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Refactor `SampleBuffer` (#6610)

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* Add refactored SampleBuffer

* Add Sample

* Add SampleLoader

* Integrate changes into AudioSampleRecorder

* Integrate changes into Oscillator

* Integrate changes into SampleClip/SamplePlayHandle

* Integrate changes into Graph

* Remove SampleBuffer include from SampleClipView

* Integrate changes into Patman

* Reduce indirection to sample buffer from Sample

* Integrate changes into AudioFileProcessor

* Remove old SampleBuffer

* Include memory header in TripleOscillator

* Include memory header in Oscillator

* Use atomic_load within SampleClip::sample

* Include memory header in EnvelopeAndLfoParameters

* Use std::atomic_load for most calls to Oscillator::userWaveSample

* Revert accidental change on SamplePlayHandle L.111

* Check if audio file is empty before loading

* Add asserts to Sample

* Add cassert include within Sample

* Adjust assert expressions in Sample

* Remove use of shared ownership for Sample
Sample does not need to be wrapped around a std::shared_ptr.
This was to work with the audio thread, but the audio thread
can instead have their own Sample separate from the UI's Sample,
so changes to the UI's Sample would not leave the audio worker thread
using freed data if it had pointed to it.

* Use ArrayVector in Sample

* Enforce std::atomic_load for users of std::shared_ptr<const SampleBuffer>

* Use requestChangesGuard in ClipView::remove
Fixes data race when deleting SampleClip

* Revert only formatting changes

* Update ClipView::remove comment

* Revert "Remove use of shared ownership for Sample"

This reverts commit 1d452331d1.
In some cases, you can infact do away with shared ownership
on Sample if there are no writes being made to either of them,
but to make sure changes are reflected to the object in cases
where writes do happen, they should work with the same one.

* Fix heap-use-after-free in Track::loadSettings

* Remove m_buffer asserts

* Refactor play functionality (again)
The responsibility of resampling the buffer
and moving the frame index is now in Sample::play, allowing the removal
of both playSampleRangeLoop and playSampleRangePingPong.

* Change copyright

* Cast processingSampleRate to float
Fixes division by zero error

* Update include/SampleLoader.h

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Update include/SampleLoader.h

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Format SampleLoader.h

* Remove SampleBuffer.h include in SampleRecordHandle.h

* Update src/core/Oscillator.cpp

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Use typeInfo<float> for float equality comparison

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Use std::min in Sample::visualize

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Move in result to m_data

* Use if block in playSampleRange

* Pass in unique_ptr to SampleClip::setSampleBuffer

* Return const QString& from SampleBuffer::audioFile

* Do not pass in unique_ptr by r-value reference

* Use isEmpty() within SampleClipView::updateSample

* Remove use of atomic_store and atomic_load

* Remove ArrayVector comment

* Use array specialization for unique_ptr when managing DrumSynth data
Also made it so that we don't create result
before checking if we failed to decode the file,
potentially saving us an allocation.

* Don't manually delete Clip if it has a Track

* Clean up generateAntiAliasUserWaveTable function
Also, make it so that we actually call this function
when necessary in TripleOscillator.

* Set user wave, even when value is empty
If the value or file is empty, I think showing a
error popup here is ideal.

* Remove whitespace in EnvelopeAndLfoParameters.cpp L#121

* Fix error in c5f7ccba49
We still have to delete the Clip's, or else we would just be eating
up memory. But we should first make sure that the Track's no longer
see this Clip in their m_clips vector. This has to happen
as it's own operation because we have to wait for the audio thread(s)
first. This would ensure that Track's do not create
PlayHandle's that would refer to a Clip that is currently
being destroyed. After that, then we call deleteLater on the Clip.

* Convert std::shared_ptr<Sample> to Sample
This conversion does not apply to Patman as there seems to be issues
with it causing heap-use-after-free issues, such as with
PatmanInstrument::unloadCurrentPatch

* Fix segfault when closing LMMS
Song should be deleted before AudioEngine.

* Construct buffer through SampleLoader in FileBrowser's previewFileItem function
+ Remove const qualification in SamplePlayHandle(const QString&)
constructor for m_sample

* Move guard out of removeClip and deleteClips
+ Revert commit 1769ed517d since
this would fix it anyway
(we don't try to lock the engine to
delete the global automation track when closing LMMS now)

* Simplify the switch in play function for loopMode

* Add SampleDecoder

* Add LMMS_HAVE_OGGVORBIS comment

* Fix unused variable error

* Include unordered_map

* Simplify SampleDecoder
Instead of using the extension (which could be wrong) for the file,
we simply loop through all the decoders available. First sndfile because
it covers a lot of formats, then the ogg decoder for the few cases where sndfile
would not work for certain audio codecs, and then the DrumSynth decoder.

* Attempt to fix Mac builds

* Attempt to fix Mac builds take 2

* Add vector include to SampleDecoder

* Add TODO comment about shared ownership with clips

Calls to ClipView::remove may occur at any point, which can cause
a problem when the Track is using the clip about to be removed.
A suitable solution would be to use shared ownership between the Track
and ClipView for the clip. Track's can then simply remove the shared
pointer in their m_clips vector, and ClipView can call reset on the
shared pointer on calls to ClipView::remove.

* Adjust TODO comment
Disregard the shared ownership idea. Since we would be modifying
the collection of Clip's in Track when removing the Clip, the Track
could be iterating said collection while this happens,
causing a bug. In this case, we do actually
want a synchronization mechanism.

However, I didn't mention another separate issue in the TODO comment
that should've been addressed: ~Clip should not be responsible for
actually removing the itself from it's Track. With calls to removeClip,
one would expect that to already occur.

* Remove Sample::playbackSize
Inside SampleClip::sampleLength, we should be using Sample::sampleSize
instead.

* Fix issues involving length of Sample's
SampleClip::sampleLength should be passing the Sample's sample rate to
Engine::framesPerTick.

I also changed sampleDuration to return a std::chrono::milliseconds
instead of an int so that the callers know what time interval
is being used.

* Simplify if condition in src/gui/FileBrowser.cpp

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Simplify if condition in src/core/SampleBuffer.cpp

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Update style in include/Oscillator.h

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Format src/core/SampleDecoder.cpp

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Set the sample rate to be that of the AudioEngine by default

I also removed some checks involving the state of the SampleBuffer.
These functions should expect a valid SampleBuffer each time.
This helps to simplify things since we don't have to validate it
in each function.

* Set single-argument constructors in Sample and SampleBuffer to be explicit

* Do not make a copy when reading result from the decoder

* Add constructor to pass in vector of sampleFrame's directly

* Do a pass by value and move in SampleBuffer.cpp

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Pass vector by value in SampleBuffer.h

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Make Sample(std::shared_ptr) constructor explicit

* Properly draw sample waveform when reversed

* Collect sample not found errors when loading project

Also return empty buffers when trying to load
either an empty file or empty Base64 string

* Use std::make_unique<SampleBuffer> in SampleLoader

* Fix loop modes

* Limit sample duration to [start, end] and not the entire buffer

* Use structured binding to access buffer

* Check if GUI exists before displaying error

* Make Base64 constructor pass in the string instead

* Remove use of QByteArray::fromBase64Encoding

* Inline simple functions in SampleBuffer

* Dynamically include supported audio file types

* Remove redundant inline specifier

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Translate file types

* Cache calls to SampleDecoder::supportedAudioTypes

* Fix translations in SampleLoader (again)
Also ensure that all the file types are listed first.
Also simplified the generation of the list a bit.

* Store static local variable for supported audio types instead of in the header

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>

* Clamp frame index depending on loop mode

* Inline member functions of PlaybackState

* Do not collect errors in SampleLoader when loading projects

Also fix conflicts with surrounding codebase

* Default construct shared pointers to SampleBuffer

* Simplify and optimize Sample::visulaize()

* Remove redundant gui:: prefix

* Rearrange Sample::visualize after optimizations by DanielKauss

* Apply amplification when visualizing sample waveforms

* Set default min and max values to 1 and -1

* Treat waveform as mono signal when visualizing

* Ensure visualization works when framesPerPixel < 1

* Simplify Sample::visualize a bit more

* Fix CPU lag in Sample by using atomics (with relaxed ordering)

Changing any of the frame markers originally took a writer
lock on a mutex.

The problem is that Sample::play took a reader lock first before
executing. Because Sample::play has to wait on the writer, this
created a lot of lag and raised the CPU meter. The solution
would to be to use atomics instead.

* Fix errors from merge

* Fix broken LFO controller functionality

The shared_ptr should have been taken by reference.

* Remove TODO

* Update EnvelopeAndLfoView.cpp

Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>

* Update src/gui/clips/SampleClipView.cpp

Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>

* Update plugins/SlicerT/SlicerT.cpp

Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>

* Update plugins/SlicerT/SlicerT.cpp

Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>

* Store shortest relative path in SampleBuffer

* Tie up a few loose ends

* Use sample_rate_t when storing sample rate in SampleBuffer

* Add missing named requirement functions and aliases

* Use sampledata attribute when loading from Base64 in AFP

* Remove initializer for m_userWave in the constructor

* Do not use trailing return syntax when return is void

* Move decoder functionality into unnamed namespace

* Remove redundant gui:: prefix

* Use PathUtil::toAbsolute to simplify code in SampleLoader::openAudioFile

* Fix translations in SampleLoader::openAudioFile

Co-authored-by: DomClark <mrdomclark@gmail.com>

* Fix formatting for ternary operator

* Remove redundant inlines

* Resolve UB when decoding from Base64 data in SampleBuffer

* Fix up SampleClip constructors

* Add AudioResampler, a wrapper class around libsamplerate

The wrapper has only been applied to Sample::PlaybackState for now.
AudioResampler should be used by other classes in the future that do
resampling with libsamplerate.

* Move buffer when moving and simplify assignment functions in Sample

* Move Sample::visualize out of Sample and into the GUI namespace

* Initialize supportedAudioTypes in static lambda

* Return shared pointer from SampleLoader

* Create and use static empty SampleBuffer by default

* Fix header guard in SampleWaveform.h

* Remove use of src_clone
CI seems to have an old version of libsamplerate and does not have this method.

* Include memory header in SampleBuffer.h

* Remove mutex and shared_mutex includes in Sample.h

* Attempt to fix string operand error within AudioResampler

* Include string header in AudioResampler.cpp

* Add LMMS_EXPORT for SampleWaveform class declaration

* Add LMMS_EXPORT for AudioResampler class declaration

* Enforce returning std::shared_ptr<const SampleBuffer>

* Restrict the size of the memcpy to the destination size, not the source size

* Do not make resample const

AudioResampler::resample, while seemingly not changing the data of the resampler, still alters its internal state and therefore should not be const.
This is because libsamplerate manages state when
resampling.

* Initialize data.end_of_input

* Add trailing new lines

* Simplify AudioResampler interface

* Fix header guard prefix to LMMS_GUI instead of LMMS

* Remove Sample::resampleSampleRange

---------

Co-authored-by: Dalton Messmer <33463986+messmerd@users.noreply.github.com>
Co-authored-by: Daniel Kauss <daniel.kauss.serna@gmail.com>
Co-authored-by: Dalton Messmer <messmer.dalton@gmail.com>
Co-authored-by: DomClark <mrdomclark@gmail.com>
This commit is contained in:
saker
2023-12-25 07:07:11 -05:00
committed by GitHub
parent 3aefe7b3d3
commit ce722dd6b6
48 changed files with 1551 additions and 2263 deletions
Download Patch File Download Diff File Expand all files Collapse all files

64
include/AudioResampler.h Normal file
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@@ -0,0 +1,64 @@
/*
* AudioResampler.h - wrapper around libsamplerate
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
* 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_AUDIO_RESAMPLER_H
#define LMMS_AUDIO_RESAMPLER_H
#include <samplerate.h>
#include "lmms_export.h"
namespace lmms {
class LMMS_EXPORT AudioResampler
{
public:
struct ProcessResult
{
int error;
long inputFramesUsed;
long outputFramesGenerated;
};
AudioResampler(int interpolationMode, int channels);
AudioResampler(const AudioResampler&) = delete;
AudioResampler(AudioResampler&&) = delete;
~AudioResampler();
AudioResampler& operator=(const AudioResampler&) = delete;
AudioResampler& operator=(AudioResampler&&) = delete;
auto resample(const float* in, long inputFrames, float* out, long outputFrames, double ratio) -> ProcessResult;
auto interpolationMode() const -> int { return m_interpolationMode; }
auto channels() const -> int { return m_channels; }
private:
int m_interpolationMode = -1;
int m_channels = 0;
int m_error = 0;
SRC_STATE* m_state = nullptr;
};
} // namespace lmms
#endif // LMMS_AUDIO_RESAMPLER_H

4
include/AudioSampleRecorder.h
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@@ -28,6 +28,7 @@
#include <QList>
#include <QPair>
#include <memory>
#include "AudioDevice.h"
@@ -44,8 +45,7 @@ public:
~AudioSampleRecorder() override;
f_cnt_t framesRecorded() const;
void createSampleBuffer( SampleBuffer** sampleBuffer );
std::shared_ptr<const SampleBuffer> createSampleBuffer();
private:
void writeBuffer( const surroundSampleFrame * _ab,

3
include/EnvelopeAndLfoParameters.h
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@@ -25,6 +25,7 @@
#ifndef LMMS_ENVELOPE_AND_LFO_PARAMETERS_H
#define LMMS_ENVELOPE_AND_LFO_PARAMETERS_H
#include <memory>
#include <vector>
#include "JournallingObject.h"
@@ -167,7 +168,7 @@ private:
sample_t * m_lfoShapeData;
sample_t m_random;
bool m_bad_lfoShapeData;
SampleBuffer m_userWave;
std::shared_ptr<const SampleBuffer> m_userWave = SampleBuffer::emptyBuffer();
enum class LfoShape
{

2
include/LfoController.h
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@@ -87,7 +87,7 @@ protected:
private:
float m_heldSample;
SampleBuffer * m_userDefSampleBuffer;
std::shared_ptr<const SampleBuffer> m_userDefSampleBuffer = SampleBuffer::emptyBuffer();
protected slots:
void updatePhase();

30
include/Oscillator.h
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@@ -28,7 +28,9 @@
#include <cassert>
#include <fftw3.h>
#include <memory>
#include <cstdlib>
#include "interpolation.h"
#include "Engine.h"
#include "lmms_constants.h"
@@ -46,7 +48,6 @@ class IntModel;
class LMMS_EXPORT Oscillator
{
MM_OPERATORS
public:
enum class WaveShape
{
@@ -91,18 +92,23 @@ public:
static void waveTableInit();
static void destroyFFTPlans();
static void generateAntiAliasUserWaveTable(SampleBuffer* sampleBuffer);
static std::unique_ptr<OscillatorConstants::waveform_t> generateAntiAliasUserWaveTable(const SampleBuffer* sampleBuffer);
inline void setUseWaveTable(bool n)
{
m_useWaveTable = n;
}
inline void setUserWave( const SampleBuffer * _wave )
void setUserWave(std::shared_ptr<const SampleBuffer> _wave)
{
m_userWave = _wave;
}
void setUserAntiAliasWaveTable(std::shared_ptr<const OscillatorConstants::waveform_t> waveform)
{
m_userAntiAliasWaveTable = waveform;
}
void update(sampleFrame* ab, const fpp_t frames, const ch_cnt_t chnl, bool modulator = false);
// now follow the wave-shape-routines...
@@ -164,9 +170,18 @@ public:
return 1.0f - fast_rand() * 2.0f / FAST_RAND_MAX;
}
inline sample_t userWaveSample( const float _sample ) const
static sample_t userWaveSample(const SampleBuffer* buffer, const float sample)
{
return m_userWave->userWaveSample( _sample );
if (buffer == nullptr || buffer->size() == 0) { return 0; }
const auto frames = buffer->size();
const auto frame = sample * frames;
auto f1 = static_cast<f_cnt_t>(frame) % frames;
if (f1 < 0)
{
f1 += frames;
}
return linearInterpolate(buffer->data()[f1][0], buffer->data()[(f1 + 1) % frames][0], fraction(frame));
}
struct wtSampleControl {
@@ -203,7 +218,7 @@ public:
table[control.band][control.f2], fraction(control.frame));
}
inline sample_t wtSample(const std::unique_ptr<OscillatorConstants::waveform_t>& table, const float sample) const
sample_t wtSample(const OscillatorConstants::waveform_t* table, const float sample) const
{
assert(table != nullptr);
wtSampleControl control = getWtSampleControl(sample);
@@ -247,7 +262,8 @@ private:
Oscillator * m_subOsc;
float m_phaseOffset;
float m_phase;
const SampleBuffer * m_userWave;
std::shared_ptr<const SampleBuffer> m_userWave = SampleBuffer::emptyBuffer();
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
// adding more explicit parameters to all of them. Can be converted to a parameter if needed.

