/*
  * 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 libavcodec/cook.c

  * 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 <math.h>
 #include <stddef.h>
 #include <stdio.h>
 

 #include "libavutil/random.h"

 #include "avcodec.h"
 #include "bitstream.h"
 #include "dsputil.h"

 #include "bytestream.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 COOKDEBUG
 
 typedef struct {

     int *now;
     int *previous;
 } cook_gains;

 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,
                        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, int chan, int16_t *out);
 

     GetBitContext       gb;
     /* stream data */
     int                 nb_channels;
     int                 joint_stereo;
     int                 bit_rate;
     int                 sample_rate;
     int                 samples_per_channel;
     int                 samples_per_frame;
     int                 subbands;

     int                 log2_numvector_size;
     int                 numvector_size;                //1 << log2_numvector_size;

     int                 js_subband_start;
     int                 total_subbands;
     int                 num_vectors;
     int                 bits_per_subpacket;

     int                 cookversion;

     /* states */

     AVRandomState       random_state;

 
     /* transform data */

     MDCTContext         mdct_ctx;

     float*              mlt_window;
 
     /* gain buffers */

     cook_gains          gains1;
     cook_gains          gains2;
     int                 gain_1[9];
     int                 gain_2[9];
     int                 gain_3[9];
     int                 gain_4[9];

 
     /* VLC data */
     int                 js_vlc_bits;
     VLC                 envelope_quant_index[13];
     VLC                 sqvh[7];          //scalar quantization
     VLC                 ccpl;             //channel coupling
 
     /* generatable tables and related variables */
     int                 gain_size_factor;
     float               gain_table[23];
 
     /* data buffers */
 
     uint8_t*            decoded_bytes_buffer;

     DECLARE_ALIGNED_16(float,mono_mdct_output[2048]);

     float               mono_previous_buffer1[1024];
     float               mono_previous_buffer2[1024];
     float               decode_buffer_1[1024];
     float               decode_buffer_2[1024];

     float               decode_buffer_0[1060]; /* static allocation for joint decode */

     const float         *cplscales[5];

 } COOKContext;
 

 static float     pow2tab[127];
 static float rootpow2tab[127];
 

 /* debug functions */
 
 #ifdef COOKDEBUG
 static void dump_float_table(float* table, int size, int delimiter) {
     int i=0;
     av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
     for (i=0 ; i<size ; i++) {
         av_log(NULL, AV_LOG_ERROR, "%5.1f, ", table[i]);
         if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
     }
 }
 
 static void dump_int_table(int* table, int size, int delimiter) {
     int i=0;
     av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
     for (i=0 ; i<size ; i++) {
         av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
         if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
     }
 }
 
 static void dump_short_table(short* table, int size, int delimiter) {
     int i=0;
     av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i);
     for (i=0 ; i<size ; i++) {
         av_log(NULL, AV_LOG_ERROR, "%d, ", table[i]);
         if ((i+1)%delimiter == 0) av_log(NULL,AV_LOG_ERROR,"\n[%d]: ",i+1);
     }
 }
 
 #endif
 
 /*************** 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(NULL,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);
     }
 
     if (q->nb_channels==2 && q->joint_stereo==1){

         result |= init_vlc (&q->ccpl, 6, (1<<q->js_vlc_bits)-1,

             ccpl_huffbits[q->js_vlc_bits-2], 1, 1,
             ccpl_huffcodes[q->js_vlc_bits-2], 2, 2, 0);
         av_log(NULL,AV_LOG_DEBUG,"Joint-stereo VLC used.\n");
     }
 
     av_log(NULL,AV_LOG_DEBUG,"VLC tables initialized.\n");
     return result;
 }
 

 static av_cold int init_cook_mlt(COOKContext *q) {

     int j;

     int mlt_size = q->samples_per_channel;

     if ((q->mlt_window = av_malloc(sizeof(float)*mlt_size)) == 0)
       return -1;

 
     /* 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 (ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size)+1, 1)) {
       av_free(q->mlt_window);
       return -1;

     }

     av_log(NULL,AV_LOG_DEBUG,"MDCT initialized, order = %d.\n",
            av_log2(mlt_size)+1);

     return 0;

 }
 

 static const float *maybe_reformat_buffer32 (COOKContext *q, const float *ptr, int n)

 {
     if (1)
         return ptr;
 }
 

 static av_cold void init_cplscales_table (COOKContext *q) {

     int i;
     for (i=0;i<5;i++)
         q->cplscales[i] = maybe_reformat_buffer32 (q, cplscales[i], (1<<(i+2))-1);
 }
 

