diff options
Diffstat (limited to 'Alc')
| -rw-r--r-- | Alc/effects/distortion.c | 72 |
1 files changed, 36 insertions, 36 deletions
diff --git a/Alc/effects/distortion.c b/Alc/effects/distortion.c index bc1e7d84..84063e84 100644 --- a/Alc/effects/distortion.c +++ b/Alc/effects/distortion.c @@ -78,25 +78,26 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCdevice ALfloat cutoff; ALfloat edge; - /* Store distorted signal attenuation settings */ + /* Store distorted signal attenuation settings. */ state->attenuation = props->Distortion.Gain; - /* Store waveshaper edge settings */ + /* Store waveshaper edge settings. */ edge = sinf(props->Distortion.Edge * (F_PI_2)); edge = minf(edge, 0.99f); state->edge_coeff = 2.0f * edge / (1.0f-edge); - /* Lowpass filter */ cutoff = props->Distortion.LowpassCutoff; - /* Bandwidth value is constant in octaves */ + /* Bandwidth value is constant in octaves. */ bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f); + /* Multiply sampling frequency by the amount of oversampling done during + * processing. + */ ALfilterState_setParams(&state->lowpass, ALfilterType_LowPass, 1.0f, cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth) ); - /* Bandpass filter */ cutoff = props->Distortion.EQCenter; - /* Convert bandwidth in Hz to octaves */ + /* Convert bandwidth in Hz to octaves. */ bandwidth = props->Distortion.EQBandwidth / (cutoff * 0.67f); ALfilterState_setParams(&state->bandpass, ALfilterType_BandPass, 1.0f, cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth) @@ -110,7 +111,6 @@ static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint Samples const ALfloat fc = state->edge_coeff; ALuint base; ALuint it; - ALuint ot; ALuint kt; for(base = 0;base < SamplesToDo;) @@ -118,49 +118,49 @@ static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint Samples float buffer[2][64 * 4]; ALuint td = minu(64, SamplesToDo-base); - /* Perform 4x oversampling to avoid aliasing. */ - /* Oversampling greatly improves distortion */ - /* quality and allows to implement lowpass and */ - /* bandpass filters using high frequencies, at */ - /* which classic IIR filters became unstable. */ + /* Perform 4x oversampling to avoid aliasing. Oversampling greatly + * improves distortion quality and allows to implement lowpass and + * bandpass filters using high frequencies, at which classic IIR + * filters became unstable. + */ - /* Fill oversample buffer using zero stuffing */ + /* Fill oversample buffer using zero stuffing. */ for(it = 0;it < td;it++) { - buffer[0][it*4 + 0] = SamplesIn[0][it+base]; + /* Multiply the sample by the amount of oversampling to maintain + * the signal's power. + */ + buffer[0][it*4 + 0] = SamplesIn[0][it+base] * 4.0f; buffer[0][it*4 + 1] = 0.0f; buffer[0][it*4 + 2] = 0.0f; buffer[0][it*4 + 3] = 0.0f; } - /* First step, do lowpass filtering of original signal, */ - /* additionally perform buffer interpolation and lowpass */ - /* cutoff for oversampling (which is fortunately first */ - /* step of distortion). So combine three operations into */ - /* the one. */ + /* First step, do lowpass filtering of original signal. Additionally + * perform buffer interpolation and lowpass cutoff for oversampling + * (which is fortunately first step of distortion). So combine three + * operations into the one. + */ ALfilterState_process(&state->lowpass, buffer[1], buffer[0], td*4); - /* Second step, do distortion using waveshaper function */ - /* to emulate signal processing during tube overdriving. */ - /* Three steps of waveshaping are intended to modify */ - /* waveform without boost/clipping/attenuation process. */ - for(it = 0;it < td;it++) + /* Second step, do distortion using waveshaper function to emulate + * signal processing during tube overdriving. Three steps of + * waveshaping are intended to modify waveform without boost/clipping/ + * attenuation process. + */ + for(it = 0;it < td*4;it++) { - for(ot = 0;ot < 4;ot++) - { - /* Restore signal power by multiplying sample by amount of oversampling */ - ALfloat smp = buffer[1][it*4 + ot] * 4.0f; + ALfloat smp = buffer[1][it]; - smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)); - smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f; - smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)); + smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)); + smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f; + smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)); - buffer[1][it*4 + ot] = smp; - } + buffer[0][it] = smp; } - /* Third step, do bandpass filtering of distorted signal */ - ALfilterState_process(&state->bandpass, buffer[0], buffer[1], td*4); + /* Third step, do bandpass filtering of distorted signal. */ + ALfilterState_process(&state->bandpass, buffer[1], buffer[0], td*4); for(kt = 0;kt < NumChannels;kt++) { @@ -172,7 +172,7 @@ static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint Samples continue; for(it = 0;it < td;it++) - SamplesOut[kt][base+it] += gain * buffer[0][it*4]; + SamplesOut[kt][base+it] += gain * buffer[1][it*4]; } base += td; |