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https://github.com/LMMS/lmms.git
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The new CPU framework replaces the old BasicOps framework. It is more flexible and the build process isn't such a mess anymore (pre-compiled assembler files etc.). It will hopefully see some improvements and extensions soon. Signed-off-by: Tobias Doerffel <tobias.doerffel@gmail.com>
1173 lines
23 KiB
C++
1173 lines
23 KiB
C++
/*
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* mixer.cpp - audio-device-independent mixer for LMMS
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*
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* Copyright (c) 2004-2009 Tobias Doerffel <tobydox/at/users.sourceforge.net>
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*
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* This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net
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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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#include <math.h>
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#include "mixer.h"
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#include "fx_mixer.h"
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#include "play_handle.h"
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#include "effect.h"
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#include "song.h"
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#include "templates.h"
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#include "envelope_and_lfo_parameters.h"
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#include "note_play_handle.h"
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#include "instrument_track.h"
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#include "debug.h"
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#include "engine.h"
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#include "config_mgr.h"
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#include "audio_port.h"
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#include "sample_play_handle.h"
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#include "piano_roll.h"
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#include "micro_timer.h"
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#include "Cpu.h"
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#include "audio_device.h"
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#include "midi_client.h"
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// platform-specific audio-interface-classes
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#include "audio_alsa.h"
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#include "audio_jack.h"
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#include "audio_oss.h"
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#include "audio_portaudio.h"
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#include "audio_pulseaudio.h"
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#include "audio_sdl.h"
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#include "audio_dummy.h"
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// platform-specific midi-interface-classes
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#include "midi_alsa_raw.h"
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#include "midi_alsa_seq.h"
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#include "midi_oss.h"
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#include "midi_winmm.h"
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#include "midi_dummy.h"
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#ifdef LMMS_HAVE_PTHREAD_H
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#include <pthread.h>
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#endif
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static QVector<fx_ch_t> __fx_channel_jobs( NumFxChannels );
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class MixerWorkerThread : public QThread
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{
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public:
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enum JobTypes
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{
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InvalidJob,
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PlayHandle,
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AudioPortEffects,
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EffectChannel,
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NumJobTypes
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} ;
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struct JobQueueItem
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{
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JobQueueItem() :
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type( InvalidJob ),
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job( NULL ),
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param( 0 ),
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done( false )
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{
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}
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JobQueueItem( JobTypes _type, void * _job, int _param = 0 ) :
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type( _type ),
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job( _job ),
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param( _param ),
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done( false )
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{
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}
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JobTypes type;
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void * job;
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int param;
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QAtomicInt done;
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} ;
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struct JobQueue
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{
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#define JOB_QUEUE_SIZE 1024
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JobQueue() :
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queueSize( 0 )
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{
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}
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JobQueueItem items[JOB_QUEUE_SIZE];
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int queueSize;
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QAtomicInt itemsDone;
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} ;
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static JobQueue s_jobQueue;
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MixerWorkerThread( int _worker_num, mixer * _mixer ) :
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QThread( _mixer ),
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m_workingBuf( CPU::allocFrames( _mixer->framesPerPeriod() ) ),
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m_workerNum( _worker_num ),
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m_quit( false ),
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m_mixer( _mixer ),
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m_queueReadyWaitCond( &m_mixer->m_queueReadyWaitCond )
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{
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}
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virtual ~MixerWorkerThread()
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{
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CPU::freeFrames( m_workingBuf );
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}
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virtual void quit( void )
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{
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m_quit = true;
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}
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void processJobQueue( void );
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private:
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virtual void run( void )
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{
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#if 0
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#ifdef LMMS_BUILD_LINUX
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#ifdef LMMS_HAVE_PTHREAD_H
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cpu_set_t mask;
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CPU_ZERO( &mask );
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CPU_SET( m_workerNum, &mask );
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pthread_setaffinity_np( pthread_self(), sizeof( mask ), &mask );
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#endif
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#endif
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#endif
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QMutex m;
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while( m_quit == false )
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{
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m.lock();
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m_queueReadyWaitCond->wait( &m );
