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-rw-r--r--Alc/effects/autowah.cpp19
-rw-r--r--Alc/effects/chorus.cpp14
-rw-r--r--Alc/effects/compressor.cpp26
-rw-r--r--Alc/effects/dedicated.cpp8
-rw-r--r--Alc/effects/distortion.cpp18
-rw-r--r--Alc/effects/echo.cpp14
-rw-r--r--Alc/effects/equalizer.cpp19
-rw-r--r--Alc/effects/fshifter.cpp18
-rw-r--r--Alc/effects/modulator.cpp17
-rw-r--r--Alc/effects/null.cpp4
-rw-r--r--Alc/effects/pshifter.cpp14
-rw-r--r--Alc/effects/reverb.cpp38
12 files changed, 103 insertions, 106 deletions
diff --git a/Alc/effects/autowah.cpp b/Alc/effects/autowah.cpp
index f7d07fe2..c391da38 100644
--- a/Alc/effects/autowah.cpp
+++ b/Alc/effects/autowah.cpp
@@ -70,7 +70,7 @@ struct ALautowahState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALautowahState)
};
@@ -126,7 +126,7 @@ void ALautowahState::update(const ALCcontext *context, const ALeffectslot *slot,
mChans[i].TargetGains);
}
-void ALautowahState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ALautowahState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
const ALfloat attack_rate = mAttackRate;
const ALfloat release_rate = mReleaseRate;
@@ -138,14 +138,14 @@ void ALautowahState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Sampl
ALsizei c, i;
env_delay = mEnvDelay;
- for(i = 0;i < SamplesToDo;i++)
+ for(i = 0;i < samplesToDo;i++)
{
ALfloat w0, sample, a;
/* Envelope follower described on the book: Audio Effects, Theory,
* Implementation and Application.
*/
- sample = peak_gain * fabsf(SamplesIn[0][i]);
+ sample = peak_gain * std::fabs(samplesIn[0][i]);
a = (sample > env_delay) ? attack_rate : release_rate;
env_delay = lerp(sample, env_delay, a);
@@ -156,7 +156,8 @@ void ALautowahState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Sampl
}
mEnvDelay = env_delay;
- for(c = 0;c < MAX_EFFECT_CHANNELS; c++)
+ ASSUME(numInput > 0);
+ for(c = 0;c < numInput;++c)
{
/* This effectively inlines BiquadFilter_setParams for a peaking
* filter and BiquadFilter_processC. The alpha and cosine components
@@ -167,7 +168,7 @@ void ALautowahState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Sampl
ALfloat z1 = mChans[c].Filter.z1;
ALfloat z2 = mChans[c].Filter.z2;
- for(i = 0;i < SamplesToDo;i++)
+ for(i = 0;i < samplesToDo;i++)
{
const ALfloat alpha = mEnv[i].alpha;
const ALfloat cos_w0 = mEnv[i].cos_w0;
@@ -181,7 +182,7 @@ void ALautowahState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Sampl
a[1] = -2.0f * cos_w0;
a[2] = 1.0f - alpha/res_gain;
- input = SamplesIn[c][i];
+ input = samplesIn[c][i];
output = input*(b[0]/a[0]) + z1;
z1 = input*(b[1]/a[0]) - output*(a[1]/a[0]) + z2;
z2 = input*(b[2]/a[0]) - output*(a[2]/a[0]);
@@ -191,8 +192,8 @@ void ALautowahState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Sampl
mChans[c].Filter.z2 = z2;
/* Now, mix the processed sound data to the output. */
- MixSamples(mBufferOut, NumChannels, SamplesOut, mChans[c].CurrentGains,
- mChans[c].TargetGains, SamplesToDo, 0, SamplesToDo);
+ MixSamples(mBufferOut, numOutput, samplesOut, mChans[c].CurrentGains,
+ mChans[c].TargetGains, samplesToDo, 0, samplesToDo);
}
}
diff --git a/Alc/effects/chorus.cpp b/Alc/effects/chorus.cpp
index 6b3053a4..758d521c 100644
--- a/Alc/effects/chorus.cpp
+++ b/Alc/effects/chorus.cpp
@@ -92,7 +92,7 @@ struct ChorusState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ChorusState)
};
@@ -189,7 +189,7 @@ void ChorusState::update(const ALCcontext *Context, const ALeffectslot *Slot, co
}
}
-void ChorusState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ChorusState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei /*numInput*/, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
const auto bufmask = static_cast<ALsizei>(mSampleBuffer.size()-1);
const ALfloat feedback{mFeedback};
@@ -199,9 +199,9 @@ void ChorusState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesI
ALsizei i, c;
ALsizei base;
- for(base = 0;base < SamplesToDo;)
+ for(base = 0;base < samplesToDo;)
{
- const ALsizei todo = mini(256, SamplesToDo-base);
+ const ALsizei todo = mini(256, samplesToDo-base);
ALint moddelays[2][256];
alignas(16) ALfloat temps[2][256];
@@ -224,7 +224,7 @@ void ChorusState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesI
for(i = 0;i < todo;i++)
{
// Feed the buffer's input first (necessary for delays < 1).
- delaybuf[offset&bufmask] = SamplesIn[0][base+i];
+ delaybuf[offset&bufmask] = samplesIn[0][base+i];
// Tap for the left output.
ALint delay{offset - (moddelays[0][i]>>FRACTIONBITS)};
@@ -246,8 +246,8 @@ void ChorusState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesI
}
for(c = 0;c < 2;c++)
- MixSamples(temps[c], NumChannels, SamplesOut, mGains[c].Current,
- mGains[c].Target, SamplesToDo-base, base, todo);
+ MixSamples(temps[c], numOutput, samplesOut, mGains[c].Current, mGains[c].Target,
+ samplesToDo-base, base, todo);
base += todo;
}
diff --git a/Alc/effects/compressor.cpp b/Alc/effects/compressor.cpp
index 93ab9bbc..23069dd7 100644
--- a/Alc/effects/compressor.cpp
+++ b/Alc/effects/compressor.cpp
@@ -50,7 +50,7 @@ struct ALcompressorState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALcompressorState)
};
@@ -83,15 +83,15 @@ void ALcompressorState::update(const ALCcontext* UNUSED(context), const ALeffect
slot->Params.Gain, mGain[i]);
}
-void ALcompressorState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ALcompressorState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
ALsizei i, j, k;
ALsizei base;
- for(base = 0;base < SamplesToDo;)
+ for(base = 0;base < samplesToDo;)
{
ALfloat gains[256];
- ALsizei td = mini(256, SamplesToDo-base);
+ ALsizei td = mini(256, samplesToDo-base);
ALfloat env = mEnvFollower;
/* Generate the per-sample gains from the signal envelope. */
@@ -102,8 +102,8 @@ void ALcompressorState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Sa
/* Clamp the absolute amplitude to the defined envelope limits,
* then attack or release the envelope to reach it.
