#include "amy.h"
#include "assert.h"

#ifndef M_PI
    #define M_PI 3.14159265358979323846
#endif

// Filters tend to get weird under this ratio -- this corresponds to 4.4Hz 
#define LOWEST_RATIO 0.0001

#define FILT_NUM_DELAYS  4    // Need 4 memories for DFI filters, if used (only 2 for DFII).

#ifdef __GNUC__
#pragma GCC diagnostic push                             // save the actual diag context
#pragma GCC diagnostic ignored "-Wuninitialized"  // disable maybe warnings
#endif
float dsps_sqrtf_f32_ansi(float f)
{
    int* f_ptr = (int*)&f;
    const int result = 0x1fbb4000 + (*f_ptr >> 1);
    float* f_result = (float*)&result;
    return *f_result;
}
#ifdef __GNUC__
#pragma GCC diagnostic pop
#endif

int8_t dsps_biquad_gen_lpf_f32(SAMPLE *coeffs, float f, float qFactor)
{
    //qFactor = sqrtf(qFactor);
    
    if (qFactor < 0.51) {
        qFactor = 0.51;
    }
    if (f > 0.45) {
        f = 0.45;
    }
    float Fs = 1;

    float w0 = 2 * M_PI * f / Fs;
    w0 = MAX(0.01, w0);  // so f >= Fs * 0.01 / (2 pi) = 70.2 Hz
    float c = cosf(w0);
    float s = sinf(w0);
    float alpha = s / (2 * qFactor);
    // sin w0 / (2 Q) < 1
    // => sin w0 < 2 Q
    // If Q >= 0.5, no problem.

    
    float b0 = (1 - c) / 2;
    float b1 = 1 - c;
    float b2 = b0;
    float a0 = 1 + alpha;
    float a1 = -2 * c;
    float a2 = 1 - alpha;

    // Where exactly are those poles?  Impose minima on (1 - r) and w0.
    float r = -99, ww = 0;
    if (false && a2 > 0) { 
        printf("before: r %f a %f %f %f\n", sqrtf(a2 / a0), a0, a1, a2);
        // Limit how close complex poles can get to the unit circle.
        r = MIN(0.99, sqrtf(a2 / a0));
        float alphadash = (1 - r * r) / (1 + r * r);
        float cosww = c / sqrtf(1 - alphadash * alphadash);
        if (fabs(cosww) < 1.0) {
            ww = acosf(cosww);
            a1 = a0 * (-2 * r * cosf(ww));
            a2 = a0 * r * r;
            printf(" after: r %f a %f %f %f\n", r, a0, a1, a2);
        }
    }

    coeffs[0] = F2S(-b0 / a0);
    coeffs[1] = F2S(-b1 / a0);
    coeffs[2] = F2S(-b2 / a0);
    coeffs[3] = F2S(a1 / a0);
    coeffs[4] = F2S(a2 / a0);

    //printf("Flpf t=%f f=%f q=%f alpha %f b0 %f b1 %f b2 %f a1 %f a2 %f r %f theta %f\n", amy_global.total_blocks*AMY_BLOCK_SIZE / (float)AMY_SAMPLE_RATE, f * AMY_SAMPLE_RATE, qFactor, alpha, 
    //       b0 / a0, b1 / a0, b2 / a0, a1 / a0, a2 / a0, r, w0);
    //printf("Slpf t=%f f=%f q=%f b0 %f b1 %f b2 %f a1 %f a2 %f\n", amy_global.total_blocks*AMY_BLOCK_SIZE / (float)AMY_SAMPLE_RATE, f * AMY_SAMPLE_RATE, qFactor,
    //       S2F(coeffs[0]), S2F(coeffs[1]), S2F(coeffs[2]), S2F(coeffs[3]), S2F(coeffs[4]));

    return 0;
}

int8_t dsps_biquad_gen_hpf_f32(SAMPLE *coeffs, float f, float qFactor)
{
    if (qFactor <= 0.0001) {
        qFactor = 0.0001;
    }
    if (f > 0.45) {
        f = 0.45;
    }
    float Fs = 1;

    float w0 = 2 * M_PI * f / Fs;
    float c = cosf(w0);
    float s = sinf(w0);
    float alpha = s / (2 * qFactor);

    float b0 = (1 + c) / 2;
    float b1 = -(1 + c);
    float b2 = b0;
    float a0 = 1 + alpha;
    float a1 = -2 * c;
    float a2 = 1 - alpha;

    coeffs[0] = F2S(b0 / a0);
    coeffs[1] = F2S(b1 / a0);
    coeffs[2] = F2S(b2 / a0);
    coeffs[3] = F2S(a1 / a0);
    coeffs[4] = F2S(a2 / a0);
    return 0;
}

int8_t dsps_biquad_gen_bpf_f32(SAMPLE *coeffs, float f, float qFactor)
{
    if (qFactor <= 0.0001) {
        qFactor = 0.0001;
    }
    if (f > 0.45) {
        f = 0.45;
    }
    float Fs = 1;

    float w0 = 2 * M_PI * f / Fs;
    float c = cosf(w0);
    float s = sinf(w0);
    float alpha = s / (2 * qFactor);

    float b0 = s / 2;
    float b1 = 0;
    float b2 = -b0;
    float a0 = 1 + alpha;
    float a1 = -2 * c;
    float a2 = 1 - alpha;

    coeffs[0] = F2S(b0 / a0);
    coeffs[1] = F2S(b1 / a0);
    coeffs[2] = F2S(b2 / a0);
    coeffs[3] = F2S(a1 / a0);
    coeffs[4] = F2S(a2 / a0);
    return 0;
}

// Stick to the faster mult for biquad, hpf etc, since the parameters aren't so sensitive, and parametric_eq was chewing major CPU.
#define FILT_MUL_SS(a, b) SMULR6(a, b)

#define FILTER_SCALEUP_BITS 0  // Apply this gain to input before filtering to avoid underflow in intermediate value.
#define FILTER_BIQUAD_SCALEUP_BITS 0  // Apply this gain to input before filtering to avoid underflow in intermediate value.
#define FILTER_BIQUAD_SCALEDOWN_BITS 0  // Extra headroom for EQ filters to avoid clipping on loud signals.

int8_t dsps_biquad_f32_ansi(const SAMPLE *input, SAMPLE *output, int len, SAMPLE *coef, SAMPLE *w) {
    AMY_PROFILE_START(DSPS_BIQUAD_F32_ANSI)

    // Zeros then poles - Direct Form I
    // We need 2 memories for input, and 2 for output.
    SAMPLE x1 = w[0];
    SAMPLE x2 = w[1];
    SAMPLE y1 = w[2];
    SAMPLE y2 = w[3];
    for (int i = 0 ; i < len ; i++) {
        SAMPLE x0 = SHIFTL(input[i], FILTER_BIQUAD_SCALEUP_BITS);
        // SAMPLE x0 = SHIFTR(input[i], FILTER_BIQUAD_SCALEDOWN_BITS);
        SAMPLE w0 = FILT_MUL_SS(coef[0], x0) + FILT_MUL_SS(coef[1], x1) + FILT_MUL_SS(coef[2], x2);
        SAMPLE y0 = w0 - FILT_MUL_SS(coef[3], y1) - FILT_MUL_SS(coef[4], y2);
        x2 = x1;
        x1 = x0;
        y2 = y1;
        y1 = y0;
        output[i] = SHIFTR(y0, FILTER_BIQUAD_SCALEUP_BITS);
        //output[i] = SHIFTL(y0, FILTER_BIQUAD_SCALEDOWN_BITS);
    }
    w[0] = x1;
    w[1] = x2;
    w[2] = y1;
    w[3] = y2;
    AMY_PROFILE_STOP(DSPS_BIQUAD_F32_ANSI)

    return 0;
}


int8_t dsps_biquad_f32_ansi_split_fb(const SAMPLE *input, SAMPLE *output, int len, SAMPLE *coef, SAMPLE *w) {
    AMY_PROFILE_START(DSPS_BIQUAD_F32_ANSI_SPLIT_FB)
    // Rewrite the feeedback coefficients as a1 = -2 + e and a2 = 1 - f
    SAMPLE x1 = w[0];
    SAMPLE x2 = w[1];
    SAMPLE y1 = w[2];
    SAMPLE y2 = w[3];
    SAMPLE e = F2S(2.0f) + coef[3];  // So coef[3] = -2 + e
    SAMPLE f = F2S(1.0f) - coef[4];  // So coef[4] = 1 - f
    //fprintf(stderr, "e=%f (%d) f=%f\n", S2F(e), (e < F2S(0.0625)), S2F(f));
    for (int i = 0 ; i < len ; i++) {
        SAMPLE x0 = SHIFTL(input[i], FILTER_SCALEUP_BITS);
        SAMPLE w0 = FILT_MUL_SS(coef[0], x0) + FILT_MUL_SS(coef[1], x1) + FILT_MUL_SS(coef[2], x2);
        SAMPLE y0 = w0 + SHIFTL(y1, 1) - y2;
        y0 = y0 - FILT_MUL_SS(e, y1) + FILT_MUL_SS(f, y2);
        x2 = x1;
        x1 = x0;
        y2 = y1;
        y1 = y0;
        output[i] = SHIFTR(y0, FILTER_SCALEUP_BITS);
    }
    w[0] = x1;
    w[1] = x2;
    w[2] = y1;
    w[3] = y2;
    AMY_PROFILE_STOP(DSPS_BIQUAD_F32_ANSI_SPLIT_FB)

    return 0;
}

// 16 bit pseudo floating-point multiply.
// See https://colab.research.google.com/drive/1_uQto5WSVMiSPHQ34cHbCC6qkF614EoN#scrollTo=njPHwSB9VIJi

static inline SAMPLE ABS(SAMPLE a) {
    if (a > 0)
        return a;
    else
        return -a;
}

