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Remove old SampleBuffer

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This commit is contained in:
sakertooth
2023-08-27 23:31:36 -04:00
parent 0017f66629
commit 53728ce70f
29 changed files with 203 additions and 2152 deletions
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4
include/AudioSampleRecorder.h
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@@ -34,7 +34,7 @@
namespace lmms
{
class SampleBuffer2;
class SampleBuffer;
class AudioSampleRecorder : public AudioDevice
@@ -44,7 +44,7 @@ public:
~AudioSampleRecorder() override;
f_cnt_t framesRecorded() const;
void createSampleBuffer(SampleBuffer2** sampleBuffer);
void createSampleBuffer(SampleBuffer** sampleBuffer);
private:

2
include/EnvelopeAndLfoParameters.h
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@@ -167,7 +167,7 @@ private:
sample_t * m_lfoShapeData;
sample_t m_random;
bool m_bad_lfoShapeData;
SampleBuffer m_userWave;
std::shared_ptr<const SampleBuffer> m_userWave;
enum class LfoShape
{

2
include/LfoController.h
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@@ -86,7 +86,7 @@ protected:
sample_t (*m_sampleFunction)( const float );
private:
std::shared_ptr<const SampleBuffer2> m_userDefSampleBuffer;
std::shared_ptr<const SampleBuffer> m_userDefSampleBuffer;
protected slots:
void updatePhase();

10
include/Oscillator.h
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@@ -36,7 +36,7 @@
#include "lmmsconfig.h"
#include "AudioEngine.h"
#include "OscillatorConstants.h"
#include "SampleBuffer2.h"
#include "SampleBuffer.h"
namespace lmms
{
@@ -91,14 +91,14 @@ public:
static void waveTableInit();
static void destroyFFTPlans();
static std::unique_ptr<OscillatorConstants::waveform_t> generateAntiAliasUserWaveTable(const SampleBuffer2 *sampleBuffer);
static std::unique_ptr<OscillatorConstants::waveform_t> generateAntiAliasUserWaveTable(const SampleBuffer *sampleBuffer);
inline void setUseWaveTable(bool n)
{
m_useWaveTable = n;
}
inline void setUserWave(std::shared_ptr<const SampleBuffer2> _wave)
inline void setUserWave(std::shared_ptr<const SampleBuffer> _wave)
{
m_userWave = _wave;
}
@@ -164,7 +164,7 @@ public:
return 1.0f - fast_rand() * 2.0f / FAST_RAND_MAX;
}
static inline sample_t userWaveSample(const SampleBuffer2* buffer, const float sample)
static inline sample_t userWaveSample(const SampleBuffer* buffer, const float sample)
{
if (buffer == nullptr || buffer->size() == 0) { return 0; }
const auto frames = buffer->size();
@@ -256,7 +256,7 @@ private:
Oscillator * m_subOsc;
float m_phaseOffset;
float m_phase;
std::shared_ptr<const SampleBuffer2> m_userWave;
std::shared_ptr<const SampleBuffer> m_userWave;
std::shared_ptr<const OscillatorConstants::waveform_t> m_userAntiAliasWaveTable;
bool m_useWaveTable;
// There are many update*() variants; the modulator flag is stored as a member variable to avoid

