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/*

 * ATRAC3 compatible decoder

 * Copyright (c) 2006-2008 Maxim Poliakovski

 * Copyright (c) 2006-2008 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

 * ATRAC3 compatible decoder.

 * This decoder handles Sony's ATRAC3 data.

 *

 * Container formats used to store ATRAC3 data:

 * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).

 *

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

 * bytes provided in the containers above.

 */

#include 

#include 

#include 

#include "libavutil/attributes.h"

#include "libavutil/float_dsp.h"

#include "libavutil/libm.h"

#include "avcodec.h"

#include "bytestream.h"

#include "fft.h"

#include "fmtconvert.h"

#include "get_bits.h"

#include "internal.h"

#include "atrac.h"

#include "atrac3data.h"

#define JOINT_STEREO    0x12

#define STEREO          0x2

#define SAMPLES_PER_FRAME 1024

#define MDCT_SIZE          512

typedef struct GainBlock {

    AtracGainInfo g_block[4];

} GainBlock;

typedef struct TonalComponent {

    int pos;

    int num_coefs;

    float coef[8];

} TonalComponent;

typedef struct ChannelUnit {

    int            bands_coded;

    int            num_components;

    float          prev_frame[SAMPLES_PER_FRAME];

    int            gc_blk_switch;

    TonalComponent components[64];

    GainBlock      gain_block[2];

    DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];

    DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];

    float          delay_buf1[46]; ///

    float          delay_buf2[46];

    float          delay_buf3[46];

} ChannelUnit;

typedef struct ATRAC3Context {

    GetBitContext gb;

    //@{

    /** stream data */

    int coding_mode;

    ChannelUnit *units;

    //@}

    //@{

    /** joint-stereo related variables */

    int matrix_coeff_index_prev[4];

    int matrix_coeff_index_now[4];

    int matrix_coeff_index_next[4];

    int weighting_delay[6];

    //@}

    //@{

    /** data buffers */

    uint8_t *decoded_bytes_buffer;

    float temp_buf[1070];

    //@}

    //@{

    /** extradata */

    int scrambled_stream;

    //@}

    AtracGCContext    gainc_ctx;

    FFTContext        mdct_ctx;

    FmtConvertContext fmt_conv;

    AVFloatDSPContext fdsp;

} ATRAC3Context;

static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];

static VLC_TYPE atrac3_vlc_table[4096][2];

static VLC   spectral_coeff_tab[7];

/**

 * Regular 512 points IMDCT without overlapping, with the exception of the

 * swapping of odd bands caused by the reverse spectra of the QMF.

 *

 * @param odd_band  1 if the band is an odd band

 */

static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)

{

    int i;

    if (odd_band) {

        /**

         * Reverse the odd bands before IMDCT, this is an effect of the QMF

         * transform or it gives better compression to do it this way.

         * FIXME: It should be possible to handle this in imdct_calc

         * for that to happen a modification of the prerotation step of

         * all SIMD code and C code is needed.

         * Or fix the functions before so they generate a pre reversed spectrum.

         */

        for (i = 0; i < 128; i++)

            FFSWAP(float, input[i], input[255 - i]);

    }

    q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);

    /* Perform windowing on the output. */

    q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE);

}

/*

 * indata descrambling, only used for data coming from the rm container

 */

static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)

{

    int i, off;

    uint32_t c;

    const uint32_t *buf;

    uint32_t *output = (uint32_t *)out;

    off = (intptr_t)input & 3;

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

    if (off)

        c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));

    else

        c = av_be2ne32(0x537F6103U);

    bytes += 3 + off;

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

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

    if (off)

        avpriv_request_sample(NULL, "Offset of %d", off);

    return off;

}

static av_cold void init_imdct_window(void)

{

    int i, j;

    /* generate the mdct window, for details see

     * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */

    for (i = 0, j = 255; i < 128; i++, j--) {

        float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;

        float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;

        float w  = 0.5 * (wi * wi + wj * wj);

        mdct_window[i] = mdct_window[511 - i] = wi / w;

        mdct_window[j] = mdct_window[511 - j] = wj / w;

    }

}

static av_cold int atrac3_decode_close(AVCodecContext *avctx)

