Files
lmms/include/AudioEngine.h
saker 7268827624 Revamp synchronization with the audio engine (#6881)
The revamp consists of one lock. When the audio thread needs to render audio or another thread wants to run a change, acquiring the lock grants mutual exclusion to do one of the two. The intention is that this will provide stronger guarantees that changes do not run concurrently with the audio thread, as well as having the synchronization mechanism itself be free of data races (verified with TSan).
2023-11-18 15:28:01 -05:00

482 lines
9.6 KiB
C++

/*
* AudioEngine.h - device-independent audio engine for LMMS
*
* Copyright (c) 2004-2014 Tobias Doerffel <tobydox/at/users.sourceforge.net>
*
* This file is part of LMMS - https://lmms.io
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* General Public License for more details.
*
* You should have received a copy of the GNU General Public
* License along with this program (see COPYING); if not, write to the
* Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
* Boston, MA 02110-1301 USA.
*
*/
#ifndef LMMS_AUDIO_ENGINE_H
#define LMMS_AUDIO_ENGINE_H
#ifdef __MINGW32__
#include <mingw.mutex.h>
#else
#include <mutex>
#endif
#include <QThread>
#include <samplerate.h>
#include <vector>
#include "lmms_basics.h"
#include "LocklessList.h"
#include "FifoBuffer.h"
#include "AudioEngineProfiler.h"
#include "PlayHandle.h"
namespace lmms
{
class AudioDevice;
class MidiClient;
class AudioPort;
class AudioEngineWorkerThread;
const fpp_t MINIMUM_BUFFER_SIZE = 32;
const fpp_t DEFAULT_BUFFER_SIZE = 256;
const int BYTES_PER_SAMPLE = sizeof( sample_t );
const int BYTES_PER_INT_SAMPLE = sizeof( int_sample_t );
const int BYTES_PER_FRAME = sizeof( sampleFrame );
const int BYTES_PER_SURROUND_FRAME = sizeof( surroundSampleFrame );
const float OUTPUT_SAMPLE_MULTIPLIER = 32767.0f;
class LMMS_EXPORT AudioEngine : public QObject
{
Q_OBJECT
public:
/**
* @brief RAII helper for requestChangesInModel.
* Used by AudioEngine::requestChangesGuard.
*/
class RequestChangesGuard {
friend class AudioEngine;
private:
RequestChangesGuard(AudioEngine* audioEngine)
: m_audioEngine{audioEngine}
{
m_audioEngine->requestChangeInModel();
}
public:
RequestChangesGuard()
: m_audioEngine{nullptr}
{
}
RequestChangesGuard(RequestChangesGuard&& other)
: RequestChangesGuard()
{
std::swap(other.m_audioEngine, m_audioEngine);
}
// Disallow copy.
RequestChangesGuard(const RequestChangesGuard&) = delete;
RequestChangesGuard& operator=(const RequestChangesGuard&) = delete;
~RequestChangesGuard() {
if (m_audioEngine) {
m_audioEngine->doneChangeInModel();
}
}
private:
AudioEngine* m_audioEngine;
};
struct qualitySettings
{
enum class Mode
{
Draft,
HighQuality,
FinalMix
} ;
enum class Interpolation
{
Linear,
SincFastest,
SincMedium,
SincBest
} ;
enum class Oversampling
{
None,
X2,
X4,
X8
} ;
Interpolation interpolation;
Oversampling oversampling;
qualitySettings(Mode m)
{
switch (m)
{
case Mode::Draft:
interpolation = Interpolation::Linear;
oversampling = Oversampling::None;
break;
case Mode::HighQuality:
interpolation =
Interpolation::SincFastest;
oversampling = Oversampling::X2;
break;
case Mode::FinalMix:
interpolation = Interpolation::SincBest;
oversampling = Oversampling::X8;
break;
}
}
qualitySettings(Interpolation i, Oversampling o) :
interpolation(i),
oversampling(o)
{
}
int sampleRateMultiplier() const
{
switch( oversampling )
{
case Oversampling::None: return 1;
case Oversampling::X2: return 2;
case Oversampling::X4: return 4;
case Oversampling::X8: return 8;
}
return 1;
}
int libsrcInterpolation() const
{
switch( interpolation )
{
case Interpolation::Linear:
return SRC_ZERO_ORDER_HOLD;
case Interpolation::SincFastest:
return SRC_SINC_FASTEST;
case Interpolation::SincMedium:
return SRC_SINC_MEDIUM_QUALITY;
case Interpolation::SincBest:
return SRC_SINC_BEST_QUALITY;
}
return SRC_LINEAR;
}
} ;
void initDevices();
void clear();
void clearNewPlayHandles();
// audio-device-stuff
bool renderOnly() const { return m_renderOnly; }
// Returns the current audio device's name. This is not necessarily
// the user's preferred audio device, in case you were thinking that.
