text: /*
   * ATRAC3 compatible decoder
   * Copyright (c) 2006-2008 Maxim Poliakovski
   * Copyright (c) 2006-2008 Benjamin Larsson
   *
   * 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
   */
  
  /**
   * @file
   * ATRAC3 compatible decoder.
   * This decoder handles Sony's ATRAC3 data.
   *
   * Container formats used to store ATRAC3 data:
   * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
   *
   * To use this decoder, a calling application must supply the extradata
   * bytes provided in the containers above.
   */
  
  #include <math.h>
  #include <stddef.h>
  #include <stdio.h>
  
  #include "libavutil/attributes.h"
  #include "libavutil/float_dsp.h"
  #include "libavutil/libm.h"
  #include "avcodec.h"
  #include "bytestream.h"
  #include "fft.h"
  #include "fmtconvert.h"
  #include "get_bits.h"
  #include "internal.h"
  
  #include "atrac.h"
  #include "atrac3data.h"
  
  #define JOINT_STEREO    0x12
  #define STEREO          0x2
  
  #define SAMPLES_PER_FRAME 1024
  #define MDCT_SIZE          512
  
  typedef struct GainBlock {
      AtracGainInfo g_block[4];
  } GainBlock;
  
  typedef struct TonalComponent {
      int pos;
      int num_coefs;
      float coef[8];
  } TonalComponent;
  
  typedef struct ChannelUnit {
      int            bands_coded;
      int            num_components;
      float          prev_frame[SAMPLES_PER_FRAME];
      int            gc_blk_switch;
      TonalComponent components[64];
      GainBlock      gain_block[2];
  
      DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
      DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
  
      float          delay_buf1[46]; ///<qmf delay buffers
      float          delay_buf2[46];
      float          delay_buf3[46];
  } ChannelUnit;
  
  typedef struct ATRAC3Context {
      GetBitContext gb;
      //@{
      /** stream data */
      int coding_mode;
  
      ChannelUnit *units;
      //@}
      //@{
      /** joint-stereo related variables */
      int matrix_coeff_index_prev[4];
      int matrix_coeff_index_now[4];
      int matrix_coeff_index_next[4];
      int weighting_delay[6];
      //@}
      //@{
      /** data buffers */
      uint8_t *decoded_bytes_buffer;
      float temp_buf[1070];
      //@}
      //@{
      /** extradata */
      int scrambled_stream;
      //@}
  
      AtracGCContext    gainc_ctx;
      FFTContext        mdct_ctx;
      FmtConvertContext fmt_conv;
      AVFloatDSPContext fdsp;
  } ATRAC3Context;
  
  static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
  static VLC_TYPE atrac3_vlc_table[4096][2];
  static VLC   spectral_coeff_tab[7];
  
  /**
   * Regular 512 points IMDCT without overlapping, with the exception of the
   * swapping of odd bands caused by the reverse spectra of the QMF.
   *
   * @param odd_band  1 if the band is an odd band
   */
  static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
  {
      int i;
  
      if (odd_band) {
          /**
           * Reverse the odd bands before IMDCT, this is an effect of the QMF
           * transform or it gives better compression to do it this way.
           * FIXME: It should be possible to handle this in imdct_calc
           * for that to happen a modification of the prerotation step of
           * all SIMD code and C code is needed.
           * Or fix the functions before so they generate a pre reversed spectrum.
           */
          for (i = 0; i < 128; i++)
              FFSWAP(float, input[i], input[255 - i]);
      }
  
      q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
  
      /* Perform windowing on the output. */
      q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE);
  }
  
