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@756:
Chris@756: #include "AudioTestData.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@1263: static QString audioDir = "svcore/data/fileio/test/testfiles";
Chris@756:
Chris@756: class AudioFileReaderTest : public QObject
Chris@756: {
Chris@756: Q_OBJECT
Chris@756:
Chris@756: const char *strOf(QString s) {
Chris@756: return strdup(s.toLocal8Bit().data());
Chris@756: }
Chris@756:
Chris@756: private slots:
Chris@756: void init()
Chris@756: {
Chris@756: if (!QDir(audioDir).exists()) {
Chris@756: cerr << "ERROR: Audio test file directory \"" << audioDir << "\" does not exist" << endl;
Chris@756: QVERIFY2(QDir(audioDir).exists(), "Audio test file directory not found");
Chris@756: }
Chris@756: }
Chris@756:
Chris@756: void read_data()
Chris@756: {
Chris@756: QTest::addColumn<QString>("audiofile");
Chris@756: QStringList files = QDir(audioDir).entryList(QDir::Files);
Chris@756: foreach (QString filename, files) {
Chris@756: QTest::newRow(strOf(filename)) << filename;
Chris@756: }
Chris@756: }
Chris@756:
Chris@756: void read()
Chris@756: {
Chris@756: QFETCH(QString, audiofile);
Chris@756:
Chris@1040: sv_samplerate_t readRate = 48000;
Chris@757:
Chris@756: AudioFileReader *reader =
Chris@756: AudioFileReaderFactory::createReader
Chris@757: (audioDir + "/" + audiofile, readRate);
Chris@757:
Chris@757: QStringList fileAndExt = audiofile.split(".");
Chris@757: QStringList bits = fileAndExt[0].split("-");
Chris@757: QString extension = fileAndExt[1];
Chris@1040: sv_samplerate_t nominalRate = bits[0].toInt();
Chris@757: int nominalChannels = bits[1].toInt();
Chris@757: int nominalDepth = 16;
Chris@757: if (bits.length() > 2) nominalDepth = bits[2].toInt();
Chris@756:
Chris@756: if (!reader) {
Chris@820: #if ( QT_VERSION >= 0x050000 )
Chris@763: QSKIP("Unsupported file, skipping");
Chris@820: #else
Chris@820: QSKIP("Unsupported file, skipping", SkipSingle);
Chris@820: #endif
Chris@756: }
Chris@756:
Chris@757: QCOMPARE((int)reader->getChannelCount(), nominalChannels);
Chris@1040: QCOMPARE(reader->getNativeRate(), nominalRate);
Chris@1040: QCOMPARE(reader->getSampleRate(), readRate);
Chris@757:
Chris@756: int channels = reader->getChannelCount();
Chris@757: AudioTestData tdata(readRate, channels);
Chris@756:
Chris@756: float *reference = tdata.getInterleavedData();
Chris@1040: sv_frame_t refFrames = tdata.getFrameCount();
Chris@756:
Chris@756: // The reader should give us exactly the expected number of
Chris@759: // frames, except for mp3/aac files. We ask for quite a lot
Chris@759: // more, though, so we can (a) check that we only get the
Chris@759: // expected number back (if this is not mp3/aac) or (b) take
Chris@759: // into account silence at beginning and end (if it is).
