diff options
author | Chris Robinson <[email protected]> | 2018-11-19 21:04:50 -0800 |
---|---|---|
committer | Chris Robinson <[email protected]> | 2018-11-19 21:04:50 -0800 |
commit | 6ac84c7a5f6d267522bdc872802c8940dcd2adec (patch) | |
tree | 3ba7b255bfaf6f6f05f76d5a7c62ffd26d0a5ae6 /Alc | |
parent | 67da3850cbf43dd6c40b41e905b446bec975663c (diff) |
Clean up the remaining effect struct member names
Diffstat (limited to 'Alc')
-rw-r--r-- | Alc/effects/equalizer.cpp | 57 | ||||
-rw-r--r-- | Alc/effects/fshifter.cpp | 108 | ||||
-rw-r--r-- | Alc/effects/modulator.cpp | 68 | ||||
-rw-r--r-- | Alc/effects/pshifter.cpp | 138 |
4 files changed, 184 insertions, 187 deletions
diff --git a/Alc/effects/equalizer.cpp b/Alc/effects/equalizer.cpp index ad101a7e..814b43a4 100644 --- a/Alc/effects/equalizer.cpp +++ b/Alc/effects/equalizer.cpp @@ -23,6 +23,8 @@ #include <math.h> #include <stdlib.h> +#include <algorithm> + #include "alMain.h" #include "alcontext.h" #include "alAuxEffectSlot.h" @@ -80,11 +82,11 @@ struct ALequalizerState final : public ALeffectState { BiquadFilter filter[4]; /* Effect gains for each channel */ - ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; - ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; - } Chans[MAX_EFFECT_CHANNELS]; + ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]{}; + ALfloat TargetGains[MAX_OUTPUT_CHANNELS]{}; + } mChans[MAX_EFFECT_CHANNELS]; - ALfloat SampleBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE]; + ALfloat mSampleBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE]{}; }; static ALvoid ALequalizerState_Destruct(ALequalizerState *state); @@ -111,14 +113,13 @@ static ALvoid ALequalizerState_Destruct(ALequalizerState *state) static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *state, ALCdevice *UNUSED(device)) { - ALsizei i, j; - - for(i = 0; i < MAX_EFFECT_CHANNELS;i++) + for(auto &e : state->mChans) { - for(j = 0;j < 4;j++) - BiquadFilter_clear(&state->Chans[i].filter[j]); - for(j = 0;j < MAX_OUTPUT_CHANNELS;j++) - state->Chans[i].CurrentGains[j] = 0.0f; + std::for_each(std::begin(e.filter), std::end(e.filter), + [](BiquadFilter &f) -> void + { BiquadFilter_clear(&f); } + ); + std::fill(std::begin(e.CurrentGains), std::end(e.CurrentGains), 0.0f); } return AL_TRUE; } @@ -136,13 +137,13 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext */ gain = maxf(sqrtf(props->Equalizer.LowGain), 0.0625f); /* Limit -24dB */ f0norm = props->Equalizer.LowCutoff/frequency; - BiquadFilter_setParams(&state->Chans[0].filter[0], BiquadType::LowShelf, + BiquadFilter_setParams(&state->mChans[0].filter[0], BiquadType::LowShelf, gain, f0norm, calc_rcpQ_from_slope(gain, 0.75f) ); gain = maxf(props->Equalizer.Mid1Gain, 0.0625f); f0norm = props->Equalizer.Mid1Center/frequency; - BiquadFilter_setParams(&state->Chans[0].filter[1], BiquadType::Peaking, + BiquadFilter_setParams(&state->mChans[0].filter[1], BiquadType::Peaking, gain, f0norm, calc_rcpQ_from_bandwidth( f0norm, props->Equalizer.Mid1Width ) @@ -150,7 +151,7 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext gain = maxf(props->Equalizer.Mid2Gain, 0.0625f); f0norm = props->Equalizer.Mid2Center/frequency; - BiquadFilter_setParams(&state->Chans[0].filter[2], BiquadType::Peaking, + BiquadFilter_setParams(&state->mChans[0].filter[2], BiquadType::Peaking, gain, f0norm, calc_rcpQ_from_bandwidth( f0norm, props->Equalizer.Mid2Width ) @@ -158,40 +159,40 @@ static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCcontext gain = maxf(sqrtf(props->Equalizer.HighGain), 0.0625f); f0norm = props->Equalizer.HighCutoff/frequency; - BiquadFilter_setParams(&state->Chans[0].filter[3], BiquadType::HighShelf, + BiquadFilter_setParams(&state->mChans[0].filter[3], BiquadType::HighShelf, gain, f0norm, calc_rcpQ_from_slope(gain, 0.75f) ); /* Copy the filter coefficients for the other input channels. */ for(i = 1;i < MAX_EFFECT_CHANNELS;i++) { - BiquadFilter_copyParams(&state->Chans[i].filter[0], &state->Chans[0].filter[0]); - BiquadFilter_copyParams(&state->Chans[i].filter[1], &state->Chans[0].filter[1]); - BiquadFilter_copyParams(&state->Chans[i].filter[2], &state->Chans[0].filter[2]); - BiquadFilter_copyParams(&state->Chans[i].filter[3], &state->Chans[0].filter[3]); + BiquadFilter_copyParams(&state->mChans[i].filter[0], &state->mChans[0].filter[0]); + BiquadFilter_copyParams(&state->mChans[i].filter[1], &state->mChans[0].filter[1]); + BiquadFilter_copyParams(&state->mChans[i].filter[2], &state->mChans[0].filter[2]); + BiquadFilter_copyParams(&state->mChans[i].filter[3], &state->mChans[0].filter[3]); } - STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer; - STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels; + state->OutBuffer = device->FOAOut.Buffer; + state->OutChannels = device->FOAOut.NumChannels; for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputePanGains(&device->FOAOut, aluMatrixf::Identity.m[i], slot->Params.Gain, - state->Chans[i].TargetGains); + state->mChans[i].TargetGains); } static ALvoid ALequalizerState_process(ALequalizerState *state, ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { - ALfloat (*RESTRICT temps)[BUFFERSIZE] = state->SampleBuffer; + ALfloat (*RESTRICT temps)[BUFFERSIZE] = state->mSampleBuffer; ALsizei c; for(c = 0;c < MAX_EFFECT_CHANNELS;c++) { - BiquadFilter_process(&state->Chans[c].filter[0], temps[0], SamplesIn[c], SamplesToDo); - BiquadFilter_process(&state->Chans[c].filter[1], temps[1], temps[0], SamplesToDo); - BiquadFilter_process(&state->Chans[c].filter[2], temps[2], temps[1], SamplesToDo); - BiquadFilter_process(&state->Chans[c].filter[3], temps[3], temps[2], SamplesToDo); + BiquadFilter_process(&state->mChans[c].filter[0], temps[0], SamplesIn[c], SamplesToDo); + BiquadFilter_process(&state->mChans[c].filter[1], temps[1], temps[0], SamplesToDo); + BiquadFilter_process(&state->mChans[c].filter[2], temps[2], temps[1], SamplesToDo); + BiquadFilter_process(&state->mChans[c].filter[3], temps[3], temps[2], SamplesToDo); MixSamples(temps[3], NumChannels, SamplesOut, - state->Chans[c].CurrentGains, state->Chans[c].TargetGains, + state->mChans[c].CurrentGains, state->mChans[c].TargetGains, SamplesToDo, 0, SamplesToDo ); } diff --git a/Alc/effects/fshifter.cpp b/Alc/effects/fshifter.cpp index 7775fafb..b4d073da 100644 --- a/Alc/effects/fshifter.cpp +++ b/Alc/effects/fshifter.cpp @@ -62,23 +62,23 @@ alignas(16) const std::array<ALdouble,HIL_SIZE> HannWindow = InitHannWindow(); struct ALfshifterState final : public ALeffectState { /* Effect parameters */ - ALsizei count; - ALsizei PhaseStep; - ALsizei Phase; - ALdouble ld_sign; + ALsizei mCount{}; + ALsizei mPhaseStep{}; + ALsizei mPhase{}; + ALdouble mLdSign{}; /*Effects buffers*/ - ALfloat InFIFO[HIL_SIZE]; - complex_d OutFIFO[HIL_SIZE]; - complex_d OutputAccum[HIL_SIZE]; - complex_d Analytic[HIL_SIZE]; - complex_d Outdata[BUFFERSIZE]; + ALfloat mInFIFO[HIL_SIZE]{}; + complex_d mOutFIFO[HIL_SIZE]{}; + complex_d mOutputAccum[HIL_SIZE]{}; + complex_d mAnalytic[HIL_SIZE]{}; + complex_d mOutdata[BUFFERSIZE]{}; - alignas(16) ALfloat BufferOut[BUFFERSIZE]; + alignas(16) ALfloat mBufferOut[BUFFERSIZE]{}; /* Effect gains for each output channel */ - ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; - ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; + ALfloat mCurrentGains[MAX_OUTPUT_CHANNELS]{}; + ALfloat mTargetGains[MAX_OUTPUT_CHANNELS]{}; }; ALvoid ALfshifterState_Destruct(ALfshifterState *state); @@ -105,113 +105,111 @@ ALvoid ALfshifterState_Destruct(ALfshifterState *state) ALboolean ALfshifterState_deviceUpdate(ALfshifterState *state, ALCdevice *UNUSED(device)) { /* (Re-)initializing parameters and clear the buffers. */ - state->count = FIFO_LATENCY; - state->PhaseStep = 0; - state->Phase = 0; - state->ld_sign = 1.0; + state->mCount = FIFO_LATENCY; + state->mPhaseStep = 0; + state->mPhase = 0; + state->mLdSign = 1.0; - std::fill(std::begin(state->InFIFO), std::end(state->InFIFO), 0.0f); - std::fill(std::begin(state->OutFIFO), std::end(state->OutFIFO), complex_d{}); - std::fill(std::begin(state->OutputAccum), std::end(state->OutputAccum), complex_d{}); - std::fill(std::begin(state->Analytic), std::end(state->Analytic), complex_d{}); + std::fill(std::begin(state->mInFIFO), std::end(state->mInFIFO), 0.0f); + std::fill(std::begin(state->mOutFIFO), std::end(state->mOutFIFO), complex_d{}); + std::fill(std::begin(state->mOutputAccum), std::end(state->mOutputAccum), complex_d{}); + std::fill(std::begin(state->mAnalytic), std::end(state->mAnalytic), complex_d{}); - std::fill(std::begin(state->CurrentGains), std::end(state->CurrentGains), 0.0f); - std::fill(std::begin(state->TargetGains), std::end(state->TargetGains), 0.0f); + std::fill(std::begin(state->mCurrentGains), std::end(state->mCurrentGains), 0.0f); + std::fill(std::begin(state->mTargetGains), std::end(state->mTargetGains), 0.0f); return AL_TRUE; } ALvoid ALfshifterState_update(ALfshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { - const ALCdevice *device = context->Device; - ALfloat coeffs[MAX_AMBI_COEFFS]; - ALfloat step; + const ALCdevice *device{context->Device}; - step = props->Fshifter.Frequency / (ALfloat)device->Frequency; - state->PhaseStep = fastf2i(minf(step, 0.5f) * FRACTIONONE); + ALfloat step{props->Fshifter.Frequency / (ALfloat)device->Frequency}; + state->mPhaseStep = fastf2i(minf(step, 0.5f) * FRACTIONONE); switch(props->Fshifter.LeftDirection) { case AL_FREQUENCY_SHIFTER_DIRECTION_DOWN: - state->ld_sign = -1.0; + state->mLdSign = -1.0; break; case AL_FREQUENCY_SHIFTER_DIRECTION_UP: - state->ld_sign = 1.0; + state->mLdSign = 1.0; break; case AL_FREQUENCY_SHIFTER_DIRECTION_OFF: - state->Phase = 0; - state->PhaseStep = 0; + state->mPhase = 0; + state->mPhaseStep = 0; break; } + ALfloat coeffs[MAX_AMBI_COEFFS]; CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs); - ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->TargetGains); + ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->mTargetGains); } ALvoid ALfshifterState_process(ALfshifterState *state, ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { static const complex_d complex_zero{0.0, 0.0}; - ALfloat *RESTRICT BufferOut = state->BufferOut; + ALfloat *RESTRICT BufferOut = state->mBufferOut; ALsizei j, k, base; for(base = 0;base < SamplesToDo;) { - ALsizei todo = mini(HIL_SIZE-state->count, SamplesToDo-base); + ALsizei todo = mini(HIL_SIZE-state->mCount, SamplesToDo-base); ASSUME(todo > 0); /* Fill FIFO buffer with samples data */ - k = state->count; + k = state->mCount; for(j = 0;j < todo;j++,k++) { - state->InFIFO[k] = SamplesIn[0][base+j]; - state->Outdata[base+j] = state->OutFIFO[k-FIFO_LATENCY]; + state->mInFIFO[k] = SamplesIn[0][base+j]; + state->mOutdata[base+j] = state->mOutFIFO[k-FIFO_LATENCY]; } - state->count += todo; + state->mCount += todo; base += todo; /* Check whether FIFO