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processJobQueue();
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m.unlock();
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}
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}
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sampleFrame * m_workingBuf;
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int m_workerNum;
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volatile bool m_quit;
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mixer * m_mixer;
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QWaitCondition * m_queueReadyWaitCond;
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} ;
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MixerWorkerThread::JobQueue MixerWorkerThread::s_jobQueue;
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void MixerWorkerThread::processJobQueue( void )
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{
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for( int i = 0; i < s_jobQueue.queueSize; ++i )
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{
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JobQueueItem * it = &s_jobQueue.items[i];
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if( it->done.fetchAndStoreOrdered( 1 ) == 0 )
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{
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switch( it->type )
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{
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case PlayHandle:
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( (playHandle *) it->job )->
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play( m_workingBuf );
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break;
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case AudioPortEffects:
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{
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audioPort * a = (audioPort *) it->job;
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const bool me = a->processEffects();
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if( me || a->m_bufferUsage != audioPort::NoUsage )
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{
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engine::getFxMixer()->mixToChannel( a->firstBuffer(),
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a->nextFxChannel() );
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a->nextPeriod();
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}
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}
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break;
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case EffectChannel:
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engine::getFxMixer()->processChannel( (fx_ch_t) it->param );
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break;
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default:
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break;
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}
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s_jobQueue.itemsDone.fetchAndAddOrdered( 1 );
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}
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}
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}
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#define FILL_JOB_QUEUE_BEGIN(_vec_type,_vec,_condition) \
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MixerWorkerThread::s_jobQueue.queueSize = 0; \
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MixerWorkerThread::s_jobQueue.itemsDone = 0; \
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for( _vec_type::Iterator it = _vec.begin(); \
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it != _vec.end(); ++it ) \
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{ \
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if( _condition ) \
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{
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#define FILL_JOB_QUEUE_END() \
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++MixerWorkerThread::s_jobQueue.queueSize; \
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} \
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}
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#define FILL_JOB_QUEUE(_vec_type,_vec,_job_type,_condition) \
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FILL_JOB_QUEUE_BEGIN(_vec_type,_vec,_condition) \
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MixerWorkerThread::s_jobQueue.items \
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[MixerWorkerThread::s_jobQueue.queueSize] = \
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MixerWorkerThread::JobQueueItem( _job_type, \
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(void *) *it ); \
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FILL_JOB_QUEUE_END()
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#define FILL_JOB_QUEUE_PARAM(_vec_type,_vec,_job_type,_condition) \
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FILL_JOB_QUEUE_BEGIN(_vec_type,_vec,_condition) \
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MixerWorkerThread::s_jobQueue.items \
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[MixerWorkerThread::s_jobQueue.queueSize] = \
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MixerWorkerThread::JobQueueItem( _job_type, \
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NULL, *it ); \
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FILL_JOB_QUEUE_END()
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#define START_JOBS() \
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m_queueReadyWaitCond.wakeAll();
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// define a pause instruction for spinlock-loop - merely useful on
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// HyperThreading systems with just one physical core (e.g. Intel Atom)
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#ifdef LMMS_HOST_X86_64
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#define SPINLOCK_PAUSE() asm( "pause" )
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#else
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#ifdef LMMS_HOST_X86_64
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#define SPINLOCK_PAUSE() asm( "pause" )
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#else
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#define SPINLOCK_PAUSE()
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#endif
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#endif
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#define WAIT_FOR_JOBS() \
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m_workers[m_numWorkers]->processJobQueue(); \
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while( MixerWorkerThread::s_jobQueue.itemsDone < \
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MixerWorkerThread::s_jobQueue.queueSize ) \
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{ \
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SPINLOCK_PAUSE(); \
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} \
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mixer::mixer( void ) :
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m_framesPerPeriod( DEFAULT_BUFFER_SIZE ),
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m_workingBuf( NULL ),
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m_inputBufferRead( 0 ),
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m_inputBufferWrite( 1 ),
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m_readBuf( NULL ),
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m_writeBuf( NULL ),
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m_cpuLoad( 0 ),
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m_workers(),
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m_numWorkers( QThread::idealThreadCount()-1 ),
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m_queueReadyWaitCond(),
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m_qualitySettings( qualitySettings::Mode_Draft ),
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m_masterGain( 1.0f ),
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m_audioDev( NULL ),
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m_oldAudioDev( NULL ),
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m_globalMutex( QMutex::Recursive )
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{
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for( int i = 0; i < 2; ++i )
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{
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m_inputBufferFrames[i] = 0;
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m_inputBufferSize[i] = DEFAULT_BUFFER_SIZE * 100;
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m_inputBuffer[i] = CPU::allocFrames(
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DEFAULT_BUFFER_SIZE * 100 );
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clearAudioBuffer( m_inputBuffer[i], m_inputBufferSize[i] );
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}
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for( int i = 1; i < NumFxChannels+1; ++i )
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{
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__fx_channel_jobs[i-1] = (fx_ch_t) i;
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}
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// just rendering?