Chris@1041: vector<float> test = reader->getInterleavedFrames(0, refFrames + 5000);
Chris@1040: sv_frame_t read = test.size() / channels;
Chris@756:
Chris@759: if (extension == "mp3" || extension == "aac" || extension == "m4a") {
Chris@759: // mp3s and aacs can have silence at start and end
Chris@759: QVERIFY(read >= refFrames);
Chris@757: } else {
Chris@759: QCOMPARE(read, refFrames);
Chris@757: }
Chris@757:
Chris@757: // Our limits are pretty relaxed -- we're not testing decoder
Chris@757: // or resampler quality here, just whether the results are
Chris@757: // plainly wrong (e.g. at wrong samplerate or with an offset)
Chris@757:
Chris@1040: double limit = 0.01;
Chris@1040: double edgeLimit = limit * 10; // in first or final edgeSize frames
Chris@759: int edgeSize = 100;
Chris@759:
Chris@757: if (nominalDepth < 16) {
Chris@757: limit = 0.02;
Chris@757: }
Chris@759: if (extension == "ogg" || extension == "mp3" ||
Chris@759: extension == "aac" || extension == "m4a") {
Chris@1296: limit = 0.1;
Chris@759: edgeLimit = limit * 3;
Chris@757: }
Chris@757:
Chris@759: // And we ignore completely the last few frames when upsampling
Chris@1040: int discard = 1 + int(round(readRate / nominalRate));
Chris@759:
Chris@759: int offset = 0;
Chris@759:
Chris@759: if (extension == "aac" || extension == "m4a") {
Chris@759: // our m4a file appears to have a fixed offset of 1024 (at
Chris@759: // file sample rate)
Chris@1040: offset = int(round((1024 / nominalRate) * readRate));
Chris@759: }
Chris@759:
Chris@759: if (extension == "mp3") {
Chris@1296: // ...while mp3s appear to vary. What we're looking for is
Chris@1296: // the first peak of the sinusoid in the first channel
Chris@1296: // (since we may have only the one channel). This should
Chris@1296: // appear at 0.4ms (see AudioTestData.h)
Chris@1296: int expectedPeak = int(0.0004 * readRate);
Chris@1296: // std::cerr << "expectedPeak = " << expectedPeak << std::endl;
Chris@1296: for (int i = 1; i < read; ++i) {
Chris@1296: if (test[i * channels] > 0.8 &&
Chris@1296: test[(i+1) * channels] < test[i * channels]) {
Chris@1296: offset = i - expectedPeak - 1;
Chris@1296: // std::cerr << "actual peak = " << i-1 << std::endl;
Chris@759: break;
Chris@759: }
Chris@759: }
Chris@759: // std::cerr << "offset = " << offset << std::endl;
Chris@759: }
Chris@756:
Chris@756: for (int c = 0; c < channels; ++c) {
Chris@756: float maxdiff = 0.f;
Chris@756: int maxAt = 0;
Chris@756: float totdiff = 0.f;
Chris@1296: for (int i = 0; i < refFrames; ++i) {
Chris@1296: int ix = i + offset;
Chris@1296: if (ix >= read) {
Chris@1296: cerr << "ERROR: audiofile " << audiofile << " reads truncated (read-rate reference frames " << i << " onward are lost)" << endl;
Chris@1296: QVERIFY(ix < read);
Chris@1296: }
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@1296: float diff = fabsf(test[(ix) * channels + c] -
Chris@756: reference[i * channels + c]);
Chris@756: totdiff += diff;
Chris@757: // in edge areas, record this only if it exceeds edgeLimit
Chris@759: if (i < edgeSize || i + edgeSize >= read - offset) {
Chris@968: if (diff > edgeLimit && diff > maxdiff) {
Chris@757: maxdiff = diff;
Chris@757: maxAt = i;
Chris@757: }
Chris@757: } else {
Chris@757: if (diff > maxdiff) {
Chris@757: maxdiff = diff;
Chris@757: maxAt = i;
Chris@757: }
Chris@756: }
Chris@756: }
Chris@1040: float meandiff = totdiff / float(read);
Chris@756: // cerr << "meandiff on channel " << c << ": " << meandiff << endl;
Chris@756: // cerr << "maxdiff on channel " << c << ": " << maxdiff << " at " << maxAt << endl;
Chris@759: if (meandiff >= limit) {
Chris@759: cerr << "ERROR: for audiofile " << audiofile << ": mean diff = " << meandiff << " for channel " << c << endl;
Chris@759: QVERIFY(meandiff < limit);
Chris@759: }
Chris@756: if (maxdiff >= limit) {
Chris@759: cerr << "ERROR: for audiofile " << audiofile << ": max diff = " << maxdiff << " at frame " << maxAt << " of " << read << " on channel " << c << " (mean diff = " << meandiff << ")" << endl;
Chris@756: QVERIFY(maxdiff < limit);
Chris@756: }
Chris@756: }
Chris@756: }
Chris@756: };
Chris@756:
Chris@756: #endif