/** * OpenAL cross platform audio library * Copyright (C) 2018 by Raul Herraiz. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include "alMain.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" #include "filters/defs.h" #define MIN_FREQ 20.0f #define MAX_FREQ 2500.0f #define Q_FACTOR 5.0f typedef struct ALautowahState { DERIVE_FROM_TYPE(ALeffectState); /* Effect parameters */ ALfloat AttackRate; ALfloat ReleaseRate; ALfloat ResonanceGain; ALfloat PeakGain; ALfloat FreqMinNorm; ALfloat BandwidthNorm; ALfloat env_delay; /* Filter components derived from the envelope. */ struct { ALfloat cos_w0; ALfloat alpha; } Env[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]; } Chans[MAX_EFFECT_CHANNELS]; /* Effects buffers */ alignas(16) ALfloat BufferOut[BUFFERSIZE]; } ALautowahState; static ALvoid ALautowahState_Destruct(ALautowahState *state); static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *device); static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props); static ALvoid ALautowahState_process(ALautowahState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels); DECLARE_DEFAULT_ALLOCATORS(ALautowahState) DEFINE_ALEFFECTSTATE_VTABLE(ALautowahState); static void ALautowahState_Construct(ALautowahState *state) { ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); SET_VTABLE2(ALautowahState, ALeffectState, state); } static ALvoid ALautowahState_Destruct(ALautowahState *state) { ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); } static ALboolean ALautowahState_deviceUpdate(ALautowahState *state, ALCdevice *UNUSED(device)) { /* (Re-)initializing parameters and clear the buffers. */ ALsizei i, j; state->AttackRate = 1.0f; state->ReleaseRate = 1.0f; state->ResonanceGain = 10.0f; state->PeakGain = 4.5f; state->FreqMinNorm = 4.5e-4f; state->BandwidthNorm = 0.05f; state->env_delay = 0.0f; memset(state->Env, 0, sizeof(state->Env)); for(i = 0;i < MAX_EFFECT_CHANNELS;i++) { 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; } static ALvoid ALautowahState_update(ALautowahState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props) { const ALCdevice *device = context->Device; ALfloat ReleaseTime; ALsizei i; 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)); /* 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; for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputeFirstOrderGains(&device->FOAOut, IdentityMatrixf.m[i], slot->Params.Gain, state->Chans[i].TargetGains); } static ALvoid ALautowahState_process(ALautowahState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels) { 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 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 = minf((bandwidth*env_delay + freq_min), 0.46f) * F_TAU; state->Env[i].cos_w0 = cosf(w0); state->Env[i].alpha = 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++) { const ALfloat alpha = state->Env[i].alpha; const ALfloat cos_w0 = state->Env[i].cos_w0; ALfloat input, output; ALfloat a[3], b[3]; 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; 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(state->BufferOut, NumChannels, SamplesOut, state->Chans[c].CurrentGains, state->Chans[c].TargetGains, SamplesToDo, 0, SamplesToDo); } } typedef struct AutowahStateFactory { DERIVE_FROM_TYPE(EffectStateFactory); } AutowahStateFactory; static ALeffectState *AutowahStateFactory_create(AutowahStateFactory *UNUSED(factory)) { ALautowahState *state; NEW_OBJ0(state, ALautowahState)(); if(!state) return NULL; return STATIC_CAST(ALeffectState, state); } DEFINE_EFFECTSTATEFACTORY_VTABLE(AutowahStateFactory); EffectStateFactory *AutowahStateFactory_getFactory(void) { static AutowahStateFactory AutowahFactory = { { GET_VTABLE2(AutowahStateFactory, EffectStateFactory) } }; return STATIC_CAST(EffectStateFactory, &AutowahFactory); } void ALautowah_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_AUTOWAH_ATTACK_TIME: if(!(val >= AL_AUTOWAH_MIN_ATTACK_TIME && val <= AL_AUTOWAH_MAX_ATTACK_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah attack time out of range"); props->Autowah.AttackTime = val; break; case AL_AUTOWAH_RELEASE_TIME: if(!(val >= AL_AUTOWAH_MIN_RELEASE_TIME && val <= AL_AUTOWAH_MAX_RELEASE_TIME)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah release time out of range"); props->Autowah.ReleaseTime = val; break; case AL_AUTOWAH_RESONANCE: if(!(val >= AL_AUTOWAH_MIN_RESONANCE && val <= AL_AUTOWAH_MAX_RESONANCE)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah resonance out of range"); props->Autowah.Resonance = val; break; case AL_AUTOWAH_PEAK_GAIN: if(!(val >= AL_AUTOWAH_MIN_PEAK_GAIN && val <= AL_AUTOWAH_MAX_PEAK_GAIN)) SETERR_RETURN(context, AL_INVALID_VALUE,,"Autowah peak gain out of range"); props->Autowah.PeakGain = val; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid autowah float property 0x%04x", param); } } void ALautowah_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALautowah_setParamf(effect, context, param, vals[0]); } void ALautowah_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer property 0x%04x", param); } void ALautowah_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer vector property 0x%04x", param); } void ALautowah_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val)) { alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer property 0x%04x", param); } void ALautowah_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals)) { alSetError(context, AL_INVALID_ENUM, "Invalid autowah integer vector property 0x%04x", param); } void ALautowah_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_AUTOWAH_ATTACK_TIME: *val = props->Autowah.AttackTime; break; case AL_AUTOWAH_RELEASE_TIME: *val = props->Autowah.ReleaseTime; break; case AL_AUTOWAH_RESONANCE: *val = props->Autowah.Resonance; break; case AL_AUTOWAH_PEAK_GAIN: *val = props->Autowah.PeakGain; break; default: alSetError(context, AL_INVALID_ENUM, "Invalid autowah float property 0x%04x", param); } } void ALautowah_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALautowah_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALautowah);