annotate data/fileio/test/AudioFileReaderTest.h @ 1686:e73baeead27f single-point

Model deletion issues - the AlignmentModel doesn't actually need to know about the input aggregate model; the document should own it instead
author Chris Cannam
date Thu, 04 Apr 2019 11:15:43 +0100
parents 6d9881e59cc2
children dbd13eb7dad1
rev   line source
Chris@756 1 /* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
Chris@756 2
Chris@756 3 /*
Chris@756 4 Sonic Visualiser
Chris@756 5 An audio file viewer and annotation editor.
Chris@756 6 Centre for Digital Music, Queen Mary, University of London.
Chris@756 7 This file copyright 2013 Chris Cannam.
Chris@756 8
Chris@756 9 This program is free software; you can redistribute it and/or
Chris@756 10 modify it under the terms of the GNU General Public License as
Chris@756 11 published by the Free Software Foundation; either version 2 of the
Chris@756 12 License, or (at your option) any later version. See the file
Chris@756 13 COPYING included with this distribution for more information.
Chris@756 14 */
Chris@756 15
Chris@756 16 #ifndef TEST_AUDIO_FILE_READER_H
Chris@756 17 #define TEST_AUDIO_FILE_READER_H
Chris@756 18
Chris@756 19 #include "../AudioFileReaderFactory.h"
Chris@756 20 #include "../AudioFileReader.h"
Chris@1313 21 #include "../WavFileWriter.h"
Chris@756 22
Chris@756 23 #include "AudioTestData.h"
Chris@756 24
Chris@756 25 #include <cmath>
Chris@756 26
Chris@756 27 #include <QObject>
Chris@756 28 #include <QtTest>
Chris@756 29 #include <QDir>
Chris@756 30
Chris@756 31 #include <iostream>
Chris@756 32
Chris@756 33 using namespace std;
Chris@756 34
Chris@756 35 class AudioFileReaderTest : public QObject
Chris@756 36 {
Chris@756 37 Q_OBJECT
Chris@756 38
Chris@1346 39 private:
Chris@1346 40 QString testDirBase;
Chris@1346 41 QString audioDir;
Chris@1346 42 QString diffDir;
Chris@1346 43
Chris@1346 44 public:
Chris@1346 45 AudioFileReaderTest(QString base) {
Chris@1346 46 if (base == "") {
Chris@1346 47 base = "svcore/data/fileio/test";
Chris@1346 48 }
Chris@1346 49 testDirBase = base;
Chris@1359 50 audioDir = base + "/audio";
Chris@1346 51 diffDir = base + "/diffs";
Chris@1346 52 }
Chris@1346 53
Chris@1346 54 private:
Chris@756 55 const char *strOf(QString s) {
Chris@756 56 return strdup(s.toLocal8Bit().data());
Chris@756 57 }
Chris@756 58
Chris@1313 59 void getFileMetadata(QString filename,
Chris@1313 60 QString &extension,
Chris@1313 61 sv_samplerate_t &rate,
Chris@1313 62 int &channels,
Chris@1313 63 int &bitdepth) {
Chris@1313 64
Chris@1313 65 QStringList fileAndExt = filename.split(".");
Chris@1313 66 QStringList bits = fileAndExt[0].split("-");
Chris@1313 67
Chris@1313 68 extension = fileAndExt[1];
Chris@1313 69 rate = bits[0].toInt();
Chris@1313 70 channels = bits[1].toInt();
Chris@1313 71 bitdepth = 16;
Chris@1313 72 if (bits.length() > 2) {
Chris@1313 73 bitdepth = bits[2].toInt();
Chris@1313 74 }
Chris@1313 75 }
Chris@1313 76
cannam@1315 77 void getExpectedThresholds(QString format,
cannam@1315 78 QString filename,
Chris@1313 79 bool resampled,
Chris@1313 80 bool gapless,
Chris@1313 81 bool normalised,
Chris@1313 82 double &maxLimit,