inline const QString & audioDevName() const
{
return m_audioDevName;
}
inline bool audioDevStartFailed() const
{
return m_audioDevStartFailed;
}
//! Set new audio device. Old device will be deleted,
//! unless it's stored using storeAudioDevice
void setAudioDevice( AudioDevice * _dev,
const struct qualitySettings & _qs,
bool _needs_fifo,
bool startNow );
void storeAudioDevice();
void restoreAudioDevice();
inline AudioDevice * audioDev()
{
return m_audioDev;
}
// audio-port-stuff
inline void addAudioPort(AudioPort * port)
{
requestChangeInModel();
m_audioPorts.push_back(port);
doneChangeInModel();
}
void removeAudioPort(AudioPort * port);
// MIDI-client-stuff
inline const QString & midiClientName() const
{
return m_midiClientName;
}
inline MidiClient * midiClient()
{
return m_midiClient;
}
// play-handle stuff
bool addPlayHandle( PlayHandle* handle );
void removePlayHandle( PlayHandle* handle );
inline PlayHandleList& playHandles()
{
return m_playHandles;
}
void removePlayHandlesOfTypes(Track * track, PlayHandle::Types types);
// methods providing information for other classes
inline fpp_t framesPerPeriod() const
{
return m_framesPerPeriod;
}
AudioEngineProfiler& profiler()
{
return m_profiler;
}
int cpuLoad() const
{
return m_profiler.cpuLoad();
}
int detailLoad(const AudioEngineProfiler::DetailType type) const
{
return m_profiler.detailLoad(type);
}
const qualitySettings & currentQualitySettings() const
{
return m_qualitySettings;
}
sample_rate_t baseSampleRate() const;
sample_rate_t outputSampleRate() const;
sample_rate_t inputSampleRate() const;
sample_rate_t processingSampleRate() const;
inline float masterGain() const
{
return m_masterGain;
}
inline void setMasterGain(const float mo)
{
m_masterGain = mo;
}
static inline sample_t clip(const sample_t s)
{
if (s > 1.0f)
{
return 1.0f;
}
else if (s < -1.0f)
{
return -1.0f;
}
return s;
}
struct StereoSample
{
StereoSample(sample_t _left, sample_t _right) : left(_left), right(_right) {}
sample_t left;
sample_t right;
};
StereoSample getPeakValues(sampleFrame * ab, const f_cnt_t _frames) const;
bool criticalXRuns() const;
inline bool hasFifoWriter() const
{
return m_fifoWriter != nullptr;
}
void pushInputFrames( sampleFrame * _ab, const f_cnt_t _frames );
inline const sampleFrame * inputBuffer()
{
return m_inputBuffer[ m_inputBufferRead ];
}
inline f_cnt_t inputBufferFrames() const
{
return m_inputBufferFrames[ m_inputBufferRead ];
}
inline const surroundSampleFrame * nextBuffer()
{
return hasFifoWriter() ? m_fifo->read() : renderNextBuffer();
}
void changeQuality(const struct qualitySettings & qs);
inline bool isMetronomeActive() const { return m_metronomeActive; }
inline void setMetronomeActive(bool value = true) { m_metronomeActive = value; }
//! Block until a change in model can be done (i.e. wait for audio thread)
void requestChangeInModel();
void doneChangeInModel();
RequestChangesGuard requestChangesGuard()
{
return RequestChangesGuard{this};
}
static bool isAudioDevNameValid(QString name);
static bool isMidiDevNameValid(QString name);
signals:
void qualitySettingsChanged();
void sampleRateChanged();
void nextAudioBuffer( const lmms::surroundSampleFrame * buffer );
private:
using Fifo = FifoBuffer<surroundSampleFrame*>;
class fifoWriter : public QThread
{
public:
fifoWriter( AudioEngine * audioEngine, Fifo * fifo );
void finish();
private:
AudioEngine * m_audioEngine;
Fifo * m_fifo;
volatile bool m_writing;
void run() override;
void write( surroundSampleFrame * buffer );
} ;
AudioEngine( bool renderOnly );
~AudioEngine() override;
void startProcessing(bool needsFifo = true);
void stopProcessing();
AudioDevice * tryAudioDevices();
MidiClient * tryMidiClients();
void renderStageNoteSetup();
void renderStageInstruments();
void renderStageEffects();
void renderStageMix();
const surroundSampleFrame * renderNextBuffer();
void swapBuffers();
void handleMetronome();
void clearInternal();
bool m_renderOnly;
std::vector<AudioPort *> m_audioPorts;
fpp_t m_framesPerPeriod;
sampleFrame * m_inputBuffer[2];
f_cnt_t m_inputBufferFrames[2];
f_cnt_t m_inputBufferSize[2];
int m_inputBufferRead;
int m_inputBufferWrite;
surroundSampleFrame * m_outputBufferRead;
surroundSampleFrame * m_outputBufferWrite;
// worker thread stuff
std::vector<AudioEngineWorkerThread *> m_workers;
int m_numWorkers;
// playhandle stuff
PlayHandleList m_playHandles;
// place where new playhandles are added temporarily
LocklessList<PlayHandle *> m_newPlayHandles;
ConstPlayHandleList m_playHandlesToRemove;
struct qualitySettings m_qualitySettings;
float m_masterGain;
// audio device stuff
void doSetAudioDevice( AudioDevice *_dev );
AudioDevice * m_audioDev;
AudioDevice * m_oldAudioDev;
QString m_audioDevName;
bool m_audioDevStartFailed;
// MIDI device stuff
MidiClient * m_midiClient;
QString m_midiClientName;
// FIFO stuff
Fifo * m_fifo;
fifoWriter * m_fifoWriter;
AudioEngineProfiler m_profiler;
bool m_metronomeActive;
bool m_clearSignal;
std::mutex m_changeMutex;
friend class Engine;
friend class AudioEngineWorkerThread;
friend class ProjectRenderer;
} ;
} // namespace lmms
#endif // LMMS_AUDIO_ENGINE_H