/** * OpenAL cross platform audio library * Copyright (C) 2009 by Chris Robinson. * 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 "alFilter.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" typedef struct ALmodulatorState { DERIVE_FROM_TYPE(ALeffectState); enum { SINUSOID, SAWTOOTH, SQUARE } Waveform; ALuint index; ALuint step; ALfloat Gain[MaxChannels]; ALfilterState Filter; } ALmodulatorState; #define WAVEFORM_FRACBITS 24 #define WAVEFORM_FRACONE (1<> (WAVEFORM_FRACBITS - 1)) & 1); } #define DECL_TEMPLATE(func) \ static void Process##func(ALmodulatorState *state, ALuint SamplesToDo, \ const ALfloat *restrict SamplesIn, \ ALfloat (*restrict SamplesOut)[BUFFERSIZE]) \ { \ const ALuint step = state->step; \ ALuint index = state->index; \ ALuint base; \ \ for(base = 0;base < SamplesToDo;) \ { \ ALfloat temps[64]; \ ALuint td = minu(SamplesToDo-base, 64); \ ALuint i, k; \ \ for(i = 0;i < td;i++) \ { \ ALfloat samp; \ samp = SamplesIn[base+i]; \ samp = ALfilterState_processSingle(&state->Filter, samp); \ \ index += step; \ index &= WAVEFORM_FRACMASK; \ temps[i] = samp * func(index); \ } \ \ for(k = 0;k < MaxChannels;k++) \ { \ ALfloat gain = state->Gain[k]; \ if(!(gain > GAIN_SILENCE_THRESHOLD)) \ continue; \ \ for(i = 0;i < td;i++) \ SamplesOut[k][base+i] += gain * temps[i]; \ } \ \ base += td; \ } \ state->index = index; \ } DECL_TEMPLATE(Sin) DECL_TEMPLATE(Saw) DECL_TEMPLATE(Square) #undef DECL_TEMPLATE static ALvoid ALmodulatorState_Destruct(ALmodulatorState *UNUSED(state)) { } static ALboolean ALmodulatorState_deviceUpdate(ALmodulatorState *UNUSED(state), ALCdevice *UNUSED(device)) { return AL_TRUE; } static ALvoid ALmodulatorState_update(ALmodulatorState *state, ALCdevice *Device, const ALeffectslot *Slot) { ALfloat gain, cw, a; if(Slot->EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SINUSOID) state->Waveform = SINUSOID; else if(Slot->EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH) state->Waveform = SAWTOOTH; else if(Slot->EffectProps.Modulator.Waveform == AL_RING_MODULATOR_SQUARE) state->Waveform = SQUARE; state->step = fastf2u(Slot->EffectProps.Modulator.Frequency*WAVEFORM_FRACONE / Device->Frequency); if(state->step == 0) state->step = 1; /* Custom filter coeffs, which match the old version instead of a low-shelf. */ cw = cosf(F_2PI * Slot->EffectProps.Modulator.HighPassCutoff / Device->Frequency); a = (2.0f-cw) - sqrtf(powf(2.0f-cw, 2.0f) - 1.0f); state->Filter.b[0] = a; state->Filter.b[1] = -a; state->Filter.b[2] = 0.0f; state->Filter.a[0] = 1.0f; state->Filter.a[1] = -a; state->Filter.a[2] = 0.0f; gain = 1.0f/Device->NumSpeakers * Slot->Gain; SetGains(Device, gain, state->Gain); } static ALvoid ALmodulatorState_process(ALmodulatorState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE]) { switch(state->Waveform) { case SINUSOID: ProcessSin(state, SamplesToDo, SamplesIn, SamplesOut); break; case SAWTOOTH: ProcessSaw(state, SamplesToDo, SamplesIn, SamplesOut); break; case SQUARE: ProcessSquare(state, SamplesToDo, SamplesIn, SamplesOut); break; } } DECLARE_DEFAULT_ALLOCATORS(ALmodulatorState) DEFINE_ALEFFECTSTATE_VTABLE(ALmodulatorState); typedef struct ALmodulatorStateFactory { DERIVE_FROM_TYPE(ALeffectStateFactory); } ALmodulatorStateFactory; static ALeffectState *ALmodulatorStateFactory_create(ALmodulatorStateFactory *UNUSED(factory)) { ALmodulatorState *state; state = ALmodulatorState_New(sizeof(*state)); if(!state) return NULL; SET_VTABLE2(ALmodulatorState, ALeffectState, state); state->index = 0; state->step = 1; ALfilterState_clear(&state->Filter); return STATIC_CAST(ALeffectState, state); } DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALmodulatorStateFactory); ALeffectStateFactory *ALmodulatorStateFactory_getFactory(void) { static ALmodulatorStateFactory ModulatorFactory = { { GET_VTABLE2(ALmodulatorStateFactory, ALeffectStateFactory) } }; return STATIC_CAST(ALeffectStateFactory, &ModulatorFactory); } void ALmodulator_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_RING_MODULATOR_FREQUENCY: if(!(val >= AL_RING_MODULATOR_MIN_FREQUENCY && val <= AL_RING_MODULATOR_MAX_FREQUENCY)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Modulator.Frequency = val; break; case AL_RING_MODULATOR_HIGHPASS_CUTOFF: if(!(val >= AL_RING_MODULATOR_MIN_HIGHPASS_CUTOFF && val <= AL_RING_MODULATOR_MAX_HIGHPASS_CUTOFF)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Modulator.HighPassCutoff = val; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALmodulator_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALmodulator_setParamf(effect, context, param, vals[0]); } void ALmodulator_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_RING_MODULATOR_FREQUENCY: case AL_RING_MODULATOR_HIGHPASS_CUTOFF: ALmodulator_setParamf(effect, context, param, (ALfloat)val); break; case AL_RING_MODULATOR_WAVEFORM: if(!(val >= AL_RING_MODULATOR_MIN_WAVEFORM && val <= AL_RING_MODULATOR_MAX_WAVEFORM)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Modulator.Waveform = val; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALmodulator_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALmodulator_setParami(effect, context, param, vals[0]); } void ALmodulator_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_RING_MODULATOR_FREQUENCY: *val = (ALint)props->Modulator.Frequency; break; case AL_RING_MODULATOR_HIGHPASS_CUTOFF: *val = (ALint)props->Modulator.HighPassCutoff; break; case AL_RING_MODULATOR_WAVEFORM: *val = props->Modulator.Waveform; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALmodulator_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALmodulator_getParami(effect, context, param, vals); } void ALmodulator_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_RING_MODULATOR_FREQUENCY: *val = props->Modulator.Frequency; break; case AL_RING_MODULATOR_HIGHPASS_CUTOFF: *val = props->Modulator.HighPassCutoff; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALmodulator_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALmodulator_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALmodulator);