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Update OpenAL Soft to 1.18.2

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This commit is contained in:
Alex Szpakowski
2017-12-10 22:34:10 -04:00
parent 75e0077566
commit b160006eb1
152 changed files with 33572 additions and 15363 deletions
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libs/openal-soft/Alc/effects/equalizer.c
+93 -54
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@@ -71,18 +71,50 @@
* 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))
static ALvoid ALequalizerState_Destruct(ALequalizerState *state);
static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *device);
static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props);
static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALequalizerState)
DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState);
static void ALequalizerState_Construct(ALequalizerState *state)
{
int it, ft;
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
SET_VTABLE2(ALequalizerState, ALeffectState, state);
/* Initialize sample history only on filter creation to avoid */
/* sound clicks if filter settings were changed in runtime. */
for(it = 0; it < 4; it++)
{
for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
ALfilterState_clear(&state->filter[it][ft]);
}
}
static ALvoid ALequalizerState_Destruct(ALequalizerState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state), ALCdevice *UNUSED(device))
@@ -90,82 +122,96 @@ static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state),
return AL_TRUE;
}
static ALvoid ALequalizerState_update(ALequalizerState *state, ALCdevice *device, const ALeffectslot *slot)
static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *device, const ALeffectslot *slot, const ALeffectProps *props)
{
ALfloat frequency = (ALfloat)device->Frequency;
ALfloat gain, freq_mult;
ALuint i;
ComputeAmbientGains(device, slot->Gain, state->Gain);
STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
ComputeFirstOrderGains(device->FOAOut, IdentityMatrixf.m[i],
slot->Params.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
* of the transition band.
*/
gain = sqrtf(slot->EffectProps.Equalizer.LowGain);
freq_mult = slot->EffectProps.Equalizer.LowCutoff/frequency;
ALfilterState_setParams(&state->filter[0], ALfilterType_LowShelf,
gain = maxf(sqrtf(props->Equalizer.LowGain), 0.0625f); /* Limit -24dB */
freq_mult = props->Equalizer.LowCutoff/frequency;
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++)
ALfilterState_copyParams(&state->filter[0][i], &state->filter[0][0]);
gain = slot->EffectProps.Equalizer.Mid1Gain;
freq_mult = slot->EffectProps.Equalizer.Mid1Center/frequency;
ALfilterState_setParams(&state->filter[1], ALfilterType_Peaking,
gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid1Width)
gain = maxf(props->Equalizer.Mid1Gain, 0.0625f);
freq_mult = props->Equalizer.Mid1Center/frequency;
ALfilterState_setParams(&state->filter[1][0], ALfilterType_Peaking,
gain, freq_mult, calc_rcpQ_from_bandwidth(
freq_mult, props->Equalizer.Mid1Width
)
);
for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
ALfilterState_copyParams(&state->filter[1][i], &state->filter[1][0]);
gain = slot->EffectProps.Equalizer.Mid2Gain;
freq_mult = slot->EffectProps.Equalizer.Mid2Center/frequency;
ALfilterState_setParams(&state->filter[2], ALfilterType_Peaking,
gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid2Width)
gain = maxf(props->Equalizer.Mid2Gain, 0.0625f);
freq_mult = props->Equalizer.Mid2Center/frequency;
ALfilterState_setParams(&state->filter[2][0], ALfilterType_Peaking,
gain, freq_mult, calc_rcpQ_from_bandwidth(
freq_mult, props->Equalizer.Mid2Width
)
);
for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
ALfilterState_copyParams(&state->filter[2][i], &state->filter[2][0]);
gain = sqrtf(slot->EffectProps.Equalizer.HighGain);
freq_mult = slot->EffectProps.Equalizer.HighCutoff/frequency;
ALfilterState_setParams(&state->filter[3], ALfilterType_HighShelf,
gain = maxf(sqrtf(props->Equalizer.HighGain), 0.0625f);
freq_mult = props->Equalizer.HighCutoff/frequency;
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++)
ALfilterState_copyParams(&state->filter[3][i], &state->filter[3][0]);
}
static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
ALuint base;
ALuint it;
ALuint kt;
ALuint ft;
ALfloat (*Samples)[MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES] = state->SampleBuffer;
ALsizei it, kt, ft;
ALsizei base;
for(base = 0;base < SamplesToDo;)
{
ALfloat temps[256];
ALuint td = minu(256, SamplesToDo-base);
ALsizei td = mini(MAX_UPDATE_SAMPLES, 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[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++)
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(it = 0;it < td;it++)
SamplesOut[kt][base+it] += gain * Samples[3][ft][it];
}
}
base += td;
}
}
DECLARE_DEFAULT_ALLOCATORS(ALequalizerState)
DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState);
typedef struct ALequalizerStateFactory {
DERIVE_FROM_TYPE(ALeffectStateFactory);
@@ -174,16 +220,9 @@ typedef struct ALequalizerStateFactory {
ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(factory))
{
ALequalizerState *state;
int it;
state = ALequalizerState_New(sizeof(*state));
NEW_OBJ0(state, ALequalizerState)();
if(!state) return NULL;
SET_VTABLE2(ALequalizerState, ALeffectState, state);
/* 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]);
return STATIC_CAST(ALeffectState, state);
}