139
include/Sample.h Normal file
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@@ -0,0 +1,139 @@
/*
* Sample.h - State for container-class SampleBuffer
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
* 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_H
#define LMMS_SAMPLE_H
#include <cmath>
#include <memory>
#include "AudioResampler.h"
#include "Note.h"
#include "SampleBuffer.h"
#include "lmms_export.h"
class QPainter;
class QRect;
namespace lmms {
class LMMS_EXPORT Sample
{
public:
// 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
static constexpr auto s_interpolationMargins = std::array<int, 5>{64, 64, 64, 4, 4};
enum class Loop
{
Off,
On,
PingPong
};
class LMMS_EXPORT PlaybackState
{
public:
PlaybackState(bool varyingPitch = false, int interpolationMode = SRC_LINEAR)
: m_resampler(interpolationMode, DEFAULT_CHANNELS)
, m_varyingPitch(varyingPitch)
{
}
auto resampler() -> AudioResampler& { return m_resampler; }
auto frameIndex() const -> f_cnt_t { return m_frameIndex; }
auto varyingPitch() const -> bool { return m_varyingPitch; }
auto backwards() const -> bool { return m_backwards; }
void setFrameIndex(f_cnt_t frameIndex) { m_frameIndex = frameIndex; }
void setVaryingPitch(bool varyingPitch) { m_varyingPitch = varyingPitch; }
void setBackwards(bool backwards) { m_backwards = backwards; }
private:
AudioResampler m_resampler;
f_cnt_t m_frameIndex = 0;
bool m_varyingPitch = false;
bool m_backwards = false;
friend class Sample;
};
Sample() = default;
Sample(const QByteArray& base64, int sampleRate = Engine::audioEngine()->processingSampleRate());
Sample(const sampleFrame* data, int numFrames, int sampleRate = Engine::audioEngine()->processingSampleRate());
Sample(const Sample& other);
Sample(Sample&& other);
explicit Sample(const QString& audioFile);
explicit Sample(std::shared_ptr<const SampleBuffer> buffer);
auto operator=(const Sample&) -> Sample&;
auto operator=(Sample&&) -> Sample&;
auto play(sampleFrame* dst, PlaybackState* state, int numFrames, float desiredFrequency = DefaultBaseFreq,
Loop loopMode = Loop::Off) -> bool;
auto sampleDuration() const -> std::chrono::milliseconds;
auto sampleFile() const -> const QString& { return m_buffer->audioFile(); }
auto sampleRate() const -> int { return m_buffer->sampleRate(); }
auto sampleSize() const -> int { return m_buffer->size(); }
auto toBase64() const -> QString { return m_buffer->toBase64(); }
auto data() const -> const sampleFrame* { return m_buffer->data(); }
auto buffer() const -> std::shared_ptr<const SampleBuffer> { return m_buffer; }
auto startFrame() const -> int { return m_startFrame.load(std::memory_order_relaxed); }
auto endFrame() const -> int { return m_endFrame.load(std::memory_order_relaxed); }
auto loopStartFrame() const -> int { return m_loopStartFrame.load(std::memory_order_relaxed); }
auto loopEndFrame() const -> int { return m_loopEndFrame.load(std::memory_order_relaxed); }
auto amplification() const -> float { return m_amplification.load(std::memory_order_relaxed); }
auto frequency() const -> float { return m_frequency.load(std::memory_order_relaxed); }
auto reversed() const -> bool { return m_reversed.load(std::memory_order_relaxed); }
void setStartFrame(int startFrame) { m_startFrame.store(startFrame, std::memory_order_relaxed); }
void setEndFrame(int endFrame) { m_endFrame.store(endFrame, std::memory_order_relaxed); }
void setLoopStartFrame(int loopStartFrame) { m_loopStartFrame.store(loopStartFrame, std::memory_order_relaxed); }
void setLoopEndFrame(int loopEndFrame) { m_loopEndFrame.store(loopEndFrame, std::memory_order_relaxed); }
void setAllPointFrames(int startFrame, int endFrame, int loopStartFrame, int loopEndFrame);
void setAmplification(float amplification) { m_amplification.store(amplification, std::memory_order_relaxed); }
void setFrequency(float frequency) { m_frequency.store(frequency, std::memory_order_relaxed); }
void setReversed(bool reversed) { m_reversed.store(reversed, std::memory_order_relaxed); }
private:
void playSampleRange(PlaybackState* state, sampleFrame* dst, size_t numFrames) const;
void amplifySampleRange(sampleFrame* src, int numFrames) const;
void copyBufferForward(sampleFrame* dst, int initialPosition, int advanceAmount) const;
void copyBufferBackward(sampleFrame* dst, int initialPosition, int advanceAmount) const;
private:
std::shared_ptr<const SampleBuffer> m_buffer = SampleBuffer::emptyBuffer();
std::atomic<int> m_startFrame = 0;
std::atomic<int> m_endFrame = 0;
std::atomic<int> m_loopStartFrame = 0;
std::atomic<int> m_loopEndFrame = 0;
std::atomic<float> m_amplification = 1.0f;
std::atomic<float> m_frequency = DefaultBaseFreq;
std::atomic<bool> m_reversed = false;
};
} // namespace lmms
#endif

353
include/SampleBuffer.h
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@@ -25,333 +25,74 @@
#ifndef LMMS_SAMPLE_BUFFER_H
#define LMMS_SAMPLE_BUFFER_H
#include <QByteArray>
#include <QString>
#include <memory>
#include <QReadWriteLock>
#include <QObject>
#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_reference = const sampleFrame&;
using iterator = std::vector<sampleFrame>::iterator;
using const_iterator = std::vector<sampleFrame>::const_iterator;
using difference_type = std::vector<sampleFrame>::difference_type;
using size_type = std::vector<sampleFrame>::size_type;
using reverse_iterator = std::vector<sampleFrame>::reverse_iterator;
using const_reverse_iterator = std::vector<sampleFrame>::const_reverse_iterator;
const f_cnt_t frameIndex() const
{
return m_frameIndex;
}
SampleBuffer() = default;
explicit SampleBuffer(const QString& audioFile);
SampleBuffer(const QString& base64, int sampleRate);
SampleBuffer(std::vector<sampleFrame> data, 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 -> const QString& { return m_audioFile; }
auto sampleRate() const -> sample_rate_t { return m_sampleRate; }
void setBackwards(bool backwards)
{
m_isBackwards = backwards;
}
auto begin() -> iterator { return m_data.begin(); }
auto end() -> iterator { return m_data.end(); }
int interpolationMode() const
{
return m_interpolationMode;
}
auto begin() const -> const_iterator { return m_data.begin(); }
auto end() const -> const_iterator { return m_data.end(); }
auto cbegin() const -> const_iterator { return m_data.cbegin(); }
auto cend() const -> const_iterator { return m_data.cend(); }
private:
f_cnt_t m_frameIndex;
const bool m_varyingPitch;
bool m_isBackwards;
SRC_STATE * m_resamplingData;
int m_interpolationMode;
auto rbegin() -> reverse_iterator { return m_data.rbegin(); }
auto rend() -> reverse_iterator { return m_data.rend(); }
friend class SampleBuffer;
auto rbegin() const -> const_reverse_iterator { return m_data.rbegin(); }
auto rend() const -> const_reverse_iterator { return m_data.rend(); }
} ;
auto crbegin() const -> const_reverse_iterator { return m_data.crbegin(); }
auto crend() const -> const_reverse_iterator { return m_data.crend(); }
auto data() const -> const sampleFrame* { return m_data.data(); }
auto size() const -> size_type { return m_data.size(); }
auto empty() const -> bool { return m_data.empty(); }
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();
static auto emptyBuffer() -> std::shared_ptr<const SampleBuffer>;
private:
static sample_rate_t audioEngineSampleRate();
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
);
std::vector<sampleFrame> m_data;
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();
} ;
sample_rate_t m_sampleRate = Engine::audioEngine()->processingSampleRate();
};
} // namespace lmms

17
include/SampleClip.h
View File

@@ -25,7 +25,9 @@
#ifndef LMMS_SAMPLE_CLIP_H
#define LMMS_SAMPLE_CLIP_H
#include <memory>
#include "Clip.h"
#include "Sample.h"
namespace lmms
{
@@ -45,14 +47,15 @@ class SampleClip : public Clip
Q_OBJECT
mapPropertyFromModel(bool,isRecord,setRecord,m_recordModel);
public:
SampleClip( Track * _track );
SampleClip(Track* track, Sample sample, bool isPlaying);
SampleClip(Track* track);
SampleClip( const SampleClip& orig );
~SampleClip() override;
SampleClip& operator=( const SampleClip& that ) = delete;
void changeLength( const TimePos & _length ) override;
const QString & sampleFile() const;
const QString& sampleFile() const;
void saveSettings( QDomDocument & _doc, QDomElement & _parent ) override;
void loadSettings( const QDomElement & _this ) override;
@@ -61,9 +64,9 @@ public:
return "sampleclip";
}
SampleBuffer* sampleBuffer()
Sample& sample()
{
return m_sampleBuffer;
return m_sample;
}
TimePos sampleLength() const;
@@ -74,10 +77,10 @@ public:
bool isPlaying() const;
void setIsPlaying(bool isPlaying);
void setSampleBuffer(std::shared_ptr<const SampleBuffer> sb);
public slots:
void setSampleBuffer( lmms::SampleBuffer* sb );
void setSampleFile( const QString & sf );
void setSampleFile(const QString& sf);
void updateLength();
void toggleRecord();
void playbackPositionChanged();
@@ -85,7 +88,7 @@ public slots:
private:
SampleBuffer* m_sampleBuffer;
Sample m_sample;
BoolModel m_recordModel;
bool m_isPlaying;

57
include/SampleDecoder.h Normal file
View File

@@ -0,0 +1,57 @@
/*
* SampleDecoder.h - Decodes audio files in various formats
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
* 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_DECODER_H
#define LMMS_SAMPLE_DECODER_H
#include <QString>
#include <functional>
#include <optional>
#include <string>
#include <vector>
#include "lmms_basics.h"
namespace lmms {
class SampleDecoder
{
public:
struct Result
{
std::vector<sampleFrame> data;
int sampleRate;
};
struct AudioType
{
std::string name;
std::string extension;
};
static auto decode(const QString& audioFile) -> std::optional<Result>;
static auto supportedAudioTypes() -> const std::vector<AudioType>&;
};
} // namespace lmms
#endif // LMMS_SAMPLE_DECODER_H

48
include/SampleLoader.h Normal file
View File

@@ -0,0 +1,48 @@
/*
* SampleLoader.h - Load audio and waveform files
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
* 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_GUI_SAMPLE_LOADER_H
#define LMMS_GUI_SAMPLE_LOADER_H
#include <QString>
#include <memory>
#include "SampleBuffer.h"
#include "lmms_export.h"
namespace lmms::gui {
class LMMS_EXPORT SampleLoader
{
public:
static QString openAudioFile(const QString& previousFile = "");
static QString openWaveformFile(const QString& previousFile = "");
static std::shared_ptr<const SampleBuffer> createBufferFromFile(const QString& filePath);
static std::shared_ptr<const SampleBuffer> createBufferFromBase64(
const QString& base64, int sampleRate = Engine::audioEngine()->processingSampleRate());
private:
static void displayError(const QString& message);
};
} // namespace lmms::gui
#endif // LMMS_GUI_SAMPLE_LOADER_H

7
include/SamplePlayHandle.h
View File

@@ -26,6 +26,7 @@
#ifndef LMMS_SAMPLE_PLAY_HANDLE_H
#define LMMS_SAMPLE_PLAY_HANDLE_H
#include "Sample.h"
#include "SampleBuffer.h"
#include "AutomatableModel.h"
#include "PlayHandle.h"
@@ -43,7 +44,7 @@ class AudioPort;
class LMMS_EXPORT SamplePlayHandle : public PlayHandle
{
public:
SamplePlayHandle( SampleBuffer* sampleBuffer , bool ownAudioPort = true );
SamplePlayHandle(Sample* sample, bool ownAudioPort = true);
SamplePlayHandle( const QString& sampleFile );
SamplePlayHandle( SampleClip* clip );
~SamplePlayHandle() override;
@@ -81,11 +82,11 @@ public:
private:
SampleBuffer * m_sampleBuffer;
Sample* m_sample;
bool m_doneMayReturnTrue;
f_cnt_t m_frame;
SampleBuffer::handleState m_state;
Sample::PlaybackState m_state;
const bool m_ownAudioPort;

3
include/SampleRecordHandle.h
View File

@@ -27,6 +27,7 @@
#include <QList>
#include <QPair>
#include <memory>
#include "PlayHandle.h"
#include "TimePos.h"
@@ -53,7 +54,7 @@ public:
bool isFromTrack( const Track * _track ) const override;
f_cnt_t framesRecorded() const;
void createSampleBuffer( SampleBuffer * * _sample_buf );
std::shared_ptr<const SampleBuffer> createSampleBuffer();
private:

41
include/SampleWaveform.h Normal file
View File

@@ -0,0 +1,41 @@
/*
* SampleWaveform.h
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
* 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_GUI_SAMPLE_WAVEFORM_H
#define LMMS_GUI_SAMPLE_WAVEFORM_H
#include <QPainter>
#include "Sample.h"
#include "lmms_export.h"
namespace lmms::gui {
class LMMS_EXPORT SampleWaveform
{
public:
static void visualize(const Sample& sample, QPainter& p, const QRect& dr, int fromFrame = 0, int toFrame = 0);
};
} // namespace lmms::gui
#endif // LMMS_GUI_SAMPLE_WAVEFORM_H

