text: /*
   * COOK compatible decoder
   * Copyright (c) 2003 Sascha Sommer
   * Copyright (c) 2005 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
   * Cook compatible decoder. Bastardization of the G.722.1 standard.
   * This decoder handles RealNetworks, RealAudio G2 data.
   * Cook is identified by the codec name cook in RM files.
   *
   * To use this decoder, a calling application must supply the extradata
   * bytes provided from the RM container; 8+ bytes for mono streams and
   * 16+ for stereo streams (maybe more).
   *
   * Codec technicalities (all this assume a buffer length of 1024):
   * Cook works with several different techniques to achieve its compression.
   * In the timedomain the buffer is divided into 8 pieces and quantized. If
   * two neighboring pieces have different quantization index a smooth
   * quantization curve is used to get a smooth overlap between the different
   * pieces.
   * To get to the transformdomain Cook uses a modulated lapped transform.
   * The transform domain has 50 subbands with 20 elements each. This
   * means only a maximum of 50*20=1000 coefficients are used out of the 1024
   * available.
   */
  
  #include "libavutil/channel_layout.h"
  #include "libavutil/lfg.h"
  
  #include "audiodsp.h"
  #include "avcodec.h"
  #include "get_bits.h"
  #include "bytestream.h"
  #include "fft.h"
  #include "internal.h"
  #include "sinewin.h"
  #include "unary.h"
  
  #include "cookdata.h"
  
  /* the different Cook versions */
  #define MONO            0x1000001
  #define STEREO          0x1000002
  #define JOINT_STEREO    0x1000003
  #define MC_COOK         0x2000000   // multichannel Cook, not supported
  
  #define SUBBAND_SIZE    20
  #define MAX_SUBPACKETS   5
  
  typedef struct cook_gains {
      int *now;
      int *previous;
  } cook_gains;
  
  typedef struct COOKSubpacket {
      int                 ch_idx;
      int                 size;
      int                 num_channels;
      int                 cookversion;
      int                 subbands;
      int                 js_subband_start;
      int                 js_vlc_bits;
      int                 samples_per_channel;
      int                 log2_numvector_size;
      unsigned int        channel_mask;
      VLC                 channel_coupling;
      int                 joint_stereo;
      int                 bits_per_subpacket;
      int                 bits_per_subpdiv;
      int                 total_subbands;
      int                 numvector_size;       // 1 << log2_numvector_size;
  
      float               mono_previous_buffer1[1024];
      float               mono_previous_buffer2[1024];
  
      cook_gains          gains1;
      cook_gains          gains2;
      int                 gain_1[9];
      int                 gain_2[9];
      int                 gain_3[9];
      int                 gain_4[9];
  } COOKSubpacket;
  
  typedef struct cook {
      /*
       * The following 5 functions provide the lowlevel arithmetic on
       * the internal audio buffers.
       */
      void (*scalar_dequant)(struct cook *q, int index, int quant_index,
                             int *subband_coef_index, int *subband_coef_sign,
                             float *mlt_p);
  
      void (*decouple)(struct cook *q,
                       COOKSubpacket *p,
                       int subband,
                       float f1, float f2,
                       float *decode_buffer,
                       float *mlt_buffer1, float *mlt_buffer2);
  
      void (*imlt_window)(struct cook *q, float *buffer1,
                          cook_gains *gains_ptr, float *previous_buffer);
  
      void (*interpolate)(struct cook *q, float *buffer,
                          int gain_index, int gain_index_next);
  
      void (*saturate_output)(struct cook *q, float *out);
  
      AVCodecContext*     avctx;
      AudioDSPContext     adsp;
      GetBitContext       gb;
      /* stream data */
      int                 num_vectors;
      int                 samples_per_channel;
      /* states */
      AVLFG               random_state;
      int                 discarded_packets;
  
      /* transform data */
      FFTContext          mdct_ctx;
      float*              mlt_window;
  
      /* VLC data */
      VLC                 envelope_quant_index[13];
      VLC                 sqvh[7];          // scalar quantization
  
      /* generatable tables and related variables */
      int                 gain_size_factor;
      float               gain_table[23];
  
      /* data buffers */
  
      uint8_t*            decoded_bytes_buffer;
      DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
      float               decode_buffer_1[1024];
      float               decode_buffer_2[1024];
      float               decode_buffer_0[1060]; /* static allocation for joint decode */
  
      const float         *cplscales[5];
      int                 num_subpackets;
      COOKSubpacket       subpacket[MAX_SUBPACKETS];
  } COOKContext;
  
  static float     pow2tab[127];
  static float rootpow2tab[127];
  
