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-rw-r--r--alc/effects/distortion.cpp24
-rw-r--r--alc/effects/equalizer.cpp34
-rw-r--r--alc/effects/modulator.cpp2
-rw-r--r--alc/effects/reverb.cpp16
-rw-r--r--alc/filters/biquad.cpp8
-rw-r--r--alc/filters/biquad.h5
-rw-r--r--alc/voice.cpp46
7 files changed, 67 insertions, 68 deletions
diff --git a/alc/effects/distortion.cpp b/alc/effects/distortion.cpp
index 0916b7c6..48fc83ee 100644
--- a/alc/effects/distortion.cpp
+++ b/alc/effects/distortion.cpp
@@ -20,11 +20,10 @@
#include "config.h"
+#include <algorithm>
#include <cmath>
#include <cstdlib>
-#include <cmath>
-
#include "al/auxeffectslot.h"
#include "alcmain.h"
#include "alcontext.h"
@@ -114,26 +113,25 @@ void DistortionState::process(const size_t samplesToDo, const al::span<const Flo
* (which is fortunately first step of distortion). So combine three
* operations into the one.
*/
- mLowpass.process(mBuffer[1], mBuffer[0], todo);
+ mLowpass.process({mBuffer[0], todo}, mBuffer[1]);
/* Second step, do distortion using waveshaper function to emulate
* signal processing during tube overdriving. Three steps of
* waveshaping are intended to modify waveform without boost/clipping/
* attenuation process.
*/
- for(size_t i{0u};i < todo;i++)
+ auto proc_sample = [fc](float smp) -> float
{
- ALfloat smp{mBuffer[1][i]};
-
- smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
- smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f;
- smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
-
- mBuffer[0][i] = smp;
- }
+ smp = (1.0f + fc) * smp/(1.0f + fc*std::abs(smp));
+ smp = (1.0f + fc) * smp/(1.0f + fc*std::abs(smp)) * -1.0f;
+ smp = (1.0f + fc) * smp/(1.0f + fc*std::abs(smp));
+ return smp;
+ };
+ std::transform(std::begin(mBuffer[1]), std::begin(mBuffer[1])+todo, std::begin(mBuffer[0]),
+ proc_sample);
/* Third step, do bandpass filtering of distorted signal. */
- mBandpass.process(mBuffer[1], mBuffer[0], todo);
+ mBandpass.process({mBuffer[0], todo}, mBuffer[1]);
todo >>= 2;
const ALfloat *outgains{mGain};
diff --git a/alc/effects/equalizer.cpp b/alc/effects/equalizer.cpp
index 11fb1498..a4204b3a 100644
--- a/alc/effects/equalizer.cpp
+++ b/alc/effects/equalizer.cpp
@@ -117,25 +117,26 @@ void EqualizerState::update(const ALCcontext *context, const ALeffectslot *slot,
/* Calculate coefficients for the each type of filter. Note that the shelf
* and peaking filters' gain is for the centerpoint of the transition band,
- * meaning its dB needs to be doubled for the shelf or peak to reach the
- * provided gain.
+ * while the effect property gains are for the shelf/peak itself. So the
+ * property gains need their dB halved (sqrt of linear gain) for the
+ * shelf/peak to reach the provided gain.
*/
- gain = maxf(std::sqrt(props->Equalizer.LowGain), 0.0625f); /* Limit -24dB */
- f0norm = props->Equalizer.LowCutoff/frequency;
+ gain = std::sqrt(props->Equalizer.LowGain);
+ f0norm = props->Equalizer.LowCutoff / frequency;
mChans[0].filter[0].setParamsFromSlope(BiquadType::LowShelf, f0norm, gain, 0.75f);
- gain = maxf(std::sqrt(props->Equalizer.Mid1Gain), 0.0625f);
- f0norm = props->Equalizer.Mid1Center/frequency;
+ gain = std::sqrt(props->Equalizer.Mid1Gain);
+ f0norm = props->Equalizer.Mid1Center / frequency;
mChans[0].filter[1].setParamsFromBandwidth(BiquadType::Peaking, f0norm, gain,
props->Equalizer.Mid1Width);
- gain = maxf(std::sqrt(props->Equalizer.Mid2Gain), 0.0625f);
- f0norm = props->Equalizer.Mid2Center/frequency;
+ gain = std::sqrt(props->Equalizer.Mid2Gain);
+ f0norm = props->Equalizer.Mid2Center / frequency;
mChans[0].filter[2].setParamsFromBandwidth(BiquadType::Peaking, f0norm, gain,
props->Equalizer.Mid2Width);
- gain = maxf(std::sqrt(props->Equalizer.HighGain), 0.0625f);
- f0norm = props->Equalizer.HighCutoff/frequency;
+ gain = std::sqrt(props->Equalizer.HighGain);
