Use constants from math_funcs rather than unstandardized C++ constants.
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7 changed files with 41 additions and 34 deletions
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@ -1,5 +1,6 @@
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#include "audio_effect_chorus.h"
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#include "audio_effect_chorus.h"
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#include "servers/audio_server.h"
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#include "servers/audio_server.h"
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#include "math_funcs.h"
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void AudioEffectChorusInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
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void AudioEffectChorusInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
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@ -61,7 +62,7 @@ void AudioEffectChorusInstance::_process_chunk(const AudioFrame *p_src_frames,Au
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//low pass filter
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//low pass filter
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if (v.cutoff==0)
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if (v.cutoff==0)
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continue;
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continue;
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float auxlp=expf(-2.0*M_PI*v.cutoff/mix_rate);
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float auxlp=expf(-2.0*Math_PI*v.cutoff/mix_rate);
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float c1=1.0-auxlp;
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float c1=1.0-auxlp;
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float c2=auxlp;
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float c2=auxlp;
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AudioFrame h=filter_h[vc];
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AudioFrame h=filter_h[vc];
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@ -83,7 +84,7 @@ void AudioEffectChorusInstance::_process_chunk(const AudioFrame *p_src_frames,Au
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float phase=(float)(local_cycles&AudioEffectChorus::CYCLES_MASK)/(float)(1<<AudioEffectChorus::CYCLES_FRAC);
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float phase=(float)(local_cycles&AudioEffectChorus::CYCLES_MASK)/(float)(1<<AudioEffectChorus::CYCLES_FRAC);
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float wave_delay=sinf(phase*2.0*M_PI)*max_depth_frames;
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float wave_delay=sinf(phase*2.0*Math_PI)*max_depth_frames;
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int wave_delay_frames=lrint(floor(wave_delay));
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int wave_delay_frames=lrint(floor(wave_delay));
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float wave_delay_frac=wave_delay-(float)wave_delay_frames;
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float wave_delay_frac=wave_delay-(float)wave_delay_frames;
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@ -1,5 +1,6 @@
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#include "audio_effect_delay.h"
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#include "audio_effect_delay.h"
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#include "servers/audio_server.h"
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#include "servers/audio_server.h"
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#include "math_funcs.h"
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void AudioEffectDelayInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
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void AudioEffectDelayInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
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@ -48,7 +49,7 @@ void AudioEffectDelayInstance::_process_chunk(const AudioFrame *p_src_frames,Aud
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tap2_vol.r*=CLAMP( 1.0 + base->tap_2_pan, 0, 1);
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tap2_vol.r*=CLAMP( 1.0 + base->tap_2_pan, 0, 1);
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// feedback lowpass here
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// feedback lowpass here
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float lpf_c=expf(-2.0*M_PI*base->feedback_lowpass/mix_rate); // 0 .. 10khz
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float lpf_c=expf(-2.0*Math_PI*base->feedback_lowpass/mix_rate); // 0 .. 10khz
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float lpf_ic=1.0-lpf_c;
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float lpf_ic=1.0-lpf_c;
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const AudioFrame *src=p_src_frames;
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const AudioFrame *src=p_src_frames;
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@ -1,5 +1,6 @@
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#include "audio_effect_distortion.h"
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#include "audio_effect_distortion.h"
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#include "servers/audio_server.h"
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#include "servers/audio_server.h"
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#include "math_funcs.h"
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@ -8,8 +9,8 @@ void AudioEffectDistortionInstance::process(const AudioFrame *p_src_frames,Audio
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const float *src = (const float*)p_src_frames;
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const float *src = (const float*)p_src_frames;
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float *dst = (float*)p_dst_frames;
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float *dst = (float*)p_dst_frames;
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//float lpf_c=expf(-2.0*M_PI*keep_hf_hz.get()/(mix_rate*(float)OVERSAMPLE));
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//float lpf_c=expf(-2.0*Math_PI*keep_hf_hz.get()/(mix_rate*(float)OVERSAMPLE));
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float lpf_c=expf(-2.0*M_PI*base->keep_hf_hz/(AudioServer::get_singleton()->get_mix_rate()));
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float lpf_c=expf(-2.0*Math_PI*base->keep_hf_hz/(AudioServer::get_singleton()->get_mix_rate()));
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float lpf_ic=1.0-lpf_c;
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float lpf_ic=1.0-lpf_c;
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float drive_f=base->drive;
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float drive_f=base->drive;
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@ -1,5 +1,6 @@
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#include "audio_effect_phaser.h"
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#include "audio_effect_phaser.h"
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#include "servers/audio_server.h"
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#include "servers/audio_server.h"
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#include "math_funcs.h"
