357 lines
12 KiB
C
357 lines
12 KiB
C
/**
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* OpenAL cross platform audio library
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* Copyright (C) 2013 by Mike Gorchak
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include <math.h>
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#include <stdlib.h>
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#include "alMain.h"
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#include "alFilter.h"
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#include "alAuxEffectSlot.h"
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#include "alError.h"
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#include "alu.h"
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/* Filters implementation is based on the "Cookbook formulae for audio *
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* EQ biquad filter coefficients" by Robert Bristow-Johnson *
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* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */
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typedef enum ALEQFilterType {
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LOWPASS,
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BANDPASS,
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} ALEQFilterType;
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typedef struct ALEQFilter {
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ALEQFilterType type;
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ALfloat x[2]; /* History of two last input samples */
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ALfloat y[2]; /* History of two last output samples */
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ALfloat a[3]; /* Transfer function coefficients "a" */
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ALfloat b[3]; /* Transfer function coefficients "b" */
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} ALEQFilter;
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typedef struct ALdistortionState {
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/* Must be first in all effects! */
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ALeffectState state;
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/* Effect gains for each channel */
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ALfloat Gain[MaxChannels];
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/* Effect parameters */
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ALEQFilter bandpass;
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ALEQFilter lowpass;
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ALfloat frequency;
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ALfloat attenuation;
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ALfloat edge;
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/* Oversample data */
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ALfloat oversample_buffer[BUFFERSIZE][4];
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} ALdistortionState;
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static ALvoid DistortionDestroy(ALeffectState *effect)
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{
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ALdistortionState *state = (ALdistortionState*)effect;
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free(state);
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}
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static ALboolean DistortionDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
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{
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ALdistortionState *state = (ALdistortionState*)effect;
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state->frequency = (ALfloat)Device->Frequency;
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return AL_TRUE;
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}
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static ALvoid DistortionUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
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{
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ALdistortionState *state = (ALdistortionState*)effect;
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ALfloat gain = sqrtf(1.0f / Device->NumChan) * Slot->Gain;
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ALuint it;
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ALfloat w0;
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ALfloat alpha;
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ALfloat bandwidth;
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ALfloat cutoff;
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for(it = 0; it < Device->NumChan; it++)
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{
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enum Channel chan = Device->Speaker2Chan[it];
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state->Gain[chan] = gain;
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}
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/* Store distorted signal attenuation settings */
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state->attenuation = Slot->effect.Distortion.Gain;
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/* Store waveshaper edge settings */
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state->edge = Slot->effect.Distortion.Edge;
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/* Lowpass filter */
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cutoff = Slot->effect.Distortion.LowpassCutoff;
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/* Bandwidth value is constant in octaves */
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bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
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w0 = 2.0f * F_PI * cutoff / (state->frequency * 4.0f);
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alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
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state->lowpass.b[0] = (1.0f - cosf(w0)) / 2.0f;
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state->lowpass.b[1] = 1.0f - cosf(w0);
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state->lowpass.b[2] = (1.0f - cosf(w0)) / 2.0f;
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state->lowpass.a[0] = 1.0f + alpha;
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state->lowpass.a[1] = -2.0f * cosf(w0);
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state->lowpass.a[2] = 1.0f - alpha;
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/* Bandpass filter */
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cutoff = Slot->effect.Distortion.EQCenter;
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/* Convert bandwidth in Hz to octaves */
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bandwidth = Slot->effect.Distortion.EQBandwidth / (cutoff * 0.67f);
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w0 = 2.0f * F_PI * cutoff / (state->frequency * 4.0f);
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alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
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state->bandpass.b[0] = alpha;
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state->bandpass.b[1] = 0;
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state->bandpass.b[2] = -alpha;
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state->bandpass.a[0] = 1.0f + alpha;
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state->bandpass.a[1] = -2.0f * cosf(w0);
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state->bandpass.a[2] = 1.0f - alpha;
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}
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static ALvoid DistortionProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
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{
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ALdistortionState *state = (ALdistortionState*)effect;
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float *RESTRICT oversample_buffer = &state->oversample_buffer[0][0];
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ALfloat tempsmp;
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ALuint it;
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ALuint kt;
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ALuint st;
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/* Perform 4x oversampling to avoid aliasing. */
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/* Oversampling greatly improves distortion */
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/* quality and allows to implement lowpass and */
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/* bandpass filters using high frequencies, at */
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/* which classic IIR filters became unstable. */
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/* Fill oversample buffer using zero stuffing */
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for(it = 0; it < SamplesToDo; it++)
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{
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oversample_buffer[it*4 + 0] = SamplesIn[it];
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oversample_buffer[it*4 + 1] = 0.0f;
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oversample_buffer[it*4 + 2] = 0.0f;
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oversample_buffer[it*4 + 3] = 0.0f;
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}
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/* First step, do lowpass filtering of original signal, */
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/* additionally perform buffer interpolation and lowpass */
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/* cutoff for oversampling (which is fortunately first */
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/* step of distortion). So combine three operations into */
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/* the one. */
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for(it = 0; it < SamplesToDo * 4; it++)
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{
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tempsmp = state->lowpass.b[0] / state->lowpass.a[0] * oversample_buffer[it] +
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state->lowpass.b[1] / state->lowpass.a[0] * state->lowpass.x[0] +
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state->lowpass.b[2] / state->lowpass.a[0] * state->lowpass.x[1] -
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state->lowpass.a[1] / state->lowpass.a[0] * state->lowpass.y[0] -
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state->lowpass.a[2] / state->lowpass.a[0] * state->lowpass.y[1];
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state->lowpass.x[1] = state->lowpass.x[0];
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state->lowpass.x[0] = oversample_buffer[it];
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state->lowpass.y[1] = state->lowpass.y[0];
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state->lowpass.y[0] = tempsmp;
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/* Restore signal power by multiplying sample by amount of oversampling */
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oversample_buffer[it] = tempsmp * 4.0f;
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}
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for(it = 0; it < SamplesToDo * 4; it++)
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{
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ALfloat smp = oversample_buffer[it];
