/*
  * Copyright (c) 2004 Gildas Bazin
  * Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
  *
  * 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 "config.h"

 #include "libavutil/attributes.h"

 #include "libavutil/intreadwrite.h"

 #include "dcadsp.h"
 

 static void decode_hf_c(float dst[DCA_SUBBANDS][8],
                         const int32_t vq_num[DCA_SUBBANDS],
                         const int8_t hf_vq[1024][32], intptr_t vq_offset,
                         int32_t scale[DCA_SUBBANDS][2],
                         intptr_t start, intptr_t end)

 {

     int i, l;
 
     for (l = start; l < end; l++) {
         /* 1 vector -> 32 samples but we only need the 8 samples
          * for this subsubframe. */
         const int8_t *ptr = &hf_vq[vq_num[l]][vq_offset];
         float fscale = scale[l][0] * (1 / 16.0);
         for (i = 0; i < 8; i++)
             dst[l][i] = ptr[i] * fscale;
     }

 }
 

 static inline void
 dca_lfe_fir(float *out, const float *in, const float *coefs,

             int decifactor)

 {

     float *out2    = out + 2 * decifactor - 1;

     int num_coeffs = 256 / decifactor;

     int j, k;
 
     /* One decimated sample generates 2*decifactor interpolated ones */
     for (k = 0; k < decifactor; k++) {
         float v0 = 0.0;
         float v1 = 0.0;

         for (j = 0; j < num_coeffs; j++, coefs++) {
             v0 += in[-j] * *coefs;
             v1 += in[j + 1 - num_coeffs] * *coefs;

         }

         *out++  = v0;

         *out2-- = v1;

     }
 }
 

 static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
                                 SynthFilterContext *synth, FFTContext *imdct,
                                 float synth_buf_ptr[512],
                                 int *synth_buf_offset, float synth_buf2[32],
                                 const float window[512], float *samples_out,
                                 float raXin[32], float scale)
 {
     int i;
     int subindex;
 
     for (i = sb_act; i < 32; i++)
         raXin[i] = 0.0;
 
     /* Reconstructed channel sample index */
     for (subindex = 0; subindex < 8; subindex++) {
         /* Load in one sample from each subband and clear inactive subbands */
         for (i = 0; i < sb_act; i++) {
             unsigned sign = (i - 1) & 2;
             uint32_t v    = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
             AV_WN32A(&raXin[i], v);
         }
 
         synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
                                   synth_buf2, window, samples_out, raXin, scale);
         samples_out += 32;
     }
 }
 

 static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)

 {

     dca_lfe_fir(out, in, coefs, 32);

 }
 

 static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)

 {

     dca_lfe_fir(out, in, coefs, 64);

 }
 

 av_cold void ff_dcadsp_init(DCADSPContext *s)

 {

     s->lfe_fir[0] = dca_lfe_fir0_c;
     s->lfe_fir[1] = dca_lfe_fir1_c;

     s->qmf_32_subbands = dca_qmf_32_subbands;

     s->decode_hf = decode_hf_c;

     if (ARCH_ARM) ff_dcadsp_init_arm(s);

     if (ARCH_X86) ff_dcadsp_init_x86(s);

 }