Chris@756: /* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */
Chris@756:
Chris@756: /*
Chris@756: Sonic Visualiser
Chris@756: An audio file viewer and annotation editor.
Chris@756: Centre for Digital Music, Queen Mary, University of London.
Chris@756: This file copyright 2013 Chris Cannam.
Chris@756:
Chris@756: This program is free software; you can redistribute it and/or
Chris@756: modify it under the terms of the GNU General Public License as
Chris@756: published by the Free Software Foundation; either version 2 of the
Chris@756: License, or (at your option) any later version. See the file
Chris@756: COPYING included with this distribution for more information.
Chris@756: */
Chris@756:
Chris@756: #ifndef TEST_AUDIO_FILE_READER_H
Chris@756: #define TEST_AUDIO_FILE_READER_H
Chris@756:
Chris@756: #include "../AudioFileReaderFactory.h"
Chris@756: #include "../AudioFileReader.h"
Chris@1313: #include "../WavFileWriter.h"
Chris@756:
Chris@756: #include "AudioTestData.h"
Chris@1698: #include "UnsupportedFormat.h"
Chris@756:
Chris@756: #include <cmath>
Chris@756:
Chris@756: #include <QObject>
Chris@756: #include <QtTest>
Chris@756: #include <QDir>
Chris@756:
Chris@756: #include <iostream>
Chris@756:
Chris@756: using namespace std;
Chris@756:
Chris@756: class AudioFileReaderTest : public QObject
Chris@756: {
Chris@756: Q_OBJECT
Chris@756:
Chris@1346: private:
Chris@1346: QString testDirBase;
Chris@1346: QString audioDir;
Chris@1346: QString diffDir;
Chris@1346:
Chris@1346: public:
Chris@1346: AudioFileReaderTest(QString base) {
Chris@1346: if (base == "") {
Chris@1346: base = "svcore/data/fileio/test";
Chris@1346: }
Chris@1346: testDirBase = base;
Chris@1359: audioDir = base + "/audio";
Chris@1346: diffDir = base + "/diffs";
Chris@1346: }
Chris@1346:
Chris@1346: private:
Chris@756: const char *strOf(QString s) {
Chris@756: return strdup(s.toLocal8Bit().data());
Chris@756: }
Chris@756:
Chris@1313: void getFileMetadata(QString filename,
Chris@1313: QString &extension,
Chris@1313: sv_samplerate_t &rate,
Chris@1313: int &channels,
Chris@1313: int &bitdepth) {
Chris@1313:
Chris@1313: QStringList fileAndExt = filename.split(".");
Chris@1313: QStringList bits = fileAndExt[0].split("-");
Chris@1313:
Chris@1313: extension = fileAndExt[1];
Chris@1313: rate = bits[0].toInt();
Chris@1313: channels = bits[1].toInt();
Chris@1313: bitdepth = 16;
Chris@1313: if (bits.length() > 2) {
Chris@1313: bitdepth = bits[2].toInt();
Chris@1313: }
Chris@1313: }
Chris@1698:
cannam@1315: void getExpectedThresholds(QString format,
cannam@1315: QString filename,
Chris@1313: bool resampled,
Chris@1313: bool gapless,
Chris@1313: bool normalised,
Chris@1313: double &maxLimit,
Chris@1313: double &rmsLimit) {
Chris@1313:
Chris@1313: QString extension;
Chris@1313: sv_samplerate_t fileRate;
Chris@1313: int channels;
Chris@1313: int bitdepth;
Chris@1313: getFileMetadata(filename, extension, fileRate, channels, bitdepth);
Chris@1313:
Chris@1313: if (normalised) {
Chris@1313:
cannam@1315: if (format == "ogg") {
Chris@1313:
Chris@1313: // Our ogg is not especially high quality and is
Chris@1313: // actually further from the original if normalised
Chris@1313:
Chris@1313: maxLimit = 0.1;
Chris@1313: rmsLimit = 0.03;