static inline int headroom(SAMPLE a) {
    // How many bits bigger can this value get before overflow?
#ifdef AMY_USE_FIXEDPOINT
    return __builtin_clz(ABS(a) | 1) - 1;  // -1 for sign bit.
#else  // !AMY_USE_FIXEDPOINT
    // How many bits can you shift before this max overflows?
    int bits = 0;
    while(a < F2S(128.0) && bits < 32) {
        a = SHIFTL(a, 1);
        ++bits;
    }
    return bits;
#endif  // AMY_USE_FIXEDPOINT
}

static inline int nheadroom16(SAMPLE a) {
    // Headroom with MAX(0, 16 - headroom(a)) precomputed.
#ifdef AMY_USE_FIXEDPOINT
    return MAX(0, 16 - (__builtin_clz(ABS(a) | 1) - 1));  // -1 for sign bit.
#else  // !AMY_USE_FIXEDPOINT
    // How many bits can you shift before this max overflows?
    int bits = 0;
    while(a < F2S(128.0) && bits < 32) {
        a = SHIFTL(a, 1);
        ++bits;
    }
    return bits;
#endif  // AMY_USE_FIXEDPOINT
}

#ifdef AMY_USE_FIXEDPOINT

SAMPLE top16SMUL(SAMPLE a, SAMPLE b) {
    // Multiply the top 15 bits of a and b.
    //int adropped = MAX(0, 16 - headroom(a));
    int adropped = nheadroom16(a);  //MAX(0, 16 - headroom(a));
    if (adropped) {
        a = SHIFTR(SHIFTR(a, adropped - 1) + 1, 1);
    }
    int resultdrop = 23 - adropped;
    int bdropped = MIN(resultdrop, nheadroom16(b)); // MAX(0, 16 - headroom(b)));
    if (bdropped) {
        //b = SHIFTR(b + (1 << (bdropped - 1)), bdropped);
        b = SHIFTR(SHIFTR(b, bdropped - 1) + 1, 1);
    }
    resultdrop -= bdropped;
    SAMPLE result = a * b;
    if (resultdrop) {
        //return SHIFTR(result + (1 << (resultdrop - 1)), resultdrop);
        return SHIFTR(SHIFTR(result, resultdrop - 1) + 1, 1);
    }
    return result;
}

SAMPLE top16SMUL_a_part(SAMPLE a, int *p_resultdrop) {
    // Just the processing of a, so we can split it out
    int adropped = nheadroom16(a);
    if (adropped) {
        //a = SHIFTR(a + (1 << (adropped - 1)), adropped);
        a = SHIFTR(SHIFTR(a, adropped - 1) + 1, 1);
    }
    *p_resultdrop = 23 - adropped;
    return a;
}

SAMPLE top16SMUL_after_a(SAMPLE a_processed, SAMPLE b, int resultdrop, int bnheadroom16) {
    // The rest of top16SMUL after a has been preprocessed by top16SML_a_part.
    //int bdropped = MIN(resultdrop, MAX(0, 16 - bheadroom));
    int bdrop = MIN(resultdrop, bnheadroom16);
    if (bdrop) {
        //b = SHIFTR(b + (1 << (bdrop - 1)), bdrop);
        b = SHIFTR(SHIFTR(b, bdrop - 1) + 1, 1);
        resultdrop -= bdrop;
    }
    SAMPLE result = a_processed * b;
    if (resultdrop) {
        //return SHIFTR(result + (1 << (resultdrop - 1)), resultdrop);
        return SHIFTR(SHIFTR(result, resultdrop - 1) + 1, 1);
    }
    return result;
}

static inline SAMPLE top16SMUL_after_a_live(SAMPLE a_processed, SAMPLE b, int resultdrop) {
    // Like top16SMUL_after_a, but measures b's actual headroom here instead of
    // trusting a bound precomputed at block start. Needed for the filter state
    // terms: the state can grow far beyond the block-start allowance within one
    // block (e.g. resonance ringing up as the cutoff sweeps toward the signal),
    // and an under-shifted b overflows the 32-bit product.
    return top16SMUL_after_a(a_processed, b, resultdrop, nheadroom16(b));
}

#else // !AMY_USE_FIXEDPOINT

// Sidestep all this logic for SAMPLE == float.

