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/*

 * COOK compatible decoder

 * Copyright (c) 2003 Sascha Sommer

 * Copyright (c) 2005 Benjamin Larsson

 *

 * This file is part of FFmpeg.

 *

 * FFmpeg is free software; you can redistribute it and/or

 * modify it under the terms of the GNU Lesser General Public

 * License as published by the Free Software Foundation; either

 * version 2.1 of the License, or (at your option) any later version.

 *

 * FFmpeg is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU

 * Lesser General Public License for more details.

 *

 * You should have received a copy of the GNU Lesser General Public

 * License along with FFmpeg; if not, write to the Free Software

 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

 */

/**

 * @file

 * Cook compatible decoder. Bastardization of the G.722.1 standard.

 * This decoder handles RealNetworks, RealAudio G2 data.

 * Cook is identified by the codec name cook in RM files.

 *

 * To use this decoder, a calling application must supply the extradata

 * bytes provided from the RM container; 8+ bytes for mono streams and

 * 16+ for stereo streams (maybe more).

 *

 * Codec technicalities (all this assume a buffer length of 1024):

 * Cook works with several different techniques to achieve its compression.

 * In the timedomain the buffer is divided into 8 pieces and quantized. If

 * two neighboring pieces have different quantization index a smooth

 * quantization curve is used to get a smooth overlap between the different

 * pieces.

 * To get to the transformdomain Cook uses a modulated lapped transform.

 * The transform domain has 50 subbands with 20 elements each. This

 * means only a maximum of 50*20=1000 coefficients are used out of the 1024

 * available.

 */

#include "libavutil/channel_layout.h"

#include "libavutil/lfg.h"

#include "audiodsp.h"

#include "avcodec.h"

#include "get_bits.h"

#include "bytestream.h"

#include "fft.h"

#include "internal.h"

#include "sinewin.h"

#include "unary.h"

#include "cookdata.h"

/* the different Cook versions */

#define MONO            0x1000001

#define STEREO          0x1000002

#define JOINT_STEREO    0x1000003

#define MC_COOK         0x2000000   // multichannel Cook, not supported

#define SUBBAND_SIZE    20

#define MAX_SUBPACKETS   5

typedef struct cook_gains {

    int *now;

    int *previous;

} cook_gains;

typedef struct COOKSubpacket {

    int                 ch_idx;

    int                 size;

    int                 num_channels;

    int                 cookversion;

    int                 subbands;

    int                 js_subband_start;

    int                 js_vlc_bits;

    int                 samples_per_channel;

    int                 log2_numvector_size;

    unsigned int        channel_mask;

    VLC                 channel_coupling;

    int                 joint_stereo;

    int                 bits_per_subpacket;

    int                 bits_per_subpdiv;

    int                 total_subbands;

    int                 numvector_size;       // 1 << log2_numvector_size;

    float               mono_previous_buffer1[1024];

    float               mono_previous_buffer2[1024];

    cook_gains          gains1;

    cook_gains          gains2;

    int                 gain_1[9];

    int                 gain_2[9];

    int                 gain_3[9];

    int                 gain_4[9];

} COOKSubpacket;

typedef struct cook {

    /*

     * The following 5 functions provide the lowlevel arithmetic on

     * the internal audio buffers.

     */

    void (*scalar_dequant)(struct cook *q, int index, int quant_index,

                           int *subband_coef_index, int *subband_coef_sign,

                           float *mlt_p);

    void (*decouple)(struct cook *q,

                     COOKSubpacket *p,

                     int subband,

                     float f1, float f2,

                     float *decode_buffer,

                     float *mlt_buffer1, float *mlt_buffer2);

    void (*imlt_window)(struct cook *q, float *buffer1,

                        cook_gains *gains_ptr, float *previous_buffer);

    void (*interpolate)(struct cook *q, float *buffer,

                        int gain_index, int gain_index_next);

    void (*saturate_output)(struct cook *q, float *out);

    AVCodecContext*     avctx;

    AudioDSPContext     adsp;

    GetBitContext       gb;