Chris@1313 83 double &rmsLimit) {
Chris@1313 84
Chris@1313 85 QString extension;
Chris@1313 86 sv_samplerate_t fileRate;
Chris@1313 87 int channels;
Chris@1313 88 int bitdepth;
Chris@1313 89 getFileMetadata(filename, extension, fileRate, channels, bitdepth);
Chris@1313 90
Chris@1313 91 if (normalised) {
Chris@1313 92
cannam@1315 93 if (format == "ogg") {
Chris@1313 94
Chris@1313 95 // Our ogg is not especially high quality and is
Chris@1313 96 // actually further from the original if normalised
Chris@1313 97
Chris@1313 98 maxLimit = 0.1;
Chris@1313 99 rmsLimit = 0.03;
Chris@1313 100
Chris@1598 101 } else if (format == "opus") {
Chris@1598 102
Chris@1598 103 maxLimit = 0.06;
Chris@1598 104 rmsLimit = 0.015;
Chris@1598 105
cannam@1315 106 } else if (format == "aac") {
Chris@1313 107
cannam@1315 108 // Terrible performance for this test, load of spill
cannam@1315 109 // from one channel to the other. I guess they know
cannam@1315 110 // what they're doing, it's perceptual after all, but
cannam@1315 111 // it does make this check a bit superfluous, you
cannam@1315 112 // could probably pass it with a signal that sounds
cannam@1315 113 // nothing like the original
cannam@1315 114 maxLimit = 0.2;
cannam@1314 115 rmsLimit = 0.1;
Chris@1313 116
Chris@1603 117 } else if (format == "wma") {
Chris@1603 118
Chris@1603 119 maxLimit = 0.05;
Chris@1603 120 rmsLimit = 0.01;
Chris@1603 121
cannam@1315 122 } else if (format == "mp3") {
Chris@1313 123
Chris@1313 124 if (resampled && !gapless) {
Chris@1313 125
Chris@1313 126 // We expect worse figures here, because the
Chris@1313 127 // combination of uncompensated encoder delay +
Chris@1313 128 // resampling results in a fractional delay which
Chris@1313 129 // means the decoded signal is slightly out of
Chris@1313 130 // phase compared to the test signal
Chris@1313 131
Chris@1313 132 maxLimit = 0.1;
Chris@1313 133 rmsLimit = 0.05;
Chris@1313 134
Chris@1313 135 } else {
Chris@1313 136
Chris@1313 137 maxLimit = 0.05;
Chris@1313 138 rmsLimit = 0.01;
Chris@1313 139 }
Chris@1313 140
Chris@1313 141 } else {
Chris@1313 142
cannam@1315 143 // lossless formats (wav, aiff, flac, apple_lossless)
Chris@1313 144
Chris@1313 145 if (bitdepth >= 16 && !resampled) {
Chris@1313 146 maxLimit = 1e-3;
Chris@1313 147 rmsLimit = 3e-4;
Chris@1313 148 } else {
Chris@1313 149 maxLimit = 0.01;
Chris@1313 150 rmsLimit = 5e-3;
Chris@1313 151 }
Chris@1313 152 }
Chris@1313 153
Chris@1313 154 } else { // !normalised
Chris@1313 155
cannam@1315 156 if (format == "ogg") {
Chris@1313 157
Chris@1313 158 maxLimit = 0.06;
Chris@1313 159 rmsLimit = 0.03;
Chris@1313 160
Chris@1598 161 } else if (format == "opus") {
Chris@1598 162
Chris@1598 163 maxLimit = 0.06;
Chris@1598 164 rmsLimit = 0.015;
Chris@1598 165
cannam@1315 166 } else if (format == "aac") {
Chris@1313 167
Chris@1603 168 maxLimit = 0.2;
cannam@1315 169 rmsLimit = 0.1;
Chris@1313 170
Chris@1603 171 } else if (format == "wma") {
Chris@1603 172
Chris@1603 173 maxLimit = 0.05;
Chris@1603 174 rmsLimit = 0.01;
Chris@1603 175
cannam@1315 176 } else if (format == "mp3") {
Chris@1313 177
Chris@1313 178 // all mp3 figures are worse when not normalising
Chris@1313 179 maxLimit = 0.1;
Chris@1313 180 rmsLimit = 0.05;
Chris@1313 181
Chris@1313 182 } else {
Chris@1313 183
cannam@1315 184 // lossless formats (wav, aiff, flac, apple_lossless)
Chris@1313 185
Chris@1313 186 if (bitdepth >= 16 && !resampled) {
Chris@1313 187 maxLimit = 1e-3;
Chris@1313 188 rmsLimit = 3e-4;
Chris@1313 189 } else {
Chris@1313 190 maxLimit = 0.02;
Chris@1313 191 rmsLimit = 0.01;
Chris@1313 192 }
Chris@1313 193 }
Chris@1313 194 }
Chris@1313 195 }
Chris@1313 196
cannam@1315 197 QString testName(QString format, QString filename, int rate, bool norm, bool gapless) {
cannam@1315 198 return QString("%1/%2 at %3%4%5")
cannam@1315 199 .arg(format)
Chris@1313 200 .arg(filename)
Chris@1313 201 .arg(rate)
Chris@1313 202 .arg(norm ? " normalised": "")
Chris@1313 203 .arg(gapless ? "" : " non-gapless");
Chris@1313 204 }
Chris@1313 205
Chris@756 206 private slots:
Chris@756 207 void init()
Chris@756 208 {
Chris@756 209 if (!QDir(audioDir).exists()) {
Chris@1346 210 QString cwd = QDir::currentPath();
Chris@1428 211 SVCERR << "ERROR: Audio test file directory \"" << audioDir << "\" does not exist (cwd = " << cwd << ")" << endl;
Chris@756 212 QVERIFY2(QDir(audioDir).exists(), "Audio test file directory not found");
Chris@756 213 }
Chris@1313 214 if (!QDir(diffDir).exists() && !QDir().mkpath(diffDir)) {
Chris@1428 215 SVCERR << "ERROR: Audio diff directory \"" << diffDir << "\" does not exist and could not be created" << endl;
Chris@1313 216 QVERIFY2(QDir(diffDir).exists(), "Audio diff directory not found and could not be created");
Chris@1313 217 }
Chris@756 218 }
Chris@756 219
Chris@756 220 void read_data()
Chris@756 221 {
cannam@1315 222 QTest::addColumn<QString>("format");
Chris@756 223 QTest::addColumn<QString>("audiofile");
Chris@1313 224 QTest::addColumn<int>("rate");
Chris@1313 225 QTest::addColumn<bool>("normalised");
Chris@1313 226 QTest::addColumn<bool>("gapless");
cannam@1315 227 QStringList dirs = QDir(audioDir).entryList(QDir::Dirs |
cannam@1315 228 QDir::NoDotAndDotDot);
cannam@1315 229 for (QString format: dirs) {
cannam@1315 230 QStringList files = QDir(QDir(audioDir).filePath(format))
cannam@1315 231 .entryList(QDir::Files);
cannam@1315 232 int readRates[] = { 44100, 48000 };
cannam@1315 233 bool norms[] = { false, true };
cannam@1315 234 bool gaplesses[] = { true, false };
cannam@1315 235 foreach (QString filename, files) {
cannam@1315 236 for (int rate: readRates) {
cannam@1315 237 for (bool norm: norms) {
cannam@1315 238 for (bool gapless: gaplesses) {
Chris@1313 239
Chris@1603 240 #ifdef Q_OS_WIN
Chris@1603 241 if (format == "aac") {
Chris@1603 242 if (gapless) {
Chris@1603 243 // Apparently no support for AAC
Chris@1603 244 // encoder delay compensation in
Chris@1603 245 // MediaFoundation, so these tests
Chris@1603 246 // are only available non-gapless
Chris@1603 247 continue;
Chris@1603 248 }
Chris@1603 249 } else if (format != "mp3") {
Chris@1603 250 if (!gapless) {
Chris@1603 251 // All other formats but mp3 are
Chris@1603 252 // intrinsically gapless, so we
Chris@1603 253 // can skip the non-gapless option