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if( !engine::hasGUI() )
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{
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m_framesPerPeriod = DEFAULT_BUFFER_SIZE;
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m_fifo = new fifo( 1 );
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}
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else if( configManager::inst()->value( "mixer", "framesperaudiobuffer"
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).toInt() >= 32 )
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{
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m_framesPerPeriod =
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(fpp_t) configManager::inst()->value( "mixer",
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"framesperaudiobuffer" ).toInt();
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if( m_framesPerPeriod > DEFAULT_BUFFER_SIZE )
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{
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m_fifo = new fifo( m_framesPerPeriod
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/ DEFAULT_BUFFER_SIZE );
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m_framesPerPeriod = DEFAULT_BUFFER_SIZE;
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}
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else
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{
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m_fifo = new fifo( 1 );
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}
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}
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else
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{
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configManager::inst()->setValue( "mixer",
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"framesperaudiobuffer",
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QString::number( m_framesPerPeriod ) );
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m_fifo = new fifo( 1 );
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}
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m_workingBuf = CPU::allocFrames( m_framesPerPeriod );
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for( Uint8 i = 0; i < 3; i++ )
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{
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m_readBuf = CPU::allocFrames( m_framesPerPeriod );
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clearAudioBuffer( m_readBuf, m_framesPerPeriod );
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m_bufferPool.push_back( m_readBuf );
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}
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for( int i = 0; i < m_numWorkers+1; ++i )
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{
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MixerWorkerThread * wt = new MixerWorkerThread( i, this );
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if( i < m_numWorkers )
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{
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wt->start( QThread::TimeCriticalPriority );
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}
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m_workers.push_back( wt );
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}
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m_poolDepth = 2;
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m_readBuffer = 0;
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m_writeBuffer = 1;
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}
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mixer::~mixer()
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{
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// distribute an empty job-queue so that worker-threads
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// get out of their processing-loop
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MixerWorkerThread::s_jobQueue.queueSize = 0;
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for( int w = 0; w < m_numWorkers; ++w )
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{
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m_workers[w]->quit();
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}
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START_JOBS();
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for( int w = 0; w < m_numWorkers; ++w )
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{
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m_workers[w]->wait( 500 );
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}
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while( m_fifo->available() )
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{
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delete[] m_fifo->read();
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}
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delete m_fifo;
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delete m_audioDev;
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delete m_midiClient;
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for( Uint8 i = 0; i < 3; i++ )
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{
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CPU::freeFrames( m_bufferPool[i] );
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}
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CPU::freeFrames( m_workingBuf );
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}
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void mixer::initDevices( void )
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{
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m_audioDev = tryAudioDevices();
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m_midiClient = tryMidiClients();
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}
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void mixer::startProcessing( bool _needs_fifo )
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{
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if( _needs_fifo )
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{
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m_fifoWriter = new fifoWriter( this, m_fifo );
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m_fifoWriter->start( QThread::HighPriority );
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}
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else
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{
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m_fifoWriter = NULL;
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}
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m_audioDev->startProcessing();
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}
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void mixer::stopProcessing( void )
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{
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if( m_fifoWriter != NULL )
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{
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m_fifoWriter->finish();
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m_audioDev->stopProcessing();
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m_fifoWriter->wait( 1000 );
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m_fifoWriter->terminate();
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delete m_fifoWriter;
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m_fifoWriter = NULL;
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}
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else
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{
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m_audioDev->stopProcessing();
