view data/fileio/test/AudioFileReaderTest.h @ 1008:d9e0e59a1581

When using an aggregate model to pass data to a transform, zero-pad the shorter input to the duration of the longer rather than truncating the longer. (This is better behaviour for e.g. MATCH, and in any case the code was previously truncating incorrectly and ending up with garbage data at the end.)

/* -*- 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);

        int readRate = 48000;

	AudioFileReader *reader =
	    AudioFileReaderFactory::createReader
	    (audioDir + "/" + audiofile, readRate);

        QStringList fileAndExt = audiofile.split(".");
        QStringList bits = fileAndExt[0].split("-");
        QString extension = fileAndExt[1];
        int 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((int)reader->getNativeRate(), nominalRate);
        QCOMPARE((int)reader->getSampleRate(), readRate);

	int channels = reader->getChannelCount();
	AudioTestData tdata(readRate, channels);
	
	float *reference = tdata.getInterleavedData();
        int refFrames = tdata.getFrameCount();
	
	vector<float> test;
	
	// 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).
	reader->getInterleavedFrames(0, refFrames + 5000, test);
	int 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)

	float limit = 0.01;
        float 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 + 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 = (1024 / float(nominalRate)) * readRate;
        }

        if (extension == "mp3") {
            // while mp3s appear to vary
            for (int i = 0; i < read; ++i) {
                bool any = false;
                float thresh = 0.01;
                for (int c = 0; c < channels; ++c) {
                    if (fabsf(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 / 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