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Reorder some loops in the equalizer and use MixSamples

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This commit is contained in:
Chris Robinson
2018-01-11 05:03:00 -08:00
parent 9c33f4aea8
commit ff231b42ff
1 changed files with 42 additions and 60 deletions
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Alc/effects/equalizer.c
+42 -60
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@@ -72,19 +72,19 @@
* 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_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
struct {
/* Effect gains for each channel */
ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
/* Effect parameters */
ALfilterState filter[4][MAX_EFFECT_CHANNELS];
/* Effect parameters */
ALfilterState filter[4];
} Chans[MAX_EFFECT_CHANNELS];
ALfloat SampleBuffer[4][MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES];
ALfloat SampleBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE];
} ALequalizerState;
static ALvoid ALequalizerState_Destruct(ALequalizerState *state);
@@ -98,18 +98,8 @@ 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)
@@ -117,8 +107,17 @@ static ALvoid ALequalizerState_Destruct(ALequalizerState *state)
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state), ALCdevice *UNUSED(device))
static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *UNUSED(device))
{
ALsizei i, j;
for(i = 0; i < MAX_EFFECT_CHANNELS;i++)
{
for(j = 0;j < 4;j++)
ALfilterState_clear(&state->Chans[i].filter[j]);
for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
state->Chans[i].CurrentGains[j] = 0.0f;
}
return AL_TRUE;
}
@@ -133,7 +132,7 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext
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]);
slot->Params.Gain, state->Chans[i].TargetGains);
/* Calculate coefficients for the each type of filter. Note that the shelf
* filters' gain is for the reference frequency, which is the centerpoint
@@ -141,75 +140,58 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext
*/
gain = maxf(sqrtf(props->Equalizer.LowGain), 0.0625f); /* Limit -24dB */
freq_mult = props->Equalizer.LowCutoff/frequency;
ALfilterState_setParams(&state->filter[0][0], ALfilterType_LowShelf,
ALfilterState_setParams(&state->Chans[0].filter[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 = maxf(props->Equalizer.Mid1Gain, 0.0625f);
freq_mult = props->Equalizer.Mid1Center/frequency;
ALfilterState_setParams(&state->filter[1][0], ALfilterType_Peaking,
ALfilterState_setParams(&state->Chans[0].filter[1], 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 = maxf(props->Equalizer.Mid2Gain, 0.0625f);
freq_mult = props->Equalizer.Mid2Center/frequency;
ALfilterState_setParams(&state->filter[2][0], ALfilterType_Peaking,
ALfilterState_setParams(&state->Chans[0].filter[2], 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 = maxf(sqrtf(props->Equalizer.HighGain), 0.0625f);
freq_mult = props->Equalizer.HighCutoff/frequency;
ALfilterState_setParams(&state->filter[3][0], ALfilterType_HighShelf,
ALfilterState_setParams(&state->Chans[0].filter[3], ALfilterType_HighShelf,
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[3][i], &state->filter[3][0]);
{
ALfilterState_copyParams(&state->Chans[i].filter[0], &state->Chans[0].filter[0]);
ALfilterState_copyParams(&state->Chans[i].filter[1], &state->Chans[0].filter[1]);
ALfilterState_copyParams(&state->Chans[i].filter[2], &state->Chans[0].filter[2]);
ALfilterState_copyParams(&state->Chans[i].filter[3], &state->Chans[0].filter[3]);
}
}
static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
ALfloat (*Samples)[MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES] = state->SampleBuffer;
ALsizei it, kt, ft;
ALsizei base;
ALfloat (*restrict temps)[BUFFERSIZE] = state->SampleBuffer;
ALsizei c;
for(base = 0;base < SamplesToDo;)
for(c = 0;c < MAX_EFFECT_CHANNELS;c++)
{
ALsizei td = mini(MAX_UPDATE_SAMPLES, SamplesToDo-base);
ALfilterState_process(&state->Chans[c].filter[0], temps[0], SamplesIn[c], SamplesToDo);
ALfilterState_process(&state->Chans[c].filter[1], temps[1], temps[0], SamplesToDo);
ALfilterState_process(&state->Chans[c].filter[2], temps[2], temps[1], SamplesToDo);
ALfilterState_process(&state->Chans[c].filter[3], temps[3], temps[2], SamplesToDo);
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++)
{
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;
MixSamples(temps[3], NumChannels, SamplesOut,
state->Chans[c].CurrentGains, state->Chans[c].TargetGains,
SamplesToDo, 0, SamplesToDo
);
}
}