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 #include "os_support.h"
  27
  28 typedef struct {
  29     /* fractional resampling */
  30     uint32_t incr; /* fractional increment */
  31     uint32_t frac;
  32     int last_sample;
  33     /* integer down sample */
  34     int iratio;  /* integer divison ratio */
  35     int icount, isum;
  36     int inv;
  37 } ReSampleChannelContext;
  38
  39 struct ReSampleContext {
  40     ReSampleChannelContext channel_ctx[2];
  41     float ratio;
  42     /* channel convert */
  43     int input_channels, output_channels, filter_channels;
  44 };
  45
  46
  47 #define FRAC_BITS 16
  48 #define FRAC (1 << FRAC_BITS)
  49
  50 static void init_mono_resample(ReSampleChannelContext *s, float ratio)
  51 {
  52     ratio = 1.0 / ratio;
  53     s->iratio = (int)floorf(ratio);
  54     if (s->iratio == 0)
  55         s->iratio = 1;
  56     s->incr = (int)((ratio / s->iratio) * FRAC);
  57     s->frac = FRAC;
  58     s->last_sample = 0;
  59     s->icount = s->iratio;
  60     s->isum = 0;
  61     s->inv = (FRAC / s->iratio);
  62 }
  63
  64 /* fractional audio resampling */
  65 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
  66 {
  67     unsigned int frac, incr;
  68     int l0, l1;
  69     short *q, *p, *pend;
  70
  71     l0 = s->last_sample;
  72     incr = s->incr;
  73     frac = s->frac;
  74
  75     p = input;
  76     pend = input + nb_samples;
  77     q = output;
  78
  79     l1 = *p++;
  80     for(;;) {
  81         /* interpolate */
  82         *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS;
  83         frac = frac + s->incr;
  84         while (frac >= FRAC) {
  85             frac -= FRAC;
  86             if (p >= pend)
  87                 goto the_end;
  88             l0 = l1;
  89             l1 = *p++;
  90         }
  91     }
  92  the_end:
  93     s->last_sample = l1;
  94     s->frac = frac;
  95     return q - output;
  96 }
  97
  98 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
  99 {
 100     short *q, *p, *pend;
 101     int c, sum;
 102
 103     p = input;
 104     pend = input + nb_samples;
 105     q = output;
 106
 107     c = s->icount;
 108     sum = s->isum;
 109
 110     for(;;) {
 111         sum += *p++;
 112         if (--c == 0) {
 113             *q++ = (sum * s->inv) >> FRAC_BITS;
 114             c = s->iratio;
 115             sum = 0;
 116         }
 117         if (p >= pend)
 118             break;
 119     }
 120     s->isum = sum;
 121     s->icount = c;
 122     return q - output;
 123 }
 124
 125 /* n1: number of samples */
 126 static void stereo_to_mono(short *output, short *input, int n1)
 127 {
 128     short *p, *q;
 129     int n = n1;
 130
 131     p = input;
 132     q = output;
 133     while (n >= 4) {
 134         q[0] = (p[0] + p[1]) >> 1;
 135         q[1] = (p[2] + p[3]) >> 1;
 136         q[2] = (p[4] + p[5]) >> 1;
 137         q[3] = (p[6] + p[7]) >> 1;
 138         q += 4;
 139         p += 8;
 140         n -= 4;
 141     }
 142     while (n > 0) {
 143         q[0] = (p[0] + p[1]) >> 1;
 144         q++;
 145         p += 2;
 146         n--;
 147     }
 148 }
 149
 150 /* n1: number of samples */
 151 static void mono_to_stereo(short *output, short *input, int n1)
 152 {
 153     short *p, *q;
 154     int n = n1;
 155     int v;
 156
 157     p = input;
 158     q = output;
 159     while (n >= 4) {
 160         v = p[0]; q[0] = v; q[1] = v;
 161         v = p[1]; q[2] = v; q[3] = v;
 162         v = p[2]; q[4] = v; q[5] = v;
 163         v = p[3]; q[6] = v; q[7] = v;
 164         q += 8;
 165         p += 4;
 166         n -= 4;
 167     }
 168     while (n > 0) {
 169         v = p[0]; q[0] = v; q[1] = v;
 170         q += 2;
 171         p += 1;
 172         n--;
 173     }
 174 }
 175
 176 /* XXX: should use more abstract 'N' channels system */
 177 static void stereo_split(short *output1, short *output2, short *input, int n)
 178 {
 179     int i;
 180
 181     for(i=0;i<n;i++) {
 182         *output1++ = *input++;
 183         *output2++ = *input++;
 184     }
 185 }
 186
 187 static void stereo_mux(short *output, short *input1, short *input2, int n)
 188 {
 189     int i;
 190
 191     for(i=0;i<n;i++) {
 192         *output++ = *input1++;