  /*************** init functions ***************/
  
  /* table generator */
  static av_cold void init_pow2table(void)
  {
      int i;
      for (i = -63; i < 64; i++) {
          pow2tab[63 + i] = pow(2, i);
          rootpow2tab[63 + i] = sqrt(pow(2, i));
      }
  }
  
  /* table generator */
  static av_cold void init_gain_table(COOKContext *q)
  {
      int i;
      q->gain_size_factor = q->samples_per_channel / 8;
      for (i = 0; i < 23; i++)
          q->gain_table[i] = pow(pow2tab[i + 52],
                                 (1.0 / (double) q->gain_size_factor));
  }
  
  
  static av_cold int init_cook_vlc_tables(COOKContext *q)
  {
      int i, result;
  
      result = 0;
      for (i = 0; i < 13; i++) {
          result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
                             envelope_quant_index_huffbits[i], 1, 1,
                             envelope_quant_index_huffcodes[i], 2, 2, 0);
      }
      av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
      for (i = 0; i < 7; i++) {
          result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
                             cvh_huffbits[i], 1, 1,
                             cvh_huffcodes[i], 2, 2, 0);
      }
  
      for (i = 0; i < q->num_subpackets; i++) {
          if (q->subpacket[i].joint_stereo == 1) {
              result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
                                 (1 << q->subpacket[i].js_vlc_bits) - 1,
                                 ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
                                 ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
              av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
          }
      }
  
      av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
      return result;
  }
  
  static av_cold int init_cook_mlt(COOKContext *q)
  {
      int j, ret;
      int mlt_size = q->samples_per_channel;
  
      if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)
          return AVERROR(ENOMEM);
  
      /* Initialize the MLT window: simple sine window. */
      ff_sine_window_init(q->mlt_window, mlt_size);
      for (j = 0; j < mlt_size; j++)
          q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
  
      /* Initialize the MDCT. */
      if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
          av_freep(&q->mlt_window);
          return ret;
      }
      av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
             av_log2(mlt_size) + 1);
  
      return 0;
  }
  
  static av_cold void init_cplscales_table(COOKContext *q)
  {
      int i;
      for (i = 0; i < 5; i++)
          q->cplscales[i] = cplscales[i];
  }
  
  /*************** init functions end ***********/
  
  #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
  #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
  
  /**
   * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
   * Why? No idea, some checksum/error detection method maybe.
   *
   * Out buffer size: extra bytes are needed to cope with
   * padding/misalignment.
   * Subpackets passed to the decoder can contain two, consecutive
   * half-subpackets, of identical but arbitrary size.
   *          1234 1234 1234 1234  extraA extraB
   * Case 1:  AAAA BBBB              0      0
   * Case 2:  AAAA ABBB BB--         3      3
   * Case 3:  AAAA AABB BBBB         2      2
   * Case 4:  AAAA AAAB BBBB BB--    1      5
   *
   * Nice way to waste CPU cycles.
   *
   * @param inbuffer  pointer to byte array of indata
   * @param out       pointer to byte array of outdata
   * @param bytes     number of bytes
   */
  static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
  {
      static const uint32_t tab[4] = {
          AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
          AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
      };
      int i, off;
      uint32_t c;
      const uint32_t *buf;
      uint32_t *obuf = (uint32_t *) out;
      /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
       * I'm too lazy though, should be something like
       * for (i = 0; i < bitamount / 64; i++)
       *     (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
       * Buffer alignment needs to be checked. */
  
      off = (intptr_t) inbuffer & 3;
      buf = (const uint32_t *) (inbuffer - off);
      c = tab[off];
      bytes += 3 + off;
      for (i = 0; i < bytes / 4; i++)
          obuf[i] = c ^ buf[i];
  
      return off;
  }
  
  static av_cold int cook_decode_close(AVCodecContext *avctx)
  {
      int i;
      COOKContext *q = avctx->priv_data;
      av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
  
      /* Free allocated memory buffers. */
      av_freep(&q->mlt_window);
      av_freep(&q->decoded_bytes_buffer);
  
      /* Free the transform. */
      ff_mdct_end(&q->mdct_ctx);
  
      /* Free the VLC tables. */
      for (i = 0; i < 13; i++)
          ff_free_vlc(&q->envelope_quant_index[i]);
      for (i = 0; i < 7; i++)
          ff_free_vlc(&q->sqvh[i]);
      for (i = 0; i < q->num_subpackets; i++)
          ff_free_vlc(&q->subpacket[i].channel_coupling);
  
      av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
  
      return 0;
  }
  
  /**
   * Fill the gain array for the timedomain quantization.
   *
   * @param gb          pointer to the GetBitContext
   * @param gaininfo    array[9] of gain indexes
   */
  static void decode_gain_info(GetBitContext *gb, int *gaininfo)
  {
      int i, n;
  
      n = get_unary(gb, 0, get_bits_left(gb));     // amount of elements*2 to update
  
      i = 0;
      while (n--) {
          int index = get_bits(gb, 3);
          int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
  
          while (i <= index)
              gaininfo[i++] = gain;
      }
      while (i <= 8)
          gaininfo[i++] = 0;
  }
  
