cook.c 44.4 KB
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/*
 * COOK compatible decoder
 * Copyright (c) 2003 Sascha Sommer
 * Copyright (c) 2005 Benjamin Larsson
 *
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 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * 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
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */

/**
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 * @file
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 * Cook compatible decoder. Bastardization of the G.722.1 standard.
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 * 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.
 */

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#include "libavutil/channel_layout.h"
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#include "libavutil/lfg.h"
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#include "audiodsp.h"
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#include "avcodec.h"
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#include "get_bits.h"
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#include "bytestream.h"
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#include "fft.h"
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#include "internal.h"
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#include "sinewin.h"
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#include "unary.h"
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#include "cookdata.h"

/* the different Cook versions */
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#define MONO            0x1000001
#define STEREO          0x1000002
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#define JOINT_STEREO    0x1000003
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#define MC_COOK         0x2000000   // multichannel Cook, not supported
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#define SUBBAND_SIZE    20
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#define MAX_SUBPACKETS   5
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typedef struct cook_gains {
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    int *now;
    int *previous;
} cook_gains;
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typedef struct COOKSubpacket {
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    int                 ch_idx;
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    int                 size;
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    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;
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    VLC                 channel_coupling;
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    int                 joint_stereo;
    int                 bits_per_subpacket;
    int                 bits_per_subpdiv;
    int                 total_subbands;
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    int                 numvector_size;       // 1 << log2_numvector_size;
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    float               mono_previous_buffer1[1024];
    float               mono_previous_buffer2[1024];
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    cook_gains          gains1;
    cook_gains          gains2;
    int                 gain_1[9];
    int                 gain_2[9];
    int                 gain_3[9];
    int                 gain_4[9];
} COOKSubpacket;

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typedef struct cook {
    /*
     * The following 5 functions provide the lowlevel arithmetic on
     * the internal audio buffers.
     */
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    void (*scalar_dequant)(struct cook *q, int index, int quant_index,
                           int *subband_coef_index, int *subband_coef_sign,
                           float *mlt_p);
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    void (*decouple)(struct cook *q,
                     COOKSubpacket *p,
                     int subband,
                     float f1, float f2,
                     float *decode_buffer,
                     float *mlt_buffer1, float *mlt_buffer2);
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    void (*imlt_window)(struct cook *q, float *buffer1,
                        cook_gains *gains_ptr, float *previous_buffer);
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    void (*interpolate)(struct cook *q, float *buffer,
                        int gain_index, int gain_index_next);
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    void (*saturate_output)(struct cook *q, float *out);
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    AVCodecContext*     avctx;
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    AudioDSPContext     adsp;
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    GetBitContext       gb;
    /* stream data */
    int                 num_vectors;
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    int                 samples_per_channel;
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    /* states */
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    AVLFG               random_state;
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    int                 discarded_packets;
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    /* transform data */
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    FFTContext          mdct_ctx;
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    float*              mlt_window;

    /* VLC data */
    VLC                 envelope_quant_index[13];
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    VLC                 sqvh[7];          // scalar quantization
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    /* generate tables and related variables */
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    int                 gain_size_factor;
    float               gain_table[23];

    /* data buffers */

    uint8_t*            decoded_bytes_buffer;
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    DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
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    float               decode_buffer_1[1024];
    float               decode_buffer_2[1024];
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    float               decode_buffer_0[1060]; /* static allocation for joint decode */
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    const float         *cplscales[5];
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    int                 num_subpackets;
    COOKSubpacket       subpacket[MAX_SUBPACKETS];
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} COOKContext;

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

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

/* table generator */
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static av_cold void init_pow2table(void)
{
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    /* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */
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    int i;
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    static const float exp2_tab[2] = {1, M_SQRT2};
    float exp2_val = powf(2, -63);
    float root_val = powf(2, -32);
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    for (i = -63; i < 64; i++) {
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        if (!(i & 1))
            root_val *= 2;
        pow2tab[63 + i] = exp2_val;
        rootpow2tab[63 + i] = root_val * exp2_tab[i & 1];
        exp2_val *= 2;
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    }
}

/* table generator */
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static av_cold void init_gain_table(COOKContext *q)
{
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    int i;
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    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));
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}


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static av_cold int init_cook_vlc_tables(COOKContext *q)
{
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    int i, result;

    result = 0;
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    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);
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    }
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    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);
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    }

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    for (i = 0; i < q->num_subpackets; i++) {
        if (q->subpacket[i].joint_stereo == 1) {
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            result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
                               (1 << q->subpacket[i].js_vlc_bits) - 1,
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                               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);
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        }
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    }

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

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static av_cold int init_cook_mlt(COOKContext *q)
{
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    int j, ret;
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    int mlt_size = q->samples_per_channel;
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    if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)
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        return AVERROR(ENOMEM);
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    /* Initialize the MLT window: simple sine window. */
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    ff_sine_window_init(q->mlt_window, mlt_size);
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    for (j = 0; j < mlt_size; j++)
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        q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
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    /* Initialize the MDCT. */
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    if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
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        av_freep(&q->mlt_window);
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        return ret;
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    }
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    av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
           av_log2(mlt_size) + 1);
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    return 0;
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}

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static av_cold void init_cplscales_table(COOKContext *q)
{
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    int i;
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    for (i = 0; i < 5; i++)
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        q->cplscales[i] = cplscales[i];
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}

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

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

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/**
 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
 * Why? No idea, some checksum/error detection method maybe.
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 *
 * Out buffer size: extra bytes are needed to cope with
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 * padding/misalignment.
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 * 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
 *
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 * Nice way to waste CPU cycles.
 *
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 * @param inbuffer  pointer to byte array of indata
 * @param out       pointer to byte array of outdata
 * @param bytes     number of bytes
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 */
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static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
{
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    static const uint32_t tab[4] = {
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        AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
        AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
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    };
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    int i, off;
    uint32_t c;
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    const uint32_t *buf;
    uint32_t *obuf = (uint32_t *) out;
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    /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
     * I'm too lazy though, should be something like
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     * for (i = 0; i < bitamount / 64; i++)
     *     (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
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     * Buffer alignment needs to be checked. */

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    off = (intptr_t) inbuffer & 3;
    buf = (const uint32_t *) (inbuffer - off);
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    c = tab[off];
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    bytes += 3 + off;
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    for (i = 0; i < bytes / 4; i++)
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        obuf[i] = c ^ buf[i];
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    return off;
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}

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static av_cold int cook_decode_close(AVCodecContext *avctx)
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{
    int i;
    COOKContext *q = avctx->priv_data;
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    av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
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    /* Free allocated memory buffers. */
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    av_freep(&q->mlt_window);
    av_freep(&q->decoded_bytes_buffer);
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    /* Free the transform. */
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    ff_mdct_end(&q->mdct_ctx);
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    /* Free the VLC tables. */
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    for (i = 0; i < 13; i++)
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        ff_free_vlc(&q->envelope_quant_index[i]);
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    for (i = 0; i < 7; i++)
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        ff_free_vlc(&q->sqvh[i]);
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    for (i = 0; i < q->num_subpackets; i++)
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        ff_free_vlc(&q->subpacket[i].channel_coupling);
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    av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
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    return 0;
}

/**
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 * Fill the gain array for the timedomain quantization.
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 *
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 * @param gb          pointer to the GetBitContext
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 * @param gaininfo    array[9] of gain indexes
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 */
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static void decode_gain_info(GetBitContext *gb, int *gaininfo)
{
    int i, n;
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    n = get_unary(gb, 0, get_bits_left(gb));     // amount of elements*2 to update
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    i = 0;
    while (n--) {
        int index = get_bits(gb, 3);
        int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
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        while (i <= index)
            gaininfo[i++] = gain;
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    }
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    while (i <= 8)
        gaininfo[i++] = 0;
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}

/**
 * Create the quant index table needed for the envelope.
 *
 * @param q                 pointer to the COOKContext
 * @param quant_index_table pointer to the array
 */
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static int decode_envelope(COOKContext *q, COOKSubpacket *p,
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                           int *quant_index_table)
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{
    int i, j, vlc_index;
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    quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
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    for (i = 1; i < p->total_subbands; i++) {
        vlc_index = i;
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        if (i >= p->js_subband_start * 2) {
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            vlc_index -= p->js_subband_start;
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        } else {
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            vlc_index /= 2;
            if (vlc_index < 1)
                vlc_index = 1;
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        }
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        if (vlc_index > 13)
            vlc_index = 13; // the VLC tables >13 are identical to No. 13
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        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
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        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);
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            return AVERROR_INVALIDDATA;
        }
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    }
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    return 0;
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}

