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 Libav.
   6  *
   7  * Libav 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  * Libav 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 Libav; 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
  24  * samplerate conversion for both audio and video
  25  */
  26
  27 #include <string.h>
  28
  29 #include "avcodec.h"
  30 #include "audioconvert.h"
  31 #include "libavutil/opt.h"
  32 #include "libavutil/mem.h"
  33 #include "libavutil/samplefmt.h"
  34
  35 #if FF_API_AVCODEC_RESAMPLE
  36
  37 #define MAX_CHANNELS 8
  38
  39 struct AVResampleContext;
  40
  41 static const char *context_to_name(void *ptr)
  42 {
  43     return "audioresample";
  44 }
  45
  46 static const AVOption options[] = {{NULL}};
  47 static const AVClass audioresample_context_class = {
  48     "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
  49 };
  50
  51 struct ReSampleContext {
  52     struct AVResampleContext *resample_context;
  53     short *temp[MAX_CHANNELS];
  54     int temp_len;
  55     float ratio;
  56     /* channel convert */
  57     int input_channels, output_channels, filter_channels;
  58     AVAudioConvert *convert_ctx[2];
  59     enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
  60     unsigned sample_size[2];           ///< size of one sample in sample_fmt
  61     short *buffer[2];                  ///< buffers used for conversion to S16
  62     unsigned buffer_size[2];           ///< sizes of allocated buffers
  63 };
  64
  65 /* n1: number of samples */
  66 static void stereo_to_mono(short *output, short *input, int n1)
  67 {
  68     short *p, *q;
  69     int n = n1;
  70
  71     p = input;
  72     q = output;
  73     while (n >= 4) {
  74         q[0] = (p[0] + p[1]) >> 1;
  75         q[1] = (p[2] + p[3]) >> 1;
  76         q[2] = (p[4] + p[5]) >> 1;
  77         q[3] = (p[6] + p[7]) >> 1;
  78         q += 4;
  79         p += 8;
  80         n -= 4;
  81     }
  82     while (n > 0) {
  83         q[0] = (p[0] + p[1]) >> 1;
  84         q++;
  85         p += 2;
  86         n--;
  87     }
  88 }
  89
  90 /* n1: number of samples */
  91 static void mono_to_stereo(short *output, short *input, int n1)
  92 {
  93     short *p, *q;
  94     int n = n1;
  95     int v;
  96
  97     p = input;
  98     q = output;
  99     while (n >= 4) {
 100         v = p[0]; q[0] = v; q[1] = v;
 101         v = p[1]; q[2] = v; q[3] = v;
 102         v = p[2]; q[4] = v; q[5] = v;
 103         v = p[3]; q[6] = v; q[7] = v;
 104         q += 8;
 105         p += 4;
 106         n -= 4;
 107     }
 108     while (n > 0) {
 109         v = p[0]; q[0] = v; q[1] = v;
 110         q += 2;
 111         p += 1;
 112         n--;
 113     }
 114 }
 115
 116 static void deinterleave(short **output, short *input, int channels, int samples)
 117 {
 118     int i, j;
 119
 120     for (i = 0; i < samples; i++) {
 121         for (j = 0; j < channels; j++) {
 122             *output[j]++ = *input++;
 123         }
 124     }
 125 }
 126
 127 static void interleave(short *output, short **input, int channels, int samples)
 128 {
 129     int i, j;
 130
 131     for (i = 0; i < samples; i++) {
 132         for (j = 0; j < channels; j++) {
 133             *output++ = *input[j]++;
 134         }
 135     }
 136 }
 137
 138 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
 139 {
 140     int i;
 141     short l, r;
 142
 143     for (i = 0; i < n; i++) {
 144         l = *input1++;
 145         r = *input2++;
 146         *output++ = l;                  /* left */
 147         *output++ = (l / 2) + (r / 2);  /* center */
 148         *output++ = r;                  /* right */
 149         *output++ = 0;                  /* left surround */
 150         *output++ = 0;                  /* right surroud */
 151         *output++ = 0;                  /* low freq */
 152     }
 153 }
 154
 155 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
 156                                         int output_rate, int input_rate,
 157                                         enum AVSampleFormat sample_fmt_out,
 158                                         enum AVSampleFormat sample_fmt_in,
 159                                         int filter_length, int log2_phase_count,
 160                                         int linear, double cutoff)
 161 {
 162     ReSampleContext *s;
 163
 164     if (input_channels > MAX_CHANNELS) {
 165         av_log(NULL, AV_LOG_ERROR,
 166                "Resampling with input channels greater than %d is unsupported.\n",
 167                MAX_CHANNELS);
 168         return NULL;
 169     }
 170     if (output_channels != input_channels &&
 171         (input_channels  > 2 ||
 172          output_channels > 2 &&
 173          !(output_channels == 6 && input_channels == 2))) {
 174         av_log(NULL, AV_LOG_ERROR,
 175                "Resampling output channel count must be 1 or 2 for mono input; 1, 2 or 6 for stereo input; or N for N channel input.\n");
 176         return NULL;
 177     }
 178
 179     s = av_mallocz(sizeof(ReSampleContext));
 180     if (!s) {
 181         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
 182         return NULL;
 183     }
 184
 185     s->ratio = (float)output_rate / (float)input_rate;
 186
 187     s->input_channels = input_channels;
 188     s->output_channels = output_channels;
 189
 190     s->filter_channels = s->input_channels;
 191     if (s->output_channels < s->filter_channels)
 192         s->filter_channels = s->output_channels;
 193
 194     s->sample_fmt[0]  = sample_fmt_in;
 195     s->sample_fmt[1]  = sample_fmt_out;
