1 /* 2 * Copyright (C) 2012 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17 #include <unistd.h> 18 #include <stdio.h> 19 #include <stdlib.h> 20 #include <fcntl.h> 21 #include <string.h> 22 #include <sys/mman.h> 23 #include <sys/stat.h> 24 #include <errno.h> 25 #include <inttypes.h> 26 #include <time.h> 27 #include <math.h> 28 #include <audio_utils/primitives.h> 29 #include <audio_utils/sndfile.h> 30 #include <utils/Vector.h> 31 #include <media/AudioBufferProvider.h> 32 #include <media/AudioResampler.h> 33 34 using namespace android; 35 36 static bool gVerbose = false; 37 38 static int usage(const char* name) { 39 fprintf(stderr,"Usage: %s [-p] [-f] [-F] [-v] [-c channels]" 40 " [-q {dq|lq|mq|hq|vhq|dlq|dmq|dhq}]" 41 " [-i input-sample-rate] [-o output-sample-rate]" 42 " [-O csv] [-P csv] [<input-file>]" 43 " <output-file>\n", name); 44 fprintf(stderr," -p enable profiling\n"); 45 fprintf(stderr," -f enable filter profiling\n"); 46 fprintf(stderr," -F enable floating point -q {dlq|dmq|dhq} only"); 47 fprintf(stderr," -v verbose : log buffer provider calls\n"); 48 fprintf(stderr," -c # channels (1-2 for lq|mq|hq; 1-8 for dlq|dmq|dhq)\n"); 49 fprintf(stderr," -q resampler quality\n"); 50 fprintf(stderr," dq : default quality\n"); 51 fprintf(stderr," lq : low quality\n"); 52 fprintf(stderr," mq : medium quality\n"); 53 fprintf(stderr," hq : high quality\n"); 54 fprintf(stderr," vhq : very high quality\n"); 55 fprintf(stderr," dlq : dynamic low quality\n"); 56 fprintf(stderr," dmq : dynamic medium quality\n"); 57 fprintf(stderr," dhq : dynamic high quality\n"); 58 fprintf(stderr," -i input file sample rate (ignored if input file is specified)\n"); 59 fprintf(stderr," -o output file sample rate\n"); 60 fprintf(stderr," -O # frames output per call to resample() in CSV format\n"); 61 fprintf(stderr," -P # frames provided per call to resample() in CSV format\n"); 62 return -1; 63 } 64 65 // Convert a list of integers in CSV format to a Vector of those values. 66 // Returns the number of elements in the list, or -1 on error. 67 int parseCSV(const char *string, Vector<int>& values) 68 { 69 // pass 1: count the number of values and do syntax check 70 size_t numValues = 0; 71 bool hadDigit = false; 72 for (const char *p = string; ; ) { 73 switch (*p++) { 74 case '0': case '1': case '2': case '3': case '4': 75 case '5': case '6': case '7': case '8': case '9': 76 hadDigit = true; 77 break; 78 case '\0': 79 if (hadDigit) { 80 // pass 2: allocate and initialize vector of values 81 values.resize(++numValues); 82 values.editItemAt(0) = atoi(p = optarg); 83 for (size_t i = 1; i < numValues; ) { 84 if (*p++ == ',') { 85 values.editItemAt(i++) = atoi(p); 86 } 87 } 88 return numValues; 89 } 90 // fall through 91 case ',': 92 if (hadDigit) { 93 hadDigit = false; 94 numValues++; 95 break; 96 } 97 // fall through 98 default: 99 return -1; 100 } 101 } 102 } 103 104 int main(int argc, char* argv[]) { 105 const char* const progname = argv[0]; 106 bool profileResample = false; 107 bool profileFilter = false; 108 bool useFloat = false; 109 int channels = 1; 110 int input_freq = 0; 111 int output_freq = 0; 112 AudioResampler::src_quality quality = AudioResampler::DEFAULT_QUALITY; 113 Vector<int> Ovalues; 114 Vector<int> Pvalues; 115 116 int ch; 117 while ((ch = getopt(argc, argv, "pfFvc:q:i:o:O:P:")) != -1) { 118 switch (ch) { 119 case 'p': 120 profileResample = true; 121 break; 122 case 'f': 123 profileFilter = true; 124 break; 125 case 'F': 126 useFloat = true; 127 break; 128 case 'v': 129 gVerbose = true; 130 break; 131 case 'c': 132 channels = atoi(optarg); 133 break; 134 case 'q': 135 if (!strcmp(optarg, "dq")) 136 quality = AudioResampler::DEFAULT_QUALITY; 137 else if (!strcmp(optarg, "lq")) 138 quality = AudioResampler::LOW_QUALITY; 139 else if (!strcmp(optarg, "mq")) 140 quality = AudioResampler::MED_QUALITY; 141 else if (!strcmp(optarg, "hq")) 142 quality = AudioResampler::HIGH_QUALITY; 143 else if (!strcmp(optarg, "vhq")) 144 quality = AudioResampler::VERY_HIGH_QUALITY; 145 else if (!strcmp(optarg, "dlq")) 146 quality = AudioResampler::DYN_LOW_QUALITY; 147 else if (!strcmp(optarg, "dmq")) 148 quality = AudioResampler::DYN_MED_QUALITY; 149 else if (!strcmp(optarg, "dhq")) 150 quality = AudioResampler::DYN_HIGH_QUALITY; 151 else { 152 usage(progname); 153 return -1; 154 } 155 break; 156 case 'i': 157 input_freq = atoi(optarg); 158 break; 159 case 'o': 160 output_freq = atoi(optarg); 161 break; 162 case 'O': 163 if (parseCSV(optarg, Ovalues) < 0) { 164 fprintf(stderr, "incorrect syntax for -O option\n"); 165 return -1; 166 } 167 break; 168 case 'P': 169 if (parseCSV(optarg, Pvalues) < 0) { 170 fprintf(stderr, "incorrect syntax for -P option\n"); 171 return -1; 172 } 173 break; 174 case '?': 175 default: 176 usage(progname); 177 return -1; 178 } 179 } 180 181 if (channels < 1 182 || channels > (quality < AudioResampler::DYN_LOW_QUALITY ? 2 : 8)) { 183 fprintf(stderr, "invalid number of audio channels %d\n", channels); 184 return -1; 185 } 186 if (useFloat && quality < AudioResampler::DYN_LOW_QUALITY) { 187 fprintf(stderr, "float processing is only possible for dynamic resamplers\n"); 188 return -1; 189 } 190 191 argc -= optind; 192 argv += optind; 193 194 const char* file_in = NULL; 195 const char* file_out = NULL; 196 if (argc == 1) { 197 file_out = argv[0]; 198 } else if (argc == 2) { 199 file_in = argv[0]; 200 file_out = argv[1]; 201 } else { 202 usage(progname); 203 return -1; 204 } 205 206 // ---------------------------------------------------------- 207 208 size_t input_size; 209 void* input_vaddr; 210 if (argc == 2) { 211 SF_INFO info; 212 info.format = 0; 213 SNDFILE *sf = sf_open(file_in, SFM_READ, &info); 214 if (sf == NULL) { 215 perror(file_in); 216 return EXIT_FAILURE; 217 } 218 input_size = info.frames * info.channels * sizeof(short); 219 input_vaddr = malloc(input_size); 220 (void) sf_readf_short(sf, (short *) input_vaddr, info.frames); 221 sf_close(sf); 222 channels = info.channels; 223 input_freq = info.samplerate; 224 } else { 225 // data for testing is exactly (input sampling rate/1000)/2 seconds 226 // so 44.1khz input is 22.05 seconds 227 double k = 1000; // Hz / s 228 double time = (input_freq / 2) / k; 229 size_t input_frames = size_t(input_freq * time); 230 input_size = channels * sizeof(int16_t) * input_frames; 231 input_vaddr = malloc(input_size); 232 int16_t* in = (int16_t*)input_vaddr; 233 for (size_t i=0 ; i<input_frames ; i++) { 234 double t = double(i) / input_freq; 235 double y = sin(M_PI * k * t * t); 236 int16_t yi = floor(y * 32767.0 + 0.5); 237 for (int j = 0; j < channels; j++) { 238 in[i*channels + j] = yi / (1 + j); 239 } 240 } 241 } 242 size_t input_framesize = channels * sizeof(int16_t); 243 size_t input_frames = input_size / input_framesize; 244 245 // For float processing, convert input int16_t to float array 246 if (useFloat) { 247 void *new_vaddr; 248 249 input_framesize = channels * sizeof(float); 250 input_size = input_frames * input_framesize; 251 new_vaddr = malloc(input_size); 252 memcpy_to_float_from_i16(reinterpret_cast<float*>(new_vaddr), 253 reinterpret_cast<int16_t*>(input_vaddr), input_frames * channels); 254 free(input_vaddr); 255 input_vaddr = new_vaddr; 256 } 257 258 // ---------------------------------------------------------- 259 260 class Provider: public AudioBufferProvider { 261 const void* mAddr; // base address 262 const size_t mNumFrames; // total frames 263 const size_t mFrameSize; // size of each frame in bytes 264 size_t mNextFrame; // index of next frame to provide 265 size_t mUnrel; // number of frames not yet released 266 const Vector<int> mPvalues; // number of frames provided per call 267 size_t mNextPidx; // index of next entry in mPvalues to use 268 public: 269 Provider(const void* addr, size_t frames, size_t frameSize, const Vector<int>& Pvalues) 270 : mAddr(addr), 271 mNumFrames(frames), 272 mFrameSize(frameSize), 273 mNextFrame(0), mUnrel(0), mPvalues(Pvalues), mNextPidx(0) { 274 } 275 virtual status_t getNextBuffer(Buffer* buffer) { 276 size_t requestedFrames = buffer->frameCount; 277 if (requestedFrames > mNumFrames - mNextFrame) { 278 buffer->frameCount = mNumFrames - mNextFrame; 279 } 280 if (!mPvalues.isEmpty()) { 281 size_t provided = mPvalues[mNextPidx++]; 282 printf("mPvalue[%zu]=%zu not %zu\n", mNextPidx-1, provided, buffer->frameCount); 283 if (provided < buffer->frameCount) { 284 buffer->frameCount = provided; 285 } 286 if (mNextPidx >= mPvalues.size()) { 287 mNextPidx = 0; 288 } 289 } 290 if (gVerbose) { 291 printf("getNextBuffer() requested %zu frames out of %zu frames available," 292 " and returned %zu frames\n", 293 requestedFrames, (size_t) (mNumFrames - mNextFrame), buffer->frameCount); 294 } 295 mUnrel = buffer->frameCount; 296 if (buffer->frameCount > 0) { 297 buffer->raw = (char *)mAddr + mFrameSize * mNextFrame; 298 return NO_ERROR; 299 } else { 300 buffer->raw = NULL; 301 return NOT_ENOUGH_DATA; 302 } 303 } 304 virtual void releaseBuffer(Buffer* buffer) { 305 if (buffer->frameCount > mUnrel) { 306 fprintf(stderr, "ERROR releaseBuffer() released %zu frames but only %zu available " 307 "to release\n", buffer->frameCount, mUnrel); 308 mNextFrame += mUnrel; 309 mUnrel = 0; 310 } else { 311 if (gVerbose) { 312 printf("releaseBuffer() released %zu frames out of %zu frames available " 313 "to release\n", buffer->frameCount, mUnrel); 314 } 315 mNextFrame += buffer->frameCount; 316 mUnrel -= buffer->frameCount; 317 } 318 buffer->frameCount = 0; 319 buffer->raw = NULL; 320 } 321 void reset() { 322 mNextFrame = 0; 323 } 324 } provider(input_vaddr, input_frames, input_framesize, Pvalues); 325 326 if (gVerbose) { 327 printf("%zu input frames\n", input_frames); 328 } 329 330 audio_format_t format = useFloat ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT; 331 int output_channels = channels > 2 ? channels : 2; // output is at least stereo samples 332 size_t output_framesize = output_channels * (useFloat ? sizeof(float) : sizeof(int32_t)); 333 size_t output_frames = ((int64_t) input_frames * output_freq) / input_freq; 334 size_t output_size = output_frames * output_framesize; 335 336 if (profileFilter) { 337 // Check how fast sample rate changes are that require filter changes. 