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}
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}
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sample_rate_t mixer::baseSampleRate( void ) const
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{
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sample_rate_t sr =
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configManager::inst()->value( "mixer", "samplerate" ).toInt();
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if( sr < 44100 )
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{
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sr = 44100;
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}
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return sr;
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}
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sample_rate_t mixer::outputSampleRate( void ) const
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{
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return m_audioDev != NULL ? m_audioDev->sampleRate() :
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baseSampleRate();
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}
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sample_rate_t mixer::inputSampleRate( void ) const
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{
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return m_audioDev != NULL ? m_audioDev->sampleRate() :
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baseSampleRate();
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}
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sample_rate_t mixer::processingSampleRate( void ) const
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{
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return outputSampleRate() * m_qualitySettings.sampleRateMultiplier();
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}
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bool mixer::criticalXRuns( void ) const
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{
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return m_cpuLoad >= 99 && engine::getSong()->realTimeTask() == true;
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}
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void mixer::pushInputFrames( sampleFrame * _ab, const f_cnt_t _frames )
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{
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lockInputFrames();
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f_cnt_t frames = m_inputBufferFrames[ m_inputBufferWrite ];
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int size = m_inputBufferSize[ m_inputBufferWrite ];
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sampleFrame * buf = m_inputBuffer[ m_inputBufferWrite ];
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if( frames + _frames > size )
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{
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size = qMax( size * 2, frames + _frames );
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sampleFrame * ab = CPU::allocFrames( size );
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CPU::memCpy( ab, buf, frames * sizeof( sampleFrame ) );
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CPU::freeFrames( buf );
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m_inputBufferSize[ m_inputBufferWrite ] = size;
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m_inputBuffer[ m_inputBufferWrite ] = ab;
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buf = ab;
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}
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CPU::memCpy( &buf[ frames ], _ab, _frames * sizeof( sampleFrame ) );
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m_inputBufferFrames[ m_inputBufferWrite ] += _frames;
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unlockInputFrames();
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}
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sampleFrameA * mixer::renderNextBuffer( void )
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{
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microTimer timer;
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static song::playPos last_metro_pos = -1;
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song::playPos p = engine::getSong()->getPlayPos(
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song::Mode_PlayPattern );
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if( engine::getSong()->playMode() == song::Mode_PlayPattern &&
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engine::getPianoRoll()->isRecording() == true &&
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p != last_metro_pos && p.getTicks() %
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(DefaultTicksPerTact / 4 ) == 0 )
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{
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addPlayHandle( new samplePlayHandle( "misc/metronome01.ogg" ) );
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last_metro_pos = p;
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}
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lockInputFrames();
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// swap buffer
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m_inputBufferWrite = ( m_inputBufferWrite + 1 ) % 2;
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m_inputBufferRead = ( m_inputBufferRead + 1 ) % 2;
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// clear new write buffer
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m_inputBufferFrames[ m_inputBufferWrite ] = 0;
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unlockInputFrames();
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// now we have to make sure no other thread does anything bad
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// while we're acting...
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lock();
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// remove all play-handles that have to be deleted and delete
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// them if they still exist...
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// maybe this algorithm could be optimized...
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ConstPlayHandleList::Iterator it_rem = m_playHandlesToRemove.begin();
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while( it_rem != m_playHandlesToRemove.end() )
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{
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PlayHandleList::Iterator it = qFind( m_playHandles.begin(),
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m_playHandles.end(), *it_rem );
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if( it != m_playHandles.end() )
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{
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delete *it;
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m_playHandles.erase( it );
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}
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it_rem = m_playHandlesToRemove.erase( it_rem );
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}
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// rotate buffers
|
|
m_writeBuffer = ( m_writeBuffer + 1 ) % m_poolDepth;
|
|
m_readBuffer = ( m_readBuffer + 1 ) % m_poolDepth;
|
|
|
|
m_writeBuf = m_bufferPool[m_writeBuffer];
|
|
m_readBuf = m_bufferPool[m_readBuffer];
|
|
|
|
// clear last audio-buffer
|
|
clearAudioBuffer( m_writeBuf, m_framesPerPeriod );
|
|
|
|
// prepare master mix (clear internal buffers etc.)
|
|
engine::getFxMixer()->prepareMasterMix();
|
|
|
|
// create play-handles for new notes, samples etc.