{

    ATRAC3Context *q = avctx->priv_data;

    av_free(q->units);

    av_free(q->decoded_bytes_buffer);

    ff_mdct_end(&q->mdct_ctx);

    return 0;

}

/**

 * Mantissa decoding

 *

 * @param selector     which table the output values are coded with

 * @param coding_flag  constant length coding or variable length coding

 * @param mantissas    mantissa output table

 * @param num_codes    number of values to get

 */

static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,

                                       int coding_flag, int *mantissas,

                                       int num_codes)

{

    int i, code, huff_symb;

    if (selector == 1)

        num_codes /= 2;

    if (coding_flag != 0) {

        /* constant length coding (CLC) */

        int num_bits = clc_length_tab[selector];

        if (selector > 1) {

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

                if (num_bits)

                    code = get_sbits(gb, num_bits);

                else

                    code = 0;

                mantissas[i] = code;

            }

        } else {

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

                if (num_bits)

                    code = get_bits(gb, num_bits); // num_bits is always 4 in this case

                else

                    code = 0;

                mantissas[i * 2    ] = mantissa_clc_tab[code >> 2];

                mantissas[i * 2 + 1] = mantissa_clc_tab[code &  3];

            }

        }

    } else {

        /* variable length coding (VLC) */

        if (selector != 1) {

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

                huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,

                                     spectral_coeff_tab[selector-1].bits, 3);

                huff_symb += 1;

                code = huff_symb >> 1;

                if (huff_symb & 1)

                    code = -code;

                mantissas[i] = code;

            }

        } else {

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

                huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,

                                     spectral_coeff_tab[selector - 1].bits, 3);

                mantissas[i * 2    ] = mantissa_vlc_tab[huff_symb * 2    ];

                mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];

            }

        }

    }

}

/**

 * Restore the quantized band spectrum coefficients

 *

 * @return subband count, fix for broken specification/files

 */

static int decode_spectrum(GetBitContext *gb, float *output)

{

    int num_subbands, coding_mode, i, j, first, last, subband_size;

    int subband_vlc_index[32], sf_index[32];

    int mantissas[128];

    float scale_factor;

    num_subbands = get_bits(gb, 5);  // number of coded subbands

    coding_mode  = get_bits1(gb);    // coding Mode: 0 - VLC/ 1-CLC

    /* get the VLC selector table for the subbands, 0 means not coded */

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

        subband_vlc_index[i] = get_bits(gb, 3);

    /* read the scale factor indexes from the stream */

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

        if (subband_vlc_index[i] != 0)

            sf_index[i] = get_bits(gb, 6);

    }

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

        first = subband_tab[i    ];

        last  = subband_tab[i + 1];

        subband_size = last - first;

        if (subband_vlc_index[i] != 0) {

            /* decode spectral coefficients for this subband */

            /* TODO: This can be done faster is several blocks share the

             * same VLC selector (subband_vlc_index) */

            read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,

                                       mantissas, subband_size);

            /* decode the scale factor for this subband */

            scale_factor = ff_atrac_sf_table[sf_index[i]] *

                           inv_max_quant[subband_vlc_index[i]];

            /* inverse quantize the coefficients */

            for (j = 0; first < last; first++, j++)

                output[first] = mantissas[j] * scale_factor;

        } else {

            /* this subband was not coded, so zero the entire subband */

            memset(output + first, 0, subband_size * sizeof(*output));

        }

    }

    /* clear the subbands that were not coded */

    first = subband_tab[i];

    memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));

    return num_subbands;

}

/**

 * Restore the quantized tonal components

 *

 * @param components tonal components

 * @param num_bands  number of coded bands

 */

static int decode_tonal_components(GetBitContext *gb,

                                   TonalComponent *components, int num_bands)

{

    int i, b, c, m;

    int nb_components, coding_mode_selector, coding_mode;

    int band_flags[4], mantissa[8];

    int component_count = 0;

    nb_components = get_bits(gb, 5);

    /* no tonal components */

    if (nb_components == 0)

        return 0;

    coding_mode_selector = get_bits(gb, 2);

    if (coding_mode_selector == 2)

        return AVERROR_INVALIDDATA;

    coding_mode = coding_mode_selector & 1;