    /* stream data */

    int                 num_vectors;

    int                 samples_per_channel;

    /* states */

    AVLFG               random_state;

    int                 discarded_packets;

    /* transform data */

    FFTContext          mdct_ctx;

    float*              mlt_window;

    /* VLC data */

    VLC                 envelope_quant_index[13];

    VLC                 sqvh[7];          // scalar quantization

    /* generatable tables and related variables */

    int                 gain_size_factor;

    float               gain_table[23];

    /* data buffers */

    uint8_t*            decoded_bytes_buffer;

    DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];

    float               decode_buffer_1[1024];

    float               decode_buffer_2[1024];

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

    const float         *cplscales[5];

    int                 num_subpackets;

    COOKSubpacket       subpacket[MAX_SUBPACKETS];

} COOKContext;

static float     pow2tab[127];

static float rootpow2tab[127];

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

/* table generator */

static av_cold void init_pow2table(void)

{

    int i;

    for (i = -63; i < 64; i++) {

        pow2tab[63 + i] = pow(2, i);

        rootpow2tab[63 + i] = sqrt(pow(2, i));

    }

}

/* table generator */

static av_cold void init_gain_table(COOKContext *q)

{

    int i;

    q->gain_size_factor = q->samples_per_channel / 8;

    for (i = 0; i < 23; i++)

        q->gain_table[i] = pow(pow2tab[i + 52],

                               (1.0 / (double) q->gain_size_factor));

}

static av_cold int init_cook_vlc_tables(COOKContext *q)

{

    int i, result;

    result = 0;

    for (i = 0; i < 13; i++) {

        result |= init_vlc(&q->envelope_quant_index[i], 9, 24,

                           envelope_quant_index_huffbits[i], 1, 1,

                           envelope_quant_index_huffcodes[i], 2, 2, 0);

    }

    av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");

    for (i = 0; i < 7; i++) {

        result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],

                           cvh_huffbits[i], 1, 1,

                           cvh_huffcodes[i], 2, 2, 0);

    }

    for (i = 0; i < q->num_subpackets; i++) {

        if (q->subpacket[i].joint_stereo == 1) {

            result |= init_vlc(&q->subpacket[i].channel_coupling, 6,

                               (1 << q->subpacket[i].js_vlc_bits) - 1,

                               ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,

                               ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);

            av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);

        }

    }

    av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");

    return result;

}

static av_cold int init_cook_mlt(COOKContext *q)

{

    int j, ret;

    int mlt_size = q->samples_per_channel;

    if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)

        return AVERROR(ENOMEM);

    /* Initialize the MLT window: simple sine window. */

    ff_sine_window_init(q->mlt_window, mlt_size);

    for (j = 0; j < mlt_size; j++)

        q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);

    /* Initialize the MDCT. */

    if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {

        av_freep(&q->mlt_window);

        return ret;

    }

    av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",

           av_log2(mlt_size) + 1);

    return 0;

}

static av_cold void init_cplscales_table(COOKContext *q)

{

    int i;

    for (i = 0; i < 5; i++)

        q->cplscales[i] = cplscales[i];

}

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

#define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)

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

/**

 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.

 * Why? No idea, some checksum/error detection method maybe.

 *

 * Out buffer size: extra bytes are needed to cope with

 * padding/misalignment.

 * Subpackets passed to the decoder can contain two, consecutive

 * half-subpackets, of identical but arbitrary size.

 *          1234 1234 1234 1234  extraA extraB

 * Case 1:  AAAA BBBB              0      0

 * Case 2:  AAAA ABBB BB--         3      3

 * Case 3:  AAAA AABB BBBB         2      2

 * Case 4:  AAAA AAAB BBBB BB--    1      5

 *

 * Nice way to waste CPU cycles.

 *

 * @param inbuffer  pointer to byte array of indata

 * @param out       pointer to byte array of outdata

 * @param bytes     number of bytes

 */

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

{

    static const uint32_t tab[4] = {

        AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),

        AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),

    };

    int i, off;

    uint32_t c;

    const uint32_t *buf;

    uint32_t *obuf = (uint32_t *) out;

    /* FIXME: 64 bit platforms would be able to do 64 bits at a time.