  /**
   * Create the quant index table needed for the envelope.
   *
   * @param q                 pointer to the COOKContext
   * @param quant_index_table pointer to the array
   */
  static int decode_envelope(COOKContext *q, COOKSubpacket *p,
                             int *quant_index_table)
  {
      int i, j, vlc_index;
  
      quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
  
      for (i = 1; i < p->total_subbands; i++) {
          vlc_index = i;
          if (i >= p->js_subband_start * 2) {
              vlc_index -= p->js_subband_start;
          } else {
              vlc_index /= 2;
              if (vlc_index < 1)
                  vlc_index = 1;
          }
          if (vlc_index > 13)
              vlc_index = 13; // the VLC tables >13 are identical to No. 13
  
          j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
                       q->envelope_quant_index[vlc_index - 1].bits, 2);
          quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
          if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
              av_log(q->avctx, AV_LOG_ERROR,
                     "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
                     quant_index_table[i], i);
              return AVERROR_INVALIDDATA;
          }
      }
  
      return 0;
  }
  
  /**
   * Calculate the category and category_index vector.
   *
   * @param q                     pointer to the COOKContext
   * @param quant_index_table     pointer to the array
   * @param category              pointer to the category array
   * @param category_index        pointer to the category_index array
   */
  static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
                         int *category, int *category_index)
  {
      int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
      int exp_index2[102] = { 0 };
      int exp_index1[102] = { 0 };
  
      int tmp_categorize_array[128 * 2] = { 0 };
      int tmp_categorize_array1_idx = p->numvector_size;
      int tmp_categorize_array2_idx = p->numvector_size;
  
      bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
  
      if (bits_left > q->samples_per_channel)
          bits_left = q->samples_per_channel +
                      ((bits_left - q->samples_per_channel) * 5) / 8;
  
      bias = -32;
  
      /* Estimate bias. */
      for (i = 32; i > 0; i = i / 2) {
          num_bits = 0;
          index    = 0;
          for (j = p->total_subbands; j > 0; j--) {
              exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);
              index++;
              num_bits += expbits_tab[exp_idx];
          }
          if (num_bits >= bits_left - 32)
              bias += i;
      }
  
      /* Calculate total number of bits. */
      num_bits = 0;
      for (i = 0; i < p->total_subbands; i++) {
          exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);
          num_bits += expbits_tab[exp_idx];
          exp_index1[i] = exp_idx;
          exp_index2[i] = exp_idx;
      }
      tmpbias1 = tmpbias2 = num_bits;
  
      for (j = 1; j < p->numvector_size; j++) {
          if (tmpbias1 + tmpbias2 > 2 * bits_left) {  /* ---> */
              int max = -999999;
              index = -1;
              for (i = 0; i < p->total_subbands; i++) {
                  if (exp_index1[i] < 7) {
                      v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
                      if (v >= max) {
                          max   = v;
                          index = i;
                      }
                  }
              }
              if (index == -1)
                  break;
              tmp_categorize_array[tmp_categorize_array1_idx++] = index;
              tmpbias1 -= expbits_tab[exp_index1[index]] -
                          expbits_tab[exp_index1[index] + 1];
              ++exp_index1[index];
          } else {  /* <--- */
              int min = 999999;
              index = -1;
              for (i = 0; i < p->total_subbands; i++) {
                  if (exp_index2[i] > 0) {
                      v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
                      if (v < min) {
                          min   = v;
                          index = i;
                      }
                  }
              }
              if (index == -1)
                  break;
              tmp_categorize_array[--tmp_categorize_array2_idx] = index;
              tmpbias2 -= expbits_tab[exp_index2[index]] -
                          expbits_tab[exp_index2[index] - 1];
              --exp_index2[index];
          }
      }
  
      for (i = 0; i < p->total_subbands; i++)
          category[i] = exp_index2[i];
  
      for (i = 0; i < p->numvector_size - 1; i++)
          category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
  }
  
  
  /**
   * Expand the category vector.
   *
   * @param q                     pointer to the COOKContext
   * @param category              pointer to the category array
   * @param category_index        pointer to the category_index array
   */
  static inline void expand_category(COOKContext *q, int *category,
                                     int *category_index)
  {
      int i;
      for (i = 0; i < q->num_vectors; i++)
      {
          int idx = category_index[i];
          if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
              --category[idx];
      }
  }
  