192
plugins/AudioFileProcessor/AudioFileProcessor.cpp
View File

@@ -29,7 +29,6 @@
#include <QPainter>
#include <QFileInfo>
#include <QDropEvent>
#include <samplerate.h>
#include "AudioEngine.h"
@@ -42,6 +41,8 @@
#include "NotePlayHandle.h"
#include "PathUtil.h"
#include "PixmapButton.h"
#include "SampleLoader.h"
#include "SampleWaveform.h"
#include "Song.h"
#include "StringPairDrag.h"
#include "Clipboard.h"
@@ -83,7 +84,6 @@ Plugin::Descriptor PLUGIN_EXPORT audiofileprocessor_plugin_descriptor =
AudioFileProcessor::AudioFileProcessor( InstrumentTrack * _instrument_track ) :
Instrument( _instrument_track, &audiofileprocessor_plugin_descriptor ),
m_sampleBuffer(),
m_ampModel( 100, 0, 500, 1, this, tr( "Amplify" ) ),
m_startPointModel( 0, 0, 1, 0.0000001f, this, tr( "Start of sample" ) ),
m_endPointModel( 1, 0, 1, 0.0000001f, this, tr( "End of sample" ) ),
@@ -131,18 +131,18 @@ void AudioFileProcessor::playNote( NotePlayHandle * _n,
// played.
if( m_stutterModel.value() == true && _n->frequency() < 20.0 )
{
m_nextPlayStartPoint = m_sampleBuffer.startFrame();
m_nextPlayStartPoint = m_sample.startFrame();
m_nextPlayBackwards = false;
return;
}
if( !_n->m_pluginData )
{
if( m_stutterModel.value() == true && m_nextPlayStartPoint >= m_sampleBuffer.endFrame() )
if (m_stutterModel.value() == true && m_nextPlayStartPoint >= m_sample.endFrame())
{
// Restart playing the note if in stutter mode, not in loop mode,
// and we're at the end of the sample.
m_nextPlayStartPoint = m_sampleBuffer.startFrame();
m_nextPlayStartPoint = m_sample.startFrame();
m_nextPlayBackwards = false;
}
// set interpolation mode for libsamplerate
@@ -159,25 +159,25 @@ void AudioFileProcessor::playNote( NotePlayHandle * _n,
srcmode = SRC_SINC_MEDIUM_QUALITY;
break;
}
_n->m_pluginData = new handleState( _n->hasDetuningInfo(), srcmode );
((handleState *)_n->m_pluginData)->setFrameIndex( m_nextPlayStartPoint );
((handleState *)_n->m_pluginData)->setBackwards( m_nextPlayBackwards );
_n->m_pluginData = new Sample::PlaybackState(_n->hasDetuningInfo(), srcmode);
static_cast<Sample::PlaybackState*>(_n->m_pluginData)->setFrameIndex(m_nextPlayStartPoint);
static_cast<Sample::PlaybackState*>(_n->m_pluginData)->setBackwards(m_nextPlayBackwards);
// debug code
/* qDebug( "frames %d", m_sampleBuffer.frames() );
qDebug( "startframe %d", m_sampleBuffer.startFrame() );
/* qDebug( "frames %d", m_sample->frames() );
qDebug( "startframe %d", m_sample->startFrame() );
qDebug( "nextPlayStartPoint %d", m_nextPlayStartPoint );*/
}
if( ! _n->isFinished() )
{
if( m_sampleBuffer.play( _working_buffer + offset,
(handleState *)_n->m_pluginData,
if (m_sample.play(_working_buffer + offset,
static_cast<Sample::PlaybackState*>(_n->m_pluginData),
frames, _n->frequency(),
static_cast<SampleBuffer::LoopMode>( m_loopModel.value() ) ) )
static_cast<Sample::Loop>(m_loopModel.value())))
{
applyRelease( _working_buffer, _n );
emit isPlaying( ((handleState *)_n->m_pluginData)->frameIndex() );
emit isPlaying(static_cast<Sample::PlaybackState*>(_n->m_pluginData)->frameIndex());
}
else
{
@@ -191,8 +191,8 @@ void AudioFileProcessor::playNote( NotePlayHandle * _n,
}
if( m_stutterModel.value() == true )
{
m_nextPlayStartPoint = ((handleState *)_n->m_pluginData)->frameIndex();
m_nextPlayBackwards = ((handleState *)_n->m_pluginData)->isBackwards();
m_nextPlayStartPoint = static_cast<Sample::PlaybackState*>(_n->m_pluginData)->frameIndex();
m_nextPlayBackwards = static_cast<Sample::PlaybackState*>(_n->m_pluginData)->backwards();
}
}
@@ -201,7 +201,7 @@ void AudioFileProcessor::playNote( NotePlayHandle * _n,
void AudioFileProcessor::deleteNotePluginData( NotePlayHandle * _n )
{
delete (handleState *)_n->m_pluginData;
delete static_cast<Sample::PlaybackState*>(_n->m_pluginData);
}
@@ -209,11 +209,10 @@ void AudioFileProcessor::deleteNotePluginData( NotePlayHandle * _n )
void AudioFileProcessor::saveSettings(QDomDocument& doc, QDomElement& elem)
{
elem.setAttribute("src", m_sampleBuffer.audioFile());
if (m_sampleBuffer.audioFile().isEmpty())
elem.setAttribute("src", m_sample.sampleFile());
if (m_sample.sampleFile().isEmpty())
{
QString s;
elem.setAttribute("sampledata", m_sampleBuffer.toBase64(s));
elem.setAttribute("sampledata", m_sample.toBase64());
}
m_reverseModel.saveSettings(doc, elem, "reversed");
m_loopModel.saveSettings(doc, elem, "looped");
@@ -230,20 +229,17 @@ void AudioFileProcessor::saveSettings(QDomDocument& doc, QDomElement& elem)
void AudioFileProcessor::loadSettings(const QDomElement& elem)
{
if (!elem.attribute("src").isEmpty())
if (auto srcFile = elem.attribute("src"); !srcFile.isEmpty())
{
setAudioFile(elem.attribute("src"), false);
QString absolutePath = PathUtil::toAbsolute(m_sampleBuffer.audioFile());
if (!QFileInfo(absolutePath).exists())
if (QFileInfo(PathUtil::toAbsolute(srcFile)).exists())
{
QString message = tr("Sample not found: %1").arg(m_sampleBuffer.audioFile());
Engine::getSong()->collectError(message);
setAudioFile(srcFile, false);
}
else { Engine::getSong()->collectError(QString("%1: %2").arg(tr("Sample not found"), srcFile)); }
}
else if (!elem.attribute("sampledata").isEmpty())
else if (auto sampleData = elem.attribute("sampledata"); !sampleData.isEmpty())
{
m_sampleBuffer.loadFromBase64(elem.attribute("srcdata"));
m_sample = Sample(gui::SampleLoader::createBufferFromBase64(sampleData));
}
m_loopModel.loadSettings(elem, "looped");
@@ -274,6 +270,7 @@ void AudioFileProcessor::loadSettings(const QDomElement& elem)
}
pointChanged();
emit sampleUpdated();
}
@@ -298,7 +295,7 @@ QString AudioFileProcessor::nodeName() const
auto AudioFileProcessor::beatLen(NotePlayHandle* note) const -> int
{
// If we can play indefinitely, use the default beat note duration
if (static_cast<SampleBuffer::LoopMode>(m_loopModel.value()) != SampleBuffer::LoopMode::Off) { return 0; }
if (static_cast<Sample::Loop>(m_loopModel.value()) != Sample::Loop::Off) { return 0; }
// Otherwise, use the remaining sample duration
const auto baseFreq = instrumentTrack()->baseFreq();
@@ -306,10 +303,10 @@ auto AudioFileProcessor::beatLen(NotePlayHandle* note) const -> int
* Engine::audioEngine()->processingSampleRate()
/ Engine::audioEngine()->baseSampleRate();
const auto startFrame = m_nextPlayStartPoint >= m_sampleBuffer.endFrame()
? m_sampleBuffer.startFrame()
const auto startFrame = m_nextPlayStartPoint >= m_sample.endFrame()
? m_sample.startFrame()
: m_nextPlayStartPoint;
const auto duration = m_sampleBuffer.endFrame() - startFrame;
const auto duration = m_sample.endFrame() - startFrame;
return static_cast<int>(std::floor(duration * freqFactor));
}
@@ -322,25 +319,22 @@ gui::PluginView* AudioFileProcessor::instantiateView( QWidget * _parent )
return new gui::AudioFileProcessorView( this, _parent );
}
void AudioFileProcessor::setAudioFile( const QString & _audio_file,
bool _rename )
void AudioFileProcessor::setAudioFile(const QString& _audio_file, bool _rename)
{
// is current channel-name equal to previous-filename??
if( _rename &&
( instrumentTrack()->name() ==
QFileInfo( m_sampleBuffer.audioFile() ).fileName() ||
m_sampleBuffer.audioFile().isEmpty() ) )
QFileInfo(m_sample.sampleFile()).fileName() ||
m_sample.sampleFile().isEmpty()))
{
// then set it to new one
instrumentTrack()->setName( PathUtil::cleanName( _audio_file ) );
}
// else we don't touch the track-name, because the user named it self
m_sampleBuffer.setAudioFile( _audio_file );
m_sample = Sample(gui::SampleLoader::createBufferFromFile(_audio_file));
loopPointChanged();
emit sampleUpdated();
}
@@ -348,9 +342,10 @@ void AudioFileProcessor::setAudioFile( const QString & _audio_file,
void AudioFileProcessor::reverseModelChanged()
{
m_sampleBuffer.setReversed( m_reverseModel.value() );
m_nextPlayStartPoint = m_sampleBuffer.startFrame();
m_sample.setReversed(m_reverseModel.value());
m_nextPlayStartPoint = m_sample.startFrame();
m_nextPlayBackwards = false;
emit sampleUpdated();
}
@@ -358,13 +353,14 @@ void AudioFileProcessor::reverseModelChanged()
void AudioFileProcessor::ampModelChanged()
{
m_sampleBuffer.setAmplification( m_ampModel.value() / 100.0f );
m_sample.setAmplification(m_ampModel.value() / 100.0f);
emit sampleUpdated();
}
void AudioFileProcessor::stutterModelChanged()
{
m_nextPlayStartPoint = m_sampleBuffer.startFrame();
m_nextPlayStartPoint = m_sample.startFrame();
m_nextPlayBackwards = false;
}
@@ -433,14 +429,14 @@ void AudioFileProcessor::loopPointChanged()
void AudioFileProcessor::pointChanged()
{
const auto f_start = static_cast<f_cnt_t>(m_startPointModel.value() * m_sampleBuffer.frames());
const auto f_end = static_cast<f_cnt_t>(m_endPointModel.value() * m_sampleBuffer.frames());
const auto f_loop = static_cast<f_cnt_t>(m_loopPointModel.value() * m_sampleBuffer.frames());
const auto f_start = static_cast<f_cnt_t>(m_startPointModel.value() * m_sample.sampleSize());
const auto f_end = static_cast<f_cnt_t>(m_endPointModel.value() * m_sample.sampleSize());
const auto f_loop = static_cast<f_cnt_t>(m_loopPointModel.value() * m_sample.sampleSize());
m_nextPlayStartPoint = f_start;
m_nextPlayBackwards = false;
m_sampleBuffer.setAllPointFrames( f_start, f_end, f_loop, f_end );
m_sample.setAllPointFrames(f_start, f_end, f_loop, f_end);
emit dataChanged();
}
@@ -601,7 +597,7 @@ void AudioFileProcessorView::newWaveView()
delete m_waveView;
m_waveView = 0;
}
m_waveView = new AudioFileProcessorWaveView( this, 245, 75, castModel<AudioFileProcessor>()->m_sampleBuffer );
m_waveView = new AudioFileProcessorWaveView(this, 245, 75, &castModel<AudioFileProcessor>()->m_sample);
m_waveView->move( 2, 172 );
m_waveView->setKnobs(
dynamic_cast<AudioFileProcessorWaveView::knob *>( m_startKnob ),
@@ -648,7 +644,8 @@ void AudioFileProcessorView::paintEvent( QPaintEvent * )
auto a = castModel<AudioFileProcessor>();
QString file_name = "";
int idx = a->m_sampleBuffer.audioFile().length();
int idx = a->m_sample.sampleFile().length();
p.setFont( pointSize<8>( font() ) );
@@ -659,7 +656,7 @@ void AudioFileProcessorView::paintEvent( QPaintEvent * )
while( idx > 0 &&
fm.size( Qt::TextSingleLine, file_name + "..." ).width() < 210 )
{
file_name = a->m_sampleBuffer.audioFile()[--idx] + file_name;
file_name = a->m_sample.sampleFile()[--idx] + file_name;
}
if( idx > 0 )
@@ -687,7 +684,7 @@ void AudioFileProcessorView::sampleUpdated()
void AudioFileProcessorView::openAudioFile()
{
QString af = castModel<AudioFileProcessor>()->m_sampleBuffer.openAudioFile();
QString af = SampleLoader::openAudioFile();
if (af.isEmpty()) { return; }
castModel<AudioFileProcessor>()->setAudioFile(af);
@@ -701,8 +698,7 @@ void AudioFileProcessorView::openAudioFile()
void AudioFileProcessorView::modelChanged()
{
auto a = castModel<AudioFileProcessor>();
connect( &a->m_sampleBuffer, SIGNAL( sampleUpdated() ),
this, SLOT( sampleUpdated() ) );
connect(a, &AudioFileProcessor::sampleUpdated, this, &AudioFileProcessorView::sampleUpdated);
m_ampKnob->setModel( &a->m_ampModel );
m_startKnob->setModel( &a->m_startPointModel );
m_endKnob->setModel( &a->m_endPointModel );
@@ -719,20 +715,20 @@ void AudioFileProcessorView::modelChanged()
void AudioFileProcessorWaveView::updateSampleRange()
{
if( m_sampleBuffer.frames() > 1 )
if (m_sample->sampleSize() > 1)
{
const f_cnt_t marging = ( m_sampleBuffer.endFrame() - m_sampleBuffer.startFrame() ) * 0.1;
m_from = qMax( 0, m_sampleBuffer.startFrame() - marging );
m_to = qMin( m_sampleBuffer.endFrame() + marging, m_sampleBuffer.frames() );
const f_cnt_t marging = (m_sample->endFrame() - m_sample->startFrame()) * 0.1;
m_from = qMax(0, m_sample->startFrame() - marging);
m_to = qMin(m_sample->endFrame() + marging, m_sample->sampleSize());
}
}
AudioFileProcessorWaveView::AudioFileProcessorWaveView( QWidget * _parent, int _w, int _h, SampleBuffer& buf ) :
AudioFileProcessorWaveView::AudioFileProcessorWaveView(QWidget * _parent, int _w, int _h, Sample* buf) :
QWidget( _parent ),
m_sampleBuffer( buf ),
m_sample(buf),
m_graph( QPixmap( _w - 2 * s_padding, _h - 2 * s_padding ) ),
m_from( 0 ),
m_to( m_sampleBuffer.frames() ),
m_to(m_sample->sampleSize()),
m_last_from( 0 ),
m_last_to( 0 ),
m_last_amp( 0 ),
@@ -880,11 +876,11 @@ void AudioFileProcessorWaveView::paintEvent( QPaintEvent * _pe )
const QRect graph_rect( s_padding, s_padding, width() - 2 * s_padding, height() - 2 * s_padding );
const f_cnt_t frames = m_to - m_from;
m_startFrameX = graph_rect.x() + ( m_sampleBuffer.startFrame() - m_from ) *
m_startFrameX = graph_rect.x() + (m_sample->startFrame() - m_from) *
double( graph_rect.width() ) / frames;
m_endFrameX = graph_rect.x() + ( m_sampleBuffer.endFrame() - m_from ) *
m_endFrameX = graph_rect.x() + (m_sample->endFrame() - m_from) *
double( graph_rect.width() ) / frames;
m_loopFrameX = graph_rect.x() + ( m_sampleBuffer.loopStartFrame() - m_from ) *
m_loopFrameX = graph_rect.x() + (m_sample->loopStartFrame() - m_from) *
double( graph_rect.width() ) / frames;
const int played_width_px = ( m_framesPlayed - m_from ) *
double( graph_rect.width() ) / frames;
@@ -959,7 +955,7 @@ void AudioFileProcessorWaveView::paintEvent( QPaintEvent * _pe )
p.setFont( pointSize<8>( font() ) );
QString length_text;
const int length = m_sampleBuffer.sampleLength();
const int length = m_sample->sampleDuration().count();
if( length > 20000 )
{
@@ -988,42 +984,37 @@ void AudioFileProcessorWaveView::updateGraph()
{
if( m_to == 1 )
{
m_to = m_sampleBuffer.frames() * 0.7;
m_to = m_sample->sampleSize() * 0.7;
slideSamplePointToFrames( Point::End, m_to * 0.7 );
}
if( m_from > m_sampleBuffer.startFrame() )
if (m_from > m_sample->startFrame())
{
m_from = m_sampleBuffer.startFrame();
m_from = m_sample->startFrame();
}
if( m_to < m_sampleBuffer.endFrame() )
if (m_to < m_sample->endFrame())
{
m_to = m_sampleBuffer.endFrame();
m_to = m_sample->endFrame();
}
if( m_sampleBuffer.reversed() != m_reversed )
if (m_sample->reversed() != m_reversed)
{
reverse();
}
else if( m_last_from == m_from && m_last_to == m_to && m_sampleBuffer.amplification() == m_last_amp )
else if (m_last_from == m_from && m_last_to == m_to && m_sample->amplification() == m_last_amp)
{
return;
}
m_last_from = m_from;
m_last_to = m_to;
m_last_amp = m_sampleBuffer.amplification();
m_last_amp = m_sample->amplification();
m_graph.fill( Qt::transparent );
QPainter p( &m_graph );
p.setPen( QColor( 255, 255, 255 ) );
m_sampleBuffer.visualize(
p,
QRect( 0, 0, m_graph.width(), m_graph.height() ),
m_from, m_to
);
SampleWaveform::visualize(*m_sample, p, QRect(0, 0, m_graph.width(), m_graph.height()), m_from, m_to);
}
@@ -1031,9 +1022,9 @@ void AudioFileProcessorWaveView::updateGraph()
void AudioFileProcessorWaveView::zoom( const bool _out )
{
const f_cnt_t start = m_sampleBuffer.startFrame();
const f_cnt_t end = m_sampleBuffer.endFrame();
const f_cnt_t frames = m_sampleBuffer.frames();
const f_cnt_t start = m_sample->startFrame();
const f_cnt_t end = m_sample->endFrame();
const f_cnt_t frames = m_sample->sampleSize();
const f_cnt_t d_from = start - m_from;
const f_cnt_t d_to = m_to - end;
@@ -1066,7 +1057,7 @@ void AudioFileProcessorWaveView::zoom( const bool _out )
);
}
if( double( new_to - new_from ) / m_sampleBuffer.sampleRate() > 0.05 )
if (static_cast<double>(new_to - new_from) / m_sample->sampleRate() > 0.05)
{
m_from = new_from;
m_to = new_to;
@@ -1085,8 +1076,8 @@ void AudioFileProcessorWaveView::slide( int _px )
step = -step;
}
f_cnt_t step_from = qBound( 0, m_from + step, m_sampleBuffer.frames() ) - m_from;
f_cnt_t step_to = qBound( m_from + 1, m_to + step, m_sampleBuffer.frames() ) - m_to;
f_cnt_t step_from = qBound(0, m_from + step, m_sample->sampleSize()) - m_from;
f_cnt_t step_to = qBound(m_from + 1, m_to + step, m_sample->sampleSize()) - m_to;
step = qAbs( step_from ) < qAbs( step_to ) ? step_from : step_to;
@@ -1147,7 +1138,7 @@ void AudioFileProcessorWaveView::slideSamplePointByFrames( Point _point, f_cnt_t
}
else
{
const double v = static_cast<double>( _frames ) / m_sampleBuffer.frames();
const double v = static_cast<double>(_frames) / m_sample->sampleSize();
if( _slide_to )
{
a_knob->slideTo( v );
@@ -1164,11 +1155,11 @@ void AudioFileProcessorWaveView::slideSamplePointByFrames( Point _point, f_cnt_t
void AudioFileProcessorWaveView::slideSampleByFrames( f_cnt_t _frames )
{
if( m_sampleBuffer.frames() <= 1 )
if (m_sample->sampleSize() <= 1)
{
return;
}
const double v = static_cast<double>( _frames ) / m_sampleBuffer.frames();
const double v = static_cast<double>( _frames ) / m_sample->sampleSize();
// update knobs in the right order
// to avoid them clamping each other
if (v < 0)
@@ -1191,14 +1182,14 @@ void AudioFileProcessorWaveView::slideSampleByFrames( f_cnt_t _frames )
void AudioFileProcessorWaveView::reverse()
{
slideSampleByFrames(
m_sampleBuffer.frames()
- m_sampleBuffer.endFrame()
- m_sampleBuffer.startFrame()
m_sample->sampleSize()
- m_sample->endFrame()
- m_sample->startFrame()
);
const f_cnt_t from = m_from;
m_from = m_sampleBuffer.frames() - m_to;
m_to = m_sampleBuffer.frames() - from;
m_from = m_sample->sampleSize() - m_to;
m_to = m_sample->sampleSize() - from;
m_reversed = ! m_reversed;
}
@@ -1240,8 +1231,7 @@ void AudioFileProcessorWaveView::knob::slideTo( double _v, bool _check_bound )
float AudioFileProcessorWaveView::knob::getValue( const QPoint & _p )
{
const double dec_fact = ! m_waveView ? 1 :
double( m_waveView->m_to - m_waveView->m_from )
/ m_waveView->m_sampleBuffer.frames();
static_cast<double>(m_waveView->m_to - m_waveView->m_from) / m_waveView->m_sample->sampleSize();
const float inc = Knob::getValue( _p ) * dec_fact;
return inc;
@@ -1262,12 +1252,12 @@ bool AudioFileProcessorWaveView::knob::checkBound( double _v ) const
return false;
const double d1 = qAbs( m_relatedKnob->model()->value() - model()->value() )
* ( m_waveView->m_sampleBuffer.frames() )
/ m_waveView->m_sampleBuffer.sampleRate();
* (m_waveView->m_sample->sampleSize())
/ m_waveView->m_sample->sampleRate();
const double d2 = qAbs( m_relatedKnob->model()->value() - _v )
* ( m_waveView->m_sampleBuffer.frames() )
/ m_waveView->m_sampleBuffer.sampleRate();
* (m_waveView->m_sample->sampleSize())
/ m_waveView->m_sample->sampleRate();
return d1 < d2 || d2 > 0.005;
}

