libavcodec/resample.c

   1 /*
   2  * samplerate conversion for both audio and video
   3  * Copyright (c) 2000 Fabrice Bellard
   4  *
   5  * This file is part of FFmpeg.
   6  *
   7  * FFmpeg is free software; you can redistribute it and/or
   8  * modify it under the terms of the GNU Lesser General Public
   9  * License as published by the Free Software Foundation; either
  10  * version 2.1 of the License, or (at your option) any later version.
  11  *
  12  * FFmpeg is distributed in the hope that it will be useful,
  13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  15  * Lesser General Public License for more details.
  16  *
  17  * You should have received a copy of the GNU Lesser General Public
  18  * License along with FFmpeg; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 /**
  23  * @file libavcodec/resample.c
  24  * samplerate conversion for both audio and video
  25  */
  26
  27 #include "avcodec.h"
  28 #include "audioconvert.h"
  29 #include "opt.h"
  30
  31 struct AVResampleContext;
  32
  33 static const char *context_to_name(void *ptr)
  34 {
  35     return "audioresample";
  36 }
  37
  38 static const AVOption options[] = {{NULL}};
  39 static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options };
  40
  41 struct ReSampleContext {
  42     struct AVResampleContext *resample_context;
  43     short *temp[2];
  44     int temp_len;
  45     float ratio;
  46     /* channel convert */
  47     int input_channels, output_channels, filter_channels;
  48     AVAudioConvert *convert_ctx[2];
  49     enum SampleFormat sample_fmt[2]; ///< input and output sample format
  50     unsigned sample_size[2];         ///< size of one sample in sample_fmt
  51     short *buffer[2];                ///< buffers used for conversion to S16
  52     unsigned buffer_size[2];         ///< sizes of allocated buffers
  53 };
  54
  55 /* n1: number of samples */
  56 static void stereo_to_mono(short *output, short *input, int n1)
  57 {
  58     short *p, *q;
  59     int n = n1;
  60
  61     p = input;
  62     q = output;
  63     while (n >= 4) {
  64         q[0] = (p[0] + p[1]) >> 1;
  65         q[1] = (p[2] + p[3]) >> 1;
  66         q[2] = (p[4] + p[5]) >> 1;
  67         q[3] = (p[6] + p[7]) >> 1;
  68         q += 4;
  69         p += 8;
  70         n -= 4;
  71     }
  72     while (n > 0) {
  73         q[0] = (p[0] + p[1]) >> 1;
  74         q++;
  75         p += 2;
  76         n--;
  77     }
  78 }
  79
  80 /* n1: number of samples */
  81 static void mono_to_stereo(short *output, short *input, int n1)
  82 {
  83     short *p, *q;
  84     int n = n1;
  85     int v;
  86
  87     p = input;
  88     q = output;
  89     while (n >= 4) {
  90         v = p[0]; q[0] = v; q[1] = v;
  91         v = p[1]; q[2] = v; q[3] = v;
  92         v = p[2]; q[4] = v; q[5] = v;
  93         v = p[3]; q[6] = v; q[7] = v;
  94         q += 8;
  95         p += 4;
  96         n -= 4;
  97     }
  98     while (n > 0) {
  99         v = p[0]; q[0] = v; q[1] = v;
 100         q += 2;
 101         p += 1;
 102         n--;
 103     }
 104 }
 105
 106 /* XXX: should use more abstract 'N' channels system */
 107 static void stereo_split(short *output1, short *output2, short *input, int n)
 108 {
 109     int i;
 110
 111     for(i=0;i<n;i++) {
 112         *output1++ = *input++;
 113         *output2++ = *input++;
 114     }
 115 }
 116
 117 static void stereo_mux(short *output, short *input1, short *input2, int n)
 118 {
 119     int i;
 120
 121     for(i=0;i<n;i++) {
 122         *output++ = *input1++;
 123         *output++ = *input2++;
 124     }
 125 }
 126
 127 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
 128 {
 129     int i;
 130     short l,r;
 131
 132     for(i=0;i<n;i++) {
 133       l=*input1++;
 134       r=*input2++;
 135       *output++ = l;           /* left */
 136       *output++ = (l/2)+(r/2); /* center */
 137       *output++ = r;           /* right */
 138       *output++ = 0;           /* left surround */
 139       *output++ = 0;           /* right surroud */
 140       *output++ = 0;           /* low freq */
 141     }
 142 }
 143
 144 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
 145                                         int output_rate, int input_rate,
 146                                         enum SampleFormat sample_fmt_out,
