/*

 * Copyright (C) 2016 foo86

 *

 * This file is part of FFmpeg.

 *

 * FFmpeg is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Lesser General Public

 * License as published by the Free Software Foundation; either

 * version 2.1 of the License, or (at your option) any later version.

 *

 * FFmpeg 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

 * Lesser General Public License for more details.

 *

 * You should have received a copy of the GNU Lesser General Public

 * License along with FFmpeg; if not, write to the Free Software

 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

 */

#include "libavutil/mem_internal.h"

#include "dcadsp.h"

#include "dcamath.h"

static void decode_hf_c(int32_t **dst,

                        const int32_t *vq_index,

                        const int8_t hf_vq[1024][32],

                        int32_t scale_factors[32][2],

                        ptrdiff_t sb_start, ptrdiff_t sb_end,

                        ptrdiff_t ofs, ptrdiff_t len)

{

    int i, j;

    for (i = sb_start; i < sb_end; i++) {

        const int8_t *coeff = hf_vq[vq_index[i]];

        int32_t scale = scale_factors[i][0];

        for (j = 0; j < len; j++)

            dst[i][j + ofs] = clip23(coeff[j] * scale + (1 << 3) >> 4);

    }

}

static void decode_joint_c(int32_t **dst, int32_t **src,

                           const int32_t *scale_factors,

                           ptrdiff_t sb_start, ptrdiff_t sb_end,

                           ptrdiff_t ofs, ptrdiff_t len)

{

    int i, j;

    for (i = sb_start; i < sb_end; i++) {

        int32_t scale = scale_factors[i];

        for (j = 0; j < len; j++)

            dst[i][j + ofs] = clip23(mul17(src[i][j + ofs], scale));

    }

}

static void lfe_fir_float_c(float *pcm_samples, int32_t *lfe_samples,

                            const float *filter_coeff, ptrdiff_t npcmblocks,

                            int dec_select)

{

    // Select decimation factor

    int factor = 64 << dec_select;

    int ncoeffs = 8 >> dec_select;

    int nlfesamples = npcmblocks >> (dec_select + 1);

    int i, j, k;

    for (i = 0; i < nlfesamples; i++) {

        // One decimated sample generates 64 or 128 interpolated ones

        for (j = 0; j < factor / 2; j++) {

            float a = 0;

            float b = 0;

            for (k = 0; k < ncoeffs; k++) {

                a += filter_coeff[      j * ncoeffs + k] * lfe_samples[-k];

                b += filter_coeff[255 - j * ncoeffs - k] * lfe_samples[-k];

            }

            pcm_samples[             j] = a;

            pcm_samples[factor / 2 + j] = b;

        }

        lfe_samples++;

        pcm_samples += factor;

    }

}

static void lfe_fir0_float_c(float *pcm_samples, int32_t *lfe_samples,

                             const float *filter_coeff, ptrdiff_t npcmblocks)

{

    lfe_fir_float_c(pcm_samples, lfe_samples, filter_coeff, npcmblocks, 0);

}

static void lfe_fir1_float_c(float *pcm_samples, int32_t *lfe_samples,

                             const float *filter_coeff, ptrdiff_t npcmblocks)

{

    lfe_fir_float_c(pcm_samples, lfe_samples, filter_coeff, npcmblocks, 1);

}

static void lfe_x96_float_c(float *dst, const float *src,

                            float *hist, ptrdiff_t len)

{

    float prev = *hist;

    int i;

    for (i = 0; i < len; i++) {

        float a = 0.25f * src[i] + 0.75f * prev;

        float b = 0.75f * src[i] + 0.25f * prev;

        prev = src[i];

        *dst++ = a;

        *dst++ = b;

    }

    *hist = prev;

}

static void sub_qmf32_float_c(SynthFilterContext *synth,

                              AVTXContext *imdct,

                              av_tx_fn imdct_fn,

                              float *pcm_samples,

                              int32_t **subband_samples_lo,

                              int32_t **subband_samples_hi,

                              float *hist1, int *offset, float *hist2,

                              const float *filter_coeff, ptrdiff_t npcmblocks,

                              float scale)