 /*************** init functions end ***********/
 
 /**
  * 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

  */

 #define DECODE_BYTES_PAD1(bytes) (3 - ((bytes)+3) % 4)
 #define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))

 static inline int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){

     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^be2me_64(int64_t)in[i]);
      * Buffer alignment needs to be checked. */
 

     off = (int)((long)inbuffer & 3);

     buf = (const uint32_t*) (inbuffer - off);

     c = be2me_32((0x37c511f2 >> (off*8)) | (0x37c511f2 << (32-(off*8))));
     bytes += 3 + off;
     for (i = 0; i < bytes/4; i++)
         obuf[i] = c ^ buf[i];

     return off;

 }
 
 /**
  * Cook uninit
  */
 

 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_free(q->mlt_window);
     av_free(q->decoded_bytes_buffer);
 
     /* Free the transform. */

     ff_mdct_end(&q->mdct_ctx);

 
     /* Free the VLC tables. */
     for (i=0 ; i<13 ; i++) {
         free_vlc(&q->envelope_quant_index[i]);
     }
     for (i=0 ; i<7 ; i++) {
         free_vlc(&q->sqvh[i]);
     }
     if(q->nb_channels==2 && q->joint_stereo==1 ){
         free_vlc(&q->ccpl);
     }
 
     av_log(NULL,AV_LOG_DEBUG,"Memory deallocated.\n");
 
     return 0;
 }
 
 /**

  * Fill the gain array for the timedomain quantization.

  *
  * @param q                 pointer to the COOKContext

  * @param gaininfo[9]       array of gain indexes

  */
 

 static void decode_gain_info(GetBitContext *gb, int *gaininfo)
 {
     int i, n;

 
     while (get_bits1(gb)) {}

     n = get_bits_count(gb) - 1;     //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 void decode_envelope(COOKContext *q, 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 < q->total_subbands ; i++){
         vlc_index=i;
         if (i >= q->js_subband_start * 2) {
             vlc_index-=q->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
     }
 }
 
 /**
  * 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, 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];
     int exp_index1[102];
 

     int tmp_categorize_array[128*2];
     int tmp_categorize_array1_idx=q->numvector_size;
     int tmp_categorize_array2_idx=q->numvector_size;

 
     bits_left =  q->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;
         //av_log(NULL, AV_LOG_ERROR, "bits_left = %d\n",bits_left);
     }
 
     memset(&exp_index1,0,102*sizeof(int));
     memset(&exp_index2,0,102*sizeof(int));

     memset(&tmp_categorize_array,0,128*2*sizeof(int));

 
     bias=-32;
 
     /* Estimate bias. */
     for (i=32 ; i>0 ; i=i/2){
         num_bits = 0;
         index = 0;
         for (j=q->total_subbands ; j>0 ; j--){

             exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);

             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<q->total_subbands ; i++) {

         exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);

         num_bits += expbits_tab[exp_idx];
         exp_index1[i] = exp_idx;
         exp_index2[i] = exp_idx;
     }

     tmpbias1 = tmpbias2 = num_bits;

 
     for (j = 1 ; j < q->numvector_size ; j++) {

         if (tmpbias1 + tmpbias2 > 2*bits_left) {  /* ---> */

             int max = -999999;
             index=-1;
             for (i=0 ; i<q->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<q->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<q->total_subbands ; i++)
         category[i] = exp_index2[i];
 

     for(i=0 ; i<q->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++){
         ++category[category_index[i]];
     }
 }
 
 /**
  * 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_random(&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, 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 (q->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) < q->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, 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<q->total_subbands ; band++){
         index = category[band];
         if(category[band] < 7){

             if(unpack_SQVH(q, category[band], subband_coef_index, subband_coef_sign)){

                 index=7;
                 for(j=0 ; j<q->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]);

     }
 
     if(q->total_subbands*SUBBAND_SIZE >= q->samples_per_channel){
         return;

     } /* FIXME: should this be removed, or moved into loop above? */

 }
 
 
 /**
  * function for decoding mono data
  *
  * @param q                 pointer to the COOKContext

  * @param mlt_buffer        pointer to mlt coefficients

  */
 
 static void mono_decode(COOKContext *q, float* mlt_buffer) {
 
     int category_index[128];
     int quant_index_table[102];
     int category[128];
 
     memset(&category, 0, 128*sizeof(int));
     memset(&category_index, 0, 128*sizeof(int));
 
     decode_envelope(q, quant_index_table);

     q->num_vectors = get_bits(&q->gb,q->log2_numvector_size);

     categorize(q, quant_index_table, category, category_index);
     expand_category(q, category, category_index);

     decode_vectors(q, category, quant_index_table, mlt_buffer);

