From c36798fd0759331caac80bb16cebe6c19a646090 Mon Sep 17 00:00:00 2001 From: Chris Robinson Date: Tue, 1 Jan 2019 02:41:27 -0800 Subject: Avoid unnecessary extra buffers for filter chains --- Alc/effects/equalizer.cpp | 22 ++++++++--------- Alc/effects/reverb.cpp | 7 +++--- Alc/uhjfilter.cpp | 61 +++++++++++++++++++++++++---------------------- 3 files changed, 46 insertions(+), 44 deletions(-) (limited to 'Alc') diff --git a/Alc/effects/equalizer.cpp b/Alc/effects/equalizer.cpp index cb421914..94c760ea 100644 --- a/Alc/effects/equalizer.cpp +++ b/Alc/effects/equalizer.cpp @@ -35,6 +35,8 @@ #include "vecmat.h" +namespace { + /* The document "Effects Extension Guide.pdf" says that low and high * * frequencies are cutoff frequencies. This is not fully correct, they * * are corner frequencies for low and high shelf filters. If they were * @@ -87,7 +89,7 @@ struct ALequalizerState final : public EffectState { ALfloat TargetGains[MAX_OUTPUT_CHANNELS]{}; } mChans[MAX_EFFECT_CHANNELS]; - ALfloat mSampleBuffer[MAX_EFFECT_CHANNELS][BUFFERSIZE]{}; + ALfloat mSampleBuffer[BUFFERSIZE]{}; ALboolean deviceUpdate(const ALCdevice *device) override; @@ -161,21 +163,19 @@ void ALequalizerState::update(const ALCcontext *context, const ALeffectslot *slo void ALequalizerState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { - ALfloat (*RESTRICT temps)[BUFFERSIZE] = mSampleBuffer; - ALsizei c; - - for(c = 0;c < MAX_EFFECT_CHANNELS;c++) + for(ALsizei c{0};c < MAX_EFFECT_CHANNELS;c++) { - mChans[c].filter[0].process(temps[0], SamplesIn[c], SamplesToDo); - mChans[c].filter[1].process(temps[1], temps[0], SamplesToDo); - mChans[c].filter[2].process(temps[2], temps[1], SamplesToDo); - mChans[c].filter[3].process(temps[3], temps[2], SamplesToDo); + mChans[c].filter[0].process(mSampleBuffer, SamplesIn[c], SamplesToDo); + mChans[c].filter[1].process(mSampleBuffer, mSampleBuffer, SamplesToDo); + mChans[c].filter[2].process(mSampleBuffer, mSampleBuffer, SamplesToDo); + mChans[c].filter[3].process(mSampleBuffer, mSampleBuffer, SamplesToDo); - MixSamples(temps[3], NumChannels, SamplesOut, mChans[c].CurrentGains, - mChans[c].TargetGains, SamplesToDo, 0, SamplesToDo); + MixSamples(mSampleBuffer, NumChannels, SamplesOut, mChans[c].CurrentGains, + mChans[c].TargetGains, SamplesToDo, 0, SamplesToDo); } } +} // namespace struct EqualizerStateFactory final : public EffectStateFactory { EffectState *create() override; diff --git a/Alc/effects/reverb.cpp b/Alc/effects/reverb.cpp index d984ceab..9bc4f8f2 100644 --- a/Alc/effects/reverb.cpp +++ b/Alc/effects/reverb.cpp @@ -1183,11 +1183,10 @@ void EarlyReflection_Faded(ReverbState *State, ALsizei offset, const ALsizei tod } /* Applies the two T60 damping filter sections. */ -static inline void LateT60Filter(ALfloat *RESTRICT samples, const ALsizei todo, T60Filter *filter) +inline void LateT60Filter(ALfloat *samples, const ALsizei todo, T60Filter *filter) { - ALfloat temp[MAX_UPDATE_SAMPLES]; - filter->HFFilter.process(temp, samples, todo); - filter->LFFilter.process(samples, temp, todo); + filter->HFFilter.process(samples, samples, todo); + filter->LFFilter.process(samples, samples, todo); } /* This generates the reverb tail using a modified feed-back delay network diff --git a/Alc/uhjfilter.cpp b/Alc/uhjfilter.cpp index 1c5c836c..64d5f76c 100644 --- a/Alc/uhjfilter.cpp +++ b/Alc/uhjfilter.cpp @@ -1,9 +1,12 @@ #include "config.h" -#include "alu.h" #include "uhjfilter.h" +#include + +#include "alu.h" + namespace { /* This is the maximum number of samples processed for each inner loop @@ -18,17 +21,17 @@ constexpr ALfloat Filter2CoeffSqr[4] = { 0.161758498368f, 0.733028932341f, 0.945349700329f, 0.990599156685f }; -void allpass_process(AllPassState *state, ALfloat *RESTRICT dst, const ALfloat *RESTRICT src, const ALfloat aa, ALsizei todo) +void allpass_process(AllPassState *state, ALfloat *dst, const ALfloat *src, const ALfloat aa, ALsizei todo) { - ALfloat z1 = state->z[0]; - ALfloat z2 = state->z[1]; - for(ALsizei i{0};i < todo;i++) + ALfloat z1{state->z[0]}; + ALfloat