Chris@1313:
Chris@1598: } else if (format == "opus") {
Chris@1598:
Chris@1598: maxLimit = 0.06;
Chris@1598: rmsLimit = 0.015;
Chris@1598:
cannam@1315: } else if (format == "aac") {
Chris@1313:
cannam@1315: // Terrible performance for this test, load of spill
cannam@1315: // from one channel to the other. I guess they know
cannam@1315: // what they're doing, it's perceptual after all, but
cannam@1315: // it does make this check a bit superfluous, you
cannam@1315: // could probably pass it with a signal that sounds
cannam@1315: // nothing like the original
cannam@1315: maxLimit = 0.2;
cannam@1314: rmsLimit = 0.1;
Chris@1313:
Chris@1603: } else if (format == "wma") {
Chris@1603:
Chris@1603: maxLimit = 0.05;
Chris@1603: rmsLimit = 0.01;
Chris@1603:
cannam@1315: } else if (format == "mp3") {
Chris@1313:
Chris@1313: if (resampled && !gapless) {
Chris@1313:
Chris@1313: // We expect worse figures here, because the
Chris@1313: // combination of uncompensated encoder delay +
Chris@1313: // resampling results in a fractional delay which
Chris@1313: // means the decoded signal is slightly out of
Chris@1313: // phase compared to the test signal
Chris@1313:
Chris@1313: maxLimit = 0.1;
Chris@1313: rmsLimit = 0.05;
Chris@1313:
Chris@1313: } else {
Chris@1313:
Chris@1313: maxLimit = 0.05;
Chris@1313: rmsLimit = 0.01;
Chris@1313: }
Chris@1313:
Chris@1313: } else {
Chris@1313:
cannam@1315: // lossless formats (wav, aiff, flac, apple_lossless)
Chris@1313:
Chris@1313: if (bitdepth >= 16 && !resampled) {
Chris@1313: maxLimit = 1e-3;
Chris@1313: rmsLimit = 3e-4;
Chris@1313: } else {
Chris@1313: maxLimit = 0.01;
Chris@1313: rmsLimit = 5e-3;
Chris@1313: }
Chris@1313: }
Chris@1313:
Chris@1313: } else { // !normalised
Chris@1313:
cannam@1315: if (format == "ogg") {
Chris@1313:
Chris@1313: maxLimit = 0.06;
Chris@1313: rmsLimit = 0.03;
Chris@1313:
Chris@1598: } else if (format == "opus") {
Chris@1598:
Chris@1598: maxLimit = 0.06;
Chris@1598: rmsLimit = 0.015;
Chris@1598:
cannam@1315: } else if (format == "aac") {
Chris@1313:
Chris@1603: maxLimit = 0.2;
cannam@1315: rmsLimit = 0.1;
Chris@1313:
Chris@1603: } else if (format == "wma") {
Chris@1603:
Chris@1603: maxLimit = 0.05;
Chris@1603: rmsLimit = 0.01;
Chris@1603:
cannam@1315: } else if (format == "mp3") {
Chris@1313:
Chris@1313: // all mp3 figures are worse when not normalising
Chris@1313: maxLimit = 0.1;
Chris@1313: rmsLimit = 0.05;
Chris@1313:
Chris@1313: } else {
Chris@1313:
cannam@1315: // lossless formats (wav, aiff, flac, apple_lossless)
Chris@1313:
Chris@1313: if (bitdepth >= 16 && !resampled) {
Chris@1313: maxLimit = 1e-3;
Chris@1313: rmsLimit = 3e-4;
Chris@1313: } else {
Chris@1313: maxLimit = 0.02;
Chris@1313: rmsLimit = 0.01;
Chris@1313: }
Chris@1313: }
Chris@1313: }
Chris@1313: }
Chris@1313:
cannam@1315: QString testName(QString format, QString filename, int rate, bool norm, bool gapless) {
cannam@1315: return QString("%1/%2 at %3%4%5")
cannam@1315: .arg(format)
Chris@1313: .arg(filename)
Chris@1313: .arg(rate)
Chris@1313: .arg(norm ? " normalised": "")
Chris@1313: .arg(gapless ? "" : " non-gapless");
Chris@1313: }
Chris@1313:
Chris@756: private slots:
Chris@756: void init()
Chris@756: {
Chris@756: if (!QDir(audioDir).exists()) {
