/**
 * 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 <math.h>
#include <stdlib.h>

#include "alMain.h"
#include "alFilter.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"

/* Filters implementation is based on the "Cookbook formulae for audio   *
 * EQ biquad filter coefficients" by Robert Bristow-Johnson              *
 * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt                   */

typedef enum ALEQFilterType {
    LOWPASS,
    BANDPASS,
} ALEQFilterType;

typedef struct ALEQFilter {
    ALEQFilterType type;
    ALfloat x[2]; /* History of two last input samples  */
    ALfloat y[2]; /* History of two last output samples */
    ALfloat a[3]; /* Transfer function coefficients "a" */
    ALfloat b[3]; /* Transfer function coefficients "b" */
} ALEQFilter;

typedef struct ALdistortionState {
    /* Must be first in all effects! */
    ALeffectState state;

    /* Effect gains for each channel */
    ALfloat Gain[MaxChannels];

    /* Effect parameters */
    ALEQFilter bandpass;
    ALEQFilter lowpass;
    ALfloat frequency;
    ALfloat attenuation;
    ALfloat edge_coeff;

    /* Oversample data */
    ALfloat oversample_buffer[BUFFERSIZE][4];
} ALdistortionState;

static ALvoid DistortionDestroy(ALeffectState *effect)
{
    ALdistortionState *state = (ALdistortionState*)effect;

    free(state);
}

static ALboolean DistortionDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
{
    ALdistortionState *state = (ALdistortionState*)effect;

    state->frequency = (ALfloat)Device->Frequency;

    return AL_TRUE;
}

static ALvoid DistortionUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
{
    ALdistortionState *state = (ALdistortionState*)effect;
    ALfloat gain = sqrtf(1.0f / Device->NumChan) * Slot->Gain;
    ALuint it;
    ALfloat w0;
    ALfloat alpha;
    ALfloat bandwidth;
    ALfloat cutoff;
    ALfloat edge;

    for(it = 0;it < MaxChannels;it++)
        state->Gain[it] = 0.0f;
    for(it = 0;it < Device->NumChan;it++)
    {
        enum Channel chan = Device->Speaker2Chan[it];
        state->Gain[chan] = gain;
    }

    /* Store distorted signal attenuation settings */
    state->attenuation = Slot->effect.Distortion.Gain;

    /* Store waveshaper edge settings */
    edge = sinf(Slot->effect.Distortion.Edge * (F_PI/2.0f));
    state->edge_coeff = 2.0f * edge / (1.0f-edge);

    /* Lowpass filter */
    cutoff = Slot->effect.Distortion.LowpassCutoff;
    /* Bandwidth value is constant in octaves */
    bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
    w0 = 2.0f * F_PI * cutoff / (state->frequency * 4.0f);
    alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
    state->lowpass.b[0] = (1.0f - cosf(w0)) / 2.0f;
    state->lowpass.b[1] = 1.0f - cosf(w0);
    state->lowpass.b[2] = (1.0f - cosf(w0)) / 2.0f;
    state->lowpass.a[0] = 1.0f + alpha;
    state->lowpass.a[1] = -2.0f * cosf(w0);
    state->lowpass.a[2] = 1.0f - alpha;

    /* Bandpass filter */
    cutoff = Slot->effect.Distortion.EQCenter;
    /* Convert bandwidth in Hz to octaves */
    bandwidth = Slot->effect.Distortion.EQBandwidth / (cutoff * 0.67f);
    w0 = 2.0f * F_PI * cutoff / (state->frequency * 4.0f);
    alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
    state->bandpass.b[0] = alpha;
    state->bandpass.b[1] = 0;
    state->bandpass.b[2] = -alpha;
    state->bandpass.a[0] = 1.0f + alpha;
    state->bandpass.a[1] = -2.0f * cosf(w0);
    state->bandpass.a[2] = 1.0f - alpha;
}

static ALvoid DistortionProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
{
    ALdistortionState *state = (ALdistortionState*)effect;
    float *RESTRICT oversample_buffer = &state->oversample_buffer[0][0];
    ALfloat tempsmp;
    ALuint it;
    ALuint kt;

    /* 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.   */

    /* Fill oversample buffer using zero stuffing */
    for(it = 0; it < SamplesToDo; it++)
    {
        oversample_buffer[it*4 + 0] = SamplesIn[it];
        oversample_buffer[it*4 + 1] = 0.0f;
        oversample_buffer[it*4 + 2] = 0.0f;
        oversample_buffer[it*4 + 3] = 0.0f;
    }

    /* First step, do lowpass filtering of original signal,  */
    /* additionally perform buffer interpolation and lowpass */
    /* cutoff for oversampling (which is fortunately first   */
    /* step of distortion). So combine three operations into */
    /* the one.                                              */
    for(it = 0; it < SamplesToDo * 4; it++)
    {
        tempsmp = state->lowpass.b[0] / state->lowpass.a[0] * oversample_buffer[it] +
                  state->lowpass.b[1] / state->lowpass.a[0] * state->lowpass.x[0] +
                  state->lowpass.b[2] / state->lowpass.a[0] * state->lowpass.x[1] -
                  state->lowpass.a[1] / state->lowpass.a[0] * state->lowpass.y[0] -
                  state->lowpass.a[2] / state->lowpass.a[0] * state->lowpass.y[1];

        state->lowpass.x[1] = state->lowpass.x[0];
        state->lowpass.x[0] = oversample_buffer[it];
        state->lowpass.y[1] = state->lowpass.y[0];
        state->lowpass.y[0] = tempsmp;
        /* Restore signal power by multiplying sample by amount of oversampling */
        oversample_buffer[it] = tempsmp * 4.0f;
    }

    for(it = 0; it < SamplesToDo * 4; it++)
    {
        ALfloat smp = oversample_buffer[it];
        ALfloat fc = state->edge_coeff;