  /*
   * indata descrambling, only used for data coming from the rm container
   */
  static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
  {
      int i, off;
      uint32_t c;
      const uint32_t *buf;
      uint32_t *output = (uint32_t *)out;
  
      off = (intptr_t)input & 3;
      buf = (const uint32_t *)(input - off);
      if (off)
          c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
      else
          c = av_be2ne32(0x537F6103U);
      bytes += 3 + off;
      for (i = 0; i < bytes / 4; i++)
          output[i] = c ^ buf[i];
  
      if (off)
          avpriv_request_sample(NULL, "Offset of %d", off);
  
      return off;
  }
  
  static av_cold void init_imdct_window(void)
  {
      int i, j;
  
      /* generate the mdct window, for details see
       * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
      for (i = 0, j = 255; i < 128; i++, j--) {
          float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
          float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
          float w  = 0.5 * (wi * wi + wj * wj);
          mdct_window[i] = mdct_window[511 - i] = wi / w;
          mdct_window[j] = mdct_window[511 - j] = wj / w;
      }
  }
  
  static av_cold int atrac3_decode_close(AVCodecContext *avctx)
  {
      ATRAC3Context *q = avctx->priv_data;
  
      av_free(q->units);
      av_free(q->decoded_bytes_buffer);
  
      ff_mdct_end(&q->mdct_ctx);
  
      return 0;
  }
  
  /**
   * Mantissa decoding
   *
   * @param selector     which table the output values are coded with
   * @param coding_flag  constant length coding or variable length coding
   * @param mantissas    mantissa output table
   * @param num_codes    number of values to get
   */
  static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
                                         int coding_flag, int *mantissas,
                                         int num_codes)
  {
      int i, code, huff_symb;
  
      if (selector == 1)
          num_codes /= 2;
  
      if (coding_flag != 0) {
          /* constant length coding (CLC) */
          int num_bits = clc_length_tab[selector];
  
          if (selector > 1) {
              for (i = 0; i < num_codes; i++) {
                  if (num_bits)
                      code = get_sbits(gb, num_bits);
                  else
                      code = 0;
                  mantissas[i] = code;
              }
          } else {
              for (i = 0; i < num_codes; i++) {
                  if (num_bits)
                      code = get_bits(gb, num_bits); // num_bits is always 4 in this case
                  else
                      code = 0;
                  mantissas[i * 2    ] = mantissa_clc_tab[code >> 2];
                  mantissas[i * 2 + 1] = mantissa_clc_tab[code &  3];
              }
          }
      } else {
          /* variable length coding (VLC) */
          if (selector != 1) {
              for (i = 0; i < num_codes; i++) {
                  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
                                       spectral_coeff_tab[selector-1].bits, 3);
                  huff_symb += 1;
                  code = huff_symb >> 1;
                  if (huff_symb & 1)
                      code = -code;
                  mantissas[i] = code;
              }
          } else {
              for (i = 0; i < num_codes; i++) {
                  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
                                       spectral_coeff_tab[selector - 1].bits, 3);
                  mantissas[i * 2    ] = mantissa_vlc_tab[huff_symb * 2    ];
                  mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
              }
          }
      }
  }
  
  /**
   * Restore the quantized band spectrum coefficients
   *
   * @return subband count, fix for broken specification/files
   */
  static int decode_spectrum(GetBitContext *gb, float *output)
  {
      int num_subbands, coding_mode, i, j, first, last, subband_size;
      int subband_vlc_index[32], sf_index[32];
      int mantissas[128];
      float scale_factor;
  
      num_subbands = get_bits(gb, 5);  // number of coded subbands
      coding_mode  = get_bits1(gb);    // coding Mode: 0 - VLC/ 1-CLC
  
      /* get the VLC selector table for the subbands, 0 means not coded */
      for (i = 0; i <= num_subbands; i++)
          subband_vlc_index[i] = get_bits(gb, 3);
  
      /* read the scale factor indexes from the stream */
      for (i = 0; i <= num_subbands; i++) {
          if (subband_vlc_index[i] != 0)
              sf_index[i] = get_bits(gb, 6);
      }
  
      for (i = 0; i <= num_subbands; i++) {
          first = subband_tab[i    ];
          last  = subband_tab[i + 1];
  
          subband_size = last - first;
  
          if (subband_vlc_index[i] != 0) {
              /* decode spectral coefficients for this subband */
              /* TODO: This can be done faster is several blocks share the
               * same VLC selector (subband_vlc_index) */
              read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
                                         mantissas, subband_size);
  