Chris@1346: QString cwd = QDir::currentPath();
Chris@1428: SVCERR << "ERROR: Audio test file directory \"" << audioDir << "\" does not exist (cwd = " << cwd << ")" << endl;
Chris@756: QVERIFY2(QDir(audioDir).exists(), "Audio test file directory not found");
Chris@756: }
Chris@1313: if (!QDir(diffDir).exists() && !QDir().mkpath(diffDir)) {
Chris@1428: SVCERR << "ERROR: Audio diff directory \"" << diffDir << "\" does not exist and could not be created" << endl;
Chris@1313: QVERIFY2(QDir(diffDir).exists(), "Audio diff directory not found and could not be created");
Chris@1313: }
Chris@756: }
Chris@756:
Chris@756: void read_data()
Chris@756: {
cannam@1315: QTest::addColumn<QString>("format");
Chris@756: QTest::addColumn<QString>("audiofile");
Chris@1313: QTest::addColumn<int>("rate");
Chris@1313: QTest::addColumn<bool>("normalised");
Chris@1313: QTest::addColumn<bool>("gapless");
cannam@1315: QStringList dirs = QDir(audioDir).entryList(QDir::Dirs |
cannam@1315: QDir::NoDotAndDotDot);
cannam@1315: for (QString format: dirs) {
cannam@1315: QStringList files = QDir(QDir(audioDir).filePath(format))
cannam@1315: .entryList(QDir::Files);
cannam@1315: int readRates[] = { 44100, 48000 };
cannam@1315: bool norms[] = { false, true };
cannam@1315: bool gaplesses[] = { true, false };
cannam@1315: foreach (QString filename, files) {
cannam@1315: for (int rate: readRates) {
cannam@1315: for (bool norm: norms) {
cannam@1315: for (bool gapless: gaplesses) {
Chris@1313:
Chris@1603: #ifdef Q_OS_WIN
Chris@1603: if (format == "aac") {
Chris@1603: if (gapless) {
Chris@1603: // Apparently no support for AAC
Chris@1603: // encoder delay compensation in
Chris@1603: // MediaFoundation, so these tests
Chris@1603: // are only available non-gapless
Chris@1603: continue;
Chris@1603: }
Chris@1603: } else if (format != "mp3") {
Chris@1603: if (!gapless) {
Chris@1603: // All other formats but mp3 are
Chris@1603: // intrinsically gapless, so we
Chris@1603: // can skip the non-gapless option
Chris@1603: continue;
Chris@1603: }
cannam@1315: }
Chris@1603: #else
Chris@1603: if (format != "mp3") {
Chris@1603: if (!gapless) {
Chris@1603: // All other formats but mp3 are
Chris@1603: // intrinsically gapless
Chris@1603: // everywhere except for Windows
Chris@1603: // (see above), so we can skip the
Chris@1603: // non-gapless option
Chris@1603: continue;
Chris@1603: }
Chris@1603: }
Chris@1603: #endif
cannam@1315:
cannam@1315: QString desc = testName
cannam@1315: (format, filename, rate, norm, gapless);
cannam@1315:
cannam@1315: QTest::newRow(strOf(desc))
cannam@1315: << format << filename << rate << norm << gapless;
Chris@1313: }
Chris@1313: }
Chris@1313: }
Chris@1313: }
Chris@756: }
Chris@756: }
Chris@756:
Chris@756: void read()
Chris@756: {
cannam@1315: QFETCH(QString, format);
Chris@756: QFETCH(QString, audiofile);
Chris@1313: QFETCH(int, rate);
Chris@1313: QFETCH(bool, normalised);
Chris@1313: QFETCH(bool, gapless);
Chris@756:
Chris@1313: sv_samplerate_t readRate(rate);
Chris@1313:
cannam@1315: // cerr << "\naudiofile = " << audiofile << endl;
Chris@1313:
Chris@1313: AudioFileReaderFactory::Parameters params;
Chris@1313: params.targetRate = readRate;
Chris@1313: params.normalisation = (normalised ?
