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Convert the equalizer effect to multichannel

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This commit is contained in:
Chris Robinson 2016-01-29 07:26:19 -08:00
parent 210b4c1fcd
commit 3ac786c6af
1 changed files with 87 additions and 30 deletions
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117
Alc/effects/equalizer.c
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@ -71,14 +71,20 @@
* filter coefficients" by Robert Bristow-Johnson * * filter coefficients" by Robert Bristow-Johnson *
* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */ * 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 { typedef struct ALequalizerState {
DERIVE_FROM_TYPE(ALeffectState); DERIVE_FROM_TYPE(ALeffectState);
/* Effect gains for each channel */ /* Effect gains for each channel */
ALfloat Gain[MAX_OUTPUT_CHANNELS]; ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
/* Effect parameters */ /* Effect parameters */
ALfilterState filter[4]; ALfilterState filter[4][MAX_EFFECT_CHANNELS];
ALfloat SampleBuffer[4][MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES];
} ALequalizerState; } ALequalizerState;
static ALvoid ALequalizerState_Destruct(ALequalizerState *UNUSED(state)) 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 frequency = (ALfloat)device->Frequency;
ALfloat gain, freq_mult; ALfloat gain, freq_mult;
aluMatrixf matrix;
ALuint i;
ComputeAmbientGains(device->AmbiCoeffs, device->NumChannels, slot->Gain, state->Gain); 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 /* Calculate coefficients for the each type of filter. Note that the shelf
* filters' gain is for the reference frequency, which is the centerpoint * 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); gain = sqrtf(slot->EffectProps.Equalizer.LowGain);
freq_mult = slot->EffectProps.Equalizer.LowCutoff/frequency; 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) 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; gain = slot->EffectProps.Equalizer.Mid1Gain;
freq_mult = slot->EffectProps.Equalizer.Mid1Center/frequency; 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) 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; gain = slot->EffectProps.Equalizer.Mid2Gain;
freq_mult = slot->EffectProps.Equalizer.Mid2Center/frequency; 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) 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); gain = sqrtf(slot->EffectProps.Equalizer.HighGain);
freq_mult = slot->EffectProps.Equalizer.HighCutoff/frequency; 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) 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) 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 base;
ALuint it;
ALuint kt;
ALuint ft;
for(base = 0;base < SamplesToDo;) for(base = 0;base < SamplesToDo;)
{ {
ALfloat temps[256]; ALuint td = minu(MAX_UPDATE_SAMPLES, SamplesToDo-base);
ALuint td = minu(256, SamplesToDo-base);
for(it = 0;it < td;it++) 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(kt = 0;kt < NumChannels;kt++)
{
ALfloat gain = state->Gain[ft][kt];
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(ft = 0;ft < 4;ft++) for(it = 0;it < td;it++)
smp = ALfilterState_processSingle(&state->filter[ft], smp); SamplesOut[kt][base+it] += gain * Samples[3][ft][it];
}
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];
} }
base += td; base += td;
@ -174,7 +228,7 @@ typedef struct ALequalizerStateFactory {
ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(factory)) ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(factory))
{ {
ALequalizerState *state; ALequalizerState *state;
int it; int it, ft;
state = ALequalizerState_New(sizeof(*state)); state = ALequalizerState_New(sizeof(*state));
if(!state) return NULL; if(!state) return NULL;
@ -183,7 +237,10 @@ ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(fa
/* Initialize sample history only on filter creation to avoid */ /* Initialize sample history only on filter creation to avoid */
/* sound clicks if filter settings were changed in runtime. */ /* sound clicks if filter settings were changed in runtime. */
for(it = 0; it < 4; it++) 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); return STATIC_CAST(ALeffectState, state);
} }