        /* Second step, do distortion using waveshaper function  */
        /* to emulate signal processing during tube overdriving. */
        /* Three steps of waveshaping are intended to modify     */
        /* waveform without boost/clipping/attenuation process.  */
        smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
        smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f;
        smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));

        /* Third step, do bandpass filtering of distorted signal */
        tempsmp = state->bandpass.b[0] / state->bandpass.a[0] * smp +
                  state->bandpass.b[1] / state->bandpass.a[0] * state->bandpass.x[0] +
                  state->bandpass.b[2] / state->bandpass.a[0] * state->bandpass.x[1] -
                  state->bandpass.a[1] / state->bandpass.a[0] * state->bandpass.y[0] -
                  state->bandpass.a[2] / state->bandpass.a[0] * state->bandpass.y[1];

        state->bandpass.x[1] = state->bandpass.x[0];
        state->bandpass.x[0] = smp;
        state->bandpass.y[1] = state->bandpass.y[0];
        state->bandpass.y[0] = tempsmp;
        smp = tempsmp;

        /* Fourth step, final, do attenuation and perform decimation, */
        /* store only one sample out of 4.                            */
        if(!(it&3))
        {
            smp *= state->attenuation;
            for(kt = 0; kt < MaxChannels; kt++)
                SamplesOut[kt][it>>2] += state->Gain[kt] * smp;
        }
    }
}

ALeffectState *DistortionCreate(void)
{
    ALdistortionState *state;

    state = malloc(sizeof(*state));
    if(!state)
        return NULL;

    state->state.Destroy = DistortionDestroy;
    state->state.DeviceUpdate = DistortionDeviceUpdate;
    state->state.Update = DistortionUpdate;
    state->state.Process = DistortionProcess;

    state->bandpass.type = BANDPASS;
    state->lowpass.type = LOWPASS;

    /* Initialize sample history only on filter creation to avoid */
    /* sound clicks if filter settings were changed in runtime.   */
    state->bandpass.x[0] = 0.0f;
    state->bandpass.x[1] = 0.0f;
    state->lowpass.y[0] = 0.0f;
    state->lowpass.y[1] = 0.0f;

    return &state->state;
}

void distortion_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
    effect=effect;
    val=val;

    switch(param)
    {
        default:
            alSetError(context, AL_INVALID_ENUM);
            break;
    }
}
void distortion_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{
    distortion_SetParami(effect, context, param, vals[0]);
}
void distortion_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
    switch(param)
    {
        case AL_DISTORTION_EDGE:
            if(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE)
                effect->Distortion.Edge = val;
            else
                alSetError(context, AL_INVALID_VALUE);
            break;

        case AL_DISTORTION_GAIN:
            if(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN)
                effect->Distortion.Gain = val;
            else
                alSetError(context, AL_INVALID_VALUE);
            break;

        case AL_DISTORTION_LOWPASS_CUTOFF:
            if(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF)
                effect->Distortion.LowpassCutoff = val;
            else
                alSetError(context, AL_INVALID_VALUE);
            break;

        case AL_DISTORTION_EQCENTER:
            if(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER)
                effect->Distortion.EQCenter = val;
            else
                alSetError(context, AL_INVALID_VALUE);
            break;

        case AL_DISTORTION_EQBANDWIDTH:
            if(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH)
                effect->Distortion.EQBandwidth = val;
            else
                alSetError(context, AL_INVALID_VALUE);
            break;

        default:
            alSetError(context, AL_INVALID_ENUM);
            break;
    }
}
void distortion_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{
    distortion_SetParamf(effect, context, param, vals[0]);
}

void distortion_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
{
    effect=effect;
    val=val;

    switch(param)
    {
        default:
            alSetError(context, AL_INVALID_ENUM);
            break;
    }
}
void distortion_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{
    distortion_GetParami(effect, context, param, vals);
}
void distortion_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
    switch(param)
    {
        case AL_DISTORTION_EDGE:
            *val = effect->Distortion.Edge;
            break;

        case AL_DISTORTION_GAIN:
            *val = effect->Distortion.Gain;
            break;

        case AL_DISTORTION_LOWPASS_CUTOFF:
            *val = effect->Distortion.LowpassCutoff;
            break;

        case AL_DISTORTION_EQCENTER:
            *val = effect->Distortion.EQCenter;
            break;

        case AL_DISTORTION_EQBANDWIDTH:
            *val = effect->Distortion.EQBandwidth;
            break;

        default:
            alSetError(context, AL_INVALID_ENUM);
            break;
    }
}
void distortion_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{
    distortion_GetParamf(effect, context, param, vals);
}