1/* 2 * Copyright (c) 2002 Fabrice Bellard 3 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> 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#include <stdint.h> 23#include <stdio.h> 24 25#include "libavutil/avstring.h" 26#include "libavutil/common.h" 27#include "libavutil/lfg.h" 28#include "libavutil/libm.h" 29#include "libavutil/log.h" 30#include "libavutil/mem.h" 31#include "libavutil/opt.h" 32#include "libavutil/samplefmt.h" 33#include "avresample.h" 34 35static double dbl_rand(AVLFG *lfg) 36{ 37 return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0; 38} 39 40#define PUT_FUNC(name, fmt, type, expr) \ 41static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\ 42 int channels, int sample, int ch, \ 43 double v_dbl) \ 44{ \ 45 type v = expr; \ 46 type **out = (type **)data; \ 47 if (av_sample_fmt_is_planar(sample_fmt)) \ 48 out[ch][sample] = v; \ 49 else \ 50 out[0][sample * channels + ch] = v; \ 51} 52 53PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128)) 54PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15)))) 55PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31)))) 56PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl) 57PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl) 58 59static void put_sample(void **data, enum AVSampleFormat sample_fmt, 60 int channels, int sample, int ch, double v_dbl) 61{ 62 switch (av_get_packed_sample_fmt(sample_fmt)) { 63 case AV_SAMPLE_FMT_U8: 64 put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl); 65 break; 66 case AV_SAMPLE_FMT_S16: 67 put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl); 68 break; 69 case AV_SAMPLE_FMT_S32: 70 put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl); 71 break; 72 case AV_SAMPLE_FMT_FLT: 73 put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl); 74 break; 75 case AV_SAMPLE_FMT_DBL: 76 put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl); 77 break; 78 } 79} 80 81static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt, 82 int channels, int sample_rate, int nb_samples) 83{ 84 int i, ch, k; 85 double v, f, a, ampa; 86 double tabf1[AVRESAMPLE_MAX_CHANNELS]; 87 double tabf2[AVRESAMPLE_MAX_CHANNELS]; 88 double taba[AVRESAMPLE_MAX_CHANNELS]; 89 90#define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v); 91 92 k = 0; 93 94 /* 1 second of single freq sine at 1000 Hz */ 95 a = 0; 96 for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { 97 v = sin(a) * 0.30; 98 for (ch = 0; ch < channels; ch++) 99 PUT_SAMPLE 100 a += M_PI * 1000.0 * 2.0 / sample_rate; 101 } 102 103 /* 1 second of varying frequency between 100 and 10000 Hz */ 104 a = 0; 105 for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { 106 v = sin(a) * 0.30; 107 for (ch = 0; ch < channels; ch++) 108 PUT_SAMPLE 109 f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate); 110 a += M_PI * f * 2.0 / sample_rate; 111 } 112 113 /* 0.5 second of low amplitude white noise */ 114 for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) { 115 v = dbl_rand(rnd) * 0.30; 116 for (ch = 0; ch < channels; ch++) 117 PUT_SAMPLE 118 } 119 120 /* 0.5 second of high amplitude white noise */ 121 for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) { 122 v = dbl_rand(rnd); 123 for (ch = 0; ch < channels; ch++) 124 PUT_SAMPLE 125 } 126 127 /* 1 second of unrelated ramps for each channel */ 128 for (ch = 0; ch < channels; ch++) { 129 taba[ch] = 0; 130 tabf1[ch] = 100 + av_lfg_get(rnd) % 5000; 131 tabf2[ch] = 100 + av_lfg_get(rnd) % 5000; 132 } 133 for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) { 134 for (ch = 0; ch < channels; ch++) { 135 v = sin(taba[ch]) * 0.30; 136 PUT_SAMPLE 137 f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate); 138 taba[ch] += M_PI * f * 2.0 / sample_rate; 139 } 140 } 141 142 /* 2 seconds of 500 Hz with varying volume */ 143 a = 0; 144 ampa = 0; 145 for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) { 146 for (ch = 0; ch < channels; ch++) { 147 double amp = (1.0 + sin(ampa)) * 0.15; 148 if (ch & 1) 149 amp = 0.30 - amp; 150 v = sin(a) * amp; 151 PUT_SAMPLE 152 a += M_PI * 500.0 * 2.0 / sample_rate; 153 ampa += M_PI * 2.0 / sample_rate; 154 } 155 } 156} 157 158/* formats, rates, and layouts are ordered for priority in testing. 159 e.g. 