SAMPLE top16SMUL(SAMPLE a, SAMPLE b) {
    return a * b;
}

SAMPLE top16SMUL_a_part(SAMPLE a, int *p_adropped) {
    *p_adropped = 0;  // dropped_bits registration unused in float.
    return a;
}

SAMPLE top16SMUL_after_a(SAMPLE a_processed, SAMPLE b, int adropped_unused, int bheadroom_unused) {
    return a_processed * b;
}

static inline SAMPLE top16SMUL_after_a_live(SAMPLE a_processed, SAMPLE b, int resultdrop_unused) {
    return a_processed * b;
}

#endif // AMY_USE_FIXEDPOINT

SAMPLE scan_max(SAMPLE* block, int len) {
    AMY_PROFILE_START(SCAN_MAX)

    // Find the max abs sample value in a block.
    SAMPLE max = 0;
    while (len--) {
        SAMPLE val = *block++;
        if (val > max) max = val;
        else if ((-val) > max) max = -val;
    }
    AMY_PROFILE_STOP(SCAN_MAX)
    return max;
}

// This is the multiply just for dsps_biquad_f32_ansi_split_fb_twice, which is only used for FILTER_LPF24.
#define FILT_MUL_SS_24(a, b) top16SMUL(a, b)

SAMPLE dsps_biquad_f32_ansi_split_fb_once(const SAMPLE *input, SAMPLE *output, int len, SAMPLE *coef, SAMPLE *w, SAMPLE max_val) {
    // Apply the filter once (for 12 dB/oct LPF)
    AMY_PROFILE_START(DSPS_BIQUAD_F32_ANSI_SPLIT_FB)
    // Rewrite the feeedback coefficients as a1 = -2 + e and a2 = 1 - f
    SAMPLE x1 = w[0];
    SAMPLE x2 = w[1];
    SAMPLE y1 = w[2];
    SAMPLE y2 = w[3];
    int abits, bbits, ebits, fbits;
    SAMPLE a = top16SMUL_a_part(coef[0], &abits);
    SAMPLE e = top16SMUL_a_part(F2S(2.0f) + coef[3], &ebits);  // So coef[3] = -2 + e
    SAMPLE f = top16SMUL_a_part(F2S(1.0f) - coef[4], &fbits);  // So coef[4] = 1 - f
    assert(FILTER_SCALEUP_BITS == 0);
    bbits = nheadroom16(max_val);
    SAMPLE max_out = 0;
    for (int i = 0 ; i < len ; i++) {
        SAMPLE x0 = top16SMUL_after_a(a, input[i], abits, bbits);  // headroom(input[i]);
        SAMPLE w0 = x0 + SHIFTL(x1, 1) + x2;
        SAMPLE y0 = w0 + SHIFTL(y1, 1) - y2;
        y0 = y0
            - top16SMUL_after_a_live(e, y1, ebits)
            + top16SMUL_after_a_live(f, y2, fbits);
        x2 = x1;
        x1 = x0;
        y2 = y1;
        y1 = y0;
        output[i] = y0;
        if (y0 < 0) y0 = -y0;
        if (y0 > 0) {
            if (y0 > max_out)
                max_out = y0;
        } else {
            if (-y0 > max_out)
                max_out = -y0;
        }
    }
    w[0] = x1;
    w[1] = x2;
    w[2] = y1;
    w[3] = y2;
    AMY_PROFILE_STOP(DSPS_BIQUAD_F32_ANSI_SPLIT_FB)

    return max_out;
}

SAMPLE dsps_biquad_f32_ansi_split_fb_twice(const SAMPLE *input, SAMPLE *output, int len, SAMPLE *coef, SAMPLE *w, SAMPLE max_val) {
    // Apply the filter twice
    AMY_PROFILE_START(DSPS_BIQUAD_F32_ANSI_SPLIT_FB_TWICE)
    // Rewrite the feeedback coefficients as a1 = -2 + e and a2 = 1 - f
    SAMPLE x1 = w[0];
    SAMPLE x2 = w[1];
    SAMPLE y1 = w[2];
    SAMPLE y2 = w[3];
    SAMPLE v1 = w[4];
    SAMPLE v2 = w[5];
    int abits, bbits, ebits, fbits;
    SAMPLE a = top16SMUL_a_part(coef[0], &abits);
    SAMPLE e = top16SMUL_a_part(F2S(2.0f) + coef[3], &ebits);  // So coef[3] = -2 + e
    SAMPLE f = top16SMUL_a_part(F2S(1.0f) - coef[4], &fbits);  // So coef[4] = 1 - f
    assert(FILTER_SCALEUP_BITS == 0);
    bbits = nheadroom16(max_val);
    SAMPLE max_out = 0;
    for (int i = 0 ; i < len ; i++) {
        SAMPLE x0, w0, v0;
        x0 = top16SMUL_after_a(a, input[i], abits, bbits);  // headroom(input[i]);
        w0 = x0 + SHIFTL(x1, 1) + x2;
        v0 = w0 + SHIFTL(v1, 1) - v2;
        v0 = v0
            - top16SMUL_after_a_live(e, v1, ebits)
            + top16SMUL_after_a_live(f, v2, fbits);
        w0 = v0 + SHIFTL(v1, 1) + v2;
        w0 = top16SMUL_after_a_live(a, w0, abits);
        SAMPLE y0 = w0 + SHIFTL(y1, 1) - y2;
        y0 = y0
            - top16SMUL_after_a_live(e, y1, ebits)
            + top16SMUL_after_a_live(f, y2, fbits);
        x2 = x1;
        x1 = x0;
        v2 = v1;
        v1 = v0;
        y2 = y1;
        y1 = y0;
        output[i] = y0;
        if (y0 < 0) y0 = -y0;
        if (y0 > 0) {
            if (y0 > max_out)
                max_out = y0;
        } else {
            if (-y0 > max_out)
                max_out = -y0;
        }
    }
    w[0] = x1;
    w[1] = x2;
    w[2] = y1;
    w[3] = y2;
    w[4] = v1;
    w[5] = v2;
    AMY_PROFILE_STOP(DSPS_BIQUAD_F32_ANSI_SPLIT_FB_TWICE)

