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Diffstat (limited to 'Alc/filters/filter.c')
| -rw-r--r-- | Alc/filters/filter.c | 129 |
1 files changed, 129 insertions, 0 deletions
diff --git a/Alc/filters/filter.c b/Alc/filters/filter.c new file mode 100644 index 00000000..2b370f89 --- /dev/null +++ b/Alc/filters/filter.c @@ -0,0 +1,129 @@ + +#include "config.h" + +#include "AL/alc.h" +#include "AL/al.h" + +#include "alMain.h" +#include "defs.h" + +extern inline void BiquadFilter_clear(BiquadFilter *filter); +extern inline void BiquadFilter_copyParams(BiquadFilter *restrict dst, const BiquadFilter *restrict src); +extern inline void BiquadFilter_passthru(BiquadFilter *filter, ALsizei numsamples); +extern inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope); +extern inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth); + + +void BiquadFilter_setParams(BiquadFilter *filter, BiquadType type, ALfloat gain, ALfloat f0norm, ALfloat rcpQ) +{ + ALfloat alpha, sqrtgain_alpha_2; + ALfloat w0, sin_w0, cos_w0; + ALfloat a[3] = { 1.0f, 0.0f, 0.0f }; + ALfloat b[3] = { 1.0f, 0.0f, 0.0f }; + + // Limit gain to -100dB + assert(gain > 0.00001f); + + w0 = F_TAU * f0norm; + sin_w0 = sinf(w0); + cos_w0 = cosf(w0); + alpha = sin_w0/2.0f * rcpQ; + + /* Calculate filter coefficients depending on filter type */ + switch(type) + { + case BiquadType_HighShelf: + sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha; + b[0] = gain*((gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2); + b[1] = -2.0f*gain*((gain-1.0f) + (gain+1.0f)*cos_w0 ); + b[2] = gain*((gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2); + a[0] = (gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2; + a[1] = 2.0f* ((gain-1.0f) - (gain+1.0f)*cos_w0 ); + a[2] = (gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2; + break; + case BiquadType_LowShelf: + sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha; + b[0] = gain*((gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2); + b[1] = 2.0f*gain*((gain-1.0f) - (gain+1.0f)*cos_w0 ); + b[2] = gain*((gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2); + a[0] = (gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2; + a[1] = -2.0f* ((gain-1.0f) + (gain+1.0f)*cos_w0 ); + a[2] = (gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2; + break; + case BiquadType_Peaking: + gain = sqrtf(gain); + b[0] = 1.0f + alpha * gain; + b[1] = -2.0f * cos_w0; + b[2] = 1.0f - alpha * gain; + a[0] = 1.0f + alpha / gain; + a[1] = -2.0f * cos_w0; + a[2] = 1.0f - alpha / gain; + break; + + case BiquadType_LowPass: + b[0] = (1.0f - cos_w0) / 2.0f; + b[1] = 1.0f - cos_w0; + b[2] = (1.0f - cos_w0) / 2.0f; + a[0] = 1.0f + alpha; + a[1] = -2.0f * cos_w0; + a[2] = 1.0f - alpha; + break; + case BiquadType_HighPass: + b[0] = (1.0f + cos_w0) / 2.0f; + b[1] = -(1.0f + cos_w0); + b[2] = (1.0f + cos_w0) / 2.0f; + a[0] = 1.0f + alpha; + a[1] = -2.0f * cos_w0; + a[2] = 1.0f - alpha; + break; + case BiquadType_BandPass: + b[0] = alpha; + b[1] = 0; + b[2] = -alpha; + a[0] = 1.0f + alpha; + a[1] = -2.0f * cos_w0; + a[2] = 1.0f - alpha; + break; + } + + filter->a1 = a[1] / a[0]; + filter->a2 = a[2] / a[0]; + filter->b0 = b[0] / a[0]; + filter->b1 = b[1] / a[0]; + filter->b2 = b[2] / a[0]; +} + + +void BiquadFilter_processC(BiquadFilter *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples) +{ + const ALfloat a1 = filter->a1; + const ALfloat a2 = filter->a2; + const ALfloat b0 = filter->b0; + const ALfloat b1 = filter->b1; + const ALfloat b2 = filter->b2; + ALfloat z1 = filter->z1; + ALfloat z2 = filter->z2; + ALsizei i; + + ASSUME(numsamples > 0); + + /* Processing loop is Transposed Direct Form II. This requires less storage + * compared to Direct Form I (only two delay components, instead of a four- + * sample history; the last two inputs and outputs), and works better for + * floating-point which favors summing similarly-sized values while being + * less bothered by overflow. + * + * See: http://www.earlevel.com/main/2003/02/28/biquads/ + */ + for(i = 0;i < numsamples;i++) + { + ALfloat input = src[i]; + ALfloat output = input*b0 + z1; + z1 = input*b1 - output*a1 + z2; + z2 = input*b2 - output*a2; + dst[i] = output; + } + + filter->z1 = z1; + filter->z2 = z2; +} |