{

    LOCAL_ALIGNED_32(float, input, [32]);

    int i, j;

    for (j = 0; j < npcmblocks; j++) {

        // Load in one sample from each subband

        for (i = 0; i < 32; i++) {

            if ((i - 1) & 2)

                input[i] = -subband_samples_lo[i][j];

            else

                input[i] =  subband_samples_lo[i][j];

        }

        // One subband sample generates 32 interpolated ones

        synth->synth_filter_float(imdct, hist1, offset,

                                  hist2, filter_coeff,

                                  pcm_samples, input, scale, imdct_fn);

        pcm_samples += 32;

    }

}

static void sub_qmf64_float_c(SynthFilterContext *synth,

                              AVTXContext *imdct,

                              av_tx_fn imdct_fn,

                              float *pcm_samples,

                              int32_t **subband_samples_lo,

                              int32_t **subband_samples_hi,

                              float *hist1, int *offset, float *hist2,

                              const float *filter_coeff, ptrdiff_t npcmblocks,

                              float scale)

{

    LOCAL_ALIGNED_32(float, input, [64]);

    int i, j;

    if (!subband_samples_hi)

        memset(&input[32], 0, sizeof(input[0]) * 32);

    for (j = 0; j < npcmblocks; j++) {

        // Load in one sample from each subband

        if (subband_samples_hi) {

            // Full 64 subbands, first 32 are residual coded

            for (i =  0; i < 32; i++) {

                if ((i - 1) & 2)

                    input[i] = -subband_samples_lo[i][j] - subband_samples_hi[i][j];

                else

                    input[i] =  subband_samples_lo[i][j] + subband_samples_hi[i][j];

            }

            for (i = 32; i < 64; i++) {

                if ((i - 1) & 2)

                    input[i] = -subband_samples_hi[i][j];

                else

                    input[i] =  subband_samples_hi[i][j];

            }

        } else {

            // Only first 32 subbands

            for (i =  0; i < 32; i++) {

                if ((i - 1) & 2)

                    input[i] = -subband_samples_lo[i][j];

                else

                    input[i] =  subband_samples_lo[i][j];

            }

        }

        // One subband sample generates 64 interpolated ones

        synth->synth_filter_float_64(imdct, hist1, offset,

                                     hist2, filter_coeff,

                                     pcm_samples, input, scale, imdct_fn);

        pcm_samples += 64;

    }

}

static void lfe_fir_fixed_c(int32_t *pcm_samples, int32_t *lfe_samples,

                            const int32_t *filter_coeff, ptrdiff_t npcmblocks)

{

    // Select decimation factor

    int nlfesamples = npcmblocks >> 1;

    int i, j, k;

    for (i = 0; i < nlfesamples; i++) {

        // One decimated sample generates 64 interpolated ones

        for (j = 0; j < 32; j++) {

            int64_t a = 0;

            int64_t b = 0;

            for (k = 0; k < 8; k++) {

                a += (int64_t)filter_coeff[      j * 8 + k] * lfe_samples[-k];

                b += (int64_t)filter_coeff[255 - j * 8 - k] * lfe_samples[-k];

            }

            pcm_samples[     j] = clip23(norm23(a));

            pcm_samples[32 + j] = clip23(norm23(b));

        }

        lfe_samples++;

        pcm_samples += 64;

    }

}

static void lfe_x96_fixed_c(int32_t *dst, const int32_t *src,

                            int32_t *hist, ptrdiff_t len)

{

    int32_t prev = *hist;

    int i;

    for (i = 0; i < len; i++) {

        int64_t a = INT64_C(2097471) * src[i] + INT64_C(6291137) * prev;

        int64_t b = INT64_C(6291137) * src[i] + INT64_C(2097471) * prev;

        prev = src[i];

        *dst++ = clip23(norm23(a));

        *dst++ = clip23(norm23(b));

    }

    *hist = prev;

}

static void sub_qmf32_fixed_c(SynthFilterContext *synth,

                              DCADCTContext *imdct,

                              int32_t *pcm_samples,

                              int32_t **subband_samples_lo,

                              int32_t **subband_samples_hi,

                              int32_t *hist1, int *offset, int32_t *hist2,

                              const int32_t *filter_coeff, ptrdiff_t npcmblocks)

