Files
lmms/include/SampleBuffer.h
Shmuel H 37f0c3f33d SampleBuffer -> Use processingSampleRate.
SampleBuffer was using baseSampleRate as the default samplerate instead of the actual processingSampleRate.
2019-06-02 20:48:58 +03:00

318 lines
7.2 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 <QtCore/QReadWriteLock>
#include <QtCore/QObject>
#include <samplerate.h>
#include "export.h"
#include "interpolation.h"
#include "lmms_basics.h"
#include "lmms_math.h"
#include "shared_object.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 EXPORT SampleBuffer : public QObject, public sharedObject
{
Q_OBJECT
MM_OPERATORS
public:
enum LoopMode {
LoopOff = 0,
LoopOn,
LoopPingPong
};
class EXPORT handleState
{
MM_OPERATORS
public:
handleState( bool _varying_pitch = false, int interpolation_mode = 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;
} ;
// constructor which either loads sample _audio_file or decodes
// base64-data out of string
SampleBuffer( const QString & _audio_file = QString(),
bool _is_base64_data = false );
SampleBuffer( const sampleFrame * _data, const f_cnt_t _frames );
SampleBuffer( const f_cnt_t _frames );
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 _from_frame = 0, f_cnt_t _to_frame = 0 );
inline void visualize( QPainter & _p, const QRect & _dr, f_cnt_t _from_frame = 0, f_cnt_t _to_frame = 0 )
{
visualize( _p, _dr, _dr, _from_frame, _to_frame );
}
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 _src_sr,
const sample_rate_t _dst_sr );
void normalizeSampleRate( const sample_rate_t _src_sr,
bool _keep_settings = 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();
}
static QString tryToMakeRelative( const QString & _file );
static QString tryToMakeAbsolute(const QString & file);
public slots:
void setAudioFile( const QString & _audio_file );
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 _keep_settings = 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 _f, sample_t * & _buf,
ch_cnt_t & _channels,
sample_rate_t & _sample_rate );
#ifdef LMMS_HAVE_OGGVORBIS
f_cnt_t decodeSampleOGGVorbis( QString _f, int_sample_t * & _buf,
ch_cnt_t & _channels,
sample_rate_t & _sample_rate );
#endif
f_cnt_t decodeSampleDS( QString _f, int_sample_t * & _buf,
ch_cnt_t & _channels,
sample_rate_t & _sample_rate );
QString m_audioFile;
sampleFrame * m_origData;
f_cnt_t m_origFrames;
sampleFrame * m_data;
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