Files
lmms/src/core/mixer.cpp
2005-09-27 13:10:34 +00:00

705 lines
14 KiB
C++

/*
* mixer.cpp - audio-device-independent mixer for LMMS
*
* Linux MultiMedia Studio
* Copyright (c) 2004-2005 Tobias Doerffel <tobydox@users.sourceforge.net>
*
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*
*/
#include "mixer.h"
#include "play_handle.h"
#include "song_editor.h"
#include "templates.h"
#include "envelope_and_lfo_widget.h"
#include "buffer_allocator.h"
#include "debug.h"
#include "config_mgr.h"
#include "audio_device.h"
#include "midi_device.h"
// platform-specific audio-interface-classes
#include "audio_alsa.h"
#include "audio_jack.h"
#include "audio_oss.h"
#include "audio_sdl.h"
#include "audio_dummy.h"
// platform-specific midi-interface-classes
#include "midi_alsa_raw.h"
#include "midi_oss.h"
#include "midi_dummy.h"
Uint32 SAMPLE_RATES[QUALITY_LEVELS] = { 44100, 88200 } ;
mixer * mixer::s_instanceOfMe = NULL;
mixer::mixer() :
#ifndef QT4
QObject(),
#endif
QThread(),
m_silence(),
#ifndef DISABLE_SURROUND
m_surroundSilence(),
#endif
m_framesPerAudioBuffer( DEFAULT_BUFFER_SIZE ),
m_buffer1( NULL ),
m_buffer2( NULL ),
m_curBuf( NULL ),
m_nextBuf( NULL ),
m_discardCurBuf( FALSE ),
m_qualityLevel( DEFAULT_QUALITY_LEVEL ),
m_masterOutput( 1.0f ),
m_quit( FALSE ),
m_audioDev( NULL ),
m_oldAudioDev( NULL )
{
// small hack because code calling mixer::inst() is called out of ctor
s_instanceOfMe = this;
if( configManager::inst()->value( "mixer", "framesperaudiobuffer"
).toInt() >= 32 )
{
m_framesPerAudioBuffer = configManager::inst()->value( "mixer",
"framesperaudiobuffer" ).toInt();
}
else
{
configManager::inst()->setValue( "mixer",
"framesperaudiobuffer",
QString::number( m_framesPerAudioBuffer ) );
}
m_buffer1 = bufferAllocator::alloc<surroundSampleFrame>(
m_framesPerAudioBuffer );
m_buffer2 = bufferAllocator::alloc<surroundSampleFrame>(
m_framesPerAudioBuffer );
m_curBuf = m_buffer1;
m_nextBuf = m_buffer2;
m_audioDev = tryAudioDevices();
m_midiDev = tryMIDIDevices();
for( int i = 0; i < MAX_SAMPLE_PACKETS; ++i )
{
m_samplePackets[i].m_buffer = NULL;
m_samplePackets[i].m_state = samplePacket::UNUSED;
}
m_silence = bufferAllocator::alloc<sampleFrame>(
m_framesPerAudioBuffer );
#ifndef DISABLE_SURROUND
m_surroundSilence = bufferAllocator::alloc<surroundSampleFrame>(
m_framesPerAudioBuffer );
#endif
for( Uint32 frame = 0; frame < m_framesPerAudioBuffer; ++frame )
{
for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
{
m_silence[frame][chnl] = 0.0f;
}
#ifndef DISABLE_SURROUND
for( Uint8 chnl = 0; chnl < SURROUND_CHANNELS; ++chnl )
{
m_surroundSilence[frame][chnl] = 0.0f;
}
#endif
}
// now clear our two output-buffers before using them...
clearAudioBuffer( m_buffer1, m_framesPerAudioBuffer );
clearAudioBuffer( m_buffer2, m_framesPerAudioBuffer );
}
mixer::~mixer()
{
delete m_audioDev;
bufferAllocator::free( m_buffer1 );
bufferAllocator::free( m_buffer2 );
for( int i = 0; i < MAX_SAMPLE_PACKETS; ++i )
{
if( m_samplePackets[i].m_state != samplePacket::UNUSED )
{
bufferAllocator::free( m_samplePackets[i].m_buffer );
}
}
bufferAllocator::free( m_silence );
#ifndef DISABLE_SURROUND
bufferAllocator::free( m_surroundSilence );
#endif
}
void mixer::quitThread( void )
{
// make sure there're no mutexes locked anymore...
m_safetySyncMutex.unlock();
m_devMutex.unlock();
// now tell mixer-thread to quit
m_quit = TRUE;
wait( 1000 );
terminate();
}
void mixer::run( void )
{
while( m_quit == FALSE )
{
// remove all play-handles that have to be deleted and delete
// them if they still exist...
// maybe this algorithm could be optimized...
while( !m_playHandlesToRemove.empty() )
{
playHandleVector::iterator it = m_playHandles.begin();
while( it != m_playHandles.end() )
{
if( *it == m_playHandlesToRemove.front() )
{
m_playHandles.erase( it );
delete m_playHandlesToRemove.front();
break;
}
++it;
}
m_playHandlesToRemove.erase(
m_playHandlesToRemove.begin() );
}
// now we have to make sure no other thread does anything bad
// while we're acting...
m_safetySyncMutex.lock();
/* following code is faster but unstable since using iterators
while deleting from vector is dangerous and often leads to
undefined results...
playHandleVector::iterator it = m_playHandles.begin();
while( it != m_playHandles.end() )
{
if( ( *it )->done() )
{
// delete all play-handles which have
// played completely now
delete *it;
m_playHandles.erase( it );
}
else
{
// play all uncompletely-played play-handles...
( *it )->play();
++it;
}
}*/
csize idx = 0;
while( idx < m_playHandles.size() )
{
register playHandle * n = m_playHandles[idx];
if( n->done() )
{
// delete all play-handles which have
// played completely now
delete n;
m_playHandles.erase( m_playHandles.begin() +
idx );
}
else
{
// play all uncompletely-played play-handles...
n->play();
++idx;
}
}
songEditor::inst()->processNextBuffer();
// check for samples-packets that have to be mixed in
// the current audio-buffer
for( int i = 0; i < MAX_SAMPLE_PACKETS; ++i )
{
if( m_samplePackets[i].m_state == samplePacket::READY )
{
if( m_samplePackets[i].m_framesAhead <=
m_framesPerAudioBuffer )
{
// found one! mix it...
mixSamplePacket( &m_samplePackets[i] );
// now this audio-sample can be used
// again
bufferAllocator::free(
m_samplePackets[i].m_buffer );
m_samplePackets[i].m_state =
samplePacket::UNUSED;
}
else
{
m_samplePackets[i].m_framesAhead -=
m_framesPerAudioBuffer;
}
}
}
if( !m_discardCurBuf )
{
m_devMutex.lock();
// write actual data to our current output-device
// (blocking!)
m_audioDev->writeBuffer( m_curBuf,
m_framesPerAudioBuffer,
SAMPLE_RATES[m_qualityLevel],
m_masterOutput );
m_devMutex.unlock();
}
else
{
m_discardCurBuf = FALSE;
}
emit nextAudioBuffer( m_curBuf, m_framesPerAudioBuffer );
m_safetySyncMutex.unlock();
// clear last audio-buffer
clearAudioBuffer( m_curBuf, m_framesPerAudioBuffer );
// now swap the buffers... current buffer becomes next (last)
// buffer and the next buffer becomes current (first) buffer
qSwap( m_curBuf, m_nextBuf );
// and trigger LFOs
envelopeAndLFOWidget::triggerLFO();
}
}
// 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 )
{
m_midiDev->noteOffAll();
for( playHandleVector::iterator it = m_playHandles.begin();
it != m_playHandles.end(); ++it )
{
m_playHandlesToRemove.push_back( *it );
}
}
void FASTCALL mixer::clearAudioBuffer( sampleFrame * _ab, Uint32 _frames )
{
if( _frames == m_framesPerAudioBuffer )
{
memcpy( _ab, m_silence, m_framesPerAudioBuffer *
BYTES_PER_FRAME );
}
else
{
for( Uint32 frame = 0; frame < _frames; ++frame )
{
for( Uint8 ch = 0; ch < DEFAULT_CHANNELS; ++ch )
{
_ab[frame][ch] = 0.0f;
}
}
}
}
#ifndef DISABLE_SURROUND
void FASTCALL mixer::clearAudioBuffer( surroundSampleFrame * _ab,
Uint32 _frames )
{
if( _frames == m_framesPerAudioBuffer )
{
memcpy( _ab, m_surroundSilence, m_framesPerAudioBuffer *
BYTES_PER_SURROUND_FRAME );
}
else
{
for( Uint32 frame = 0; frame < _frames; ++frame )
{
for( Uint8 ch = 0; ch < DEFAULT_CHANNELS; ++ch )
{
_ab[frame][ch] = 0.0f;
}
}
}
}
#endif
void FASTCALL mixer::addBuffer( sampleFrame * _buf, Uint32 _frames,
Uint32 _frames_ahead,
volumeVector & _volume_vector )
{
#ifdef LMMS_DEBUG
bool success = FALSE;
#endif
for ( Uint16 i = 0; i < MAX_SAMPLE_PACKETS; ++i )
{
if( m_samplePackets[i].m_state == samplePacket::UNUSED )
{
m_samplePackets[i].m_state = samplePacket::FILLING;
m_samplePackets[i].m_frames = _frames;//m_framesPerAudioBuffer;
m_samplePackets[i].m_framesDone = 0;
m_samplePackets[i].m_framesAhead = _frames_ahead;
m_samplePackets[i].m_buffer =
bufferAllocator::alloc<surroundSampleFrame>(
m_framesPerAudioBuffer );
// now we have to make a surround-buffer out of a
// stereo-buffer (could be done more easily if there
// would be no volume-vector...)
for( Uint32 frame = 0; frame < _frames/*m_framesPerAudioBuffer*/;
++frame )
{
for( Uint8 chnl = 0; chnl < SURROUND_CHANNELS;
++chnl )
{
m_samplePackets[i].m_buffer[frame][chnl] =
_buf[frame][chnl%DEFAULT_CHANNELS] *
_volume_vector.vol[chnl];
}
}
m_samplePackets[i].m_state = samplePacket::READY;
#ifdef LMMS_DEBUG
success = TRUE;
#endif
break;
}
}
#ifdef LMMS_DEBUG
if( success == FALSE )
{
qWarning( "No sample-packets left in mixer::addBuffer(...)!\n" );
}
#endif
}
void mixer::setHighQuality( bool _hq_on )
{
m_safetySyncMutex.lock();
// delete (= close) our audio-device
delete m_audioDev;
// set new quality-level...