{

    LOCAL_ALIGNED_32(int32_t, input, [32]);

    int i, j;

    for (j = 0; j < npcmblocks; j++) {

        // Load in one sample from each subband

        for (i = 0; i < 32; i++)

            input[i] = subband_samples_lo[i][j];

        // One subband sample generates 32 interpolated ones

        synth->synth_filter_fixed(imdct, hist1, offset,

                                  hist2, filter_coeff,

                                  pcm_samples, input);

        pcm_samples += 32;

    }

}

static void sub_qmf64_fixed_c(SynthFilterContext *synth,

                              DCADCTContext *imdct,

                              int32_t *pcm_samples,

                              int32_t **subband_samples_lo,

                              int32_t **subband_samples_hi,

                              int32_t *hist1, int *offset, int32_t *hist2,

                              const int32_t *filter_coeff, ptrdiff_t npcmblocks)

{

    LOCAL_ALIGNED_32(int32_t, input, [64]);

    int i, j;

    if (!subband_samples_hi)

        memset(&input[32], 0, sizeof(input[0]) * 32);

    for (j = 0; j < npcmblocks; j++) {

        // Load in one sample from each subband

        if (subband_samples_hi) {

            // Full 64 subbands, first 32 are residual coded

            for (i =  0; i < 32; i++)

                input[i] = subband_samples_lo[i][j] + subband_samples_hi[i][j];

            for (i = 32; i < 64; i++)

                input[i] = subband_samples_hi[i][j];

        } else {

            // Only first 32 subbands

            for (i =  0; i < 32; i++)

                input[i] = subband_samples_lo[i][j];

        }

        // One subband sample generates 64 interpolated ones

        synth->synth_filter_fixed_64(imdct, hist1, offset,

                                     hist2, filter_coeff,

                                     pcm_samples, input);

        pcm_samples += 64;

    }

}

static void decor_c(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)

{

    int i;

    for (i = 0; i < len; i++)

        dst[i] += (SUINT)((int)(src[i] * (SUINT)coeff + (1 << 2)) >> 3);

}

static void dmix_sub_xch_c(int32_t *dst1, int32_t *dst2,

                           const int32_t *src, ptrdiff_t len)

{

    int i;

    for (i = 0; i < len; i++) {

        int32_t cs = mul23(src[i], 5931520 /* M_SQRT1_2 * (1 << 23) */);

        dst1[i] -= cs;

        dst2[i] -= cs;

    }

}

static void dmix_sub_c(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)

{

    int i;

    for (i = 0; i < len; i++)

        dst[i] -= (unsigned)mul15(src[i], coeff);

}

static void dmix_add_c(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)

{

    int i;

    for (i = 0; i < len; i++)

        dst[i] += (unsigned)mul15(src[i], coeff);

}

static void dmix_scale_c(int32_t *dst, int scale, ptrdiff_t len)

{

    int i;

    for (i = 0; i < len; i++)

        dst[i] = mul15(dst[i], scale);

}

static void dmix_scale_inv_c(int32_t *dst, int scale_inv, ptrdiff_t len)

{

    int i;

    for (i = 0; i < len; i++)

        dst[i] = mul16(dst[i], scale_inv);

}

static void filter0(SUINT32 *dst, const int32_t *src, int32_t coeff, ptrdiff_t len)

{

    int i;

    for (i = 0; i < len; i++)

        dst[i] -= mul22(src[i], coeff);

}

static void filter1(SUINT32 *dst, const int32_t *src, int32_t coeff, ptrdiff_t len)

{

    int i;

    for (i = 0; i < len; i++)

        dst[i] -= mul23(src[i], coeff);

}

static void assemble_freq_bands_c(int32_t *dst, int32_t *src0, int32_t *src1,

                                  const int32_t *coeff, ptrdiff_t len)

