making act segmenter

1 files changed, 1292 insertions, 0 deletions

diff --git a/ffmpeg/libavcodec/cook.c b/ffmpeg/libavcodec/cook.c
new file mode 100644
index 0000000..08cd401
--- /dev/null
+++ b/ffmpeg/libavcodec/cook.c
@@ -0,0 +1,1292 @@
+/*
+ * COOK compatible decoder
+ * Copyright (c) 2003 Sascha Sommer
+ * Copyright (c) 2005 Benjamin Larsson
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * Cook compatible decoder. Bastardization of the G.722.1 standard.
+ * This decoder handles RealNetworks, RealAudio G2 data.
+ * Cook is identified by the codec name cook in RM files.
+ *
+ * To use this decoder, a calling application must supply the extradata
+ * bytes provided from the RM container; 8+ bytes for mono streams and
+ * 16+ for stereo streams (maybe more).
+ *
+ * Codec technicalities (all this assume a buffer length of 1024):
+ * Cook works with several different techniques to achieve its compression.
+ * In the timedomain the buffer is divided into 8 pieces and quantized. If
+ * two neighboring pieces have different quantization index a smooth
+ * quantization curve is used to get a smooth overlap between the different
+ * pieces.
+ * To get to the transformdomain Cook uses a modulated lapped transform.
+ * The transform domain has 50 subbands with 20 elements each. This
+ * means only a maximum of 50*20=1000 coefficients are used out of the 1024
+ * available.
+ */
+
+#include "libavutil/channel_layout.h"
+#include "libavutil/lfg.h"
+#include "avcodec.h"
+#include "get_bits.h"
+#include "dsputil.h"
+#include "bytestream.h"
+#include "fft.h"
+#include "internal.h"
+#include "sinewin.h"
+
+#include "cookdata.h"
+
+/* the different Cook versions */
+#define MONO            0x1000001
+#define STEREO          0x1000002
+#define JOINT_STEREO    0x1000003
+#define MC_COOK         0x2000000   // multichannel Cook, not supported
+
+#define SUBBAND_SIZE    20
+#define MAX_SUBPACKETS   5
+
+typedef struct {
+    int *now;
+    int *previous;
+} cook_gains;
+
+typedef struct {
+    int                 ch_idx;
+    int                 size;
+    int                 num_channels;
+    int                 cookversion;
+    int                 subbands;
+    int                 js_subband_start;
+    int                 js_vlc_bits;
+    int                 samples_per_channel;
+    int                 log2_numvector_size;
+    unsigned int        channel_mask;
+    VLC                 channel_coupling;
+    int                 joint_stereo;
+    int                 bits_per_subpacket;
+    int                 bits_per_subpdiv;
+    int                 total_subbands;
+    int                 numvector_size;       // 1 << log2_numvector_size;
+
+    float               mono_previous_buffer1[1024];
+    float               mono_previous_buffer2[1024];
+
+    cook_gains          gains1;
+    cook_gains          gains2;
+    int                 gain_1[9];
+    int                 gain_2[9];
+    int                 gain_3[9];
+    int                 gain_4[9];
+} COOKSubpacket;
+
+typedef struct cook {
+    /*
+     * The following 5 functions provide the lowlevel arithmetic on
+     * the internal audio buffers.
+     */
+    void (*scalar_dequant)(struct cook *q, int index, int quant_index,
+                           int *subband_coef_index, int *subband_coef_sign,
+                           float *mlt_p);
+
+    void (*decouple)(struct cook *q,
+                     COOKSubpacket *p,
+                     int subband,
+                     float f1, float f2,
+                     float *decode_buffer,
+                     float *mlt_buffer1, float *mlt_buffer2);
+
+    void (*imlt_window)(struct cook *q, float *buffer1,
+                        cook_gains *gains_ptr, float *previous_buffer);
+
+    void (*interpolate)(struct cook *q, float *buffer,
+                        int gain_index, int gain_index_next);
+
+    void (*saturate_output)(struct cook *q, float *out);
+
+    AVCodecContext*     avctx;
+    DSPContext          dsp;
+    GetBitContext       gb;
+    /* stream data */
+    int                 num_vectors;
+    int                 samples_per_channel;
+    /* states */
+    AVLFG               random_state;
+    int                 discarded_packets;
+
+    /* transform data */
+    FFTContext          mdct_ctx;
+    float*              mlt_window;
+
+    /* VLC data */
+    VLC                 envelope_quant_index[13];
+    VLC                 sqvh[7];          // scalar quantization
+
+    /* generatable tables and related variables */
+    int                 gain_size_factor;
+    float               gain_table[23];
+
+    /* data buffers */
+
+    uint8_t*            decoded_bytes_buffer;
+    DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
+    float               decode_buffer_1[1024];