    return max_out;
}

int8_t dsps_biquad_f32_ansi_commuted(const SAMPLE *input, SAMPLE *output, int len, SAMPLE *coef, SAMPLE *w) {
    AMY_PROFILE_START(DSPS_BIQUAD_F32_ANSI_COMMUTED)
    // poles before zeros, for Direct Form II
    for (int i = 0 ; i < len ; i++) {
        SAMPLE d0 = input[i] - FILT_MUL_SS(coef[3], w[0]) - FILT_MUL_SS(coef[4], w[1]);
        output[i] = FILT_MUL_SS(coef[0], d0) + FILT_MUL_SS(coef[1], w[0]) + FILT_MUL_SS(coef[2], w[1]);
        w[1] = w[0];
        w[0] = d0;
    }
    AMY_PROFILE_STOP(DSPS_BIQUAD_F32_ANSI_COMMUTED)

    return 0;
}

void filters_deinit(uint8_t bus) {
    SAMPLE **eq_coeffs = amy_global.bus[bus]->eq.eq_coeffs;
    SAMPLE ***eq_delay = amy_global.bus[bus]->eq.eq_delay;
    for(uint16_t i=0;i<AMY_NCHANS;i++) {
        for(uint16_t j=0;j<3;j++) free(eq_delay[i][j]);
        free(eq_delay[i]);
    }
    for(uint16_t i=0;i<3;i++) free(eq_coeffs[i]);
    free(eq_coeffs);
    free(eq_delay);
    amy_global.bus[bus]->eq.eq_coeffs = NULL;
    amy_global.bus[bus]->eq.eq_delay = NULL;
}


void filters_init(uint8_t bus) {
    SAMPLE ***p_eq_coeffs = &amy_global.bus[bus]->eq.eq_coeffs;
    SAMPLE ****p_eq_delay = &amy_global.bus[bus]->eq.eq_delay;
    *p_eq_coeffs = malloc_caps(sizeof(SAMPLE*)*3, amy_global.config.ram_caps_fbl);
    *p_eq_delay = malloc_caps(sizeof(SAMPLE**)*AMY_NCHANS, amy_global.config.ram_caps_fbl);
    for(uint16_t i=0;i<3;i++) {
        (*p_eq_coeffs)[i] = malloc_caps(sizeof(SAMPLE) * 5, amy_global.config.ram_caps_fbl);
    }
    for(uint16_t i=0;i<AMY_NCHANS;i++) {
        (*p_eq_delay)[i] = malloc_caps(sizeof(SAMPLE*) * 3, amy_global.config.ram_caps_fbl);
        for(uint16_t j=0;j<3;j++) {
            (*p_eq_delay)[i][j] = malloc_caps(sizeof(SAMPLE) * FILT_NUM_DELAYS, amy_global.config.ram_caps_fbl);

        }
    }

    // update the parametric filters 
    dsps_biquad_gen_lpf_f32((*p_eq_coeffs)[0], EQ_CENTER_LOW /(float)AMY_SAMPLE_RATE, 0.707);
    dsps_biquad_gen_bpf_f32((*p_eq_coeffs)[1], EQ_CENTER_MED /(float)AMY_SAMPLE_RATE, 1.000);
    dsps_biquad_gen_hpf_f32((*p_eq_coeffs)[2], EQ_CENTER_HIGH/(float)AMY_SAMPLE_RATE, 0.707);
    for (int c = 0; c < AMY_NCHANS; ++c) {
        for (int b = 0; b < 3; ++b) {
            for (int d = 0; d < FILT_NUM_DELAYS; ++d) {
                (*p_eq_delay)[c][b][d] = 0;
            }
        }
    }
}


#define FILT_MUL_SS_EQ(a, b) top16SMUL(a, b)
//#define FILT_MUL_SS_EQ(a, b) SMULR6(a, b)

inline static SAMPLE MAXABS2(SAMPLE a, SAMPLE b) {
    if (a < 0) a = -a;
    if (b < 0) b = -b;
    if (a > b) return a;
    return b;
}

void parametric_eq_process(uint8_t bus, SAMPLE *block) {
    // was void parametric_eq_process_top16block
    // Optimized to run all 3 filters interleaved, to avoid extra buffers/buf accesses.
    AMY_PROFILE_START(PARAMETRIC_EQ_PROCESS)
    SAMPLE **eq_coeffs = amy_global.bus[bus]->eq.eq_coeffs;
    SAMPLE ***eq_delay = amy_global.bus[bus]->eq.eq_delay;
        
