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Inline the autowah peaking filter processing

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This commit is contained in:
Chris Robinson
2018-07-28 13:48:53 -07:00
parent 39a5d0cb94
commit 8ab448b119
1 changed files with 72 additions and 37 deletions
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Alc/effects/autowah.c
+72 -37
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@@ -45,17 +45,23 @@ typedef struct ALautowahState {
ALfloat BandwidthNorm;
ALfloat env_delay;
/* Filter components derived from the envelope. */
ALfloat Alpha[BUFFERSIZE];
ALfloat CosW0[BUFFERSIZE];
struct {
/* Effect filters' history. */
struct {
ALfloat z1, z2;
} Filter;
/* Effect gains for each output channel */
ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
/* Effect filters */
BiquadFilter Filter;
} Chans[MAX_EFFECT_CHANNELS];
/*Effects buffers*/
alignas(16) ALfloat BufferOut[MAX_EFFECT_CHANNELS][BUFFERSIZE];
/* Effects buffers */
alignas(16) ALfloat BufferOut[BUFFERSIZE];
} ALautowahState;
static ALvoid ALautowahState_Destruct(ALautowahState *state);
@@ -66,18 +72,6 @@ DECLARE_DEFAULT_ALLOCATORS(ALautowahState)
DEFINE_ALEFFECTSTATE_VTABLE(ALautowahState);
/*Envelope follewer described on the book: Audio Effects, Theory, Implementation and Application*/
static inline ALfloat envelope_follower(ALautowahState *state, ALfloat SampleIn)
{
ALfloat alpha, Sample;
Sample = state->PeakGain*fabsf(SampleIn);
alpha = (Sample > state->env_delay) ? state->AttackRate : state->ReleaseRate;
state->env_delay = alpha*state->env_delay + (1.0f-alpha)*Sample;
return state->env_delay;
}
static void ALautowahState_Construct(ALautowahState *state)
{
ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
@@ -104,9 +98,10 @@ static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *U
for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
{
BiquadFilter_clear(&state->Chans[i].Filter);
for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
state->Chans[i].CurrentGains[j] = 0.0f;
state->Chans[i].Filter.z1 = 0.0f;
state->Chans[i].Filter.z2 = 0.0f;
}
return AL_TRUE;
@@ -116,16 +111,17 @@ static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *con
{
const ALCdevice *device = context->Device;
ALfloat ReleaseTime;
ALuint i;
ALsizei i;
ReleaseTime = clampf(props->Autowah.ReleaseTime,0.001f,1.0f);
ReleaseTime = clampf(props->Autowah.ReleaseTime, 0.001f, 1.0f);
state->AttackRate = expf(-1.0f/(props->Autowah.AttackTime*device->Frequency));
state->ReleaseRate = expf(-1.0f/(ReleaseTime*device->Frequency));
state->ResonanceGain = 10.0f/3.0f*log10f(props->Autowah.Resonance);/*0-20dB Resonance Peak gain*/
state->PeakGain = 1.0f -log10f(props->Autowah.PeakGain/AL_AUTOWAH_MAX_PEAK_GAIN);
state->FreqMinNorm = MIN_FREQ/device->Frequency;
state->BandwidthNorm = (MAX_FREQ - MIN_FREQ)/device->Frequency;
state->AttackRate = expf(-1.0f / (props->Autowah.AttackTime*device->Frequency));
state->ReleaseRate = expf(-1.0f / (ReleaseTime*device->Frequency));
/* 0-20dB Resonance Peak gain */
state->ResonanceGain = log10f(props->Autowah.Resonance)*10.0f / 3.0f;
state->PeakGain = 1.0f - log10f(props->Autowah.PeakGain/AL_AUTOWAH_MAX_PEAK_GAIN);
state->FreqMinNorm = MIN_FREQ / device->Frequency;
state->BandwidthNorm = (MAX_FREQ-MIN_FREQ) / device->Frequency;
STATIC_CAST(ALeffectState,state)->OutBuffer = device->FOAOut.Buffer;
STATIC_CAST(ALeffectState,state)->OutChannels = device->FOAOut.NumChannels;
@@ -136,31 +132,70 @@ static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *con
static ALvoid ALautowahState_process(ALautowahState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
ALfloat (*restrict BufferOut)[BUFFERSIZE] = state->BufferOut;
ALfloat f0norm[BUFFERSIZE];
const ALfloat peak_gain = state->PeakGain;
const ALfloat attack_rate = state->AttackRate;
const ALfloat release_rate = state->ReleaseRate;
const ALfloat freq_min = state->FreqMinNorm;
const ALfloat bandwidth = state->BandwidthNorm;
ALfloat env_delay;
ALsizei c, i;
env_delay = state->env_delay;
for(i = 0;i < SamplesToDo;i++)
{
ALfloat env_out;
env_out = envelope_follower(state, SamplesIn[0][i]);
f0norm[i] = state->BandwidthNorm*env_out + state->FreqMinNorm;
ALfloat w0, sample, a;
/* Envelope follower described on the book: Audio Effects, Theory,
* Implementation and Application.
*/
sample = peak_gain * fabsf(SamplesIn[0][i]);
a = (sample > env_delay) ? attack_rate : release_rate;
env_delay = lerp(sample, env_delay, a);
/* Calculate the cos and alpha components for this sample's filter. */
w0 = (bandwidth*env_delay + freq_min) * F_TAU;
state->CosW0[i] = cosf(w0);
state->Alpha[i] = sinf(w0)/(2.0f * Q_FACTOR);
}
state->env_delay = env_delay;
for(c = 0;c < MAX_EFFECT_CHANNELS; c++)
{
/* This effectively inlines BiquadFilter_setParams for a peaking
* filter and BiquadFilter_processC. The alpha and cosine components
* for the filter coefficients were previously calculated with the
* envelope. Because the filter changes for each sample, the
* coefficients are transient and don't need to be held.
*/
const ALfloat res_gain = sqrtf(state->ResonanceGain);
ALfloat z1 = state->Chans[c].Filter.z1;
ALfloat z2 = state->Chans[c].Filter.z2;
for(i = 0;i < SamplesToDo;i++)
{
ALfloat temp;
const ALfloat alpha = state->Alpha[i];
const ALfloat cos_w0 = state->CosW0[i];
ALfloat input, output;
ALfloat a[3], b[3];
BiquadFilter_setParams(&state->Chans[c].Filter, BiquadType_Peaking,
state->ResonanceGain, f0norm[i], 1.0f/Q_FACTOR);
BiquadFilter_process(&state->Chans[c].Filter, &temp, &SamplesIn[c][i], 1);
b[0] = 1.0f + alpha*res_gain;
b[1] = -2.0f * cos_w0;
b[2] = 1.0f - alpha*res_gain;
a[0] = 1.0f + alpha/res_gain;
a[1] = -2.0f * cos_w0;
a[2] = 1.0f - alpha/res_gain;
BufferOut[c][i] = temp;
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]);
state->BufferOut[i] = output;
}
state->Chans[c].Filter.z1 = z1;
state->Chans[c].Filter.z2 = z2;
/* Now, mix the processed sound data to the output. */
MixSamples(BufferOut[c], NumChannels, SamplesOut, state->Chans[c].CurrentGains,
MixSamples(state->BufferOut, NumChannels, SamplesOut, state->Chans[c].CurrentGains,
state->Chans[c].TargetGains, SamplesToDo, 0, SamplesToDo);
}
}