/** * OpenAL cross platform audio library * Copyright (C) 2013 by Mike Gorchak * 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, 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 ALchorusState { DERIVE_FROM_TYPE(ALeffectState); ALfloat *SampleBufferLeft; ALfloat *SampleBufferRight; ALuint BufferLength; ALint offset; ALfloat lfo_coeff; ALint lfo_disp; /* Gains for left and right sides */ ALfloat Gain[2][MaxChannels]; /* effect parameters */ ALint waveform; ALint phase; ALfloat rate; ALfloat depth; ALfloat feedback; ALfloat delay; ALfloat frequency; } ALchorusState; static ALvoid ChorusDestroy(ALeffectState *effect) { ALchorusState *state = GET_PARENT_TYPE(ALchorusState, ALeffectState, effect); if(state) { free(state->SampleBufferLeft); state->SampleBufferLeft = NULL; free(state->SampleBufferRight); state->SampleBufferRight = NULL; free(state); } } static ALboolean ChorusDeviceUpdate(ALeffectState *effect, ALCdevice *Device) { ALchorusState *state = GET_PARENT_TYPE(ALchorusState, ALeffectState, effect); ALuint maxlen; ALuint it; maxlen = fastf2u(AL_CHORUS_MAX_DELAY * 3.0f * Device->Frequency) + 1; maxlen = NextPowerOf2(maxlen); if (maxlen != state->BufferLength) { void *temp; temp = realloc(state->SampleBufferLeft, maxlen * sizeof(ALfloat)); if (!temp) { return AL_FALSE; } state->SampleBufferLeft = temp; temp = realloc(state->SampleBufferRight, maxlen * sizeof(ALfloat)); if (!temp) { return AL_FALSE; } state->SampleBufferRight = temp; state->BufferLength = maxlen; } for (it = 0; it < state->BufferLength; it++) { state->SampleBufferLeft[it] = 0.0f; state->SampleBufferRight[it] = 0.0f; } state->frequency=(ALfloat)Device->Frequency; return AL_TRUE; } static ALvoid ChorusUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot) { ALchorusState *state = GET_PARENT_TYPE(ALchorusState, ALeffectState, effect); ALuint it; for (it = 0; it < MaxChannels; it++) { state->Gain[0][it] = 0.0f; state->Gain[1][it] = 0.0f; } state->waveform = Slot->effect.Chorus.Waveform; state->phase = Slot->effect.Chorus.Phase; state->rate = Slot->effect.Chorus.Rate; state->depth = Slot->effect.Chorus.Depth; state->feedback = Slot->effect.Chorus.Feedback; state->delay = Slot->effect.Chorus.Delay; state->frequency=(ALfloat)Device->Frequency; /* Gains for left and right sides */ ComputeAngleGains(Device, atan2f(-1.0f, 0.0f), 0.0f, Slot->Gain, state->Gain[0]); ComputeAngleGains(Device, atan2f(+1.0f, 0.0f), 0.0f, Slot->Gain, state->Gain[1]); /* Calculate LFO coefficient */ switch (state->waveform) { case AL_CHORUS_WAVEFORM_TRIANGLE: if (state->rate == 0.0f) { state->lfo_coeff = 0.0f; } else { state->lfo_coeff = 1.0f / ((ALfloat)Device->Frequency / state->rate); } break; case AL_CHORUS_WAVEFORM_SINUSOID: if (state->rate == 0.0f) { state->lfo_coeff = 0.0f; } else { state->lfo_coeff = F_PI * 2.0f / ((ALfloat)Device->Frequency / state->rate); } break; } /* Calculate lfo phase displacement */ if ((state->phase == 0) || (state->rate == 0.0f)) { state->lfo_disp = 0; } else { state->lfo_disp = (ALint) ((ALfloat)Device->Frequency / state->rate / (360.0f / (ALfloat)state->phase)); } } static ALvoid ChorusProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE]) { ALchorusState *state = GET_PARENT_TYPE(ALchorusState, ALeffectState, effect); const ALuint mask = state->BufferLength-1; ALuint it; ALuint kt; ALint offset; ALfloat lfo_value_left = 0.0f; ALfloat lfo_value_right = 0.0f; ALint delay_left = 0; ALint delay_right = 0; ALfloat smp; offset=state->offset; switch (state->waveform) { case AL_CHORUS_WAVEFORM_TRIANGLE: for (it = 0; it < SamplesToDo; it++, offset++) { lfo_value_left = 2.0f - fabsf(2.0f - fmodf(state->lfo_coeff * offset * 4.0f, 4.0f)); lfo_value_left *= state->depth * state->delay; lfo_value_left += state->delay; delay_left = (ALint)(lfo_value_left * state->frequency); lfo_value_right = 2.0f - fabsf(2.0f - fmodf(state->lfo_coeff * (offset + state->lfo_disp) * 4.0f, 4.0f)); lfo_value_right *= state->depth * state->delay; lfo_value_right += state->delay; delay_right = (ALint)(lfo_value_right * state->frequency); smp = state->SampleBufferLeft[(offset-delay_left) & mask]; for (kt = 0; kt < MaxChannels; kt++) { SamplesOut[kt][it] += smp * state->Gain[0][kt]; } state->SampleBufferLeft[offset & mask] = (smp + SamplesIn[it]) * state->feedback; smp = state->SampleBufferRight[(offset-delay_right) & mask]; for (kt = 0; kt < MaxChannels; kt++) { SamplesOut[kt][it] += smp * state->Gain[1][kt]; } state->SampleBufferRight[offset & mask] = (smp + SamplesIn[it]) * state->feedback; } break; case AL_CHORUS_WAVEFORM_SINUSOID: for (it = 0; it < SamplesToDo; it++, offset++) { lfo_value_left = 1.0f + sinf(fmodf(state->lfo_coeff * offset, 2 * F_PI)); lfo_value_left *= state->depth * state->delay; lfo_value_left += state->delay; delay_left = (ALint)(lfo_value_left * state->frequency); lfo_value_right = 1.0f + sinf(fmodf(state->lfo_coeff * (offset + state->lfo_disp), 2 * F_PI)); lfo_value_right *= state->depth * state->delay; lfo_value_right += state->delay; delay_right = (ALint)(lfo_value_right * state->frequency); smp = state->SampleBufferLeft[(offset-delay_left) & mask]; for (kt = 0; kt < MaxChannels; kt++) { SamplesOut[kt][it] += smp * state->Gain[0][kt]; } state->SampleBufferLeft[offset & mask] = (smp + SamplesIn[it]) * state->feedback; smp = state->SampleBufferRight[(offset-delay_right) & mask]; for (kt = 0; kt < MaxChannels; kt++) { SamplesOut[kt][it] += smp * state->Gain[1][kt]; } state->SampleBufferRight[offset & mask] = (smp + SamplesIn[it]) * state->feedback; } break; } state->offset=offset; } ALeffectState *ChorusCreate(void) { ALchorusState *state; state = malloc(sizeof(*state)); if(!state) return NULL; GET_DERIVED_TYPE(ALeffectState, state)->Destroy = ChorusDestroy; GET_DERIVED_TYPE(ALeffectState, state)->DeviceUpdate = ChorusDeviceUpdate; GET_DERIVED_TYPE(ALeffectState, state)->Update = ChorusUpdate; GET_DERIVED_TYPE(ALeffectState, state)->Process = ChorusProcess; state->BufferLength = 0; state->SampleBufferLeft = NULL; state->SampleBufferRight = NULL; state->offset = 0; return GET_DERIVED_TYPE(ALeffectState, state); } void chorus_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { switch(param) { case AL_CHORUS_WAVEFORM: if(val >= AL_CHORUS_MIN_WAVEFORM && val <= AL_CHORUS_MAX_WAVEFORM) effect->Chorus.Waveform = val; else alSetError(context, AL_INVALID_VALUE); break; case AL_CHORUS_PHASE: if(val >= AL_CHORUS_MIN_PHASE && val <= AL_CHORUS_MAX_PHASE) effect->Chorus.Phase = val; else alSetError(context, AL_INVALID_VALUE); break; default: alSetError(context, AL_INVALID_ENUM); break; } } void chorus_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { chorus_SetParami(effect, context, param, vals[0]); } void chorus_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { switch(param) { case AL_CHORUS_RATE: if(val >= AL_CHORUS_MIN_RATE && val <= AL_CHORUS_MAX_RATE) effect->Chorus.Rate = val; else alSetError(context, AL_INVALID_VALUE); break; case AL_CHORUS_DEPTH: if(val >= AL_CHORUS_MIN_DEPTH && val <= AL_CHORUS_MAX_DEPTH) effect->Chorus.Depth = val; else alSetError(context, AL_INVALID_VALUE); break; case AL_CHORUS_FEEDBACK: if(val >= AL_CHORUS_MIN_FEEDBACK && val <= AL_CHORUS_MAX_FEEDBACK) effect->Chorus.Feedback = val; else alSetError(context, AL_INVALID_VALUE); break; case AL_CHORUS_DELAY: if(val >= AL_CHORUS_MIN_DELAY && val <= AL_CHORUS_MAX_DELAY) effect->Chorus.Delay = val; else alSetError(context, AL_INVALID_VALUE); break; default: alSetError(context, AL_INVALID_ENUM); break; } } void chorus_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { chorus_SetParamf(effect, context, param, vals[0]); } void chorus_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { switch(param) { case AL_CHORUS_WAVEFORM: *val = effect->Chorus.Waveform; break; case AL_CHORUS_PHASE: *val = effect->Chorus.Phase; break; default: alSetError(context, AL_INVALID_ENUM); break; } } void chorus_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { chorus_GetParami(effect, context, param, vals); } void chorus_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { switch(param) { case AL_CHORUS_RATE: *val = effect->Chorus.Rate; break; case AL_CHORUS_DEPTH: *val = effect->Chorus.Depth; break; case AL_CHORUS_FEEDBACK: *val = effect->Chorus.Feedback; break; case AL_CHORUS_DELAY: *val = effect->Chorus.Delay; break; default: alSetError(context, AL_INVALID_ENUM); break; } } void chorus_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { chorus_GetParamf(effect, context, param, vals); }