 }
 
 
 /**
  * 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;
         }
         return;
     } 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;
         }
         return;
     }
 }
 

 /**
  * 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 *buffer1,
                                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++){
         buffer1[i] = buffer1[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 */

     ff_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, sizeof(float)*q->samples_per_channel);

 }
 
 
 /**
  * function for getting the jointstereo coupling information
  *
  * @param q                 pointer to the COOKContext
  * @param decouple_tab      decoupling array
  *
  */
 
 static void decouple_info(COOKContext *q, int* decouple_tab){
     int length, i;
 
     if(get_bits1(&q->gb)) {
         if(cplband[q->js_subband_start] > cplband[q->subbands-1]) return;
 
         length = cplband[q->subbands-1] - cplband[q->js_subband_start] + 1;
         for (i=0 ; i<length ; i++) {
             decouple_tab[cplband[q->js_subband_start] + i] = get_vlc2(&q->gb, q->ccpl.table, q->ccpl.bits, 2);
         }
         return;
     }
 
     if(cplband[q->js_subband_start] > cplband[q->subbands-1]) return;
 
     length = cplband[q->subbands-1] - cplband[q->js_subband_start] + 1;
     for (i=0 ; i<length ; i++) {
        decouple_tab[cplband[q->js_subband_start] + i] = get_bits(&q->gb, q->js_vlc_bits);
     }
     return;
 }
 

 /*
  * 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,
                             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 = ((q->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 void joint_decode(COOKContext *q, float* mlt_buffer1,
                          float* mlt_buffer2) {
     int i,j;
     int decouple_tab[SUBBAND_SIZE];

     float *decode_buffer = q->decode_buffer_0;

     int idx, cpl_tmp;

     float f1,f2;

     const float* cplscale;

 
     memset(decouple_tab, 0, sizeof(decouple_tab));
     memset(decode_buffer, 0, sizeof(decode_buffer));
 
     /* Make sure the buffers are zeroed out. */
     memset(mlt_buffer1,0, 1024*sizeof(float));
     memset(mlt_buffer2,0, 1024*sizeof(float));
     decouple_info(q, decouple_tab);
     mono_decode(q, decode_buffer);
 
     /* The two channels are stored interleaved in decode_buffer. */
     for (i=0 ; i<q->js_subband_start ; i++) {
         for (j=0 ; j<SUBBAND_SIZE ; j++) {
             mlt_buffer1[i*20+j] = decode_buffer[i*40+j];
             mlt_buffer2[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 << q->js_vlc_bits) - 1;

     for (i=q->js_subband_start ; i<q->subbands ; i++) {
         cpl_tmp = cplband[i];
         idx -=decouple_tab[cpl_tmp];

         cplscale = q->cplscales[q->js_vlc_bits-2];  //choose decoupler table

         f1 = cplscale[decouple_tab[cpl_tmp]];
         f2 = cplscale[idx-1];

         q->decouple (q, i, f1, f2, decode_buffer, mlt_buffer1, mlt_buffer2);

         idx = (1 << q->js_vlc_bits) - 1;

     }
 }
 
 /**

  * 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 gain_ptr          array of current/prev gain pointers
  */
 
 static inline void

 decode_bytes_and_gain(COOKContext *q, const uint8_t *inbuffer,

                       cook_gains *gains_ptr)

 {
     int offset;
 
     offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
                           q->bits_per_subpacket/8);
     init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
                   q->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 to signed 16bit integers.
  *
  * @param q                 pointer to the COOKContext
  * @param chan              channel to saturate
  * @param out               pointer to the output vector
  */
 static void
 saturate_output_float (COOKContext *q, int chan, int16_t *out)
 {
     int j;

     float *output = q->mono_mdct_output + q->samples_per_channel;

     /* Clip and convert floats to 16 bits.
      */
     for (j = 0; j < q->samples_per_channel; j++) {
         out[chan + q->nb_channels * j] =

           av_clip_int16(lrintf(output[j]));

     }
 }
 

 /**
  * 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 gain_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
  * @param chan              0: left or single channel, 1: right channel
  */
 
 static inline void
 mlt_compensate_output(COOKContext *q, float *decode_buffer,

                       cook_gains *gains, float *previous_buffer,

                       int16_t *out, int chan)
 {

     imlt_gain(q, decode_buffer, gains, previous_buffer);

     q->saturate_output (q, chan, 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 sub_packet_size   subpacket size
  * @param outbuffer         pointer to the outbuffer
  */
 