+    float               decode_buffer_2[1024];
+    float               decode_buffer_0[1060]; /* static allocation for joint decode */
+
+    const float         *cplscales[5];
+    int                 num_subpackets;
+    COOKSubpacket       subpacket[MAX_SUBPACKETS];
+} COOKContext;
+
+static float     pow2tab[127];
+static float rootpow2tab[127];
+
+/*************** init functions ***************/
+
+/* table generator */
+static av_cold void init_pow2table(void)
+{
+    int i;
+    for (i = -63; i < 64; i++) {
+        pow2tab[63 + i] = pow(2, i);
+        rootpow2tab[63 + i] = sqrt(pow(2, i));
+    }
+}
+
+/* table generator */
+static av_cold void init_gain_table(COOKContext *q)
+{
+    int i;
+    q->gain_size_factor = q->samples_per_channel / 8;
+    for (i = 0; i < 23; i++)
+        q->gain_table[i] = pow(pow2tab[i + 52],
+                               (1.0 / (double) q->gain_size_factor));
+}
+
+
+static av_cold int init_cook_vlc_tables(COOKContext *q)
+{
+    int i, result;
+
+    result = 0;
+    for (i = 0; i < 13; i++) {
+        result |= init_vlc(&q->envelope_quant_index[i], 9, 24,
+                           envelope_quant_index_huffbits[i], 1, 1,
+                           envelope_quant_index_huffcodes[i], 2, 2, 0);
+    }
+    av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
+    for (i = 0; i < 7; i++) {
+        result |= init_vlc(&q->sqvh[i], vhvlcsize_tab[i], vhsize_tab[i],
+                           cvh_huffbits[i], 1, 1,
+                           cvh_huffcodes[i], 2, 2, 0);
+    }
+
+    for (i = 0; i < q->num_subpackets; i++) {
+        if (q->subpacket[i].joint_stereo == 1) {
+            result |= init_vlc(&q->subpacket[i].channel_coupling, 6,
+                               (1 << q->subpacket[i].js_vlc_bits) - 1,
+                               ccpl_huffbits[q->subpacket[i].js_vlc_bits - 2], 1, 1,
+                               ccpl_huffcodes[q->subpacket[i].js_vlc_bits - 2], 2, 2, 0);
+            av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
+        }
+    }
+
+    av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
+    return result;
+}
+
+static av_cold int init_cook_mlt(COOKContext *q)
+{
+    int j, ret;
+    int mlt_size = q->samples_per_channel;
+
+    if ((q->mlt_window = av_malloc(mlt_size * sizeof(*q->mlt_window))) == 0)
+        return AVERROR(ENOMEM);
+
+    /* Initialize the MLT window: simple sine window. */
+    ff_sine_window_init(q->mlt_window, mlt_size);
+    for (j = 0; j < mlt_size; j++)
+        q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
+
+    /* Initialize the MDCT. */
+    if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
+        av_free(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_free(q->mlt_window);
+    av_free(q->decoded_bytes_buffer);
+
+    /* Free the transform. */
+    ff_mdct_end(&q->mdct_ctx);
+
+    /* Free the VLC tables. */
+    for (i = 0; i < 13; i++)
+        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;
+
+    while (get_bits1(gb)) {
+        /* NOTHING */
+    }
+
+    n = get_bits_count(gb) - 1;     // amount of elements*2 to update
+
+    i = 0;
+    while (n--) {
+        int index = get_bits(gb, 3);
+        int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
+
+        while (i <= index)
+            gaininfo[i++] = gain;
+    }
+    while (i <= 8)
+        gaininfo[i++] = 0;
+}
+
+/**
+ * Create the quant index table needed for the envelope.
+ *
+ * @param q                 pointer to the COOKContext
+ * @param quant_index_table pointer to the array
+ */
+static int decode_envelope(COOKContext *q, COOKSubpacket *p,
+                           int *quant_index_table)
+{
+    int i, j, vlc_index;
+
+    quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
+
+    for (i = 1; i < p->total_subbands; i++) {
+        vlc_index = i;
+        if (i >= p->js_subband_start * 2) {
+            vlc_index -= p->js_subband_start;
+        } else {
+            vlc_index /= 2;
+            if (vlc_index < 1)
+                vlc_index = 1;
+        }
+        if (vlc_index > 13)
+            vlc_index = 13; // the VLC tables >13 are identical to No. 13
+
+        j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
+                     q->envelope_quant_index[vlc_index - 1].bits, 2);
+        quant_index_table[i] = quant_index_table[i - 1] + j - 12; // differential encoding
+        if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
+            av_log(q->avctx, AV_LOG_ERROR,
+                   "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
+                   quant_index_table[i], i);
+            return AVERROR_INVALIDDATA;
+        }
+    }
+
+    return 0;
+}
+
+/**
+ * Calculate the category and category_index vector.