 196     s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
 197     s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
 198
 199     if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
 200         if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
 201                                                          s->sample_fmt[0], 1, NULL, 0))) {
 202             av_log(s, AV_LOG_ERROR,
 203                    "Cannot convert %s sample format to s16 sample format\n",
 204                    av_get_sample_fmt_name(s->sample_fmt[0]));
 205             av_free(s);
 206             return NULL;
 207         }
 208     }
 209
 210     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
 211         if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
 212                                                          AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
 213             av_log(s, AV_LOG_ERROR,
 214                    "Cannot convert s16 sample format to %s sample format\n",
 215                    av_get_sample_fmt_name(s->sample_fmt[1]));
 216             av_audio_convert_free(s->convert_ctx[0]);
 217             av_free(s);
 218             return NULL;
 219         }
 220     }
 221
 222     s->resample_context = av_resample_init(output_rate, input_rate,
 223                                            filter_length, log2_phase_count,
 224                                            linear, cutoff);
 225
 226     *(const AVClass**)s->resample_context = &audioresample_context_class;
 227
 228     return s;
 229 }
 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[MAX_CHANNELS];
 237     short *bufout[MAX_CHANNELS];
 238     short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
 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] != AV_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,
 270                    "Audio sample format conversion failed\n");
 271             return 0;
 272         }
 273
 274         input = s->buffer[0];
 275     }
 276
 277     lenout = 4 * nb_samples * s->ratio + 16;
 278
 279     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
 280         int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
 281                        s->output_channels;
 282         output_bak = output;
 283
 284         if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
 285             av_free(s->buffer[1]);
 286             s->buffer_size[1] = out_size;
 287             s->buffer[1] = av_malloc(s->buffer_size[1]);
 288             if (!s->buffer[1]) {
 289                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
 290                 return 0;
 291             }
 292         }
 293
 294         output = s->buffer[1];
 295     }
 296
 297     /* XXX: move those malloc to resample init code */
 298     for (i = 0; i < s->filter_channels; i++) {
 299         bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
 300         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
 301         buftmp2[i] = bufin[i] + s->temp_len;
 302         bufout[i] = av_malloc(lenout * sizeof(short));
 303     }
 304
 305     if (s->input_channels == 2 && s->output_channels == 1) {
 306         buftmp3[0] = output;
 307         stereo_to_mono(buftmp2[0], input, nb_samples);
 308     } else if (s->output_channels >= 2 && s->input_channels == 1) {
 309         buftmp3[0] = bufout[0];
 310         memcpy(buftmp2[0], input, nb_samples * sizeof(short));
 311     } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
 312         for (i = 0; i < s->input_channels; i++) {
 313             buftmp3[i] = bufout[i];
 314         }
 315         deinterleave(buftmp2, input, s->input_channels, 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],
 330                                   &consumed, nb_samples, lenout, is_last);
 331         s->temp_len = nb_samples - consumed;
 332         s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short));
 333         memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
 334     }
 335
 336     if (s->output_channels == 2 && s->input_channels == 1) {
 337         mono_to_stereo(output, buftmp3[0], nb_samples1);
 338     } else if (s->output_channels == 6 && s->input_channels == 2) {
 339         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 340     } else if (s->output_channels == s->input_channels && s->input_channels >= 2) {
 341         interleave(output, buftmp3, s->output_channels, nb_samples1);
 342     }
 343
 344     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
 345         int istride[1] = { 2 };
 346         int ostride[1] = { s->sample_size[1] };
 347         const void *ibuf[1] = { output };
 348         void       *obuf[1] = { output_bak };
 349
 350         if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
 351                              ibuf, istride, nb_samples1 * s->output_channels) < 0) {
 352             av_log(s->resample_context, AV_LOG_ERROR,
 353                    "Audio sample format conversion failed\n");
 354             return 0;
 355         }
 356     }
 357
 358     for (i = 0; i < s->filter_channels; i++) {
 359         av_free(bufin[i]);
 360         av_free(bufout[i]);
 361     }
 362
 363     return nb_samples1;
 364 }
 365
 366 void audio_resample_close(ReSampleContext *s)
 367 {
 368     int i;
 369     av_resample_close(s->resample_context);
 370     for (i = 0; i < s->filter_channels; i++)
 371         av_freep(&s->temp[i]);
 372     av_freep(&s->buffer[0]);
 373     av_freep(&s->buffer[1]);
 374     av_audio_convert_free(s->convert_ctx[0]);
 375     av_audio_convert_free(s->convert_ctx[1]);
 376     av_free(s);
 377 }
 378
 379 #endif