 193         *output++ = *input2++;
 194     }
 195 }
 196
 197 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples)
 198 {
 199     short *buf1;
 200     short *buftmp;
 201
 202     buf1= (short*)av_malloc( nb_samples * sizeof(short) );
 203
 204     /* first downsample by an integer factor with averaging filter */
 205     if (s->iratio > 1) {
 206         buftmp = buf1;
 207         nb_samples = integer_downsample(s, buftmp, input, nb_samples);
 208     } else {
 209         buftmp = input;
 210     }
 211
 212     /* then do a fractional resampling with linear interpolation */
 213     if (s->incr != FRAC) {
 214         nb_samples = fractional_resample(s, output, buftmp, nb_samples);
 215     } else {
 216         memcpy(output, buftmp, nb_samples * sizeof(short));
 217     }
 218     av_free(buf1);
 219     return nb_samples;
 220 }
 221
 222 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
 223                                       int output_rate, int input_rate)
 224 {
 225     ReSampleContext *s;
 226     int i;
 227
 228     if (output_channels > 2 || input_channels > 2)
 229         return NULL;
 230
 231     s = av_mallocz(sizeof(ReSampleContext));
 232     if (!s)
 233         return NULL;
 234
 235     s->ratio = (float)output_rate / (float)input_rate;
 236
 237     s->input_channels = input_channels;
 238     s->output_channels = output_channels;
 239
 240     s->filter_channels = s->input_channels;
 241     if (s->output_channels < s->filter_channels)
 242         s->filter_channels = s->output_channels;
 243
 244     for(i=0;i<s->filter_channels;i++) {
 245         init_mono_resample(&s->channel_ctx[i], s->ratio);
 246     }
 247     return s;
 248 }
 249
 250 /* resample audio. 'nb_samples' is the number of input samples */
 251 /* XXX: optimize it ! */
 252 /* XXX: do it with polyphase filters, since the quality here is
 253    HORRIBLE. Return the number of samples available in output */
 254 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
 255 {
 256     int i, nb_samples1;
 257     short *bufin[2];
 258     short *bufout[2];
 259     short *buftmp2[2], *buftmp3[2];
 260     int lenout;
 261
 262     if (s->input_channels == s->output_channels && s->ratio == 1.0) {
 263         /* nothing to do */
 264         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
 265         return nb_samples;
 266     }
 267
 268     /* XXX: move those malloc to resample init code */
 269     bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) );
 270     bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) );
 271
 272     /* make some zoom to avoid round pb */
 273     lenout= (int)(nb_samples * s->ratio) + 16;
 274     bufout[0]= (short*) av_malloc( lenout * sizeof(short) );
 275     bufout[1]= (short*) av_malloc( lenout * sizeof(short) );
 276
 277     if (s->input_channels == 2 &&
 278         s->output_channels == 1) {
 279         buftmp2[0] = bufin[0];
 280         buftmp3[0] = output;
 281         stereo_to_mono(buftmp2[0], input, nb_samples);
 282     } else if (s->output_channels == 2 && s->input_channels == 1) {
 283         buftmp2[0] = input;
 284         buftmp3[0] = bufout[0];
 285     } else if (s->output_channels == 2) {
 286         buftmp2[0] = bufin[0];
 287         buftmp2[1] = bufin[1];
 288         buftmp3[0] = bufout[0];
 289         buftmp3[1] = bufout[1];
 290         stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
 291     } else {
 292         buftmp2[0] = input;
 293         buftmp3[0] = output;
 294     }
 295
 296     /* resample each channel */
 297     nb_samples1 = 0; /* avoid warning */
 298     for(i=0;i<s->filter_channels;i++) {
 299         nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples);
 300     }
 301
 302     if (s->output_channels == 2 && s->input_channels == 1) {
 303         mono_to_stereo(output, buftmp3[0], nb_samples1);
 304     } else if (s->output_channels == 2) {
 305         stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 306     }
 307
 308     av_free(bufin[0]);
 309     av_free(bufin[1]);
 310
 311     av_free(bufout[0]);
 312     av_free(bufout[1]);
 313     return nb_samples1;
 314 }
 315
 316 void audio_resample_close(ReSampleContext *s)
 317 {
 318     av_free(s);
 319 }