/**
 * 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
 */
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static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
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                       int *category, int *category_index)
{
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    int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
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    int exp_index2[102] = { 0 };
    int exp_index1[102] = { 0 };
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    int tmp_categorize_array[128 * 2] = { 0 };
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    int tmp_categorize_array1_idx = p->numvector_size;
    int tmp_categorize_array2_idx = p->numvector_size;
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    bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
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    if (bits_left > q->samples_per_channel)
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        bits_left = q->samples_per_channel +
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                    ((bits_left - q->samples_per_channel) * 5) / 8;
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    bias = -32;
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    /* Estimate bias. */
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    for (i = 32; i > 0; i = i / 2) {
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        num_bits = 0;
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        index    = 0;
        for (j = p->total_subbands; j > 0; j--) {
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            exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);
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            index++;
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            num_bits += expbits_tab[exp_idx];
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        }
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        if (num_bits >= bits_left - 32)
            bias += i;
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    }

    /* Calculate total number of bits. */
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    num_bits = 0;
    for (i = 0; i < p->total_subbands; i++) {
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        exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);
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        num_bits += expbits_tab[exp_idx];
        exp_index1[i] = exp_idx;
        exp_index2[i] = exp_idx;
    }
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    tmpbias1 = tmpbias2 = num_bits;
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    for (j = 1; j < p->numvector_size; j++) {
        if (tmpbias1 + tmpbias2 > 2 * bits_left) {  /* ---> */
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            int max = -999999;
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            index = -1;
            for (i = 0; i < p->total_subbands; i++) {
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                if (exp_index1[i] < 7) {
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                    v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
                    if (v >= max) {
                        max   = v;
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                        index = i;
                    }
                }
            }
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            if (index == -1)
                break;
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            tmp_categorize_array[tmp_categorize_array1_idx++] = index;
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            tmpbias1 -= expbits_tab[exp_index1[index]] -
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                        expbits_tab[exp_index1[index] + 1];
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            ++exp_index1[index];
        } else {  /* <--- */
            int min = 999999;
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            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;
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                        index = i;
                    }
                }
            }
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            if (index == -1)
                break;
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            tmp_categorize_array[--tmp_categorize_array2_idx] = index;
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            tmpbias2 -= expbits_tab[exp_index2[index]] -
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                        expbits_tab[exp_index2[index] - 1];
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            --exp_index2[index];
        }
    }

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    for (i = 0; i < p->total_subbands; i++)
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        category[i] = exp_index2[i];

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    for (i = 0; i < p->numvector_size - 1; i++)
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        category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
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}


/**
 * 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
 */
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static inline void expand_category(COOKContext *q, int *category,
                                   int *category_index)
{
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    int i;
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    for (i = 0; i < q->num_vectors; i++)
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    {
        int idx = category_index[i];
        if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
            --category[idx];
    }
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}

/**
 * The real requantization of the mltcoefs
 *
 * @param q                     pointer to the COOKContext
 * @param index                 index
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 * @param quant_index           quantisation index
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 * @param subband_coef_index    array of indexes to quant_centroid_tab
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 * @param subband_coef_sign     signs of coefficients
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 * @param mlt_p                 pointer into the mlt buffer
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 */
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static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
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                                 int *subband_coef_index, int *subband_coef_sign,
                                 float *mlt_p)
{
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    int i;
    float f1;