              /* decode the scale factor for this subband */
              scale_factor = ff_atrac_sf_table[sf_index[i]] *
                             inv_max_quant[subband_vlc_index[i]];
  
              /* inverse quantize the coefficients */
              for (j = 0; first < last; first++, j++)
                  output[first] = mantissas[j] * scale_factor;
          } else {
              /* this subband was not coded, so zero the entire subband */
              memset(output + first, 0, subband_size * sizeof(*output));
          }
      }
  
      /* clear the subbands that were not coded */
      first = subband_tab[i];
      memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
      return num_subbands;
  }
  
  /**
   * Restore the quantized tonal components
   *
   * @param components tonal components
   * @param num_bands  number of coded bands
   */
  static int decode_tonal_components(GetBitContext *gb,
                                     TonalComponent *components, int num_bands)
  {
      int i, b, c, m;
      int nb_components, coding_mode_selector, coding_mode;
      int band_flags[4], mantissa[8];
      int component_count = 0;
  
      nb_components = get_bits(gb, 5);
  
      /* no tonal components */
      if (nb_components == 0)
          return 0;
  
      coding_mode_selector = get_bits(gb, 2);
      if (coding_mode_selector == 2)
          return AVERROR_INVALIDDATA;
  
      coding_mode = coding_mode_selector & 1;
  
      for (i = 0; i < nb_components; i++) {
          int coded_values_per_component, quant_step_index;
  
          for (b = 0; b <= num_bands; b++)
              band_flags[b] = get_bits1(gb);
  
          coded_values_per_component = get_bits(gb, 3);
  
          quant_step_index = get_bits(gb, 3);
          if (quant_step_index <= 1)
              return AVERROR_INVALIDDATA;
  
          if (coding_mode_selector == 3)
              coding_mode = get_bits1(gb);
  
          for (b = 0; b < (num_bands + 1) * 4; b++) {
              int coded_components;
  
              if (band_flags[b >> 2] == 0)
                  continue;
  
              coded_components = get_bits(gb, 3);
  
              for (c = 0; c < coded_components; c++) {
                  TonalComponent *cmp = &components[component_count];
                  int sf_index, coded_values, max_coded_values;
                  float scale_factor;
  
                  sf_index = get_bits(gb, 6);
                  if (component_count >= 64)
                      return AVERROR_INVALIDDATA;
  
                  cmp->pos = b * 64 + get_bits(gb, 6);
  
                  max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
                  coded_values     = coded_values_per_component + 1;
                  coded_values     = FFMIN(max_coded_values, coded_values);
  
                  scale_factor = ff_atrac_sf_table[sf_index] *
                                 inv_max_quant[quant_step_index];
  
                  read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
                                             mantissa, coded_values);
  
                  cmp->num_coefs = coded_values;
  
                  /* inverse quant */
                  for (m = 0; m < coded_values; m++)
                      cmp->coef[m] = mantissa[m] * scale_factor;
  
                  component_count++;
              }
          }
      }
  
      return component_count;
  }
  
  /**
   * Decode gain parameters for the coded bands
   *
   * @param block      the gainblock for the current band
   * @param num_bands  amount of coded bands
   */
  static int decode_gain_control(GetBitContext *gb, GainBlock *block,
                                 int num_bands)
  {
      int b, j;
      int *level, *loc;
  
      AtracGainInfo *gain = block->g_block;
  
      for (b = 0; b <= num_bands; b++) {
          gain[b].num_points = get_bits(gb, 3);
          level              = gain[b].lev_code;
          loc                = gain[b].loc_code;
  
          for (j = 0; j < gain[b].num_points; j++) {
              level[j] = get_bits(gb, 4);
              loc[j]   = get_bits(gb, 5);
              if (j && loc[j] <= loc[j - 1])
                  return AVERROR_INVALIDDATA;
          }
      }
  