Chris@1603 254 continue;
Chris@1603 255 }
cannam@1315 256 }
Chris@1603 257 #else
Chris@1603 258 if (format != "mp3") {
Chris@1603 259 if (!gapless) {
Chris@1603 260 // All other formats but mp3 are
Chris@1603 261 // intrinsically gapless
Chris@1603 262 // everywhere except for Windows
Chris@1603 263 // (see above), so we can skip the
Chris@1603 264 // non-gapless option
Chris@1603 265 continue;
Chris@1603 266 }
Chris@1603 267 }
Chris@1603 268 #endif
cannam@1315 269
cannam@1315 270 QString desc = testName
cannam@1315 271 (format, filename, rate, norm, gapless);
cannam@1315 272
cannam@1315 273 QTest::newRow(strOf(desc))
cannam@1315 274 << format << filename << rate << norm << gapless;
Chris@1313 275 }
Chris@1313 276 }
Chris@1313 277 }
Chris@1313 278 }
Chris@756 279 }
Chris@756 280 }
Chris@756 281
Chris@756 282 void read()
Chris@756 283 {
cannam@1315 284 QFETCH(QString, format);
Chris@756 285 QFETCH(QString, audiofile);
Chris@1313 286 QFETCH(int, rate);
Chris@1313 287 QFETCH(bool, normalised);
Chris@1313 288 QFETCH(bool, gapless);
Chris@756 289
Chris@1313 290 sv_samplerate_t readRate(rate);
Chris@1313 291
cannam@1315 292 // cerr << "\naudiofile = " << audiofile << endl;
Chris@1313 293
Chris@1313 294 AudioFileReaderFactory::Parameters params;
Chris@1313 295 params.targetRate = readRate;
Chris@1313 296 params.normalisation = (normalised ?
Chris@1313 297 AudioFileReaderFactory::Normalisation::Peak :
Chris@1313 298 AudioFileReaderFactory::Normalisation::None);
Chris@1313 299 params.gaplessMode = (gapless ?
Chris@1313 300 AudioFileReaderFactory::GaplessMode::Gapless :
Chris@1313 301 AudioFileReaderFactory::GaplessMode::Gappy);
Chris@757 302
Chris@1429 303 AudioFileReader *reader =
Chris@1429 304 AudioFileReaderFactory::createReader
Chris@1429 305 (audioDir + "/" + format + "/" + audiofile, params);
Chris@1313 306
Chris@1429 307 if (!reader) {
Chris@820 308 #if ( QT_VERSION >= 0x050000 )
Chris@1429 309 QSKIP("Unsupported file, skipping");
Chris@820 310 #else
Chris@1429 311 QSKIP("Unsupported file, skipping", SkipSingle);
Chris@820 312 #endif
Chris@1429 313 }
Chris@756 314
Chris@1313 315 QString extension;
Chris@1313 316 sv_samplerate_t fileRate;
Chris@1313 317 int channels;
Chris@1313 318 int fileBitdepth;
Chris@1313 319 getFileMetadata(audiofile, extension, fileRate, channels, fileBitdepth);
Chris@1313 320
Chris@1313 321 QCOMPARE((int)reader->getChannelCount(), channels);
Chris@1313 322 QCOMPARE(reader->getNativeRate(), fileRate);
Chris@1040 323 QCOMPARE(reader->getSampleRate(), readRate);
Chris@757 324
Chris@1429 325 AudioTestData tdata(readRate, channels);
Chris@1429 326
Chris@1429 327 float *reference = tdata.getInterleavedData();
Chris@1040 328 sv_frame_t refFrames = tdata.getFrameCount();
Chris@1429 329
Chris@1429 330 // The reader should give us exactly the expected number of
Chris@1429 331 // frames, except for mp3/aac files. We ask for quite a lot
Chris@1429 332 // more, though, so we can (a) check that we only get the
Chris@1429 333 // expected number back (if this is not mp3/aac) or (b) take
Chris@1429 334 // into account silence at beginning and end (if it is).