10
include/Sample.h
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@@ -1,5 +1,5 @@
/*
* Sample.h - State for container-class SampleBuffer2
* Sample.h - State for container-class SampleBuffer
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
@@ -29,7 +29,7 @@
#include <memory>
#include "Note.h"
#include "SampleBuffer2.h"
#include "SampleBuffer.h"
#include "lmms_export.h"
#ifdef __MINGW32__
@@ -88,7 +88,7 @@ public:
Sample(const QString& audioFile);
Sample(const QByteArray& base64, int sampleRate = Engine::audioEngine()->processingSampleRate());
Sample(const sampleFrame* data, int numFrames, int sampleRate = Engine::audioEngine()->processingSampleRate());
Sample(std::shared_ptr<const SampleBuffer2> buffer);
Sample(std::shared_ptr<const SampleBuffer> buffer);
Sample(const Sample& other);
Sample(Sample&& other) noexcept;
@@ -107,7 +107,7 @@ public:
auto toBase64() const -> QString;
auto buffer() const -> std::shared_ptr<const SampleBuffer2>;
auto buffer() const -> std::shared_ptr<const SampleBuffer>;
auto startFrame() const -> int;
auto endFrame() const -> int;
auto loopStartFrame() const -> int;
@@ -144,7 +144,7 @@ private:
auto amplifySampleRange(sampleFrame* src, int numFrames) const -> void;
private:
std::shared_ptr<const SampleBuffer2> m_buffer = std::make_shared<SampleBuffer2>();
std::shared_ptr<const SampleBuffer> m_buffer = std::make_shared<SampleBuffer>();
int m_startFrame = 0;
int m_endFrame = 0;
int m_loopStartFrame = 0;

351
include/SampleBuffer.h
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@@ -25,333 +25,58 @@
#ifndef LMMS_SAMPLE_BUFFER_H
#define LMMS_SAMPLE_BUFFER_H
#include <memory>
#include <QReadWriteLock>
#include <QObject>
#include <QByteArray>
#include <QString>
#include <optional>
#include <samplerate.h>
#include <vector>
#include "lmms_export.h"
#include "interpolation.h"
#include "AudioEngine.h"
#include "Engine.h"
#include "lmms_basics.h"
#include "lmms_math.h"
#include "shared_object.h"
#include "OscillatorConstants.h"
#include "MemoryManager.h"
#include "lmms_export.h"
class QPainter;
class QRect;
namespace lmms
namespace lmms {
class LMMS_EXPORT SampleBuffer
{
// 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 class LoopMode {
Off = 0,
On,
PingPong
};
class LMMS_EXPORT handleState
{
MM_OPERATORS
public:
handleState(bool varyingPitch = false, int interpolationMode = SRC_LINEAR);
virtual ~handleState();
using value_type = sampleFrame;
using reference = sampleFrame&;
using const_iterator = std::vector<sampleFrame>::const_iterator;
using const_reverse_iterator = std::vector<sampleFrame>::const_reverse_iterator;
using difference_type = std::vector<sampleFrame>::difference_type;
using size_type = std::vector<sampleFrame>::size_type;
const f_cnt_t frameIndex() const
{
return m_frameIndex;
}
SampleBuffer() = default;
SampleBuffer(const QString& audioFile);
SampleBuffer(const QByteArray& base64Data, int sampleRate);
SampleBuffer(
const sampleFrame* data, int numFrames, int sampleRate = Engine::audioEngine()->processingSampleRate());
void setFrameIndex(f_cnt_t index)
{
m_frameIndex = index;
}
friend void swap(SampleBuffer& first, SampleBuffer& second) noexcept;
auto toBase64() const -> QString;
bool isBackwards() const
{
return m_isBackwards;
}
auto audioFile() const -> QString;
auto sampleRate() const -> sample_rate_t;
void setBackwards(bool backwards)
{
m_isBackwards = backwards;
}
auto begin() const -> const_iterator;
auto end() const -> const_iterator;
auto rbegin() const -> const_reverse_iterator;
auto rend() const -> const_reverse_iterator;
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);
~SampleBuffer() override;
bool play(
sampleFrame * ab,
handleState * state,
const fpp_t frames,
const float freq,
const LoopMode loopMode = LoopMode::Off
);
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 lmms::f_cnt_t s);
void setEndFrame(const lmms::f_cnt_t e);
void setAmplification(float a);
void setReversed(bool on);
void sampleRateChanged();
auto data() const -> const sampleFrame*;
auto size() const -> size_type;
bool empty() const;
private:
static sample_rate_t audioEngineSampleRate();
void decodeSampleSF(const QString& fileName);
void decodeSampleDS(const QString& fileName);
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();
} ;
private:
std::vector<sampleFrame> m_data;
std::optional<QString> m_audioFile;
int m_sampleRate = 0;
};
} // namespace lmms