Chris@1313: AudioFileReaderFactory::Normalisation::Peak :
Chris@1313: AudioFileReaderFactory::Normalisation::None);
Chris@1313: params.gaplessMode = (gapless ?
Chris@1313: AudioFileReaderFactory::GaplessMode::Gapless :
Chris@1313: AudioFileReaderFactory::GaplessMode::Gappy);
Chris@757:
Chris@1429: AudioFileReader *reader =
Chris@1429: AudioFileReaderFactory::createReader
Chris@1429: (audioDir + "/" + format + "/" + audiofile, params);
Chris@1313:
Chris@1429: if (!reader) {
Chris@1714: if (UnsupportedFormat::isLegitimatelyUnsupported(format)) {
Chris@820: #if ( QT_VERSION >= 0x050000 )
Chris@1698: QSKIP("Unsupported file, skipping");
Chris@820: #else
Chris@1698: QSKIP("Unsupported file, skipping", SkipSingle);
Chris@820: #endif
Chris@1698: }
Chris@1429: }
Chris@756:
cannam@1700: QVERIFY(reader != nullptr);
cannam@1700:
Chris@1313: QString extension;
Chris@1313: sv_samplerate_t fileRate;
Chris@1313: int channels;
Chris@1313: int fileBitdepth;
Chris@1313: getFileMetadata(audiofile, extension, fileRate, channels, fileBitdepth);
Chris@1313:
Chris@1313: QCOMPARE((int)reader->getChannelCount(), channels);
Chris@1313: QCOMPARE(reader->getNativeRate(), fileRate);
Chris@1040: QCOMPARE(reader->getSampleRate(), readRate);
Chris@757:
Chris@1429: AudioTestData tdata(readRate, channels);
Chris@1429:
Chris@1429: float *reference = tdata.getInterleavedData();
Chris@1040: sv_frame_t refFrames = tdata.getFrameCount();
Chris@1429:
Chris@1429: // The reader should give us exactly the expected number of
Chris@1429: // frames, except for mp3/aac files. We ask for quite a lot
Chris@1429: // more, though, so we can (a) check that we only get the
Chris@1429: // expected number back (if this is not mp3/aac) or (b) take
Chris@1429: // into account silence at beginning and end (if it is).
Chris@1429: floatvec_t test = reader->getInterleavedFrames(0, refFrames + 5000);
Chris@1402:
Chris@1402: delete reader;
Chris@1402: reader = 0;
Chris@1402:
Chris@1429: sv_frame_t read = test.size() / channels;
Chris@756:
Chris@1313: bool perceptual = (extension == "mp3" ||
Chris@1313: extension == "aac" ||
Chris@1598: extension == "m4a" ||
Chris@1603: extension == "wma" ||
Chris@1598: extension == "opus");
Chris@1313:
Chris@1313: if (perceptual && !gapless) {
Chris@1313: // allow silence at start and end
Chris@759: QVERIFY(read >= refFrames);
Chris@757: } else {
Chris@759: QCOMPARE(read, refFrames);
Chris@757: }
Chris@757:
Chris@1313: bool resampled = readRate != fileRate;
Chris@1313: double maxLimit, rmsLimit;
cannam@1315: getExpectedThresholds(format,
cannam@1315: audiofile,
Chris@1313: resampled,
Chris@1313: gapless,
Chris@1313: normalised,
Chris@1313: maxLimit, rmsLimit);
Chris@1313:
Chris@1313: double edgeLimit = maxLimit * 3; // in first or final edgeSize frames
Chris@1313: if (resampled && edgeLimit < 0.1) edgeLimit = 0.1;
Chris@759: int edgeSize = 100;
Chris@759:
Chris@759: // And we ignore completely the last few frames when upsampling
Chris@1313: int discard = 1 + int(round(readRate / fileRate));
Chris@759:
Chris@759: int offset = 0;
Chris@759:
Chris@1313: if (perceptual) {
Chris@759:
cannam@1314: // Look for an initial offset.