z2{state->z[1]}; + auto proc_sample = [aa,&z1,&z2](ALfloat input) noexcept -> ALfloat { - ALfloat input = src[i]; ALfloat output = input*aa + z1; z1 = z2; z2 = output*aa - input; - dst[i] = output; - } + return output; + }; + std::transform(src, src+todo, dst, proc_sample); state->z[0] = z1; state->z[1] = z2; } @@ -59,7 +62,7 @@ void allpass_process(AllPassState *state, ALfloat *RESTRICT dst, const ALfloat * void Uhj2Encoder::encode(ALfloat *LeftOut, ALfloat *RightOut, ALfloat (*InSamples)[BUFFERSIZE], const ALsizei SamplesToDo) { alignas(16) ALfloat D[MAX_UPDATE_SAMPLES], S[MAX_UPDATE_SAMPLES]; - alignas(16) ALfloat temp[2][MAX_UPDATE_SAMPLES]; + alignas(16) ALfloat temp[MAX_UPDATE_SAMPLES]; ASSUME(SamplesToDo > 0); @@ -71,43 +74,43 @@ void Uhj2Encoder::encode(ALfloat *LeftOut, ALfloat *RightOut, ALfloat (*InSample /* D = 0.6554516*Y */ const ALfloat *RESTRICT input{al::assume_aligned<16>(InSamples[2]+base)}; for(ALsizei i{0};i < todo;i++) - temp[0][i] = 0.6554516f*input[i]; - allpass_process(&mFilter1_Y[0], temp[1], temp[0], Filter1CoeffSqr[0], todo); - allpass_process(&mFilter1_Y[1], temp[0], temp[1], Filter1CoeffSqr[1], todo); - allpass_process(&mFilter1_Y[2], temp[1], temp[0], Filter1CoeffSqr[2], todo); - allpass_process(&mFilter1_Y[3], temp[0], temp[1], Filter1CoeffSqr[3], todo); + temp[i] = 0.6554516f*input[i]; + allpass_process(&mFilter1_Y[0], temp, temp, Filter1CoeffSqr[0], todo); + allpass_process(&mFilter1_Y[1], temp, temp, Filter1CoeffSqr[1], todo); + allpass_process(&mFilter1_Y[2], temp, temp, Filter1CoeffSqr[2], todo); + allpass_process(&mFilter1_Y[3], temp, temp, Filter1CoeffSqr[3], todo); /* NOTE: Filter1 requires a 1 sample delay for the final output, so * take the last processed sample from the previous run as the first * output sample. */ D[0] = mLastY; for(ALsizei i{1};i < todo;i++) - D[i] = temp[0][i-1]; - mLastY = temp[0][todo-1]; + D[i] = temp[i-1]; + mLastY = temp[todo-1]; /* D += j(-0.3420201*W + 0.5098604*X) */ const ALfloat *RESTRICT input0{al::assume_aligned<16>(InSamples[0]+base)}; const ALfloat *RESTRICT input1{al::assume_aligned<16>(InSamples[1]+base)}; for(ALsizei i{0};i < todo;i++) - temp[0][i] = -0.3420201f*input0[i] + 0.5098604f*input1[i]; - allpass_process(&mFilter2_WX[0], temp[1], temp[0], Filter2CoeffSqr[0], todo); - allpass_process(&mFilter2_WX[1], temp[0], temp[1], Filter2CoeffSqr[1], todo); - allpass_process(&mFilter2_WX[2], temp[1], temp[0], Filter2CoeffSqr[2], todo); - allpass_process(&mFilter2_WX[3], temp[0], temp[1], Filter2CoeffSqr[3], todo); + temp[i] = -0.3420201f*input0[i] + 0.5098604f*input1[i]; + allpass_process(&mFilter2_WX[0], temp, temp, Filter2CoeffSqr[0], todo); + allpass_process(&mFilter2_WX[1], temp, temp, Filter2CoeffSqr[1], todo); + allpass_process(&mFilter2_WX[2], temp, temp, Filter2CoeffSqr[2], todo); + allpass_process(&mFilter2_WX[3], temp, temp, Filter2CoeffSqr[3], todo); for(ALsizei i{0};i < todo;i++) - D[i] += temp[0][i]; + D[i] += temp[i]; /* S = 0.9396926*W + 0.1855740*X */ for(ALsizei i{0};i < todo;i++) - temp[0][i] = 0.9396926f*input0[i] + 0.1855740f*input1[i]; - allpass_process(&mFilter1_WX[0], temp[1], temp[0], Filter1CoeffSqr[0], todo); - allpass_process(&mFilter1_WX[1], temp[0], temp[1], Filter1CoeffSqr[1], todo); - allpass_process(&mFilter1_WX[2], temp[1], temp[0], Filter1CoeffSqr[2], todo); - allpass_process(&mFilter1_WX[3], temp[0], temp[1], Filter1CoeffSqr[3], todo); + temp[i] = 0.9396926f*input0[i] + 0.1855740f*input1[i]; + allpass_process(&mFilter1_WX[0], temp, temp, Filter1CoeffSqr[0], todo); + allpass_process(&mFilter1_WX[1], temp, temp, Filter1CoeffSqr[1], todo); + allpass_process(&mFilter1_WX[2], temp, temp, Filter1CoeffSqr[2], todo); + allpass_process(&mFilter1_WX[3], temp, temp, Filter1CoeffSqr[3], todo); S[0] = mLastWX; for(ALsizei i{1};i < todo;i++) - S[i] = temp[0][i-1]; - mLastWX = temp[0][todo-1]; + S[i] = temp[i-1]; + mLastWX = temp[todo-1]; /* Left = (S + D)/2.0 */ ALfloat *RESTRICT left = al::assume_aligned<16>(LeftOut+base); 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