|
|
engine::getSong()->processNextBuffer();
|
|
|
|
|
|
// STAGE 1: run and render all play handles
|
|
FILL_JOB_QUEUE(PlayHandleList,m_playHandles,
|
|
MixerWorkerThread::PlayHandle,
|
|
!( *it )->done());
|
|
START_JOBS();
|
|
WAIT_FOR_JOBS();
|
|
|
|
// removed all play handles which are done
|
|
for( PlayHandleList::Iterator it = m_playHandles.begin();
|
|
it != m_playHandles.end(); )
|
|
{
|
|
if( ( *it )->affinityMatters() &&
|
|
( *it )->affinity() != QThread::currentThread() )
|
|
{
|
|
++it;
|
|
continue;
|
|
}
|
|
if( ( *it )->done() )
|
|
{
|
|
delete *it;
|
|
it = m_playHandles.erase( it );
|
|
}
|
|
else
|
|
{
|
|
++it;
|
|
}
|
|
}
|
|
|
|
|
|
// STAGE 2: process effects of all instrument- and sampletracks
|
|
FILL_JOB_QUEUE(QVector<audioPort*>,m_audioPorts,
|
|
MixerWorkerThread::AudioPortEffects,1);
|
|
START_JOBS();
|
|
WAIT_FOR_JOBS();
|
|
|
|
|
|
// STAGE 3: process effects in FX mixer
|
|
FILL_JOB_QUEUE_PARAM(QVector<fx_ch_t>,__fx_channel_jobs,
|
|
MixerWorkerThread::EffectChannel,1);
|
|
START_JOBS();
|
|
WAIT_FOR_JOBS();
|
|
|
|
|
|
// STAGE 4: do master mix in FX mixer
|
|
engine::getFxMixer()->masterMix( m_writeBuf );
|
|
|
|
unlock();
|
|
|
|
|
|
emit nextAudioBuffer();
|
|
|
|
// and trigger LFOs
|
|
envelopeAndLFOParameters::triggerLFO();
|
|
controller::triggerFrameCounter();
|
|
|
|
const float new_cpu_load = timer.elapsed() / 10000.0f *
|
|
processingSampleRate() / m_framesPerPeriod;
|
|
m_cpuLoad = tLimit( (int) ( new_cpu_load * 0.1f + m_cpuLoad * 0.9f ), 0,
|
|
100 );
|
|
|
|
return m_readBuf;
|
|
}
|
|
|
|
|
|
|
|
|
|
// removes all play-handles. this is neccessary, when the song is stopped ->
|
|
// all remaining notes etc. would be played until their end
|
|
void mixer::clear( void )
|
|
{
|
|
// TODO: m_midiClient->noteOffAll();
|
|
lock();
|
|
for( PlayHandleList::Iterator it = m_playHandles.begin();
|
|
it != m_playHandles.end(); ++it )
|
|
{
|
|
// we must not delete instrument-play-handles as they exist
|
|
// during the whole lifetime of an instrument
|
|
if( ( *it )->type() != playHandle::InstrumentPlayHandle )
|
|
{
|
|
m_playHandlesToRemove.push_back( *it );
|
|
}
|
|
}
|
|
unlock();
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::bufferToPort( const sampleFrame * _buf,
|
|
const fpp_t _frames,
|
|
const f_cnt_t _offset,
|
|
stereoVolumeVector _vv,
|
|
audioPort * _port )
|
|
{
|
|
const int start_frame = _offset % m_framesPerPeriod;
|
|
int end_frame = start_frame + _frames;
|
|
const int loop1_frame = qMin<int>( end_frame, m_framesPerPeriod );
|
|
|
|
_port->lockFirstBuffer();
|
|
CPU::unalignedBufMixLRCoeff( _port->firstBuffer() + start_frame,
|
|
_buf, _vv.vol[0], _vv.vol[1],
|
|
loop1_frame - start_frame );
|
|
_port->unlockFirstBuffer();
|
|
|
|
_port->lockSecondBuffer();
|
|
if( end_frame > m_framesPerPeriod )
|
|
{
|
|
const int frames_done = m_framesPerPeriod - start_frame;
|
|
end_frame -= m_framesPerPeriod;
|
|
end_frame = qMin<int>( end_frame, m_framesPerPeriod );
|
|
CPU::unalignedBufMixLRCoeff( _port->secondBuffer(),
|
|
_buf+frames_done,
|
|
_vv.vol[0], _vv.vol[1],
|
|
end_frame );
|
|
// we used both buffers so set flags
|
|
_port->m_bufferUsage = audioPort::BothBuffers;
|
|
}
|
|
else if( _port->m_bufferUsage == audioPort::NoUsage )
