view data/fileio/test/AudioFileReaderTest.h @ 1455:ec9e65fcf749

The use of the begin/end pairs here just seems to cause too many rows to be deleted (from the visual representation, not the underlying model). Things apparently work better if we just modify the underlying model and let the change signals percolate back up again. To that end, update the change handlers so as to cover their proper ranges with dataChanged signals.

/* -*- 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 "../WavFileWriter.h"

#include "AudioTestData.h"

#include <cmath>

#include <QObject>
#include <QtTest>
#include <QDir>

#include <iostream>

using namespace std;

class AudioFileReaderTest : public QObject
{
    Q_OBJECT

private:
    QString testDirBase;
    QString audioDir;
    QString diffDir;

public:
    AudioFileReaderTest(QString base) {
        if (base == "") {
            base = "svcore/data/fileio/test";
        }
        testDirBase = base;
        audioDir = base + "/audio";
        diffDir = base + "/diffs";
    }

private:
    const char *strOf(QString s) {
        return strdup(s.toLocal8Bit().data());
    }

    void getFileMetadata(QString filename,
                         QString &extension,
                         sv_samplerate_t &rate,
                         int &channels,
                         int &bitdepth) {

        QStringList fileAndExt = filename.split(".");
        QStringList bits = fileAndExt[0].split("-");

        extension = fileAndExt[1];
        rate = bits[0].toInt();
        channels = bits[1].toInt();
        bitdepth = 16;
        if (bits.length() > 2) {
            bitdepth = bits[2].toInt();
        }
    }
    
    void getExpectedThresholds(QString format,
                               QString filename,
                               bool resampled,
                               bool gapless,
                               bool normalised,
                               double &maxLimit,
                               double &rmsLimit) {

        QString extension;
        sv_samplerate_t fileRate;
        int channels;
        int bitdepth;
        getFileMetadata(filename, extension, fileRate, channels, bitdepth);
        
        if (normalised) {

            if (format == "ogg") {

                // Our ogg is not especially high quality and is
                // actually further from the original if normalised

                maxLimit = 0.1;
                rmsLimit = 0.03;

            } else if (format == "aac") {

                // Terrible performance for this test, load of spill
                // from one channel to the other. I guess they know
                // what they're doing, it's perceptual after all, but
                // it does make this check a bit superfluous, you
                // could probably pass it with a signal that sounds
                // nothing like the original
                maxLimit = 0.2;
                rmsLimit = 0.1;

            } else if (format == "mp3") {

                if (resampled && !gapless) {

                    // We expect worse figures here, because the
                    // combination of uncompensated encoder delay +
                    // resampling results in a fractional delay which
                    // means the decoded signal is slightly out of
                    // phase compared to the test signal

                    maxLimit = 0.1;
                    rmsLimit = 0.05;

                } else {

                    maxLimit = 0.05;
                    rmsLimit = 0.01;
                }

            } else {

                // lossless formats (wav, aiff, flac, apple_lossless)
                
                if (bitdepth >= 16 && !resampled) {
                    maxLimit = 1e-3;
                    rmsLimit = 3e-4;
                } else {
                    maxLimit = 0.01;
                    rmsLimit = 5e-3;
                }
            }
            
        } else { // !normalised
            
            if (format == "ogg") {

                maxLimit = 0.06;
                rmsLimit = 0.03;

            } else if (format == "aac") {

                maxLimit = 0.1;
                rmsLimit = 0.1;

            } else if (format == "mp3") {

                // all mp3 figures are worse when not normalising
                maxLimit = 0.1;
                rmsLimit = 0.05;

            } else {

                // lossless formats (wav, aiff, flac, apple_lossless)
                
                if (bitdepth >= 16 && !resampled) {
                    maxLimit = 1e-3;
                    rmsLimit = 3e-4;
                } else {
                    maxLimit = 0.02;
                    rmsLimit = 0.01;
                }
            }
        }
    }

    QString testName(QString format, QString filename, int rate, bool norm, bool gapless) {
        return QString("%1/%2 at %3%4%5")
            .arg(format)
            .arg(filename)
            .arg(rate)
            .arg(norm ? " normalised": "")
            .arg(gapless ? "" : " non-gapless");
    }

private slots:
    void init()
    {
        if (!QDir(audioDir).exists()) {
            QString cwd = QDir::currentPath();
            SVCERR << "ERROR: Audio test file directory \"" << audioDir << "\" does not exist (cwd = " << cwd << ")" << endl;
            QVERIFY2(QDir(audioDir).exists(), "Audio test file directory not found");
        }
        if (!QDir(diffDir).exists() && !QDir().mkpath(diffDir)) {
            SVCERR << "ERROR: Audio diff directory \"" << diffDir << "\" does not exist and could not be created" << endl;
            QVERIFY2(QDir(diffDir).exists(), "Audio diff directory not found and could not be created");
        }
    }

    void read_data()
    {
        QTest::addColumn<QString>("format");
        QTest::addColumn<QString>("audiofile");
        QTest::addColumn<int>("rate");
        QTest::addColumn<bool>("normalised");
        QTest::addColumn<bool>("gapless");
        QStringList dirs = QDir(audioDir).entryList(QDir::Dirs |
                                                    QDir::NoDotAndDotDot);
        for (QString format: dirs) {
            QStringList files = QDir(QDir(audioDir).filePath(format))
                .entryList(QDir::Files);
            int readRates[] = { 44100, 48000 };
            bool norms[] = { false, true };
            bool gaplesses[] = { true, false };
            foreach (QString filename, files) {
                for (int rate: readRates) {
                    for (bool norm: norms) {
                        for (bool gapless: gaplesses) {

                            if (format != "mp3" && !gapless) {
                                continue;
                            }
                        
                            QString desc = testName
                                (format, filename, rate, norm, gapless);

                            QTest::newRow(strOf(desc))
                                << format << filename << rate << norm << gapless;
                        }
                    }
                }
            }
        }
    }

    void read()
    {
        QFETCH(QString, format);
        QFETCH(QString, audiofile);
        QFETCH(int, rate);
        QFETCH(bool, normalised);
        QFETCH(bool, gapless);

        sv_samplerate_t readRate(rate);
        
//        cerr << "\naudiofile = " << audiofile << endl;

        AudioFileReaderFactory::Parameters params;
        params.targetRate = readRate;
        params.normalisation = (normalised ?
                                AudioFileReaderFactory::Normalisation::Peak :
                                AudioFileReaderFactory::Normalisation::None);
        params.gaplessMode = (gapless ?
                              AudioFileReaderFactory::GaplessMode::Gapless :
                              AudioFileReaderFactory::GaplessMode::Gappy);

        AudioFileReader *reader =
            AudioFileReaderFactory::createReader
            (audioDir + "/" + format + "/" + audiofile, params);
        
        if (!reader) {
#if ( QT_VERSION >= 0x050000 )
            QSKIP("Unsupported file, skipping");
#else
            QSKIP("Unsupported file, skipping", SkipSingle);
#endif
        }

        QString extension;
        sv_samplerate_t fileRate;
        int channels;
        int fileBitdepth;
        getFileMetadata(audiofile, extension, fileRate, channels, fileBitdepth);
        
        QCOMPARE((int)reader->getChannelCount(), channels);
        QCOMPARE(reader->getNativeRate(), fileRate);
        QCOMPARE(reader->getSampleRate(), readRate);

        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).
        floatvec_t test = reader->getInterleavedFrames(0, refFrames + 5000);

        delete reader;
        reader = 0;
        
        sv_frame_t read = test.size() / channels;

        bool perceptual = (extension == "mp3" ||
                           extension == "aac" ||
                           extension == "m4a");
        
        if (perceptual && !gapless) {
            // allow silence at start and end
            QVERIFY(read >= refFrames);
        } else {
            QCOMPARE(read, refFrames);
        }

        bool resampled = readRate != fileRate;
        double maxLimit, rmsLimit;
        getExpectedThresholds(format,
                              audiofile,
                              resampled,
                              gapless,
                              normalised,
                              maxLimit, rmsLimit);
        
        double edgeLimit = maxLimit * 3; // in first or final edgeSize frames
        if (resampled && edgeLimit < 0.1) edgeLimit = 0.1;
        int edgeSize = 100; 

        // And we ignore completely the last few frames when upsampling
        int discard = 1 + int(round(readRate / fileRate));

        int offset = 0;

        if (perceptual) {

            // Look for an initial offset.
            //
            // We know the first channel has a sinusoid in it. It
            // should have a peak at 0.4ms (see AudioTestData.h) but
            // that might have been clipped, which would make it
            // imprecise. We can tell if it's clipped, though, as
            // there will be samples having exactly identical
            // values. So what we look for is the peak if it's not
            // clipped and, if it is, the first zero crossing after
            // the peak, which should be at 0.8ms.

            int expectedPeak = int(0.0004 * readRate);
            int expectedZC = int(0.0008 * readRate);
            bool foundPeak = false;
            for (int i = 1; i+1 < read; ++i) {
                float prevSample = test[(i-1) * channels];
                float thisSample = test[i * channels];
                float nextSample = test[(i+1) * channels];
                if (thisSample > 0.8 && nextSample < thisSample) {
                    foundPeak = true;
                    if (thisSample > prevSample) {
                        // not clipped
                        offset = i - expectedPeak - 1;
                        break;
                    }
                }
                if (foundPeak && (thisSample >= 0.0 && nextSample < 0.0)) {
//                    cerr << "thisSample = " << thisSample << ", nextSample = "
//                         << nextSample << endl;
                    offset = i - expectedZC - 1;
                    break;
                }
            }

