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 31struct AVResampleContext; 32 33static const char *context_to_name(void *ptr) 34{ 35 return "audioresample"; 36} 37 38static const AVOption options[] = {{NULL}}; 39static const AVClass audioresample_context_class = { "ReSampleContext", context_to_name, options }; 40 41struct ReSampleContext { 42 const AVClass *av_class; 43 struct AVResampleContext *resample_context; 44 short *temp[2]; 45 int temp_len; 46 float ratio; 47 /* channel convert */ 48 int input_channels, output_channels, filter_channels; 49 AVAudioConvert *convert_ctx[2]; 50 enum SampleFormat sample_fmt[2]; ///< input and output sample format 51 unsigned sample_size[2]; ///< size of one sample in sample_fmt 52 short *buffer[2]; ///< buffers used for conversion to S16 53 unsigned buffer_size[2]; ///< sizes of allocated buffers 54}; 55 56/* n1: number of samples */ 57static void stereo_to_mono(short *output, short *input, int n1) 58{ 59 short *p, *q; 60 int n = n1; 61 62 p = input; 63 q = output; 64 while (n >= 4) { 65 q[0] = (p[0] + p[1]) >> 1; 66 q[1] = (p[2] + p[3]) >> 1; 67 q[2] = (p[4] + p[5]) >> 1; 68 q[3] = (p[6] + p[7]) >> 1; 69 q += 4; 70 p += 8; 71 n -= 4; 72 } 73 while (n > 0) { 74 q[0] = (p[0] + p[1]) >> 1; 75 q++; 76 p += 2; 77 n--; 78 } 79} 80 81/* n1: number of samples */ 82static void mono_to_stereo(short *output, short *input, int n1) 83{ 84 short *p, *q; 85 int n = n1; 86 int v; 87 88 p = input; 89 q = output; 90 while (n >= 4) { 91 v = p[0]; q[0] = v; q[1] = v; 92 v = p[1]; q[2] = v; q[3] = v; 93 v = p[2]; q[4] = v; q[5] = v; 94 v = p[3]; q[6] = v; q[7] = v; 95 q += 8; 96 p += 4; 97 n -= 4; 98 } 99 while (n > 0) { 100 v = p[0]; q[0] = v; q[1] = v; 101 q += 2; 102 p += 1; 103 n--; 104 } 105} 106 107/* XXX: should use more abstract 'N' channels system */ 108static void stereo_split(short *output1, short *output2, short *input, int n) 109{ 110 int i; 111 112 for(i=0;i<n;i++) { 113 *output1++ = *input++; 114 *output2++ = *input++; 115 } 116} 117 118static void stereo_mux(short *output, short *input1, short *input2, int n) 119{ 120 int i; 121 122 for(i=0;i<n;i++) { 123 *output++ = *input1++; 124 *output++ = *input2++; 125 } 126} 127 128static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) 129{ 130 int i; 131 short l,r; 132 133 for(i=0;i<n;i++) { 134 l=*input1++; 135 r=*input2++; 136 *output++ = l; /* left */ 137 *output++ = (l/2)+(r/2); /* center */ 138 *output++ = r; /* right */ 139 *output++ = 0; /* left surround */ 140 *output++ = 0; /* right surroud */ 141 *output++ = 0; /* low freq */ 142 } 143} 144 145ReSampleContext *av_audio_resample_init(int output_channels, int input_channels, 146 int output_rate, int input_rate, 147 enum SampleFormat sample_fmt_out, 148 enum SampleFormat sample_fmt_in, 149 int filter_length, int log2_phase_count, 150 int linear, double cutoff) 151{ 152 ReSampleContext *s; 153 154 if ( input_channels > 2) 155 { 156 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported.\n"); 157 return NULL; 158 } 159 160 s = av_mallocz(sizeof(ReSampleContext)); 161 if (!s) 162 { 163 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context.\n"); 164 return NULL; 165 } 166 167 s->ratio = (float)output_rate / (float)input_rate; 168 169 s->input_channels = input_channels; 170 s->output_channels = output_channels; 171 172 s->filter_channels = s->input_channels; 173 if (s->output_channels < s->filter_channels) 174 s->filter_channels = s->output_channels; 175 176 s->sample_fmt [0] = sample_fmt_in; 177 s->sample_fmt [1] = sample_fmt_out; 178 s->sample_size[0] = av_get_bits_per_sample_format(s->sample_fmt[0])>>3; 179 s->sample_size[1] = av_get_bits_per_sample_format(s->sample_fmt[1])>>3; 180 181 if (s->sample_fmt[0] != SAMPLE_FMT_S16) { 182 if (!(s->convert_ctx[0] = av_audio_convert_alloc(SAMPLE_FMT_S16, 1, 183 s->sample_fmt[0], 1, NULL, 0))) { 184 av_log(s, AV_LOG_ERROR, 185 "Cannot convert %s sample format to s16 sample format\n", 186 avcodec_get_sample_fmt_name(s->sample_fmt[0])); 187 av_free(s); 188 return NULL; 189 } 190 } 191 192 if (s->sample_fmt[1] != SAMPLE_FMT_S16) { 193 if (!(s->convert_ctx[1] = av_audio_convert_alloc(s->sample_fmt[1], 1, 194 SAMPLE_FMT_S16, 1, NULL, 0))) { 195 av_log(s, AV_LOG_ERROR, 196 "Cannot convert s16 sample format to %s sample format\n", 197 avcodec_get_sample_fmt_name(s->sample_fmt[1])); 198 av_audio_convert_free(s->convert_ctx[0]); 199 av_free(s); 200 return NULL; 201 } 202 } 203 204/* 205 * AC-3 output is the only case where filter_channels could be greater than 2. 