diff --git a/include/basic_filters.h b/include/basic_filters.h index 207e35c86..52465e17d 100644 --- a/include/basic_filters.h +++ b/include/basic_filters.h @@ -72,6 +72,8 @@ public: Lowpass_SV, Bandpass_SV, Highpass_SV, + Notch_SV, + FastFormant, NumFilters } ; @@ -202,21 +204,26 @@ public: } // 4-pole state-variant lowpass filter, adapted from Nekobee source code + // and extended to other SV filter types // /* Hal Chamberlin's state variable filter */ case Lowpass_SV: case Bandpass_SV: { m_sva[_chnl] += ( qAbs( _in0 ) - m_sva[_chnl] ) * m_svsr; + float highpass; - m_delay2[_chnl] = m_delay2[_chnl] + m_svf1 * m_delay1[_chnl]; /* delay2/4 = lowpass output */ - float highpass = _in0 - m_delay2[_chnl] - m_svq * m_delay1[_chnl]; - m_delay1[_chnl] = m_svf1 * highpass + m_delay1[_chnl]; /* delay1/3 = bandpass output */ + for( int i = 0; i < 2; ++i ) // 2x oversample + { + m_delay2[_chnl] = m_delay2[_chnl] + m_svf1 * m_delay1[_chnl]; /* delay2/4 = lowpass output */ + highpass = _in0 - m_delay2[_chnl] - m_svq * m_delay1[_chnl]; + m_delay1[_chnl] = m_svf1 * highpass + m_delay1[_chnl]; /* delay1/3 = bandpass output */ + + m_delay4[_chnl] = m_delay4[_chnl] + m_svf2 * m_delay3[_chnl]; + highpass = m_delay2[_chnl] - m_delay4[_chnl] - m_svq * m_delay3[_chnl]; + m_delay3[_chnl] = m_svf2 * highpass + m_delay3[_chnl]; + } - m_delay4[_chnl] = m_delay4[_chnl] + m_svf2 * m_delay3[_chnl]; - highpass = m_delay2[_chnl] - m_delay4[_chnl] - m_svq * m_delay3[_chnl]; - m_delay3[_chnl] = m_svf2 * highpass + m_delay3[_chnl]; - /* mix filter output into output buffer */ out = m_type == Lowpass_SV ? atanf( 3.0f * m_delay4[_chnl] * m_sva[_chnl] ) @@ -227,14 +234,39 @@ public: case Highpass_SV: { m_sva[_chnl] += ( qAbs( _in0 ) - m_sva[_chnl] ) * m_svsr; - - m_delay2[_chnl] = m_delay2[_chnl] + m_svf1 * m_delay1[_chnl]; - float hp = _in0 - m_delay2[_chnl] - m_svq * m_delay1[_chnl]; - m_delay1[_chnl] = m_svf1 * hp + m_delay1[_chnl]; + float hp; + + for( int i = 0; i < 2; ++i ) // 2x oversample + { + m_delay2[_chnl] = m_delay2[_chnl] + m_svf1 * m_delay1[_chnl]; + hp = _in0 - m_delay2[_chnl] - m_svq * m_delay1[_chnl]; + m_delay1[_chnl] = m_svf1 * hp + m_delay1[_chnl]; + } out = atanf( 3.0f * hp * m_sva[_chnl] ); break; } + + case Notch_SV: + { + m_sva[_chnl] += ( qAbs( _in0 ) - m_sva[_chnl] ) * m_svsr; + float hp1, hp2; + + for( int i = 0; i < 2; ++i ) // 2x oversample + { + m_delay2[_chnl] = m_delay2[_chnl] + m_svf1 * m_delay1[_chnl]; /* delay2/4 = lowpass output */ + hp1 = _in0 - m_delay2[_chnl] - m_svq * m_delay1[_chnl]; + m_delay1[_chnl] = m_svf1 * hp1 + m_delay1[_chnl]; /* delay1/3 = bandpass output */ + + m_delay4[_chnl] = m_delay4[_chnl] + m_svf2 * m_delay3[_chnl]; + hp2 = m_delay2[_chnl] - m_delay4[_chnl] - m_svq * m_delay3[_chnl]; + m_delay3[_chnl] = m_svf2 * hp2 + m_delay3[_chnl]; + } + + /* mix filter output into output buffer */ + out = atanf( 1.5f * ( m_delay4[_chnl] + hp2 ) * m_sva[_chnl] ); + break; + } // 4-times oversampled simulation of an active RC-Bandpass,-Lowpass,-Highpass- @@ -376,11 +408,13 @@ public: } case Formantfilter: + case FastFormant: { sample_t hp, bp, in; out = 0; - for(int o=0; o<4; o++) + const int os = m_type == FastFormant ? 1 : 4; // no oversampling for fast formant + for( int o = 0; o < os; ++o ) { // first formant in = _in0 + m_vfbp[0][_chnl] * m_vfq; @@ -466,7 +500,7 @@ public: out += bp; } - return( out/2.0f ); + return( out * 0.5f ); break; } @@ -511,16 +545,17 @@ public: { _freq = qBound( 50.0f, _freq, 20000.0f ); - m_rca = 1.0f - (1.0f/(m_sampleRate*4)) / ( (1.0f/(_freq*2.0f*M_PI)) + (1.0f/(m_sampleRate*4)) ); + m_rca = 1.0f - (1.0f/(m_sampleRate*4)) / ( (1.0f/(_freq*2.0f*F_PI)) + (1.0f/(m_sampleRate*4)) ); m_rcb = 1.0f - m_rca; - m_rcc = (1.0f/(_freq*2.0f*M_PI)) / ( (1.0f/(_freq*2.0f*M_PI)) + (1.0f/(m_sampleRate*4)) ); + m_rcc = (1.0f/(_freq*2.0f*F_PI)) / ( (1.0f/(_freq*2.0f*F_PI)) + (1.0f/(m_sampleRate*4)) ); // Stretch Q/resonance, as self-oscillation reliably starts at a q of ~2.5 - ~2.6 m_rcq = _q * 0.25f; return; } - if( m_type == Formantfilter ) + if( m_type == Formantfilter || + m_type == FastFormant ) { _freq = qBound( minFreq(), _freq, 20000.0f ); // limit freq and q for not getting bad noise out of the filter... @@ -543,21 +578,19 @@ public: const float f0 = linearInterpolate( _f[vowel+0][0], _f[vowel+1][0], fract ); const float f1 = linearInterpolate( _f[vowel+0][1], _f[vowel+1][1], fract ); - m_vfa[0] = 1.0f - (1.0f/(m_sampleRate*4)) / - ( (1.0f/(f0*2.0f*M_PI)) + - (1.0f/(m_sampleRate*4)) ); - m_vfb[0] = 1.0f - m_vfa[0]; - m_vfc[0] = (1.0f/(f0*2.0f*M_PI)) / - ( (1.0f/(f0*2.0f*M_PI)) + - (1.0f/(m_sampleRate*4)) ); + // samplerate coeff: depends on oversampling + const float sr = m_type == FastFormant ? m_sampleRatio : m_sampleRatio * 0.25f; - m_vfa[1] = 1.0f - (1.0f/(m_sampleRate*4)) / - ( (1.0f/(f1*2.0f*M_PI)) + - (1.0f/(m_sampleRate*4)) ); + m_vfa[0] = 1.0f - sr / + ( ( 1.0f / ( f0 * 2.0f * F_PI ) ) + sr ); + m_vfb[0] = 1.0f - m_vfa[0]; + m_vfc[0] = ( 1.0f / ( f0 * 2.0f * F_PI ) ) / + ( ( 1.0f / ( f0 *2.0f * F_PI ) ) + sr ); + m_vfa[1] = 1.0f - sr / + ( ( 1.0f / ( f1 * 2.0f * F_PI ) ) + sr ); m_vfb[1] = 1.0f - m_vfa[1]; - m_vfc[1] = (1.0f/(f1*2.0f*M_PI)) / - ( (1.0f/(f1*2.0f*M_PI)) + - (1.0f/(m_sampleRate*4)) ); + m_vfc[1] = ( 1.0f / ( f1 * 2.0f * F_PI ) ) / + ( ( 1.0f / ( f1 * 2.0f * F_PI ) ) + sr ); return; } @@ -584,11 +617,12 @@ public: if( m_type == Lowpass_SV || m_type == Bandpass_SV || - m_type == Highpass_SV ) + m_type == Highpass_SV || + m_type == Notch_SV ) { - const float f = qMax( minFreq(), _freq ) * m_sampleRatio; - m_svf1 = qMin( f * 2.0f, 0.825f ); - m_svf2 = qMin( f * 4.0f, 0.825f ); + const float f = sinf( qMax( minFreq(), _freq ) * m_sampleRatio * F_PI ); + m_svf1 = qMin( f, 0.825f ); + m_svf2 = qMin( f * 2.0f, 0.825f ); m_svq = qMax( 0.0001f, 2.0f - ( _q * 0.1995f ) ); return; } @@ -598,12 +632,7 @@ public: const float omega = F_2PI * _freq * m_sampleRatio; const float tsin = sinf( omega ); const float tcos = cosf( omega ); - //float alpha; - //if (q_is_bandwidth) - //alpha = tsin*sinhf(logf(2.0f)/2.0f*q*omega/ - // tsin); - //else const float alpha = 0.5f * tsin / _q; const float a0 = 1.0f / ( 1.0f + alpha ); diff --git a/src/core/InstrumentSoundShaping.cpp b/src/core/InstrumentSoundShaping.cpp index 02d3182bb..4e32c9740 100644 --- a/src/core/InstrumentSoundShaping.cpp +++ b/src/core/InstrumentSoundShaping.cpp @@ -98,6 +98,8 @@ InstrumentSoundShaping::InstrumentSoundShaping( m_filterModel.addItem( tr( "SV LowPass" ), new PixmapLoader( "filter_lp" ) ); m_filterModel.addItem( tr( "SV BandPass" ), new PixmapLoader( "filter_bp" ) ); m_filterModel.addItem( tr( "SV HighPass" ), new PixmapLoader( "filter_hp" ) ); + m_filterModel.addItem( tr( "SV Notch" ), new PixmapLoader( "filter_notch" ) ); + m_filterModel.addItem( tr( "Fast Formant" ), new PixmapLoader( "filter_hp" ) ); }