if( _hq_on == TRUE )
{
m_qualityLevel = HIGH_QUALITY_LEVEL;
}
else
{
m_qualityLevel = DEFAULT_QUALITY_LEVEL;
}
// and re-open device
m_audioDev = tryAudioDevices();
m_safetySyncMutex.unlock();
emit( sampleRateChanged() );
}
void FASTCALL mixer::setAudioDevice( audioDevice * _dev, bool _hq )
{
m_devMutex.lock();
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;
}
m_qualityLevel = _hq ? 1 : 0;
emit sampleRateChanged();
m_devMutex.unlock();
}
void mixer::restoreAudioDevice( void )
{
m_devMutex.lock();
if( m_oldAudioDev != NULL )
{
delete m_audioDev;
m_audioDev = m_oldAudioDev;
for( Uint8 qli = 0; qli < QUALITY_LEVELS; ++qli )
{
if( SAMPLE_RATES[qli] == m_audioDev->sampleRate() )
{
m_qualityLevel = qli;
emit sampleRateChanged();
break;
}
}
m_oldAudioDev = NULL;
m_discardCurBuf = TRUE;
}
m_devMutex.unlock();
}
void mixer::checkValidityOfPlayHandles( void )
{
playHandleVector::iterator it = m_playHandles.begin();
while( it != m_playHandles.end() )
{
( *it )->checkValidity();
++it;
}
}
void FASTCALL mixer::mixSamplePacket( samplePacket * _sp )
{
Uint32 start_frame = _sp->m_framesAhead % m_framesPerAudioBuffer;
Uint32 end_frame = start_frame + _sp->m_frames;//m_framesPerAudioBuffer;
if( end_frame <= m_framesPerAudioBuffer )
{
for( Uint32 frame = start_frame; frame < end_frame; ++frame )
{
for( Uint8 chnl = 0; chnl < SURROUND_CHANNELS; ++chnl )
{
m_curBuf[frame][chnl] +=
_sp->m_buffer[frame-start_frame][chnl];
}
}
}
else
{
for( Uint32 frame = start_frame; frame <
m_framesPerAudioBuffer; ++frame )
{
for( Uint8 chnl = 0; chnl < SURROUND_CHANNELS; ++chnl )
{
m_curBuf[frame][chnl] +=
_sp->m_buffer[frame-start_frame][chnl];
}
}
Uint32 frames_done = m_framesPerAudioBuffer - start_frame;
end_frame = tMin( end_frame -= m_framesPerAudioBuffer,
m_framesPerAudioBuffer );
for( Uint32 frame = 0; frame < end_frame; ++frame )
{
for( Uint8 chnl = 0; chnl < SURROUND_CHANNELS; ++chnl )
{
m_nextBuf[frame][chnl] +=
_sp->m_buffer[frames_done+frame][chnl];
}
}
}
}
audioDevice * mixer::tryAudioDevices( void )
{
//m_discardCurBuf = TRUE;
bool success_ful = FALSE;
audioDevice * dev = NULL;
QString dev_name = configManager::inst()->value( "mixer", "audiodev" );
#ifdef OSS_SUPPORT
if( dev_name == audioOSS::name() || dev_name == "" )
{
dev = new audioOSS( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL],
success_ful );
if( success_ful )
{
m_audioDevName = audioOSS::name();
return( dev );
}
delete dev;
}
#endif
#ifdef ALSA_SUPPORT
if( dev_name == audioALSA::name() || dev_name == "" )
{
dev = new audioALSA( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL],
success_ful );
if( success_ful )
{
m_audioDevName = audioALSA::name();
return( dev );
}
delete dev;
}
#endif
#ifdef JACK_SUPPORT
if( dev_name == audioJACK::name() || dev_name == "" )
{
dev = new audioJACK( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL],
success_ful );
if( success_ful )
{
m_audioDevName = audioJACK::name();
return( dev );
}
delete dev;
}
#endif
#ifdef SDL_AUDIO_SUPPORT
if( dev_name == audioSDL::name() || dev_name == "" )
{
dev = new audioSDL( SAMPLE_RATES[DEFAULT_QUALITY_LEVEL],
success_ful );
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 );
//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( SAMPLE_RATES[m_qualityLevel], success_ful ) );
}
midiDevice * mixer::tryMIDIDevices( void )
{
QString dev_name = configManager::inst()->value( "mixer", "mididev" );
#ifdef ALSA_SUPPORT
if( dev_name == midiALSARaw::name() || dev_name == "" )
{
midiALSARaw * malsa = new midiALSARaw();
if( malsa->isRunning() )
{
m_midiDevName = midiALSARaw::name();
return( malsa );
}
delete malsa;
}
#endif
#ifdef OSS_SUPPORT
if( dev_name == midiOSS::name() || dev_name == "" )
{
midiOSS * moss = new midiOSS();
if( moss->isRunning() )
{
m_midiDevName = midiOSS::name();
return( moss );
}
delete moss;
}
#endif
printf( "Couldn't open a MIDI-device, neither with ALSA nor with "
"OSS. Will use dummy-MIDI-device.\n" );
m_midiDevName = midiDummy::name();
return( new midiDummy() );
}
#include "mixer.moc"