    for(int c = 0; c < AMY_NCHANS; ++c) {
        SAMPLE *cblock = block + c * AMY_BLOCK_SIZE;
        // Zeros then poles - Direct Form I
        // We need 2 memories for input, and 2 for output.
        SAMPLE x1 = eq_delay[c][0][0];
        SAMPLE x2 = eq_delay[c][0][1];
        SAMPLE y01 = eq_delay[c][0][2];
        SAMPLE y02 = eq_delay[c][0][3];
        SAMPLE y11 = eq_delay[c][1][2];
        SAMPLE y12 = eq_delay[c][1][3];
        SAMPLE y21 = eq_delay[c][2][2];
        SAMPLE y22 = eq_delay[c][2][3];
        int c00bits, c03bits, c04bits;
        int c10bits, c13bits, c14bits;
        int c20bits, c23bits, c24bits;
        // int c21bits, c22bits, c11bits, c12bits, c01bits, c02bits;
        // Fold the global EQ parameters into the forward-gains of each stage.
        SAMPLE c00 = top16SMUL_a_part(top16SMUL(amy_global.bus[bus]->eq.eq[0], eq_coeffs[0][0]), &c00bits);
        SAMPLE c03 = top16SMUL_a_part(eq_coeffs[0][3], &c03bits);
        SAMPLE c04 = top16SMUL_a_part(eq_coeffs[0][4], &c04bits);
        SAMPLE c10 = top16SMUL_a_part(top16SMUL(amy_global.bus[bus]->eq.eq[1], eq_coeffs[1][0]), &c10bits);
        SAMPLE c13 = top16SMUL_a_part(eq_coeffs[1][3], &c13bits);
        SAMPLE c14 = top16SMUL_a_part(eq_coeffs[1][4], &c14bits);
        SAMPLE c20 = top16SMUL_a_part(top16SMUL(amy_global.bus[bus]->eq.eq[2], eq_coeffs[2][0]), &c20bits);
        SAMPLE c23 = top16SMUL_a_part(eq_coeffs[2][3], &c23bits);
        SAMPLE c24 = top16SMUL_a_part(eq_coeffs[2][4], &c24bits);
        for (int i = 0 ; i < AMY_BLOCK_SIZE ; i++) {
            SAMPLE x0 = cblock[i];
            SAMPLE x1times2 = SHIFTL(x1, 1);
            int xbits = nheadroom16(MAXABS2(x0, MAXABS2(x1times2, x2)));
            // Optimize the FIR multiplies for the known structure of the zeros in LPF/BPF/HPF.
            SAMPLE w00 = top16SMUL_after_a(c00, x0 + x1times2 + x2, c00bits, xbits);
            int y0bits = nheadroom16(MAXABS2(y01, y02));
            SAMPLE y00 = w00 - top16SMUL_after_a(c03, y01, c03bits, y0bits) - top16SMUL_after_a(c04, y02, c04bits, y0bits);
            SAMPLE w10 = top16SMUL_after_a(c10, x0 - x2, c10bits, xbits);
            int y1bits = nheadroom16(MAXABS2(y11, y12));
            SAMPLE y10 = w10 - top16SMUL_after_a(c13, y11, c13bits, y1bits) - top16SMUL_after_a(c14, y12, c14bits, y1bits);
            SAMPLE w20 = top16SMUL_after_a(c20, x0 - x1times2 + x2, c20bits, xbits);
            int y2bits = nheadroom16(MAXABS2(y21, y22));
            SAMPLE y20 = w20 - top16SMUL_after_a(c23, y21, c23bits, y2bits) - top16SMUL_after_a(c24, y22, c24bits, y2bits);
            x2 = x1;
            x1 = x0;
            y02 = y01;
            y01 = y00;
            y12 = y11;
            y11 = y10;
            y22 = y21;
            y21 = y20;
            cblock[i] = y00 - y10 + y20;
        }
        eq_delay[c][0][0] = x1;
        eq_delay[c][0][1] = x2;
        eq_delay[c][0][2] = y01;
        eq_delay[c][0][3] = y02;
        eq_delay[c][1][2] = y11;
        eq_delay[c][1][3] = y12;
        eq_delay[c][2][2] = y21;
        eq_delay[c][2][3] = y22;
    }
    AMY_PROFILE_STOP(PARAMETRIC_EQ_PROCESS)
}

void hpf_buf(SAMPLE *buf, SAMPLE *state) {
    AMY_PROFILE_START(HPF_BUF)

    // Implement a dc-blocking HPF with a pole at (decay + 0j) and a zero at (1 + 0j).
    SAMPLE pole = F2S(0.99);
    SAMPLE xn1 = state[0];
    SAMPLE yn1 = state[1];
    for (uint16_t i = 0; i < AMY_BLOCK_SIZE; ++i) {
        SAMPLE w = buf[i] - xn1;
        xn1 = buf[i];
        buf[i] = w + FILT_MUL_SS(pole, yn1);
        yn1 = buf[i];
    }
    state[0] = xn1;
    state[1] = yn1;
    AMY_PROFILE_STOP(HPF_BUF)