buffer is filled */ - if(state->count < HIL_SIZE) continue; - - state->count = FIFO_LATENCY; + if(state->mCount < HIL_SIZE) continue; + state->mCount = FIFO_LATENCY; /* Real signal windowing and store in Analytic buffer */ for(k = 0;k < HIL_SIZE;k++) { - state->Analytic[k].real(state->InFIFO[k] * HannWindow[k]); - state->Analytic[k].imag(0.0); + state->mAnalytic[k].real(state->mInFIFO[k] * HannWindow[k]); + state->mAnalytic[k].imag(0.0); } /* Processing signal by Discrete Hilbert Transform (analytical signal). */ - complex_hilbert(state->Analytic, HIL_SIZE); + complex_hilbert(state->mAnalytic, HIL_SIZE); /* Windowing and add to output accumulator */ for(k = 0;k < HIL_SIZE;k++) - state->OutputAccum[k] += 2.0/OVERSAMP*HannWindow[k]*state->Analytic[k]; + state->mOutputAccum[k] += 2.0/OVERSAMP*HannWindow[k]*state->mAnalytic[k]; /* Shift accumulator, input & output FIFO */ - for(k = 0;k < HIL_STEP;k++) state->OutFIFO[k] = state->OutputAccum[k]; - for(j = 0;k < HIL_SIZE;k++,j++) state->OutputAccum[j] = state->OutputAccum[k]; - for(;j < HIL_SIZE;j++) state->OutputAccum[j] = complex_zero; + for(k = 0;k < HIL_STEP;k++) state->mOutFIFO[k] = state->mOutputAccum[k]; + for(j = 0;k < HIL_SIZE;k++,j++) state->mOutputAccum[j] = state->mOutputAccum[k]; + for(;j < HIL_SIZE;j++) state->mOutputAccum[j] = complex_zero; for(k = 0;k < FIFO_LATENCY;k++) - state->InFIFO[k] = state->InFIFO[k+HIL_STEP]; + state->mInFIFO[k] = state->mInFIFO[k+HIL_STEP]; } /* Process frequency shifter using the analytic signal obtained. */ for(k = 0;k < SamplesToDo;k++) { - double phase = state->Phase * ((1.0/FRACTIONONE) * 2.0*M_PI); - BufferOut[k] = (float)(state->Outdata[k].real()*std::cos(phase) + - state->Outdata[k].imag()*std::sin(phase)*state->ld_sign); + double phase = state->mPhase * ((1.0/FRACTIONONE) * 2.0*M_PI); + BufferOut[k] = (float)(state->mOutdata[k].real()*std::cos(phase) + + state->mOutdata[k].imag()*std::sin(phase)*state->mLdSign); - state->Phase += state->PhaseStep; - state->Phase &= FRACTIONMASK; + state->mPhase += state->mPhaseStep; + state->mPhase &= FRACTIONMASK; } /* Now, mix the processed sound data to the output. */ - MixSamples(BufferOut, NumChannels, SamplesOut, state->CurrentGains, state->TargetGains, + MixSamples(BufferOut, NumChannels, SamplesOut, state->mCurrentGains, state->mTargetGains, maxi(SamplesToDo, 512), 0, SamplesToDo); } diff --git a/Alc/effects/modulator.cpp b/Alc/effects/modulator.cpp index b7ee97d9..0062a779 100644 --- a/Alc/effects/modulator.cpp +++ b/Alc/effects/modulator.cpp @@ -23,6 +23,9 @@ #include <math.h> #include <stdlib.h> +#include <cmath> +#include <algorithm> + #include "alMain.h" #include "alcontext.h" #include "alAuxEffectSlot.h" @@ -35,17 +38,17 @@ #define MAX_UPDATE_SAMPLES 128 struct ALmodulatorState final : public ALeffectState { - void (*GetSamples)(ALfloat*RESTRICT, ALsizei, const ALsizei, ALsizei); + void (*mGetSamples)(ALfloat*RESTRICT, ALsizei, const ALsizei, ALsizei){}; - ALsizei index; - ALsizei step; + ALsizei mIndex{0}; + ALsizei mStep{1}; struct { BiquadFilter Filter; - ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; - ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; - } Chans[MAX_EFFECT_CHANNELS]; + ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]{}; + ALfloat TargetGains[MAX_OUTPUT_CHANNELS]{}; + } mChans[MAX_EFFECT_CHANNELS]; }; static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state); @@ -63,7 +66,7 @@ DEFINE_ALEFFECTSTATE_VTABLE(ALmodulatorState); static inline ALfloat Sin(ALsizei index) { - return sinf((ALfloat)index * (F_TAU / WAVEFORM_FRACONE)); + return std::sin((ALfloat)index * (F_TAU / (ALfloat)WAVEFORM_FRACONE)); } static inline ALfloat Saw(ALsizei index) @@ -107,9 +110,6 @@ static void ALmodulatorState_Construct(ALmodulatorState *state) new (state) ALmodulatorState{}; ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALmodulatorState, ALeffectState, state); - - state->index = 0; - state->step = 1; } static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state) @@ -120,12 +120,10 @@ static ALvoid ALmodulatorState_Destruct(ALmodulatorState *state) static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *state, ALCdevice *UNUSED(device)) { - ALsizei i, j; - for(i = 0;i < MAX_EFFECT_CHANNELS;i++) + for(auto &e : state->mChans) { - BiquadFilter_clear(&state->Chans[i].Filter); - for(j = 0;j < MAX_OUTPUT_CHANNELS;j++) - state->Chans[i].CurrentGains[j] = 0.0f; + BiquadFilter_clear(&e.Filter); + std::fill(std::begin(e.CurrentGains), std::end(e.CurrentGains), 0.0f); } return AL_TRUE; } @@ -136,37 +134,37 @@ static ALvoid ALmodulatorState_update(ALmodulatorState *state, const ALCcontext ALfloat f0norm; ALsizei i; - state->step = fastf2i(props->Modulator.Frequency / (ALfloat)device->Frequency * - WAVEFORM_FRACONE); - state->step = clampi(state->step, 0, WAVEFORM_FRACONE-1); + state->mStep = fastf2i(props->Modulator.Frequency / (ALfloat)device->Frequency * + WAVEFORM_FRACONE); + state->mStep = clampi(state->mStep, 0, WAVEFORM_FRACONE-1); - if(state->step == 0) - state->GetSamples = ModulateOne; + if(state->mStep == 0) + state->mGetSamples = ModulateOne; else if(props->Modulator.Waveform == AL_RING_MODULATOR_SINUSOID) - state->GetSamples = ModulateSin; + state->mGetSamples = ModulateSin; else if(props->Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH) - state->GetSamples = ModulateSaw; + state->mGetSamples = ModulateSaw; else /*if(Slot->Params.EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)*/ - state->GetSamples = ModulateSquare; + state->mGetSamples = ModulateSquare; f0norm = props->Modulator.HighPassCutoff / (ALfloat)device->Frequency; f0norm = clampf(f0norm, 1.0f/512.0f, 0.49f); /* Bandwidth value is constant in octaves. */ - BiquadFilter_setParams(&state->Chans[0].Filter, BiquadType::HighPass, 1.0f, + BiquadFilter_setParams(&state->mChans[0].Filter, BiquadType::HighPass, 1.0f, f0norm, calc_rcpQ_from_bandwidth(f0norm, 0.75f)); for(i = 1;i < MAX_EFFECT_CHANNELS;i++) - BiquadFilter_copyParams(&state->Chans[i].Filter, &state->Chans[0].Filter); + BiquadFilter_copyParams(&state->mChans[i].Filter, &state->mChans[0].Filter); - STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer; - STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels; + state->OutBuffer = device->FOAOut.Buffer; + state->OutChannels = device->FOAOut.NumChannels; for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputePanGains(&device->FOAOut, aluMatrixf::Identity.m[i], slot->Params.Gain, - state->Chans[i].TargetGains); + state->mChans[i].TargetGains); } static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { - const ALsizei step = state->step; + const ALsizei step = state->mStep; ALsizei base; for(base = 0;base < SamplesToDo;) @@ -175,20 +173,20 @@ static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALsizei SamplesT ALsizei td = mini(MAX_UPDATE_SAMPLES, SamplesToDo-base); ALsizei c, i; - state->GetSamples(modsamples, state->index, step, td); - state->index += (step*td) & WAVEFORM_FRACMASK; - state->index &= WAVEFORM_FRACMASK; + state->mGetSamples(modsamples, state->mIndex, step, td); + state->mIndex += (step*td) & WAVEFORM_FRACMASK; + state->mIndex &= WAVEFORM_FRACMASK; for(c = 0;c < MAX_EFFECT_CHANNELS;c++) { alignas(16) ALfloat