 

 static int decode_subpacket(COOKContext *q, const uint8_t *inbuffer,

                             int sub_packet_size, int16_t *outbuffer) {
     /* packet dump */
 //    for (i=0 ; i<sub_packet_size ; i++) {
 //        av_log(NULL, AV_LOG_ERROR, "%02x", inbuffer[i]);
 //    }
 //    av_log(NULL, AV_LOG_ERROR, "\n");
 

     decode_bytes_and_gain(q, inbuffer, &q->gains1);

     if (q->joint_stereo) {
         joint_decode(q, q->decode_buffer_1, q->decode_buffer_2);
     } else {
         mono_decode(q, q->decode_buffer_1);

         if (q->nb_channels == 2) {

             decode_bytes_and_gain(q, inbuffer + sub_packet_size/2, &q->gains2);

             mono_decode(q, q->decode_buffer_2);
         }
     }

     mlt_compensate_output(q, q->decode_buffer_1, &q->gains1,

                           q->mono_previous_buffer1, outbuffer, 0);

     if (q->nb_channels == 2) {
         if (q->joint_stereo) {

             mlt_compensate_output(q, q->decode_buffer_2, &q->gains1,

                                   q->mono_previous_buffer2, outbuffer, 1);
         } else {

             mlt_compensate_output(q, q->decode_buffer_2, &q->gains2,

                                   q->mono_previous_buffer2, outbuffer, 1);

         }
     }

     return q->samples_per_frame * sizeof(int16_t);

 }
 
 
 /**
  * Cook frame decoding
  *
  * @param avctx     pointer to the AVCodecContext
  */
 
 static int cook_decode_frame(AVCodecContext *avctx,
             void *data, int *data_size,

             const uint8_t *buf, int buf_size) {

     COOKContext *q = avctx->priv_data;
 
     if (buf_size < avctx->block_align)
         return buf_size;
 
     *data_size = decode_subpacket(q, buf, avctx->block_align, data);
 

     /* Discard the first two frames: no valid audio. */
     if (avctx->frame_number < 2) *data_size = 0;
 

     return avctx->block_align;
 }

 #ifdef COOKDEBUG

 static void dump_cook_context(COOKContext *q)

 {
     //int i=0;
 #define PRINT(a,b) av_log(NULL,AV_LOG_ERROR," %s = %d\n", a, b);
     av_log(NULL,AV_LOG_ERROR,"COOKextradata\n");

     av_log(NULL,AV_LOG_ERROR,"cookversion=%x\n",q->cookversion);
     if (q->cookversion > STEREO) {
         PRINT("js_subband_start",q->js_subband_start);
         PRINT("js_vlc_bits",q->js_vlc_bits);

     }
     av_log(NULL,AV_LOG_ERROR,"COOKContext\n");
     PRINT("nb_channels",q->nb_channels);
     PRINT("bit_rate",q->bit_rate);
     PRINT("sample_rate",q->sample_rate);
     PRINT("samples_per_channel",q->samples_per_channel);
     PRINT("samples_per_frame",q->samples_per_frame);
     PRINT("subbands",q->subbands);
     PRINT("random_state",q->random_state);
     PRINT("js_subband_start",q->js_subband_start);

     PRINT("log2_numvector_size",q->log2_numvector_size);

     PRINT("numvector_size",q->numvector_size);
     PRINT("total_subbands",q->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;

 
     /* Take care of the codec specific extradata. */
     if (avctx->extradata_size <= 0) {

         av_log(avctx,AV_LOG_ERROR,"Necessary extradata missing!\n");

         return -1;
     } else {
         /* 8 for mono, 16 for stereo, ? for multichannel
            Swap to right endianness so we don't need to care later on. */

         av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);

         if (avctx->extradata_size >= 8){

             q->cookversion = bytestream_get_be32(&edata_ptr);
             q->samples_per_frame =  bytestream_get_be16(&edata_ptr);
             q->subbands = bytestream_get_be16(&edata_ptr);

         }
         if (avctx->extradata_size >= 16){

             bytestream_get_be32(&edata_ptr);    //Unknown unused
             q->js_subband_start = bytestream_get_be16(&edata_ptr);
             q->js_vlc_bits = bytestream_get_be16(&edata_ptr);

         }
     }
 
     /* Take data from the AVCodecContext (RM container). */
     q->sample_rate = avctx->sample_rate;
     q->nb_channels = avctx->channels;
     q->bit_rate = avctx->bit_rate;
 

     /* Initialize RNG. */

     av_random_init(&q->random_state, 1);

