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 "avcodec.h"
  28 #include "audioconvert.h"
  29 #include "libavutil/opt.h"
  30 #include "libavutil/samplefmt.h"
  31
  32 #define MAX_CHANNELS 8
  33
  34 struct AVResampleContext;
  35
  36 static const char *context_to_name(void *ptr)
  37 {
  38     return "audioresample";
  39 }
  40
  41 static const AVOption options[] = {{NULL}};
  42 static const AVClass audioresample_context_class = {
  43     "ReSampleContext", context_to_name, options, LIBAVUTIL_VERSION_INT
  44 };
  45
  46 struct ReSampleContext {
  47     struct AVResampleContext *resample_context;
  48     short *temp[MAX_CHANNELS];
  49     int temp_len;
  50     float ratio;
  51     /* channel convert */
  52     int input_channels, output_channels, filter_channels;
  53     AVAudioConvert *convert_ctx[2];
  54     enum AVSampleFormat sample_fmt[2]; ///< input and output sample format
  55     unsigned sample_size[2];           ///< size of one sample in sample_fmt
  56     short *buffer[2];                  ///< buffers used for conversion to S16
  57     unsigned buffer_size[2];           ///< sizes of allocated buffers
  58 };
  59
  60 /* n1: number of samples */
  61 static void stereo_to_mono(short *output, short *input, int n1)
  62 {
  63     short *p, *q;
  64     int n = n1;
  65
  66     p = input;
  67     q = output;
  68     while (n >= 4) {
  69         q[0] = (p[0] + p[1]) >> 1;
  70         q[1] = (p[2] + p[3]) >> 1;
  71         q[2] = (p[4] + p[5]) >> 1;
  72         q[3] = (p[6] + p[7]) >> 1;
  73         q += 4;
  74         p += 8;
  75         n -= 4;
  76     }
  77     while (n > 0) {
  78         q[0] = (p[0] + p[1]) >> 1;
  79         q++;
  80         p += 2;
  81         n--;
  82     }
  83 }
  84
  85 /* n1: number of samples */
  86 static void mono_to_stereo(short *output, short *input, int n1)
  87 {
  88     short *p, *q;
  89     int n = n1;
  90     int v;
  91
  92     p = input;
  93     q = output;
  94     while (n >= 4) {
  95         v = p[0]; q[0] = v; q[1] = v;
  96         v = p[1]; q[2] = v; q[3] = v;
  97         v = p[2]; q[4] = v; q[5] = v;
  98         v = p[3]; q[6] = v; q[7] = v;
  99         q += 8;
 100         p += 4;
 101         n -= 4;
 102     }
 103     while (n > 0) {
 104         v = p[0]; q[0] = v; q[1] = v;
 105         q += 2;
 106         p += 1;
 107         n--;
 108     }
 109 }
 110
 111 static void deinterleave(short **output, short *input, int channels, int samples)
 112 {
 113     int i, j;
 114
 115     for (i = 0; i < samples; i++) {
 116         for (j = 0; j < channels; j++) {
 117             *output[j]++ = *input++;
 118         }
 119     }
 120 }
 121
 122 static void interleave(short *output, short **input, int channels, int samples)
 123 {
 124     int i, j;
 125
 126     for (i = 0; i < samples; i++) {
 127         for (j = 0; j < channels; j++) {
 128             *output++ = *input[j]++;
 129         }
 130     }
 131 }
 132
 133 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n)
 134 {
 135     int i;
 136     short l, r;
 137
 138     for (i = 0; i < n; i++) {
 139         l = *input1++;
 140         r = *input2++;
 141         *output++ = l;                  /* left */
 142         *output++ = (l / 2) + (r / 2);  /* center */
 143         *output++ = r;                  /* right */
 144         *output++ = 0;                  /* left surround */
 145         *output++ = 0;                  /* right surroud */
 146         *output++ = 0;                  /* low freq */
 147     }
 148 }
 149
 150 ReSampleContext *av_audio_resample_init(int output_channels, int input_channels,
 151                                         int output_rate, int input_rate,
 152                                         enum AVSampleFormat sample_fmt_out,
 153                                         enum AVSampleFormat sample_fmt_in,
 154                                         int filter_length, int log2_phase_count,
 155                                         int linear, double cutoff)
 156 {
 157     ReSampleContext *s;
 158
 159     if (input_channels > MAX_CHANNELS) {
 160         av_log(NULL, AV_LOG_ERROR,
 161                "Resampling with input channels greater than %d is unsupported.\n",