temps[MAX_UPDATE_SAMPLES]; - BiquadFilter_process(&state->Chans[c].Filter, temps, &SamplesIn[c][base], td); + BiquadFilter_process(&state->mChans[c].Filter, temps, &SamplesIn[c][base], td); for(i = 0;i < td;i++) temps[i] *= modsamples[i]; - MixSamples(temps, NumChannels, SamplesOut, state->Chans[c].CurrentGains, - state->Chans[c].TargetGains, SamplesToDo-base, base, td); + MixSamples(temps, NumChannels, SamplesOut, state->mChans[c].CurrentGains, + state->mChans[c].TargetGains, SamplesToDo-base, base, td); } base += td; diff --git a/Alc/effects/pshifter.cpp b/Alc/effects/pshifter.cpp index 217b021e..ec473678 100644 --- a/Alc/effects/pshifter.cpp +++ b/Alc/effects/pshifter.cpp @@ -119,28 +119,28 @@ inline complex_d polar2rect(const ALphasor &number) struct ALpshifterState final : public ALeffectState { /* Effect parameters */ - ALsizei count; - ALsizei PitchShiftI; - ALfloat PitchShift; - ALfloat FreqPerBin; + ALsizei mCount; + ALsizei mPitchShiftI; + ALfloat mPitchShift; + ALfloat mFreqPerBin; - /*Effects buffers*/ - ALfloat InFIFO[STFT_SIZE]; - ALfloat OutFIFO[STFT_STEP]; - ALdouble LastPhase[STFT_HALF_SIZE+1]; - ALdouble SumPhase[STFT_HALF_SIZE+1]; - ALdouble OutputAccum[STFT_SIZE]; + /* Effects buffers */ + ALfloat mInFIFO[STFT_SIZE]; + ALfloat mOutFIFO[STFT_STEP]; + ALdouble mLastPhase[STFT_HALF_SIZE+1]; + ALdouble mSumPhase[STFT_HALF_SIZE+1]; + ALdouble mOutputAccum[STFT_SIZE]; - complex_d FFTbuffer[STFT_SIZE]; + complex_d mFFTbuffer[STFT_SIZE]; - ALfrequencyDomain Analysis_buffer[STFT_HALF_SIZE+1]; - ALfrequencyDomain Syntesis_buffer[STFT_HALF_SIZE+1]; + ALfrequencyDomain mAnalysis_buffer[STFT_HALF_SIZE+1]; + ALfrequencyDomain mSyntesis_buffer[STFT_HALF_SIZE+1]; - alignas(16) ALfloat BufferOut[BUFFERSIZE]; + alignas(16) ALfloat mBufferOut[BUFFERSIZE]; /* Effect gains for each output channel */ - ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]; - ALfloat TargetGains[MAX_OUTPUT_CHANNELS]; + ALfloat mCurrentGains[MAX_OUTPUT_CHANNELS]; + ALfloat mTargetGains[MAX_OUTPUT_CHANNELS]; }; static ALvoid ALpshifterState_Destruct(ALpshifterState *state); @@ -167,22 +167,22 @@ ALvoid ALpshifterState_Destruct(ALpshifterState *state) ALboolean ALpshifterState_deviceUpdate(ALpshifterState *state, ALCdevice *device) { /* (Re-)initializing parameters and clear the buffers. */ - state->count = FIFO_LATENCY; - state->PitchShiftI = FRACTIONONE; - state->PitchShift = 1.0f; - state->FreqPerBin = device->Frequency / (ALfloat)STFT_SIZE; - - std::fill(std::begin(state->InFIFO), std::end(state->InFIFO), 0.0f); - std::fill(std::begin(state->OutFIFO), std::end(state->OutFIFO), 0.0f); - std::fill(std::begin(state->LastPhase), std::end(state->LastPhase), 0.0); - std::fill(std::begin(state->SumPhase), std::end(state->SumPhase), 0.0); - std::fill(std::begin(state->OutputAccum), std::end(state->OutputAccum), 0.0); - std::fill(std::begin(state->FFTbuffer), std::end(state->FFTbuffer), complex_d{}); - std::fill(std::begin(state->Analysis_buffer), std::end(state->Analysis_buffer), ALfrequencyDomain{}); - std::fill(std::begin(state->Syntesis_buffer), std::end(state->Syntesis_buffer), ALfrequencyDomain{}); - - std::fill(std::begin(state->CurrentGains), std::end(state->CurrentGains), 0.0f); - std::fill(std::begin(state->TargetGains), std::end(state->TargetGains), 0.0f); + state->mCount = FIFO_LATENCY; + state->mPitchShiftI = FRACTIONONE; + state->mPitchShift = 1.0f; + state->mFreqPerBin = device->Frequency / (ALfloat)STFT_SIZE; + + std::fill(std::begin(state->mInFIFO), std::end(state->mInFIFO), 0.0f); + std::fill(std::begin(state->mOutFIFO), std::end(state->mOutFIFO), 