libavcodec/resample.c

   1 /*
   2  * Sample rate convertion for both audio and video
   3  * Copyright (c) 2000 Fabrice Bellard.
   4  *
   5  * This library is free software; you can redistribute it and/or
   6  * modify it under the terms of the GNU Lesser General Public
   7  * License as published by the Free Software Foundation; either
   8  * version 2 of the License, or (at your option) any later version.
   9  *
  10  * This library is distributed in the hope that it will be useful,
  11  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  12  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  13  * Lesser General Public License for more details.
  14  *
  15  * You should have received a copy of the GNU Lesser General Public
  16  * License along with this library; if not, write to the Free Software
  17  * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  18  */
  19
  20 /**
  21  * @file resample.c
  22  * Sample rate convertion for both audio and video.
  23  */
  24
  25 #include "avcodec.h"
  26
  27 #if defined (CONFIG_OS2)
  28 #define floorf(n) floor(n)
  29 #endif
  30
  31 typedef struct {
  32     /* fractional resampling */
  33     uint32_t incr; /* fractional increment */
  34     uint32_t frac;
  35     int last_sample;
  36     /* integer down sample */
  37     int iratio;  /* integer divison ratio */
  38     int icount, isum;
  39     int inv;
  40 } ReSampleChannelContext;
  41
  42 struct ReSampleContext {
  43     ReSampleChannelContext channel_ctx[2];
  44     float ratio;
  45     /* channel convert */
  46     int input_channels, output_channels, filter_channels;
  47 };
  48
  49
  50 #define FRAC_BITS 16
  51 #define FRAC (1 << FRAC_BITS)
  52
  53 static void init_mono_resample(ReSampleChannelContext *s, float ratio)
  54 {
  55     ratio = 1.0 / ratio;
  56     s->iratio = (int)floorf(ratio);
  57     if (s->iratio == 0)
  58         s->iratio = 1;
  59     s->incr = (int)((ratio / s->iratio) * FRAC);
  60     s->frac = FRAC;
  61     s->last_sample = 0;
  62     s->icount = s->iratio;
  63     s->isum = 0;
  64     s->inv = (FRAC / s->iratio);
  65 }
  66
  67 /* fractional audio resampling */
  68 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
  69 {
  70     unsigned int frac, incr;
  71     int l0, l1;
  72     short *q, *p, *pend;
  73
  74     l0 = s->last_sample;
  75     incr = s->incr;
  76     frac = s->frac;
  77
  78     p = input;
  79     pend = input + nb_samples;
  80     q = output;
  81
  82     l1 = *p++;
  83     for(;;) {
  84         /* interpolate */
  85         *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
  86         frac = frac + s->incr;
  87         while (frac >= FRAC) {
  88             frac -= FRAC;
  89             if (p >= pend)
  90                 goto the_end;
  91             l0 = l1;
  92             l1 = *p++;
  93         }
  94     }
  95  the_end:
  96     s->last_sample = l1;
  97     s->frac = frac;
  98     return q - output;
  99 }
 100
 101 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
 102 {
 103     short *q, *p, *pend;
 104     int c, sum;
 105
 106     p = input;
 107     pend = input + nb_samples;
 108     q = output;
 109
 110     c = s->icount;
 111     sum = s->isum;
 112
 113     for(;;) {
 114         sum += *p++;
 115         if (--c == 0) {
 116             *q++ = (sum * s->inv) >> FRAC_BITS;
 117             c = s->iratio;
 118             sum = 0;
 119         }
 120         if (p >= pend)
 121             break;
 122     }
 123     s->isum = sum;
 124     s->icount = c;
 125     return q - output;
 126 }
 127
 128 /* n1: number of samples */
 129 static void stereo_to_mono(short *output, short *input, int n1)
 130 {
 131     short *p, *q;
 132     int n = n1;
 133
 134     p = input;
 135     q = output;
 136     while (n >= 4) {
 137         q[0] = (p[0] + p[1]) >> 1;
 138         q[1] = (p[2] + p[3]) >> 1;
 139         q[2] = (p[4] + p[5]) >> 1;
 140         q[3] = (p[6] + p[7]) >> 1;
 141         q += 4;
 142         p += 8;
 143         n -= 4;
 144     }
 145     while (n > 0) {
 146         q[0] = (p[0] + p[1]) >> 1;
 147         q++;
 148         p += 2;
 149         n--;
 150     }
 151 }
 152
 153 /* n1: number of samples */
 154 static void mono_to_stereo(short *output, short *input, int n1)
 155 {
 156     short *p, *q;
 157     int n = n1;
 158     int v;
 159
 160     p = input;
 161     q = output;
 162     while (n >= 4) {
 163         v = p[0]; q[0] = v; q[1] = v;
 164         v = p[1]; q[2] = v; q[3] = v;
 165         v = p[2]; q[4] = v; q[5] = v;
 166         v = p[3]; q[6] = v; q[7] = v;
 167         q += 8;
 168         p += 4;
 169         n -= 4;
 170     }
 171     while (n > 0) {
 172         v = p[0]; q[0] = v; q[1] = v;
 173         q += 2;
