mirror of
https://github.com/LMMS/lmms.git
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430 lines
8.2 KiB
C++
430 lines
8.2 KiB
C++
/*
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* Mixer.h - audio-device-independent mixer for LMMS
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*
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* Copyright (c) 2004-2014 Tobias Doerffel <tobydox/at/users.sourceforge.net>
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*
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* This file is part of LMMS - http://lmms.io
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* General Public License for more details.
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*
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* You should have received a copy of the GNU General Public
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* License along with this program (see COPYING); if not, write to the
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* Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
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* Boston, MA 02110-1301 USA.
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*
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*/
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#ifndef MIXER_H
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#define MIXER_H
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#include "denormals.h"
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#include "lmmsconfig.h"
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#include <QtCore/QMutex>
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#include <QtCore/QThread>
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#include <QtCore/QVector>
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#include <QtCore/QWaitCondition>
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#include <samplerate.h>
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#include "lmms_basics.h"
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#include "LocklessList.h"
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#include "Note.h"
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#include "fifo_buffer.h"
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#include "MixerProfiler.h"
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class AudioDevice;
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class MidiClient;
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class AudioPort;
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const fpp_t MINIMUM_BUFFER_SIZE = 32;
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const fpp_t DEFAULT_BUFFER_SIZE = 256;
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const int BYTES_PER_SAMPLE = sizeof( sample_t );
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const int BYTES_PER_INT_SAMPLE = sizeof( int_sample_t );
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const int BYTES_PER_FRAME = sizeof( sampleFrame );
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const int BYTES_PER_SURROUND_FRAME = sizeof( surroundSampleFrame );
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const float OUTPUT_SAMPLE_MULTIPLIER = 32767.0f;
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const float BaseFreq = 440.0f;
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const Keys BaseKey = Key_A;
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const Octaves BaseOctave = DefaultOctave;
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#include "PlayHandle.h"
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class MixerWorkerThread;
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class EXPORT Mixer : public QObject
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{
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Q_OBJECT
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public:
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struct qualitySettings
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{
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enum Mode
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{
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Mode_Draft,
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Mode_HighQuality,
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Mode_FinalMix
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} ;
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enum Interpolation
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{
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Interpolation_Linear,
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Interpolation_SincFastest,
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Interpolation_SincMedium,
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Interpolation_SincBest
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} ;
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enum Oversampling
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{
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Oversampling_None,
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Oversampling_2x,
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Oversampling_4x,
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Oversampling_8x
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} ;
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Interpolation interpolation;
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Oversampling oversampling;
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qualitySettings( Mode _m )
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{
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switch( _m )
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{
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case Mode_Draft:
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interpolation = Interpolation_Linear;
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oversampling = Oversampling_None;
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break;
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case Mode_HighQuality:
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interpolation =
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Interpolation_SincFastest;
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oversampling = Oversampling_2x;
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break;
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case Mode_FinalMix:
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interpolation = Interpolation_SincBest;
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oversampling = Oversampling_8x;
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break;
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}
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}
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qualitySettings( Interpolation _i, Oversampling _o ) :
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interpolation( _i ),
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oversampling( _o )
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{
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}
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int sampleRateMultiplier() const
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{
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switch( oversampling )
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{
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case Oversampling_None: return 1;
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case Oversampling_2x: return 2;
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case Oversampling_4x: return 4;
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case Oversampling_8x: return 8;
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}
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return 1;
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}
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int libsrcInterpolation() const
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{
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switch( interpolation )
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{
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case Interpolation_Linear:
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return SRC_ZERO_ORDER_HOLD;
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case Interpolation_SincFastest:
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return SRC_SINC_FASTEST;
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case Interpolation_SincMedium:
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return SRC_SINC_MEDIUM_QUALITY;
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case Interpolation_SincBest:
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return SRC_SINC_BEST_QUALITY;
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}
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return SRC_LINEAR;
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}
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} ;
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void initDevices();
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void clear();
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// audio-device-stuff
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inline const QString & audioDevName() const
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{
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return m_audioDevName;
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}
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inline bool audioDevStartFailed() const
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{
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return m_audioDevStartFailed;
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}
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void setAudioDevice( AudioDevice * _dev );
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void setAudioDevice( AudioDevice * _dev,
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const struct qualitySettings & _qs,
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bool _needs_fifo );
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void restoreAudioDevice();
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inline AudioDevice * audioDev()
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{
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return m_audioDev;
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}
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// audio-port-stuff
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inline void addAudioPort( AudioPort * _port )
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{
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requestChangeInModel();
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m_audioPorts.push_back( _port );
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doneChangeInModel();
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}
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void removeAudioPort( AudioPort * _port );
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// MIDI-client-stuff
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inline const QString & midiClientName() const
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{
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return m_midiClientName;
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}
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inline MidiClient * midiClient()
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{
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return m_midiClient;