+ *
+ * @param q                     pointer to the COOKContext
+ * @param quant_index_table     pointer to the array
+ * @param category              pointer to the category array
+ * @param category_index        pointer to the category_index array
+ */
+static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
+                       int *category, int *category_index)
+{
+    int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
+    int exp_index2[102] = { 0 };
+    int exp_index1[102] = { 0 };
+
+    int tmp_categorize_array[128 * 2] = { 0 };
+    int tmp_categorize_array1_idx = p->numvector_size;
+    int tmp_categorize_array2_idx = p->numvector_size;
+
+    bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
+
+    if (bits_left > q->samples_per_channel)
+        bits_left = q->samples_per_channel +
+                    ((bits_left - q->samples_per_channel) * 5) / 8;
+
+    bias = -32;
+
+    /* Estimate bias. */
+    for (i = 32; i > 0; i = i / 2) {
+        num_bits = 0;
+        index    = 0;
+        for (j = p->total_subbands; j > 0; j--) {
+            exp_idx = av_clip((i - quant_index_table[index] + bias) / 2, 0, 7);
+            index++;
+            num_bits += expbits_tab[exp_idx];
+        }
+        if (num_bits >= bits_left - 32)
+            bias += i;
+    }
+
+    /* Calculate total number of bits. */
+    num_bits = 0;
+    for (i = 0; i < p->total_subbands; i++) {
+        exp_idx = av_clip((bias - quant_index_table[i]) / 2, 0, 7);
+        num_bits += expbits_tab[exp_idx];
+        exp_index1[i] = exp_idx;
+        exp_index2[i] = exp_idx;
+    }
+    tmpbias1 = tmpbias2 = num_bits;
+
+    for (j = 1; j < p->numvector_size; j++) {
+        if (tmpbias1 + tmpbias2 > 2 * bits_left) {  /* ---> */
+            int max = -999999;
+            index = -1;
+            for (i = 0; i < p->total_subbands; i++) {
+                if (exp_index1[i] < 7) {
+                    v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
+                    if (v >= max) {
+                        max   = v;
+                        index = i;
+                    }
+                }
+            }
+            if (index == -1)
+                break;
+            tmp_categorize_array[tmp_categorize_array1_idx++] = index;
+            tmpbias1 -= expbits_tab[exp_index1[index]] -
+                        expbits_tab[exp_index1[index] + 1];
+            ++exp_index1[index];
+        } else {  /* <--- */
+            int min = 999999;
+            index = -1;
+            for (i = 0; i < p->total_subbands; i++) {
+                if (exp_index2[i] > 0) {
+                    v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
+                    if (v < min) {
+                        min   = v;
+                        index = i;
+                    }
+                }
+            }
+            if (index == -1)
+                break;
+            tmp_categorize_array[--tmp_categorize_array2_idx] = index;
+            tmpbias2 -= expbits_tab[exp_index2[index]] -
+                        expbits_tab[exp_index2[index] - 1];
+            --exp_index2[index];
+        }
+    }
+
+    for (i = 0; i < p->total_subbands; i++)
+        category[i] = exp_index2[i];
+
+    for (i = 0; i < p->numvector_size - 1; i++)
+        category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
+}
+
+
+/**
+ * Expand the category vector.
+ *
+ * @param q                     pointer to the COOKContext
+ * @param category              pointer to the category array
+ * @param category_index        pointer to the category_index array
+ */
+static inline void expand_category(COOKContext *q, int *category,
+                                   int *category_index)
+{
+    int i;
+    for (i = 0; i < q->num_vectors; i++)
+    {
+        int idx = category_index[i];
+        if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
+            --category[idx];
+    }
+}
+
+/**
+ * The real requantization of the mltcoefs
+ *
+ * @param q                     pointer to the COOKContext
+ * @param index                 index
+ * @param quant_index           quantisation index
+ * @param subband_coef_index    array of indexes to quant_centroid_tab
+ * @param subband_coef_sign     signs of coefficients
+ * @param mlt_p                 pointer into the mlt buffer
+ */
+static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
+                                 int *subband_coef_index, int *subband_coef_sign,
+                                 float *mlt_p)
+{
+    int i;
+    float f1;
+
+    for (i = 0; i < SUBBAND_SIZE; i++) {
+        if (subband_coef_index[i]) {
+            f1 = quant_centroid_tab[index][subband_coef_index[i]];
+            if (subband_coef_sign[i])
+                f1 = -f1;
+        } else {
+            /* noise coding if subband_coef_index[i] == 0 */
+            f1 = dither_tab[index];
+            if (av_lfg_get(&q->random_state) < 0x80000000)
+                f1 = -f1;
+        }
+        mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
+    }
+}
+/**
+ * Unpack the subband_coef_index and subband_coef_sign vectors.
