/*
  * 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 "avcodec.h"

 #include "get_bits.h"

 #include "dsputil.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 {

     int *now;
     int *previous;
 } cook_gains;

 typedef struct {
     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;

     DSPContext          dsp;

     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(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((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 < p->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 < 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->dsp.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 <= 0) {

         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_dsputil_init(&q->dsp, avctx);
 

     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 == NULL)

         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 },

 };