*/
- ALfloat amplitude = clampf(fabsf(SamplesIn[0][base+i]),
- AMP_ENVELOPE_MIN, AMP_ENVELOPE_MAX);
+ const ALfloat amplitude{clampf(std::fabs(samplesIn[0][base+i]), AMP_ENVELOPE_MIN,
+ AMP_ENVELOPE_MAX)};
if(amplitude > env)
env = minf(env*mAttackMult, amplitude);
else if(amplitude < env)
@@ -123,7 +123,7 @@ void ALcompressorState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Sa
*/
for(i = 0;i < td;++i)
{
- ALfloat amplitude = 1.0f;
+ const ALfloat amplitude{1.0f};
if(amplitude > env)
env = minf(env*mAttackMult, amplitude);
else if(amplitude < env)
@@ -135,16 +135,18 @@ void ALcompressorState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Sa
mEnvFollower = env;
/* Now compress the signal amplitude to output. */
- for(j = 0;j < MAX_EFFECT_CHANNELS;j++)
+ ASSUME(numInput > 0);
+ for(j = 0;j < numInput;j++)
{
- for(k = 0;k < NumChannels;k++)
+ ASSUME(numOutput > 0);
+ for(k = 0;k < numOutput;k++)
{
- ALfloat gain = mGain[j][k];
- if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+ const ALfloat gain{mGain[j][k]};
+ if(!(std::fabs(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(i = 0;i < td;i++)
- SamplesOut[k][base+i] += SamplesIn[j][base+i] * gains[i] * gain;
+ samplesOut[k][base+i] += samplesIn[j][base+i] * gains[i] * gain;
}
}
diff --git a/Alc/effects/dedicated.cpp b/Alc/effects/dedicated.cpp
index 1804895a..8cc4b936 100644
--- a/Alc/effects/dedicated.cpp
+++ b/Alc/effects/dedicated.cpp
@@ -38,7 +38,7 @@ struct ALdedicatedState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALdedicatedState)
};
@@ -88,10 +88,10 @@ void ALdedicatedState::update(const ALCcontext* UNUSED(context), const ALeffects
}
}
-void ALdedicatedState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ALdedicatedState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei /*numInput*/, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
- MixSamples(SamplesIn[0], NumChannels, SamplesOut, mCurrentGains,
- mTargetGains, SamplesToDo, 0, SamplesToDo);
+ MixSamples(samplesIn[0], numOutput, samplesOut, mCurrentGains, mTargetGains, samplesToDo, 0,
+ samplesToDo);
}
diff --git a/Alc/effects/distortion.cpp b/Alc/effects/distortion.cpp
index 72ce7062..8e21681e 100644
--- a/Alc/effects/distortion.cpp
+++ b/Alc/effects/distortion.cpp
@@ -48,7 +48,7 @@ struct ALdistortionState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALdistortionState)
};
@@ -95,27 +95,27 @@ void ALdistortionState::update(const ALCcontext *context, const ALeffectslot *sl
ComputePanGains(target.Main, coeffs, slot->Params.Gain*props->Distortion.Gain, mGain);
}
-void ALdistortionState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ALdistortionState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
ALfloat (*RESTRICT buffer)[BUFFERSIZE] = mBuffer;
const ALfloat fc = mEdgeCoeff;
ALsizei base;
ALsizei i, k;
- for(base = 0;base < SamplesToDo;)
+ for(base = 0;base < samplesToDo;)
{
/* Perform 4x oversampling to avoid aliasing. Oversampling greatly
* improves distortion quality and allows to implement lowpass and
* bandpass filters using high frequencies, at which classic IIR
* filters became unstable.
*/
- ALsizei todo = mini(BUFFERSIZE, (SamplesToDo-base) * 4);
+ ALsizei todo{mini(BUFFERSIZE, (samplesToDo-base) * 4)};
/* Fill oversample buffer using zero stuffing. Multiply the sample by
* the amount of oversampling to maintain the signal's power.
*/
for(i = 0;i < todo;i++)
- buffer[0][i] = !(i&3) ? SamplesIn[0][(i>>2)+base] * 4.0f : 0.0f;
+ buffer[0][i] = !(i&3) ? samplesIn[0][(i>>2)+base] * 4.0f : 0.0f;
/* First step, do lowpass filtering of original signal. Additionally
* perform buffer interpolation and lowpass cutoff for oversampling
@@ -144,17 +144,17 @@ void ALdistortionState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Sa
mBandpass.process(buffer[1], buffer[0], todo);
todo >>= 2;
- for(k = 0;k < NumChannels;k++)
+ for(k = 0;k < numOutput;k++)
{
/* Fourth step, final, do attenuation and perform decimation,
* storing only one sample out of four.
