Files
lmms/include/SampleBuffer.h
Martin Pavelek 6f8c6dba82 Alias-free oscillators (#5826)
Add a band-limited, alias-free wavetable oscillator option to the
`Oscillator` class. Use it by default for Triple Oscillator.

Savefiles which do not have this feature enabled (e.g. old
savefiles) will be loaded without this feature to keep the sound
consistent.

Original author: @curlymorphic.
Fixed: @he29-net.
2021-07-04 13:14:59 +02:00

356 lines
7.3 KiB
C++

/*
* SampleBuffer.h - container-class SampleBuffer
*
* Copyright (c) 2005-2014 Tobias Doerffel <tobydox/at/users.sourceforge.net>
*
* This file is part of LMMS - https://lmms.io
*
* 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.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public
* License along with this program (see COPYING); if not, write to the
* Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
* Boston, MA 02110-1301 USA.
*
*/
#ifndef SAMPLE_BUFFER_H
#define SAMPLE_BUFFER_H
#include <memory>
#include <QtCore/QReadWriteLock>
#include <QtCore/QObject>
#include <samplerate.h>
#include "lmms_export.h"
#include "interpolation.h"
#include "lmms_basics.h"
#include "lmms_math.h"
#include "shared_object.h"
#include "OscillatorConstants.h"
#include "MemoryManager.h"
class QPainter;
class QRect;
// values for buffer margins, used for various libsamplerate interpolation modes
// the array positions correspond to the converter_type parameter values in libsamplerate
// if there appears problems with playback on some interpolation mode, then the value for that mode
// may need to be higher - conversely, to optimize, some may work with lower values
const f_cnt_t MARGIN[] = { 64, 64, 64, 4, 4 };
class LMMS_EXPORT SampleBuffer : public QObject, public sharedObject
{
Q_OBJECT
MM_OPERATORS
public:
enum LoopMode {
LoopOff = 0,
LoopOn,
LoopPingPong
};
class LMMS_EXPORT handleState
{
MM_OPERATORS
public:
handleState(bool varyingPitch = false, int interpolationMode = SRC_LINEAR);
virtual ~handleState();
const f_cnt_t frameIndex() const
{
return m_frameIndex;
}
void setFrameIndex(f_cnt_t index)
{
m_frameIndex = index;
}
bool isBackwards() const
{
return m_isBackwards;
}
void setBackwards(bool backwards)
{
m_isBackwards = backwards;
}
int interpolationMode() const
{
return m_interpolationMode;
}
private:
f_cnt_t m_frameIndex;
const bool m_varyingPitch;
bool m_isBackwards;
SRC_STATE * m_resamplingData;
int m_interpolationMode;
friend class SampleBuffer;
} ;
SampleBuffer();
// constructor which either loads sample _audio_file or decodes
// base64-data out of string
SampleBuffer(const QString & audioFile, bool isBase64Data = false);
SampleBuffer(const sampleFrame * data, const f_cnt_t frames);
explicit SampleBuffer(const f_cnt_t frames);
SampleBuffer(const SampleBuffer & orig);
friend void swap(SampleBuffer & first, SampleBuffer & second) noexcept;
SampleBuffer& operator= (const SampleBuffer that);
virtual ~SampleBuffer();
bool play(
sampleFrame * ab,
handleState * state,
const fpp_t frames,
const float freq,
const LoopMode loopMode = LoopOff
);
void visualize(
QPainter & p,
const QRect & dr,
const QRect & clip,
f_cnt_t fromFrame = 0,
f_cnt_t toFrame = 0
);
inline void visualize(
QPainter & p,
const QRect & dr,
f_cnt_t fromFrame = 0,
f_cnt_t toFrame = 0
)
{
visualize(p, dr, dr, fromFrame, toFrame);
}
inline const QString & audioFile() const
{
return m_audioFile;
}
inline f_cnt_t startFrame() const
{
return m_startFrame;
}
inline f_cnt_t endFrame() const
{
return m_endFrame;
}
inline f_cnt_t loopStartFrame() const
{
return m_loopStartFrame;
}
inline f_cnt_t loopEndFrame() const
{
return m_loopEndFrame;
}
void setLoopStartFrame(f_cnt_t start)
{
m_loopStartFrame = start;
}
void setLoopEndFrame(f_cnt_t end)
{
m_loopEndFrame = end;
}
void setAllPointFrames(
f_cnt_t start,
f_cnt_t end,
f_cnt_t loopStart,
f_cnt_t loopEnd
)
{
m_startFrame = start;
m_endFrame = end;
m_loopStartFrame = loopStart;
m_loopEndFrame = loopEnd;
}
inline f_cnt_t frames() const
{
return m_frames;
}
inline float amplification() const
{
return m_amplification;
}
inline bool reversed() const
{
return m_reversed;
}
inline float frequency() const
{
return m_frequency;
}
sample_rate_t sampleRate() const
{
return m_sampleRate;
}
int sampleLength() const
{
return double(m_endFrame - m_startFrame) / m_sampleRate * 1000;
}
inline void setFrequency(float freq)
{
m_frequency = freq;
}
inline void setSampleRate(sample_rate_t rate)
{
m_sampleRate = rate;
}
inline const sampleFrame * data() const
{
return m_data;
}
QString openAudioFile() const;
QString openAndSetAudioFile();
QString openAndSetWaveformFile();
QString & toBase64(QString & dst) const;
// protect calls from the GUI to this function with dataReadLock() and
// dataUnlock()
SampleBuffer * resample(const sample_rate_t srcSR, const sample_rate_t dstSR);
void normalizeSampleRate(const sample_rate_t srcSR, bool keepSettings = false);
// protect calls from the GUI to this function with dataReadLock() and
// dataUnlock(), out of loops for efficiency
inline sample_t userWaveSample(const float sample) const
{
f_cnt_t frames = m_frames;
sampleFrame * data = m_data;
const float frame = sample * frames;
f_cnt_t f1 = static_cast<f_cnt_t>(frame) % frames;
if (f1 < 0)
{
f1 += frames;
}
return linearInterpolate(data[f1][0], data[(f1 + 1) % frames][0], fraction(frame));
}
void dataReadLock()
{
m_varLock.lockForRead();
}
void dataUnlock()
{
m_varLock.unlock();
}
std::unique_ptr<OscillatorConstants::waveform_t> m_userAntiAliasWaveTable;
public slots:
void setAudioFile(const QString & audioFile);
void loadFromBase64(const QString & data);
void setStartFrame(const f_cnt_t s);
void setEndFrame(const f_cnt_t e);
void setAmplification(float a);
void setReversed(bool on);
void sampleRateChanged();
private:
static sample_rate_t mixerSampleRate();
void update(bool keepSettings = false);
void convertIntToFloat(int_sample_t * & ibuf, f_cnt_t frames, int channels);
void directFloatWrite(sample_t * & fbuf, f_cnt_t frames, int channels);
f_cnt_t decodeSampleSF(
QString fileName,
sample_t * & buf,
ch_cnt_t & channels,
sample_rate_t & samplerate
);
#ifdef LMMS_HAVE_OGGVORBIS
f_cnt_t decodeSampleOGGVorbis(
QString fileName,
int_sample_t * & buf,
ch_cnt_t & channels,
sample_rate_t & samplerate
);
#endif
f_cnt_t decodeSampleDS(
QString fileName,
int_sample_t * & buf,
ch_cnt_t & channels,
sample_rate_t & samplerate
);
QString m_audioFile;
sampleFrame * m_origData;
f_cnt_t m_origFrames;
sampleFrame * m_data;
mutable QReadWriteLock m_varLock;
f_cnt_t m_frames;
f_cnt_t m_startFrame;
f_cnt_t m_endFrame;
f_cnt_t m_loopStartFrame;
f_cnt_t m_loopEndFrame;
float m_amplification;
bool m_reversed;
float m_frequency;
sample_rate_t m_sampleRate;
sampleFrame * getSampleFragment(
f_cnt_t index,
f_cnt_t frames,
LoopMode loopMode,
sampleFrame * * tmp,
bool * backwards,
f_cnt_t loopStart,
f_cnt_t loopEnd,
f_cnt_t end
) const;
f_cnt_t getLoopedIndex(f_cnt_t index, f_cnt_t startf, f_cnt_t endf) const;
f_cnt_t getPingPongIndex(f_cnt_t index, f_cnt_t startf, f_cnt_t endf) const;
signals:
void sampleUpdated();
} ;
#endif