206 * input channels can't be greater than 2, so resample the 2 channels and then 207 * expand to 6 channels after the resampling. 208 */ 209 if(s->filter_channels>2) 210 s->filter_channels = 2; 211 212#define TAPS 16 213 s->resample_context= av_resample_init(output_rate, input_rate, 214 filter_length, log2_phase_count, linear, cutoff); 215 216 s->av_class= &audioresample_context_class; 217 218 return s; 219} 220 221#if LIBAVCODEC_VERSION_MAJOR < 53 222ReSampleContext *audio_resample_init(int output_channels, int input_channels, 223 int output_rate, int input_rate) 224{ 225 return av_audio_resample_init(output_channels, input_channels, 226 output_rate, input_rate, 227 SAMPLE_FMT_S16, SAMPLE_FMT_S16, 228 TAPS, 10, 0, 0.8); 229} 230#endif 231 232/* resample audio. 'nb_samples' is the number of input samples */ 233/* XXX: optimize it ! */ 234int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) 235{ 236 int i, nb_samples1; 237 short *bufin[2]; 238 short *bufout[2]; 239 short *buftmp2[2], *buftmp3[2]; 240 short *output_bak = NULL; 241 int lenout; 242 243 if (s->input_channels == s->output_channels && s->ratio == 1.0 && 0) { 244 /* nothing to do */ 245 memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); 246 return nb_samples; 247 } 248 249 if (s->sample_fmt[0] != SAMPLE_FMT_S16) { 250 int istride[1] = { s->sample_size[0] }; 251 int ostride[1] = { 2 }; 252 const void *ibuf[1] = { input }; 253 void *obuf[1]; 254 unsigned input_size = nb_samples*s->input_channels*2; 255 256 if (!s->buffer_size[0] || s->buffer_size[0] < input_size) { 257 av_free(s->buffer[0]); 258 s->buffer_size[0] = input_size; 259 s->buffer[0] = av_malloc(s->buffer_size[0]); 260 if (!s->buffer[0]) { 261 av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n"); 262 return 0; 263 } 264 } 265 266 obuf[0] = s->buffer[0]; 267 268 if (av_audio_convert(s->convert_ctx[0], obuf, ostride, 269 ibuf, istride, nb_samples*s->input_channels) < 0) { 270 av_log(s, AV_LOG_ERROR, "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] != SAMPLE_FMT_S16) { 280 output_bak = output; 281 282 if (!s->buffer_size[1] || s->buffer_size[1] < lenout) { 283 av_free(s->buffer[1]); 284 s->buffer_size[1] = lenout; 285 s->buffer[1] = av_malloc(s->buffer_size[1]); 286 if (!s->buffer[1]) { 287 av_log(s, AV_LOG_ERROR, "Could not allocate buffer\n"); 288 return 0; 289 } 290 } 291 292 output = s->buffer[1]; 293 } 294 295 /* XXX: move those malloc to resample init code */ 296 for(i=0; i<s->filter_channels; i++){ 297 bufin[i]= av_malloc( (nb_samples + s->temp_len) * sizeof(short) ); 298 memcpy(bufin[i], s->temp[i], s->temp_len * sizeof(short)); 299 buftmp2[i] = bufin[i] + s->temp_len; 300 } 301 302 /* make some zoom to avoid round pb */ 303 bufout[0]= av_malloc( lenout * sizeof(short) ); 304 bufout[1]= av_malloc( lenout * sizeof(short) ); 305 306 if (s->input_channels == 2 && 307 s->output_channels == 1) { 308 buftmp3[0] = output; 309 stereo_to_mono(buftmp2[0], input, nb_samples); 310 } else if (s->output_channels >= 2 && s->input_channels == 1) { 311 buftmp3[0] = bufout[0]; 312 memcpy(buftmp2[0], input, nb_samples*sizeof(short)); 313 } else if (s->output_channels >= 2) { 314 buftmp3[0] = bufout[0]; 315 buftmp3[1] = bufout[1]; 316 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); 317 } else { 318 buftmp3[0] = output; 319 memcpy(buftmp2[0], input, nb_samples*sizeof(short)); 320 } 321 322 nb_samples += s->temp_len; 323 324 /* resample each channel */ 325 nb_samples1 = 0; /* avoid warning */ 326 for(i=0;i<s->filter_channels;i++) { 327 int consumed; 328 int is_last= i+1 == s->filter_channels; 329 330 nb_samples1 = av_resample(s->resample_context, buftmp3[i], bufin[i], &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 == 2) { 339 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); 340 } else if (s->output_channels == 6) { 341 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); 342 } 343 344 if (s->sample_fmt[1] != 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, AV_LOG_ERROR, "Audio sample format convertion failed\n"); 353 return 0; 354 } 355 } 356 357 for(i=0; i<s->filter_channels; i++) 358 av_free(bufin[i]); 359 360 av_free(bufout[0]); 361 av_free(bufout[1]); 362 return nb_samples1; 363} 364 365void audio_resample_close(ReSampleContext *s) 366{ 367 av_resample_close(s->resample_context); 368 av_freep(&s->temp[0]); 369 av_freep(&s->temp[1]); 370 av_freep(&s->buffer[0]); 371 av_freep(&s->buffer[1]); 372 av_audio_convert_free(s->convert_ctx[0]); 373 av_audio_convert_free(s->convert_ctx[1]); 374 av_free(s); 375} 376