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Pass the number of input channels to EffectState::process

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This commit is contained in:
Chris Robinson 2019-02-21 02:57:39 -08:00
parent 7e00f646d9
commit 4b4041319d
14 changed files with 105 additions and 108 deletions
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2
Alc/alu.cpp
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@ -1529,7 +1529,7 @@ void ProcessContext(ALCcontext *ctx, const ALsizei SamplesToDo)
EffectState *state{slot->Params.mEffectState};
state->process(SamplesToDo,
&reinterpret_cast<const ALfloat(&)[BUFFERSIZE]>(slot->WetBuffer[0]),
state->mOutBuffer, state->mOutChannels);
slot->WetBuffer.size(), state->mOutBuffer, state->mOutChannels);
}
);
}

19
Alc/effects/autowah.cpp
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@ -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);
}
}

14
Alc/effects/chorus.cpp
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@ -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;
}

26
Alc/effects/compressor.cpp
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@ -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;
}
}

8
Alc/effects/dedicated.cpp
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@ -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);
}

18
Alc/effects/distortion.cpp
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@ -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;

14
Alc/effects/echo.cpp
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@ -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;
}

19
Alc/effects/equalizer.cpp
View File

@ -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);
}
}

18
Alc/effects/fshifter.cpp
View File

@ -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

17
Alc/effects/modulator.cpp
View File

@ -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;

4
Alc/effects/null.cpp
View File

@ -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*/)
{
}

14
Alc/effects/pshifter.cpp
View File

@ -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

38
Alc/effects/reverb.cpp
View File

@ -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. */

2
OpenAL32/Include/alAuxEffectSlot.h
View File

@ -30,7 +30,7 @@ struct EffectState {
virtual ALboolean deviceUpdate(const ALCdevice *device) = 0;
virtual void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) = 0;
virtual void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) = 0;
virtual void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) = 0;
void IncRef() noexcept;
void DecRef() noexcept;