      /* Clear the unused blocks. */
      for (; b < 4 ; b++)
          gain[b].num_points = 0;
  
      return 0;
  }
  
  /**
   * Combine the tonal band spectrum and regular band spectrum
   *
   * @param spectrum        output spectrum buffer
   * @param num_components  number of tonal components
   * @param components      tonal components for this band
   * @return                position of the last tonal coefficient
   */
  static int add_tonal_components(float *spectrum, int num_components,
                                  TonalComponent *components)
  {
      int i, j, last_pos = -1;
      float *input, *output;
  
      for (i = 0; i < num_components; i++) {
          last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
          input    = components[i].coef;
          output   = &spectrum[components[i].pos];
  
          for (j = 0; j < components[i].num_coefs; j++)
              output[j] += input[j];
      }
  
      return last_pos;
  }
  
  #define INTERPOLATE(old, new, nsample) \
      ((old) + (nsample) * 0.125 * ((new) - (old)))
  
  static void reverse_matrixing(float *su1, float *su2, int *prev_code,
                                int *curr_code)
  {
      int i, nsample, band;
      float mc1_l, mc1_r, mc2_l, mc2_r;
  
      for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
          int s1 = prev_code[i];
          int s2 = curr_code[i];
          nsample = band;
  
          if (s1 != s2) {
              /* Selector value changed, interpolation needed. */
              mc1_l = matrix_coeffs[s1 * 2    ];
              mc1_r = matrix_coeffs[s1 * 2 + 1];
              mc2_l = matrix_coeffs[s2 * 2    ];
              mc2_r = matrix_coeffs[s2 * 2 + 1];
  
              /* Interpolation is done over the first eight samples. */
              for (; nsample < band + 8; nsample++) {
                  float c1 = su1[nsample];
                  float c2 = su2[nsample];
                  c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
                       c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
                  su1[nsample] = c2;
                  su2[nsample] = c1 * 2.0 - c2;
              }
          }
  
          /* Apply the matrix without interpolation. */
          switch (s2) {
          case 0:     /* M/S decoding */
              for (; nsample < band + 256; nsample++) {
                  float c1 = su1[nsample];
                  float c2 = su2[nsample];
                  su1[nsample] =  c2       * 2.0;
                  su2[nsample] = (c1 - c2) * 2.0;
              }
              break;
          case 1:
              for (; nsample < band + 256; nsample++) {
                  float c1 = su1[nsample];
                  float c2 = su2[nsample];
                  su1[nsample] = (c1 + c2) *  2.0;
                  su2[nsample] =  c2       * -2.0;
              }
              break;
          case 2:
          case 3:
              for (; nsample < band + 256; nsample++) {
                  float c1 = su1[nsample];
                  float c2 = su2[nsample];
                  su1[nsample] = c1 + c2;
                  su2[nsample] = c1 - c2;
              }
              break;
          default:
              av_assert1(0);
          }
      }
  }
  
  static void get_channel_weights(int index, int flag, float ch[2])
  {
      if (index == 7) {
          ch[0] = 1.0;
          ch[1] = 1.0;
      } else {
          ch[0] = (index & 7) / 7.0;
          ch[1] = sqrt(2 - ch[0] * ch[0]);
          if (flag)
              FFSWAP(float, ch[0], ch[1]);
      }
  }
  
  static void channel_weighting(float *su1, float *su2, int *p3)
  {
      int band, nsample;
      /* w[x][y] y=0 is left y=1 is right */
      float w[2][2];
  
      if (p3[1] != 7 || p3[3] != 7) {
          get_channel_weights(p3[1], p3[0], w[0]);
          get_channel_weights(p3[3], p3[2], w[1]);
  
          for (band = 256; band < 4 * 256; band += 256) {
              for (nsample = band; nsample < band + 8; nsample++) {
                  su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
                  su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
              }
              for(; nsample < band + 256; nsample++) {
                  su1[nsample] *= w[1][0];
                  su2[nsample] *= w[1][1];
              }
          }
      }
  }
  
  /**
   * Decode a Sound Unit
   *
   * @param snd           the channel unit to be used
   * @param output        the decoded samples before IQMF in float representation
   * @param channel_num   channel number
   * @param coding_mode   the coding mode (JOINT_STEREO or regular stereo/mono)
   */
  static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
                                       ChannelUnit *snd, float *output,
                                       int channel_num, int coding_mode)
  {
      int band, ret, num_subbands, last_tonal, num_bands;
      GainBlock *gain1 = &snd->gain_block[    snd->gc_blk_switch];
      GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
  
      if (coding_mode == JOINT_STEREO && channel_num == 1) {
          if (get_bits(gb, 2) != 3) {
              av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
              return AVERROR_INVALIDDATA;
          }
      } else {
          if (get_bits(gb, 6) != 0x28) {
              av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
              return AVERROR_INVALIDDATA;
          }
      }
  
      /* number of coded QMF bands */
      snd->bands_coded = get_bits(gb, 2);
  
      ret = decode_gain_control(gb, gain2, snd->bands_coded);
      if (ret)
          return ret;
  
      snd->num_components = decode_tonal_components(gb, snd->components,
                                                    snd->bands_coded);
      if (snd->num_components < 0)
          return snd->num_components;
  
      num_subbands = decode_spectrum(gb, snd->spectrum);
  
      /* Merge the decoded spectrum and tonal components. */
      last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
                                        snd->components);
  
  
      /* calculate number of used MLT/QMF bands according to the amount of coded
         spectral lines */
      num_bands = (subband_tab[num_subbands] - 1) >> 8;
      if (last_tonal >= 0)
          num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
  
  
      /* Reconstruct time domain samples. */
      for (band = 0; band < 4; band++) {
          /* Perform the IMDCT step without overlapping. */
          if (band <= num_bands)
              imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
          else
              memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
  
          /* gain compensation and overlapping */
          ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
                                     &snd->prev_frame[band * 256],
                                     &gain1->g_block[band], &gain2->g_block[band],
                                     256, &output[band * 256]);
      }
  
      /* Swap the gain control buffers for the next frame. */
      snd->gc_blk_switch ^= 1;
  
      return 0;
  }
  
  static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
                          float **out_samples)
  {
      ATRAC3Context *q = avctx->priv_data;
      int ret, i;
      uint8_t *ptr1;
  
      if (q->coding_mode == JOINT_STEREO) {
          /* channel coupling mode */
          /* decode Sound Unit 1 */
          init_get_bits(&q->gb, databuf, avctx->block_align * 8);
  
          ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
                                          JOINT_STEREO);
          if (ret != 0)
              return ret;
  
          /* Framedata of the su2 in the joint-stereo mode is encoded in
           * reverse byte order so we need to swap it first. */
          if (databuf == q->decoded_bytes_buffer) {
              uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
              ptr1          = q->decoded_bytes_buffer;
              for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
                  FFSWAP(uint8_t, *ptr1, *ptr2);
          } else {
              const uint8_t *ptr2 = databuf + avctx->block_align - 1;
              for (i = 0; i < avctx->block_align; i++)
                  q->decoded_bytes_buffer[i] = *ptr2--;
          }
  
          /* Skip the sync codes (0xF8). */
          ptr1 = q->decoded_bytes_buffer;
          for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
              if (i >= avctx->block_align)
                  return AVERROR_INVALIDDATA;
          }
  
  
          /* set the bitstream reader at the start of the second Sound Unit*/
          init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1);
  