+ *
+ * @param q                     pointer to the COOKContext
+ * @param category              pointer to the category array
+ * @param subband_coef_index    array of indexes to quant_centroid_tab
+ * @param subband_coef_sign     signs of coefficients
+ */
+static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
+                       int *subband_coef_index, int *subband_coef_sign)
+{
+    int i, j;
+    int vlc, vd, tmp, result;
+
+    vd = vd_tab[category];
+    result = 0;
+    for (i = 0; i < vpr_tab[category]; i++) {
+        vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
+        if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
+            vlc = 0;
+            result = 1;
+        }
+        for (j = vd - 1; j >= 0; j--) {
+            tmp = (vlc * invradix_tab[category]) / 0x100000;
+            subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
+            vlc = tmp;
+        }
+        for (j = 0; j < vd; j++) {
+            if (subband_coef_index[i * vd + j]) {
+                if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
+                    subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
+                } else {
+                    result = 1;
+                    subband_coef_sign[i * vd + j] = 0;
+                }
+            } else {
+                subband_coef_sign[i * vd + j] = 0;
+            }
+        }
+    }
+    return result;
+}
+
+
+/**
+ * Fill the mlt_buffer with mlt coefficients.
+ *
+ * @param q                 pointer to the COOKContext
+ * @param category          pointer to the category array
+ * @param quant_index_table pointer to the array
+ * @param mlt_buffer        pointer to mlt coefficients
+ */
+static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
+                           int *quant_index_table, float *mlt_buffer)
+{
+    /* A zero in this table means that the subband coefficient is
+       random noise coded. */
+    int subband_coef_index[SUBBAND_SIZE];
+    /* A zero in this table means that the subband coefficient is a
+       positive multiplicator. */
+    int subband_coef_sign[SUBBAND_SIZE];
+    int band, j;
+    int index = 0;
+
+    for (band = 0; band < p->total_subbands; band++) {
+        index = category[band];
+        if (category[band] < 7) {
+            if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
+                index = 7;
+                for (j = 0; j < p->total_subbands; j++)
+                    category[band + j] = 7;
+            }
+        }
+        if (index >= 7) {
+            memset(subband_coef_index, 0, sizeof(subband_coef_index));
+            memset(subband_coef_sign,  0, sizeof(subband_coef_sign));
+        }
+        q->scalar_dequant(q, index, quant_index_table[band],
+                          subband_coef_index, subband_coef_sign,
+                          &mlt_buffer[band * SUBBAND_SIZE]);
+    }
+
+    /* FIXME: should this be removed, or moved into loop above? */
+    if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
+        return;
+}
+
+
+static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
+{
+    int category_index[128] = { 0 };
+    int category[128]       = { 0 };
+    int quant_index_table[102];
+    int res, i;
+
+    if ((res = decode_envelope(q, p, quant_index_table)) < 0)
+        return res;
+    q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);
+    categorize(q, p, quant_index_table, category, category_index);
+    expand_category(q, category, category_index);
+    for (i=0; i<p->total_subbands; i++) {
+        if (category[i] > 7)
+            return AVERROR_INVALIDDATA;
+    }
+    decode_vectors(q, p, category, quant_index_table, mlt_buffer);
+
+    return 0;
+}
+
+
+/**
+ * the actual requantization of the timedomain samples
+ *
+ * @param q                 pointer to the COOKContext
+ * @param buffer            pointer to the timedomain buffer
+ * @param gain_index        index for the block multiplier
+ * @param gain_index_next   index for the next block multiplier
+ */
+static void interpolate_float(COOKContext *q, float *buffer,
+                              int gain_index, int gain_index_next)
+{
+    int i;
+    float fc1, fc2;
+    fc1 = pow2tab[gain_index + 63];
+
+    if (gain_index == gain_index_next) {             // static gain
+        for (i = 0; i < q->gain_size_factor; i++)
+            buffer[i] *= fc1;
+    } else {                                        // smooth gain
+        fc2 = q->gain_table[11 + (gain_index_next - gain_index)];
+        for (i = 0; i < q->gain_size_factor; i++) {
+            buffer[i] *= fc1;
+            fc1       *= fc2;
+        }
+    }
+}
+
+/**
+ * Apply transform window, overlap buffers.
+ *
+ * @param q                 pointer to the COOKContext
+ * @param inbuffer          pointer to the mltcoefficients
+ * @param gains_ptr         current and previous gains
+ * @param previous_buffer   pointer to the previous buffer to be used for overlapping
+ */
+static void imlt_window_float(COOKContext *q, float *inbuffer,
+                              cook_gains *gains_ptr, float *previous_buffer)
+{
+    const float fc = pow2tab[gains_ptr->previous[0] + 63];
+    int i;
+    /* The weird thing here, is that the two halves of the time domain
+     * buffer are swapped. Also, the newest data, that we save away for
+     * next frame, has the wrong sign. Hence the subtraction below.
+     * Almost sounds like a complex conjugate/reverse data/FFT effect.
+     */
+
+    /* Apply window and overlap */
+    for (i = 0; i < q->samples_per_channel; i++)
+        inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
+                      previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
+}
+
+/**
+ * The modulated lapped transform, this takes transform coefficients
+ * and transforms them into timedomain samples.
+ * Apply transform window, overlap buffers, apply gain profile
+ * and buffer management.