     * I'm too lazy though, should be something like

     * for (i = 0; i < bitamount / 64; i++)

     *     (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);

     * Buffer alignment needs to be checked. */

    off = (intptr_t) inbuffer & 3;

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

    c = tab[off];

    bytes += 3 + off;

    for (i = 0; i < bytes / 4; i++)

        obuf[i] = c ^ buf[i];

    return off;

}

static av_cold int cook_decode_close(AVCodecContext *avctx)

{

    int i;

    COOKContext *q = avctx->priv_data;

    av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");

    /* Free allocated memory buffers. */

    av_freep(&q->mlt_window);

    av_freep(&q->decoded_bytes_buffer);

    /* Free the transform. */

    ff_mdct_end(&q->mdct_ctx);

    /* Free the VLC tables. */

    for (i = 0; i < 13; i++)

        ff_free_vlc(&q->envelope_quant_index[i]);

    for (i = 0; i < 7; i++)

        ff_free_vlc(&q->sqvh[i]);

    for (i = 0; i < q->num_subpackets; i++)

        ff_free_vlc(&q->subpacket[i].channel_coupling);

    av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");

    return 0;

}

/**

 * Fill the gain array for the timedomain quantization.

 *

 * @param gb          pointer to the GetBitContext

 * @param gaininfo    array[9] of gain indexes

 */

static void decode_gain_info(GetBitContext *gb, int *gaininfo)

{

    int i, n;

    n = get_unary(gb, 0, get_bits_left(gb));     // amount of elements*2 to update

    i = 0;

    while (n--) {

        int index = get_bits(gb, 3);

        int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;

        while (i <= index)

            gaininfo[i++] = gain;

    }

    while (i <= 8)

        gaininfo[i++] = 0;

}

/**

 * Create the quant index table needed for the envelope.

 *

 * @param q                 pointer to the COOKContext

 * @param quant_index_table pointer to the array

 */

static int decode_envelope(COOKContext *q, COOKSubpacket *p,

                           int *quant_index_table)

{

    int i, j, vlc_index;

    quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize

    for (i = 1; i < p->total_subbands; i++) {

        vlc_index = i;

        if (i >= p->js_subband_start * 2) {

            vlc_index -= p->js_subband_start;

        } else {

            vlc_index /= 2;

            if (vlc_index < 1)

                vlc_index = 1;

        }

        if (vlc_index > 13)

            vlc_index = 13; // the VLC tables >13 are identical to No. 13

        j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,

                     q->envelope_quant_index[vlc_index - 1].bits, 2);

        quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding

        if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {

            av_log(q->avctx, AV_LOG_ERROR,

                   "Invalid quantizer %d at position %d, outside [-63, 63] range\n",

                   quant_index_table[i], i);

            return AVERROR_INVALIDDATA;

        }

    }

    return 0;

}

/**

 * Calculate the category and category_index vector.

 *

 * @param q                     pointer to the COOKContext

 * @param quant_index_table     pointer to the array

 * @param category              pointer to the category array

 * @param category_index        pointer to the category_index array

 */

static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,

                       int *category, int *category_index)

{

    int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;

    int exp_index2[102] = { 0 };

    int exp_index1[102] = { 0 };

    int tmp_categorize_array[128 * 2] = { 0 };

    int tmp_categorize_array1_idx = p->numvector_size;

    int tmp_categorize_array2_idx = p->numvector_size;

    bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);

    if (bits_left > q->samples_per_channel)

        bits_left = q->samples_per_channel +

                    ((bits_left - q->samples_per_channel) * 5) / 8;

    bias = -32;

    /* Estimate bias. */

    for (i = 32; i > 0; i = i / 2) {

        num_bits = 0;

        index    = 0;

        for (j = p->total_subbands; j > 0; j--) {

            exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);

            index++;

            num_bits += expbits_tab[exp_idx];

        }

        if (num_bits >= bits_left - 32)

            bias += i;

    }

    /* Calculate total number of bits. */

    num_bits = 0;

    for (i = 0; i < p->total_subbands; i++) {

        exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);

        num_bits += expbits_tab[exp_idx];

        exp_index1[i] = exp_idx;

        exp_index2[i] = exp_idx;

    }

    tmpbias1 = tmpbias2 = num_bits;

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

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

            int max = -999999;

            index = -1;

            for (i = 0; i < p->total_subbands; i++) {

                if (exp_index1[i] < 7) {

                    v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;

                    if (v >= max) {

                        max   = v;

                        index = i;

                    }

                }

            }

            if (index == -1)

                break;

            tmp_categorize_array[tmp_categorize_array1_idx++] = index;

            tmpbias1 -= expbits_tab[exp_index1[index]] -

                        expbits_tab[exp_index1[index] + 1];

            ++exp_index1[index];

        } else {  /* <--- */

            int min = 999999;

            index = -1;

            for (i = 0; i < p->total_subbands; i++) {

                if (exp_index2[i] > 0) {

                    v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;

                    if (v < min) {

                        min   = v;

                        index = i;

                    }

                }

            }

            if (index == -1)

                break;

            tmp_categorize_array[--tmp_categorize_array2_idx] = index;

            tmpbias2 -= expbits_tab[exp_index2[index]] -

                        expbits_tab[exp_index2[index] - 1];

            --exp_index2[index];

        }

    }

    for (i = 0; i < p->total_subbands; i++)

        category[i] = exp_index2[i];

    for (i = 0; i < p->numvector_size - 1; i++)

        category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];

}

/**

 * Expand the category vector.

 *

 * @param q                     pointer to the COOKContext

 * @param category              pointer to the category array

 * @param category_index        pointer to the category_index array

 */

static inline void expand_category(COOKContext *q, int *category,

                                   int *category_index)

{

    int i;

    for (i = 0; i < q->num_vectors; i++)

    {

        int idx = category_index[i];

        if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))

            --category[idx];

    }

}

/**

 * The real requantization of the mltcoefs

 *

 * @param q                     pointer to the COOKContext

 * @param index                 index

 * @param quant_index           quantisation index

 * @param subband_coef_index    array of indexes to quant_centroid_tab

 * @param subband_coef_sign     signs of coefficients

 * @param mlt_p                 pointer into the mlt buffer

 */

static void scalar_dequant_float(COOKContext *q, int index, int quant_index,

                                 int *subband_coef_index, int *subband_coef_sign,

                                 float *mlt_p)

{

    int i;

    float f1;

    for (i = 0; i < SUBBAND_SIZE; i++) {

        if (subband_coef_index[i]) {

            f1 = quant_centroid_tab[index][subband_coef_index[i]];

            if (subband_coef_sign[i])

                f1 = -f1;

        } else {

            /* noise coding if subband_coef_index[i] == 0 */

            f1 = dither_tab[index];

            if (av_lfg_get(&q->random_state) < 0x80000000)

                f1 = -f1;

        }

        mlt_p[i] = f1 * rootpow2tab[quant_index + 63];

    }

}

/**

 * Unpack the subband_coef_index and subband_coef_sign vectors.

 *

 * @param q                     pointer to the COOKContext

 * @param category              pointer to the category array

 * @param subband_coef_index    array of indexes to quant_centroid_tab

 * @param subband_coef_sign     signs of coefficients

 */

static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,

                       int *subband_coef_index, int *subband_coef_sign)

{

    int i, j;

    int vlc, vd, tmp, result;

    vd = vd_tab[category];

    result = 0;

    for (i = 0; i < vpr_tab[category]; i++) {

        vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);

        if (p->bits_per_subpacket < get_bits_count(&q->gb)) {

            vlc = 0;

            result = 1;

        }

        for (j = vd - 1; j >= 0; j--) {

            tmp = (vlc * invradix_tab[category]) / 0x100000;

            subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);

            vlc = tmp;

        }

        for (j = 0; j < vd; j++) {

            if (subband_coef_index[i * vd + j]) {

                if (get_bits_count(&q->gb) < p->bits_per_subpacket) {

                    subband_coef_sign[i * vd + j] = get_bits1(&q->gb);

                } else {

                    result = 1;

                    subband_coef_sign[i * vd + j] = 0;

                }

            } else {

                subband_coef_sign[i * vd + j] = 0;

            }

        }

    }

    return result;

}

/**

 * Fill the mlt_buffer with mlt coefficients.