          /* Fill the Weighting coeffs delay buffer */
          memmove(q->weighting_delay, &q->weighting_delay[2],
                  4 * sizeof(*q->weighting_delay));
          q->weighting_delay[4] = get_bits1(&q->gb);
          q->weighting_delay[5] = get_bits(&q->gb, 3);
  
          for (i = 0; i < 4; i++) {
              q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
              q->matrix_coeff_index_now[i]  = q->matrix_coeff_index_next[i];
              q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
          }
  
          /* Decode Sound Unit 2. */
          ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
                                          out_samples[1], 1, JOINT_STEREO);
          if (ret != 0)
              return ret;
  
          /* Reconstruct the channel coefficients. */
          reverse_matrixing(out_samples[0], out_samples[1],
                            q->matrix_coeff_index_prev,
                            q->matrix_coeff_index_now);
  
          channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
      } else {
          /* normal stereo mode or mono */
          /* Decode the channel sound units. */
          for (i = 0; i < avctx->channels; i++) {
              /* Set the bitstream reader at the start of a channel sound unit. */
              init_get_bits(&q->gb,
                            databuf + i * avctx->block_align / avctx->channels,
                            avctx->block_align * 8 / avctx->channels);
  
              ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
                                              out_samples[i], i, q->coding_mode);
              if (ret != 0)
                  return ret;
          }
      }
  
      /* Apply the iQMF synthesis filter. */
      for (i = 0; i < avctx->channels; i++) {
          float *p1 = out_samples[i];
          float *p2 = p1 + 256;
          float *p3 = p2 + 256;
          float *p4 = p3 + 256;
          ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
          ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
          ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
      }
  
      return 0;
  }
  
  static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
                                 int *got_frame_ptr, AVPacket *avpkt)
  {
      AVFrame *frame     = data;
      const uint8_t *buf = avpkt->data;
      int buf_size = avpkt->size;
      ATRAC3Context *q = avctx->priv_data;
      int ret;
      const uint8_t *databuf;
  
      if (buf_size < avctx->block_align) {
          av_log(avctx, AV_LOG_ERROR,
                 "Frame too small (%d bytes). Truncated file?\n", buf_size);
          return AVERROR_INVALIDDATA;
      }
  
      /* get output buffer */
      frame->nb_samples = SAMPLES_PER_FRAME;
      if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
          return ret;
  
      /* Check if we need to descramble and what buffer to pass on. */
      if (q->scrambled_stream) {
          decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
          databuf = q->decoded_bytes_buffer;
      } else {
          databuf = buf;
      }
  
      ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
      if (ret) {
          av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
          return ret;
      }
  
      *got_frame_ptr = 1;
  
      return avctx->block_align;
  }
  
  static av_cold void atrac3_init_static_data(void)
  {
      int i;
  
      init_imdct_window();
      ff_atrac_generate_tables();
  
      /* Initialize the VLC tables. */
      for (i = 0; i < 7; i++) {
          spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
          spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
                                                  atrac3_vlc_offs[i    ];
          init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
                   huff_bits[i],  1, 1,
                   huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
      }
  }
  
  static av_cold int atrac3_decode_init(AVCodecContext *avctx)
  {
      static int static_init_done;
      int i, ret;
      int version, delay, samples_per_frame, frame_factor;
      const uint8_t *edata_ptr = avctx->extradata;
      ATRAC3Context *q = avctx->priv_data;
  
      if (avctx->channels <= 0 || avctx->channels > 2) {
          av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
          return AVERROR(EINVAL);
      }
  
      if (!static_init_done)
          atrac3_init_static_data();
      static_init_done = 1;
  
      /* Take care of the codec-specific extradata. */
      if (avctx->extradata_size == 14) {
          /* Parse the extradata, WAV format */
          av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
                 bytestream_get_le16(&edata_ptr));  // Unknown value always 1
          edata_ptr += 4;                             // samples per channel
          q->coding_mode = bytestream_get_le16(&edata_ptr);
          av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
                 bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
          frame_factor = bytestream_get_le16(&edata_ptr);  // Unknown always 1
          av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
                 bytestream_get_le16(&edata_ptr));  // Unknown always 0
  