83
include/SampleBuffer2.h
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@@ -1,83 +0,0 @@
/*
* SampleBuffer2.h - container-class SampleBuffer2
*
* 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 LMMS_SAMPLE_BUFFER2_H
#define LMMS_SAMPLE_BUFFER2_H
#include <QByteArray>
#include <QString>
#include <optional>
#include <samplerate.h>
#include <vector>
#include "AudioEngine.h"
#include "Engine.h"
#include "lmms_basics.h"
#include "lmms_export.h"
namespace lmms {
class LMMS_EXPORT SampleBuffer2
{
public:
using value_type = sampleFrame;
using reference = sampleFrame&;
using const_iterator = std::vector<sampleFrame>::const_iterator;
using const_reverse_iterator = std::vector<sampleFrame>::const_reverse_iterator;
using difference_type = std::vector<sampleFrame>::difference_type;
using size_type = std::vector<sampleFrame>::size_type;
SampleBuffer2() = default;
SampleBuffer2(const QString& audioFile);
SampleBuffer2(const QByteArray& base64Data, int sampleRate);
SampleBuffer2(
const sampleFrame* data, int numFrames, int sampleRate = Engine::audioEngine()->processingSampleRate());
friend void swap(SampleBuffer2& first, SampleBuffer2& second) noexcept;
auto toBase64() const -> QString;
auto audioFile() const -> QString;
auto sampleRate() const -> sample_rate_t;
auto begin() const -> const_iterator;
auto end() const -> const_iterator;
auto rbegin() const -> const_reverse_iterator;
auto rend() const -> const_reverse_iterator;
auto data() const -> const sampleFrame*;
auto size() const -> size_type;
bool empty() const;
private:
void decodeSampleSF(const QString& fileName);
void decodeSampleDS(const QString& fileName);
private:
std::vector<sampleFrame> m_data;
std::optional<QString> m_audioFile;
int m_sampleRate = 0;
};
} // namespace lmms
#endif // LMMS_SAMPLE_BUFFER2_H

2
include/SampleClip.h
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@@ -78,7 +78,7 @@ public:
void setIsPlaying(bool isPlaying);
public slots:
void setSampleBuffer(SampleBuffer2* sb);
void setSampleBuffer(SampleBuffer* sb);
void setSampleFile( const QString & _sf );
void updateLength();
void toggleRecord();

6
include/SampleLoader.h
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@@ -28,7 +28,7 @@
#include <QString>
#include <memory>
#include "SampleBuffer2.h"
#include "SampleBuffer.h"
#include "lmms_export.h"
namespace lmms::gui {
@@ -37,8 +37,8 @@ class LMMS_EXPORT SampleLoader
public:
static QString openAudioFile(const QString& previousFile = "");
static QString openWaveformFile(const QString& previousFile = "");
static std::unique_ptr<SampleBuffer2> createBufferFromFile(const QString& filePath);
static std::unique_ptr<SampleBuffer2> createBufferFromBase64(const QString& base64, int sampleRate = Engine::audioEngine()->processingSampleRate());
static std::unique_ptr<SampleBuffer> createBufferFromFile(const QString& filePath);
static std::unique_ptr<SampleBuffer> createBufferFromBase64(const QString& base64, int sampleRate = Engine::audioEngine()->processingSampleRate());
private:
static void displayError(const QString& message);
};

4
include/SampleRecordHandle.h
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@@ -29,7 +29,7 @@
#include <QPair>
#include "PlayHandle.h"
#include "SampleBuffer2.h"
#include "SampleBuffer.h"
#include "TimePos.h"
namespace lmms
@@ -54,7 +54,7 @@ public:
bool isFromTrack( const Track * _track ) const override;
f_cnt_t framesRecorded() const;
void createSampleBuffer(SampleBuffer2** _sample_buf);
void createSampleBuffer(SampleBuffer** _sample_buf);
private: