Convert the equalizer effect to multichannel
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Chris Robinson
parent
210b4c1fcd
commit
3ac786c6af
@ -71,14 +71,20 @@
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* filter coefficients" by Robert Bristow-Johnson *
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* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */
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/* The maximum number of sample frames per update. */
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#define MAX_UPDATE_SAMPLES 256
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typedef struct ALequalizerState {
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DERIVE_FROM_TYPE(ALeffectState);
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/* Effect gains for each channel */
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ALfloat Gain[MAX_OUTPUT_CHANNELS];
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ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
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/* Effect parameters */
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ALfilterState filter[4];
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ALfilterState filter[4][MAX_EFFECT_CHANNELS];
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ALfloat SampleBuffer[4][MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES];
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} ALequalizerState;
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static ALvoid ALequalizerState_Destruct(ALequalizerState *UNUSED(state))
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@ -94,8 +100,19 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *
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{
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ALfloat frequency = (ALfloat)device->Frequency;
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ALfloat gain, freq_mult;
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aluMatrixf matrix;
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ALuint i;
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ComputeAmbientGains(device->AmbiCoeffs, device->NumChannels, slot->Gain, state->Gain);
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gain = device->AmbiScale;
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aluMatrixfSet(&matrix,
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1.0f, 0.0f, 0.0f, 0.0f,
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0.0f, gain, 0.0f, 0.0f,
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0.0f, 0.0f, gain, 0.0f,
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0.0f, 0.0f, 0.0f, gain
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);
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for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
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ComputeBFormatGains(device->AmbiCoeffs, device->NumChannels,
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matrix.m[i], slot->Gain, state->Gain[i]);
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/* Calculate coefficients for the each type of filter. Note that the shelf
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* filters' gain is for the reference frequency, which is the centerpoint
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@ -103,59 +120,96 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *
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*/
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gain = sqrtf(slot->EffectProps.Equalizer.LowGain);
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freq_mult = slot->EffectProps.Equalizer.LowCutoff/frequency;
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ALfilterState_setParams(&state->filter[0], ALfilterType_LowShelf,
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ALfilterState_setParams(&state->filter[0][0], ALfilterType_LowShelf,
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gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
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);
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/* Copy the filter coefficients for the other input channels. */
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for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
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{
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state->filter[0][i].a1 = state->filter[0][0].a1;
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state->filter[0][i].a2 = state->filter[0][0].a2;
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state->filter[0][i].b1 = state->filter[0][0].b1;
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state->filter[0][i].b2 = state->filter[0][0].b2;
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state->filter[0][i].input_gain = state->filter[0][0].input_gain;
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state->filter[0][i].process = state->filter[0][0].process;
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}
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gain = slot->EffectProps.Equalizer.Mid1Gain;
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freq_mult = slot->EffectProps.Equalizer.Mid1Center/frequency;
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ALfilterState_setParams(&state->filter[1], ALfilterType_Peaking,
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ALfilterState_setParams(&state->filter[1][0], ALfilterType_Peaking,
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gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid1Width)
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);
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for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
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{
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state->filter[1][i].a1 = state->filter[1][0].a1;
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state->filter[1][i].a2 = state->filter[1][0].a2;
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state->filter[1][i].b1 = state->filter[1][0].b1;
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state->filter[1][i].b2 = state->filter[1][0].b2;
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state->filter[1][i].input_gain = state->filter[1][0].input_gain;
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state->filter[1][i].process = state->filter[1][0].process;
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}
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gain = slot->EffectProps.Equalizer.Mid2Gain;
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freq_mult = slot->EffectProps.Equalizer.Mid2Center/frequency;
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ALfilterState_setParams(&state->filter[2], ALfilterType_Peaking,
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ALfilterState_setParams(&state->filter[2][0], ALfilterType_Peaking,
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gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid2Width)
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);
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for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
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{
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state->filter[2][i].a1 = state->filter[2][0].a1;
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state->filter[2][i].a2 = state->filter[2][0].a2;
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state->filter[2][i].b1 = state->filter[2][0].b1;
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state->filter[2][i].b2 = state->filter[2][0].b2;
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state->filter[2][i].input_gain = state->filter[2][0].input_gain;
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state->filter[2][i].process = state->filter[2][0].process;
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}
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gain = sqrtf(slot->EffectProps.Equalizer.HighGain);
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freq_mult = slot->EffectProps.Equalizer.HighCutoff/frequency;
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ALfilterState_setParams(&state->filter[3], ALfilterType_HighShelf,
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ALfilterState_setParams(&state->filter[3][0], ALfilterType_HighShelf,
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gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
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);
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for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
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{
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state->filter[3][i].a1 = state->filter[3][0].a1;
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state->filter[3][i].a2 = state->filter[3][0].a2;
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state->filter[3][i].b1 = state->filter[3][0].b1;
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state->filter[3][i].b2 = state->filter[3][0].b2;
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state->filter[3][i].input_gain = state->filter[3][0].input_gain;
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state->filter[3][i].process = state->filter[3][0].process;
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}
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}
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static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
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{
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ALfloat (*Samples)[MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES] = state->SampleBuffer;
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ALuint it, kt, ft;
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ALuint base;
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ALuint it;
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ALuint kt;
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ALuint ft;
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for(base = 0;base < SamplesToDo;)
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{
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ALfloat temps[256];
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ALuint td = minu(256, SamplesToDo-base);
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ALuint td = minu(MAX_UPDATE_SAMPLES, SamplesToDo-base);
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for(it = 0;it < td;it++)
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_process(&state->filter[0][ft], Samples[0][ft], &SamplesIn[ft][base], td);
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_process(&state->filter[1][ft], Samples[1][ft], Samples[0][ft], td);
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_process(&state->filter[2][ft], Samples[2][ft], Samples[1][ft], td);
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_process(&state->filter[3][ft], Samples[3][ft], Samples[2][ft], td);
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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{
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ALfloat smp = SamplesIn[0][base+it];
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for(kt = 0;kt < NumChannels;kt++)
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{
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ALfloat gain = state->Gain[ft][kt];
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if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
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continue;
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for(ft = 0;ft < 4;ft++)
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smp = ALfilterState_processSingle(&state->filter[ft], smp);
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temps[it] = smp;
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}
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for(kt = 0;kt < NumChannels;kt++)
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{
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ALfloat gain = state->Gain[kt];
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if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
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continue;
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for(it = 0;it < td;it++)
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SamplesOut[kt][base+it] += gain * temps[it];
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for(it = 0;it < td;it++)
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SamplesOut[kt][base+it] += gain * Samples[3][ft][it];
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}
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}
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base += td;
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@ -174,7 +228,7 @@ typedef struct ALequalizerStateFactory {
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ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(factory))
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{
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ALequalizerState *state;
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int it;
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int it, ft;
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state = ALequalizerState_New(sizeof(*state));
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if(!state) return NULL;
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@ -183,7 +237,10 @@ ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(fa
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/* Initialize sample history only on filter creation to avoid */
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/* sound clicks if filter settings were changed in runtime. */
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for(it = 0; it < 4; it++)
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ALfilterState_clear(&state->filter[it]);
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{
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_clear(&state->filter[it][ft]);
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}
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return STATIC_CAST(ALeffectState, state);
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}
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