diff options
| -rw-r--r-- | Alc/effects/equalizer.c | 123 |
1 files changed, 90 insertions, 33 deletions
diff --git a/Alc/effects/equalizer.c b/Alc/effects/equalizer.c index 8ef52986..0a1ff1db 100644 --- a/Alc/effects/equalizer.c +++ b/Alc/effects/equalizer.c @@ -71,14 +71,20 @@ * filter coefficients" by Robert Bristow-Johnson * * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */ + +/* The maximum number of sample frames per update. */ +#define MAX_UPDATE_SAMPLES 256 + typedef struct ALequalizerState { DERIVE_FROM_TYPE(ALeffectState); /* Effect gains for each channel */ - ALfloat Gain[MAX_OUTPUT_CHANNELS]; + ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS]; /* Effect parameters */ - ALfilterState filter[4]; + ALfilterState filter[4][MAX_EFFECT_CHANNELS]; + + ALfloat SampleBuffer[4][MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES]; } ALequalizerState; static ALvoid ALequalizerState_Destruct(ALequalizerState *UNUSED(state)) @@ -94,8 +100,19 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice * { ALfloat frequency = (ALfloat)device->Frequency; ALfloat gain, freq_mult; - - ComputeAmbientGains(device->AmbiCoeffs, device->NumChannels, slot->Gain, state->Gain); + aluMatrixf matrix; + ALuint i; + + gain = device->AmbiScale; + aluMatrixfSet(&matrix, + 1.0f, 0.0f, 0.0f, 0.0f, + 0.0f, gain, 0.0f, 0.0f, + 0.0f, 0.0f, gain, 0.0f, + 0.0f, 0.0f, 0.0f, gain + ); + for(i = 0;i < MAX_EFFECT_CHANNELS;i++) + ComputeBFormatGains(device->AmbiCoeffs, device->NumChannels, + matrix.m[i], slot->Gain, state->Gain[i]); /* Calculate coefficients for the each type of filter. Note that the shelf * filters' gain is for the reference frequency, which is the centerpoint @@ -103,59 +120,96 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice * */ gain = sqrtf(slot->EffectProps.Equalizer.LowGain); freq_mult = slot->EffectProps.Equalizer.LowCutoff/frequency; - ALfilterState_setParams(&state->filter[0], ALfilterType_LowShelf, + ALfilterState_setParams(&state->filter[0][0], ALfilterType_LowShelf, gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f) ); + /* Copy the filter coefficients for the other input channels. */ + for(i = 1;i < MAX_EFFECT_CHANNELS;i++) + { + state->filter[0][i].a1 = state->filter[0][0].a1; + state->filter[0][i].a2 = state->filter[0][0].a2; + state->filter[0][i].b1 = state->filter[0][0].b1; + state->filter[0][i].b2 = state->filter[0][0].b2; + state->filter[0][i].input_gain = state->filter[0][0].input_gain; + state->filter[0][i].process = state->filter[0][0].process; + } gain = slot->EffectProps.Equalizer.Mid1Gain; freq_mult = slot->EffectProps.Equalizer.Mid1Center/frequency; - ALfilterState_setParams(&state->filter[1], ALfilterType_Peaking, + ALfilterState_setParams(&state->filter[1][0], ALfilterType_Peaking, gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid1Width) ); + for(i = 1;i < MAX_EFFECT_CHANNELS;i++) + { + state->filter[1][i].a1 = state->filter[1][0].a1; + state->filter[1][i].a2 = state->filter[1][0].a2; + state->filter[1][i].b1 = state->filter[1][0].b1; + state->filter[1][i].b2 = state->filter[1][0].b2; + state->filter[1][i].input_gain = state->filter[1][0].input_gain; + state->filter[1][i].process = state->filter[1][0].process; + } gain = slot->EffectProps.Equalizer.Mid2Gain; freq_mult = slot->EffectProps.Equalizer.Mid2Center/frequency; - ALfilterState_setParams(&state->filter[2], ALfilterType_Peaking, + ALfilterState_setParams(&state->filter[2][0], ALfilterType_Peaking, gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid2Width) ); + for(i = 1;i < MAX_EFFECT_CHANNELS;i++) + { + state->filter[2][i].a1 = state->filter[2][0].a1; + state->filter[2][i].a2 = state->filter[2][0].a2; + state->filter[2][i].b1 = state->filter[2][0].b1; + state->filter[2][i].b2 = state->filter[2][0].b2; + state->filter[2][i].input_gain = state->filter[2][0].input_gain; + state->filter[2][i].process = state->filter[2][0].process; + } gain = sqrtf(slot->EffectProps.Equalizer.HighGain); freq_mult = slot->EffectProps.Equalizer.HighCutoff/frequency; - ALfilterState_setParams(&state->filter[3], ALfilterType_HighShelf, + ALfilterState_setParams(&state->filter[3][0], ALfilterType_HighShelf, gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f) ); + for(i = 1;i < MAX_EFFECT_CHANNELS;i++) + { + state->filter[3][i].a1 = state->filter[3][0].a1; + state->filter[3][i].a2 = state->filter[3][0].a2; + state->filter[3][i].b1 = state->filter[3][0].b1; + state->filter[3][i].b2 = state->filter[3][0].b2; + state->filter[3][i].input_gain = state->filter[3][0].input_gain; + state->filter[3][i].process = state->filter[3][0].process; + } } static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { + ALfloat (*Samples)[MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES] = state->SampleBuffer; + ALuint it, kt, ft; ALuint base; - ALuint it; - ALuint kt; - ALuint ft; for(base = 0;base < SamplesToDo;) { - ALfloat temps[256]; - ALuint td = minu(256, SamplesToDo-base); - - for(it = 0;it < td;it++) + ALuint td = minu(MAX_UPDATE_SAMPLES, SamplesToDo-base); + + for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) + ALfilterState_process(&state->filter[0][ft], Samples[0][ft], &SamplesIn[ft][base], td); + for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) + ALfilterState_process(&state->filter[1][ft], Samples[1][ft], Samples[0][ft], td); + for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) + ALfilterState_process(&state->filter[2][ft], Samples[2][ft], Samples[1][ft], td); + for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) + ALfilterState_process(&state->filter[3][ft], Samples[3][ft], Samples[2][ft], td); + + for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) { - ALfloat smp = SamplesIn[0][base+it]; - - for(ft = 0;ft < 4;ft++) - smp = ALfilterState_processSingle(&state->filter[ft], smp); - - temps[it] = smp; - } - - for(kt = 0;kt < NumChannels;kt++) - { - ALfloat gain = state->Gain[kt]; - if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) - continue; - - for(it = 0;it < td;it++) - SamplesOut[kt][base+it] += gain * temps[it]; + for(kt = 0;kt < NumChannels;kt++) + { + ALfloat gain = state->Gain[ft][kt]; + if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) + continue; + + for(it = 0;it < td;it++) + SamplesOut[kt][base+it] += gain * Samples[3][ft][it]; + } } base += td; @@ -174,7 +228,7 @@ typedef struct ALequalizerStateFactory { ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(factory)) { ALequalizerState *state; - int it; + int it, ft; state = ALequalizerState_New(sizeof(*state)); if(!state) return NULL; @@ -183,7 +237,10 @@ ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(fa /* Initialize sample history only on filter creation to avoid */ /* sound clicks if filter settings were changed in runtime. */ for(it = 0; it < 4; it++) - ALfilterState_clear(&state->filter[it]); + { + for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) + ALfilterState_clear(&state->filter[it][ft]); + } return STATIC_CAST(ALeffectState, state); } |