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}
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// play-handle stuff
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bool addPlayHandle( PlayHandle* handle );
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void removePlayHandle( PlayHandle* handle );
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inline PlayHandleList& playHandles()
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{
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return m_playHandles;
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}
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void removePlayHandlesOfTypes( Track * _track, const quint8 types );
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// methods providing information for other classes
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inline fpp_t framesPerPeriod() const
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{
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return m_framesPerPeriod;
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}
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MixerProfiler& profiler()
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{
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return m_profiler;
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}
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int cpuLoad() const
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{
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return m_profiler.cpuLoad();
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}
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const qualitySettings & currentQualitySettings() const
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{
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return m_qualitySettings;
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}
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sample_rate_t baseSampleRate() const;
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sample_rate_t outputSampleRate() const;
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sample_rate_t inputSampleRate() const;
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sample_rate_t processingSampleRate() const;
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inline float masterGain() const
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{
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return m_masterGain;
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}
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inline void setMasterGain( const float _mo )
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{
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m_masterGain = _mo;
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}
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static inline sample_t clip( const sample_t _s )
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{
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if( _s > 1.0f )
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{
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return 1.0f;
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}
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else if( _s < -1.0f )
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{
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return -1.0f;
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}
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return _s;
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}
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void getPeakValues( sampleFrame * _ab, const f_cnt_t _frames, float & peakLeft, float & peakRight ) const;
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bool criticalXRuns() const;
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inline bool hasFifoWriter() const
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{
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return m_fifoWriter != NULL;
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}
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void pushInputFrames( sampleFrame * _ab, const f_cnt_t _frames );
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inline const sampleFrame * inputBuffer()
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{
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return m_inputBuffer[ m_inputBufferRead ];
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}
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inline f_cnt_t inputBufferFrames() const
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{
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return m_inputBufferFrames[ m_inputBufferRead ];
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}
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inline const surroundSampleFrame * nextBuffer()
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{
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return hasFifoWriter() ? m_fifo->read() : renderNextBuffer();
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}
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void changeQuality( const struct qualitySettings & _qs );
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inline bool isMetronomeActive() const { return m_metronomeActive; }
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inline void setMetronomeActive(bool value = true) { m_metronomeActive = value; }
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void requestChangeInModel();
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void doneChangeInModel();
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signals:
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void qualitySettingsChanged();
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void sampleRateChanged();
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void nextAudioBuffer( const surroundSampleFrame * buffer );
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private:
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typedef fifoBuffer<surroundSampleFrame *> fifo;
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class fifoWriter : public QThread
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{
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public:
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fifoWriter( Mixer * _mixer, fifo * _fifo );
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void finish();
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private:
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Mixer * m_mixer;
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fifo * m_fifo;
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volatile bool m_writing;
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virtual void run();
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void write( surroundSampleFrame * buffer );
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} ;
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Mixer( bool renderOnly );
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virtual ~Mixer();
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void startProcessing( bool _needs_fifo = true );
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void stopProcessing();
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AudioDevice * tryAudioDevices();
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MidiClient * tryMidiClients();
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const surroundSampleFrame * renderNextBuffer();
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void clearInternal();
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void runChangesInModel();
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QVector<AudioPort *> m_audioPorts;
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fpp_t m_framesPerPeriod;
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sampleFrame * m_inputBuffer[2];
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f_cnt_t m_inputBufferFrames[2];
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f_cnt_t m_inputBufferSize[2];
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int m_inputBufferRead;
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int m_inputBufferWrite;
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surroundSampleFrame * m_readBuf;
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surroundSampleFrame * m_writeBuf;
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QVector<surroundSampleFrame *> m_bufferPool;
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int m_readBuffer;
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int m_writeBuffer;
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int m_poolDepth;
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// worker thread stuff
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QVector<MixerWorkerThread *> m_workers;
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int m_numWorkers;
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// playhandle stuff
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PlayHandleList m_playHandles;
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// place where new playhandles are added temporarily
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LocklessList<PlayHandle *> m_newPlayHandles;
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ConstPlayHandleList m_playHandlesToRemove;
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struct qualitySettings m_qualitySettings;
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float m_masterGain;
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bool m_isProcessing;
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// audio device stuff
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AudioDevice * m_audioDev;
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AudioDevice * m_oldAudioDev;
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QString m_audioDevName;
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bool m_audioDevStartFailed;
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// MIDI device stuff
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MidiClient * m_midiClient;
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QString m_midiClientName;
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// FIFO stuff
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fifo * m_fifo;
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fifoWriter * m_fifoWriter;
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MixerProfiler m_profiler;
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bool m_metronomeActive;
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bool m_clearSignal;
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bool m_changesSignal;
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unsigned int m_changes;
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QMutex m_changesMutex;
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QMutex m_doChangesMutex;
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QMutex m_waitChangesMutex;
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QWaitCondition m_changesMixerCondition;
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QWaitCondition m_changesRequestCondition;
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bool m_waitingForWrite;
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friend class LmmsCore;
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friend class MixerWorkerThread;
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} ;
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#endif
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