0.0f); + std::fill(std::begin(state->mLastPhase), std::end(state->mLastPhase), 0.0); + std::fill(std::begin(state->mSumPhase), std::end(state->mSumPhase), 0.0); + std::fill(std::begin(state->mOutputAccum), std::end(state->mOutputAccum), 0.0); + std::fill(std::begin(state->mFFTbuffer), std::end(state->mFFTbuffer), complex_d{}); + std::fill(std::begin(state->mAnalysis_buffer), std::end(state->mAnalysis_buffer), ALfrequencyDomain{}); + std::fill(std::begin(state->mSyntesis_buffer), std::end(state->mSyntesis_buffer), ALfrequencyDomain{}); + + std::fill(std::begin(state->mCurrentGains), std::end(state->mCurrentGains), 0.0f); + std::fill(std::begin(state->mTargetGains), std::end(state->mTargetGains), 0.0f); return AL_TRUE; } @@ -196,11 +196,11 @@ ALvoid ALpshifterState_update(ALpshifterState *state, const ALCcontext *context, pitch = std::pow(2.0f, (ALfloat)(props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune) / 1200.0f ); - state->PitchShiftI = fastf2i(pitch*FRACTIONONE); - state->PitchShift = state->PitchShiftI * (1.0f/FRACTIONONE); + state->mPitchShiftI = fastf2i(pitch*FRACTIONONE); + state->mPitchShift = state->mPitchShiftI * (1.0f/FRACTIONONE); CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs); - ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->TargetGains); + ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->mTargetGains); } ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels) @@ -210,16 +210,16 @@ ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, cons */ static constexpr ALdouble expected{M_PI*2.0 / OVERSAMP}; - const ALdouble freq_per_bin{state->FreqPerBin}; - ALfloat *RESTRICT bufferOut{state->BufferOut}; - ALsizei count{state->count}; + const ALdouble freq_per_bin{state->mFreqPerBin}; + ALfloat *RESTRICT bufferOut{state->mBufferOut}; + ALsizei count{state->mCount}; for(ALsizei i{0};i < SamplesToDo;) { do { /* Fill FIFO buffer with samples data */ - state->InFIFO[count] = SamplesIn[0][i]; - bufferOut[i] = state->OutFIFO[count - FIFO_LATENCY]; + state->mInFIFO[count] = SamplesIn[0][i]; + bufferOut[i] = state->mOutFIFO[count - FIFO_LATENCY]; count++; } while(++i < SamplesToDo && count < STFT_SIZE); @@ -231,13 +231,13 @@ ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, cons /* Real signal windowing and store in FFTbuffer */ for(ALsizei k{0};k < STFT_SIZE;k++) { - state->FFTbuffer[k].real(state->InFIFO[k] * HannWindow[k]); - state->FFTbuffer[k].imag(0.0); + state->mFFTbuffer[k].real(state->mInFIFO[k] * HannWindow[k]); + state->mFFTbuffer[k].imag(0.0); } /* ANALYSIS */ /* Apply FFT to FFTbuffer data */ - complex_fft(state->FFTbuffer, STFT_SIZE, -1.0); + complex_fft(state->mFFTbuffer, STFT_SIZE, -1.0); /* Analyze the obtained data. Since the real FFT is symmetric, only * STFT_HALF_SIZE+1 samples are needed. @@ -245,10 +245,10 @@ ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, cons for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++) { /* Compute amplitude and phase */ - ALphasor component{rect2polar(state->FFTbuffer[k])}; + ALphasor component{rect2polar(state->mFFTbuffer[k])}; /* Compute phase difference and subtract expected phase difference */ - double tmp{(component.Phase - state->LastPhase[k]) - k*expected}; + double tmp{(component.Phase - state->mLastPhase[k]) - k*expected}; /* Map delta phase into +/- Pi interval */ int qpd{double2int(tmp / M_PI)}; @@ -261,29 +261,29 @@ ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, cons * for maintain the gain (because half of bins are used) and store * amplitude and true frequency in analysis buffer. */ - state->Analysis_buffer[k].Amplitude = 2.0 * component.Amplitude; - state->Analysis_buffer[k].Frequency = (k + tmp) * freq_per_bin; + state->mAnalysis_buffer[k].Amplitude = 2.0 * component.Amplitude; + state->mAnalysis_buffer[k].Frequency = (k + tmp) * freq_per_bin; /* Store actual phase[k] for the calculations in the next frame*/ - state->LastPhase[k] = component.Phase; + state->mLastPhase[k] = component.Phase; } /* PROCESSING */ /* pitch shifting */ for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++) { - state->Syntesis_buffer[k].Amplitude = 0.0; - state->Syntesis_buffer[k].Frequency = 0.0; + state->mSyntesis_buffer[k].Amplitude = 0.0; + state->mSyntesis_buffer[k].Frequency = 0.0; } for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++) { - ALsizei j{(k*state->PitchShiftI) >> FRACTIONBITS}; + ALsizei j{(k*state->mPitchShiftI) >> FRACTIONBITS}; if(j >= STFT_HALF_SIZE+1) break; - state->Syntesis_buffer[j].Amplitude += state->Analysis_buffer[k].Amplitude; - state->Syntesis_buffer[j].Frequency = state->Analysis_buffer[k].Frequency * - state->PitchShift; + state->mSyntesis_buffer[j].Amplitude += state->mAnalysis_buffer[k].Amplitude; + state->mSyntesis_buffer[j].Frequency = state->mAnalysis_buffer[k].Frequency * + state->mPitchShift; } /* SYNTHESIS */ @@ -294,41 +294,41 @@ ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, cons ALdouble tmp; /* Compute bin deviation from scaled freq */ - tmp = state->Syntesis_buffer[k].Frequency/freq_per_bin - k; + tmp = state->mSyntesis_buffer[k].Frequency/freq_per_bin - k; /* Calculate actual delta phase and accumulate it to get bin phase */ - state->SumPhase[k] += (k + tmp) * expected; + state->mSumPhase[k] += (k + tmp) * expected; - component.Amplitude = state->Syntesis_buffer[k].Amplitude; - component.Phase = state->SumPhase[k]; + component.Amplitude = state->mSyntesis_buffer[k].Amplitude; + component.Phase = state->mSumPhase[k]; /* Compute phasor component to cartesian complex number and storage it into FFTbuffer*/ - state->FFTbuffer[k] = polar2rect(component); + state->mFFTbuffer[k] = polar2rect(component); } /* zero negative frequencies for recontruct a real signal */ for(ALsizei k{STFT_HALF_SIZE+1};k < STFT_SIZE;k++) - state->FFTbuffer[k] = complex_d{}; + state->mFFTbuffer[k] = complex_d{}; /* Apply iFFT to buffer data */ - complex_fft(state->FFTbuffer, STFT_SIZE, 1.0); + complex_fft(state->mFFTbuffer, STFT_SIZE, 1.0); /* Windowing and add to output */ for(ALsizei k{0};k < STFT_SIZE;k++) - state->OutputAccum[k] += HannWindow[k] * state->FFTbuffer[k].real() / - (0.5 * STFT_HALF_SIZE * OVERSAMP); + state->mOutputAccum[k] += HannWindow[k] * state->mFFTbuffer[k].real() / + (0.5 * STFT_HALF_SIZE * OVERSAMP); /* Shift accumulator, input & output FIFO */ ALsizei j, k; - for(k = 0;k < STFT_STEP;k++) state->OutFIFO[k] = (ALfloat)state->OutputAccum[k]; - for(j = 0;k < STFT_SIZE;k++,j++) state->OutputAccum[j] = state->OutputAccum[k]; - for(;j < STFT_SIZE;j++) state->OutputAccum[j] = 0.0; + for(k = 0;k < STFT_STEP;k++) state->mOutFIFO[k] = (ALfloat)state->mOutputAccum[k]; + for(j = 0;k < STFT_SIZE;k++,j++) state->mOutputAccum[j] = state->mOutputAccum[k]; + for(;j < STFT_SIZE;j++) state->mOutputAccum[j] = 0.0; for(k = 0;k < FIFO_LATENCY;k++) - state->InFIFO[k] = state->InFIFO[k+STFT_STEP]; + state->mInFIFO[k] = state->mInFIFO[k+STFT_STEP]; } - state->count = count; + state->mCount = count; /* Now, mix the processed sound data to the output. */ - MixSamples(bufferOut, NumChannels, SamplesOut, state->CurrentGains, state->TargetGains, + MixSamples(bufferOut, NumChannels, SamplesOut, state->mCurrentGains, state->mTargetGains, maxi(SamplesToDo, 512), 0, SamplesToDo); } |