 162                MAX_CHANNELS);
 163         return NULL;
 164     }
 165     if (output_channels != input_channels &&
 166         (input_channels  > 2 ||
 167          output_channels > 2 &&
 168          !(output_channels == 6 && input_channels == 2))) {
 169         av_log(NULL, AV_LOG_ERROR,
 170                "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");
 171         return NULL;
 172     }
 173
 174     s = av_mallocz(sizeof(ReSampleContext));
 175     if (!s) {
 176         av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n");
 177         return NULL;
 178     }
 179
 180     s->ratio = (float)output_rate / (float)input_rate;
 181
 182     s->input_channels = input_channels;
 183     s->output_channels = output_channels;
 184
 185     s->filter_channels = s->input_channels;
 186     if (s->output_channels < s->filter_channels)
 187         s->filter_channels = s->output_channels;
 188
 189     s->sample_fmt[0]  = sample_fmt_in;
 190     s->sample_fmt[1]  = sample_fmt_out;
 191     s->sample_size[0] = av_get_bytes_per_sample(s->sample_fmt[0]);
 192     s->sample_size[1] = av_get_bytes_per_sample(s->sample_fmt[1]);
 193
 194     if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
 195         if (!(s->convert_ctx[0] = av_audio_convert_alloc(AV_SAMPLE_FMT_S16, 1,
 196                                                          s->sample_fmt[0], 1, NULL, 0))) {
 197             av_log(s, AV_LOG_ERROR,
 198                    "Cannot convert %s sample format to s16 sample format\n",
 199                    av_get_sample_fmt_name(s->sample_fmt[0]));
 200             av_free(s);
 201             return NULL;
 202         }
 203     }
 204
 205     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
 206         if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1,
 207                                                          AV_SAMPLE_FMT_S16, 1, NULL, 0))) {
 208             av_log(s, AV_LOG_ERROR,
 209                    "Cannot convert s16 sample format to %s sample format\n",
 210                    av_get_sample_fmt_name(s->sample_fmt[1]));
 211             av_audio_convert_free(s->convert_ctx[0]);
 212             av_free(s);
 213             return NULL;
 214         }
 215     }
 216
 217     s->resample_context = av_resample_init(output_rate, input_rate,
 218                                            filter_length, log2_phase_count,
 219                                            linear, cutoff);
 220
 221     *(const AVClass**)s->resample_context = &audioresample_context_class;
 222
 223     return s;
 224 }
 225
 226 /* resample audio. 'nb_samples' is the number of input samples */
 227 /* XXX: optimize it ! */
 228 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples)
 229 {
 230     int i, nb_samples1;
 231     short *bufin[MAX_CHANNELS];
 232     short *bufout[MAX_CHANNELS];
 233     short *buftmp2[MAX_CHANNELS], *buftmp3[MAX_CHANNELS];
 234     short *output_bak = NULL;
 235     int lenout;
 236
 237     if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) {
 238         /* nothing to do */
 239         memcpy(output, input, nb_samples * s->input_channels * sizeof(short));
 240         return nb_samples;
 241     }
 242
 243     if (s->sample_fmt[0] != AV_SAMPLE_FMT_S16) {
 244         int istride[1] = { s->sample_size[0] };
 245         int ostride[1] = { 2 };
 246         const void *ibuf[1] = { input };
 247         void       *obuf[1];
 248         unsigned input_size = nb_samples * s->input_channels * 2;
 249
 250         if (!s->buffer_size[0] || s->buffer_size[0] < input_size) {
 251             av_free(s->buffer[0]);
 252             s->buffer_size[0] = input_size;
 253             s->buffer[0] = av_malloc(s->buffer_size[0]);
 254             if (!s->buffer[0]) {
 255                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
 256                 return 0;
 257             }
 258         }
 259
 260         obuf[0] = s->buffer[0];
 261
 262         if (av_audio_convert(s->convert_ctx[0], obuf, ostride,
 263                              ibuf, istride, nb_samples * s->input_channels) < 0) {
 264             av_log(s->resample_context, AV_LOG_ERROR,
 265                    "Audio sample format conversion failed\n");