  /**
   * The real requantization of the mltcoefs
   *
   * @param q                     pointer to the COOKContext
   * @param index                 index
   * @param quant_index           quantisation index
   * @param subband_coef_index    array of indexes to quant_centroid_tab
   * @param subband_coef_sign     signs of coefficients
   * @param mlt_p                 pointer into the mlt buffer
   */
  static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
                                   int *subband_coef_index, int *subband_coef_sign,
                                   float *mlt_p)
  {
      int i;
      float f1;
  
      for (i = 0; i < SUBBAND_SIZE; i++) {
          if (subband_coef_index[i]) {
              f1 = quant_centroid_tab[index][subband_coef_index[i]];
              if (subband_coef_sign[i])
                  f1 = -f1;
          } else {
              /* noise coding if subband_coef_index[i] == 0 */
              f1 = dither_tab[index];
              if (av_lfg_get(&q->random_state) < 0x80000000)
                  f1 = -f1;
          }
          mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
      }
  }
  /**
   * Unpack the subband_coef_index and subband_coef_sign vectors.
   *
   * @param q                     pointer to the COOKContext
   * @param category              pointer to the category array
   * @param subband_coef_index    array of indexes to quant_centroid_tab
   * @param subband_coef_sign     signs of coefficients
   */
  static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
                         int *subband_coef_index, int *subband_coef_sign)
  {
      int i, j;
      int vlc, vd, tmp, result;
  
      vd = vd_tab[category];
      result = 0;
      for (i = 0; i < vpr_tab[category]; i++) {
          vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
          if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
              vlc = 0;
              result = 1;
          }
          for (j = vd - 1; j >= 0; j--) {
              tmp = (vlc * invradix_tab[category]) / 0x100000;
              subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
              vlc = tmp;
          }
          for (j = 0; j < vd; j++) {
              if (subband_coef_index[i * vd + j]) {
                  if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
                      subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
                  } else {
                      result = 1;
                      subband_coef_sign[i * vd + j] = 0;
                  }
              } else {
                  subband_coef_sign[i * vd + j] = 0;
              }
          }
      }
      return result;
  }
  
  
  /**
   * Fill the mlt_buffer with mlt coefficients.
   *
   * @param q                 pointer to the COOKContext
   * @param category          pointer to the category array
   * @param quant_index_table pointer to the array
   * @param mlt_buffer        pointer to mlt coefficients
   */
  static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
                             int *quant_index_table, float *mlt_buffer)
  {
      /* A zero in this table means that the subband coefficient is
         random noise coded. */
      int subband_coef_index[SUBBAND_SIZE];
      /* A zero in this table means that the subband coefficient is a
         positive multiplicator. */
      int subband_coef_sign[SUBBAND_SIZE];
      int band, j;
      int index = 0;
  
      for (band = 0; band < p->total_subbands; band++) {
          index = category[band];
          if (category[band] < 7) {
              if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
                  index = 7;
                  for (j = 0; j < p->total_subbands; j++)
                      category[band + j] = 7;
              }
          }
          if (index >= 7) {
              memset(subband_coef_index, 0, sizeof(subband_coef_index));
              memset(subband_coef_sign,  0, sizeof(subband_coef_sign));
          }
          q->scalar_dequant(q, index, quant_index_table[band],
                            subband_coef_index, subband_coef_sign,
                            &mlt_buffer[band * SUBBAND_SIZE]);
      }
  
      /* FIXME: should this be removed, or moved into loop above? */
      if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
          return;
  }
  
  
  static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
  {
      int category_index[128] = { 0 };
      int category[128]       = { 0 };
      int quant_index_table[102];
      int res, i;
  
      if ((res = decode_envelope(q, p, quant_index_table)) < 0)
          return res;
      q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);
      categorize(q, p, quant_index_table, category, category_index);
      expand_category(q, category, category_index);
      for (i=0; i<p->total_subbands; i++) {
          if (category[i] > 7)
              return AVERROR_INVALIDDATA;
      }
      decode_vectors(q, p, category, quant_index_table, mlt_buffer);
  
      return 0;
  }
  
  
  /**
   * the actual requantization of the timedomain samples
   *
   * @param q                 pointer to the COOKContext
   * @param buffer            pointer to the timedomain buffer
   * @param gain_index        index for the block multiplier
   * @param gain_index_next   index for the next block multiplier
   */
  static void interpolate_float(COOKContext *q, float *buffer,
                                int gain_index, int gain_index_next)
  {
      int i;
      float fc1, fc2;
      fc1 = pow2tab[gain_index + 63];
  
      if (gain_index == gain_index_next) {             // static gain
          for (i = 0; i < q->gain_size_factor; i++)
              buffer[i] *= fc1;
      } else {                                        // smooth gain
          fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
          for (i = 0; i < q->gain_size_factor; i++) {
              buffer[i] *= fc1;
              fc1       *= fc2;
          }
      }
  }
  
  /**
   * Apply transform window, overlap buffers.
   *
   * @param q                 pointer to the COOKContext
   * @param inbuffer          pointer to the mltcoefficients
   * @param gains_ptr         current and previous gains
   * @param previous_buffer   pointer to the previous buffer to be used for overlapping
   */
  static void imlt_window_float(COOKContext *q, float *inbuffer,
                                cook_gains *gains_ptr, float *previous_buffer)
  {
      const float fc = pow2tab[gains_ptr->previous[0] + 63];
      int i;
      /* The weird thing here, is that the two halves of the time domain
       * buffer are swapped. Also, the newest data, that we save away for
       * next frame, has the wrong sign. Hence the subtraction below.
       * Almost sounds like a complex conjugate/reverse data/FFT effect.
       */
  
      /* Apply window and overlap */
      for (i = 0; i < q->samples_per_channel; i++)
          inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
                        previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
  }
  
  /**
   * The modulated lapped transform, this takes transform coefficients
   * and transforms them into timedomain samples.
   * Apply transform window, overlap buffers, apply gain profile
   * and buffer management.
   *
   * @param q                 pointer to the COOKContext
   * @param inbuffer          pointer to the mltcoefficients
   * @param gains_ptr         current and previous gains
   * @param previous_buffer   pointer to the previous buffer to be used for overlapping
   */
  static void imlt_gain(COOKContext *q, float *inbuffer,
                        cook_gains *gains_ptr, float *previous_buffer)
  {
      float *buffer0 = q->mono_mdct_output;
      float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
      int i;
  
      /* Inverse modified discrete cosine transform */
      q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
  
      q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
  
      /* Apply gain profile */
      for (i = 0; i < 8; i++)
          if (gains_ptr->now[i] || gains_ptr->now[i + 1])
              q->interpolate(q, &buffer1[q->gain_size_factor * i],
                             gains_ptr->now[i], gains_ptr->now[i + 1]);
  
      /* Save away the current to be previous block. */
      memcpy(previous_buffer, buffer0,
             q->samples_per_channel * sizeof(*previous_buffer));
  }
  
  
  /**
   * function for getting the jointstereo coupling information
   *
   * @param q                 pointer to the COOKContext
   * @param decouple_tab      decoupling array
   */
  static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
  {
      int i;
      int vlc    = get_bits1(&q->gb);
      int start  = cplband[p->js_subband_start];
      int end    = cplband[p->subbands - 1];
      int length = end - start + 1;
  
      if (start > end)
          return 0;
  
      if (vlc)
          for (i = 0; i < length; i++)
              decouple_tab[start + i] = get_vlc2(&q->gb,
                                                 p->channel_coupling.table,
                                                 p->channel_coupling.bits, 2);
      else
          for (i = 0; i < length; i++) {
              int v = get_bits(&q->gb, p->js_vlc_bits);
              if (v == (1<<p->js_vlc_bits)-1) {
                  av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
                  return AVERROR_INVALIDDATA;
              }
              decouple_tab[start + i] = v;
          }
      return 0;
  }
  
  /**
   * function decouples a pair of signals from a single signal via multiplication.
   *
   * @param q                 pointer to the COOKContext
   * @param subband           index of the current subband
   * @param f1                multiplier for channel 1 extraction
   * @param f2                multiplier for channel 2 extraction
   * @param decode_buffer     input buffer
   * @param mlt_buffer1       pointer to left channel mlt coefficients
   * @param mlt_buffer2       pointer to right channel mlt coefficients
   */
  static void decouple_float(COOKContext *q,
                             COOKSubpacket *p,
                             int subband,
                             float f1, float f2,
                             float *decode_buffer,
                             float *mlt_buffer1, float *mlt_buffer2)
  {
      int j, tmp_idx;
      for (j = 0; j < SUBBAND_SIZE; j++) {
          tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
          mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
          mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
      }
  }
  
  /**
   * function for decoding joint stereo data
   *
   * @param q                 pointer to the COOKContext
   * @param mlt_buffer1       pointer to left channel mlt coefficients
   * @param mlt_buffer2       pointer to right channel mlt coefficients
   */
  static int joint_decode(COOKContext *q, COOKSubpacket *p,
                          float *mlt_buffer_left, float *mlt_buffer_right)
  {
      int i, j, res;
      int decouple_tab[SUBBAND_SIZE] = { 0 };
      float *decode_buffer = q->decode_buffer_0;
      int idx, cpl_tmp;
      float f1, f2;
      const float *cplscale;
  
      memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
  
      /* Make sure the buffers are zeroed out. */
      memset(mlt_buffer_left,  0, 1024 * sizeof(*mlt_buffer_left));
      memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
      if ((res = decouple_info(q, p, decouple_tab)) < 0)
          return res;
      if ((res = mono_decode(q, p, decode_buffer)) < 0)
          return res;
      /* The two channels are stored interleaved in decode_buffer. */
      for (i = 0; i < p->js_subband_start; i++) {
          for (j = 0; j < SUBBAND_SIZE; j++) {
              mlt_buffer_left[i  * 20 + j] = decode_buffer[i * 40 + j];
              mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
          }
      }
  