16
plugins/AudioFileProcessor/AudioFileProcessor.h
View File

@@ -31,6 +31,7 @@
#include "ComboBoxModel.h"
#include "Instrument.h"
#include "InstrumentView.h"
#include "Sample.h"
#include "SampleBuffer.h"
#include "Knob.h"
@@ -78,8 +79,7 @@ public:
public slots:
void setAudioFile( const QString & _audio_file, bool _rename = true );
void setAudioFile(const QString& _audio_file, bool _rename = true);
private slots:
void reverseModelChanged();
@@ -93,12 +93,10 @@ private slots:
signals:
void isPlaying( lmms::f_cnt_t _current_frame );
void sampleUpdated();
private:
using handleState = SampleBuffer::handleState;
SampleBuffer m_sampleBuffer;
Sample m_sample;
FloatModel m_ampModel;
FloatModel m_startPointModel;
@@ -246,7 +244,7 @@ private:
SampleLoop
} ;
SampleBuffer& m_sampleBuffer;
Sample* m_sample;
QPixmap m_graph;
f_cnt_t m_from;
f_cnt_t m_to;
@@ -266,8 +264,10 @@ private:
f_cnt_t m_framesPlayed;
bool m_animation;
friend class AudioFileProcessorView;
public:
AudioFileProcessorWaveView( QWidget * _parent, int _w, int _h, SampleBuffer& buf );
AudioFileProcessorWaveView(QWidget * _parent, int _w, int _h, Sample* buf);
void setKnobs(knob *_start, knob *_end, knob *_loop );

4
plugins/GigPlayer/GigPlayer.cpp
View File

@@ -46,7 +46,7 @@
#include "Knob.h"
#include "NotePlayHandle.h"
#include "PathUtil.h"
#include "SampleBuffer.h"
#include "Sample.h"
#include "Song.h"
#include "PatchesDialog.h"
@@ -437,7 +437,7 @@ void GigInstrument::play( sampleFrame * _working_buffer )
if (sample.region->PitchTrack == true) { freq_factor *= sample.freqFactor; }
// We need a bit of margin so we don't get glitching
samples = frames / freq_factor + MARGIN[m_interpolation];
samples = frames / freq_factor + Sample::s_interpolationMargins[m_interpolation];
}
// Load this note's data

26
plugins/Patman/Patman.cpp
View File

@@ -153,8 +153,8 @@ void PatmanInstrument::playNote( NotePlayHandle * _n,
float play_freq = hdata->tuned ? _n->frequency() :
hdata->sample->frequency();
if( hdata->sample->play( _working_buffer + offset, hdata->state, frames,
play_freq, m_loopedModel.value() ? SampleBuffer::LoopMode::On : SampleBuffer::LoopMode::Off ) )
if (hdata->sample->play(_working_buffer + offset, hdata->state, frames,
play_freq, m_loopedModel.value() ? Sample::Loop::On : Sample::Loop::Off))
{
applyRelease( _working_buffer, _n );
}
@@ -170,7 +170,6 @@ void PatmanInstrument::playNote( NotePlayHandle * _n,
void PatmanInstrument::deleteNotePluginData( NotePlayHandle * _n )
{
auto hdata = (handle_data*)_n->m_pluginData;
sharedObject::unref( hdata->sample );
delete hdata->state;
delete hdata;
}
@@ -356,9 +355,8 @@ PatmanInstrument::LoadError PatmanInstrument::loadPatch(
}
}
auto psample = new SampleBuffer(data, frames);
psample->setFrequency( root_freq / 1000.0f );
psample->setSampleRate( sample_rate );
auto psample = std::make_shared<Sample>(data, frames, sample_rate);
psample->setFrequency(root_freq / 1000.0f);
if( modes & MODES_LOOPING )
{
@@ -366,7 +364,7 @@ PatmanInstrument::LoadError PatmanInstrument::loadPatch(
psample->setLoopEndFrame( loop_end );
}
m_patchSamples.push_back( psample );
m_patchSamples.push_back(psample);
delete[] wave_samples;
delete[] data;
@@ -382,7 +380,6 @@ void PatmanInstrument::unloadCurrentPatch()
{
while( !m_patchSamples.empty() )
{
sharedObject::unref( m_patchSamples.back() );
m_patchSamples.pop_back();
}
}
@@ -395,7 +392,7 @@ void PatmanInstrument::selectSample( NotePlayHandle * _n )
const float freq = _n->frequency();
float min_dist = HUGE_VALF;
SampleBuffer* sample = nullptr;
std::shared_ptr<Sample> sample = nullptr;
for (const auto& patchSample : m_patchSamples)
{
@@ -412,15 +409,8 @@ void PatmanInstrument::selectSample( NotePlayHandle * _n )
auto hdata = new handle_data;
hdata->tuned = m_tunedModel.value();
if( sample )
{
hdata->sample = sharedObject::ref( sample );
}
else
{
hdata->sample = new SampleBuffer( nullptr, 0 );
}
hdata->state = new SampleBuffer::handleState( _n->hasDetuningInfo() );
hdata->sample = sample ? sample : std::make_shared<Sample>();
hdata->state = new Sample::PlaybackState(_n->hasDetuningInfo());
_n->m_pluginData = hdata;
}

7
plugins/Patman/Patman.h
View File

@@ -28,6 +28,7 @@
#include "Instrument.h"
#include "InstrumentView.h"
#include "Sample.h"
#include "SampleBuffer.h"
#include "AutomatableModel.h"
#include "MemoryManager.h"
@@ -87,13 +88,13 @@ private:
struct handle_data
{
MM_OPERATORS
SampleBuffer::handleState* state;
Sample::PlaybackState* state;
bool tuned;
SampleBuffer* sample;
std::shared_ptr<Sample> sample;
};
QString m_patchFile;
QVector<SampleBuffer *> m_patchSamples;
QVector<std::shared_ptr<Sample>> m_patchSamples;
BoolModel m_loopedModel;
BoolModel m_tunedModel;

55
plugins/SlicerT/SlicerT.cpp
View File

@@ -31,6 +31,7 @@
#include "Engine.h"
#include "InstrumentTrack.h"
#include "PathUtil.h"
#include "SampleLoader.h"
#include "Song.h"
#include "embed.h"
#include "lmms_constants.h"
@@ -76,7 +77,7 @@ SlicerT::SlicerT(InstrumentTrack* instrumentTrack)
void SlicerT::playNote(NotePlayHandle* handle, sampleFrame* workingBuffer)
{
if (m_originalSample.frames() <= 1) { return; }
if (m_originalSample.sampleSize() <= 1) { return; }
int noteIndex = handle->key() - m_parentTrack->baseNote();
const fpp_t frames = handle->framesLeftForCurrentPeriod();
@@ -115,24 +116,24 @@ void SlicerT::playNote(NotePlayHandle* handle, sampleFrame* workingBuffer)
if (noteLeft > 0)
{
int noteFrame = noteDone * m_originalSample.frames();
int noteFrame = noteDone * m_originalSample.sampleSize();
SRC_STATE* resampleState = playbackState->resamplingState();
SRC_DATA resampleData;
resampleData.data_in = (m_originalSample.data() + noteFrame)->data();
resampleData.data_out = (workingBuffer + offset)->data();
resampleData.input_frames = noteLeft * m_originalSample.frames();
resampleData.input_frames = noteLeft * m_originalSample.sampleSize();
resampleData.output_frames = frames;
resampleData.src_ratio = speedRatio;
src_process(resampleState, &resampleData);
float nextNoteDone = noteDone + frames * (1.0f / speedRatio) / m_originalSample.frames();
float nextNoteDone = noteDone + frames * (1.0f / speedRatio) / m_originalSample.sampleSize();
playbackState->setNoteDone(nextNoteDone);
// exponential fade out, applyRelease() not used since it extends the note length
int fadeOutFrames = m_fadeOutFrames.value() / 1000.0f * Engine::audioEngine()->processingSampleRate();
int noteFramesLeft = noteLeft * m_originalSample.frames() * speedRatio;
int noteFramesLeft = noteLeft * m_originalSample.sampleSize() * speedRatio;
for (int i = 0; i < frames; i++)
{
float fadeValue = static_cast<float>(noteFramesLeft - i) / fadeOutFrames;
@@ -160,7 +161,7 @@ void SlicerT::deleteNotePluginData(NotePlayHandle* handle)
// http://www.iro.umontreal.ca/~pift6080/H09/documents/papers/bello_onset_tutorial.pdf
void SlicerT::findSlices()
{
if (m_originalSample.frames() <= 1) { return; }
if (m_originalSample.sampleSize() <= 1) { return; }
m_slicePoints = {};
const int windowSize = 512;
@@ -170,8 +171,8 @@ void SlicerT::findSlices()
int minDist = sampleRate * minBeatLength;
float maxMag = -1;
std::vector<float> singleChannel(m_originalSample.frames(), 0);
for (int i = 0; i < m_originalSample.frames(); i++)
std::vector<float> singleChannel(m_originalSample.sampleSize(), 0);
for (int i = 0; i < m_originalSample.sampleSize(); i++)
{
singleChannel[i] = (m_originalSample.data()[i][0] + m_originalSample.data()[i][1]) / 2;
maxMag = std::max(maxMag, singleChannel[i]);
@@ -232,7 +233,7 @@ void SlicerT::findSlices()
spectralFlux = 1E-10; // again for no divison by zero
}
m_slicePoints.push_back(m_originalSample.frames());
m_slicePoints.push_back(m_originalSample.sampleSize());
for (float& sliceValue : m_slicePoints)
{
@@ -255,7 +256,7 @@ void SlicerT::findSlices()
for (float& sliceIndex : m_slicePoints)
{
sliceIndex /= m_originalSample.frames();
sliceIndex /= m_originalSample.sampleSize();
}
m_slicePoints[0] = 0;
@@ -268,10 +269,10 @@ void SlicerT::findSlices()
// and lies in the 100 - 200 bpm range
void SlicerT::findBPM()
{
if (m_originalSample.frames() <= 1) { return; }
if (m_originalSample.sampleSize() <= 1) { return; }
float sampleRate = m_originalSample.sampleRate();
float totalFrames = m_originalSample.frames();
float totalFrames = m_originalSample.sampleSize();
float sampleLength = totalFrames / sampleRate;
float bpmEstimate = 240.0f / sampleLength;
@@ -295,7 +296,7 @@ std::vector<Note> SlicerT::getMidi()
std::vector<Note> outputNotes;
float speedRatio = static_cast<float>(m_originalBPM.value()) / Engine::getSong()->getTempo();
float outFrames = m_originalSample.frames() * speedRatio;
float outFrames = m_originalSample.sampleSize() * speedRatio;
float framesPerTick = Engine::framesPerTick();
float totalTicks = outFrames / framesPerTick;
@@ -320,7 +321,7 @@ std::vector<Note> SlicerT::getMidi()
void SlicerT::updateFile(QString file)
{
m_originalSample.setAudioFile(file);
if (auto buffer = gui::SampleLoader::createBufferFromFile(file)) { m_originalSample = Sample(std::move(buffer)); }
findBPM();
findSlices();
@@ -336,11 +337,10 @@ void SlicerT::updateSlices()
void SlicerT::saveSettings(QDomDocument& document, QDomElement& element)
{
element.setAttribute("version", "1");
element.setAttribute("src", m_originalSample.audioFile());
if (m_originalSample.audioFile().isEmpty())
element.setAttribute("src", m_originalSample.sampleFile());
if (m_originalSample.sampleFile().isEmpty())
{
QString s;
element.setAttribute("sampledata", m_originalSample.toBase64(s));
element.setAttribute("sampledata", m_originalSample.toBase64());
}
element.setAttribute("totalSlices", static_cast<int>(m_slicePoints.size()));
@@ -357,20 +357,23 @@ void SlicerT::saveSettings(QDomDocument& document, QDomElement& element)
void SlicerT::loadSettings(const QDomElement& element)
{
if (!element.attribute("src").isEmpty())
if (auto srcFile = element.attribute("src"); !srcFile.isEmpty())
{
m_originalSample.setAudioFile(element.attribute("src"));
QString absolutePath = PathUtil::toAbsolute(m_originalSample.audioFile());
if (!QFileInfo(absolutePath).exists())
if (QFileInfo(PathUtil::toAbsolute(srcFile)).exists())
{
QString message = tr("Sample not found: %1").arg(m_originalSample.audioFile());
auto buffer = gui::SampleLoader::createBufferFromFile(srcFile);
m_originalSample = Sample(std::move(buffer));
}
else
{
QString message = tr("Sample not found: %1").arg(srcFile);
Engine::getSong()->collectError(message);
}
}
else if (!element.attribute("sampledata").isEmpty())
else if (auto sampleData = element.attribute("sampledata"); !sampleData.isEmpty())
{
m_originalSample.loadFromBase64(element.attribute("srcdata"));
auto buffer = gui::SampleLoader::createBufferFromBase64(sampleData);
m_originalSample = Sample(std::move(buffer));
}
if (!element.attribute("totalSlices").isEmpty())