*/
- ALfloat gain = mGain[k];
- if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
+ const ALfloat gain{mGain[k]};
+ if(!(std::fabs(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(i = 0;i < todo;i++)
- SamplesOut[k][base+i] += gain * buffer[1][i*4];
+ samplesOut[k][base+i] += gain * buffer[1][i*4];
}
base += todo;
diff --git a/Alc/effects/echo.cpp b/Alc/effects/echo.cpp
index ad368b5b..1325e015 100644
--- a/Alc/effects/echo.cpp
+++ b/Alc/effects/echo.cpp
@@ -58,7 +58,7 @@ struct ALechoState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALechoState)
};
@@ -125,7 +125,7 @@ void ALechoState::update(const ALCcontext *context, const ALeffectslot *slot, co
ComputePanGains(target.Main, coeffs[1], slot->Params.Gain, mGains[1].Target);
}
-void ALechoState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ALechoState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei /*numInput*/, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
const auto mask = static_cast<ALsizei>(mSampleBuffer.size()-1);
const ALsizei tap1{mTap[0].delay};
@@ -137,15 +137,15 @@ void ALechoState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesI
ALsizei c, i;
std::tie(z1, z2) = mFilter.getComponents();
- for(base = 0;base < SamplesToDo;)
+ for(base = 0;base < samplesToDo;)
{
alignas(16) ALfloat temps[2][128];
- ALsizei td = mini(128, SamplesToDo-base);
+ ALsizei td = mini(128, samplesToDo-base);
for(i = 0;i < td;i++)
{
/* Feed the delay buffer's input first. */
- delaybuf[offset&mask] = SamplesIn[0][i+base];
+ delaybuf[offset&mask] = samplesIn[0][i+base];
/* First tap */
temps[0][i] = delaybuf[(offset-tap1) & mask];
@@ -162,8 +162,8 @@ void ALechoState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesI
}
for(c = 0;c < 2;c++)
- MixSamples(temps[c], NumChannels, SamplesOut, mGains[c].Current,
- mGains[c].Target, SamplesToDo-base, base, td);
+ MixSamples(temps[c], numOutput, samplesOut, mGains[c].Current, mGains[c].Target,
+ samplesToDo-base, base, td);
base += td;
}
diff --git a/Alc/effects/equalizer.cpp b/Alc/effects/equalizer.cpp
index fc00d00f..942d6139 100644
--- a/Alc/effects/equalizer.cpp
+++ b/Alc/effects/equalizer.cpp
@@ -94,7 +94,7 @@ struct ALequalizerState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALequalizerState)
};
@@ -161,17 +161,18 @@ void ALequalizerState::update(const ALCcontext *context, const ALeffectslot *slo
mChans[i].TargetGains);
}
-void ALequalizerState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ALequalizerState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
- for(ALsizei c{0};c < MAX_EFFECT_CHANNELS;c++)
+ ASSUME(numInput > 0);
+ for(ALsizei c{0};c < numInput;c++)
{
- 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);
+ 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(mSampleBuffer, NumChannels, SamplesOut, mChans[c].CurrentGains,
- mChans[c].TargetGains, SamplesToDo, 0, SamplesToDo);
+ MixSamples(mSampleBuffer, numOutput, samplesOut, mChans[c].CurrentGains,
+ mChans[c].TargetGains, samplesToDo, 0, samplesToDo);
}
}
diff --git a/Alc/effects/fshifter.cpp b/Alc/effects/fshifter.cpp
index 5c808a4b..21d3a706 100644
--- a/Alc/effects/fshifter.cpp
+++ b/Alc/effects/fshifter.cpp
@@ -83,7 +83,7 @@ struct ALfshifterState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALfshifterState)
};
@@ -138,15 +138,15 @@ void ALfshifterState::update(const ALCcontext *context, const ALeffectslot *slot
ComputePanGains(target.Main, coeffs, slot->Params.Gain, mTargetGains);
}