+ *
+ * @param q                 pointer to the COOKContext
+ * @param inbuffer          pointer to the mltcoefficients
+ * @param gains_ptr         current and previous gains
+ * @param previous_buffer   pointer to the previous buffer to be used for overlapping
+ */
+static void imlt_gain(COOKContext *q, float *inbuffer,
+                      cook_gains *gains_ptr, float *previous_buffer)
+{
+    float *buffer0 = q->mono_mdct_output;
+    float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
+    int i;
+
+    /* Inverse modified discrete cosine transform */
+    q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
+
+    q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
+
+    /* Apply gain profile */
+    for (i = 0; i < 8; i++)
+        if (gains_ptr->now[i] || gains_ptr->now[i + 1])
+            q->interpolate(q, &buffer1[q->gain_size_factor * i],
+                           gains_ptr->now[i], gains_ptr->now[i + 1]);
+
+    /* Save away the current to be previous block. */
+    memcpy(previous_buffer, buffer0,
+           q->samples_per_channel * sizeof(*previous_buffer));
+}
+
+
+/**
+ * function for getting the jointstereo coupling information
+ *
+ * @param q                 pointer to the COOKContext
+ * @param decouple_tab      decoupling array
+ */
+static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
+{
+    int i;
+    int vlc    = get_bits1(&q->gb);
+    int start  = cplband[p->js_subband_start];
+    int end    = cplband[p->subbands - 1];
+    int length = end - start + 1;
+
+    if (start > end)
+        return 0;
+
+    if (vlc)
+        for (i = 0; i < length; i++)
+            decouple_tab[start + i] = get_vlc2(&q->gb,
+                                               p->channel_coupling.table,
+                                               p->channel_coupling.bits, 2);
+    else
+        for (i = 0; i < length; i++) {
+            int v = get_bits(&q->gb, p->js_vlc_bits);
+            if (v == (1<<p->js_vlc_bits)-1) {
+                av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
+                return AVERROR_INVALIDDATA;
+            }
+            decouple_tab[start + i] = v;
+        }
+    return 0;
+}
+
+/**
+ * function decouples a pair of signals from a single signal via multiplication.
+ *
+ * @param q                 pointer to the COOKContext
+ * @param subband           index of the current subband
+ * @param f1                multiplier for channel 1 extraction
+ * @param f2                multiplier for channel 2 extraction
+ * @param decode_buffer     input buffer
+ * @param mlt_buffer1       pointer to left channel mlt coefficients
+ * @param mlt_buffer2       pointer to right channel mlt coefficients
+ */
+static void decouple_float(COOKContext *q,
+                           COOKSubpacket *p,
+                           int subband,
+                           float f1, float f2,
+                           float *decode_buffer,
+                           float *mlt_buffer1, float *mlt_buffer2)
+{
+    int j, tmp_idx;
+    for (j = 0; j < SUBBAND_SIZE; j++) {
+        tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
+        mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
+        mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
+    }
+}
+
+/**
+ * function for decoding joint stereo data
+ *
+ * @param q                 pointer to the COOKContext
+ * @param mlt_buffer1       pointer to left channel mlt coefficients
+ * @param mlt_buffer2       pointer to right channel mlt coefficients
+ */
+static int joint_decode(COOKContext *q, COOKSubpacket *p,
+                        float *mlt_buffer_left, float *mlt_buffer_right)
+{
+    int i, j, res;
+    int decouple_tab[SUBBAND_SIZE] = { 0 };
+    float *decode_buffer = q->decode_buffer_0;
+    int idx, cpl_tmp;
+    float f1, f2;
+    const float *cplscale;
+
+    memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
+
+    /* Make sure the buffers are zeroed out. */
+    memset(mlt_buffer_left,  0, 1024 * sizeof(*mlt_buffer_left));
+    memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
+    if ((res = decouple_info(q, p, decouple_tab)) < 0)
+        return res;
+    if ((res = mono_decode(q, p, decode_buffer)) < 0)
+        return res;
+    /* The two channels are stored interleaved in decode_buffer. */
+    for (i = 0; i < p->js_subband_start; i++) {
+        for (j = 0; j < SUBBAND_SIZE; j++) {
+            mlt_buffer_left[i  * 20 + j] = decode_buffer[i * 40 + j];
+            mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
+        }
+    }
+
+    /* When we reach js_subband_start (the higher frequencies)
+       the coefficients are stored in a coupling scheme. */
+    idx = (1 << p->js_vlc_bits) - 1;
+    for (i = p->js_subband_start; i < p->subbands; i++) {
+        cpl_tmp = cplband[i];
+        idx -= decouple_tab[cpl_tmp];
+        cplscale = q->cplscales[p->js_vlc_bits - 2];  // choose decoupler table
+        f1 = cplscale[decouple_tab[cpl_tmp] + 1];
+        f2 = cplscale[idx];
+        q->decouple(q, p, i, f1, f2, decode_buffer,
+                    mlt_buffer_left, mlt_buffer_right);
+        idx = (1 << p->js_vlc_bits) - 1;
+    }
+
+    return 0;
+}
+
+/**
+ * First part of subpacket decoding:
+ *  decode raw stream bytes and read gain info.