 147                                         enum SampleFormat sample_fmt_in,
 148                                         int filter_length, int log2_phase_count,
 149                                         int linear, double cutoff)
 150 {
 151     ReSampleContext *s;
 152
 153     if ( input_channels > 2)
 154       {
 155         av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n");
 156         return NULL;
 157       }
 158
 159     s = av_mallocz(sizeof(ReSampleContext));
 160     if (!s)
 161       {
 162         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
 163         return NULL;
 164       }
 165
 166     s->ratio = (float)output_rate / (float)input_rate;
 167
 168     s->input_channels = input_channels;
 169     s->output_channels = output_channels;
 170
 171     s->filter_channels = s->input_channels;
 172     if (s->output_channels < s->filter_channels)
 173         s->filter_channels = s->output_channels;
 174
 175     s->sample_fmt [0] = sample_fmt_in;
 176     s->sample_fmt [1] = sample_fmt_out;
 177     s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3;
 178     s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3;
 179
 180     if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
 181         if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1,
 182                                                          s->sample_fmt[0], 1, NULL, 0))) {
 183             av_log(s, AV_LOG_ERROR,
 184                    "Cannot convert %s sample format to s16 sample format\n",
 185                    avcodec_get_sample_fmt_name(s->sample_fmt[0]));
 186             av_free(s);
 187             return NULL;
 188         }
 189     }
 190
 191     if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
 192         if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
 193                                                          SAMPLE_FMT_S16, 1, NULL, 0))) {
 194             av_log(s, AV_LOG_ERROR,
 195                    "Cannot convert s16 sample format to %s sample format\n",
 196                    avcodec_get_sample_fmt_name(s->sample_fmt[1]));
 197             av_audio_convert_free(s->convert_ctx[0]);
 198             av_free(s);
 199             return NULL;
 200         }
 201     }
 202
 203 /*
 204  * AC-3 output is the only case where filter_channels could be greater than 2.
 205  * input channels can't be greater than 2, so resample the 2 channels and then
 206  * expand to 6 channels after the resampling.
 207  */
 208     if(s->filter_channels>2)
 209       s->filter_channels = 2;
 210
 211 #define TAPS 16
 212     s->resample_context= av_resample_init(output_rate, input_rate,
 213                          filter_length, log2_phase_count, linear, cutoff);
 214
 215     *(const AVClass**)s->resample_context = &audioresample_context_class;
 216
 217     return s;
 218 }
 219
 220 #if LIBAVCODEC_VERSION_MAJOR < 53
 221 ReSampleContext *audio_resample_init(int output_channels, int input_channels,
 222                                      int output_rate, int input_rate)
 223 {
 224     return av_audio_resample_init(output_channels, input_channels,
 225                                   output_rate, input_rate,
 226                                   SAMPLE_FMT_S16, SAMPLE_FMT_S16,
 227                                   TAPS, 10, 0, 0.8);
 228 }
 229 #endif
 230
 231 /* resample audio. 'nb_samples' is the number of input samples */
 232 /* XXX: optimize it ! */
 233 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
 234 {
 235     int i, nb_samples1;
 236     short *bufin[2];
 237     short *bufout[2];
 238     short *buftmp2[2], *buftmp3[2];
 239     short *output_bak = NULL;
 240     int lenout;
 241
 242     if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
 243         /* nothing to do */
 244         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
 245         return nb_samples;
 246     }
 247
 248     if (s->sample_fmt[0] != SAMPLE_FMT_S16) {
 249         int istride[1] = { s->sample_size[0] };
 250         int ostride[1] = { 2 };
 251         const void *ibuf[1] = { input };
 252         void       *obuf[1];
 253         unsigned input_size = nb_samples*s->input_channels*2;
 254
 255         if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
 256             av_free(s->buffer[0]);
 257             s->buffer_size[0] = input_size;
 258             s->buffer[0] = av_malloc(s->buffer_size[0]);