}

void check_overflow(SAMPLE* block, int osc, char *msg) {
    // Search for overflow in a sample buffer as an unusually large sample-to-sample delta.
#ifdef AMY_DEBUG
    SAMPLE last = block[0];
    SAMPLE max = 0;
    SAMPLE maxdiff = 0;
    int len = AMY_BLOCK_SIZE;
    while (len--) {
        SAMPLE val = *block++;
        SAMPLE diff = val - last;
        if (diff < 0)  diff = -diff;
        if (diff > maxdiff) maxdiff = diff;
        last = val;
        if (val < 0)  val = -val;
        if (val > max)  max = val;
    }
    if (maxdiff > F2S(0.2f))
        fprintf(stderr, "Overflow at timeframe %.3f max=%.3f maxdiff=%.3f osc=%d msg=%s\n",
                (float)(amy_global.total_blocks*AMY_BLOCK_SIZE) / AMY_SAMPLE_RATE,
                S2F(max), S2F(maxdiff), osc, msg);

#endif // AMY_DEBUG
}

SAMPLE block_norm(SAMPLE* block, int len, int bits) {
    AMY_PROFILE_START(BLOCK_NORM)

    SAMPLE max_val = 0;
    if (bits >= 0) {
        // do this even if bits == 0 to ensure max_val is set.
        while(len--) {
            *block = SHIFTL(*block, bits);
            if (*block > 0) {
                if (*block > max_val)
                    max_val = *block;
            } else {
                if (-*block > max_val)
                    max_val = -*block;
            }
            ++block;
        }
    } else {
        // bits is negative - right-shift.
        bits = -bits;
        while(len--) {
            *block = SHIFTR(*block, bits);
            if (*block > 0) {
                if (*block > max_val)
                    max_val = *block;
            } else {
                if (-*block > max_val)
                    max_val = -*block;
            }
            ++block;
        }
    }
    AMY_PROFILE_STOP(BLOCK_NORM)
    return max_val;
}

SAMPLE block_denorm(SAMPLE* block, int len, int bits) {
    return block_norm(block, len, -bits);
}

SAMPLE filter_process(SAMPLE * block, uint16_t osc, SAMPLE max_val) {

    SAMPLE coeffs[5];

    SAMPLE filtmax = scan_max(synth[osc]->filter_delay, 2 * FILT_NUM_DELAYS);
    if (max_val == 0 && filtmax == 0) return 0;

    AMY_PROFILE_START(FILTER_PROCESS)

    AMY_PROFILE_START(FILTER_PROCESS_STAGE0)

    float ratio = freq_of_logfreq(msynth[osc]->filter_logfreq)/(float)AMY_SAMPLE_RATE;
    if(ratio < LOWEST_RATIO) ratio = LOWEST_RATIO;
    if(synth[osc]->filter_type==FILTER_LPF || synth[osc]->filter_type==FILTER_LPF24)
        dsps_biquad_gen_lpf_f32(coeffs, ratio, msynth[osc]->resonance);
    else if(synth[osc]->filter_type==FILTER_BPF)
        dsps_biquad_gen_bpf_f32(coeffs, ratio, msynth[osc]->resonance);
    else if(synth[osc]->filter_type==FILTER_HPF)
        dsps_biquad_gen_hpf_f32(coeffs, ratio, msynth[osc]->resonance);
    else {
        fprintf(stderr, "Unrecognized filter type %d\n", synth[osc]->filter_type);
        return 0;
    }
    AMY_PROFILE_STOP(FILTER_PROCESS_STAGE0)

#ifdef NOTDEF
    printf("FlPr t=%.3f f=%.3f q=%.3f %.3f %.3f %.3f %.3f %.3f ST %.6f %.6f %.6f %.6f %.6f %.6f %.6f %.6f B %.6f %.6f %.6f %.6f\n",
           amy_global.total_blocks*AMY_BLOCK_SIZE / (float)AMY_SAMPLE_RATE,
           ratio * AMY_SAMPLE_RATE, msynth[osc]->resonance, 
           S2F(coeffs[0]), S2F(coeffs[1]), S2F(coeffs[2]), S2F(coeffs[3]), S2F(coeffs[4]),
           S2F(synth[osc]->filter_delay[0]), 
           S2F(synth[osc]->filter_delay[1]), 
           S2F(synth[osc]->filter_delay[2]), 
           S2F(synth[osc]->filter_delay[3]),
           S2F(synth[osc]->filter_delay[4]), 
           S2F(synth[osc]->filter_delay[5]), 
           S2F(synth[osc]->filter_delay[6]),
           S2F(synth[osc]->filter_delay[7]),
           S2F(block[0]), S2F(block[1]), S2F(block[2]), S2F(block[3])
           );
#endif