      /* When we reach js_subband_start (the higher frequencies)
         the coefficients are stored in a coupling scheme. */
      idx = (1 << p->js_vlc_bits) - 1;
      for (i = p->js_subband_start; i < p->subbands; i++) {
          cpl_tmp = cplband[i];
          idx -= decouple_tab[cpl_tmp];
          cplscale = q->cplscales[p->js_vlc_bits - 2];  // choose decoupler table
          f1 = cplscale[decouple_tab[cpl_tmp] + 1];
          f2 = cplscale[idx];
          q->decouple(q, p, i, f1, f2, decode_buffer,
                      mlt_buffer_left, mlt_buffer_right);
          idx = (1 << p->js_vlc_bits) - 1;
      }
  
      return 0;
  }
  
  /**
   * First part of subpacket decoding:
   *  decode raw stream bytes and read gain info.
   *
   * @param q                 pointer to the COOKContext
   * @param inbuffer          pointer to raw stream data
   * @param gains_ptr         array of current/prev gain pointers
   */
  static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
                                           const uint8_t *inbuffer,
                                           cook_gains *gains_ptr)
  {
      int offset;
  
      offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
                            p->bits_per_subpacket / 8);
      init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
                    p->bits_per_subpacket);
      decode_gain_info(&q->gb, gains_ptr->now);
  
      /* Swap current and previous gains */
      FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
  }
  
  /**
   * Saturate the output signal and interleave.
   *
   * @param q                 pointer to the COOKContext
   * @param out               pointer to the output vector
   */
  static void saturate_output_float(COOKContext *q, float *out)
  {
      q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
                           -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
  }
  
  
  /**
   * Final part of subpacket decoding:
   *  Apply modulated lapped transform, gain compensation,
   *  clip and convert to integer.
   *
   * @param q                 pointer to the COOKContext
   * @param decode_buffer     pointer to the mlt coefficients
   * @param gains_ptr         array of current/prev gain pointers
   * @param previous_buffer   pointer to the previous buffer to be used for overlapping
   * @param out               pointer to the output buffer
   */
  static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
                                           cook_gains *gains_ptr, float *previous_buffer,
                                           float *out)
  {
      imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
      if (out)
          q->saturate_output(q, out);
  }
  
  
  /**
   * Cook subpacket decoding. This function returns one decoded subpacket,
   * usually 1024 samples per channel.
   *
   * @param q                 pointer to the COOKContext
   * @param inbuffer          pointer to the inbuffer
   * @param outbuffer         pointer to the outbuffer
   */
  static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
                              const uint8_t *inbuffer, float **outbuffer)
  {
      int sub_packet_size = p->size;
      int res;
  
      memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
      decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
  
      if (p->joint_stereo) {
          if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
              return res;
      } else {
          if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
              return res;
  
          if (p->num_channels == 2) {
              decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
              if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
                  return res;
          }
      }
  
      mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
                            p->mono_previous_buffer1,
                            outbuffer ? outbuffer[p->ch_idx] : NULL);
  
      if (p->num_channels == 2) {
          if (p->joint_stereo)
              mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
                                    p->mono_previous_buffer2,
                                    outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
          else
              mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
                                    p->mono_previous_buffer2,
                                    outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
      }
  
      return 0;
  }
  
  
  static int cook_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;
      COOKContext *q = avctx->priv_data;
      float **samples = NULL;
      int i, ret;
      int offset = 0;
      int chidx = 0;
  
      if (buf_size < avctx->block_align)
          return buf_size;
  
      /* get output buffer */
      if (q->discarded_packets >= 2) {
          frame->nb_samples = q->samples_per_channel;
          if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
              return ret;
          samples = (float **)frame->extended_data;
      }
  
      /* estimate subpacket sizes */
      q->subpacket[0].size = avctx->block_align;
  
      for (i = 1; i < q->num_subpackets; i++) {
          q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
          q->subpacket[0].size -= q->subpacket[i].size + 1;
          if (q->subpacket[0].size < 0) {
              av_log(avctx, AV_LOG_DEBUG,
                     "frame subpacket size total > avctx->block_align!\n");
              return AVERROR_INVALIDDATA;
          }
      }
  