|
|
{
|
|
// only first buffer touched
|
|
_port->m_bufferUsage = audioPort::FirstBuffer;
|
|
}
|
|
_port->unlockSecondBuffer();
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::clearAudioBuffer( sampleFrame * _ab, const f_cnt_t _frames,
|
|
const f_cnt_t _offset )
|
|
{
|
|
if( likely( (size_t)( _ab+_offset ) % 16 == 0 && _frames % 8 == 0 ) )
|
|
{
|
|
CPU::memClear( _ab+_offset, sizeof( *_ab ) * _frames );
|
|
}
|
|
else
|
|
{
|
|
memset( _ab+_offset, 0, sizeof( *_ab ) * _frames );
|
|
}
|
|
}
|
|
|
|
|
|
|
|
#ifndef LMMS_DISABLE_SURROUND
|
|
void mixer::clearAudioBuffer( surroundSampleFrame * _ab, const f_cnt_t _frames,
|
|
const f_cnt_t _offset )
|
|
{
|
|
memset( _ab+_offset, 0, sizeof( *_ab ) * _frames );
|
|
}
|
|
#endif
|
|
|
|
|
|
|
|
|
|
float mixer::peakValueLeft( sampleFrame * _ab, const f_cnt_t _frames )
|
|
{
|
|
float p = 0.0f;
|
|
for( f_cnt_t f = 0; f < _frames; ++f )
|
|
{
|
|
if( _ab[f][0] > p )
|
|
{
|
|
p = _ab[f][0];
|
|
}
|
|
else if( -_ab[f][0] > p )
|
|
{
|
|
p = -_ab[f][0];
|
|
}
|
|
}
|
|
return p;
|
|
}
|
|
|
|
|
|
|
|
|
|
float mixer::peakValueRight( sampleFrame * _ab, const f_cnt_t _frames )
|
|
{
|
|
float p = 0.0f;
|
|
for( f_cnt_t f = 0; f < _frames; ++f )
|
|
{
|
|
if( _ab[f][1] > p )
|
|
{
|
|
p = _ab[f][1];
|
|
}
|
|
else if( -_ab[f][1] > p )
|
|
{
|
|
p = -_ab[f][1];
|
|
}
|
|
}
|
|
return p;
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::changeQuality( const struct qualitySettings & _qs )
|
|
{
|
|
// don't delete the audio-device
|
|
stopProcessing();
|
|
|
|
m_qualitySettings = _qs;
|
|
m_audioDev->applyQualitySettings();
|
|
|
|
emit sampleRateChanged();
|
|
emit qualitySettingsChanged();
|
|
|
|
startProcessing();
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::setAudioDevice( audioDevice * _dev )
|
|
{
|
|
stopProcessing();
|
|
|
|
m_oldAudioDev = m_audioDev;
|
|
|
|
if( _dev == NULL )
|
|
{
|
|
printf( "param _dev == NULL in mixer::setAudioDevice(...). "
|
|
"Trying any working audio-device\n" );
|
|
m_audioDev = tryAudioDevices();
|
|
}
|
|
else
|
|
{
|
|
m_audioDev = _dev;
|
|
}
|
|
|
|
emit sampleRateChanged();
|
|
|
|
startProcessing();
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::setAudioDevice( audioDevice * _dev,
|
|
const struct qualitySettings & _qs,
|
|
bool _needs_fifo )
|
|
{
|
|
// don't delete the audio-device
|
|
stopProcessing();
|
|
|
|
m_qualitySettings = _qs;
|
|
m_oldAudioDev = m_audioDev;
|
|
|
|
if( _dev == NULL )
|
|
{
|
|
printf( "param _dev == NULL in mixer::setAudioDevice(...). "
|
|
"Trying any working audio-device\n" );
|
|
m_audioDev = tryAudioDevices();
|
|
}
|
|
else
|
|
{
|
|
m_audioDev = _dev;
|
|
}
|
|
|
|
emit qualitySettingsChanged();
|
|
emit sampleRateChanged();
|
|
|
|
startProcessing( _needs_fifo );
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::restoreAudioDevice( void )
|
|
{
|
|
if( m_oldAudioDev != NULL )
|
|
{
|
|
stopProcessing();
|
|
delete m_audioDev;
|
|
|
|
m_audioDev = m_oldAudioDev;
|
|
emit sampleRateChanged();
|
|
|
|
m_oldAudioDev = NULL;
|
|
startProcessing();
|
|
}
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::removeAudioPort( audioPort * _port )
|
|
{
|
|
QVector<audioPort *>::Iterator it = qFind( m_audioPorts.begin(),