Chris@1429 335 floatvec_t test = reader->getInterleavedFrames(0, refFrames + 5000);
Chris@1402 336
Chris@1402 337 delete reader;
Chris@1402 338 reader = 0;
Chris@1402 339
Chris@1429 340 sv_frame_t read = test.size() / channels;
Chris@756 341
Chris@1313 342 bool perceptual = (extension == "mp3" ||
Chris@1313 343 extension == "aac" ||
Chris@1598 344 extension == "m4a" ||
Chris@1603 345 extension == "wma" ||
Chris@1598 346 extension == "opus");
Chris@1313 347
Chris@1313 348 if (perceptual && !gapless) {
Chris@1313 349 // allow silence at start and end
Chris@759 350 QVERIFY(read >= refFrames);
Chris@757 351 } else {
Chris@759 352 QCOMPARE(read, refFrames);
Chris@757 353 }
Chris@757 354
Chris@1313 355 bool resampled = readRate != fileRate;
Chris@1313 356 double maxLimit, rmsLimit;
cannam@1315 357 getExpectedThresholds(format,
cannam@1315 358 audiofile,
Chris@1313 359 resampled,
Chris@1313 360 gapless,
Chris@1313 361 normalised,
Chris@1313 362 maxLimit, rmsLimit);
Chris@1313 363
Chris@1313 364 double edgeLimit = maxLimit * 3; // in first or final edgeSize frames
Chris@1313 365 if (resampled && edgeLimit < 0.1) edgeLimit = 0.1;
Chris@759 366 int edgeSize = 100;
Chris@759 367
Chris@759 368 // And we ignore completely the last few frames when upsampling
Chris@1313 369 int discard = 1 + int(round(readRate / fileRate));
Chris@759 370
Chris@759 371 int offset = 0;
Chris@759 372
Chris@1313 373 if (perceptual) {
Chris@759 374
cannam@1314 375 // Look for an initial offset.
cannam@1314 376 //
cannam@1314 377 // We know the first channel has a sinusoid in it. It
cannam@1314 378 // should have a peak at 0.4ms (see AudioTestData.h) but
cannam@1314 379 // that might have been clipped, which would make it
cannam@1314 380 // imprecise. We can tell if it's clipped, though, as
cannam@1314 381 // there will be samples having exactly identical
cannam@1314 382 // values. So what we look for is the peak if it's not
cannam@1314 383 // clipped and, if it is, the first zero crossing after
cannam@1314 384 // the peak, which should be at 0.8ms.
cannam@1314 385
Chris@1296 386 int expectedPeak = int(0.0004 * readRate);
cannam@1314 387 int expectedZC = int(0.0008 * readRate);
cannam@1314 388 bool foundPeak = false;
cannam@1314 389 for (int i = 1; i+1 < read; ++i) {
cannam@1314 390 float prevSample = test[(i-1) * channels];
cannam@1314 391 float thisSample = test[i * channels];
cannam@1314 392 float nextSample = test[(i+1) * channels];
cannam@1314 393 if (thisSample > 0.8 && nextSample < thisSample) {
cannam@1314 394 foundPeak = true;
cannam@1314 395 if (thisSample > prevSample) {
cannam@1314 396 // not clipped
cannam@1314 397 offset = i - expectedPeak - 1;
cannam@1314 398 break;
cannam@1314 399 }
cannam@1314 400 }
cannam@1314 401 if (foundPeak && (thisSample >= 0.0 && nextSample < 0.0)) {
cannam@1315 402 // cerr << "thisSample = " << thisSample << ", nextSample = "
cannam@1315 403 // << nextSample << endl;
cannam@1314 404 offset = i - expectedZC - 1;
Chris@759 405 break;
Chris@759 406 }
Chris@759 407 }
Chris@1313 408
cannam@1315 409 // int fileRateEquivalent = int((offset / readRate) * fileRate);
cannam@1315 410 // std::cerr << "offset = " << offset << std::endl;
cannam@1315 411 // std::cerr << "at file rate would be " << fileRateEquivalent << std::endl;
Chris@1313 412
Chris@1313 413 // Previously our m4a test file had a fixed offset of 1024
Chris@1313 414 // at the file sample rate -- this may be because it was
Chris@1313 415 // produced by FAAC which did not write in the delay as
Chris@1313 416 // metadata? We now have an m4a produced by Core Audio
Chris@1313 417 // which gives a 0 offset. What to do...