{

    int i;

    filter0(src0, src1, coeff[0], len);

    filter0(src1, src0, coeff[1], len);

    filter0(src0, src1, coeff[2], len);

    filter0(src1, src0, coeff[3], len);

    for (i = 0; i < 8; i++, src0--) {

        filter1(src0, src1, coeff[i +  4], len);

        filter1(src1, src0, coeff[i + 12], len);

        filter1(src0, src1, coeff[i +  4], len);

    }

    for (i = 0; i < len; i++) {

        *dst++ = *src1++;

        *dst++ = *++src0;

    }

}

static void lbr_bank_c(float output[32][4], float **input,

                       const float *coeff, ptrdiff_t ofs, ptrdiff_t len)

{

    float SW0 = coeff[0];

    float SW1 = coeff[1];

    float SW2 = coeff[2];

    float SW3 = coeff[3];

    float C1  = coeff[4];

    float C2  = coeff[5];

    float C3  = coeff[6];

    float C4  = coeff[7];

    float AL1 = coeff[8];

    float AL2 = coeff[9];

    int i;

    // Short window and 8 point forward MDCT

    for (i = 0; i < len; i++) {

        float *src = input[i] + ofs;

        float a = src[-4] * SW0 - src[-1] * SW3;

        float b = src[-3] * SW1 - src[-2] * SW2;

        float c = src[ 2] * SW1 + src[ 1] * SW2;

        float d = src[ 3] * SW0 + src[ 0] * SW3;

        output[i][0] = C1 * b - C2 * c + C4 * a - C3 * d;

        output[i][1] = C1 * d - C2 * a - C4 * b - C3 * c;

        output[i][2] = C3 * b + C2 * d - C4 * c + C1 * a;

        output[i][3] = C3 * a - C2 * b + C4 * d - C1 * c;

    }

    // Aliasing cancellation for high frequencies

    for (i = 12; i < len - 1; i++) {

        float a = output[i  ][3] * AL1;

        float b = output[i+1][0] * AL1;

        output[i  ][3] += b - a;

        output[i+1][0] -= b + a;

        a = output[i  ][2] * AL2;

        b = output[i+1][1] * AL2;

        output[i  ][2] += b - a;

        output[i+1][1] -= b + a;

    }

}

static void lfe_iir_c(float *output, const float *input,

                      const float iir[5][4], float hist[5][2],

                      ptrdiff_t factor)

{

    float res, tmp;

    int i, j, k;

    for (i = 0; i < 64; i++) {

        res = *input++;

        for (j = 0; j < factor; j++) {

            for (k = 0; k < 5; k++) {

                tmp = hist[k][0] * iir[k][0] + hist[k][1] * iir[k][1] + res;

                res = hist[k][0] * iir[k][2] + hist[k][1] * iir[k][3] + tmp;

                hist[k][0] = hist[k][1];

                hist[k][1] = tmp;

            }

            *output++ = res;

            res = 0;

        }

    }

}

av_cold void ff_dcadsp_init(DCADSPContext *s)

{

    s->decode_hf     = decode_hf_c;

    s->decode_joint  = decode_joint_c;

    s->lfe_fir_float[0] = lfe_fir0_float_c;

    s->lfe_fir_float[1] = lfe_fir1_float_c;

    s->lfe_x96_float    = lfe_x96_float_c;

    s->sub_qmf_float[0] = sub_qmf32_float_c;

    s->sub_qmf_float[1] = sub_qmf64_float_c;

    s->lfe_fir_fixed    = lfe_fir_fixed_c;

    s->lfe_x96_fixed    = lfe_x96_fixed_c;

    s->sub_qmf_fixed[0] = sub_qmf32_fixed_c;

    s->sub_qmf_fixed[1] = sub_qmf64_fixed_c;

    s->decor   = decor_c;

    s->dmix_sub_xch   = dmix_sub_xch_c;

    s->dmix_sub       = dmix_sub_c;

    s->dmix_add       = dmix_add_c;

    s->dmix_scale     = dmix_scale_c;

    s->dmix_scale_inv = dmix_scale_inv_c;

    s->assemble_freq_bands = assemble_freq_bands_c;

    s->lbr_bank = lbr_bank_c;

    s->lfe_iir = lfe_iir_c;

#if ARCH_X86

    ff_dcadsp_init_x86(s);

#endif

}