3
plugins/SlicerT/SlicerT.h
View File

@@ -33,6 +33,7 @@
#include "Instrument.h"
#include "InstrumentView.h"
#include "Note.h"
#include "Sample.h"
#include "SampleBuffer.h"
#include "SlicerTView.h"
#include "lmms_basics.h"
@@ -95,7 +96,7 @@ private:
ComboBoxModel m_sliceSnap;
BoolModel m_enableSync;
SampleBuffer m_originalSample;
Sample m_originalSample;
std::vector<float> m_slicePoints;

5
plugins/SlicerT/SlicerTView.cpp
View File

@@ -31,6 +31,7 @@
#include "DataFile.h"
#include "Engine.h"
#include "InstrumentTrack.h"
#include "SampleLoader.h"
#include "SlicerT.h"
#include "Song.h"
#include "StringPairDrag.h"
@@ -108,7 +109,7 @@ Knob* SlicerTView::createStyledKnob()
void SlicerTView::exportMidi()
{
using namespace Clipboard;
if (m_slicerTParent->m_originalSample.frames() <= 1) { return; }
if (m_slicerTParent->m_originalSample.sampleSize() <= 1) { return; }
DataFile dataFile(DataFile::Type::ClipboardData);
QDomElement noteList = dataFile.createElement("note-list");
@@ -129,7 +130,7 @@ void SlicerTView::exportMidi()
void SlicerTView::openFiles()
{
QString audioFile = m_slicerTParent->m_originalSample.openAudioFile();
const auto audioFile = SampleLoader::openAudioFile();
if (audioFile.isEmpty()) { return; }
m_slicerTParent->updateFile(audioFile);
}

25
plugins/SlicerT/SlicerTWaveform.cpp
View File

@@ -26,6 +26,7 @@
#include <QBitmap>
#include "SampleWaveform.h"
#include "SlicerT.h"
#include "SlicerTView.h"
#include "embed.h"
@@ -84,12 +85,13 @@ SlicerTWaveform::SlicerTWaveform(int totalWidth, int totalHeight, SlicerT* instr
void SlicerTWaveform::drawSeekerWaveform()
{
m_seekerWaveform.fill(s_waveformBgColor);
if (m_slicerTParent->m_originalSample.frames() <= 1) { return; }
if (m_slicerTParent->m_originalSample.sampleSize() <= 1) { return; }
QPainter brush(&m_seekerWaveform);
brush.setPen(s_waveformColor);
m_slicerTParent->m_originalSample.visualize(brush, QRect(0, 0, m_seekerWaveform.width(), m_seekerWaveform.height()),
0, m_slicerTParent->m_originalSample.frames());
SampleWaveform::visualize(m_slicerTParent->m_originalSample, brush,
QRect(0, 0, m_seekerWaveform.width(), m_seekerWaveform.height()), 0,
m_slicerTParent->m_originalSample.sampleSize());
// increase brightness in inner color
QBitmap innerMask = m_seekerWaveform.createMaskFromColor(s_waveformMaskColor, Qt::MaskMode::MaskOutColor);
@@ -100,7 +102,7 @@ void SlicerTWaveform::drawSeekerWaveform()
void SlicerTWaveform::drawSeeker()
{
m_seeker.fill(s_emptyColor);
if (m_slicerTParent->m_originalSample.frames() <= 1) { return; }
if (m_slicerTParent->m_originalSample.sampleSize() <= 1) { return; }
QPainter brush(&m_seeker);
brush.setPen(s_sliceColor);
@@ -134,16 +136,17 @@ void SlicerTWaveform::drawSeeker()
void SlicerTWaveform::drawEditorWaveform()
{
m_editorWaveform.fill(s_emptyColor);
if (m_slicerTParent->m_originalSample.frames() <= 1) { return; }
if (m_slicerTParent->m_originalSample.sampleSize() <= 1) { return; }
QPainter brush(&m_editorWaveform);
float startFrame = m_seekerStart * m_slicerTParent->m_originalSample.frames();
float endFrame = m_seekerEnd * m_slicerTParent->m_originalSample.frames();
float startFrame = m_seekerStart * m_slicerTParent->m_originalSample.sampleSize();
float endFrame = m_seekerEnd * m_slicerTParent->m_originalSample.sampleSize();
brush.setPen(s_waveformColor);
float zoomOffset = (m_editorHeight - m_zoomLevel * m_editorHeight) / 2;
m_slicerTParent->m_originalSample.visualize(
brush, QRect(0, zoomOffset, m_editorWidth, m_zoomLevel * m_editorHeight), startFrame, endFrame);
SampleWaveform::visualize(m_slicerTParent->m_originalSample, brush,
QRect(0, zoomOffset, m_editorWidth, m_zoomLevel * m_editorHeight), startFrame, endFrame);
// increase brightness in inner color
QBitmap innerMask = m_editorWaveform.createMaskFromColor(s_waveformMaskColor, Qt::MaskMode::MaskOutColor);
@@ -157,7 +160,7 @@ void SlicerTWaveform::drawEditor()
QPainter brush(&m_sliceEditor);
// No sample loaded
if (m_slicerTParent->m_originalSample.frames() <= 1)
if (m_slicerTParent->m_originalSample.sampleSize() <= 1)
{
brush.setPen(s_playHighlightColor);
brush.setFont(QFont(brush.font().family(), 9.0f, -1, false));
@@ -306,7 +309,7 @@ void SlicerTWaveform::mousePressEvent(QMouseEvent* me)
drawEditorWaveform();
break;
case Qt::MouseButton::LeftButton:
if (m_slicerTParent->m_originalSample.frames() <= 1) { static_cast<SlicerTView*>(parent())->openFiles(); }
if (m_slicerTParent->m_originalSample.sampleSize() <= 1) { static_cast<SlicerTView*>(parent())->openFiles(); }
// update seeker middle for correct movement
m_seekerMiddle = static_cast<float>(me->x() - s_seekerHorMargin) / m_seekerWidth;
break;

36
plugins/TripleOscillator/TripleOscillator.cpp
View File

@@ -23,8 +23,8 @@
*/
#include <QDomElement>
#include <QFileInfo>
#include "TripleOscillator.h"
#include "AudioEngine.h"
@@ -35,9 +35,11 @@
#include "Knob.h"
#include "NotePlayHandle.h"
#include "Oscillator.h"
#include "PathUtil.h"
#include "PixmapButton.h"
#include "SampleBuffer.h"
#include "SampleLoader.h"
#include "Song.h"
#include "embed.h"
#include "plugin_export.h"
@@ -133,22 +135,13 @@ OscillatorObject::OscillatorObject( Model * _parent, int _idx ) :
}
OscillatorObject::~OscillatorObject()
{
sharedObject::unref( m_sampleBuffer );
}
void OscillatorObject::oscUserDefWaveDblClick()
{
QString af = m_sampleBuffer->openAndSetWaveformFile();
auto af = gui::SampleLoader::openWaveformFile();
if( af != "" )
{
m_sampleBuffer = gui::SampleLoader::createBufferFromFile(af);
m_userAntiAliasWaveTable = Oscillator::generateAntiAliasUserWaveTable(m_sampleBuffer.get());
// TODO:
//m_usrWaveBtn->setToolTip(m_sampleBuffer->audioFile());
}
@@ -289,8 +282,16 @@ void TripleOscillator::loadSettings( const QDomElement & _this )
"modalgo" + QString::number( i+1 ) );
m_osc[i]->m_useWaveTableModel.loadSettings( _this,
"useWaveTable" + QString::number (i+1 ) );
m_osc[i]->m_sampleBuffer->setAudioFile( _this.attribute(
"userwavefile" + is ) );
if (auto userWaveFile = _this.attribute("userwavefile" + is); !userWaveFile.isEmpty())
{
if (QFileInfo(PathUtil::toAbsolute(userWaveFile)).exists())
{
m_osc[i]->m_sampleBuffer = gui::SampleLoader::createBufferFromFile(userWaveFile);
m_osc[i]->m_userAntiAliasWaveTable = Oscillator::generateAntiAliasUserWaveTable(m_osc[i]->m_sampleBuffer.get());
}
else { Engine::getSong()->collectError(QString("%1: %2").arg(tr("Sample not found"), userWaveFile)); }
}
}
}
@@ -360,7 +361,8 @@ void TripleOscillator::playNote( NotePlayHandle * _n,
oscs_l[i]->setUserWave( m_osc[i]->m_sampleBuffer );
oscs_r[i]->setUserWave( m_osc[i]->m_sampleBuffer );
oscs_l[i]->setUserAntiAliasWaveTable(m_osc[i]->m_userAntiAliasWaveTable);
oscs_r[i]->setUserAntiAliasWaveTable(m_osc[i]->m_userAntiAliasWaveTable);
}
_n->m_pluginData = new oscPtr;

10
plugins/TripleOscillator/TripleOscillator.h
View File

@@ -26,9 +26,13 @@
#ifndef _TRIPLE_OSCILLATOR_H
#define _TRIPLE_OSCILLATOR_H
#include <memory>
#include "Instrument.h"
#include "InstrumentView.h"
#include "AutomatableModel.h"
#include "OscillatorConstants.h"
#include "SampleBuffer.h"
namespace lmms
{
@@ -57,9 +61,6 @@ class OscillatorObject : public Model
Q_OBJECT
public:
OscillatorObject( Model * _parent, int _idx );
~OscillatorObject() override;
private:
FloatModel m_volumeModel;
FloatModel m_panModel;
@@ -71,7 +72,8 @@ private:
IntModel m_waveShapeModel;
IntModel m_modulationAlgoModel;
BoolModel m_useWaveTableModel;
SampleBuffer* m_sampleBuffer;
std::shared_ptr<const SampleBuffer> m_sampleBuffer = SampleBuffer::emptyBuffer();
std::shared_ptr<const OscillatorConstants::waveform_t> m_userAntiAliasWaveTable;
float m_volumeLeft;
float m_volumeRight;

64
src/core/AudioResampler.cpp Normal file
View File

@@ -0,0 +1,64 @@
/*
* AudioResampler.cpp - wrapper for libsamplerate
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
* 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.
*
*/
#include "AudioResampler.h"
#include <samplerate.h>
#include <stdexcept>
#include <string>
namespace lmms {
AudioResampler::AudioResampler(int interpolationMode, int channels)
: m_interpolationMode(interpolationMode)
, m_channels(channels)
, m_state(src_new(interpolationMode, channels, &m_error))
{
if (!m_state)
{
const auto errorMessage = std::string{src_strerror(m_error)};
const auto fullMessage = std::string{"Failed to create an AudioResampler: "} + errorMessage;
throw std::runtime_error{fullMessage};
}
}
AudioResampler::~AudioResampler()
{
src_delete(m_state);
}
auto AudioResampler::resample(const float* in, long inputFrames, float* out, long outputFrames, double ratio)
-> ProcessResult
{
auto data = SRC_DATA{};
data.data_in = in;
data.input_frames = inputFrames;
data.data_out = out;
data.output_frames = outputFrames;
data.src_ratio = ratio;
data.end_of_input = 0;
return {src_process(m_state, &data), data.input_frames_used, data.output_frames_gen};
}
} // namespace lmms

3
src/core/CMakeLists.txt
View File

@@ -4,6 +4,7 @@ set(LMMS_SRCS
core/AudioEngine.cpp
core/AudioEngineProfiler.cpp
core/AudioEngineWorkerThread.cpp
core/AudioResampler.cpp
core/AutomatableModel.cpp
core/AutomationClip.cpp
core/AutomationNode.cpp
@@ -65,8 +66,10 @@ set(LMMS_SRCS
core/RemotePlugin.cpp
core/RenderManager.cpp
core/RingBuffer.cpp
core/Sample.cpp
core/SampleBuffer.cpp
core/SampleClip.cpp
core/SampleDecoder.cpp
core/SamplePlayHandle.cpp
core/SampleRecordHandle.cpp
core/Scale.cpp

25
src/core/EnvelopeAndLfoParameters.cpp
View File

@@ -22,13 +22,17 @@
*
*/
#include <QDomElement>
#include "EnvelopeAndLfoParameters.h"
#include <QDomElement>
#include <QFileInfo>
#include "AudioEngine.h"
#include "Engine.h"
#include "Oscillator.h"
#include "PathUtil.h"
#include "SampleLoader.h"
#include "Song.h"
namespace lmms
{
@@ -118,7 +122,7 @@ EnvelopeAndLfoParameters::EnvelopeAndLfoParameters(
m_controlEnvAmountModel( false, this, tr( "Modulate env amount" ) ),
m_lfoFrame( 0 ),
m_lfoAmountIsZero( false ),
m_lfoShapeData( nullptr )
m_lfoShapeData(nullptr)
{
m_amountModel.setCenterValue( 0 );
m_lfoAmountModel.setCenterValue( 0 );
@@ -221,7 +225,7 @@ inline sample_t EnvelopeAndLfoParameters::lfoShapeSample( fpp_t _frame_offset )
shape_sample = Oscillator::sawSample( phase );
break;
case LfoShape::UserDefinedWave:
shape_sample = m_userWave.userWaveSample( phase );
shape_sample = Oscillator::userWaveSample(m_userWave.get(), phase);
break;
case LfoShape::RandomWave:
if( frame == 0 )
@@ -354,7 +358,7 @@ void EnvelopeAndLfoParameters::saveSettings( QDomDocument & _doc,
m_lfoAmountModel.saveSettings( _doc, _parent, "lamt" );
m_x100Model.saveSettings( _doc, _parent, "x100" );
m_controlEnvAmountModel.saveSettings( _doc, _parent, "ctlenvamt" );
_parent.setAttribute( "userwavefile", m_userWave.audioFile() );
_parent.setAttribute("userwavefile", m_userWave->audioFile());
}
@@ -386,7 +390,14 @@ void EnvelopeAndLfoParameters::loadSettings( const QDomElement & _this )
m_sustainModel.setValue( 1.0 - m_sustainModel.value() );
}
m_userWave.setAudioFile( _this.attribute( "userwavefile" ) );
if (const auto userWaveFile = _this.attribute("userwavefile"); !userWaveFile.isEmpty())
{
if (QFileInfo(PathUtil::toAbsolute(userWaveFile)).exists())
{
m_userWave = gui::SampleLoader::createBufferFromFile(_this.attribute("userwavefile"));
}
else { Engine::getSong()->collectError(QString("%1: %2").arg(tr("Sample not found"), userWaveFile)); }
}
updateSampleVars();
}

29
src/core/LfoController.cpp
View File

@@ -23,13 +23,15 @@
*
*/
#include <QDomElement>
#include "LfoController.h"
#include "AudioEngine.h"
#include "Song.h"
#include <QDomElement>
#include <QFileInfo>
#include "AudioEngine.h"
#include "PathUtil.h"
#include "SampleLoader.h"
#include "Song.h"
namespace lmms
{
@@ -48,7 +50,7 @@ LfoController::LfoController( Model * _parent ) :
m_phaseOffset( 0 ),
m_currentPhase( 0 ),
m_sampleFunction( &Oscillator::sinSample ),
m_userDefSampleBuffer( new SampleBuffer )
m_userDefSampleBuffer(std::make_shared<SampleBuffer>())
{
setSampleExact( true );
connect( &m_waveModel, SIGNAL(dataChanged()),
@@ -74,7 +76,6 @@ LfoController::LfoController( Model * _parent ) :
LfoController::~LfoController()
{
sharedObject::unref( m_userDefSampleBuffer );
m_baseModel.disconnect( this );
m_speedModel.disconnect( this );
m_amountModel.disconnect( this );
@@ -122,7 +123,7 @@ void LfoController::updateValueBuffer()
}
case Oscillator::WaveShape::UserDefined:
{
currentSample = m_userDefSampleBuffer->userWaveSample(phase);
currentSample = Oscillator::userWaveSample(m_userDefSampleBuffer.get(), phase);
break;
}
default:
@@ -222,7 +223,7 @@ void LfoController::saveSettings( QDomDocument & _doc, QDomElement & _this )
m_phaseModel.saveSettings( _doc, _this, "phase" );
m_waveModel.saveSettings( _doc, _this, "wave" );
m_multiplierModel.saveSettings( _doc, _this, "multiplier" );
_this.setAttribute( "userwavefile" , m_userDefSampleBuffer->audioFile() );
_this.setAttribute("userwavefile", m_userDefSampleBuffer->audioFile());
}
@@ -237,7 +238,15 @@ void LfoController::loadSettings( const QDomElement & _this )
m_phaseModel.loadSettings( _this, "phase" );
m_waveModel.loadSettings( _this, "wave" );
m_multiplierModel.loadSettings( _this, "multiplier" );
m_userDefSampleBuffer->setAudioFile( _this.attribute("userwavefile" ) );
if (const auto userWaveFile = _this.attribute("userwavefile"); !userWaveFile.isEmpty())
{
if (QFileInfo(PathUtil::toAbsolute(userWaveFile)).exists())
{
m_userDefSampleBuffer = gui::SampleLoader::createBufferFromFile(_this.attribute("userwavefile"));
}
else { Engine::getSong()->collectError(QString("%1: %2").arg(tr("Sample not found"), userWaveFile)); }
}
updateSampleFunction();
}