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    for (i = 0; i < SUBBAND_SIZE; i++) {
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        if (subband_coef_index[i]) {
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            f1 = quant_centroid_tab[index][subband_coef_index[i]];
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            if (subband_coef_sign[i])
                f1 = -f1;
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        } else {
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            /* noise coding if subband_coef_index[i] == 0 */
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            f1 = dither_tab[index];
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            if (av_lfg_get(&q->random_state) < 0x80000000)
                f1 = -f1;
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        }
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        mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
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    }
}
/**
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 * Unpack the subband_coef_index and subband_coef_sign vectors.
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 *
 * @param q                     pointer to the COOKContext
 * @param category              pointer to the category array
 * @param subband_coef_index    array of indexes to quant_centroid_tab
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 * @param subband_coef_sign     signs of coefficients
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 */
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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;
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    vd = vd_tab[category];
    result = 0;
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    for (i = 0; i < vpr_tab[category]; i++) {
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        vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
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        if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
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            vlc = 0;
            result = 1;
        }
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        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);
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            vlc = tmp;
        }
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        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);
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                } else {
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                    result = 1;
                    subband_coef_sign[i * vd + j] = 0;
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                }
            } else {
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                subband_coef_sign[i * vd + j] = 0;
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            }
        }
    }
    return result;
}


/**
 * Fill the mlt_buffer with mlt coefficients.
 *
 * @param q                 pointer to the COOKContext
 * @param category          pointer to the category array
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 * @param quant_index_table pointer to the array
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 * @param mlt_buffer        pointer to mlt coefficients
 */
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static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
                           int *quant_index_table, float *mlt_buffer)
{
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    /* A zero in this table means that the subband coefficient is
       random noise coded. */
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    int subband_coef_index[SUBBAND_SIZE];
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    /* A zero in this table means that the subband coefficient is a
       positive multiplicator. */
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    int subband_coef_sign[SUBBAND_SIZE];
605
    int band, j;
606
    int index = 0;
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    for (band = 0; band < p->total_subbands; band++) {
609
        index = category[band];
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        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;
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            }
        }
617
        if (index >= 7) {
618
            memset(subband_coef_index, 0, sizeof(subband_coef_index));
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            memset(subband_coef_sign,  0, sizeof(subband_coef_sign));
620
        }
621
        q->scalar_dequant(q, index, quant_index_table[band],
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                          subband_coef_index, subband_coef_sign,
                          &mlt_buffer[band * SUBBAND_SIZE]);
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    }

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    /* FIXME: should this be removed, or moved into loop above? */
    if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
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        return;
}


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static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
633
{
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    int category_index[128] = { 0 };
    int category[128]       = { 0 };
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    int quant_index_table[102];
637
    int res, i;
638

639 640
    if ((res = decode_envelope(q, p, quant_index_table)) < 0)
        return res;
641
    q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);
642
    categorize(q, p, quant_index_table, category, category_index);
643
    expand_category(q, category, category_index);
644 645 646 647
    for (i=0; i<p->total_subbands; i++) {
        if (category[i] > 7)
            return AVERROR_INVALIDDATA;
    }
648
    decode_vectors(q, p, category, quant_index_table, mlt_buffer);
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    return 0;
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}


/**
 * 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
 */
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static void interpolate_float(COOKContext *q, float *buffer,
                              int gain_index, int gain_index_next)
{
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    int i;
    float fc1, fc2;
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    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;
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        }
    }
}

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/**
 * Apply transform window, overlap buffers.
 *
 * @param q                 pointer to the COOKContext
685
 * @param inbuffer          pointer to the mltcoefficients
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 * @param gains_ptr         current and previous gains
 * @param previous_buffer   pointer to the previous buffer to be used for overlapping
 */
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static void imlt_window_float(COOKContext *q, float *inbuffer,
                              cook_gains *gains_ptr, float *previous_buffer)
691
{
692
    const float fc = pow2tab[gains_ptr->previous[0] + 63];
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    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 */
701
    for (i = 0; i < q->samples_per_channel; i++)
702
        inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
703
                      previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
704
}
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/**
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 * 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.
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 *
 * @param q                 pointer to the COOKContext
713
 * @param inbuffer          pointer to the mltcoefficients
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 * @param gains_ptr         current and previous gains
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 * @param previous_buffer   pointer to the previous buffer to be used for overlapping
 */
717
static void imlt_gain(COOKContext *q, float *inbuffer,
718
                      cook_gains *gains_ptr, float *previous_buffer)
719
{
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    float *buffer0 = q->mono_mdct_output;
    float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
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    int i;

724
    /* Inverse modified discrete cosine transform */
725
    q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
726