|
|
m_audioPorts.end(),
|
|
_port );
|
|
if( it != m_audioPorts.end() )
|
|
{
|
|
lock();
|
|
m_audioPorts.erase( it );
|
|
unlock();
|
|
}
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::removePlayHandle( playHandle * _ph )
|
|
{
|
|
lock();
|
|
// check thread affinity as we must not delete play-handles
|
|
// which were created in a thread different than mixer thread
|
|
if( _ph->affinityMatters() &&
|
|
_ph->affinity() == QThread::currentThread() )
|
|
{
|
|
PlayHandleList::Iterator it =
|
|
qFind( m_playHandles.begin(),
|
|
m_playHandles.end(), _ph );
|
|
if( it != m_playHandles.end() )
|
|
{
|
|
m_playHandles.erase( it );
|
|
delete _ph;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
m_playHandlesToRemove.push_back( _ph );
|
|
}
|
|
unlock();
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::removePlayHandles( track * _track, playHandle::Type _type )
|
|
{
|
|
lock();
|
|
PlayHandleList::Iterator it = m_playHandles.begin();
|
|
while( it != m_playHandles.end() )
|
|
{
|
|
if( ( _type == playHandle::NumPlayHandleTypes ||
|
|
( *it )->type() == _type ) &&
|
|
( *it )->isFromTrack( _track ) )
|
|
{
|
|
delete *it;
|
|
it = m_playHandles.erase( it );
|
|
}
|
|
else
|
|
{
|
|
++it;
|
|
}
|
|
}
|
|
unlock();
|
|
}
|
|
|
|
|
|
|
|
|
|
audioDevice * mixer::tryAudioDevices( void )
|
|
{
|
|
bool success_ful = false;
|
|
audioDevice * dev = NULL;
|
|
QString dev_name = configManager::inst()->value( "mixer", "audiodev" );
|
|
|
|
if( dev_name == audioDummy::name() )
|
|
{
|
|
dev_name = "";
|
|
}
|
|
|
|
#ifdef LMMS_HAVE_ALSA
|
|
if( dev_name == audioALSA::name() || dev_name == "" )
|
|
{
|
|
dev = new audioALSA( success_ful, this );
|
|
if( success_ful )
|
|
{
|
|
m_audioDevName = audioALSA::name();
|
|
return dev;
|
|
}
|
|
delete dev;
|
|
}
|
|
#endif
|
|
|
|
|
|
#ifdef LMMS_HAVE_PORTAUDIO
|
|
if( dev_name == audioPortAudio::name() || dev_name == "" )
|
|
{
|
|
dev = new audioPortAudio( success_ful, this );
|
|
if( success_ful )
|
|
{
|
|
m_audioDevName = audioPortAudio::name();
|
|
return dev;
|
|
}
|
|
delete dev;
|
|
}
|
|
#endif
|
|
|
|
|
|
#ifdef LMMS_HAVE_PULSEAUDIO
|
|
if( dev_name == audioPulseAudio::name() || dev_name == "" )
|
|
{
|
|
dev = new audioPulseAudio( success_ful, this );
|
|
if( success_ful )
|
|
{
|
|
m_audioDevName = audioPulseAudio::name();
|
|
return dev;
|
|
}
|
|
delete dev;
|
|
}
|
|
#endif
|
|
|
|
|
|
#ifdef LMMS_HAVE_OSS
|
|
if( dev_name == audioOSS::name() || dev_name == "" )
|
|
{
|
|
dev = new audioOSS( success_ful, this );
|
|
if( success_ful )
|
|
{
|
|
m_audioDevName = audioOSS::name();
|
|
return dev;
|
|
}
|
|
delete dev;
|
|
}
|
|
#endif
|
|
|
|
|
|
#ifdef LMMS_HAVE_JACK
|
|
if( dev_name == audioJACK::name() || dev_name == "" )
|
|
{
|
|
dev = new audioJACK( success_ful, this );
|
|
if( success_ful )
|
|
{
|
|
m_audioDevName = audioJACK::name();
|
|
return dev;
|
|
}
|
|
delete dev;
|
|
}
|
|
#endif
|
|
|
|
|
|
#ifdef LMMS_HAVE_SDL
|
|
if( dev_name == audioSDL::name() || dev_name == "" )
|
|
{
|
|
dev = new audioSDL( success_ful, this );
|
|
if( success_ful )
|
|
{
|
|
m_audioDevName = audioSDL::name();
|
|
return dev;
|
|
}
|
|
delete dev;
|
|
}
|
|
#endif
|
|
|
|
// add more device-classes here...