cannam@1314: //
cannam@1314: // We know the first channel has a sinusoid in it. It
cannam@1314: // should have a peak at 0.4ms (see AudioTestData.h) but
cannam@1314: // that might have been clipped, which would make it
cannam@1314: // imprecise. We can tell if it's clipped, though, as
cannam@1314: // there will be samples having exactly identical
cannam@1314: // values. So what we look for is the peak if it's not
cannam@1314: // clipped and, if it is, the first zero crossing after
cannam@1314: // the peak, which should be at 0.8ms.
cannam@1314:
Chris@1296: int expectedPeak = int(0.0004 * readRate);
cannam@1314: int expectedZC = int(0.0008 * readRate);
cannam@1314: bool foundPeak = false;
cannam@1314: for (int i = 1; i+1 < read; ++i) {
cannam@1314: float prevSample = test[(i-1) * channels];
cannam@1314: float thisSample = test[i * channels];
cannam@1314: float nextSample = test[(i+1) * channels];
cannam@1314: if (thisSample > 0.8 && nextSample < thisSample) {
cannam@1314: foundPeak = true;
cannam@1314: if (thisSample > prevSample) {
cannam@1314: // not clipped
cannam@1314: offset = i - expectedPeak - 1;
cannam@1314: break;
cannam@1314: }
cannam@1314: }
cannam@1314: if (foundPeak && (thisSample >= 0.0 && nextSample < 0.0)) {
cannam@1315: // cerr << "thisSample = " << thisSample << ", nextSample = "
cannam@1315: // << nextSample << endl;
cannam@1314: offset = i - expectedZC - 1;
Chris@759: break;
Chris@759: }
Chris@759: }
Chris@1313:
cannam@1315: // int fileRateEquivalent = int((offset / readRate) * fileRate);
cannam@1315: // std::cerr << "offset = " << offset << std::endl;
cannam@1315: // std::cerr << "at file rate would be " << fileRateEquivalent << std::endl;
Chris@1313:
Chris@1313: // Previously our m4a test file had a fixed offset of 1024
Chris@1313: // at the file sample rate -- this may be because it was
Chris@1313: // produced by FAAC which did not write in the delay as
Chris@1313: // metadata? We now have an m4a produced by Core Audio
Chris@1313: // which gives a 0 offset. What to do...
Chris@1313:
Chris@1313: // Anyway, mp3s should have 0 offset in gapless mode and
Chris@1313: // "something else" otherwise.
Chris@1313:
Chris@1313: if (gapless) {
Chris@1603: if (format == "aac"
Chris@1603: #ifdef Q_OS_WIN
Chris@1603: || (format == "mp3" && (readRate != fileRate))
Chris@1603: #endif
Chris@1603: ) {
cannam@1315: // ouch!
cannam@1315: if (offset == -1) offset = 0;
cannam@1315: }
Chris@1313: QCOMPARE(offset, 0);
Chris@1313: }
Chris@759: }
Chris@756:
cannam@1315: {
cannam@1315: // Write the diff file now, so that it's already been written
cannam@1315: // even if the comparison fails. We aren't checking anything
cannam@1315: // here except as necessary to avoid buffer overruns etc
cannam@1315:
cannam@1315: QString diffFile =
cannam@1315: testName(format, audiofile, rate, normalised, gapless);
cannam@1315: diffFile.replace("/", "_");
cannam@1315: diffFile.replace(".", "_");
cannam@1315: diffFile.replace(" ", "_");
cannam@1315: diffFile += ".wav";
cannam@1315: diffFile = QDir(diffDir).filePath(diffFile);
cannam@1315: WavFileWriter diffWriter(diffFile, readRate, channels,
Chris@1359: WavFileWriter::WriteToTemporary);
cannam@1315: QVERIFY(diffWriter.isOK());
cannam@1315:
cannam@1315: vector<vector<float>> diffs(channels);
cannam@1315: for (int c = 0; c < channels; ++c) {