//            int fileRateEquivalent = int((offset / readRate) * fileRate);
//            std::cerr << "offset = " << offset << std::endl;
//            std::cerr << "at file rate would be " << fileRateEquivalent << std::endl;

            // Previously our m4a test file had a fixed offset of 1024
            // at the file sample rate -- this may be because it was
            // produced by FAAC which did not write in the delay as
            // metadata? We now have an m4a produced by Core Audio
            // which gives a 0 offset. What to do...

            // Anyway, mp3s should have 0 offset in gapless mode and
            // "something else" otherwise.
            
            if (gapless) {
                if (format == "aac") {
                    // ouch!
                    if (offset == -1) offset = 0;
                }
                QCOMPARE(offset, 0);
            }
        }

        {
            // Write the diff file now, so that it's already been written
            // even if the comparison fails. We aren't checking anything
            // here except as necessary to avoid buffer overruns etc

            QString diffFile =
                testName(format, audiofile, rate, normalised, gapless);
            diffFile.replace("/", "_");
            diffFile.replace(".", "_");
            diffFile.replace(" ", "_");
            diffFile += ".wav";
            diffFile = QDir(diffDir).filePath(diffFile);
            WavFileWriter diffWriter(diffFile, readRate, channels,
                                     WavFileWriter::WriteToTemporary);
            QVERIFY(diffWriter.isOK());

            vector<vector<float>> diffs(channels);
            for (int c = 0; c < channels; ++c) {
                for (int i = 0; i < refFrames; ++i) {
                    int ix = i + offset;
                    if (ix < read) {
                        float signeddiff =
                            test[ix * channels + c] -
                            reference[i * channels + c];
                        diffs[c].push_back(signeddiff);
                    }
                }
            }
            float **ptrs = new float*[channels];
            for (int c = 0; c < channels; ++c) {
                ptrs[c] = diffs[c].data();
            }
            diffWriter.writeSamples(ptrs, refFrames);
            delete[] ptrs;
        }
            
        for (int c = 0; c < channels; ++c) {

            double maxDiff = 0.0;
            double totalDiff = 0.0;
            double totalSqrDiff = 0.0;
            int maxIndex = 0;

            for (int i = 0; i < refFrames; ++i) {
                int ix = i + offset;
                if (ix >= read) {
                    SVCERR << "ERROR: audiofile " << audiofile << " reads truncated (read-rate reference frames " << i << " onward, of " << refFrames << ", are lost)" << endl;
                    QVERIFY(ix < read);
                }

                if (ix + discard >= read) {
                    // we forgive the very edge samples when
                    // resampling (discard > 0)
                    continue;
                }
                
                double diff = fabs(test[ix * channels + c] -
                                   reference[i * channels + c]);

                totalDiff += diff;
                totalSqrDiff += diff * diff;
                
                // in edge areas, record this only if it exceeds edgeLimit
                if (i < edgeSize || i + edgeSize >= refFrames) {
                    if (diff > edgeLimit && diff > maxDiff) {
                        maxDiff = diff;
                        maxIndex = i;
                    }
                } else {
                    if (diff > maxDiff) {
                        maxDiff = diff;
                        maxIndex = i;
                    }
                }
            }
                
            double meanDiff = totalDiff / double(refFrames);
            double rmsDiff = sqrt(totalSqrDiff / double(refFrames));

            /*
        cerr << "channel " << c << ": mean diff " << meanDiff << endl;
            cerr << "channel " << c << ":  rms diff " << rmsDiff << endl;
            cerr << "channel " << c << ":  max diff " << maxDiff << " at " << maxIndex << endl;
            */            
            if (rmsDiff >= rmsLimit) {
                SVCERR << "ERROR: for audiofile " << audiofile << ": RMS diff = " << rmsDiff << " for channel " << c << " (limit = " << rmsLimit << ")" << endl;
                QVERIFY(rmsDiff < rmsLimit);
            }
            if (maxDiff >= maxLimit) {
                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;
                QVERIFY(maxDiff < maxLimit);
            }

            // and check for spurious material at end
            
            for (sv_frame_t i = refFrames; i + offset < read; ++i) {
                sv_frame_t ix = i + offset;
                float quiet = 0.1f; //!!! allow some ringing - but let's come back to this, it should tail off
                float mag = fabsf(test[ix * channels + c]);
                if (mag > quiet) {
                    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;
                    QVERIFY(mag < quiet);
                }
            }
        }
    }
};

#endif