-void ALfshifterState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ALfshifterState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei /*numInput*/, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
static constexpr complex_d complex_zero{0.0, 0.0};
ALfloat *RESTRICT BufferOut = mBufferOut;
ALsizei j, k, base;
- for(base = 0;base < SamplesToDo;)
+ for(base = 0;base < samplesToDo;)
{
- const ALsizei todo{mini(HIL_SIZE-mCount, SamplesToDo-base)};
+ const ALsizei todo{mini(HIL_SIZE-mCount, samplesToDo-base)};
ASSUME(todo > 0);
@@ -154,7 +154,7 @@ void ALfshifterState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Samp
k = mCount;
for(j = 0;j < todo;j++,k++)
{
- mInFIFO[k] = SamplesIn[0][base+j];
+ mInFIFO[k] = samplesIn[0][base+j];
mOutdata[base+j] = mOutFIFO[k-FIFO_LATENCY];
}
mCount += todo;
@@ -187,19 +187,19 @@ void ALfshifterState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Samp
}
/* Process frequency shifter using the analytic signal obtained. */
- for(k = 0;k < SamplesToDo;k++)
+ for(k = 0;k < samplesToDo;k++)
{
double phase = mPhase * ((1.0/FRACTIONONE) * al::MathDefs<double>::Tau());
BufferOut[k] = static_cast<float>(mOutdata[k].real()*std::cos(phase) +
- mOutdata[k].imag()*std::sin(phase)*mLdSign);
+ mOutdata[k].imag()*std::sin(phase)*mLdSign);
mPhase += mPhaseStep;
mPhase &= FRACTIONMASK;
}
/* Now, mix the processed sound data to the output. */
- MixSamples(BufferOut, NumChannels, SamplesOut, mCurrentGains, mTargetGains,
- maxi(SamplesToDo, 512), 0, SamplesToDo);
+ MixSamples(BufferOut, numOutput, samplesOut, mCurrentGains, mTargetGains,
+ maxi(samplesToDo, 512), 0, samplesToDo);
}
} // namespace
diff --git a/Alc/effects/modulator.cpp b/Alc/effects/modulator.cpp
index d9a3046e..62d5a7ad 100644
--- a/Alc/effects/modulator.cpp
+++ b/Alc/effects/modulator.cpp
@@ -90,7 +90,7 @@ struct ALmodulatorState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALmodulatorState)
};
@@ -138,31 +138,32 @@ void ALmodulatorState::update(const ALCcontext *context, const ALeffectslot *slo
mChans[i].TargetGains);
}
-void ALmodulatorState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ALmodulatorState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
const ALsizei step = mStep;
ALsizei base;
- for(base = 0;base < SamplesToDo;)
+ for(base = 0;base < samplesToDo;)
{
alignas(16) ALfloat modsamples[MAX_UPDATE_SAMPLES];
- ALsizei td = mini(MAX_UPDATE_SAMPLES, SamplesToDo-base);
+ ALsizei td = mini(MAX_UPDATE_SAMPLES, samplesToDo-base);
ALsizei c, i;
mGetSamples(modsamples, mIndex, step, td);
mIndex += (step*td) & WAVEFORM_FRACMASK;
mIndex &= WAVEFORM_FRACMASK;
- for(c = 0;c < MAX_EFFECT_CHANNELS;c++)
+ ASSUME(numInput > 0);
+ for(c = 0;c < numInput;c++)
{
alignas(16) ALfloat temps[MAX_UPDATE_SAMPLES];
- mChans[c].Filter.process(temps, &SamplesIn[c][base], td);
+ mChans[c].Filter.process(temps, &samplesIn[c][base], td);
for(i = 0;i < td;i++)
temps[i] *= modsamples[i];
- MixSamples(temps, NumChannels, SamplesOut, mChans[c].CurrentGains,
- mChans[c].TargetGains, SamplesToDo-base, base, td);
+ MixSamples(temps, numOutput, samplesOut, mChans[c].CurrentGains,
+ mChans[c].TargetGains, samplesToDo-base, base, td);
}
base += td;
diff --git a/Alc/effects/null.cpp b/Alc/effects/null.cpp
index 96cb3114..6c9e3008 100644
--- a/Alc/effects/null.cpp
+++ b/Alc/effects/null.cpp
@@ -17,7 +17,7 @@ struct ALnullState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALnullState)
};
@@ -52,7 +52,7 @@ void ALnullState::update(const ALCcontext* UNUSED(context), const ALeffectslot*
* input to the output buffer. The result should be added to the output buffer,
* not replace it.