+ f0norm = props->Equalizer.HighCutoff / frequency;
mChans[0].filter[3].setParamsFromSlope(BiquadType::HighShelf, f0norm, gain, 0.75f);
/* Copy the filter coefficients for the other input channels. */
@@ -157,16 +158,17 @@ void EqualizerState::update(const ALCcontext *context, const ALeffectslot *slot,
void EqualizerState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
+ const al::span<float> buffer{mSampleBuffer, samplesToDo};
auto chandata = std::addressof(mChans[0]);
for(const auto &input : samplesIn)
{
- chandata->filter[0].process(mSampleBuffer, input.data(), samplesToDo);
- chandata->filter[1].process(mSampleBuffer, mSampleBuffer, samplesToDo);
- chandata->filter[2].process(mSampleBuffer, mSampleBuffer, samplesToDo);
- chandata->filter[3].process(mSampleBuffer, mSampleBuffer, samplesToDo);
+ chandata->filter[0].process({input.data(), samplesToDo}, buffer.begin());
+ chandata->filter[1].process(buffer, buffer.begin());
+ chandata->filter[2].process(buffer, buffer.begin());
+ chandata->filter[3].process(buffer, buffer.begin());
- MixSamples({mSampleBuffer, samplesToDo}, samplesOut, chandata->CurrentGains,
- chandata->TargetGains, samplesToDo, 0);
+ MixSamples(buffer, samplesOut, chandata->CurrentGains, chandata->TargetGains, samplesToDo,
+ 0u);
++chandata;
}
}
diff --git a/alc/effects/modulator.cpp b/alc/effects/modulator.cpp
index aa339dea..00afa052 100644
--- a/alc/effects/modulator.cpp
+++ b/alc/effects/modulator.cpp
@@ -146,7 +146,7 @@ void ModulatorState::process(const size_t samplesToDo, const al::span<const Floa
{
alignas(16) ALfloat temps[MAX_UPDATE_SAMPLES];
- chandata->Filter.process(temps, &input[base], td);
+ chandata->Filter.process({&input[base], td}, temps);
for(size_t i{0u};i < td;i++)
temps[i] *= modsamples[i];
diff --git a/alc/effects/reverb.cpp b/alc/effects/reverb.cpp
index 6a5503f5..c8ebba68 100644
--- a/alc/effects/reverb.cpp
+++ b/alc/effects/reverb.cpp
@@ -286,10 +286,10 @@ struct T60Filter {
const ALfloat hfDecayTime, const ALfloat lf0norm, const ALfloat hf0norm);
/* Applies the two T60 damping filter sections. */
- void process(ALfloat *samples, const size_t todo)
+ void process(const al::span<float> samples)
{
- HFFilter.process(samples, samples, todo);
- LFFilter.process(samples, samples, todo);
+ HFFilter.process(samples, samples.begin());
+ LFFilter.process(samples, samples.begin());
}
};
@@ -1359,7 +1359,7 @@ void ReverbState::lateUnfaded(const size_t offset, const size_t todo)
late_delay.Line[late_feedb_tap++][j]*midGain;
} while(--td);
}
- mLate.T60[j].process(mTempSamples[j].data(), todo);
+ mLate.T60[j].process({mTempSamples[j].data(), todo});
}
/* Apply a vector all-pass to improve micro-surface diffusion, and write
@@ -1420,7 +1420,7 @@ void ReverbState::lateFaded(const size_t offset, const size_t todo, const ALfloa
late_delay.Line[late_feedb_tap1++][j]*gfade1;
} while(--td);
}
- mLate.T60[j].process(mTempSamples[j].data(), todo);
+ mLate.T60[j].process({mTempSamples[j].data(), todo});
}
mLate.VecAp.processFaded(mTempSamples, offset, mixX, mixY, fade, fadeStep, todo);
@@ -1445,9 +1445,9 @@ void ReverbState::process(const size_t samplesToDo, const al::span<const FloatBu
MixRowSamples(tmpspan, {B2A[c], numInput}, samplesIn[0].data(), samplesIn[0].size());
/* Band-pass the incoming samples and feed the initial delay line. */
- mFilter[c].Lp.process(mTempLine.data(), mTempLine.data(), samplesToDo);
- mFilter[c].Hp.process(mTempLine.data(), mTempLine.data(), samplesToDo);
- mDelay.write(offset, c, mTempLine.data(), samplesToDo);
+ mFilter[c].Lp.process(tmpspan, tmpspan.begin());
+ mFilter[c].Hp.process(tmpspan, tmpspan.begin());
+ mDelay.write(offset, c, tmpspan.cbegin(), samplesToDo);
}
/* Process reverb for these samples. */
diff --git a/alc/filters/biquad.cpp b/alc/filters/biquad.cpp
index 13cbca3f..271ca696 100644