'avresample-test 4 2 2' will test all input/output combinations of 160 S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */ 161 162static const enum AVSampleFormat formats[] = { 163 AV_SAMPLE_FMT_S16, 164 AV_SAMPLE_FMT_FLTP, 165 AV_SAMPLE_FMT_S16P, 166 AV_SAMPLE_FMT_FLT, 167 AV_SAMPLE_FMT_S32P, 168 AV_SAMPLE_FMT_S32, 169 AV_SAMPLE_FMT_U8P, 170 AV_SAMPLE_FMT_U8, 171 AV_SAMPLE_FMT_DBLP, 172 AV_SAMPLE_FMT_DBL, 173}; 174 175static const int rates[] = { 176 48000, 177 44100, 178 16000 179}; 180 181static const uint64_t layouts[] = { 182 AV_CH_LAYOUT_STEREO, 183 AV_CH_LAYOUT_MONO, 184 AV_CH_LAYOUT_5POINT1, 185 AV_CH_LAYOUT_7POINT1, 186}; 187 188int main(int argc, char **argv) 189{ 190 AVAudioResampleContext *s; 191 AVLFG rnd; 192 int ret = 0; 193 uint8_t *in_buf = NULL; 194 uint8_t *out_buf = NULL; 195 unsigned int in_buf_size; 196 unsigned int out_buf_size; 197 uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 }; 198 uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 }; 199 int in_linesize; 200 int out_linesize; 201 uint64_t in_ch_layout; 202 int in_channels; 203 enum AVSampleFormat in_fmt; 204 int in_rate; 205 uint64_t out_ch_layout; 206 int out_channels; 207 enum AVSampleFormat out_fmt; 208 int out_rate; 209 int num_formats, num_rates, num_layouts; 210 int i, j, k, l, m, n; 211 212 num_formats = 2; 213 num_rates = 2; 214 num_layouts = 2; 215 if (argc > 1) { 216 if (!av_strncasecmp(argv[1], "-h", 3)) { 217 av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> " 218 "[<num sample rates> [<num channel layouts>]]]\n" 219 "Default is 2 2 2\n"); 220 return 0; 221 } 222 num_formats = strtol(argv[1], NULL, 0); 223 num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats)); 224 } 225 if (argc > 2) { 226 num_rates = strtol(argv[2], NULL, 0); 227 num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates)); 228 } 229 if (argc > 3) { 230 num_layouts = strtol(argv[3], NULL, 0); 231 num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts)); 232 } 233 234 av_log_set_level(AV_LOG_DEBUG); 235 236 av_lfg_init(&rnd, 0xC0FFEE); 237 238 in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6, 239 AV_SAMPLE_FMT_DBLP, 0); 240 out_buf_size = in_buf_size; 241 242 in_buf = av_malloc(in_buf_size); 243 if (!in_buf) 244 goto end; 245 out_buf = av_malloc(out_buf_size); 246 if (!out_buf) 247 goto end; 248 249 s = avresample_alloc_context(); 250 if (!s) { 251 av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n"); 252 ret = 1; 253 goto end; 254 } 255 256 for (i = 0; i < num_formats; i++) { 257 in_fmt = formats[i]; 258 for (k = 0; k < num_layouts; k++) { 259 in_ch_layout = layouts[k]; 260 in_channels = av_get_channel_layout_nb_channels(in_ch_layout); 261 for (m = 0; m < num_rates; m++) { 262 in_rate = rates[m]; 263 264 ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf, 265 in_channels, in_rate * 6, 266 in_fmt, 0); 267 if (ret < 0) { 268 av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n"); 269 goto end; 270 } 271 audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6); 272 273 for (j = 0; j < num_formats; j++) { 274 out_fmt = formats[j]; 275 for (l = 0; l < num_layouts; l++) { 276 out_ch_layout = layouts[l]; 277 out_channels = av_get_channel_layout_nb_channels(out_ch_layout); 278 for (n = 0; n < num_rates; n++) { 279 out_rate = rates[n]; 280 281 av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n", 282 av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt), 283 in_channels, out_channels, in_rate, out_rate); 284 285 ret = av_samples_fill_arrays(out_data, &out_linesize, 286 out_buf, out_channels, 287 out_rate * 6, out_fmt, 0); 288 if (ret < 0) { 289 av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n"); 290 goto end; 291 } 292 293 av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0); 294 av_opt_set_int(s, "in_sample_fmt", in_fmt, 0); 295 av_opt_set_int(s, "in_sample_rate", in_rate, 0); 296 av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0); 297 av_opt_set_int(s, "out_sample_fmt", out_fmt, 0); 298 av_opt_set_int(s, "out_sample_rate", out_rate, 0); 299 300 av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0); 301 302 ret = avresample_open(s); 303 if (ret < 0) { 304 av_log(s, AV_LOG_ERROR, "Error opening context\n"); 305 goto end; 306 } 307 308 ret = avresample_convert(s, out_data, out_linesize, out_rate * 6, 309 in_data, in_linesize, in_rate * 6); 310 if (ret < 0) { 311 char errbuf[256]; 312 av_strerror(ret, errbuf, sizeof(errbuf)); 313 av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf); 314 goto end; 315 } 316 av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n", 317 in_rate * 6, ret); 318 if (avresample_get_delay(s) > 0) 319 av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n", 320 avresample_get_delay(s)); 321 if (avresample_available(s) > 0) 322 av_log(NULL, AV_LOG_INFO, "%d samples available for output\n", 323 avresample_available(s)); 324 av_log(NULL, AV_LOG_INFO, "\n"); 325 326 avresample_close(s); 327 } 328 } 329 } 330 } 331 } 332 } 333 334 ret = 0; 335 336end: 337 av_freep(&in_buf); 338 av_freep(&out_buf); 339 avresample_free(&s); 340 return ret; 341} 342