Chris@1313 418
Chris@1313 419 // Anyway, mp3s should have 0 offset in gapless mode and
Chris@1313 420 // "something else" otherwise.
Chris@1313 421
Chris@1313 422 if (gapless) {
Chris@1603 423 if (format == "aac"
Chris@1603 424 #ifdef Q_OS_WIN
Chris@1603 425 || (format == "mp3" && (readRate != fileRate))
Chris@1603 426 #endif
Chris@1603 427 ) {
cannam@1315 428 // ouch!
cannam@1315 429 if (offset == -1) offset = 0;
cannam@1315 430 }
Chris@1313 431 QCOMPARE(offset, 0);
Chris@1313 432 }
Chris@759 433 }
Chris@756 434
cannam@1315 435 {
cannam@1315 436 // Write the diff file now, so that it's already been written
cannam@1315 437 // even if the comparison fails. We aren't checking anything
cannam@1315 438 // here except as necessary to avoid buffer overruns etc
cannam@1315 439
cannam@1315 440 QString diffFile =
cannam@1315 441 testName(format, audiofile, rate, normalised, gapless);
cannam@1315 442 diffFile.replace("/", "_");
cannam@1315 443 diffFile.replace(".", "_");
cannam@1315 444 diffFile.replace(" ", "_");
cannam@1315 445 diffFile += ".wav";
cannam@1315 446 diffFile = QDir(diffDir).filePath(diffFile);
cannam@1315 447 WavFileWriter diffWriter(diffFile, readRate, channels,
Chris@1359 448 WavFileWriter::WriteToTemporary);
cannam@1315 449 QVERIFY(diffWriter.isOK());
cannam@1315 450
cannam@1315 451 vector<vector<float>> diffs(channels);
cannam@1315 452 for (int c = 0; c < channels; ++c) {
cannam@1315 453 for (int i = 0; i < refFrames; ++i) {
cannam@1315 454 int ix = i + offset;
cannam@1315 455 if (ix < read) {
cannam@1315 456 float signeddiff =
cannam@1315 457 test[ix * channels + c] -
cannam@1315 458 reference[i * channels + c];
cannam@1315 459 diffs[c].push_back(signeddiff);
cannam@1315 460 }
cannam@1315 461 }
cannam@1315 462 }
cannam@1315 463 float **ptrs = new float*[channels];
cannam@1315 464 for (int c = 0; c < channels; ++c) {
cannam@1315 465 ptrs[c] = diffs[c].data();
cannam@1315 466 }
cannam@1315 467 diffWriter.writeSamples(ptrs, refFrames);
cannam@1315 468 delete[] ptrs;
cannam@1315 469 }
Chris@1313 470
Chris@1346 471 for (int c = 0; c < channels; ++c) {
Chris@1313 472
Chris@1313 473 double maxDiff = 0.0;
Chris@1313 474 double totalDiff = 0.0;
Chris@1313 475 double totalSqrDiff = 0.0;
Chris@1346 476 int maxIndex = 0;
Chris@1313 477
Chris@1346 478 for (int i = 0; i < refFrames; ++i) {
Chris@1296 479 int ix = i + offset;
Chris@1296 480 if (ix >= read) {
Chris@1428 481 SVCERR << "ERROR: audiofile " << audiofile << " reads truncated (read-rate reference frames " << i << " onward, of " << refFrames << ", are lost)" << endl;
Chris@1296 482 QVERIFY(ix < read);
Chris@1296 483 }
Chris@1313 484
Chris@1296 485 if (ix + discard >= read) {
Chris@1296 486 // we forgive the very edge samples when