*/
-void ALnullState::process(ALsizei UNUSED(samplesToDo), const ALfloat (*RESTRICT UNUSED(samplesIn))[BUFFERSIZE], ALfloat (*RESTRICT UNUSED(samplesOut))[BUFFERSIZE], ALsizei UNUSED(numChannels))
+void ALnullState::process(ALsizei /*samplesToDo*/, const ALfloat (*RESTRICT /*samplesIn*/)[BUFFERSIZE], const ALsizei /*numInput*/, ALfloat (*RESTRICT /*samplesOut*/)[BUFFERSIZE], const ALsizei /*numOutput*/)
{
}
diff --git a/Alc/effects/pshifter.cpp b/Alc/effects/pshifter.cpp
index 31d40abc..c76cd6a8 100644
--- a/Alc/effects/pshifter.cpp
+++ b/Alc/effects/pshifter.cpp
@@ -145,7 +145,7 @@ struct ALpshifterState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ALpshifterState)
};
@@ -189,7 +189,7 @@ void ALpshifterState::update(const ALCcontext* UNUSED(context), const ALeffectsl
ComputePanGains(target.Main, coeffs, slot->Params.Gain, mTargetGains);
}
-void ALpshifterState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ALpshifterState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei /*numInput*/, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
/* Pitch shifter engine based on the work of Stephan Bernsee.
* http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
@@ -200,15 +200,15 @@ void ALpshifterState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Samp
ALfloat *RESTRICT bufferOut{mBufferOut};
ALsizei count{mCount};
- for(ALsizei i{0};i < SamplesToDo;)
+ for(ALsizei i{0};i < samplesToDo;)
{
do {
/* Fill FIFO buffer with samples data */
- mInFIFO[count] = SamplesIn[0][i];
+ mInFIFO[count] = samplesIn[0][i];
bufferOut[i] = mOutFIFO[count - FIFO_LATENCY];
count++;
- } while(++i < SamplesToDo && count < STFT_SIZE);
+ } while(++i < samplesToDo && count < STFT_SIZE);
/* Check whether FIFO buffer is filled */
if(count < STFT_SIZE) break;
@@ -313,8 +313,8 @@ void ALpshifterState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT Samp
mCount = count;
/* Now, mix the processed sound data to the output. */
- MixSamples(bufferOut, NumChannels, SamplesOut, mCurrentGains, mTargetGains,
- maxi(SamplesToDo, 512), 0, SamplesToDo);
+ MixSamples(bufferOut, numOutput, samplesOut, mCurrentGains, mTargetGains,
+ maxi(samplesToDo, 512), 0, samplesToDo);
}
} // namespace
diff --git a/Alc/effects/reverb.cpp b/Alc/effects/reverb.cpp
index 764fd376..fe673421 100644
--- a/Alc/effects/reverb.cpp
+++ b/Alc/effects/reverb.cpp
@@ -341,7 +341,7 @@ struct ReverbState final : public EffectState {
ALboolean deviceUpdate(const ALCdevice *device) override;
void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;
+ void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
DEF_NEWDEL(ReverbState)
};
@@ -1316,19 +1316,19 @@ void LateReverb_Faded(ReverbState *State, ALsizei offset, const ALsizei todo, co
VectorScatterRevDelayIn(&late_delay, offset, mixX, mixY, temps, todo);
}
-void ReverbState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
+void ReverbState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
{
ALfloat (*RESTRICT afmt)[MAX_UPDATE_SAMPLES]{mTempSamples};
ALfloat (*RESTRICT samples)[MAX_UPDATE_SAMPLES]{mMixBuffer};
ALsizei fadeCount{mFadeCount};
ALsizei offset{mOffset};
- ASSUME(SamplesToDo > 0);
+ ASSUME(samplesToDo > 0);
/* Process reverb for these samples. */
- for(ALsizei base{0};base < SamplesToDo;)
+ for(ALsizei base{0};base < samplesToDo;)
{
- ALsizei todo{SamplesToDo - base};
+ ALsizei todo{samplesToDo - base};
/* If cross-fading, don't do more samples than there are to fade. */
if(FADE_SAMPLES-fadeCount > 0)
{
@@ -1339,14 +1339,14 @@ void ReverbState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesI
/* If this is not the final update, ensure the update size is a
* multiple of 4 for the SIMD mixers.