338 // The delta sample rate changes must indicate a downsampling ratio, 339 // and must be larger than 10% changes. 340 // 341 // On fast devices, filters should be generated between 0.1ms - 1ms. 342 // (single threaded). 343 AudioResampler* resampler = AudioResampler::create(format, channels, 344 8000, quality); 345 int looplimit = 100; 346 timespec start, end; 347 clock_gettime(CLOCK_MONOTONIC, &start); 348 for (int i = 0; i < looplimit; ++i) { 349 resampler->setSampleRate(9000); 350 resampler->setSampleRate(12000); 351 resampler->setSampleRate(20000); 352 resampler->setSampleRate(30000); 353 } 354 clock_gettime(CLOCK_MONOTONIC, &end); 355 int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec; 356 int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec; 357 int64_t time = end_ns - start_ns; 358 printf("%.2f sample rate changes with filter calculation/sec\n", 359 looplimit * 4 / (time / 1e9)); 360 361 // Check how fast sample rate changes are without filter changes. 362 // This should be very fast, probably 0.1us - 1us per sample rate 363 // change. 364 resampler->setSampleRate(1000); 365 looplimit = 1000; 366 clock_gettime(CLOCK_MONOTONIC, &start); 367 for (int i = 0; i < looplimit; ++i) { 368 resampler->setSampleRate(1000+i); 369 } 370 clock_gettime(CLOCK_MONOTONIC, &end); 371 start_ns = start.tv_sec * 1000000000LL + start.tv_nsec; 372 end_ns = end.tv_sec * 1000000000LL + end.tv_nsec; 373 time = end_ns - start_ns; 374 printf("%.2f sample rate changes without filter calculation/sec\n", 375 looplimit / (time / 1e9)); 376 resampler->reset(); 377 delete resampler; 378 } 379 380 void* output_vaddr = malloc(output_size); 381 AudioResampler* resampler = AudioResampler::create(format, channels, 382 output_freq, quality); 383 384 resampler->setSampleRate(input_freq); 385 resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT); 386 387 if (profileResample) { 388 /* 389 * For profiling on mobile devices, upon experimentation 390 * it is better to run a few trials with a shorter loop limit, 391 * and take the minimum time. 392 * 393 * Long tests can cause CPU temperature to build up and thermal throttling 394 * to reduce CPU frequency. 395 * 396 * For frequency checks (index=0, or 1, etc.): 397 * "cat /sys/devices/system/cpu/cpu${index}/cpufreq/scaling_*_freq" 398 * 399 * For temperature checks (index=0, or 1, etc.): 400 * "cat /sys/class/thermal/thermal_zone${index}/temp" 401 * 402 * Another way to avoid thermal throttling is to fix the CPU frequency 403 * at a lower level which prevents excessive temperatures. 404 */ 405 const int trials = 4; 406 const int looplimit = 4; 407 timespec start, end; 408 int64_t time = 0; 409 410 for (int n = 0; n < trials; ++n) { 411 clock_gettime(CLOCK_MONOTONIC, &start); 412 for (int i = 0; i < looplimit; ++i) { 413 resampler->resample((int*) output_vaddr, output_frames, &provider); 414 provider.reset(); // during benchmarking reset only the provider 415 } 416 clock_gettime(CLOCK_MONOTONIC, &end); 417 int64_t start_ns = start.tv_sec * 1000000000LL + start.tv_nsec; 418 int64_t end_ns = end.tv_sec * 1000000000LL + end.tv_nsec; 419 int64_t diff_ns = end_ns - start_ns; 420 if (n == 0 || diff_ns < time) { 421 time = diff_ns; // save the best out of our trials. 422 } 423 } 424 // Mfrms/s is "Millions of output frames per second". 425 printf("quality: %d channels: %d msec: %" PRId64 " Mfrms/s: %.2lf\n", 426 quality, channels, time/1000000, output_frames * looplimit / (time / 1e9) / 1e6); 427 resampler->reset(); 428 429 // TODO fix legacy bug: reset does not clear buffers. 430 // delete and recreate resampler here. 431 delete resampler; 432 resampler = AudioResampler::create(format, channels, 433 output_freq, quality); 434 resampler->setSampleRate(input_freq); 435 resampler->setVolume(AudioResampler::UNITY_GAIN_FLOAT, AudioResampler::UNITY_GAIN_FLOAT); 436 } 437 438 memset(output_vaddr, 0, output_size); 439 if (gVerbose) { 440 printf("resample() %zu output frames\n", output_frames); 441 } 442 if (Ovalues.isEmpty()) { 443 Ovalues.push(output_frames); 444 } 445 for (size_t i = 0, j = 0; i < output_frames; ) { 446 size_t thisFrames = Ovalues[j++]; 447 if (j >= Ovalues.size()) { 448 j = 0; 449 } 450 if (thisFrames == 0 || thisFrames > output_frames - i) { 451 thisFrames = output_frames - i; 452 } 453 resampler->resample((int*) output_vaddr + output_channels*i, thisFrames, &provider); 454 i += thisFrames; 455 } 456 if (gVerbose) { 457 printf("resample() complete\n"); 458 } 459 resampler->reset(); 460 if (gVerbose) { 461 printf("reset() complete\n"); 462 } 463 delete resampler; 464 resampler = NULL; 465 466 // For float processing, convert output format from float to Q4.27, 467 // which is then converted to int16_t for final storage. 468 if (useFloat) { 469 memcpy_to_q4_27_from_float(reinterpret_cast<int32_t*>(output_vaddr), 470 reinterpret_cast<float*>(output_vaddr), output_frames * output_channels); 471 } 472 473 // mono takes left channel only (out of stereo output pair) 474 // stereo and multichannel preserve all channels. 475 int32_t* out = (int32_t*) output_vaddr; 476 int16_t* convert = (int16_t*) malloc(output_frames * channels * sizeof(int16_t)); 477 478 const int volumeShift = 12; // shift requirement for Q4.27 to Q.15 479 // round to half towards zero and saturate at int16 (non-dithered) 480 const int roundVal = (1<<(volumeShift-1)) - 1; // volumePrecision > 0 481 482 for (size_t i = 0; i < output_frames; i++) { 483 for (int j = 0; j < channels; j++) { 484 int32_t s = out[i * output_channels + j] + roundVal; // add offset here 485 if (s < 0) { 486 s = (s + 1) >> volumeShift; // round to 0 487 if (s < -32768) { 488 s = -32768; 489 } 490 } else { 491 s = s >> volumeShift; 492 if (s > 32767) { 493 s = 32767; 494 } 495 } 496 convert[i * channels + j] = int16_t(s); 497 } 498 } 499 500 // write output to disk 501 SF_INFO info; 502 info.frames = 0; 503 info.samplerate = output_freq; 504 info.channels = channels; 505 info.format = SF_FORMAT_WAV | SF_FORMAT_PCM_16; 506 SNDFILE *sf = sf_open(file_out, SFM_WRITE, &info); 507 if (sf == NULL) { 508 perror(file_out); 509 return EXIT_FAILURE; 510 } 511 (void) sf_writef_short(sf, convert, output_frames); 512 sf_close(sf); 513 514 return EXIT_SUCCESS; 515 } 516
