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Avoid writing to the same buffer that's read from

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Also clean up comment formatting a bit.
This commit is contained in:
Chris Robinson 2016-12-23 12:37:48 -08:00
parent 40f359d159
commit fcdf1cea70
1 changed files with 36 additions and 36 deletions
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72
Alc/effects/distortion.c
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@ -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;