Use ALfilterState for the distortion effect filters
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Chris Robinson
parent
dcefeac6e6
commit
647398d7c6
@ -37,23 +37,6 @@ typedef struct ALdistortionStateFactory {
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static ALdistortionStateFactory DistortionFactory;
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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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DERIVE_FROM_TYPE(ALeffectState);
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@ -61,8 +44,8 @@ typedef struct ALdistortionState {
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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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ALfilterState lowpass;
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ALfilterState bandpass;
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ALfloat attenuation;
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ALfloat edge_coeff;
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} ALdistortionState;
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@ -84,8 +67,6 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, ALCdevice *Devi
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ALfloat gain = sqrtf(1.0f / Device->NumChan) * Slot->Gain;
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ALfloat frequency = (ALfloat)Device->Frequency;
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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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ALfloat edge;
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@ -109,34 +90,21 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, ALCdevice *Devi
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cutoff = Slot->EffectProps.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 / (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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ALfilterState_setParams(&state->lowpass, ALfilterType_LowPass, 1.0f,
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cutoff / (frequency*4.0f), bandwidth);
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/* Bandpass filter */
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cutoff = Slot->EffectProps.Distortion.EQCenter;
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/* Convert bandwidth in Hz to octaves */
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bandwidth = Slot->EffectProps.Distortion.EQBandwidth / (cutoff * 0.67f);
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w0 = 2.0f*F_PI * cutoff / (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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ALfilterState_setParams(&state->bandpass, ALfilterType_BandPass, 1.0f,
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cutoff / (frequency*4.0f), bandwidth);
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}
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static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
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{
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const ALfloat fc = state->edge_coeff;
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float oversample_buffer[64][4];
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ALfloat tempsmp;
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ALuint base;
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ALuint it;
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ALuint ot;
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@ -171,18 +139,11 @@ static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint Samples
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{
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for(ot = 0;ot < 4;ot++)
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{
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tempsmp = state->lowpass.b[0] / state->lowpass.a[0] * oversample_buffer[it][ot] +
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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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ALfloat smp;
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smp = ALfilterState_processSingle(&state->lowpass, oversample_buffer[it][ot]);
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state->lowpass.x[1] = state->lowpass.x[0];
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state->lowpass.x[0] = oversample_buffer[it][ot];
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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][ot] = tempsmp * 4.0f;
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oversample_buffer[it][ot] = smp * 4.0f;
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}
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}
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@ -201,18 +162,8 @@ static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint Samples
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smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
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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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oversample_buffer[it][ot] = tempsmp;
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smp = ALfilterState_processSingle(&state->bandpass, smp);
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oversample_buffer[it][ot] = smp;
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}
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/* Fourth step, final, do attenuation and perform decimation, */
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@ -251,15 +202,8 @@ static ALeffectState *ALdistortionStateFactory_create(ALdistortionStateFactory *
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if(!state) return NULL;
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SET_VTABLE2(ALdistortionState, ALeffectState, state);
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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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ALfilterState_clear(&state->lowpass);
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ALfilterState_clear(&state->bandpass);
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return STATIC_CAST(ALeffectState, state);
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}
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@ -9,10 +9,18 @@ extern "C" {
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#define LOWPASSFREQREF (5000)
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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 ALfilterType {
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ALfilterType_HighShelf,
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ALfilterType_LowShelf,
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ALfilterType_Peaking,
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ALfilterType_LowPass,
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ALfilterType_BandPass,
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} ALfilterType;
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typedef struct ALfilterState {
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@ -392,6 +392,25 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
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filter->a[1] = -2.0f * cosf(w0);
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filter->a[2] = 1.0f - alpha / gain;
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break;
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case ALfilterType_LowPass:
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alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
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filter->b[0] = (1.0f - cosf(w0)) / 2.0f;
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filter->b[1] = 1.0f - cosf(w0);
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filter->b[2] = (1.0f - cosf(w0)) / 2.0f;
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filter->a[0] = 1.0f + alpha;
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filter->a[1] = -2.0f * cosf(w0);
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filter->a[2] = 1.0f - alpha;
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break;
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case ALfilterType_BandPass:
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alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
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filter->b[0] = alpha;
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filter->b[1] = 0;
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filter->b[2] = -alpha;
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filter->a[0] = 1.0f + alpha;
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filter->a[1] = -2.0f * cosf(w0);
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filter->a[2] = 1.0f - alpha;
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break;
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}
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filter->b[2] /= filter->a[0];
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