      /* decode supbackets */
      for (i = 0; i < q->num_subpackets; i++) {
          q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
                                                q->subpacket[i].bits_per_subpdiv;
          q->subpacket[i].ch_idx = chidx;
          av_log(avctx, AV_LOG_DEBUG,
                 "subpacket[%i] size %i js %i %i block_align %i\n",
                 i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
                 avctx->block_align);
  
          if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
              return ret;
          offset += q->subpacket[i].size;
          chidx += q->subpacket[i].num_channels;
          av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
                 i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
      }
  
      /* Discard the first two frames: no valid audio. */
      if (q->discarded_packets < 2) {
          q->discarded_packets++;
          *got_frame_ptr = 0;
          return avctx->block_align;
      }
  
      *got_frame_ptr = 1;
  
      return avctx->block_align;
  }
  
  #ifdef DEBUG
  static void dump_cook_context(COOKContext *q)
  {
      //int i=0;
  #define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
      av_dlog(q->avctx, "COOKextradata\n");
      av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
      if (q->subpacket[0].cookversion > STEREO) {
          PRINT("js_subband_start", q->subpacket[0].js_subband_start);
          PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
      }
      av_dlog(q->avctx, "COOKContext\n");
      PRINT("nb_channels", q->avctx->channels);
      PRINT("bit_rate", q->avctx->bit_rate);
      PRINT("sample_rate", q->avctx->sample_rate);
      PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
      PRINT("subbands", q->subpacket[0].subbands);
      PRINT("js_subband_start", q->subpacket[0].js_subband_start);
      PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
      PRINT("numvector_size", q->subpacket[0].numvector_size);
      PRINT("total_subbands", q->subpacket[0].total_subbands);
  }
  #endif
  
  /**
   * Cook initialization
   *
   * @param avctx     pointer to the AVCodecContext
   */
  static av_cold int cook_decode_init(AVCodecContext *avctx)
  {
      COOKContext *q = avctx->priv_data;
      const uint8_t *edata_ptr = avctx->extradata;
      const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
      int extradata_size = avctx->extradata_size;
      int s = 0;
      unsigned int channel_mask = 0;
      int samples_per_frame = 0;
      int ret;
      q->avctx = avctx;
  
      /* Take care of the codec specific extradata. */
      if (extradata_size < 8) {
          av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
          return AVERROR_INVALIDDATA;
      }
      av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
  
      /* Take data from the AVCodecContext (RM container). */
      if (!avctx->channels) {
          av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
          return AVERROR_INVALIDDATA;
      }
  
      /* Initialize RNG. */
      av_lfg_init(&q->random_state, 0);
  
      ff_audiodsp_init(&q->adsp);
  
      while (edata_ptr < edata_ptr_end) {
          /* 8 for mono, 16 for stereo, ? for multichannel
             Swap to right endianness so we don't need to care later on. */
          if (extradata_size >= 8) {
              q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
              samples_per_frame           = bytestream_get_be16(&edata_ptr);
              q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
              extradata_size -= 8;
          }
          if (extradata_size >= 8) {
              bytestream_get_be32(&edata_ptr);    // Unknown unused
              q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
              if (q->subpacket[s].js_subband_start >= 51) {
                  av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
                  return AVERROR_INVALIDDATA;
              }
  
              q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
              extradata_size -= 8;
          }
  
          /* Initialize extradata related variables. */
          q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
          q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
  
          /* Initialize default data states. */
          q->subpacket[s].log2_numvector_size = 5;
          q->subpacket[s].total_subbands = q->subpacket[s].subbands;
          q->subpacket[s].num_channels = 1;
  
          /* Initialize version-dependent variables */
  
          av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
                 q->subpacket[s].cookversion);
          q->subpacket[s].joint_stereo = 0;
          switch (q->subpacket[s].cookversion) {
          case MONO:
              if (avctx->channels != 1) {
                  avpriv_request_sample(avctx, "Container channels != 1");
                  return AVERROR_PATCHWELCOME;
              }
              av_log(avctx, AV_LOG_DEBUG, "MONO\n");
              break;
          case STEREO:
              if (avctx->channels != 1) {
                  q->subpacket[s].bits_per_subpdiv = 1;
                  q->subpacket[s].num_channels = 2;
              }
              av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
              break;
          case JOINT_STEREO:
              if (avctx->channels != 2) {
                  avpriv_request_sample(avctx, "Container channels != 2");
                  return AVERROR_PATCHWELCOME;
              }
              av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
              if (avctx->extradata_size >= 16) {
                  q->subpacket[s].total_subbands = q->subpacket[s].subbands +
                                                   q->subpacket[s].js_subband_start;
                  q->subpacket[s].joint_stereo = 1;
                  q->subpacket[s].num_channels = 2;
              }
              if (q->subpacket[s].samples_per_channel > 256) {
                  q->subpacket[s].log2_numvector_size = 6;
              }
              if (q->subpacket[s].samples_per_channel > 512) {
                  q->subpacket[s].log2_numvector_size = 7;
              }
              break;
          case MC_COOK:
              av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
              if (extradata_size >= 4)
                  channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
  