+ *
+ * @param q                 pointer to the COOKContext
+ * @param inbuffer          pointer to raw stream data
+ * @param gains_ptr         array of current/prev gain pointers
+ */
+static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
+                                         const uint8_t *inbuffer,
+                                         cook_gains *gains_ptr)
+{
+    int offset;
+
+    offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
+                          p->bits_per_subpacket / 8);
+    init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
+                  p->bits_per_subpacket);
+    decode_gain_info(&q->gb, gains_ptr->now);
+
+    /* Swap current and previous gains */
+    FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
+}
+
+/**
+ * Saturate the output signal and interleave.
+ *
+ * @param q                 pointer to the COOKContext
+ * @param out               pointer to the output vector
+ */
+static void saturate_output_float(COOKContext *q, float *out)
+{
+    q->dsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
+                        -1.0f, 1.0f, FFALIGN(q->samples_per_channel, 8));
+}
+
+
+/**
+ * Final part of subpacket decoding:
+ *  Apply modulated lapped transform, gain compensation,
+ *  clip and convert to integer.
+ *
+ * @param q                 pointer to the COOKContext
+ * @param decode_buffer     pointer to the mlt coefficients
+ * @param gains_ptr         array of current/prev gain pointers
+ * @param previous_buffer   pointer to the previous buffer to be used for overlapping
+ * @param out               pointer to the output buffer
+ */
+static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
+                                         cook_gains *gains_ptr, float *previous_buffer,
+                                         float *out)
+{
+    imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
+    if (out)
+        q->saturate_output(q, out);
+}
+
+
+/**
+ * Cook subpacket decoding. This function returns one decoded subpacket,
+ * usually 1024 samples per channel.
+ *
+ * @param q                 pointer to the COOKContext
+ * @param inbuffer          pointer to the inbuffer
+ * @param outbuffer         pointer to the outbuffer
+ */
+static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
+                            const uint8_t *inbuffer, float **outbuffer)
+{
+    int sub_packet_size = p->size;
+    int res;
+
+    memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
+    decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
+
+    if (p->joint_stereo) {
+        if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
+            return res;
+    } else {
+        if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
+            return res;
+
+        if (p->num_channels == 2) {
+            decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
+            if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
+                return res;
+        }
+    }
+
+    mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
+                          p->mono_previous_buffer1,
+                          outbuffer ? outbuffer[p->ch_idx] : NULL);
+
+    if (p->num_channels == 2) {
+        if (p->joint_stereo)
+            mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
+                                  p->mono_previous_buffer2,
+                                  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
+        else
+            mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
+                                  p->mono_previous_buffer2,
+                                  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
+    }
+
+    return 0;
+}
+
+
+static int cook_decode_frame(AVCodecContext *avctx, void *data,
+                             int *got_frame_ptr, AVPacket *avpkt)
+{
+    AVFrame *frame     = data;
+    const uint8_t *buf = avpkt->data;
+    int buf_size = avpkt->size;
+    COOKContext *q = avctx->priv_data;
+    float **samples = NULL;
+    int i, ret;
+    int offset = 0;
+    int chidx = 0;
+
+    if (buf_size < avctx->block_align)
+        return buf_size;
+
+    /* get output buffer */
+    if (q->discarded_packets >= 2) {
+        frame->nb_samples = q->samples_per_channel;
+        if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
+            return ret;
+        samples = (float **)frame->extended_data;
+    }
+
+    /* estimate subpacket sizes */
+    q->subpacket[0].size = avctx->block_align;
+
+    for (i = 1; i < q->num_subpackets; i++) {
+        q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
+        q->subpacket[0].size -= q->subpacket[i].size + 1;
+        if (q->subpacket[0].size < 0) {
+            av_log(avctx, AV_LOG_DEBUG,
+                   "frame subpacket size total > avctx->block_align!\n");
+            return AVERROR_INVALIDDATA;
+        }
+    }
+
+    /* decode supbackets */
+    for (i = 0; i < q->num_subpackets; i++) {
+        q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
+                                              q->subpacket[i].bits_per_subpdiv;
+        q->subpacket[i].ch_idx = chidx;
+        av_log(avctx, AV_LOG_DEBUG,
+               "subpacket[%i] size %i js %i %i block_align %i\n",
+               i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
+               avctx->block_align);
+
+        if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
+            return ret;
+        offset += q->subpacket[i].size;