 266             return 0;
 267         }
 268
 269         input = s->buffer[0];
 270     }
 271
 272     lenout = 4 * nb_samples * s->ratio + 16;
 273
 274     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
 275         int out_size = lenout * av_get_bytes_per_sample(s->sample_fmt[1]) *
 276                        s->output_channels;
 277         output_bak = output;
 278
 279         if (!s->buffer_size[1] || s->buffer_size[1] < out_size) {
 280             av_free(s->buffer[1]);
 281             s->buffer_size[1] = out_size;
 282             s->buffer[1] = av_malloc(s->buffer_size[1]);
 283             if (!s->buffer[1]) {
 284                 av_log(s->resample_context, AV_LOG_ERROR, "Could not allocate buffer\n");
 285                 return 0;
 286             }
 287         }
 288
 289         output = s->buffer[1];
 290     }
 291
 292     /* XXX: move those malloc to resample init code */
 293     for (i = 0; i < s->filter_channels; i++) {
 294         bufin[i] = av_malloc((nb_samples + s->temp_len) * sizeof(short));
 295         memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short));
 296         buftmp2[i] = bufin[i] + s->temp_len;
 297         bufout[i] = av_malloc(lenout * sizeof(short));
 298     }
 299
 300     if (s->input_channels == 2 && s->output_channels == 1) {
 301         buftmp3[0] = output;
 302         stereo_to_mono(buftmp2[0], input, nb_samples);
 303     } else if (s->output_channels >= 2 && s->input_channels == 1) {
 304         buftmp3[0] = bufout[0];
 305         memcpy(buftmp2[0], input, nb_samples * sizeof(short));
 306     } else if (s->output_channels >= s->input_channels && s->input_channels >= 2) {
 307         for (i = 0; i < s->input_channels; i++) {
 308             buftmp3[i] = bufout[i];
 309         }
 310         deinterleave(buftmp2, input, s->input_channels, nb_samples);
 311     } else {
 312         buftmp3[0] = output;
 313         memcpy(buftmp2[0], input, nb_samples * sizeof(short));
 314     }
 315
 316     nb_samples += s->temp_len;
 317
 318     /* resample each channel */
 319     nb_samples1 = 0; /* avoid warning */
 320     for (i = 0; i < s->filter_channels; i++) {
 321         int consumed;
 322         int is_last = i + 1 == s->filter_channels;
 323
 324         nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i],
 325                                   &consumed, nb_samples, lenout, is_last);
 326         s->temp_len = nb_samples - consumed;
 327         s->temp[i] = av_realloc(s->temp[i], s->temp_len * sizeof(short));
 328         memcpy(s->temp[i], bufin[i] + consumed, s->temp_len * sizeof(short));
 329     }
 330
 331     if (s->output_channels == 2 && s->input_channels == 1) {
 332         mono_to_stereo(output, buftmp3[0], nb_samples1);
 333     } else if (s->output_channels == 6 && s->input_channels == 2) {
 334         ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1);
 335     } else if (s->output_channels == s->input_channels && s->input_channels >= 2) {
 336         interleave(output, buftmp3, s->output_channels, nb_samples1);
 337     }
 338
 339     if (s->sample_fmt[1] != AV_SAMPLE_FMT_S16) {
 340         int istride[1] = { 2 };
 341         int ostride[1] = { s->sample_size[1] };
 342         const void *ibuf[1] = { output };
 343         void       *obuf[1] = { output_bak };
 344
 345         if (av_audio_convert(s->convert_ctx[1], obuf, ostride,
 346                              ibuf, istride, nb_samples1 * s->output_channels) < 0) {
 347             av_log(s->resample_context, AV_LOG_ERROR,
 348                    "Audio sample format convertion failed\n");
 349             return 0;
 350         }
 351     }
 352
 353     for (i = 0; i < s->filter_channels; i++) {
 354         av_free(bufin[i]);
 355         av_free(bufout[i]);
 356     }
 357
 358     return nb_samples1;
 359 }
 360
 361 void audio_resample_close(ReSampleContext *s)
 362 {
 363     int i;
 364     av_resample_close(s->resample_context);
 365     for (i = 0; i < s->filter_channels; i++)
 366         av_freep(&s->temp[i]);
 367     av_freep(&s->buffer[0]);
 368     av_freep(&s->buffer[1]);
 369     av_audio_convert_free(s->convert_ctx[0]);
 370     av_audio_convert_free(s->convert_ctx[1]);
 371     av_free(s);
 372 }