22
src/core/Oscillator.cpp
View File

@@ -182,19 +182,23 @@ void Oscillator::generateFromFFT(int bands, sample_t* table)
normalize(s_sampleBuffer.data(), table, OscillatorConstants::WAVETABLE_LENGTH, 2*OscillatorConstants::WAVETABLE_LENGTH + 1);
}
void Oscillator::generateAntiAliasUserWaveTable(SampleBuffer *sampleBuffer)
std::unique_ptr<OscillatorConstants::waveform_t> Oscillator::generateAntiAliasUserWaveTable(const SampleBuffer* sampleBuffer)
{
if (sampleBuffer->m_userAntiAliasWaveTable == nullptr) {return;}
auto userAntiAliasWaveTable = std::make_unique<OscillatorConstants::waveform_t>();
for (int i = 0; i < OscillatorConstants::WAVE_TABLES_PER_WAVEFORM_COUNT; ++i)
{
for (int i = 0; i < OscillatorConstants::WAVETABLE_LENGTH; ++i)
// TODO: This loop seems to be doing the same thing for each iteration of the outer loop,
// and could probably be moved out of it
for (int j = 0; j < OscillatorConstants::WAVETABLE_LENGTH; ++j)
{
s_sampleBuffer[i] = sampleBuffer->userWaveSample((float)i / (float)OscillatorConstants::WAVETABLE_LENGTH);
s_sampleBuffer[j] = Oscillator::userWaveSample(
sampleBuffer, static_cast<float>(j) / OscillatorConstants::WAVETABLE_LENGTH);
}
fftwf_execute(s_fftPlan);
Oscillator::generateFromFFT(OscillatorConstants::MAX_FREQ / freqFromWaveTableBand(i), (*(sampleBuffer->m_userAntiAliasWaveTable))[i].data());
Oscillator::generateFromFFT(OscillatorConstants::MAX_FREQ / freqFromWaveTableBand(i), (*userAntiAliasWaveTable)[i].data());
}
return userAntiAliasWaveTable;
}
@@ -807,13 +811,13 @@ template<>
inline sample_t Oscillator::getSample<Oscillator::WaveShape::UserDefined>(
const float _sample )
{
if (m_useWaveTable && !m_isModulator)
if (m_useWaveTable && m_userAntiAliasWaveTable && !m_isModulator)
{
return wtSample(m_userWave->m_userAntiAliasWaveTable, _sample);
return wtSample(m_userAntiAliasWaveTable.get(), _sample);
}
else
{
return userWaveSample(_sample);
return userWaveSample(m_userWave.get(), _sample);
}
}

230
src/core/Sample.cpp Normal file
View File

@@ -0,0 +1,230 @@
/*
* Sample.cpp - State for container-class SampleBuffer
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
* 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.
*
*/
#include "Sample.h"
#include <QPainter>
#include <QRect>
namespace lmms {
Sample::Sample(const QString& audioFile)
: m_buffer(std::make_shared<SampleBuffer>(audioFile))
, m_startFrame(0)
, m_endFrame(m_buffer->size())
, m_loopStartFrame(0)
, m_loopEndFrame(m_buffer->size())
{
}
Sample::Sample(const QByteArray& base64, int sampleRate)
: m_buffer(std::make_shared<SampleBuffer>(base64, sampleRate))
, m_startFrame(0)
, m_endFrame(m_buffer->size())
, m_loopStartFrame(0)
, m_loopEndFrame(m_buffer->size())
{
}
Sample::Sample(const sampleFrame* data, int numFrames, int sampleRate)
: m_buffer(std::make_shared<SampleBuffer>(data, numFrames, sampleRate))
, m_startFrame(0)
, m_endFrame(m_buffer->size())
, m_loopStartFrame(0)
, m_loopEndFrame(m_buffer->size())
{
}
Sample::Sample(std::shared_ptr<const SampleBuffer> buffer)
: m_buffer(buffer)
, m_startFrame(0)
, m_endFrame(m_buffer->size())
, m_loopStartFrame(0)
, m_loopEndFrame(m_buffer->size())
{
}
Sample::Sample(const Sample& other)
: m_buffer(other.m_buffer)
, m_startFrame(other.startFrame())
, m_endFrame(other.endFrame())
, m_loopStartFrame(other.loopStartFrame())
, m_loopEndFrame(other.loopEndFrame())
, m_amplification(other.amplification())
, m_frequency(other.frequency())
, m_reversed(other.reversed())
{
}
Sample::Sample(Sample&& other)
: m_buffer(std::move(other.m_buffer))
, m_startFrame(other.startFrame())
, m_endFrame(other.endFrame())
, m_loopStartFrame(other.loopStartFrame())
, m_loopEndFrame(other.loopEndFrame())
, m_amplification(other.amplification())
, m_frequency(other.frequency())
, m_reversed(other.reversed())
{
}
auto Sample::operator=(const Sample& other) -> Sample&
{
m_buffer = other.m_buffer;
m_startFrame = other.startFrame();
m_endFrame = other.endFrame();
m_loopStartFrame = other.loopStartFrame();
m_loopEndFrame = other.loopEndFrame();
m_amplification = other.amplification();
m_frequency = other.frequency();
m_reversed = other.reversed();
return *this;
}
auto Sample::operator=(Sample&& other) -> Sample&
{
m_buffer = std::move(other.m_buffer);
m_startFrame = other.startFrame();
m_endFrame = other.endFrame();
m_loopStartFrame = other.loopStartFrame();
m_loopEndFrame = other.loopEndFrame();
m_amplification = other.amplification();
m_frequency = other.frequency();
m_reversed = other.reversed();
return *this;
}
bool Sample::play(sampleFrame* dst, PlaybackState* state, int numFrames, float desiredFrequency, Loop loopMode)
{
if (numFrames <= 0 || desiredFrequency <= 0) { return false; }
auto resampleRatio = static_cast<float>(Engine::audioEngine()->processingSampleRate()) / m_buffer->sampleRate();
resampleRatio *= frequency() / desiredFrequency;
auto playBuffer = std::vector<sampleFrame>(numFrames / resampleRatio);
if (!typeInfo<float>::isEqual(resampleRatio, 1.0f))
{
playBuffer.resize(playBuffer.size() + s_interpolationMargins[state->resampler().interpolationMode()]);
}
const auto start = startFrame();
const auto end = endFrame();
const auto loopStart = loopStartFrame();
const auto loopEnd = loopEndFrame();
switch (loopMode)
{
case Loop::Off:
state->m_frameIndex = std::clamp(state->m_frameIndex, start, end);
if (state->m_frameIndex == end) { return false; }
break;
case Loop::On:
state->m_frameIndex = std::clamp(state->m_frameIndex, start, loopEnd);
if (state->m_frameIndex == loopEnd) { state->m_frameIndex = loopStart; }
break;
case Loop::PingPong:
state->m_frameIndex = std::clamp(state->m_frameIndex, start, loopEnd);
if (state->m_frameIndex == loopEnd)
{
state->m_frameIndex = loopEnd - 1;
state->m_backwards = true;
}
else if (state->m_frameIndex <= loopStart && state->m_backwards)
{
state->m_frameIndex = loopStart;
state->m_backwards = false;
}
break;
}
playSampleRange(state, playBuffer.data(), playBuffer.size());
const auto result
= state->resampler().resample(&playBuffer[0][0], playBuffer.size(), &dst[0][0], numFrames, resampleRatio);
if (result.error != 0) { return false; }
state->m_frameIndex += (state->m_backwards ? -1 : 1) * result.inputFramesUsed;
amplifySampleRange(dst, result.outputFramesGenerated);
return true;
}
auto Sample::sampleDuration() const -> std::chrono::milliseconds
{
const auto numFrames = endFrame() - startFrame();
const auto duration = numFrames / static_cast<float>(m_buffer->sampleRate()) * 1000;
return std::chrono::milliseconds{static_cast<int>(duration)};
}
void Sample::setAllPointFrames(int startFrame, int endFrame, int loopStartFrame, int loopEndFrame)
{
setStartFrame(startFrame);
setEndFrame(endFrame);
setLoopStartFrame(loopStartFrame);
setLoopEndFrame(loopEndFrame);
}
void Sample::playSampleRange(PlaybackState* state, sampleFrame* dst, size_t numFrames) const
{
auto framesToCopy = 0;
if (state->m_backwards)
{
framesToCopy = std::min<int>(state->m_frameIndex - startFrame(), numFrames);
copyBufferBackward(dst, state->m_frameIndex, framesToCopy);
}
else
{
framesToCopy = std::min<int>(endFrame() - state->m_frameIndex, numFrames);
copyBufferForward(dst, state->m_frameIndex, framesToCopy);
}
if (framesToCopy < numFrames) { std::fill_n(dst + framesToCopy, numFrames - framesToCopy, sampleFrame{0, 0}); }
}
void Sample::copyBufferForward(sampleFrame* dst, int initialPosition, int advanceAmount) const
{
reversed() ? std::copy_n(m_buffer->rbegin() + initialPosition, advanceAmount, dst)
: std::copy_n(m_buffer->begin() + initialPosition, advanceAmount, dst);
}
void Sample::copyBufferBackward(sampleFrame* dst, int initialPosition, int advanceAmount) const
{
reversed() ? std::reverse_copy(
m_buffer->rbegin() + initialPosition - advanceAmount, m_buffer->rbegin() + initialPosition, dst)
: std::reverse_copy(
m_buffer->begin() + initialPosition - advanceAmount, m_buffer->begin() + initialPosition, dst);
}
void Sample::amplifySampleRange(sampleFrame* src, int numFrames) const
{
const auto amplification = m_amplification.load(std::memory_order_relaxed);
for (int i = 0; i < numFrames; ++i)
{
src[i][0] *= amplification;
src[i][1] *= amplification;
}
}
} // namespace lmms

1616
src/core/SampleBuffer.cpp
View File

File diff suppressed because it is too large Load Diff

88
src/core/SampleClip.cpp
View File

@@ -25,21 +25,22 @@
#include "SampleClip.h"
#include <QDomElement>
#include <QFileInfo>
#include "PathUtil.h"
#include "SampleBuffer.h"
#include "SampleClipView.h"
#include "SampleLoader.h"
#include "SampleTrack.h"
#include "TimeLineWidget.h"
namespace lmms
{
SampleClip::SampleClip( Track * _track ) :
Clip( _track ),
m_sampleBuffer( new SampleBuffer ),
m_isPlaying( false )
SampleClip::SampleClip(Track* _track, Sample sample, bool isPlaying)
: Clip(_track)
, m_sample(std::move(sample))
, m_isPlaying(false)
{
saveJournallingState( false );
setSampleFile( "" );
@@ -81,14 +82,14 @@ SampleClip::SampleClip( Track * _track ) :
updateTrackClips();
}
SampleClip::SampleClip(const SampleClip& orig) :
SampleClip(orig.getTrack())
SampleClip::SampleClip(Track* track)
: SampleClip(track, Sample(), false)
{
}
SampleClip::SampleClip(const SampleClip& orig) :
SampleClip(orig.getTrack(), orig.m_sample, orig.m_isPlaying)
{
// TODO: This creates a new SampleBuffer for the new Clip, eating up memory
// & eventually causing performance issues. Letting tracks share buffers
// when they're identical would fix this, but isn't possible right now.
*m_sampleBuffer = *orig.m_sampleBuffer;
m_isPlaying = orig.m_isPlaying;
}
@@ -101,9 +102,6 @@ SampleClip::~SampleClip()
{
sampletrack->updateClips();
}
Engine::audioEngine()->requestChangeInModel();
sharedObject::unref( m_sampleBuffer );
Engine::audioEngine()->doneChangeInModel();
}
@@ -117,33 +115,30 @@ void SampleClip::changeLength( const TimePos & _length )
const QString & SampleClip::sampleFile() const
const QString& SampleClip::sampleFile() const
{
return m_sampleBuffer->audioFile();
return m_sample.sampleFile();
}
void SampleClip::setSampleBuffer( SampleBuffer* sb )
void SampleClip::setSampleBuffer(std::shared_ptr<const SampleBuffer> sb)
{
Engine::audioEngine()->requestChangeInModel();
sharedObject::unref( m_sampleBuffer );
Engine::audioEngine()->doneChangeInModel();
m_sampleBuffer = sb;
{
const auto guard = Engine::audioEngine()->requestChangesGuard();
m_sample = Sample(std::move(sb));
}
updateLength();
emit sampleChanged();
}
void SampleClip::setSampleFile(const QString & sf)
void SampleClip::setSampleFile(const QString& sf)
{
int length = 0;
if (!sf.isEmpty())
{
m_sampleBuffer->setAudioFile(sf);
//Otherwise set it to the sample's length
m_sample = Sample(gui::SampleLoader::createBufferFromFile(sf));
length = sampleLength();
}
@@ -222,7 +217,7 @@ void SampleClip::updateLength()
TimePos SampleClip::sampleLength() const
{
return (int)( m_sampleBuffer->frames() / Engine::framesPerTick() );
return static_cast<int>(m_sample.sampleSize() / Engine::framesPerTick(m_sample.sampleRate()));
}
@@ -230,7 +225,7 @@ TimePos SampleClip::sampleLength() const
void SampleClip::setSampleStartFrame(f_cnt_t startFrame)
{
m_sampleBuffer->setStartFrame( startFrame );
m_sample.setStartFrame(startFrame);
}
@@ -238,7 +233,7 @@ void SampleClip::setSampleStartFrame(f_cnt_t startFrame)
void SampleClip::setSamplePlayLength(f_cnt_t length)
{
m_sampleBuffer->setEndFrame( length );
m_sample.setEndFrame(length);
}
@@ -261,15 +256,15 @@ void SampleClip::saveSettings( QDomDocument & _doc, QDomElement & _this )
if( sampleFile() == "" )
{
QString s;
_this.setAttribute( "data", m_sampleBuffer->toBase64( s ) );
_this.setAttribute("data", m_sample.toBase64());
}
_this.setAttribute( "sample_rate", m_sampleBuffer->sampleRate());
_this.setAttribute( "sample_rate", m_sample.sampleRate());
if (const auto& c = color())
{
_this.setAttribute("color", c->name());
}
if (m_sampleBuffer->reversed())
if (m_sample.reversed())
{
_this.setAttribute("reversed", "true");
}
@@ -285,14 +280,23 @@ void SampleClip::loadSettings( const QDomElement & _this )
{
movePosition( _this.attribute( "pos" ).toInt() );
}
setSampleFile( _this.attribute( "src" ) );
if (const auto srcFile = _this.attribute("src"); !srcFile.isEmpty())
{
if (QFileInfo(PathUtil::toAbsolute(srcFile)).exists())
{
setSampleFile(srcFile);
}
else { Engine::getSong()->collectError(QString("%1: %2").arg(tr("Sample not found"), srcFile)); }
}
if( sampleFile().isEmpty() && _this.hasAttribute( "data" ) )
{
m_sampleBuffer->loadFromBase64( _this.attribute( "data" ) );
if (_this.hasAttribute("sample_rate"))
{
m_sampleBuffer->setSampleRate(_this.attribute("sample_rate").toInt());
}
auto sampleRate = _this.hasAttribute("sample_rate") ? _this.attribute("sample_rate").toInt() :
Engine::audioEngine()->processingSampleRate();
auto buffer = gui::SampleLoader::createBufferFromBase64(_this.attribute("data"), sampleRate);
m_sample = Sample(std::move(buffer));
}
changeLength( _this.attribute( "len" ).toInt() );
setMuted( _this.attribute( "muted" ).toInt() );
@@ -305,7 +309,7 @@ void SampleClip::loadSettings( const QDomElement & _this )
if(_this.hasAttribute("reversed"))
{
m_sampleBuffer->setReversed(true);
m_sample.setReversed(true);
emit wasReversed(); // tell SampleClipView to update the view
}
}