727
    q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
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    /* Apply gain profile */
730
    for (i = 0; i < 8; i++)
731
        if (gains_ptr->now[i] || gains_ptr->now[i + 1])
732
            q->interpolate(q, &buffer1[q->gain_size_factor * i],
733
                           gains_ptr->now[i], gains_ptr->now[i + 1]);
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    /* Save away the current to be previous block. */
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    memcpy(previous_buffer, buffer0,
           q->samples_per_channel * sizeof(*previous_buffer));
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}


/**
 * function for getting the jointstereo coupling information
 *
 * @param q                 pointer to the COOKContext
 * @param decouple_tab      decoupling array
 */
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static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
748 749 750 751
{
    int i;
    int vlc    = get_bits1(&q->gb);
    int start  = cplband[p->js_subband_start];
752
    int end    = cplband[p->subbands - 1];
753
    int length = end - start + 1;
754

755
    if (start > end)
756
        return 0;
757

758
    if (vlc)
759
        for (i = 0; i < length; i++)
760 761 762
            decouple_tab[start + i] = get_vlc2(&q->gb,
                                               p->channel_coupling.table,
                                               p->channel_coupling.bits, 2);
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    else
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        for (i = 0; i < length; i++) {
            int v = get_bits(&q->gb, p->js_vlc_bits);
            if (v == (1<<p->js_vlc_bits)-1) {
                av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
                return AVERROR_INVALIDDATA;
            }
            decouple_tab[start + i] = v;
        }
    return 0;
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}

775
/**
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 * 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
 */
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static void decouple_float(COOKContext *q,
                           COOKSubpacket *p,
                           int subband,
                           float f1, float f2,
                           float *decode_buffer,
                           float *mlt_buffer1, float *mlt_buffer2)
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{
    int j, tmp_idx;
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    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];
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    }
}
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/**
 * 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
 */
808 809
static int joint_decode(COOKContext *q, COOKSubpacket *p,
                        float *mlt_buffer_left, float *mlt_buffer_right)
810
{
811
    int i, j, res;
812
    int decouple_tab[SUBBAND_SIZE] = { 0 };
813
    float *decode_buffer = q->decode_buffer_0;
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814
    int idx, cpl_tmp;
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    float f1, f2;
    const float *cplscale;
817

818
    memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
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    /* Make sure the buffers are zeroed out. */
821 822
    memset(mlt_buffer_left,  0, 1024 * sizeof(*mlt_buffer_left));
    memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
823 824
    if ((res = decouple_info(q, p, decouple_tab)) < 0)
        return res;
825 826
    if ((res = mono_decode(q, p, decode_buffer)) < 0)
        return res;
827
    /* The two channels are stored interleaved in decode_buffer. */
828 829
    for (i = 0; i < p->js_subband_start; i++) {
        for (j = 0; j < SUBBAND_SIZE; j++) {
830 831
            mlt_buffer_left[i  * 20 + j] = decode_buffer[i * 40 + j];
            mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
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        }
    }

    /* When we reach js_subband_start (the higher frequencies)
       the coefficients are stored in a coupling scheme. */
837
    idx = (1 << p->js_vlc_bits) - 1;
838
    for (i = p->js_subband_start; i < p->subbands; i++) {
839
        cpl_tmp = cplband[i];
840 841
        idx -= decouple_tab[cpl_tmp];
        cplscale = q->cplscales[p->js_vlc_bits - 2];  // choose decoupler table
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        f1 = cplscale[decouple_tab[cpl_tmp] + 1];
        f2 = cplscale[idx];
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        q->decouple(q, p, i, f1, f2, decode_buffer,
                    mlt_buffer_left, mlt_buffer_right);
846
        idx = (1 << p->js_vlc_bits) - 1;
847
    }
848

849
    return 0;
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}

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/**
 * 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
858
 * @param gains_ptr         array of current/prev gain pointers
859
 */
860 861 862
static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
                                         const uint8_t *inbuffer,
                                         cook_gains *gains_ptr)
863 864 865 866
{
    int offset;

    offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
867
                          p->bits_per_subpacket / 8);
868
    init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
869
                  p->bits_per_subpacket);
870
    decode_gain_info(&q->gb, gains_ptr->now);
871 872

    /* Swap current and previous gains */
873
    FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
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}

876
/**
877
 * Saturate the output signal and interleave.
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 *
 * @param q                 pointer to the COOKContext
 * @param out               pointer to the output vector
 */
882
static void saturate_output_float(COOKContext *q, float *out)
883
{
884
    q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
885
                         FFALIGN(q->samples_per_channel, 8), -1.0f, 1.0f);
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}

888

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/**
 * 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
896
 * @param gains_ptr         array of current/prev gain pointers
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 * @param previous_buffer   pointer to the previous buffer to be used for overlapping
 * @param out               pointer to the output buffer
 */
900 901
static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
                                         cook_gains *gains_ptr, float *previous_buffer,
902
                                         float *out)
903
{
904
    imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
905
    if (out)
906
        q->saturate_output(q, out);
907 908 909
}


910 911 912 913 914 915 916 917
/**
 * 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
 */
918
static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
919
                            const uint8_t *inbuffer, float **outbuffer)
920
{
921
    int sub_packet_size = p->size;
922
    int res;
923

924
    memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
925
    decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
926

927
    if (p->joint_stereo) {
928 929
        if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
            return res;
930
    } else {
931 932
        if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
            return res;
933

934
        if (p->num_channels == 2) {
935
            decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
936 937
            if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
                return res;
938 939
        }
    }
940

941
    mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
942 943
                          p->mono_previous_buffer1,
                          outbuffer ? outbuffer[p->ch_idx] : NULL);
944

945
    if (p->num_channels == 2) {
946
        if (p->joint_stereo)
947
            mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
948 949
                                  p->mono_previous_buffer2,
                                  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
950
        else
951
            mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
952 953
                                  p->mono_previous_buffer2,
                                  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
954
    }
955

956
    return 0;
957 958 959
}


960 961 962
static int cook_decode_frame(AVCodecContext *avctx, void *data,
                             int *got_frame_ptr, AVPacket *avpkt)
{
963
    AVFrame *frame     = data;
964 965
    const uint8_t *buf = avpkt->data;
    int buf_size = avpkt->size;
966
    COOKContext *q = avctx->priv_data;
967
    float **samples = NULL;
968
    int i, ret;
969 970
    int offset = 0;
    int chidx = 0;
971 972 973 974

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

975 976
    /* get output buffer */
    if (q->discarded_packets >= 2) {
977
        frame->nb_samples = q->samples_per_channel;
978
        if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
979
            return ret;
980
        samples = (float **)frame->extended_data;
981 982
    }

983 984 985
    /* estimate subpacket sizes */
    q->subpacket[0].size = avctx->block_align;

986
    for (i = 1; i < q->num_subpackets; i++) {
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987
        q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
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988
        q->subpacket[0].size -= q->subpacket[i].size + 1;
989
        if (q->subpacket[0].size < 0) {
990 991
            av_log(avctx, AV_LOG_DEBUG,
                   "frame subpacket size total > avctx->block_align!\n");
992
            return AVERROR_INVALIDDATA;
993
        }
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994
    }
995

996
    /* decode supbackets */
997 998 999
    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;
1000
        q->subpacket[i].ch_idx = chidx;
1001 1002 1003 1004 1005
        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);

1006 1007
        if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
            return ret;
1008 1009
        offset += q->subpacket[i].size;
        chidx += q->subpacket[i].num_channels;
1010 1011
        av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
               i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
1012
    }
1013

1014
    /* Discard the first two frames: no valid audio. */
1015 1016 1017 1018 1019 1020
    if (q->discarded_packets < 2) {
        q->discarded_packets++;
        *got_frame_ptr = 0;
        return avctx->block_align;
    }

1021
    *got_frame_ptr = 1;
1022

1023 1024
    return avctx->block_align;
}
1025

1026
static void dump_cook_context(COOKContext *q)
1027 1028
{
    //int i=0;
1029 1030 1031
#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);
1032
    if (q->subpacket[0].cookversion > STEREO) {
1033 1034
        PRINT("js_subband_start", q->subpacket[0].js_subband_start);
        PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1035
    }
1036
    ff_dlog(q->avctx, "COOKContext\n");
1037
    PRINT("nb_channels", q->avctx->channels);
1038
    PRINT("bit_rate", (int)q->avctx->bit_rate);
1039
    PRINT("sample_rate", q->avctx->sample_rate);
1040 1041 1042 1043 1044 1045
    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);
1046
}
1047

1048 1049 1050 1051 1052
/**
 * Cook initialization
 *
 * @param avctx     pointer to the AVCodecContext
 */
1053
static av_cold int cook_decode_init(AVCodecContext *avctx)
1054 1055
{
    COOKContext *q = avctx->priv_data;
1056
    GetByteContext gb;
1057
    int s = 0;
1058
    unsigned int channel_mask = 0;
1059
    int samples_per_frame = 0;
1060
    int ret;
1061
    q->avctx = avctx;
1062 1063

    /* Take care of the codec specific extradata. */
1064
    if (avctx->extradata_size < 8) {
1065
        av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1066
        return AVERROR_INVALIDDATA;
1067
    }
1068
    av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1069

1070 1071
    bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);

1072
    /* Take data from the AVCodecContext (RM container). */
1073
    if (!avctx->channels) {
1074 1075 1076
        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
        return AVERROR_INVALIDDATA;
    }
1077

1078
    /* Initialize RNG. */
1079
    av_lfg_init(&q->random_state, 0);
1080

1081
    ff_audiodsp_init(&q->adsp);
1082

1083
    while (bytestream2_get_bytes_left(&gb)) {
1084 1085
        /* 8 for mono, 16 for stereo, ? for multichannel
           Swap to right endianness so we don't need to care later on. */
1086 1087 1088 1089 1090
        q->subpacket[s].cookversion      = bytestream2_get_be32(&gb);
        samples_per_frame                = bytestream2_get_be16(&gb);
        q->subpacket[s].subbands         = bytestream2_get_be16(&gb);
        bytestream2_get_be32(&gb);    // Unknown unused
        q->subpacket[s].js_subband_start = bytestream2_get_be16(&gb);
1091 1092 1093
        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;
1094
        }
1095
        q->subpacket[s].js_vlc_bits      = bytestream2_get_be16(&gb);
1096 1097

        /* Initialize extradata related variables. */
1098
        q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1099 1100 1101 1102 1103 1104 1105 1106 1107
        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 */

1108 1109
        av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
               q->subpacket[s].cookversion);
1110 1111
        q->subpacket[s].joint_stereo = 0;
        switch (q->subpacket[s].cookversion) {
1112
        case MONO:
1113
            if (avctx->channels != 1) {
1114
                avpriv_request_sample(avctx, "Container channels != 1");
1115 1116 1117 1118 1119
                return AVERROR_PATCHWELCOME;
            }
            av_log(avctx, AV_LOG_DEBUG, "MONO\n");
            break;
        case STEREO:
1120
            if (avctx->channels != 1) {
1121 1122 1123 1124 1125 1126
                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:
1127
            if (avctx->channels != 2) {
1128
                avpriv_request_sample(avctx, "Container channels != 2");
1129 1130 1131 1132 1133 1134 1135 1136 1137 1138 1139 1140 1141 1142 1143 1144 1145 1146
                return AVERROR_PATCHWELCOME;
            }
            av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
            if (avctx->extradata_size >= 16) {
                q->subpacket[s].total_subbands = q->subpacket[s].subbands +
                                                 q->subpacket[s].js_subband_start;
                q->subpacket[s].joint_stereo = 1;
                q->subpacket[s].num_channels = 2;
            }
            if (q->subpacket[s].samples_per_channel > 256) {
                q->subpacket[s].log2_numvector_size = 6;
            }
            if (q->subpacket[s].samples_per_channel > 512) {
                q->subpacket[s].log2_numvector_size = 7;
            }
            break;
        case MC_COOK:
            av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1147
            channel_mask |= q->subpacket[s].channel_mask = bytestream2_get_be32(&gb);
1148

1149
            if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {
1150 1151 1152 1153
                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;
1154
                q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1155

1156
                if (q->subpacket[s].samples_per_channel > 256) {
1157
                    q->subpacket[s].log2_numvector_size = 6;
1158 1159
                }
                if (q->subpacket[s].samples_per_channel > 512) {
1160
                    q->subpacket[s].log2_numvector_size = 7;
1161
                }
1162
            } else
1163
                q->subpacket[s].samples_per_channel = samples_per_frame;
1164

1165 1166
            break;
        default:
1167 1168
            avpriv_request_sample(avctx, "Cook version %d",
                                  q->subpacket[s].cookversion);
1169
            return AVERROR_PATCHWELCOME;
1170 1171
        }

1172 1173
        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");
1174
            return AVERROR_INVALIDDATA;
1175 1176 1177 1178 1179 1180 1181
        } 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);

1182
        /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1183
        if (q->subpacket[s].total_subbands > 53) {
1184
            avpriv_request_sample(avctx, "total_subbands > 53");
1185
            return AVERROR_PATCHWELCOME;
1186 1187
        }

1188 1189 1190 1191
        if ((q->subpacket[s].js_vlc_bits > 6) ||
            (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
            av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
                   q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1192
            return AVERROR_INVALIDDATA;
1193
        }
1194

1195
        if (q->subpacket[s].subbands > 50) {
1196
            avpriv_request_sample(avctx, "subbands > 50");
1197
            return AVERROR_PATCHWELCOME;
1198
        }
1199
        if (q->subpacket[s].subbands == 0) {
1200
            avpriv_request_sample(avctx, "subbands = 0");
1201 1202
            return AVERROR_PATCHWELCOME;
        }
1203 1204 1205 1206
        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;
1207

1208 1209
        if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
            av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
1210 1211 1212
            return AVERROR_INVALIDDATA;
        }

1213 1214
        q->num_subpackets++;
        s++;
1215 1216
        if (s > FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
            avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
1217
            return AVERROR_PATCHWELCOME;
1218
        }
1219
    }
1220
    /* Generate tables */
1221
    init_pow2table();
1222
    init_gain_table(q);
1223
    init_cplscales_table(q);
1224

1225 1226
    if ((ret = init_cook_vlc_tables(q)))
        return ret;
1227

1228

1229
    if (avctx->block_align >= UINT_MAX / 2)
1230
        return AVERROR(EINVAL);
1231

1232 1233
    /* Pad the databuffer with:
       DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1234
       AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1235 1236 1237
    q->decoded_bytes_buffer =
        av_mallocz(avctx->block_align
                   + DECODE_BYTES_PAD1(avctx->block_align)
1238
                   + AV_INPUT_BUFFER_PADDING_SIZE);
1239
    if (!q->decoded_bytes_buffer)
1240
        return AVERROR(ENOMEM);
1241 1242

    /* Initialize transform. */
1243 1244
    if ((ret = init_cook_mlt(q)))
        return ret;
1245

1246 1247
    /* Initialize COOK signal arithmetic handling */
    if (1) {
1248
        q->scalar_dequant  = scalar_dequant_float;
1249 1250
        q->decouple        = decouple_float;
        q->imlt_window     = imlt_window_float;
1251
        q->interpolate     = interpolate_float;
1252 1253 1254
        q->saturate_output = saturate_output_float;
    }

1255
    /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1256 1257
    if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
        q->samples_per_channel != 1024) {
1258
        avpriv_request_sample(avctx, "samples_per_channel = %d",
1259
                              q->samples_per_channel);
1260
        return AVERROR_PATCHWELCOME;
1261
    }
1262

1263
    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1264 1265 1266
    if (channel_mask)
        avctx->channel_layout = channel_mask;
    else
1267
        avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1268

1269

1270
    dump_cook_context(q);
1271

1272 1273 1274
    return 0;
}

1275 1276
AVCodec ff_cook_decoder = {
    .name           = "cook",
1277
    .long_name      = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1278
    .type           = AVMEDIA_TYPE_AUDIO,
1279
    .id             = AV_CODEC_ID_COOK,
1280
    .priv_data_size = sizeof(COOKContext),
1281 1282 1283
    .init           = cook_decode_init,
    .close          = cook_decode_close,
    .decode         = cook_decode_frame,
1284
    .capabilities   = AV_CODEC_CAP_DR1,
1285
    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,
1286 1287
    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
                                                      AV_SAMPLE_FMT_NONE },
1288
};