              if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {
                  q->subpacket[s].total_subbands = q->subpacket[s].subbands +
                                                   q->subpacket[s].js_subband_start;
                  q->subpacket[s].joint_stereo = 1;
                  q->subpacket[s].num_channels = 2;
                  q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
  
                  if (q->subpacket[s].samples_per_channel > 256) {
                      q->subpacket[s].log2_numvector_size = 6;
                  }
                  if (q->subpacket[s].samples_per_channel > 512) {
                      q->subpacket[s].log2_numvector_size = 7;
                  }
              } else
                  q->subpacket[s].samples_per_channel = samples_per_frame;
  
              break;
          default:
              avpriv_request_sample(avctx, "Cook version %d",
                                    q->subpacket[s].cookversion);
              return AVERROR_PATCHWELCOME;
          }
  
          if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
              av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
              return AVERROR_INVALIDDATA;
          } else
              q->samples_per_channel = q->subpacket[0].samples_per_channel;
  
  
          /* Initialize variable relations */
          q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
  
          /* Try to catch some obviously faulty streams, othervise it might be exploitable */
          if (q->subpacket[s].total_subbands > 53) {
              avpriv_request_sample(avctx, "total_subbands > 53");
              return AVERROR_PATCHWELCOME;
          }
  
          if ((q->subpacket[s].js_vlc_bits > 6) ||
              (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
              av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
                     q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
              return AVERROR_INVALIDDATA;
          }
  
          if (q->subpacket[s].subbands > 50) {
              avpriv_request_sample(avctx, "subbands > 50");
              return AVERROR_PATCHWELCOME;
          }
          if (q->subpacket[s].subbands == 0) {
              avpriv_request_sample(avctx, "subbands = 0");
              return AVERROR_PATCHWELCOME;
          }
          q->subpacket[s].gains1.now      = q->subpacket[s].gain_1;
          q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
          q->subpacket[s].gains2.now      = q->subpacket[s].gain_3;
          q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
  
          if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
              av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
              return AVERROR_INVALIDDATA;
          }
  
          q->num_subpackets++;
          s++;
          if (s > FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
              avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
              return AVERROR_PATCHWELCOME;
          }
      }
      /* Generate tables */
      init_pow2table();
      init_gain_table(q);
      init_cplscales_table(q);
  
      if ((ret = init_cook_vlc_tables(q)))
          return ret;
  
  
      if (avctx->block_align >= UINT_MAX / 2)
          return AVERROR(EINVAL);
  
      /* Pad the databuffer with:
         DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
         FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
      q->decoded_bytes_buffer =
          av_mallocz(avctx->block_align
                     + DECODE_BYTES_PAD1(avctx->block_align)
                     + FF_INPUT_BUFFER_PADDING_SIZE);
      if (!q->decoded_bytes_buffer)
          return AVERROR(ENOMEM);
  
      /* Initialize transform. */
      if ((ret = init_cook_mlt(q)))
          return ret;
  
      /* Initialize COOK signal arithmetic handling */
      if (1) {
          q->scalar_dequant  = scalar_dequant_float;
          q->decouple        = decouple_float;
          q->imlt_window     = imlt_window_float;
          q->interpolate     = interpolate_float;
          q->saturate_output = saturate_output_float;
      }
  
      /* Try to catch some obviously faulty streams, othervise it might be exploitable */
      if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
          q->samples_per_channel != 1024) {
          avpriv_request_sample(avctx, "samples_per_channel = %d",
                                q->samples_per_channel);
          return AVERROR_PATCHWELCOME;
      }
  
      avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
      if (channel_mask)
          avctx->channel_layout = channel_mask;
      else
          avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
  
  #ifdef DEBUG
      dump_cook_context(q);
  #endif
      return 0;
  }
  
  AVCodec ff_cook_decoder = {
      .name           = "cook",
      .long_name      = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
      .type           = AVMEDIA_TYPE_AUDIO,
      .id             = AV_CODEC_ID_COOK,
      .priv_data_size = sizeof(COOKContext),
      .init           = cook_decode_init,
      .close          = cook_decode_close,
      .decode         = cook_decode_frame,
      .capabilities   = CODEC_CAP_DR1,
      .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                        AV_SAMPLE_FMT_NONE },
  };
  