184
src/core/SampleDecoder.cpp Normal file
View File

@@ -0,0 +1,184 @@
#include "SampleDecoder.h"
#include <QFile>
#include <QFileInfo>
#include <QString>
#include <memory>
#include <sndfile.h>
#ifdef LMMS_HAVE_OGGVORBIS
#include <vorbis/vorbisfile.h>
#endif
#include "AudioEngine.h"
#include "DrumSynth.h"
#include "Engine.h"
#include "lmms_basics.h"
namespace lmms {
namespace {
using Decoder = std::optional<SampleDecoder::Result>(*)(const QString&);
auto decodeSampleSF(const QString& audioFile) -> std::optional<SampleDecoder::Result>;
auto decodeSampleDS(const QString& audioFile) -> std::optional<SampleDecoder::Result>;
#ifdef LMMS_HAVE_OGGVORBIS
auto decodeSampleOggVorbis(const QString& audioFile) -> std::optional<SampleDecoder::Result>;
#endif
static constexpr std::array<Decoder, 3> decoders = {&decodeSampleSF,
#ifdef LMMS_HAVE_OGGVORBIS
&decodeSampleOggVorbis,
#endif
&decodeSampleDS};
auto decodeSampleSF(const QString& audioFile) -> std::optional<SampleDecoder::Result>
{
SNDFILE* sndFile = nullptr;
auto sfInfo = SF_INFO{};
// Use QFile to handle unicode file names on Windows
auto file = QFile{audioFile};
if (!file.open(QIODevice::ReadOnly)) { return std::nullopt; }
sndFile = sf_open_fd(file.handle(), SFM_READ, &sfInfo, false);
if (sf_error(sndFile) != 0) { return std::nullopt; }
auto buf = std::vector<sample_t>(sfInfo.channels * sfInfo.frames);
sf_read_float(sndFile, buf.data(), buf.size());
sf_close(sndFile);
file.close();
auto result = std::vector<sampleFrame>(sfInfo.frames);
for (int i = 0; i < static_cast<int>(result.size()); ++i)
{
if (sfInfo.channels == 1)
{
// Upmix from mono to stereo
result[i] = {buf[i], buf[i]};
}
else if (sfInfo.channels > 1)
{
// TODO: Add support for higher number of channels (i.e., 5.1 channel systems)
// The current behavior assumes stereo in all cases excluding mono.
// This may not be the expected behavior, given some audio files with a higher number of channels.
result[i] = {buf[i * sfInfo.channels], buf[i * sfInfo.channels + 1]};
}
}
return SampleDecoder::Result{std::move(result), static_cast<int>(sfInfo.samplerate)};
}
auto decodeSampleDS(const QString& audioFile) -> std::optional<SampleDecoder::Result>
{
// Populated by DrumSynth::GetDSFileSamples
int_sample_t* dataPtr = nullptr;
auto ds = DrumSynth{};
const auto engineRate = Engine::audioEngine()->processingSampleRate();
const auto frames = ds.GetDSFileSamples(audioFile, dataPtr, DEFAULT_CHANNELS, engineRate);
const auto data = std::unique_ptr<int_sample_t[]>{dataPtr}; // NOLINT, we have to use a C-style array here
if (frames <= 0 || !data) { return std::nullopt; }
auto result = std::vector<sampleFrame>(frames);
src_short_to_float_array(data.get(), &result[0][0], frames * DEFAULT_CHANNELS);
return SampleDecoder::Result{std::move(result), static_cast<int>(engineRate)};
}
#ifdef LMMS_HAVE_OGGVORBIS
auto decodeSampleOggVorbis(const QString& audioFile) -> std::optional<SampleDecoder::Result>
{
auto vorbisFile = OggVorbis_File{};
const auto openError = ov_fopen(audioFile.toLocal8Bit(), &vorbisFile);
if (openError != 0) { return std::nullopt; }
const auto vorbisInfo = ov_info(&vorbisFile, -1);
const auto numChannels = vorbisInfo->channels;
const auto sampleRate = vorbisInfo->rate;
const auto numSamples = ov_pcm_total(&vorbisFile, -1);
auto buffer = std::vector<float>(numSamples);
auto output = static_cast<float**>(nullptr);
auto totalSamplesRead = 0;
while (true)
{
auto samplesRead = ov_read_float(&vorbisFile, &output, numSamples, 0);
if (samplesRead < 0) { return std::nullopt; }
else if (samplesRead == 0) { break; }
std::copy_n(*output, samplesRead, buffer.begin() + totalSamplesRead);
totalSamplesRead += samplesRead;
}
ov_clear(&vorbisFile);
auto result = std::vector<sampleFrame>(numSamples / numChannels);
for (int i = 0; i < buffer.size(); ++i)
{
if (numChannels == 1) { result[i] = {buffer[i], buffer[i]}; }
else if (numChannels > 1) { result[i] = {buffer[i * numChannels], buffer[i * numChannels + 1]}; }
}
return SampleDecoder::Result{std::move(result), static_cast<int>(sampleRate)};
}
#endif // LMMS_HAVE_OGGVORBIS
} // namespace
auto SampleDecoder::supportedAudioTypes() -> const std::vector<AudioType>&
{
static const auto s_audioTypes = []
{
auto types = std::vector<AudioType>();
// Add DrumSynth by default since that support comes from us
types.push_back(AudioType{"DrumSynth", "ds"});
auto sfFormatInfo = SF_FORMAT_INFO{};
auto simpleTypeCount = 0;
sf_command(nullptr, SFC_GET_SIMPLE_FORMAT_COUNT, &simpleTypeCount, sizeof(int));
// TODO: Ideally, this code should be iterating over the major formats, but some important extensions such as *.ogg
// are not included. This is planned for future versions of sndfile.
for (int simple = 0; simple < simpleTypeCount; ++simple)
{
sfFormatInfo.format = simple;
sf_command(nullptr, SFC_GET_SIMPLE_FORMAT, &sfFormatInfo, sizeof(sfFormatInfo));
auto it = std::find_if(types.begin(), types.end(),
[&](const AudioType& type) { return sfFormatInfo.extension == type.extension; });
if (it != types.end()) { continue; }
auto name = std::string{sfFormatInfo.extension};
std::transform(name.begin(), name.end(), name.begin(), [](unsigned char ch) { return std::toupper(ch); });
types.push_back(AudioType{std::move(name), sfFormatInfo.extension});
return types;
}
std::sort(types.begin(), types.end(),
[&](const AudioType& a, const AudioType& b) { return a.name < b.name; });
return types;
}();
return s_audioTypes;
}
auto SampleDecoder::decode(const QString& audioFile) -> std::optional<Result>
{
auto result = std::optional<Result>{};
for (const auto& decoder : decoders)
{
result = decoder(audioFile);
if (result) { break; }
}
return result;
}
} // namespace lmms

17
src/core/SamplePlayHandle.cpp
View File

@@ -35,9 +35,9 @@ namespace lmms
{
SamplePlayHandle::SamplePlayHandle( SampleBuffer* sampleBuffer , bool ownAudioPort ) :
SamplePlayHandle::SamplePlayHandle(Sample* sample, bool ownAudioPort) :
PlayHandle( Type::SamplePlayHandle ),
m_sampleBuffer( sharedObject::ref( sampleBuffer ) ),
m_sample(sample),
m_doneMayReturnTrue( true ),
m_frame( 0 ),
m_ownAudioPort( ownAudioPort ),
@@ -56,16 +56,15 @@ SamplePlayHandle::SamplePlayHandle( SampleBuffer* sampleBuffer , bool ownAudioPo
SamplePlayHandle::SamplePlayHandle( const QString& sampleFile ) :
SamplePlayHandle( new SampleBuffer( sampleFile ) , true)
SamplePlayHandle(new Sample(sampleFile), true)
{
sharedObject::unref( m_sampleBuffer );
}
SamplePlayHandle::SamplePlayHandle( SampleClip* clip ) :
SamplePlayHandle( clip->sampleBuffer() , false)
SamplePlayHandle(&clip->sample(), false)
{
m_track = clip->getTrack();
setAudioPort( ( (SampleTrack *)clip->getTrack() )->audioPort() );
@@ -76,10 +75,10 @@ SamplePlayHandle::SamplePlayHandle( SampleClip* clip ) :
SamplePlayHandle::~SamplePlayHandle()
{
sharedObject::unref( m_sampleBuffer );
if( m_ownAudioPort )
{
delete audioPort();
delete m_sample;
}
}
@@ -115,7 +114,7 @@ void SamplePlayHandle::play( sampleFrame * buffer )
m_volumeModel->value() / DefaultVolume } };*/
// SamplePlayHandle always plays the sample at its original pitch;
// it is used only for previews, SampleTracks and the metronome.
if (!m_sampleBuffer->play(workingBuffer, &m_state, frames, DefaultBaseFreq))
if (!m_sample->play(workingBuffer, &m_state, frames, DefaultBaseFreq))
{
memset(workingBuffer, 0, frames * sizeof(sampleFrame));
}
@@ -145,8 +144,8 @@ bool SamplePlayHandle::isFromTrack( const Track * _track ) const
f_cnt_t SamplePlayHandle::totalFrames() const
{
return ( m_sampleBuffer->endFrame() - m_sampleBuffer->startFrame() ) *
( Engine::audioEngine()->processingSampleRate() / m_sampleBuffer->sampleRate() );
return (m_sample->endFrame() - m_sample->startFrame()) *
(static_cast<float>(Engine::audioEngine()->processingSampleRate()) / m_sample->sampleRate());
}

31
src/core/SampleRecordHandle.cpp
View File

@@ -51,13 +51,8 @@ SampleRecordHandle::SampleRecordHandle( SampleClip* clip ) :
SampleRecordHandle::~SampleRecordHandle()
{
if( !m_buffers.empty() )
{
SampleBuffer* sb;
createSampleBuffer( &sb );
m_clip->setSampleBuffer( sb );
}
if (!m_buffers.empty()) { m_clip->setSampleBuffer(createSampleBuffer()); }
while( !m_buffers.empty() )
{
delete[] m_buffers.front().first;
@@ -111,28 +106,22 @@ f_cnt_t SampleRecordHandle::framesRecorded() const
void SampleRecordHandle::createSampleBuffer( SampleBuffer** sampleBuf )
std::shared_ptr<const SampleBuffer> SampleRecordHandle::createSampleBuffer()
{
const f_cnt_t frames = framesRecorded();
// create buffer to store all recorded buffers in
auto data = new sampleFrame[frames];
// make sure buffer is cleaned up properly at the end...
sampleFrame * data_ptr = data;
assert( data != nullptr );
auto bigBuffer = std::vector<sampleFrame>(frames);
// now copy all buffers into big buffer
for( bufferList::const_iterator it = m_buffers.begin(); it != m_buffers.end(); ++it )
auto framesCopied = 0;
for (const auto& [buf, numFrames] : m_buffers)
{
memcpy( data_ptr, ( *it ).first, ( *it ).second *
sizeof( sampleFrame ) );
data_ptr += ( *it ).second;
std::copy_n(buf, numFrames, bigBuffer.begin() + framesCopied);
framesCopied += numFrames;
}
// create according sample-buffer out of big buffer
*sampleBuf = new SampleBuffer( data, frames );
( *sampleBuf)->setSampleRate( Engine::audioEngine()->inputSampleRate() );
delete[] data;
return std::make_shared<const SampleBuffer>(std::move(bigBuffer), Engine::audioEngine()->inputSampleRate());
}

5
src/core/Track.cpp
View File

@@ -283,10 +283,9 @@ void Track::loadSettings( const QDomElement & element )
return;
}
while( !m_clips.empty() )
{
delete m_clips.front();
// m_clips.erase( m_clips.begin() );
auto guard = Engine::audioEngine()->requestChangesGuard();
deleteClips();
}
QDomNode node = element.firstChild();

26
src/core/audio/AudioSampleRecorder.cpp
View File

@@ -67,32 +67,22 @@ f_cnt_t AudioSampleRecorder::framesRecorded() const
return frames;
}
void AudioSampleRecorder::createSampleBuffer( SampleBuffer** sampleBuf )
std::shared_ptr<const SampleBuffer> AudioSampleRecorder::createSampleBuffer()
{
const f_cnt_t frames = framesRecorded();
// create buffer to store all recorded buffers in
auto data = new sampleFrame[frames];
// make sure buffer is cleaned up properly at the end...
sampleFrame * data_ptr = data;
assert( data != nullptr );
auto bigBuffer = std::vector<sampleFrame>(frames);
// now copy all buffers into big buffer
for( BufferList::ConstIterator it = m_buffers.begin();
it != m_buffers.end(); ++it )
auto framesCopied = 0;
for (const auto& [buf, numFrames] : m_buffers)
{
memcpy( data_ptr, ( *it ).first, ( *it ).second *
sizeof( sampleFrame ) );
data_ptr += ( *it ).second;
std::copy_n(buf, numFrames, bigBuffer.begin() + framesCopied);
framesCopied += numFrames;
}
// create according sample-buffer out of big buffer
*sampleBuf = new SampleBuffer( data, frames );
( *sampleBuf )->setSampleRate( sampleRate() );
delete[] data;
return std::make_shared<const SampleBuffer>(std::move(bigBuffer), sampleRate());
}

2
src/gui/CMakeLists.txt
View File

@@ -34,7 +34,9 @@ SET(LMMS_SRCS
gui/PluginBrowser.cpp
gui/ProjectNotes.cpp
gui/RowTableView.cpp
gui/SampleLoader.cpp
gui/SampleTrackWindow.cpp
gui/SampleWaveform.cpp
gui/SendButtonIndicator.cpp
gui/SideBar.cpp
gui/SideBarWidget.cpp

11
src/gui/FileBrowser.cpp
View File

@@ -57,7 +57,9 @@
#include "PatternStore.h"
#include "PluginFactory.h"
#include "PresetPreviewPlayHandle.h"
#include "Sample.h"
#include "SampleClip.h"
#include "SampleLoader.h"
#include "SamplePlayHandle.h"
#include "SampleTrack.h"
#include "Song.h"
@@ -715,9 +717,12 @@ void FileBrowserTreeWidget::previewFileItem(FileItem* file)
embed::getIconPixmap("sample_file", 24, 24), 0);
// TODO: this can be removed once we do this outside the event thread
qApp->processEvents(QEventLoop::ExcludeUserInputEvents);
auto s = new SamplePlayHandle(fileName);
s->setDoneMayReturnTrue(false);
newPPH = s;
if (auto buffer = SampleLoader::createBufferFromFile(fileName))
{
auto s = new SamplePlayHandle(new lmms::Sample{std::move(buffer)});
s->setDoneMayReturnTrue(false);
newPPH = s;
}
delete tf;
}
else if (

17
src/gui/LfoControllerDialog.cpp
View File

@@ -31,6 +31,7 @@
#include "Knob.h"
#include "TempoSyncKnob.h"
#include "PixmapButton.h"
#include "SampleLoader.h"
namespace lmms::gui
{
@@ -210,14 +211,14 @@ LfoControllerDialog::~LfoControllerDialog()
void LfoControllerDialog::askUserDefWave()
{
SampleBuffer * sampleBuffer = dynamic_cast<LfoController*>(this->model())->
m_userDefSampleBuffer;
QString fileName = sampleBuffer->openAndSetWaveformFile();
if( fileName.isEmpty() == false )
{
// TODO:
m_userWaveBtn->setToolTip(sampleBuffer->audioFile());
}
const auto fileName = SampleLoader::openWaveformFile();
if (fileName.isEmpty()) { return; }
auto lfoModel = dynamic_cast<LfoController*>(model());
auto& buffer = lfoModel->m_userDefSampleBuffer;
buffer = SampleLoader::createBufferFromFile(fileName);
m_userWaveBtn->setToolTip(buffer->audioFile());
}