    AMY_PROFILE_START(FILTER_PROCESS_STAGE1)
    //SAMPLE max_val = scan_max(block, AMY_BLOCK_SIZE);
    // Also have to consider the filter state.
#define USE_BLOCK_FLOATING_POINT
#ifdef USE_BLOCK_FLOATING_POINT
    int filtnormbits = synth[osc]->last_filt_norm_bits + headroom(filtmax);
#define HEADROOM_BITS 6
#define STATE_HEADROOM_BITS 2
    int normbits = MIN(MAX(0, headroom(max_val) - HEADROOM_BITS), MAX(0, filtnormbits - STATE_HEADROOM_BITS));
    normbits = MIN(normbits, synth[osc]->last_filt_norm_bits + 1);  // Increase at most one bit per block.
    normbits = MIN(8, normbits);  // Without this, I get a weird sign flip at the end of TestLFO - intermediate overflow?
#else  // No block floating point
    const int normbits = 0;  // defeat BFP
#endif
    //printf("time %f max_val %f filtmax %f lastfiltnormbits %d filtnormbits %d normbits %d\n", amy_global.total_blocks*AMY_BLOCK_SIZE / (float)AMY_SAMPLE_RATE, S2F(max_val), S2F(filtmax), synth[osc]->last_filt_norm_bits, filtnormbits, normbits);
    if(synth[osc]->filter_type==FILTER_LPF24) {
        // 24 dB/oct by running the same filter twice.
        max_val = dsps_biquad_f32_ansi_split_fb_twice(block, block, AMY_BLOCK_SIZE, coeffs, synth[osc]->filter_delay, max_val);
    //} else if(synth[osc]->filter_type==FILTER_LPF) {
        // Optimized block-floating point 12 dB/oct LPF
        //max_val = dsps_biquad_f32_ansi_split_fb_once(block, block, AMY_BLOCK_SIZE, coeffs, synth[osc]->filter_delay, max_val);
    } else {
        block_norm(synth[osc]->filter_delay, 2 * FILT_NUM_DELAYS, normbits - synth[osc]->last_filt_norm_bits);
        block_norm(block, AMY_BLOCK_SIZE, normbits);
        dsps_biquad_f32_ansi_split_fb(block, block, AMY_BLOCK_SIZE, coeffs, synth[osc]->filter_delay);
        max_val = block_denorm(block, AMY_BLOCK_SIZE, normbits);
        synth[osc]->last_filt_norm_bits = normbits;
    }
    //dsps_biquad_f32_ansi_commuted(block, block, AMY_BLOCK_SIZE, coeffs, synth[osc]->filter_delay);
    //block_denorm(synth[osc]->filter_delay, 2 * FILT_NUM_DELAYS, normbits);
    // Final high-pass to remove residual DC offset from sub-fundamental LPF.  (Not needed now source waveforms are zero-mean).
    AMY_PROFILE_STOP(FILTER_PROCESS_STAGE1)
    AMY_PROFILE_STOP(FILTER_PROCESS)
#ifdef NOTDEF
    printf("FlP2 t=%.3f f=%.3f q=%.3f %.3f %.3f %.3f %.3f %.3f ST %.6f %.6f %.6f %.6f %.6f %.6f %.6f %.6f B %.6f %.6f %.6f %.6f\n",
           amy_global.total_blocks*AMY_BLOCK_SIZE / (float)AMY_SAMPLE_RATE,
           ratio * AMY_SAMPLE_RATE, msynth[osc]->resonance, 
           S2F(coeffs[0]), S2F(coeffs[1]), S2F(coeffs[2]), S2F(coeffs[3]), S2F(coeffs[4]),
           S2F(synth[osc]->filter_delay[0]), 
           S2F(synth[osc]->filter_delay[1]), 
           S2F(synth[osc]->filter_delay[2]), 
           S2F(synth[osc]->filter_delay[3]),
           S2F(synth[osc]->filter_delay[4]), 
           S2F(synth[osc]->filter_delay[5]), 
           S2F(synth[osc]->filter_delay[6]),
           S2F(synth[osc]->filter_delay[7]),
           S2F(block[0]), S2F(block[1]), S2F(block[2]), S2F(block[3])
           );
#endif
    return max_val;
}


void reset_filter(uint16_t osc) {
    // Reset all the filter state to zero.
    // The LPF has typically accumulated a large DC offset, so you have to reset both
    // the LPF *and* the dc-blocking HPF at the same time.
    for(int i = 0; i < 2 * FILT_NUM_DELAYS; ++i) synth[osc]->filter_delay[i] = 0;
    synth[osc]->last_filt_norm_bits = 0;
}


void reset_parametric(uint8_t bus) {
    SAMPLE ***eq_delay = amy_global.bus[bus]->eq.eq_delay;
    for (int c = 0; c < AMY_NCHANS; ++c) {
        for (int b = 0; b < 3; ++b) {
            for (int d = 0; d < FILT_NUM_DELAYS; ++d) {
                eq_delay[c][b][d] = 0;
            }
        }
    }
}