|
|
//dev = new audioXXXX( SAMPLE_RATES[m_qualityLevel], success_ful, this );
|
|
//if( sucess_ful )
|
|
//{
|
|
// return dev;
|
|
//}
|
|
//delete dev
|
|
|
|
printf( "No audio-driver working - falling back to dummy-audio-"
|
|
"driver\nYou can render your songs and listen to the output "
|
|
"files...\n" );
|
|
|
|
m_audioDevName = audioDummy::name();
|
|
|
|
return new audioDummy( success_ful, this );
|
|
}
|
|
|
|
|
|
|
|
|
|
midiClient * mixer::tryMidiClients( void )
|
|
{
|
|
QString client_name = configManager::inst()->value( "mixer",
|
|
"mididev" );
|
|
|
|
#ifdef LMMS_HAVE_ALSA
|
|
if( client_name == midiALSASeq::name() || client_name == "" )
|
|
{
|
|
midiALSASeq * malsas = new midiALSASeq;
|
|
if( malsas->isRunning() )
|
|
{
|
|
m_midiClientName = midiALSASeq::name();
|
|
return malsas;
|
|
}
|
|
delete malsas;
|
|
}
|
|
|
|
if( client_name == midiALSARaw::name() || client_name == "" )
|
|
{
|
|
midiALSARaw * malsar = new midiALSARaw;
|
|
if( malsar->isRunning() )
|
|
{
|
|
m_midiClientName = midiALSARaw::name();
|
|
return malsar;
|
|
}
|
|
delete malsar;
|
|
}
|
|
#endif
|
|
|
|
#ifdef LMMS_HAVE_OSS
|
|
if( client_name == midiOSS::name() || client_name == "" )
|
|
{
|
|
midiOSS * moss = new midiOSS;
|
|
if( moss->isRunning() )
|
|
{
|
|
m_midiClientName = midiOSS::name();
|
|
return moss;
|
|
}
|
|
delete moss;
|
|
}
|
|
#endif
|
|
|
|
#ifdef LMMS_BUILD_WIN32
|
|
if( client_name == midiWinMM::name() || client_name == "" )
|
|
{
|
|
midiWinMM * mwmm = new midiWinMM;
|
|
// if( moss->isRunning() )
|
|
{
|
|
m_midiClientName = midiWinMM::name();
|
|
return mwmm;
|
|
}
|
|
delete mwmm;
|
|
}
|
|
#endif
|
|
|
|
printf( "Couldn't create MIDI-client, neither with ALSA nor with "
|
|
"OSS. Will use dummy-MIDI-client.\n" );
|
|
|
|
m_midiClientName = midiDummy::name();
|
|
|
|
return new midiDummy;
|
|
}
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
mixer::fifoWriter::fifoWriter( mixer * _mixer, fifo * _fifo ) :
|
|
m_mixer( _mixer ),
|
|
m_fifo( _fifo ),
|
|
m_writing( true )
|
|
{
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::fifoWriter::finish( void )
|
|
{
|
|
m_writing = false;
|
|
}
|
|
|
|
|
|
|
|
|
|
void mixer::fifoWriter::run( void )
|
|
{
|
|
#if 0
|
|
#ifdef LMMS_BUILD_LINUX
|
|
#ifdef LMMS_HAVE_PTHREAD_H
|
|
cpu_set_t mask;
|
|
CPU_ZERO( &mask );
|
|
CPU_SET( 0, &mask );
|
|
pthread_setaffinity_np( pthread_self(), sizeof( mask ), &mask );
|
|
#endif
|
|
#endif
|
|
#endif
|
|
|
|
const fpp_t frames = m_mixer->framesPerPeriod();
|
|
while( m_writing )
|
|
{
|
|
sampleFrameA * buffer = CPU::allocFrames( frames );
|
|
const sampleFrameA * b = m_mixer->renderNextBuffer();
|
|
CPU::memCpy( buffer, b, frames * sizeof( sampleFrameA ) );
|
|
m_fifo->write( buffer );
|
|
}
|
|
|
|
m_fifo->write( NULL );
|
|
}
|
|
|
|
|
|
|
|
|
|
#include "moc_mixer.cxx"
|
|
|