--- a/alc/filters/biquad.cpp
+++ b/alc/filters/biquad.cpp
@@ -89,10 +89,8 @@ void BiquadFilterR<Real>::setParams(BiquadType type, Real f0norm, Real gain, Rea
}
template<typename Real>
-void BiquadFilterR<Real>::process(Real *dst, const Real *src, const size_t numsamples)
+void BiquadFilterR<Real>::process(const al::span<const Real> src, Real *dst)
{
- ASSUME(numsamples > 0);
-
const Real b0{mB0};
const Real b1{mB1};
const Real b2{mB2};
@@ -111,12 +109,12 @@ void BiquadFilterR<Real>::process(Real *dst, const Real *src, const size_t numsa
*/
auto proc_sample = [b0,b1,b2,a1,a2,&z1,&z2](Real input) noexcept -> Real
{
- Real output = input*b0 + z1;
+ const Real output{input*b0 + z1};
z1 = input*b1 - output*a1 + z2;
z2 = input*b2 - output*a2;
return output;
};
- std::transform(src, src+numsamples, dst, proc_sample);
+ std::transform(src.cbegin(), src.cend(), dst, proc_sample);
mZ1 = z1;
mZ2 = z2;
diff --git a/alc/filters/biquad.h b/alc/filters/biquad.h
index a8bc86e7..30eed57d 100644
--- a/alc/filters/biquad.h
+++ b/alc/filters/biquad.h
@@ -6,6 +6,7 @@
#include <cstddef>
#include <utility>
+#include "alspan.h"
#include "math_defs.h"
@@ -114,14 +115,14 @@ public:
}
- void process(Real *dst, const Real *src, const size_t numsamples);
+ void process(const al::span<const Real> src, Real *dst);
/* Rather hacky. It's just here to support "manual" processing. */
std::pair<Real,Real> getComponents() const noexcept { return {mZ1, mZ2}; }
void setComponents(Real z1, Real z2) noexcept { mZ1 = z1; mZ2 = z2; }
Real processOne(const Real in, Real &z1, Real &z2) const noexcept
{
- Real out{in*mB0 + z1};
+ const Real out{in*mB0 + z1};
z1 = in*mB1 - out*mA1 + z2;
z2 = in*mB2 - out*mA2;
return out;
diff --git a/alc/voice.cpp b/alc/voice.cpp
index 14680a78..4697cc56 100644
--- a/alc/voice.cpp
+++ b/alc/voice.cpp
@@ -284,31 +284,31 @@ void SendSourceStoppedEvent(ALCcontext *context, ALuint id)
}
-const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter, ALfloat *dst,
- const ALfloat *src, const size_t numsamples, int type)
+const float *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter, float *dst,
+ const al::span<const float> src, int type)
{
switch(type)
{
- case AF_None:
- lpfilter->clear();
- hpfilter->clear();
- break;
-
- case AF_LowPass:
- lpfilter->process(dst, src, numsamples);
- hpfilter->clear();
- return dst;
- case AF_HighPass:
- lpfilter->clear();
- hpfilter->process(dst, src, numsamples);
- return dst;
-
- case AF_BandPass:
- lpfilter->process(dst, src, numsamples);
- hpfilter->process(dst, dst, numsamples);
- return dst;
+ case AF_None:
+ lpfilter->clear();
+ hpfilter->clear();
+ break;
+
+ case AF_LowPass:
+ lpfilter->process(src, dst);
+ hpfilter->clear();
+ return dst;
+ case AF_HighPass:
+ lpfilter->clear();
+ hpfilter->process(src, dst);
+ return dst;
+
+ case AF_BandPass:
+ lpfilter->process(src, dst);
+ hpfilter->process({dst, src.size()}, dst);
+ return dst;
}
- return src;
+ return src.data();
}
@@ -694,7 +694,7 @@ void ALvoice::mix(const State vstate, ALCcontext *Context, const ALuint SamplesT
{
DirectParams &parms = chandata.mDryParams;
const ALfloat *samples{DoFilters(&parms.LowPass, &parms.HighPass, FilterBuf,
- ResampledData, DstBufferSize, mDirect.FilterType)};
+ {ResampledData, DstBufferSize}, mDirect.FilterType)};
if((mFlags&VOICE_HAS_HRTF))
{
@@ -726,7 +726,7 @@ void ALvoice::mix(const State vstate, ALCcontext *Context, const ALuint SamplesT
SendParams &parms = chandata.mWetParams[send];
const ALfloat *samples{DoFilters(&parms.LowPass, &parms.HighPass, FilterBuf,
- ResampledData, DstBufferSize, mSend[send].FilterType)};
+ {ResampledData, DstBufferSize}, mSend[send].FilterType)};
const float *TargetGains{UNLIKELY(vstate == ALvoice::Stopping) ?
SilentTarget.data() : parms.Gains.Target.data()};
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-04 12:43:33 +0000