126
src/gui/SampleLoader.cpp Normal file
View File

@@ -0,0 +1,126 @@
/*
* SampleLoader.cpp - Static functions that open audio files
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
* 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.
*
*/
#include "SampleLoader.h"
#include <QFileInfo>
#include <QMessageBox>
#include <memory>
#include "ConfigManager.h"
#include "FileDialog.h"
#include "GuiApplication.h"
#include "PathUtil.h"
#include "SampleDecoder.h"
#include "Song.h"
namespace lmms::gui {
QString SampleLoader::openAudioFile(const QString& previousFile)
{
auto openFileDialog = FileDialog(nullptr, QObject::tr("Open audio file"));
auto dir = !previousFile.isEmpty() ? PathUtil::toAbsolute(previousFile) : ConfigManager::inst()->userSamplesDir();
// change dir to position of previously opened file
openFileDialog.setDirectory(dir);
openFileDialog.setFileMode(FileDialog::ExistingFiles);
// set filters
auto fileTypes = QStringList{};
auto allFileTypes = QStringList{};
auto nameFilters = QStringList{};
const auto& supportedAudioTypes = SampleDecoder::supportedAudioTypes();
for (const auto& audioType : supportedAudioTypes)
{
const auto name = QString::fromStdString(audioType.name);
const auto extension = QString::fromStdString(audioType.extension);
const auto displayExtension = QString{"*.%1"}.arg(extension);
fileTypes.append(QString{"%1 (%2)"}.arg(FileDialog::tr("%1 files").arg(name), displayExtension));
allFileTypes.append(displayExtension);
}
nameFilters.append(QString{"%1 (%2)"}.arg(FileDialog::tr("All audio files"), allFileTypes.join(" ")));
nameFilters.append(fileTypes);
nameFilters.append(QString("%1 (*)").arg(FileDialog::tr("Other files")));
openFileDialog.setNameFilters(nameFilters);
if (!previousFile.isEmpty())
{
// select previously opened file
openFileDialog.selectFile(QFileInfo{previousFile}.fileName());
}
if (openFileDialog.exec() == QDialog::Accepted)
{
if (openFileDialog.selectedFiles().isEmpty()) { return ""; }
return PathUtil::toShortestRelative(openFileDialog.selectedFiles()[0]);
}
return "";
}
QString SampleLoader::openWaveformFile(const QString& previousFile)
{
return openAudioFile(
previousFile.isEmpty() ? ConfigManager::inst()->factorySamplesDir() + "waveforms/10saw.flac" : previousFile);
}
std::shared_ptr<const SampleBuffer> SampleLoader::createBufferFromFile(const QString& filePath)
{
if (filePath.isEmpty()) { return SampleBuffer::emptyBuffer(); }
try
{
return std::make_shared<SampleBuffer>(filePath);
}
catch (const std::runtime_error& error)
{
if (getGUI()) { displayError(QString::fromStdString(error.what())); }
return SampleBuffer::emptyBuffer();
}
}
std::shared_ptr<const SampleBuffer> SampleLoader::createBufferFromBase64(const QString& base64, int sampleRate)
{
if (base64.isEmpty()) { return SampleBuffer::emptyBuffer(); }
try
{
return std::make_shared<SampleBuffer>(base64, sampleRate);
}
catch (const std::runtime_error& error)
{
if (getGUI()) { displayError(QString::fromStdString(error.what())); }
return SampleBuffer::emptyBuffer();
}
}
void SampleLoader::displayError(const QString& message)
{
QMessageBox::critical(nullptr, QObject::tr("Error loading sample"), message);
}
} // namespace lmms::gui

94
src/gui/SampleWaveform.cpp Normal file
View File

@@ -0,0 +1,94 @@
/*
* SampleWaveform.cpp
*
* Copyright (c) 2023 saker <sakertooth@gmail.com>
*
* 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.
*
*/
#include "SampleWaveform.h"
namespace lmms::gui {
void SampleWaveform::visualize(const Sample& sample, QPainter& p, const QRect& dr, int fromFrame, int toFrame)
{
if (sample.sampleSize() == 0) { return; }
const auto x = dr.x();
const auto height = dr.height();
const auto width = dr.width();
const auto centerY = dr.center().y();
const auto halfHeight = height / 2;
const auto buffer = sample.data() + fromFrame;
const auto color = p.pen().color();
const auto rmsColor = color.lighter(123);
auto numFrames = toFrame - fromFrame;
if (numFrames == 0) { numFrames = sample.sampleSize(); }
const auto framesPerPixel = std::max(1, numFrames / width);
constexpr auto maxFramesPerPixel = 512;
const auto resolution = std::max(1, framesPerPixel / maxFramesPerPixel);
const auto framesPerResolution = framesPerPixel / resolution;
const auto numPixels = std::min(numFrames, width);
auto min = std::vector<float>(numPixels, 1);
auto max = std::vector<float>(numPixels, -1);
auto squared = std::vector<float>(numPixels);
const auto maxFrames = numPixels * framesPerPixel;
for (int i = 0; i < maxFrames; i += resolution)
{
const auto pixelIndex = i / framesPerPixel;
const auto value = std::accumulate(buffer[i].begin(), buffer[i].end(), 0.0f) / buffer[i].size();
if (value > max[pixelIndex]) { max[pixelIndex] = value; }
if (value < min[pixelIndex]) { min[pixelIndex] = value; }
squared[pixelIndex] += value * value;
}
const auto amplification = sample.amplification();
const auto reversed = sample.reversed();
for (int i = 0; i < numPixels; i++)
{
const auto lineY1 = centerY - max[i] * halfHeight * amplification;
const auto lineY2 = centerY - min[i] * halfHeight * amplification;
auto lineX = i + x;
if (reversed) { lineX = width - lineX; }
p.drawLine(lineX, lineY1, lineX, lineY2);
const auto rms = std::sqrt(squared[i] / framesPerResolution);
const auto maxRMS = std::clamp(rms, min[i], max[i]);
const auto minRMS = std::clamp(-rms, min[i], max[i]);
const auto rmsLineY1 = centerY - maxRMS * halfHeight * amplification;
const auto rmsLineY2 = centerY - minRMS * halfHeight * amplification;
p.setPen(rmsColor);
p.drawLine(lineX, rmsLineY1, lineX, rmsLineY2);
p.setPen(color);
}
}
} // namespace lmms::gui

11
src/gui/clips/ClipView.cpp
View File

@@ -294,6 +294,17 @@ void ClipView::remove()
// delete ourself
close();
if (m_clip->getTrack())
{
auto guard = Engine::audioEngine()->requestChangesGuard();
m_clip->getTrack()->removeClip(m_clip);
}
// TODO: Clip::~Clip should not be responsible for removing the Clip from the Track.
// One would expect that a call to Track::removeClip would already do that for you, as well
// as actually deleting the Clip with the deleteLater function. That being said, it shouldn't
// be possible to make a Clip without a Track (i.e., Clip::getTrack is never nullptr).
m_clip->deleteLater();
}

26
src/gui/clips/SampleClipView.cpp
View File

@@ -32,8 +32,9 @@
#include "AutomationEditor.h"
#include "embed.h"
#include "PathUtil.h"
#include "SampleBuffer.h"
#include "SampleClip.h"
#include "SampleLoader.h"
#include "SampleWaveform.h"
#include "Song.h"
#include "StringPairDrag.h"
@@ -62,9 +63,11 @@ void SampleClipView::updateSample()
update();
// set tooltip to filename so that user can see what sample this
// sample-clip contains
setToolTip(m_clip->m_sampleBuffer->audioFile() != "" ?
PathUtil::toAbsolute(m_clip->m_sampleBuffer->audioFile()) :
tr( "Double-click to open sample" ) );
setToolTip(
!m_clip->m_sample.sampleFile().isEmpty()
? PathUtil::toAbsolute(m_clip->m_sample.sampleFile())
: tr("Double-click to open sample")
);
}
@@ -120,8 +123,7 @@ void SampleClipView::dropEvent( QDropEvent * _de )
}
else if( StringPairDrag::decodeKey( _de ) == "sampledata" )
{
m_clip->m_sampleBuffer->loadFromBase64(
StringPairDrag::decodeValue( _de ) );
m_clip->setSampleBuffer(SampleLoader::createBufferFromBase64(StringPairDrag::decodeValue(_de)));
m_clip->updateLength();
update();
_de->accept();
@@ -179,12 +181,12 @@ void SampleClipView::mouseReleaseEvent(QMouseEvent *_me)
void SampleClipView::mouseDoubleClickEvent( QMouseEvent * )
{
QString af = m_clip->m_sampleBuffer->openAudioFile();
QString af = SampleLoader::openAudioFile();
if ( af.isEmpty() ) {} //Don't do anything if no file is loaded
else if ( af == m_clip->m_sampleBuffer->audioFile() )
else if (af == m_clip->m_sample.sampleFile())
{ //Instead of reloading the existing file, just reset the size
int length = (int) ( m_clip->m_sampleBuffer->frames() / Engine::framesPerTick() );
int length = static_cast<int>(m_clip->m_sample.sampleSize() / Engine::framesPerTick());
m_clip->changeLength(length);
}
else
@@ -267,9 +269,9 @@ void SampleClipView::paintEvent( QPaintEvent * pe )
float offset = m_clip->startTimeOffset() / ticksPerBar * pixelsPerBar();
QRect r = QRect( offset, spacing,
qMax( static_cast<int>( m_clip->sampleLength() * ppb / ticksPerBar ), 1 ), rect().bottom() - 2 * spacing );
m_clip->m_sampleBuffer->visualize( p, r, pe->rect() );
SampleWaveform::visualize(m_clip->m_sample, p, r);
QString name = PathUtil::cleanName(m_clip->m_sampleBuffer->audioFile());
QString name = PathUtil::cleanName(m_clip->m_sample.sampleFile());
paintTextLabel(name, p);
// disable antialiasing for borders, since its not needed
@@ -322,7 +324,7 @@ void SampleClipView::paintEvent( QPaintEvent * pe )
void SampleClipView::reverseSample()
{
m_clip->sampleBuffer()->setReversed(!m_clip->sampleBuffer()->reversed());
m_clip->m_sample.setReversed(!m_clip->m_sample.reversed());
Engine::getSong()->setModified();
update();
}

7
src/gui/editors/AutomationEditor.cpp
View File

@@ -39,6 +39,7 @@
#include <cmath>
#include "SampleClip.h"
#include "SampleWaveform.h"
#ifndef __USE_XOPEN
#define __USE_XOPEN
@@ -1235,9 +1236,9 @@ void AutomationEditor::paintEvent(QPaintEvent * pe )
}
// draw ghost sample
if (m_ghostSample != nullptr && m_ghostSample->sampleBuffer()->frames() > 1 && m_renderSample)
if (m_ghostSample != nullptr && m_ghostSample->sample().sampleSize() > 1 && m_renderSample)
{
int sampleFrames = m_ghostSample->sampleBuffer()->frames();
int sampleFrames = m_ghostSample->sample().sampleSize();
int length = static_cast<float>(sampleFrames) / Engine::framesPerTick();
int editorHeight = grid_bottom - TOP_MARGIN;
@@ -1247,7 +1248,7 @@ void AutomationEditor::paintEvent(QPaintEvent * pe )
int yOffset = (editorHeight - sampleHeight) / 2.0f + TOP_MARGIN;
p.setPen(m_ghostSampleColor);
m_ghostSample->sampleBuffer()->visualize(p, QRect(startPos, yOffset, sampleWidth, sampleHeight), 0, sampleFrames);
SampleWaveform::visualize(m_ghostSample->sample(), p, QRect(startPos, yOffset, sampleWidth, sampleHeight), 0, sampleFrames);
}
// draw ghost notes

17
src/gui/instrument/EnvelopeAndLfoView.cpp
View File

@@ -28,6 +28,7 @@
#include "EnvelopeAndLfoView.h"
#include "EnvelopeAndLfoParameters.h"
#include "SampleLoader.h"
#include "embed.h"
#include "Engine.h"
#include "gui_templates.h"
@@ -306,8 +307,7 @@ void EnvelopeAndLfoView::dropEvent( QDropEvent * _de )
QString value = StringPairDrag::decodeValue( _de );
if( type == "samplefile" )
{
m_params->m_userWave.setAudioFile(
StringPairDrag::decodeValue( _de ) );
m_params->m_userWave = SampleLoader::createBufferFromFile(value);
m_userLfoBtn->model()->setValue( true );
m_params->m_lfoWaveModel.setValue(static_cast<int>(EnvelopeAndLfoParameters::LfoShape::UserDefinedWave));
_de->accept();
@@ -316,9 +316,10 @@ void EnvelopeAndLfoView::dropEvent( QDropEvent * _de )
else if( type == QString( "clip_%1" ).arg( static_cast<int>(Track::Type::Sample) ) )
{
DataFile dataFile( value.toUtf8() );
m_params->m_userWave.setAudioFile( dataFile.content().
auto file = dataFile.content().
firstChildElement().firstChildElement().
firstChildElement().attribute( "src" ) );
firstChildElement().attribute("src");
m_params->m_userWave = SampleLoader::createBufferFromFile(file);
m_userLfoBtn->model()->setValue( true );
m_params->m_lfoWaveModel.setValue(static_cast<int>(EnvelopeAndLfoParameters::LfoShape::UserDefinedWave));
_de->accept();
@@ -428,8 +429,6 @@ void EnvelopeAndLfoView::paintEvent( QPaintEvent * )
osc_frames *= 100.0f;
}
// userWaveSample() may be used, called out of loop for efficiency
m_params->m_userWave.dataReadLock();
float old_y = 0;
for( int x = 0; x <= LFO_GRAPH_W; ++x )
{
@@ -465,8 +464,7 @@ void EnvelopeAndLfoView::paintEvent( QPaintEvent * )
val = m_randomGraph;
break;
case EnvelopeAndLfoParameters::LfoShape::UserDefinedWave:
val = m_params->m_userWave.
userWaveSample( phase );
val = Oscillator::userWaveSample(m_params->m_userWave.get(), phase);
break;
}
if( static_cast<f_cnt_t>( cur_sample ) <=
@@ -481,7 +479,6 @@ void EnvelopeAndLfoView::paintEvent( QPaintEvent * )
graph_y_base + cur_y ) );
old_y = cur_y;
}
m_params->m_userWave.dataUnlock();
p.setPen( QColor( 201, 201, 225 ) );
int ms_per_osc = static_cast<int>( SECS_PER_LFO_OSCILLATION *
@@ -499,7 +496,7 @@ void EnvelopeAndLfoView::lfoUserWaveChanged()
if( static_cast<EnvelopeAndLfoParameters::LfoShape>(m_params->m_lfoWaveModel.value()) ==
EnvelopeAndLfoParameters::LfoShape::UserDefinedWave )
{
if( m_params->m_userWave.frames() <= 1 )
if (m_params->m_userWave->size() <= 1)
{
TextFloat::displayMessage( tr( "Hint" ),
tr( "Drag and drop a sample into this window." ),

12
src/gui/widgets/Graph.cpp
View File

@@ -26,6 +26,7 @@
#include <QPainter>
#include "Graph.h"
#include "SampleLoader.h"
#include "StringPairDrag.h"
#include "SampleBuffer.h"
#include "Oscillator.h"
@@ -588,21 +589,16 @@ void graphModel::setWaveToNoise()
QString graphModel::setWaveToUser()
{
auto sampleBuffer = new SampleBuffer;
QString fileName = sampleBuffer->openAndSetWaveformFile();
QString fileName = gui::SampleLoader::openWaveformFile();
if( fileName.isEmpty() == false )
{
sampleBuffer->dataReadLock();
auto sampleBuffer = gui::SampleLoader::createBufferFromFile(fileName);
for( int i = 0; i < length(); i++ )
{
m_samples[i] = sampleBuffer->userWaveSample(
i / static_cast<float>( length() ) );
m_samples[i] = Oscillator::userWaveSample(sampleBuffer.get(), i / static_cast<float>(length()));
}
sampleBuffer->dataUnlock();
}
sharedObject::unref( sampleBuffer );
emit samplesChanged( 0, length() - 1 );
return fileName;
};

4
src/tracks/SampleTrack.cpp
View File

@@ -108,10 +108,10 @@ bool SampleTrack::play( const TimePos & _start, const fpp_t _frames,
{
if( sClip->isPlaying() == false && _start >= (sClip->startPosition() + sClip->startTimeOffset()) )
{
auto bufferFramesPerTick = Engine::framesPerTick (sClip->sampleBuffer ()->sampleRate ());
auto bufferFramesPerTick = Engine::framesPerTick(sClip->sample().sampleRate());
f_cnt_t sampleStart = bufferFramesPerTick * ( _start - sClip->startPosition() - sClip->startTimeOffset() );
f_cnt_t clipFrameLength = bufferFramesPerTick * ( sClip->endPosition() - sClip->startPosition() - sClip->startTimeOffset() );
f_cnt_t sampleBufferLength = sClip->sampleBuffer()->frames();
f_cnt_t sampleBufferLength = sClip->sample().sampleSize();
//if the Clip smaller than the sample length we play only until Clip end
//else we play the sample to the end but nothing more
f_cnt_t samplePlayLength = clipFrameLength > sampleBufferLength ? sampleBufferLength : clipFrameLength;