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ALfloat edge = sinf(state->edge * (F_PI / 2.0f));
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/* Second step, do distortion using waveshaper function */
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/* to emulate signal processing during tube overdriving. */
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/* Three steps of waveshaping are intended to modify */
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/* waveform without boost/clipping/attenuation process. */
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for(st = 0; st < 3; st++)
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{
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smp = (1.0f + 2.0f * edge / (1.0f - edge)) * smp / (1.0f + 2.0f * edge / (1.0f - edge) * fabsf(smp));
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if((st & 0x00000001) == 0x00000001)
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smp *= -1.0f;
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}
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/* Third step, do bandpass filtering of distorted signal */
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tempsmp = state->bandpass.b[0] / state->bandpass.a[0] * smp +
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state->bandpass.b[1] / state->bandpass.a[0] * state->bandpass.x[0] +
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state->bandpass.b[2] / state->bandpass.a[0] * state->bandpass.x[1] -
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state->bandpass.a[1] / state->bandpass.a[0] * state->bandpass.y[0] -
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state->bandpass.a[2] / state->bandpass.a[0] * state->bandpass.y[1];
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state->bandpass.x[1] = state->bandpass.x[0];
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state->bandpass.x[0] = smp;
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state->bandpass.y[1] = state->bandpass.y[0];
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state->bandpass.y[0] = tempsmp;
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smp = tempsmp;
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/* Fourth step, final, do attenuation and perform decimation, */
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/* store only one sample out of 4. */
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if(!(it & 0x00000003))
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{
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smp *= state->attenuation;
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for(kt = 0; kt < MaxChannels; kt++)
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SamplesOut[kt][it>>2] += state->Gain[kt] * smp;
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}
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}
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}
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ALeffectState *DistortionCreate(void)
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{
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ALdistortionState *state;
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state = malloc(sizeof(*state));
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if(!state)
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return NULL;
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state->state.Destroy = DistortionDestroy;
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state->state.DeviceUpdate = DistortionDeviceUpdate;
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state->state.Update = DistortionUpdate;
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state->state.Process = DistortionProcess;
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state->bandpass.type = BANDPASS;
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state->lowpass.type = LOWPASS;
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/* Initialize sample history only on filter creation to avoid */
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/* sound clicks if filter settings were changed in runtime. */
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state->bandpass.x[0] = 0.0f;
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state->bandpass.x[1] = 0.0f;
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state->lowpass.y[0] = 0.0f;
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state->lowpass.y[1] = 0.0f;
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return &state->state;
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}
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void distortion_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
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{
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effect=effect;
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val=val;
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switch(param)
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{
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default:
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alSetError(context, AL_INVALID_ENUM);
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break;
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}
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}
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void distortion_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
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{
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distortion_SetParami(effect, context, param, vals[0]);
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}
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void distortion_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
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{
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switch(param)
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{
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case AL_DISTORTION_EDGE:
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if(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE)
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effect->Distortion.Edge = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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break;
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case AL_DISTORTION_GAIN:
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if(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN)
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effect->Distortion.Gain = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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break;
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case AL_DISTORTION_LOWPASS_CUTOFF:
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if(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF)
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effect->Distortion.LowpassCutoff = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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break;
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case AL_DISTORTION_EQCENTER:
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if(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER)
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effect->Distortion.EQCenter = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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break;
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case AL_DISTORTION_EQBANDWIDTH:
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if(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH)
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effect->Distortion.EQBandwidth = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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break;
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default:
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alSetError(context, AL_INVALID_ENUM);
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break;
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}
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}
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void distortion_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
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{
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distortion_SetParamf(effect, context, param, vals[0]);
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}
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void distortion_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
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{
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effect=effect;
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val=val;
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switch(param)
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{
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default:
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alSetError(context, AL_INVALID_ENUM);
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break;
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}
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}
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void distortion_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
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{
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distortion_GetParami(effect, context, param, vals);
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}
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void distortion_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
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{
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switch(param)
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{
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case AL_DISTORTION_EDGE:
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*val = effect->Distortion.Edge;
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break;
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case AL_DISTORTION_GAIN:
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*val = effect->Distortion.Gain;
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break;
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case AL_DISTORTION_LOWPASS_CUTOFF:
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*val = effect->Distortion.LowpassCutoff;
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break;
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case AL_DISTORTION_EQCENTER:
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*val = effect->Distortion.EQCenter;
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break;
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case AL_DISTORTION_EQBANDWIDTH:
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*val = effect->Distortion.EQBandwidth;
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break;
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default:
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alSetError(context, AL_INVALID_ENUM);
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break;
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}
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}
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void distortion_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
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{
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distortion_GetParamf(effect, context, param, vals);
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}
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