 174         p += 1;
 175         n--;
 176     }
 177 }
 178
 179 /* XXX: should use more abstract 'N' channels system */
 180 static void stereo_split(short *output1, short *output2, short *input, int n)
 181 {
 182     int i;
 183
 184     for(i=0;i<n;i++) {
 185         *output1++ = *input++;
 186         *output2++ = *input++;
 187     }
 188 }
 189
 190 static void stereo_mux(short *output, short *input1, short *input2, int n)
 191 {
 192     int i;
 193
 194     for(i=0;i<n;i++) {
 195         *output++ = *input1++;
 196         *output++ = *input2++;
 197     }
 198 }
 199
 200 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
 201 {
 202     short *buf1;
 203     short *buftmp;
 204
 205     buf1= (short*)av_malloc( nb_samples * sizeof(short) );
 206
 207     /* first downsample by an integer factor with averaging filter */
 208     if (s->iratio > 1) {
 209         buftmp = buf1;
 210         nb_samples = integer_downsample(s, buftmp, input, nb_samples);
 211     } else {
 212         buftmp = input;
 213     }
 214
 215     /* then do a fractional resampling with linear interpolation */
 216     if (s->incr != FRAC) {
 217         nb_samples = fractional_resample(s, output, buftmp, nb_samples);
 218     } else {
 219         memcpy(output, buftmp, nb_samples * sizeof(short));
 220     }
 221     av_free(buf1);
 222     return nb_samples;
 223 }
 224
 225 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
 226                                       int output_rate, int input_rate)
 227 {
 228     ReSampleContext *s;
 229     int i;
 230
 231     if (output_channels > 2 || input_channels > 2)
 232         return NULL;
 233
 234     s = av_mallocz(sizeof(ReSampleContext));
 235     if (!s)
 236         return NULL;
 237
 238     s->ratio = (float)output_rate / (float)input_rate;
 239
 240     s->input_channels = input_channels;
 241     s->output_channels = output_channels;
 242
 243     s->filter_channels = s->input_channels;
 244     if (s->output_channels < s->filter_channels)
 245         s->filter_channels = s->output_channels;
 246
 247     for(i=0;i<s->filter_channels;i++) {
 248         init_mono_resample(&s->channel_ctx[i], s->ratio);
 249     }
 250     return s;
 251 }
 252
 253 /* resample audio. 'nb_samples' is the number of input samples */
 254 /* XXX: optimize it ! */
 255 /* XXX: do it with polyphase filters, since the quality here is
 256    HORRIBLE. Return the number of samples available in output */
 257 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
 258 {
 259     int i, nb_samples1;
 260     short *bufin[2];
 261     short *bufout[2];
 262     short *buftmp2[2], *buftmp3[2];
 263     int lenout;
 264
 265     if (s->input_channels == s->output_channels && s->ratio == 1.0) {
 266         /* nothing to do */
 267         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
 268         return nb_samples;
 269     }
 270
 271     /* XXX: move those malloc to resample init code */
 272     bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
 273     bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
 274
 275     /* make some zoom to avoid round pb */
 276     lenout= (int)(nb_samples * s->ratio) + 16;
 277     bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
 278     bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
 279
 280     if (s->input_channels == 2 &&
 281         s->output_channels == 1) {
 282         buftmp2[0] = bufin[0];
 283         buftmp3[0] = output;
 284         stereo_to_mono(buftmp2[0], input, nb_samples);
 285     } else if (s->output_channels == 2 && s->input_channels == 1) {
 286         buftmp2[0] = input;
 287         buftmp3[0] = bufout[0];
 288     } else if (s->output_channels == 2) {
 289         buftmp2[0] = bufin[0];
 290         buftmp2[1] = bufin[1];
 291         buftmp3[0] = bufout[0];
 292         buftmp3[1] = bufout[1];
 293         stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
 294     } else {
 295         buftmp2[0] = input;
 296         buftmp3[0] = output;
 297     }
 298
 299     /* resample each channel */
 300     nb_samples1 = 0; /* avoid warning */
 301     for(i=0;i<s->filter_channels;i++) {
 302         nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
 303     }
 304
 305     if (s->output_channels == 2 && s->input_channels == 1) {
 306         mono_to_stereo(output, buftmp3[0], nb_samples1);
 307     } else if (s->output_channels == 2) {
 308         stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 309     }
 310
 311     av_free(bufin[0]);
 312     av_free(bufin[1]);
 313
 314     av_free(bufout[0]);
 315     av_free(bufout[1]);
 316     return nb_samples1;
 317 }
 318
 319 void audio_resample_close(ReSampleContext *s)
 320 {
 321     av_free(s);
 322 }