annotate cook.c @ 6920:d02af7474bff libavcodec

Prevent 128*1<<trellis from becoming 0 and creating 0 sized arrays. fixes CID84 RUN2 CID85 RUN2 CID86 RUN2 CID87 RUN2 CID88 RUN2 CID89 RUN2 CID90 RUN2 CID91 RUN2 CID92 RUN2 CID93 RUN2 CID94 RUN2 CID95 RUN2 CID96 RUN2 CID97 RUN2 CID98 RUN2 CID99 RUN2 CID100 RUN2 CID101 RUN2 CID102 RUN2 CID103 RUN2 CID104 RUN2 CID105 RUN2 CID106 RUN2

/*
 * 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.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;
    DECLARE_ALIGNED_16(FFTSample, mdct_tmp[1024]);  /* temporary storage for imlt */
    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 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 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 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 int init_cook_mlt(COOKContext *q) {
    int j;
    float alpha;
    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. */
    alpha = M_PI / (2.0 * (float)mlt_size);
    for(j=0 ; j<mlt_size ; j++)
        q->mlt_window[j] = sin((j + 0.5) * alpha) * 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 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 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 */
    q->mdct_ctx.fft.imdct_calc(&q->mdct_ctx, q->mono_mdct_output,
                               inbuffer, q->mdct_tmp);

    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 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_init_random(1, &q->random_state);

    /* 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;
    }

#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 = "COOK",
};