/*
 * Copyright (C) 2010 Google Inc. All rights reserved.
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 *
 * 1.  Redistributions of source code must retain the above copyright
 *     notice, this list of conditions and the following disclaimer.
 * 2.  Redistributions in binary form must reproduce the above copyright
 *     notice, this list of conditions and the following disclaimer in the
 *     documentation and/or other materials provided with the distribution.
 * 3.  Neither the name of Apple Inc. ("Apple") nor the names of
 *     its contributors may be used to endorse or promote products derived
 *     from this software without specific prior written permission.
 *
 * THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
 * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
 * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
 * DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
 * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
 * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
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 * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 */

#ifndef Biquad_h
#define Biquad_h

#include "AudioArray.h"
#include <complex>
#include <sys/types.h>

namespace WebCore {

// A basic biquad (two-zero / two-pole digital filter)
//
// It can be configured to a number of common and very useful filters:
//    lowpass, highpass, shelving, parameteric, notch, allpass, ...

class Biquad final {
    WTF_MAKE_FAST_ALLOCATED;
public:   
    Biquad();
    ~Biquad();

    void process(const float* sourceP, float* destP, size_t framesToProcess);

    bool hasSampleAccurateValues() const { return m_hasSampleAccurateValues; }
    void setHasSampleAccurateValues(bool hasSampleAccurateValues) { m_hasSampleAccurateValues = hasSampleAccurateValues; }

    // frequency is 0 - 1 normalized, resonance and dbGain are in decibels.
    // Q is a unitless quality factor.
    void setLowpassParams(size_t index, double frequency, double resonance);
    void setHighpassParams(size_t index, double frequency, double resonance);
    void setBandpassParams(size_t index, double frequency, double Q);
    void setLowShelfParams(size_t index, double frequency, double dbGain);
    void setHighShelfParams(size_t index, double frequency, double dbGain);
    void setPeakingParams(size_t index, double frequency, double Q, double dbGain);
    void setAllpassParams(size_t index, double frequency, double Q);
    void setNotchParams(size_t index, double frequency, double Q);

    // Resets filter state
    void reset();

    // Filter response at a set of n frequencies. The magnitude and
    // phase response are returned in magResponse and phaseResponse.
    // The phase response is in radians.
    void getFrequencyResponse(unsigned nFrequencies, const float* frequency, float* magResponse, float* phaseResponse);

    // Compute tail frame based on the filter coefficents at index
    // |coefIndex|. The tail frame is the frame number where the
    // impulse response of the filter falls below a threshold value.
    // The maximum allowed frame value is given by |maxFrame|. This
    // limits how much work is done in computing the frame number.
    double tailFrame(size_t coefIndex, double maxFrame);

private:
    void setNormalizedCoefficients(size_t index, double b0, double b1, double b2, double a0, double a1, double a2);

    // Filter coefficients. The filter is defined as
    //
    // y[n] + m_a1*y[n-1] + m_a2*y[n-2] = m_b0*x[n] + m_b1*x[n-1] + m_b2*x[n-2].
    AudioDoubleArray m_b0;
    AudioDoubleArray m_b1;
    AudioDoubleArray m_b2;
    AudioDoubleArray m_a1;
    AudioDoubleArray m_a2;

#if USE(ACCELERATE)
    void processFast(const float* sourceP, float* destP, size_t framesToProcess);
    void processSliceFast(double* sourceP, double* destP, double* coefficientsP, size_t framesToProcess);
    AudioDoubleArray m_inputBuffer;
    AudioDoubleArray m_outputBuffer;
#endif

    // Filter memory
    double m_x1; // input delayed by 1 sample
    double m_x2; // input delayed by 2 samples
    double m_y1; // output delayed by 1 sample
    double m_y2; // output delayed by 2 samples

    bool m_hasSampleAccurateValues { false };
};

} // namespace WebCore

#endif // Biquad_h