renamed "mixer::framesPerAudioBuffer()" to "mixer::framesPerPeriod()" and type "fpab_t" to "fpp_t"
git-svn-id: https://lmms.svn.sf.net/svnroot/lmms/trunk/lmms@502 0778d3d1-df1d-0410-868b-ea421aaaa00d
This commit is contained in:
@@ -69,7 +69,7 @@ bassBoosterEffect::~bassBoosterEffect()
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bool FASTCALL bassBoosterEffect::processAudioBuffer( surroundSampleFrame * _buf,
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const fpab_t _frames )
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const fpp_t _frames )
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{
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if( isBypassed() || !isRunning () )
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{
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@@ -77,7 +77,7 @@ bool FASTCALL bassBoosterEffect::processAudioBuffer( surroundSampleFrame * _buf,
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}
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double out_sum = 0.0;
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for( fpab_t f = 0; f < _frames; ++f )
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for( fpp_t f = 0; f < _frames; ++f )
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{
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sample_t s[2] = { _buf[f][0], _buf[f][1] };
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m_bbFX.nextSample( s[0], s[1] );
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@@ -41,7 +41,7 @@ public:
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bassBoosterEffect( const descriptor::subPluginFeatures::key * _key );
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virtual ~bassBoosterEffect();
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virtual bool FASTCALL processAudioBuffer( surroundSampleFrame * _buf,
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const fpab_t _frames );
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const fpp_t _frames );
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inline virtual QString nodeName( void ) const
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{
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return( "bassboostereffect" );
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@@ -701,11 +701,11 @@ void bitInvader::playNote( notePlayHandle * _n, bool )
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engine::getMixer()->sampleRate() );
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}
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const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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sampleFrame * buf = new sampleFrame[frames];
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bSynth * ps = static_cast<bSynth *>( _n->m_pluginData );
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for( fpab_t frame = 0; frame < frames; ++frame )
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for( fpp_t frame = 0; frame < frames; ++frame )
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{
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const sample_t cur = ps->nextStringSample();
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for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
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@@ -192,12 +192,13 @@ void kickerInstrument::playNote( notePlayHandle * _n, bool )
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//const float freq = getInstrumentTrack()->frequency( _n ) / 2;
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const float fdiff = m_endFreqKnob->value() - m_startFreqKnob->value();
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const fpab_t frames = _n->released() ?
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/* const fpp_t frames = _n->released() ?
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tMax( tMin<f_cnt_t>( desiredReleaseFrames() -
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_n->releaseFramesDone(),
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engine::getMixer()->framesPerAudioBuffer() ), 0 )
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:
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engine::getMixer()->framesPerAudioBuffer();
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engine::getMixer()->framesPerAudioBuffer();*/
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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const float f1 = m_startFreqKnob->value() + tfp * fdiff / decfr;
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const float f2 = m_startFreqKnob->value() + (frames+tfp-1)*fdiff/decfr;
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@@ -209,7 +210,7 @@ void kickerInstrument::playNote( notePlayHandle * _n, bool )
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if( _n->released() )
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{
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for( fpab_t f = 0; f < frames; ++f )
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for( fpp_t f = 0; f < frames; ++f )
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{
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const float fac = 1.0f -
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(float)( _n->releaseFramesDone()+f ) /
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@@ -116,7 +116,7 @@ ladspaEffect::ladspaEffect( const descriptor::subPluginFeatures::key * _key ) :
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// during cleanup. It was easier to troubleshoot with the
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// memory management all taking place in one file.
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p->buffer =
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new LADSPA_Data[engine::getMixer()->framesPerAudioBuffer()];
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new LADSPA_Data[engine::getMixer()->framesPerPeriod()];
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if( p->name.toUpper().contains( "IN" ) &&
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m_ladspa->isPortInput( m_key, port ) )
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@@ -334,7 +334,7 @@ ladspaEffect::~ladspaEffect()
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bool FASTCALL ladspaEffect::processAudioBuffer( surroundSampleFrame * _buf,
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const fpab_t _frames )
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const fpp_t _frames )
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{
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if( !isOkay() || dontRun() || !isRunning() || isBypassed() )
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{
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@@ -353,7 +353,7 @@ bool FASTCALL ladspaEffect::processAudioBuffer( surroundSampleFrame * _buf,
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switch( m_ports[proc][port]->rate )
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{
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case CHANNEL_IN:
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for( fpab_t frame = 0;
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for( fpp_t frame = 0;
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frame < _frames; frame++ )
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{
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m_ports[proc][port]->buffer[frame] =
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@@ -369,7 +369,7 @@ bool FASTCALL ladspaEffect::processAudioBuffer( surroundSampleFrame * _buf,
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// This only supports control rate ports, so the audio rates are
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// treated as though they were control rate by setting the
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// port buffer to all the same value.
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for( fpab_t frame = 0;
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for( fpp_t frame = 0;
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frame < _frames; frame++ )
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{
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m_ports[proc][port]->buffer[frame] =
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@@ -419,7 +419,7 @@ bool FASTCALL ladspaEffect::processAudioBuffer( surroundSampleFrame * _buf,
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case CONTROL_RATE_INPUT:
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break;
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case CHANNEL_OUT:
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for( fpab_t frame = 0;
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for( fpp_t frame = 0;
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frame < _frames; frame++ )
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{
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_buf[frame][channel] =
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@@ -50,7 +50,7 @@ public:
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virtual ~ladspaEffect();
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virtual bool FASTCALL processAudioBuffer( surroundSampleFrame * _buf,
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const fpab_t _frames );
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const fpp_t _frames );
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void FASTCALL setControl( Uint16 _control, LADSPA_Data _data );
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@@ -89,7 +89,7 @@ private:
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ladspa2LMMS * m_ladspa;
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Uint16 m_effectChannels;
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Uint16 m_portCount;
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fpab_t m_bufferSize;
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fpp_t m_bufferSize;
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const LADSPA_Descriptor * m_descriptor;
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vvector<LADSPA_Handle> m_handles;
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@@ -738,7 +738,7 @@ void lb302Synth::playNote( notePlayHandle * _n, bool )
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}
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}
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const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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sampleFrame *buf = new sampleFrame[frames];
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process(buf, frames);
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@@ -359,7 +359,7 @@ void organicInstrument::playNote( notePlayHandle * _n, bool )
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oscillator * osc_r = static_cast<oscPtr *>( _n->m_pluginData
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)->oscRight;
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const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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sampleFrame * buf = new sampleFrame[frames];
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osc_l->update( buf, frames, 0 );
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@@ -213,7 +213,7 @@ QString patmanSynth::nodeName( void ) const
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void patmanSynth::playNote( notePlayHandle * _n, bool )
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{
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const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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sampleFrame * buf = new sampleFrame[frames];
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if( !_n->m_pluginData )
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@@ -143,11 +143,11 @@ void pluckedStringSynth::playNote( notePlayHandle * _n, bool )
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engine::getMixer()->sampleRate() );
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}
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const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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sampleFrame * buf = new sampleFrame[frames];
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pluckSynth * ps = static_cast<pluckSynth *>( _n->m_pluginData );
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for( Uint32 frame = 0; frame < frames; ++frame )
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for( fpp_t frame = 0; frame < frames; ++frame )
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{
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const sample_t cur = ps->nextStringSample();
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for( Uint8 chnl = 0; chnl < DEFAULT_CHANNELS; ++chnl )
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@@ -551,7 +551,7 @@ void polyb302Synth::playNote( notePlayHandle * _n, bool )
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}
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}
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const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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sampleFrame * buf = new sampleFrame[frames];
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hstate->process( buf, frames, _n->frequency() );
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@@ -145,7 +145,7 @@ singerBot::~singerBot()
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void singerBot::playNote( notePlayHandle * _n, bool )
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{
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const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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if( !_n->m_pluginData )
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{
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@@ -293,7 +293,7 @@ void singerBot::createWave( notePlayHandle * _n )
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void singerBot::play( sampleFrame * _ab, handle_data * _hdata,
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const fpab_t _frames )
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const fpp_t _frames )
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{
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const f_cnt_t offset = _hdata->wave->num_samples()
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- _hdata->remaining_frames;
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@@ -144,7 +144,7 @@ private:
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void createWave( notePlayHandle * _n );
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void play( sampleFrame * _ab, handle_data * _hdata,
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const fpab_t _frames );
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const fpp_t _frames );
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void updateWords( void );
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} ;
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@@ -94,7 +94,7 @@ mallets::mallets( instrumentTrack * _channel_track ) :
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m_spread->move( 178, 173 );
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m_spread->setHintText( tr( "Spread:" ) + " ", "" );
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m_buffer = new sampleFrame[engine::getMixer()->framesPerAudioBuffer()];
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m_buffer = new sampleFrame[engine::getMixer()->framesPerPeriod()];
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}
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@@ -441,7 +441,7 @@ void mallets::playNote( notePlayHandle * _n, bool )
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}
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}
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const Uint32 frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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malletsSynth * ps = static_cast<malletsSynth *>( _n->m_pluginData );
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sample_t add_scale = 0.0f;
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@@ -450,7 +450,7 @@ void mallets::playNote( notePlayHandle * _n, bool )
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add_scale =
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static_cast<sample_t>( m_strike->value() ) * freq * 2.5f;
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}
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for( Uint32 frame = 0; frame < frames; ++frame )
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for( fpp_t frame = 0; frame < frames; ++frame )
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{
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const sample_t left = ps->nextSampleLeft() *
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( m_scalers[m_presets->value()] + add_scale );
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@@ -686,7 +686,7 @@ void tripleOscillator::playNote( notePlayHandle * _n, bool )
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oscillator * osc_r = static_cast<oscPtr *>( _n->m_pluginData
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)->oscRight;
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const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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sampleFrame * buf = new sampleFrame[frames];
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osc_l->update( buf, frames, 0 );
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@@ -302,7 +302,7 @@ void vestigeInstrument::waitForWorkerThread( void )
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return;
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}
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const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = engine::getMixer()->framesPerPeriod();
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sampleFrame * buf = new sampleFrame[frames];
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if( m_plugin->waitForProcessingFinished( buf ) )
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@@ -358,7 +358,7 @@ void vestigeInstrument::playNote( notePlayHandle * _n, bool )
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m_noteStates[k] = ON;
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}
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m_plugin->enqueueMidiEvent( midiEvent( NOTE_ON, 0, k,
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_n->getVolume() ), _n->framesAhead() );
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_n->getVolume() ), _n->offset() );
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// notify when the handle stops, call to deleteNotePluginData
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_n->m_pluginData = _n;
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}
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@@ -575,13 +575,13 @@ void vibed::playNote( notePlayHandle * _n, bool )
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}
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}
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const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = _n->framesLeftForCurrentPeriod();
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stringContainer * ps = static_cast<stringContainer *>(
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_n->m_pluginData );
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sampleFrame * buf = new sampleFrame[frames];
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for( fpab_t i = 0; i < frames; ++i )
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for( fpp_t i = 0; i < frames; ++i )
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{
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buf[i][0] = 0.0f;
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buf[i][1] = 0.0f;
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@@ -289,7 +289,7 @@ void remoteVSTPlugin::hideEditor( void )
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bool remoteVSTPlugin::process( const sampleFrame * _in_buf,
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sampleFrame * _out_buf, bool _wait )
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{
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const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = engine::getMixer()->framesPerPeriod();
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if( m_shm == NULL )
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{
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@@ -317,7 +317,7 @@ bool remoteVSTPlugin::process( const sampleFrame * _in_buf,
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{
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for( ch_cnt_t ch = 0; ch < inputs; ++ch )
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{
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for( fpab_t frame = 0; frame < frames; ++frame )
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for( fpp_t frame = 0; frame < frames; ++frame )
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{
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m_shm[ch * frames + frame] = _in_buf[frame][ch];
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}
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@@ -353,7 +353,7 @@ bool remoteVSTPlugin::waitForProcessingFinished( sampleFrame * _out_buf )
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usleep( 10 );
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}
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const fpab_t frames = engine::getMixer()->framesPerAudioBuffer();
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const fpp_t frames = engine::getMixer()->framesPerPeriod();
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const ch_cnt_t outputs = tMax<ch_cnt_t>( m_outputCount,
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DEFAULT_CHANNELS );
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if( outputs != DEFAULT_CHANNELS )
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@@ -364,7 +364,7 @@ bool remoteVSTPlugin::waitForProcessingFinished( sampleFrame * _out_buf )
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for( ch_cnt_t ch = 0; ch < outputs; ++ch )
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{
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for( fpab_t frame = 0; frame < frames; ++frame )
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for( fpp_t frame = 0; frame < frames; ++frame )
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{
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_out_buf[frame][ch] = m_shm[(m_inputCount+ch)*
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frames+frame];
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@@ -571,8 +571,8 @@ Sint16 remoteVSTPlugin::processNextMessage( void )
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case VST_GET_BUFFER_SIZE:
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writeValueS<Sint16>( VST_BUFFER_SIZE );
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// handle is the same
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writeValueS<fpab_t>(
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engine::getMixer()->framesPerAudioBuffer() );
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writeValueS<fpp_t>(
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engine::getMixer()->framesPerPeriod() );
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break;
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case VST_SHM_KEY_AND_SIZE:
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@@ -161,7 +161,7 @@ public:
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}
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// set given block-size for plugin
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void setBlockSize( const fpab_t _bsize );
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void setBlockSize( const fpp_t _bsize );
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// set given tempo
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void setBPM( const bpm_t _bpm )
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@@ -244,7 +244,7 @@ private:
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DWORD m_guiThreadID;
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fpab_t m_blockSize;
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fpp_t m_blockSize;
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float * m_shm;
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float * * m_inputs;
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@@ -532,7 +532,7 @@ void VSTPlugin::enqueueMidiEvent( const midiEvent & _event,
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void VSTPlugin::setBlockSize( const fpab_t _bsize )
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void VSTPlugin::setBlockSize( const fpp_t _bsize )
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{
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if( _bsize == m_blockSize )
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{
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@@ -1219,7 +1219,7 @@ int main( void )
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case VST_BUFFER_SIZE:
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plugin->setBlockSize( readValue<fpab_t>() );
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plugin->setBlockSize( readValue<fpp_t>() );
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break;
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case VST_BPM:
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@@ -88,7 +88,7 @@ vstEffect::~vstEffect()
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bool FASTCALL vstEffect::processAudioBuffer( surroundSampleFrame * _buf,
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const fpab_t _frames )
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const fpp_t _frames )
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{
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if( isBypassed() || !isRunning () )
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{
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@@ -98,7 +98,7 @@ bool FASTCALL vstEffect::processAudioBuffer( surroundSampleFrame * _buf,
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if( m_plugin )
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{
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sampleFrame * buf = new sampleFrame[_frames];
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for( fpab_t f = 0; f < _frames; ++f )
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for( fpp_t f = 0; f < _frames; ++f )
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{
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for( ch_cnt_t ch = 0; ch < DEFAULT_CHANNELS; ++ch )
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{
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@@ -109,7 +109,7 @@ bool FASTCALL vstEffect::processAudioBuffer( surroundSampleFrame * _buf,
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m_plugin->process( buf, buf, TRUE );
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m_pluginMutex.unlock();
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double out_sum = 0.0;
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for( fpab_t f = 0; f < _frames; ++f )
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for( fpp_t f = 0; f < _frames; ++f )
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{
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for( ch_cnt_t ch = 0; ch < SURROUND_CHANNELS; ++ch )
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{
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@@ -53,7 +53,7 @@ public:
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virtual ~vstEffect();
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virtual bool FASTCALL processAudioBuffer( surroundSampleFrame * _buf,
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const fpab_t _frames );
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const fpp_t _frames );
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virtual inline QString publicName( void ) const
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{
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