cannam@1315: for (int i = 0; i < refFrames; ++i) {
cannam@1315: int ix = i + offset;
cannam@1315: if (ix < read) {
cannam@1315: float signeddiff =
cannam@1315: test[ix * channels + c] -
cannam@1315: reference[i * channels + c];
cannam@1315: diffs[c].push_back(signeddiff);
cannam@1315: }
cannam@1315: }
cannam@1315: }
cannam@1315: float **ptrs = new float*[channels];
cannam@1315: for (int c = 0; c < channels; ++c) {
cannam@1315: ptrs[c] = diffs[c].data();
cannam@1315: }
cannam@1315: diffWriter.writeSamples(ptrs, refFrames);
cannam@1315: delete[] ptrs;
cannam@1315: }
Chris@1313:
Chris@1346: for (int c = 0; c < channels; ++c) {
Chris@1313:
Chris@1313: double maxDiff = 0.0;
Chris@1313: double totalDiff = 0.0;
Chris@1313: double totalSqrDiff = 0.0;
Chris@1346: int maxIndex = 0;
Chris@1313:
Chris@1346: for (int i = 0; i < refFrames; ++i) {
Chris@1296: int ix = i + offset;
Chris@1296: if (ix >= read) {
Chris@1428: SVCERR << "ERROR: audiofile " << audiofile << " reads truncated (read-rate reference frames " << i << " onward, of " << refFrames << ", are lost)" << endl;
Chris@1296: QVERIFY(ix < read);
Chris@1296: }
Chris@1313:
Chris@1296: if (ix + discard >= read) {
Chris@1296: // we forgive the very edge samples when
Chris@1296: // resampling (discard > 0)
Chris@1296: continue;
Chris@1296: }
Chris@1313:
Chris@1346: double diff = fabs(test[ix * channels + c] -
cannam@1315: reference[i * channels + c]);
Chris@1313:
Chris@1346: totalDiff += diff;
Chris@1313: totalSqrDiff += diff * diff;
Chris@1313:
Chris@757: // in edge areas, record this only if it exceeds edgeLimit
Chris@1313: if (i < edgeSize || i + edgeSize >= refFrames) {
Chris@1313: if (diff > edgeLimit && diff > maxDiff) {
Chris@1313: maxDiff = diff;
Chris@1313: maxIndex = i;
Chris@757: }
Chris@757: } else {
Chris@1313: if (diff > maxDiff) {
Chris@1313: maxDiff = diff;
Chris@1313: maxIndex = i;
Chris@757: }
Chris@1346: }
Chris@1346: }
Chris@1313:
Chris@1346: double meanDiff = totalDiff / double(refFrames);
Chris@1313: double rmsDiff = sqrt(totalSqrDiff / double(refFrames));
cannam@1308:
cannam@1314: /*
Chris@1346: cerr << "channel " << c << ": mean diff " << meanDiff << endl;
Chris@1429: cerr << "channel " << c << ": rms diff " << rmsDiff << endl;
Chris@1429: cerr << "channel " << c << ": max diff " << maxDiff << " at " << maxIndex << endl;
cannam@1314: */
Chris@1313: if (rmsDiff >= rmsLimit) {
Chris@1428: SVCERR << "ERROR: for audiofile " << audiofile << ": RMS diff = " << rmsDiff << " for channel " << c << " (limit = " << rmsLimit << ")" << endl;
Chris@1313: QVERIFY(rmsDiff < rmsLimit);
Chris@1313: }
Chris@1346: if (maxDiff >= maxLimit) {
Chris@1428: 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: QVERIFY(maxDiff < maxLimit);
Chris@1346: }
Chris@1313:
Chris@1313: // and check for spurious material at end
Chris@1313:
Chris@1309: for (sv_frame_t i = refFrames; i + offset < read; ++i) {
Chris@1309: sv_frame_t ix = i + offset;
Chris@1323: float quiet = 0.1f; //!!! allow some ringing - but let's come back to this, it should tail off
cannam@1308: float mag = fabsf(test[ix * channels + c]);
cannam@1308: if (mag > quiet) {
Chris@1428: 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: QVERIFY(mag < quiet);
cannam@1308: }
cannam@1308: }
Chris@1429: }
Chris@756: }
Chris@756: };
Chris@756:
Chris@756: #endif