+        chidx += q->subpacket[i].num_channels;
+        av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
+               i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
+    }
+
+    /* Discard the first two frames: no valid audio. */
+    if (q->discarded_packets < 2) {
+        q->discarded_packets++;
+        *got_frame_ptr = 0;
+        return avctx->block_align;
+    }
+
+    *got_frame_ptr = 1;
+
+    return avctx->block_align;
+}
+
+#ifdef DEBUG
+static void dump_cook_context(COOKContext *q)
+{
+    //int i=0;
+#define PRINT(a, b) av_dlog(q->avctx, " %s = %d\n", a, b);
+    av_dlog(q->avctx, "COOKextradata\n");
+    av_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
+    if (q->subpacket[0].cookversion > STEREO) {
+        PRINT("js_subband_start", q->subpacket[0].js_subband_start);
+        PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
+    }
+    av_dlog(q->avctx, "COOKContext\n");
+    PRINT("nb_channels", q->avctx->channels);
+    PRINT("bit_rate", q->avctx->bit_rate);
+    PRINT("sample_rate", q->avctx->sample_rate);
+    PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
+    PRINT("subbands", q->subpacket[0].subbands);
+    PRINT("js_subband_start", q->subpacket[0].js_subband_start);
+    PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
+    PRINT("numvector_size", q->subpacket[0].numvector_size);
+    PRINT("total_subbands", q->subpacket[0].total_subbands);
+}
+#endif
+
+/**
+ * Cook initialization
+ *
+ * @param avctx     pointer to the AVCodecContext
+ */
+static av_cold int cook_decode_init(AVCodecContext *avctx)
+{
+    COOKContext *q = avctx->priv_data;
+    const uint8_t *edata_ptr = avctx->extradata;
+    const uint8_t *edata_ptr_end = edata_ptr + avctx->extradata_size;
+    int extradata_size = avctx->extradata_size;
+    int s = 0;
+    unsigned int channel_mask = 0;
+    int samples_per_frame = 0;
+    int ret;
+    q->avctx = avctx;
+
+    /* Take care of the codec specific extradata. */
+    if (extradata_size <= 0) {
+        av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
+        return AVERROR_INVALIDDATA;
+    }
+    av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
+
+    /* Take data from the AVCodecContext (RM container). */
+    if (!avctx->channels) {
+        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
+        return AVERROR_INVALIDDATA;
+    }
+
+    /* Initialize RNG. */
+    av_lfg_init(&q->random_state, 0);
+
+    ff_dsputil_init(&q->dsp, avctx);
+
+    while (edata_ptr < edata_ptr_end) {
+        /* 8 for mono, 16 for stereo, ? for multichannel
+           Swap to right endianness so we don't need to care later on. */
+        if (extradata_size >= 8) {
+            q->subpacket[s].cookversion = bytestream_get_be32(&edata_ptr);
+            samples_per_frame           = bytestream_get_be16(&edata_ptr);
+            q->subpacket[s].subbands = bytestream_get_be16(&edata_ptr);
+            extradata_size -= 8;
+        }
+        if (extradata_size >= 8) {
+            bytestream_get_be32(&edata_ptr);    // Unknown unused
+            q->subpacket[s].js_subband_start = bytestream_get_be16(&edata_ptr);
+            if (q->subpacket[s].js_subband_start >= 51) {
+                av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
+                return AVERROR_INVALIDDATA;
+            }
+
+            q->subpacket[s].js_vlc_bits = bytestream_get_be16(&edata_ptr);
+            extradata_size -= 8;
+        }
+
+        /* Initialize extradata related variables. */
+        q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
+        q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
+
+        /* Initialize default data states. */
+        q->subpacket[s].log2_numvector_size = 5;
+        q->subpacket[s].total_subbands = q->subpacket[s].subbands;
+        q->subpacket[s].num_channels = 1;
+
+        /* Initialize version-dependent variables */
+
+        av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
+               q->subpacket[s].cookversion);
+        q->subpacket[s].joint_stereo = 0;
+        switch (q->subpacket[s].cookversion) {
+        case MONO:
+            if (avctx->channels != 1) {
+                avpriv_request_sample(avctx, "Container channels != 1");
+                return AVERROR_PATCHWELCOME;
+            }
+            av_log(avctx, AV_LOG_DEBUG, "MONO\n");
+            break;
+        case STEREO:
+            if (avctx->channels != 1) {
+                q->subpacket[s].bits_per_subpdiv = 1;
+                q->subpacket[s].num_channels = 2;
+            }
+            av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
+            break;
+        case JOINT_STEREO:
+            if (avctx->channels != 2) {
+                avpriv_request_sample(avctx, "Container channels != 2");
+                return AVERROR_PATCHWELCOME;
+            }
+            av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
+            if (avctx->extradata_size >= 16) {
+                q->subpacket[s].total_subbands = q->subpacket[s].subbands +
+                                                 q->subpacket[s].js_subband_start;
+                q->subpacket[s].joint_stereo = 1;
+                q->subpacket[s].num_channels = 2;
+            }
+            if (q->subpacket[s].samples_per_channel > 256) {
+                q->subpacket[s].log2_numvector_size = 6;
+            }
+            if (q->subpacket[s].samples_per_channel > 512) {
+                q->subpacket[s].log2_numvector_size = 7;