Chris@1296 487 // resampling (discard > 0)
Chris@1296 488 continue;
Chris@1296 489 }
Chris@1313 490
Chris@1346 491 double diff = fabs(test[ix * channels + c] -
cannam@1315 492 reference[i * channels + c]);
Chris@1313 493
Chris@1346 494 totalDiff += diff;
Chris@1313 495 totalSqrDiff += diff * diff;
Chris@1313 496
Chris@757 497 // in edge areas, record this only if it exceeds edgeLimit
Chris@1313 498 if (i < edgeSize || i + edgeSize >= refFrames) {
Chris@1313 499 if (diff > edgeLimit && diff > maxDiff) {
Chris@1313 500 maxDiff = diff;
Chris@1313 501 maxIndex = i;
Chris@757 502 }
Chris@757 503 } else {
Chris@1313 504 if (diff > maxDiff) {
Chris@1313 505 maxDiff = diff;
Chris@1313 506 maxIndex = i;
Chris@757 507 }
Chris@1346 508 }
Chris@1346 509 }
Chris@1313 510
Chris@1346 511 double meanDiff = totalDiff / double(refFrames);
Chris@1313 512 double rmsDiff = sqrt(totalSqrDiff / double(refFrames));
cannam@1308 513
cannam@1314 514 /*
Chris@1346 515 cerr << "channel " << c << ": mean diff " << meanDiff << endl;
Chris@1429 516 cerr << "channel " << c << ": rms diff " << rmsDiff << endl;
Chris@1429 517 cerr << "channel " << c << ": max diff " << maxDiff << " at " << maxIndex << endl;
cannam@1314 518 */
Chris@1313 519 if (rmsDiff >= rmsLimit) {
Chris@1428 520 SVCERR << "ERROR: for audiofile " << audiofile << ": RMS diff = " << rmsDiff << " for channel " << c << " (limit = " << rmsLimit << ")" << endl;
Chris@1313 521 QVERIFY(rmsDiff < rmsLimit);
Chris@1313 522 }
Chris@1346 523 if (maxDiff >= maxLimit) {
Chris@1428 524 SVCERR << "ERROR: for audiofile " << audiofile << ": max diff = " << maxDiff << " at frame " << maxIndex << " of " << read << " on channel " << c << " (limit = " << maxLimit << ", edge limit = " << edgeLimit << ", mean diff = " << meanDiff << ", rms = " << rmsDiff << ")" << endl;
Chris@1346 525 QVERIFY(maxDiff < maxLimit);
Chris@1346 526 }
Chris@1313 527
Chris@1313 528 // and check for spurious material at end
Chris@1313 529
Chris@1309 530 for (sv_frame_t i = refFrames; i + offset < read; ++i) {
Chris@1309 531 sv_frame_t ix = i + offset;
Chris@1323 532 float quiet = 0.1f; //!!! allow some ringing - but let's come back to this, it should tail off
cannam@1308 533 float mag = fabsf(test[ix * channels + c]);
cannam@1308 534 if (mag > quiet) {
Chris@1428 535 SVCERR << "ERROR: audiofile " << audiofile << " contains spurious data after end of reference (found sample " << test[ix * channels + c] << " at index " << ix << " of channel " << c << " after reference+offset ended at " << refFrames+offset << ")" << endl;
cannam@1308 536 QVERIFY(mag < quiet);
cannam@1308 537 }
cannam@1308 538 }
Chris@1429 539 }
Chris@756 540 }
Chris@756 541 };
Chris@756 542
Chris@756 543 #endif