 *

 * @param q                 pointer to the COOKContext

 * @param category          pointer to the category array

 * @param quant_index_table pointer to the array

 * @param mlt_buffer        pointer to mlt coefficients

 */

static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,

                           int *quant_index_table, float *mlt_buffer)

{

    /* A zero in this table means that the subband coefficient is

       random noise coded. */

    int subband_coef_index[SUBBAND_SIZE];

    /* A zero in this table means that the subband coefficient is a

       positive multiplicator. */

    int subband_coef_sign[SUBBAND_SIZE];

    int band, j;

    int index = 0;

    for (band = 0; band < p->total_subbands; band++) {

        index = category[band];

        if (category[band] < 7) {

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

                index = 7;

                for (j = 0; j < p->total_subbands; j++)

                    category[band + j] = 7;

            }

        }

        if (index >= 7) {

            memset(subband_coef_index, 0, sizeof(subband_coef_index));

            memset(subband_coef_sign,  0, sizeof(subband_coef_sign));

        }

        q->scalar_dequant(q, index, quant_index_table[band],

                          subband_coef_index, subband_coef_sign,

                          &mlt_buffer[band * SUBBAND_SIZE]);

    }

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

    if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)

        return;

}

static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)

{

    int category_index[128] = { 0 };

    int category[128]       = { 0 };

    int quant_index_table[102];

    int res, i;

    if ((res = decode_envelope(q, p, quant_index_table)) < 0)

        return res;

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

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

    expand_category(q, category, category_index);

    for (i=0; itotal_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<js_vlc_bits)-1) {

                av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");

                return AVERROR_INVALIDDATA;

            }

            decouple_tab[start + i] = v;

        }

    return 0;

}

/**

 * function decouples a pair of signals from a single signal via multiplication.

 *

 * @param q                 pointer to the COOKContext

 * @param subband           index of the current subband

 * @param f1                multiplier for channel 1 extraction

 * @param f2                multiplier for channel 2 extraction

 * @param decode_buffer     input buffer

 * @param mlt_buffer1       pointer to left channel mlt coefficients

 * @param mlt_buffer2       pointer to right channel mlt coefficients

 */

static void decouple_float(COOKContext *q,

                           COOKSubpacket *p,

                           int subband,

                           float f1, float f2,

                           float *decode_buffer,

                           float *mlt_buffer1, float *mlt_buffer2)

{

    int j, tmp_idx;

    for (j = 0; j < SUBBAND_SIZE; j++) {

        tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;

        mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];

        mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];

    }

}

/**

 * function for decoding joint stereo data

 *

 * @param q                 pointer to the COOKContext

 * @param mlt_buffer1       pointer to left channel mlt coefficients

 * @param mlt_buffer2       pointer to right channel mlt coefficients

 */

static int joint_decode(COOKContext *q, COOKSubpacket *p,

                        float *mlt_buffer_left, float *mlt_buffer_right)

{

    int i, j, res;

    int decouple_tab[SUBBAND_SIZE] = { 0 };

    float *decode_buffer = q->decode_buffer_0;

    int idx, cpl_tmp;

    float f1, f2;

    const float *cplscale;

    memset(decode_buffer, 0, sizeof(q->decode_buffer_0));

    /* Make sure the buffers are zeroed out. */

    memset(mlt_buffer_left,  0, 1024 * sizeof(*mlt_buffer_left));

    memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));

    if ((res = decouple_info(q, p, decouple_tab)) < 0)

        return res;

    if ((res = mono_decode(q, p, decode_buffer)) < 0)

        return res;

    /* The two channels are stored interleaved in decode_buffer. */

    for (i = 0; i < p->js_subband_start; i++) {

        for (j = 0; j < SUBBAND_SIZE; j++) {

            mlt_buffer_left[i  * 20 + j] = decode_buffer[i * 40 + j];

            mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];

        }

    }

    /* When we reach js_subband_start (the higher frequencies)

       the coefficients are stored in a coupling scheme. */

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

    for (i = p->js_subband_start; i < p->subbands; i++) {

        cpl_tmp = cplband[i];

        idx -= decouple_tab[cpl_tmp];

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

        f1 = cplscale[decouple_tab[cpl_tmp] + 1];

        f2 = cplscale[idx];

        q->decouple(q, p, i, f1, f2, decode_buffer,

                    mlt_buffer_left, mlt_buffer_right);

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

    }

    return 0;

}

/**

 * First part of subpacket decoding:

 *  decode raw stream bytes and read gain info.

 *

 * @param q                 pointer to the COOKContext

 * @param inbuffer          pointer to raw stream data

 * @param gains_ptr         array of current/prev gain pointers

 */

static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,

                                         const uint8_t *inbuffer,

                                         cook_gains *gains_ptr)

{

    int offset;

    offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,

                          p->bits_per_subpacket / 8);

    init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,

                  p->bits_per_subpacket);

    decode_gain_info(&q->gb, gains_ptr->now);

    /* Swap current and previous gains */

    FFSWAP(int *, gains_ptr->now, gains_ptr->previous);

}

/**

 * Saturate the output signal and interleave.

 *

 * @param q                 pointer to the COOKContext

 * @param out               pointer to the output vector

 */

static void saturate_output_float(COOKContext *q, float *out)

{

    q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,

                         -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));

}

/**

 * Final part of subpacket decoding:

 *  Apply modulated lapped transform, gain compensation,

 *  clip and convert to integer.

 *

 * @param q                 pointer to the COOKContext

 * @param decode_buffer     pointer to the mlt coefficients

 * @param gains_ptr         array of current/prev gain pointers

 * @param previous_buffer   pointer to the previous buffer to be used for overlapping

 * @param out               pointer to the output buffer

 */

static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,

                                         cook_gains *gains_ptr, float *previous_buffer,

                                         float *out)

{

    imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);

    if (out)

        q->saturate_output(q, out);

}

/**

 * Cook subpacket decoding. This function returns one decoded subpacket,

 * usually 1024 samples per channel.

 *

 * @param q                 pointer to the COOKContext

 * @param inbuffer          pointer to the inbuffer

 * @param outbuffer         pointer to the outbuffer

 */

static int decode_subpacket(COOKContext *q, COOKSubpacket *p,

                            const uint8_t *inbuffer, float **outbuffer)

{

    int sub_packet_size = p->size;

    int res;

    memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));

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

    if (p->joint_stereo) {

        if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)

            return res;

    } else {

        if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)

            return res;

        if (p->num_channels == 2) {

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

            if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)

                return res;

        }

    }

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

                          p->mono_previous_buffer1,

                          outbuffer ? outbuffer[p->ch_idx] : NULL);

    if (p->num_channels == 2) {

        if (p->joint_stereo)

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

                                  p->mono_previous_buffer2,

                                  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);

        else

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

                                  p->mono_previous_buffer2,

                                  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);

    }

    return 0;

}

static int cook_decode_frame(AVCodecContext *avctx, void *data,

                             int *got_frame_ptr, AVPacket *avpkt)

{

    AVFrame *frame     = data;

    const uint8_t *buf = avpkt->data;

    int buf_size = avpkt->size;

    COOKContext *q = avctx->priv_data;

    float **samples = NULL;

    int i, ret;

    int offset = 0;

    int chidx = 0;

    if (buf_size < avctx->block_align)

        return buf_size;

    /* get output buffer */

    if (q->discarded_packets >= 2) {

        frame->nb_samples = q->samples_per_channel;

        if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)

            return ret;

        samples = (float **)frame->extended_data;