 259             if (!s->buffer[0]) {
 260                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
 261                 return 0;
 262             }
 263         }
 264
 265         obuf[0] = s->buffer[0];
 266
 267         if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
 268                              ibuf, istride, nb_samples*s->input_channels) < 0) {
 269             av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format conversion failed\n");
 270             return 0;
 271         }
 272
 273         input  = s->buffer[0];
 274     }
 275
 276     lenout= 4*nb_samples * s->ratio + 16;
 277
 278     if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
 279         output_bak = output;
 280
 281         if (!s->buffer_size[1] || s->buffer_size[1] < lenout) {
 282             av_free(s->buffer[1]);
 283             s->buffer_size[1] = lenout;
 284             s->buffer[1] = av_malloc(s->buffer_size[1]);
 285             if (!s->buffer[1]) {
 286                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
 287                 return 0;
 288             }
 289         }
 290
 291         output = s->buffer[1];
 292     }
 293
 294     /* XXX: move those malloc to resample init code */
 295     for(i=0; i<s->filter_channels; i++){
 296         bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) );
 297         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
 298         buftmp2[i] = bufin[i] + s->temp_len;
 299     }
 300
 301     /* make some zoom to avoid round pb */
 302     bufout[0]= av_malloc( lenout * sizeof(short) );
 303     bufout[1]= av_malloc( lenout * sizeof(short) );
 304
 305     if (s->input_channels == 2 &&
 306         s->output_channels == 1) {
 307         buftmp3[0] = output;
 308         stereo_to_mono(buftmp2[0], input, nb_samples);
 309     } else if (s->output_channels >= 2 && s->input_channels == 1) {
 310         buftmp3[0] = bufout[0];
 311         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
 312     } else if (s->output_channels >= 2) {
 313         buftmp3[0] = bufout[0];
 314         buftmp3[1] = bufout[1];
 315         stereo_split(buftmp2[0], buftmp2[1], input, nb_samples);
 316     } else {
 317         buftmp3[0] = output;
 318         memcpy(buftmp2[0], input, nb_samples*sizeof(short));
 319     }
 320
 321     nb_samples += s->temp_len;
 322
 323     /* resample each channel */
 324     nb_samples1 = 0; /* avoid warning */
 325     for(i=0;i<s->filter_channels;i++) {
 326         int consumed;
 327         int is_last= i+1 == s->filter_channels;
 328
 329         nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &consumed, nb_samples, lenout, is_last);
 330         s->temp_len= nb_samples - consumed;
 331         s->temp[i]= av_realloc(s->temp[i], s->temp_len*sizeof(short));
 332         memcpy(s->temp[i], bufin[i] + consumed, s->temp_len*sizeof(short));
 333     }
 334
 335     if (s->output_channels == 2 && s->input_channels == 1) {
 336         mono_to_stereo(output, buftmp3[0], nb_samples1);
 337     } else if (s->output_channels == 2) {
 338         stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 339     } else if (s->output_channels == 6) {
 340         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 341     }
 342
 343     if (s->sample_fmt[1] != SAMPLE_FMT_S16) {
 344         int istride[1] = { 2 };
 345         int ostride[1] = { s->sample_size[1] };
 346         const void *ibuf[1] = { output };
 347         void       *obuf[1] = { output_bak };
 348
 349         if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
 350                              ibuf, istride, nb_samples1*s->output_channels) < 0) {
 351             av_log(s->resample_context, AV_LOG_ERROR, "Audio sample format convertion failed\n");
 352             return 0;
 353         }
 354     }
 355
 356     for(i=0; i<s->filter_channels; i++)
 357         av_free(bufin[i]);
 358
 359     av_free(bufout[0]);
 360     av_free(bufout[1]);
 361     return nb_samples1;
 362 }
 363
 364 void audio_resample_close(ReSampleContext *s)
 365 {
 366     av_resample_close(s->resample_context);
 367     av_freep(&s->temp[0]);
 368     av_freep(&s->temp[1]);
 369     av_freep(&s->buffer[0]);
 370     av_freep(&s->buffer[1]);
 371     av_audio_convert_free(s->convert_ctx[0]);
 372     av_audio_convert_free(s->convert_ctx[1]);
 373     av_free(s);
 374 }