*/
- if(todo < SamplesToDo-base)
+ if(todo < samplesToDo-base)
todo &= ~3;
/* Convert B-Format to A-Format for processing. */
for(ALsizei c{0};c < NUM_LINES;c++)
{
std::fill(std::begin(afmt[c]), std::end(afmt[c]), 0.0f);
- MixRowSamples(afmt[c], mInputConv[c], SamplesIn, mNumInputs, base, todo);
+ MixRowSamples(afmt[c], mInputConv[c], samplesIn, numInput, base, todo);
}
/* Process the samples for reverb. */
@@ -1370,18 +1370,14 @@ void ReverbState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesI
* to B-Format.
*/
for(ALsizei c{0};c < NUM_LINES;c++)
- MixSamples(samples[c], NumChannels, SamplesOut,
- mEarly.CurrentGain[c], mEarly.PanGain[c],
- SamplesToDo-base, base, todo
- );
+ MixSamples(samples[c], numOutput, samplesOut, mEarly.CurrentGain[c],
+ mEarly.PanGain[c], samplesToDo-base, base, todo);
/* Generate and mix late reverb. */
LateReverb_Faded(this, offset, todo, fade, samples);
for(ALsizei c{0};c < NUM_LINES;c++)
- MixSamples(samples[c], NumChannels, SamplesOut,
- mLate.CurrentGain[c], mLate.PanGain[c],
- SamplesToDo-base, base, todo
- );
+ MixSamples(samples[c], numOutput, samplesOut, mLate.CurrentGain[c],
+ mLate.PanGain[c], samplesToDo-base, base, todo);
/* Step fading forward. */
fadeCount += todo;
@@ -1410,18 +1406,14 @@ void ReverbState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesI
/* Generate and mix early reflections. */
EarlyReflection_Unfaded(this, offset, todo, samples);
for(ALsizei c{0};c < NUM_LINES;c++)
- MixSamples(samples[c], NumChannels, SamplesOut,
- mEarly.CurrentGain[c], mEarly.PanGain[c],
- SamplesToDo-base, base, todo
- );
+ MixSamples(samples[c], numOutput, samplesOut, mEarly.CurrentGain[c],
+ mEarly.PanGain[c], samplesToDo-base, base, todo);
/* Generate and mix late reverb. */
LateReverb_Unfaded(this, offset, todo, samples);
for(ALsizei c{0};c < NUM_LINES;c++)
- MixSamples(samples[c], NumChannels, SamplesOut,
- mLate.CurrentGain[c], mLate.PanGain[c],
- SamplesToDo-base, base, todo
- );
+ MixSamples(samples[c], numOutput, samplesOut, mLate.CurrentGain[c],
+ mLate.PanGain[c], samplesToDo-base, base, todo);
}
/* Step all delays forward. */
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-04 03:57:14 +0000