          /* setup */
          samples_per_frame    = SAMPLES_PER_FRAME * avctx->channels;
          version              = 4;
          delay                = 0x88E;
          q->coding_mode       = q->coding_mode ? JOINT_STEREO : STEREO;
          q->scrambled_stream  = 0;
  
          if (avctx->block_align !=  96 * avctx->channels * frame_factor &&
              avctx->block_align != 152 * avctx->channels * frame_factor &&
              avctx->block_align != 192 * avctx->channels * frame_factor) {
              av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
                     "configuration %d/%d/%d\n", avctx->block_align,
                     avctx->channels, frame_factor);
              return AVERROR_INVALIDDATA;
          }
      } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
          /* Parse the extradata, RM format. */
          version                = bytestream_get_be32(&edata_ptr);
          samples_per_frame      = bytestream_get_be16(&edata_ptr);
          delay                  = bytestream_get_be16(&edata_ptr);
          q->coding_mode         = bytestream_get_be16(&edata_ptr);
          q->scrambled_stream    = 1;
  
      } else {
          av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
                 avctx->extradata_size);
          return AVERROR(EINVAL);
      }
  
      /* Check the extradata */
  
      if (version != 4) {
          av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
          return AVERROR_INVALIDDATA;
      }
  
      if (samples_per_frame != SAMPLES_PER_FRAME &&
          samples_per_frame != SAMPLES_PER_FRAME * 2) {
          av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
                 samples_per_frame);
          return AVERROR_INVALIDDATA;
      }
  
      if (delay != 0x88E) {
          av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
                 delay);
          return AVERROR_INVALIDDATA;
      }
  
      if (q->coding_mode == STEREO)
          av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
      else if (q->coding_mode == JOINT_STEREO) {
          if (avctx->channels != 2) {
              av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");
              return AVERROR_INVALIDDATA;
          }
          av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
      } else {
          av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
                 q->coding_mode);
          return AVERROR_INVALIDDATA;
      }
  
      if (avctx->block_align >= UINT_MAX / 2)
          return AVERROR(EINVAL);
  
      q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
                                           FF_INPUT_BUFFER_PADDING_SIZE);
      if (!q->decoded_bytes_buffer)
          return AVERROR(ENOMEM);
  
      avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
  
      /* initialize the MDCT transform */
      if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
          av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
          av_freep(&q->decoded_bytes_buffer);
          return ret;
      }
  
      /* init the joint-stereo decoding data */
      q->weighting_delay[0] = 0;
      q->weighting_delay[1] = 7;
      q->weighting_delay[2] = 0;
      q->weighting_delay[3] = 7;
      q->weighting_delay[4] = 0;
      q->weighting_delay[5] = 7;
  
      for (i = 0; i < 4; i++) {
          q->matrix_coeff_index_prev[i] = 3;
          q->matrix_coeff_index_now[i]  = 3;
          q->matrix_coeff_index_next[i] = 3;
      }
  
      ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
      avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
      ff_fmt_convert_init(&q->fmt_conv, avctx);
  
      q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
      if (!q->units) {
          atrac3_decode_close(avctx);
          return AVERROR(ENOMEM);
      }
  
      return 0;
  }
  
  AVCodec ff_atrac3_decoder = {
      .name             = "atrac3",
      .long_name        = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
      .type             = AVMEDIA_TYPE_AUDIO,
      .id               = AV_CODEC_ID_ATRAC3,
      .priv_data_size   = sizeof(ATRAC3Context),
      .init             = atrac3_decode_init,
      .close            = atrac3_decode_close,
      .decode           = atrac3_decode_frame,
      .capabilities     = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
      .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                          AV_SAMPLE_FMT_NONE },
  };
  