+            }
+            break;
+        case MC_COOK:
+            av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
+            if (extradata_size >= 4)
+                channel_mask |= q->subpacket[s].channel_mask = bytestream_get_be32(&edata_ptr);
+
+            if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {
+                q->subpacket[s].total_subbands = q->subpacket[s].subbands +
+                                                 q->subpacket[s].js_subband_start;
+                q->subpacket[s].joint_stereo = 1;
+                q->subpacket[s].num_channels = 2;
+                q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
+
+                if (q->subpacket[s].samples_per_channel > 256) {
+                    q->subpacket[s].log2_numvector_size = 6;
+                }
+                if (q->subpacket[s].samples_per_channel > 512) {
+                    q->subpacket[s].log2_numvector_size = 7;
+                }
+            } else
+                q->subpacket[s].samples_per_channel = samples_per_frame;
+
+            break;
+        default:
+            avpriv_request_sample(avctx, "Cook version %d",
+                                  q->subpacket[s].cookversion);
+            return AVERROR_PATCHWELCOME;
+        }
+
+        if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
+            av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
+            return AVERROR_INVALIDDATA;
+        } else
+            q->samples_per_channel = q->subpacket[0].samples_per_channel;
+
+
+        /* Initialize variable relations */
+        q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
+
+        /* Try to catch some obviously faulty streams, othervise it might be exploitable */
+        if (q->subpacket[s].total_subbands > 53) {
+            avpriv_request_sample(avctx, "total_subbands > 53");
+            return AVERROR_PATCHWELCOME;
+        }
+
+        if ((q->subpacket[s].js_vlc_bits > 6) ||
+            (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
+            av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
+                   q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
+            return AVERROR_INVALIDDATA;
+        }
+
+        if (q->subpacket[s].subbands > 50) {
+            avpriv_request_sample(avctx, "subbands > 50");
+            return AVERROR_PATCHWELCOME;
+        }
+        if (q->subpacket[s].subbands == 0) {
+            avpriv_request_sample(avctx, "subbands = 0");
+            return AVERROR_PATCHWELCOME;
+        }
+        q->subpacket[s].gains1.now      = q->subpacket[s].gain_1;
+        q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
+        q->subpacket[s].gains2.now      = q->subpacket[s].gain_3;
+        q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
+
+        if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
+            av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
+            return AVERROR_INVALIDDATA;
+        }
+
+        q->num_subpackets++;
+        s++;
+        if (s > MAX_SUBPACKETS) {
+            avpriv_request_sample(avctx, "subpackets > %d", MAX_SUBPACKETS);
+            return AVERROR_PATCHWELCOME;
+        }
+    }
+    /* Generate tables */
+    init_pow2table();
+    init_gain_table(q);
+    init_cplscales_table(q);
+
+    if ((ret = init_cook_vlc_tables(q)))
+        return ret;
+
+
+    if (avctx->block_align >= UINT_MAX / 2)
+        return AVERROR(EINVAL);
+
+    /* Pad the databuffer with:
+       DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
+       FF_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
+    q->decoded_bytes_buffer =
+        av_mallocz(avctx->block_align
+                   + DECODE_BYTES_PAD1(avctx->block_align)
+                   + FF_INPUT_BUFFER_PADDING_SIZE);
+    if (q->decoded_bytes_buffer == NULL)
+        return AVERROR(ENOMEM);
+
+    /* Initialize transform. */
+    if ((ret = init_cook_mlt(q)))
+        return ret;
+
+    /* Initialize COOK signal arithmetic handling */
+    if (1) {
+        q->scalar_dequant  = scalar_dequant_float;
+        q->decouple        = decouple_float;
+        q->imlt_window     = imlt_window_float;
+        q->interpolate     = interpolate_float;
+        q->saturate_output = saturate_output_float;
+    }
+
+    /* Try to catch some obviously faulty streams, othervise it might be exploitable */
+    if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
+        q->samples_per_channel != 1024) {
+        avpriv_request_sample(avctx, "samples_per_channel = %d",
+                              q->samples_per_channel);
+        return AVERROR_PATCHWELCOME;
+    }
+
+    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
+    if (channel_mask)
+        avctx->channel_layout = channel_mask;
+    else
+        avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
+
+#ifdef DEBUG
+    dump_cook_context(q);
+#endif
+    return 0;
+}
+
+AVCodec ff_cook_decoder = {
+    .name           = "cook",
+    .type           = AVMEDIA_TYPE_AUDIO,
+    .id             = AV_CODEC_ID_COOK,
+    .priv_data_size = sizeof(COOKContext),
+    .init           = cook_decode_init,
+    .close          = cook_decode_close,
+    .decode         = cook_decode_frame,
+    .capabilities   = CODEC_CAP_DR1,
+    .long_name      = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
+    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
+                                                      AV_SAMPLE_FMT_NONE },
+};