    }

    /* estimate subpacket sizes */

    q->subpacket[0].size = avctx->block_align;

    for (i = 1; i < q->num_subpackets; i++) {

        q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];

        q->subpacket[0].size -= q->subpacket[i].size + 1;

        if (q->subpacket[0].size < 0) {

            av_log(avctx, AV_LOG_DEBUG,

                   "frame subpacket size total > avctx->block_align!\n");

            return AVERROR_INVALIDDATA;

        }

    }

    /* decode supbackets */

    for (i = 0; i < q->num_subpackets; i++) {

        q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>

                                              q->subpacket[i].bits_per_subpdiv;

        q->subpacket[i].ch_idx = chidx;

        av_log(avctx, AV_LOG_DEBUG,

               "subpacket[%i] size %i js %i %i block_align %i\n",

               i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,

               avctx->block_align);

        if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)

            return ret;

        offset += q->subpacket[i].size;

        chidx += q->subpacket[i].num_channels;

        av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",

               i, q->subpacket[i].size * 8, get_bits_count(&q->gb));

    }

    /* Discard the first two frames: no valid audio. */

    if (q->discarded_packets < 2) {

        q->discarded_packets++;

        *got_frame_ptr = 0;

        return avctx->block_align;

    }

    *got_frame_ptr = 1;

    return avctx->block_align;

}

static void dump_cook_context(COOKContext *q)

{

    //int i=0;

#define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);

    ff_dlog(q->avctx, "COOKextradata\n");

    ff_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);

    }

    ff_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);

}

/**

 * Cook initialization

 *

 * @param avctx     pointer to the AVCodecContext

 */

static av_cold int cook_decode_init(AVCodecContext *avctx)

{

    COOKContext *q = avctx->priv_data;

    const uint8_t *edata_ptr = avctx->extradata;

    const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;

    int extradata_size = avctx->extradata_size;

    int s = 0;

    unsigned int channel_mask = 0;

    int samples_per_frame = 0;

    int ret;

    q->avctx = avctx;

    /* Take care of the codec specific extradata. */

    if (extradata_size < 8) {

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

        return AVERROR_INVALIDDATA;

    }

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

    /* Take data from the AVCodecContext (RM container). */

    if (!avctx->channels) {

        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");

        return AVERROR_INVALIDDATA;

    }

    /* Initialize RNG. */

    av_lfg_init(&q->random_state, 0);

    ff_audiodsp_init(&q->adsp);

    while (edata_ptr < edata_ptr_end) {

        /* 8 for mono, 16 for stereo, ? for multichannel

           Swap to right endianness so we don't need to care later on. */

        if (extradata_size >= 8) {

            q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);

            samples_per_frame           = bytestream_get_be16(&edata_ptr);

            q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);

            extradata_size -= 8;

        }

        if (extradata_size >= 8) {

            bytestream_get_be32(&edata_ptr);    // Unknown unused

            q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);

            if (q->subpacket[s].js_subband_start >= 51) {

                av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);

                return AVERROR_INVALIDDATA;

            }

            q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);

            extradata_size -= 8;

        }

        /* Initialize extradata related variables. */

        q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;

        q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;

        /* Initialize default data states. */

        q->subpacket[s].log2_numvector_size = 5;

        q->subpacket[s].total_subbands = q->subpacket[s].subbands;

        q->subpacket[s].num_channels = 1;

        /* Initialize version-dependent variables */

        av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,

               q->subpacket[s].cookversion);

        q->subpacket[s].joint_stereo = 0;

        switch (q->subpacket[s].cookversion) {

        case MONO:

            if (avctx->channels != 1) {

                avpriv_request_sample(avctx, "Container channels != 1");

                return AVERROR_PATCHWELCOME;

            }

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

            break;

        case STEREO:

            if (avctx->channels != 1) {

                q->subpacket[s].bits_per_subpdiv = 1;

                q->subpacket[s].num_channels = 2;

            }

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

            break;

        case JOINT_STEREO:

            if (avctx->channels != 2) {

                avpriv_request_sample(avctx, "Container channels != 2");

                return AVERROR_PATCHWELCOME;