Mercurial > hg > svcore
view data/fileio/test/AudioFileReaderTest.h @ 1104:393134235fa0 simple-fft-model
Two-channel test
author | Chris Cannam |
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date | Tue, 16 Jun 2015 20:32:44 +0100 |
parents | 843f67be0ed9 |
children | abfc498c52bc |
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/* -*- c-basic-offset: 4 indent-tabs-mode: nil -*- vi:set ts=8 sts=4 sw=4: */ /* Sonic Visualiser An audio file viewer and annotation editor. Centre for Digital Music, Queen Mary, University of London. This file copyright 2013 Chris Cannam. This program is free software; you can redistribute it and/or modify it under the terms of the GNU General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. See the file COPYING included with this distribution for more information. */ #ifndef TEST_AUDIO_FILE_READER_H #define TEST_AUDIO_FILE_READER_H #include "../AudioFileReaderFactory.h" #include "../AudioFileReader.h" #include "AudioTestData.h" #include <cmath> #include <QObject> #include <QtTest> #include <QDir> #include <iostream> using namespace std; static QString audioDir = "testfiles"; class AudioFileReaderTest : public QObject { Q_OBJECT const char *strOf(QString s) { return strdup(s.toLocal8Bit().data()); } private slots: void init() { if (!QDir(audioDir).exists()) { cerr << "ERROR: Audio test file directory \"" << audioDir << "\" does not exist" << endl; QVERIFY2(QDir(audioDir).exists(), "Audio test file directory not found"); } } void read_data() { QTest::addColumn<QString>("audiofile"); QStringList files = QDir(audioDir).entryList(QDir::Files); foreach (QString filename, files) { QTest::newRow(strOf(filename)) << filename; } } void read() { QFETCH(QString, audiofile); sv_samplerate_t readRate = 48000; AudioFileReader *reader = AudioFileReaderFactory::createReader (audioDir + "/" + audiofile, readRate); QStringList fileAndExt = audiofile.split("."); QStringList bits = fileAndExt[0].split("-"); QString extension = fileAndExt[1]; sv_samplerate_t nominalRate = bits[0].toInt(); int nominalChannels = bits[1].toInt(); int nominalDepth = 16; if (bits.length() > 2) nominalDepth = bits[2].toInt(); if (!reader) { #if ( QT_VERSION >= 0x050000 ) QSKIP("Unsupported file, skipping"); #else QSKIP("Unsupported file, skipping", SkipSingle); #endif } QCOMPARE((int)reader->getChannelCount(), nominalChannels); QCOMPARE(reader->getNativeRate(), nominalRate); QCOMPARE(reader->getSampleRate(), readRate); int channels = reader->getChannelCount(); AudioTestData tdata(readRate, channels); float *reference = tdata.getInterleavedData(); sv_frame_t refFrames = tdata.getFrameCount(); // The reader should give us exactly the expected number of // frames, except for mp3/aac files. We ask for quite a lot // more, though, so we can (a) check that we only get the // expected number back (if this is not mp3/aac) or (b) take // into account silence at beginning and end (if it is). vector<float> test = reader->getInterleavedFrames(0, refFrames + 5000); sv_frame_t read = test.size() / channels; if (extension == "mp3" || extension == "aac" || extension == "m4a") { // mp3s and aacs can have silence at start and end QVERIFY(read >= refFrames); } else { QCOMPARE(read, refFrames); } // Our limits are pretty relaxed -- we're not testing decoder // or resampler quality here, just whether the results are // plainly wrong (e.g. at wrong samplerate or with an offset) double limit = 0.01; double edgeLimit = limit * 10; // in first or final edgeSize frames int edgeSize = 100; if (nominalDepth < 16) { limit = 0.02; } if (extension == "ogg" || extension == "mp3" || extension == "aac" || extension == "m4a") { limit = 0.2; edgeLimit = limit * 3; } // And we ignore completely the last few frames when upsampling int discard = 1 + int(round(readRate / nominalRate)); int offset = 0; if (extension == "aac" || extension == "m4a") { // our m4a file appears to have a fixed offset of 1024 (at // file sample rate) offset = int(round((1024 / nominalRate) * readRate)); } if (extension == "mp3") { // while mp3s appear to vary for (int i = 0; i < read; ++i) { bool any = false; double thresh = 0.01; for (int c = 0; c < channels; ++c) { if (fabs(test[i * channels + c]) > thresh) { any = true; break; } } if (any) { offset = i; break; } } // std::cerr << "offset = " << offset << std::endl; } for (int c = 0; c < channels; ++c) { float maxdiff = 0.f; int maxAt = 0; float totdiff = 0.f; for (int i = 0; i < read - offset - discard && i < refFrames; ++i) { float diff = fabsf(test[(i + offset) * channels + c] - reference[i * channels + c]); totdiff += diff; // in edge areas, record this only if it exceeds edgeLimit if (i < edgeSize || i + edgeSize >= read - offset) { if (diff > edgeLimit && diff > maxdiff) { maxdiff = diff; maxAt = i; } } else { if (diff > maxdiff) { maxdiff = diff; maxAt = i; } } } float meandiff = totdiff / float(read); // cerr << "meandiff on channel " << c << ": " << meandiff << endl; // cerr << "maxdiff on channel " << c << ": " << maxdiff << " at " << maxAt << endl; if (meandiff >= limit) { cerr << "ERROR: for audiofile " << audiofile << ": mean diff = " << meandiff << " for channel " << c << endl; QVERIFY(meandiff < limit); } if (maxdiff >= limit) { cerr << "ERROR: for audiofile " << audiofile << ": max diff = " << maxdiff << " at frame " << maxAt << " of " << read << " on channel " << c << " (mean diff = " << meandiff << ")" << endl; QVERIFY(maxdiff < limit); } } } }; #endif