Unable to design prototype filter.\n");
248 return AF_ERROR;
250 // Copy data from prototype to polyphase filter
251 wt=w;
252 for(j=0;j<L;j++){//Columns
253 for(i=0;i<s->up;i++){//Rows
254 if((s->setup & RSMP_MASK) == RSMP_INT){
255 float t=(float)s->up*32767.0*(*wt);
256 ((int16_t*)s->w)[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
258 else
259 ((float*)s->w)[i*L+j] = (float)s->up*(*wt);
260 wt++;
263 free(w);
264 af_msg(AF_MSG_VERBOSE,"[resample] New filter designed up: %i "
265 "down: %i\n", s->up, s->dn);
268 // Set multiplier and delay
269 af->delay = 0; // not set correctly, but shouldn't be too large anyway
270 af->mul = (double)s->up / s->dn;
271 return rv;
273 case AF_CONTROL_COMMAND_LINE:{
274 af_resample_t* s = (af_resample_t*)af->setup;
275 int rate=0;
276 int type=RSMP_INT;
277 int sloppy=1;
278 sscanf((char*)arg,"%i:%i:%i", &rate, &sloppy, &type);
279 s->setup = (sloppy?FREQ_SLOPPY:FREQ_EXACT) |
280 (clamp(type,RSMP_LIN,RSMP_FLOAT));
281 return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate);
283 case AF_CONTROL_POST_CREATE:
284 if((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT)
285 ((af_resample_t*)af->setup)->setup = RSMP_FLOAT;
286 return AF_OK;
287 case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
288 // Reinit must be called after this function has been called
290 // Sanity check
291 if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
292 af_msg(AF_MSG_ERROR,"[resample] The output sample frequency "
293 "must be between 8kHz and 192kHz. Current value is %i \n",
294 ((int*)arg)[0]);
295 return AF_ERROR;
298 af->data->rate=((int*)arg)[0];
299 af_msg(AF_MSG_VERBOSE,"[resample] Changing sample rate "
300 "to %iHz\n",af->data->rate);
301 return AF_OK;
303 return AF_UNKNOWN;
306 // Deallocate memory
307 static void uninit(struct af_instance_s* af)
309 if(af->data)
310 free(af->data->audio);
311 free(af->data);
314 // Filter data through filter
315 static af_data_t* play(struct af_instance_s* af, af_data_t* data)
317 int len = 0; // Length of output data
318 af_data_t* c = data; // Current working data
319 af_data_t* l = af->data; // Local data
320 af_resample_t* s = (af_resample_t*)af->setup;
322 if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
323 return NULL;
325 // Run resampling
326 switch(s->setup & RSMP_MASK){
327 case(RSMP_INT):
328 # define FORMAT_I 1
329 if(s->up>s->dn){
330 # define UP
331 # include "af_resample_template.c"
332 # undef UP
334 else{
335 # define DN
336 # include "af_resample_template.c"
337 # undef DN
339 break;
340 case(RSMP_FLOAT):
341 # undef FORMAT_I
342 # define FORMAT_F 1
343 if(s->up>s->dn){
344 # define UP
345 # include "af_resample_template.c"
346 # undef UP
348 else{
349 # define DN
350 # include "af_resample_template.c"
351 # undef DN
353 break;
354 case(RSMP_LIN):
355 len = linint(c, l, s);
356 break;
359 // Set output data
360 c->audio = l->audio;
361 c->len = len*l->bps;
362 c->rate = l->rate;
364 return c;
367 // Allocate memory and set function pointers
368 static int af_open(af_instance_t* af){
369 af->control=control;
370 af->uninit=uninit;
371 af->play=play;
372 af->mul=1;
373 af->data=calloc(1,sizeof(af_data_t));
374 af->setup=calloc(1,sizeof(af_resample_t));
375 if(af->data == NULL || af->setup == NULL)
376 return AF_ERROR;
377 ((af_resample_t*)af->setup)->setup = RSMP_INT | FREQ_SLOPPY;
378 return AF_OK;
381 // Description of this plugin
382 af_info_t af_info_resample = {
383 "Sample frequency conversion",
384 "resample",
385 "Anders",
387 AF_FLAGS_REENTRANT,
388 af_open