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void AudioEffectPhaserInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
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void AudioEffectPhaserInstance::process(const AudioFrame *p_src_frames,AudioFrame *p_dst_frames,int p_frame_count) {
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@ -8,14 +9,14 @@ void AudioEffectPhaserInstance::process(const AudioFrame *p_src_frames,AudioFram
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float dmin = base->range_min / (sampling_rate/2.0);
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float dmin = base->range_min / (sampling_rate/2.0);
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float dmax = base->range_max / (sampling_rate/2.0);
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float dmax = base->range_max / (sampling_rate/2.0);
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float increment = 2.f * M_PI * (base->rate / sampling_rate);
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float increment = 2.f * Math_PI * (base->rate / sampling_rate);
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for(int i=0;i<p_frame_count;i++) {
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for(int i=0;i<p_frame_count;i++) {
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phase += increment;
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phase += increment;
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while ( phase >= M_PI * 2.f ) {
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while ( phase >= Math_PI * 2.f ) {
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phase -= M_PI * 2.f;
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phase -= Math_PI * 2.f;
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}
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}
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float d = dmin + (dmax-dmin) * ((sin( phase ) + 1.f)/2.f);
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float d = dmin + (dmax-dmin) * ((sin( phase ) + 1.f)/2.f);
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@ -1,5 +1,6 @@
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#include "audio_effect_pitch_shift.h"
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#include "audio_effect_pitch_shift.h"
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#include "servers/audio_server.h"
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#include "servers/audio_server.h"
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#include "math_funcs.h"
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/****************************************************************************
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/****************************************************************************
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*
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*
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* NAME: smbPitchShift.cpp
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* NAME: smbPitchShift.cpp
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@ -57,7 +58,7 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
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fftFrameSize2 = fftFrameSize/2;
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fftFrameSize2 = fftFrameSize/2;
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stepSize = fftFrameSize/osamp;
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stepSize = fftFrameSize/osamp;
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freqPerBin = sampleRate/(double)fftFrameSize;
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freqPerBin = sampleRate/(double)fftFrameSize;
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expct = 2.*M_PI*(double)stepSize/(double)fftFrameSize;
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expct = 2.*Math_PI*(double)stepSize/(double)fftFrameSize;
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inFifoLatency = fftFrameSize-stepSize;
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inFifoLatency = fftFrameSize-stepSize;
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if (gRover == 0) gRover = inFifoLatency;
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if (gRover == 0) gRover = inFifoLatency;
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@ -77,7 +78,7 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
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/* do windowing and re,im interleave */
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/* do windowing and re,im interleave */
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for (k = 0; k < fftFrameSize;k++) {
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for (k = 0; k < fftFrameSize;k++) {
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window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
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window = -.5*cos(2.*Math_PI*(double)k/(double)fftFrameSize)+.5;
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gFFTworksp[2*k] = gInFIFO[k] * window;
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gFFTworksp[2*k] = gInFIFO[k] * window;
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gFFTworksp[2*k+1] = 0.;
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gFFTworksp[2*k+1] = 0.;
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}
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}
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@ -106,13 +107,13 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
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tmp -= (double)k*expct;
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tmp -= (double)k*expct;
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/* map delta phase into +/- Pi interval */
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/* map delta phase into +/- Pi interval */
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qpd = tmp/M_PI;
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qpd = tmp/Math_PI;
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if (qpd >= 0) qpd += qpd&1;
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if (qpd >= 0) qpd += qpd&1;
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else qpd -= qpd&1;
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else qpd -= qpd&1;
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tmp -= M_PI*(double)qpd;
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tmp -= Math_PI*(double)qpd;
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/* get deviation from bin frequency from the +/- Pi interval */
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/* get deviation from bin frequency from the +/- Pi interval */
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tmp = osamp*tmp/(2.*M_PI);
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tmp = osamp*tmp/(2.*Math_PI);
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/* compute the k-th partials' true frequency */
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/* compute the k-th partials' true frequency */
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tmp = (double)k*freqPerBin + tmp*freqPerBin;
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tmp = (double)k*freqPerBin + tmp*freqPerBin;
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@ -150,7 +151,7 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
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tmp /= freqPerBin;
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tmp /= freqPerBin;
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/* take osamp into account */
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/* take osamp into account */
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tmp = 2.*M_PI*tmp/osamp;
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tmp = 2.*Math_PI*tmp/osamp;
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/* add the overlap phase advance back in */
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/* add the overlap phase advance back in */
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tmp += (double)k*expct;
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tmp += (double)k*expct;
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@ -172,7 +173,7 @@ void SMBPitchShift::PitchShift(float pitchShift, long numSampsToProcess, long ff
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/* do windowing and add to output accumulator */
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/* do windowing and add to output accumulator */
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for(k=0; k < fftFrameSize; k++) {
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for(k=0; k < fftFrameSize; k++) {
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window = -.5*cos(2.*M_PI*(double)k/(double)fftFrameSize)+.5;
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window = -.5*cos(2.*Math_PI*(double)k/(double)fftFrameSize)+.5;
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gOutputAccum[k] += 2.*window*gFFTworksp[2*k]/(fftFrameSize2*osamp);
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gOutputAccum[k] += 2.*window*gFFTworksp[2*k]/(fftFrameSize2*osamp);
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}
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}
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for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
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for (k = 0; k < stepSize; k++) gOutFIFO[k] = gOutputAccum[k];
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@ -224,7 +225,7 @@ void SMBPitchShift::smbFft(float *fftBuffer, long fftFrameSize, long sign)
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le2 = le>>1;
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le2 = le>>1;
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ur = 1.0;
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ur = 1.0;
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ui = 0.0;
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ui = 0.0;
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arg = M_PI / (le2>>1);
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arg = Math_PI / (le2>>1);
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wr = cos(arg);
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wr = cos(arg);
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wi = sign*sin(arg);
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wi = sign*sin(arg);
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for (j = 0; j < le2; j += 2) {
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for (j = 0; j < le2; j += 2) {
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@ -12,6 +12,7 @@
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#include "eq.h"
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#include "eq.h"
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#include <math.h>
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#include <math.h>
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#include "error_macros.h"
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#include "error_macros.h"
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#include "math_funcs.h"
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#define POW2(v) ((v)*(v))
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#define POW2(v) ((v)*(v))
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double side_gain2=POW2(1.0/M_SQRT2);
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double side_gain2=POW2(Math_SQRT12);
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double th=2.0*M_PI*frq/mix_rate;
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double th=2.0*Math_PI*frq/mix_rate;
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double th_l=2.0*M_PI*frq_l/mix_rate;
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double th_l=2.0*Math_PI*frq_l/mix_rate;
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double c2a=side_gain2 * POW2(cos(th))
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double c2a=side_gain2 * POW2(cos(th))
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- 2.0 * side_gain2 * cos(th_l) * cos(th)
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- 2.0 * side_gain2 * cos(th_l) * cos(th)
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@ -12,6 +12,7 @@
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#include "reverb.h"
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#include "reverb.h"
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#include <math.h>
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#include <math.h>
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#include "math_funcs.h"
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const float Reverb::comb_tunings[MAX_COMBS]={
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const float Reverb::comb_tunings[MAX_COMBS]={
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@ -68,7 +69,7 @@ void Reverb::process(float *p_src,float *p_dst,int p_frames) {
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}
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}
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if (params.hpf>0) {
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if (params.hpf>0) {
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float hpaux=expf(-2.0*M_PI*params.hpf*6000/params.mix_rate);
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float hpaux=expf(-2.0*Math_PI*params.hpf*6000/params.mix_rate);
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float hp_a1=(1.0+hpaux)/2.0;
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float hp_a1=(1.0+hpaux)/2.0;
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float hp_a2=-(1.0+hpaux)/2.0;
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float hp_a2=-(1.0+hpaux)/2.0;
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float hp_b1=hpaux;
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float hp_b1=hpaux;
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float auxdmp=params.damp/2.0+0.5; //only half the range (0.5 .. 1.0 is enough)
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float auxdmp=params.damp/2.0+0.5; //only half the range (0.5 .. 1.0 is enough)
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auxdmp*=auxdmp;
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auxdmp*=auxdmp;
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c.damp=expf(-2.0*M_PI*auxdmp*10000/params.mix_rate); // 0 .. 10khz
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c.damp=expf(-2.0*Math_PI*auxdmp*10000/params.mix_rate); // 0 .. 10khz
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}
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}
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}
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}
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