 
     /* Initialize extradata related variables. */

     q->samples_per_channel = q->samples_per_frame / q->nb_channels;

     q->bits_per_subpacket = avctx->block_align * 8;
 
     /* Initialize default data states. */

     q->log2_numvector_size = 5;

     q->total_subbands = q->subbands;
 
     /* Initialize version-dependent variables */

     av_log(NULL,AV_LOG_DEBUG,"q->cookversion=%x\n",q->cookversion);

     q->joint_stereo = 0;

     switch (q->cookversion) {

         case MONO:

             if (q->nb_channels != 1) {

                 av_log(avctx,AV_LOG_ERROR,"Container channels != 1, report sample!\n");

                 return -1;
             }

             av_log(avctx,AV_LOG_DEBUG,"MONO\n");

             break;

         case STEREO:

             if (q->nb_channels != 1) {

                 q->bits_per_subpacket = q->bits_per_subpacket/2;

             }

             av_log(avctx,AV_LOG_DEBUG,"STEREO\n");

             break;
         case JOINT_STEREO:
             if (q->nb_channels != 2) {

                 av_log(avctx,AV_LOG_ERROR,"Container channels != 2, report sample!\n");

                 return -1;
             }

             av_log(avctx,AV_LOG_DEBUG,"JOINT_STEREO\n");

             if (avctx->extradata_size >= 16){

                 q->total_subbands = q->subbands + q->js_subband_start;

                 q->joint_stereo = 1;
             }
             if (q->samples_per_channel > 256) {

                 q->log2_numvector_size  = 6;

             }
             if (q->samples_per_channel > 512) {

                 q->log2_numvector_size  = 7;

             }
             break;
         case MC_COOK:

             av_log(avctx,AV_LOG_ERROR,"MC_COOK not supported!\n");

             return -1;
             break;
         default:

             av_log(avctx,AV_LOG_ERROR,"Unknown Cook version, report sample!\n");

             return -1;
             break;
     }
 
     /* Initialize variable relations */

     q->numvector_size = (1 << q->log2_numvector_size);

 
     /* Generate tables */

     init_pow2table();

     init_gain_table(q);

     init_cplscales_table(q);

 
     if (init_cook_vlc_tables(q) != 0)
         return -1;
 

 
     if(avctx->block_align >= UINT_MAX/2)
         return -1;
 

     /* Pad the databuffer with:
        DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
        FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
     if (q->nb_channels==2 && q->joint_stereo==0) {
         q->decoded_bytes_buffer =
           av_mallocz(avctx->block_align/2
                      + DECODE_BYTES_PAD2(avctx->block_align/2)
                      + FF_INPUT_BUFFER_PADDING_SIZE);
     } else {
         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 == NULL)

         return -1;
 

     q->gains1.now      = q->gain_1;
     q->gains1.previous = q->gain_2;
     q->gains2.now      = q->gain_3;
     q->gains2.previous = q->gain_4;

 
     /* Initialize transform. */

     if ( init_cook_mlt(q) != 0 )

         return -1;

     /* 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->total_subbands > 53) {

         av_log(avctx,AV_LOG_ERROR,"total_subbands > 53, report sample!\n");

         return -1;
     }
     if (q->subbands > 50) {

         av_log(avctx,AV_LOG_ERROR,"subbands > 50, report sample!\n");

         return -1;
     }

     if ((q->samples_per_channel == 256) || (q->samples_per_channel == 512) || (q->samples_per_channel == 1024)) {
     } else {

         av_log(avctx,AV_LOG_ERROR,"unknown amount of samples_per_channel = %d, report sample!\n",q->samples_per_channel);

         return -1;
     }

     if ((q->js_vlc_bits > 6) || (q->js_vlc_bits < 0)) {
         av_log(avctx,AV_LOG_ERROR,"q->js_vlc_bits = %d, only >= 0 and <= 6 allowed!\n",q->js_vlc_bits);
         return -1;
     }

     avctx->sample_fmt = SAMPLE_FMT_S16;

     avctx->channel_layout = (avctx->channels==2) ? CH_LAYOUT_STEREO : CH_LAYOUT_MONO;

 #ifdef COOKDEBUG

     dump_cook_context(q);

 #endif

     return 0;
 }
 
 
 AVCodec cook_decoder =
 {
     .name = "cook",
     .type = CODEC_TYPE_AUDIO,
     .id = CODEC_ID_COOK,
     .priv_data_size = sizeof(COOKContext),
     .init = cook_decode_init,
     .close = cook_decode_close,
     .decode = cook_decode_frame,

     .long_name = NULL_IF_CONFIG_SMALL("COOK"),

 };