annotate cook.c @ 12501:b3f9612d4ea7 libavcodec

Remove pointless semicolon

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

#include "libavutil/lfg.h"
#include "libavutil/random_seed.h"
#include "avcodec.h"
#include "get_bits.h"
#include "dsputil.h"
#include "bytestream.h"
#include "fft.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
//#define COOKDEBUG

typedef struct {
    int *now;
    int *previous;
} cook_gains;

typedef struct {
    int                 ch_idx;
    int                 size;
    int                 num_channels;
    int                 cookversion;
    int                 samples_per_frame;
    int                 subbands;
    int                 js_subband_start;
    int                 js_vlc_bits;
    int                 samples_per_channel;
    int                 log2_numvector_size;
    unsigned int        channel_mask;
    VLC                 ccpl;                 ///< 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];
    /** gain buffers */
    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, int chan, int16_t *out);

    AVCodecContext*     avctx;
    GetBitContext       gb;
    /* stream data */
    int                 nb_channels;
    int                 bit_rate;
    int                 sample_rate;
    int                 num_vectors;
    int                 samples_per_channel;
    /* states */
    AVLFG               random_state;

    /* 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(16, 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];

/* 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(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].ccpl, 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;
    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, 1.0)) {
      av_free(q->mlt_window);
      return -1;
    }
    av_log(q->avctx,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 ***********/

#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){
    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 = av_be2ne32((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]);
    }
    for (i=0 ; i<q->num_subpackets ; i++) {
        free_vlc(&q->subpacket[i].ccpl);
    }

    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[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, 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
    }
}

/**
 * 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, 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=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;
        //av_log(q->avctx, 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=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++){
        ++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_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]);
    }

    if(p->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, COOKSubpacket *p, 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, p, quant_index_table);
    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);
    decode_vectors(q, p, 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 *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 */
    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, COOKSubpacket *p, int* decouple_tab){
    int length, i;

    if(get_bits1(&q->gb)) {
        if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;

        length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
        for (i=0 ; i<length ; i++) {
            decouple_tab[cplband[p->js_subband_start] + i] = get_vlc2(&q->gb, p->ccpl.table, p->ccpl.bits, 2);
        }
        return;
    }

    if(cplband[p->js_subband_start] > cplband[p->subbands-1]) return;

    length = cplband[p->subbands-1] - cplband[p->js_subband_start] + 1;
    for (i=0 ; i<length ; i++) {
       decouple_tab[cplband[p->js_subband_start] + i] = get_bits(&q->gb, p->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,
                            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 void joint_decode(COOKContext *q, COOKSubpacket *p, 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, p, decouple_tab);
    mono_decode(q, p, decode_buffer);

    /* 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_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 << 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]];
        f2 = cplscale[idx-1];
        q->decouple (q, p, i, f1, f2, decode_buffer, mlt_buffer1, mlt_buffer2);
        idx = (1 << p->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 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 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 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
 * @param chan              0: left or single channel, 1: right channel
 */

static inline void
mlt_compensate_output(COOKContext *q, float *decode_buffer,
                      cook_gains *gains_ptr, float *previous_buffer,
                      int16_t *out, int chan)
{
    imlt_gain(q, decode_buffer, gains_ptr, 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 outbuffer         pointer to the outbuffer
 */
static void decode_subpacket(COOKContext *q, COOKSubpacket* p, const uint8_t *inbuffer, int16_t *outbuffer) {
    int sub_packet_size = p->size;
    /* packet dump */
//    for (i=0 ; i<sub_packet_size ; i++) {
//        av_log(q->avctx, AV_LOG_ERROR, "%02x", inbuffer[i]);
//    }
//    av_log(q->avctx, AV_LOG_ERROR, "\n");
    memset(q->decode_buffer_1,0,sizeof(q->decode_buffer_1));
    decode_bytes_and_gain(q, p, inbuffer, &p->gains1);

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

        if (p->num_channels == 2) {
            decode_bytes_and_gain(q, p, inbuffer + sub_packet_size/2, &p->gains2);
            mono_decode(q, p, q->decode_buffer_2);
        }
    }

    mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
                          p->mono_previous_buffer1, outbuffer, p->ch_idx);

    if (p->num_channels == 2) {
        if (p->joint_stereo) {
            mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
                                  p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
         } else {
            mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
                                  p->mono_previous_buffer2, outbuffer, p->ch_idx + 1);
         }
     }

}


/**
 * Cook frame decoding
 *
 * @param avctx     pointer to the AVCodecContext
 */

static int cook_decode_frame(AVCodecContext *avctx,
            void *data, int *data_size,
            AVPacket *avpkt) {
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
    COOKContext *q = avctx->priv_data;
    int i;
    int offset = 0;
    int chidx = 0;

    if (buf_size < avctx->block_align)
        return buf_size;

    /* 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 -1;
        }
    }

    /* decode supbackets */
    *data_size = 0;
    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);
        decode_subpacket(q, &q->subpacket[i], buf + offset, (int16_t*)data);
        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));
    }
    *data_size = sizeof(int16_t) * q->nb_channels * q->samples_per_channel;

    /* 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(q->avctx,AV_LOG_ERROR," %s = %d\n", a, b);
    av_log(q->avctx,AV_LOG_ERROR,"COOKextradata\n");
    av_log(q->avctx,AV_LOG_ERROR,"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_log(q->avctx,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->subpacket[0].samples_per_channel);
    PRINT("samples_per_frame",q->subpacket[0].samples_per_frame);
    PRINT("subbands",q->subpacket[0].subbands);
    PRINT("random_state",q->random_state);
    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

static av_cold int cook_count_channels(unsigned int mask){
    int i;
    int channels = 0;
    for(i = 0;i<32;i++){
        if(mask & (1<<i))
            ++channels;
    }
    return channels;
}

/**
 * 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;
    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 -1;
    }
    av_log(avctx,AV_LOG_DEBUG,"codecdata_length=%d\n",avctx->extradata_size);

    /* 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_lfg_init(&q->random_state, 0);

    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);
            q->subpacket[s].samples_per_frame =  bytestream_get_be16(&edata_ptr);
            q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
            extradata_size -= 8;
        }
        if (avctx->extradata_size >= 8){
            bytestream_get_be32(&edata_ptr);    //Unknown unused
            q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
            q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
            extradata_size -= 8;
        }

        /* Initialize extradata related variables. */
        q->subpacket[s].samples_per_channel = q->subpacket[s].samples_per_frame / q->nb_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 (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->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 (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->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(cook_count_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 = q->subpacket[s].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 = q->subpacket[s].samples_per_frame;

                break;
            default:
                av_log(avctx,AV_LOG_ERROR,"Unknown Cook version, report sample!\n");
                return -1;
                break;
        }

        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 -1;
        } 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) {
            av_log(avctx,AV_LOG_ERROR,"total_subbands > 53, report sample!\n");
            return -1;
        }

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

        if (q->subpacket[s].subbands > 50) {
            av_log(avctx,AV_LOG_ERROR,"subbands > 50, report sample!\n");
            return -1;
        }
        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;

        q->num_subpackets++;
        s++;
        if (s > MAX_SUBPACKETS) {
            av_log(avctx,AV_LOG_ERROR,"Too many subpackets > 5, report file!\n");
            return -1;
        }
    }
    /* 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. */
        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;

    /* 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->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;
    }

    avctx->sample_fmt = SAMPLE_FMT_S16;
    if (channel_mask)
        avctx->channel_layout = channel_mask;
    else
        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 = AVMEDIA_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"),
};