view data/fileio/test/AudioFileReaderTest.h @ 1196:c7b9c902642f spectrogram-minor-refactor

Fix threshold in spectrogram -- it wasn't working in the last release.
There is a new protocol for this. Formerly the threshold parameter had a
range from -50dB to 0 with the default at -50, and -50 treated internally
as "no threshold". However, there was a hardcoded, hidden internal threshold
for spectrogram colour mapping at -80dB with anything below this being rounded
to zero. Now the threshold parameter has range -81 to -1 with the default
at -80, -81 is treated internally as "no threshold", and there is no hidden
internal threshold. So the default behaviour is the same as before, an
effective -80dB threshold, but it is now possible to change this in both
directions. Sessions reloaded from prior versions may look slightly different
because, if the session says there should be no threshold, there will now
actually be no threshold instead of having the hidden internal one.

Still need to do something in the UI to make it apparent that the -81dB
setting removes the threshold entirely. This is at least no worse than the
previous, also obscured, magic -50dB setting.

/* -*- 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