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

        int coded_values_per_component, quant_step_index;

        for (b = 0; b <= num_bands; b++)

            band_flags[b] = get_bits1(gb);

        coded_values_per_component = get_bits(gb, 3);

        quant_step_index = get_bits(gb, 3);

        if (quant_step_index <= 1)

            return AVERROR_INVALIDDATA;

        if (coding_mode_selector == 3)

            coding_mode = get_bits1(gb);

        for (b = 0; b < (num_bands + 1) * 4; b++) {

            int coded_components;

            if (band_flags[b >> 2] == 0)

                continue;

            coded_components = get_bits(gb, 3);

            for (c = 0; c < coded_components; c++) {

                TonalComponent *cmp = &components[component_count];

                int sf_index, coded_values, max_coded_values;

                float scale_factor;

                sf_index = get_bits(gb, 6);

                if (component_count >= 64)

                    return AVERROR_INVALIDDATA;

                cmp->pos = b * 64 + get_bits(gb, 6);

                max_coded_values = SAMPLES_PER_FRAME - cmp->pos;

                coded_values     = coded_values_per_component + 1;

                coded_values     = FFMIN(max_coded_values, coded_values);

                scale_factor = ff_atrac_sf_table[sf_index] *

                               inv_max_quant[quant_step_index];

                read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,

                                           mantissa, coded_values);

                cmp->num_coefs = coded_values;

                /* inverse quant */

                for (m = 0; m < coded_values; m++)

                    cmp->coef[m] = mantissa[m] * scale_factor;

                component_count++;

            }

        }

    }

    return component_count;

}

/**

 * Decode gain parameters for the coded bands

 *

 * @param block      the gainblock for the current band

 * @param num_bands  amount of coded bands

 */

static int decode_gain_control(GetBitContext *gb, GainBlock *block,

                               int num_bands)

{

    int b, j;

    int *level, *loc;

    AtracGainInfo *gain = block->g_block;

    for (b = 0; b <= num_bands; b++) {

        gain[b].num_points = get_bits(gb, 3);

        level              = gain[b].lev_code;

        loc                = gain[b].loc_code;

        for (j = 0; j < gain[b].num_points; j++) {

            level[j] = get_bits(gb, 4);

            loc[j]   = get_bits(gb, 5);

            if (j && loc[j] <= loc[j - 1])

                return AVERROR_INVALIDDATA;

        }

    }

    /* Clear the unused blocks. */

    for (; b < 4 ; b++)

        gain[b].num_points = 0;

    return 0;

}

/**

 * Combine the tonal band spectrum and regular band spectrum

 *

 * @param spectrum        output spectrum buffer

 * @param num_components  number of tonal components

 * @param components      tonal components for this band

 * @return                position of the last tonal coefficient

 */

static int add_tonal_components(float *spectrum, int num_components,

                                TonalComponent *components)

{

    int i, j, last_pos = -1;

    float *input, *output;

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

        last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);

        input    = components[i].coef;

        output   = &spectrum[components[i].pos];

        for (j = 0; j < components[i].num_coefs; j++)

            output[j] += input[j];

    }

    return last_pos;

}

#define INTERPOLATE(old, new, nsample) \

    ((old) + (nsample) * 0.125 * ((new) - (old)))

static void reverse_matrixing(float *su1, float *su2, int *prev_code,

                              int *curr_code)

{

    int i, nsample, band;

    float mc1_l, mc1_r, mc2_l, mc2_r;

    for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {

        int s1 = prev_code[i];

        int s2 = curr_code[i];

        nsample = band;

        if (s1 != s2) {

            /* Selector value changed, interpolation needed. */

            mc1_l = matrix_coeffs[s1 * 2    ];

            mc1_r = matrix_coeffs[s1 * 2 + 1];

            mc2_l = matrix_coeffs[s2 * 2    ];

            mc2_r = matrix_coeffs[s2 * 2 + 1];

            /* Interpolation is done over the first eight samples. */

            for (; nsample < band + 8; nsample++) {

                float c1 = su1[nsample];

                float c2 = su2[nsample];

                c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +

                     c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);

                su1[nsample] = c2;

                su2[nsample] = c1 * 2.0 - c2;

            }

        }

        /* Apply the matrix without interpolation. */

        switch (s2) {

        case 0:     /* M/S decoding */

            for (; nsample < band + 256; nsample++) {

                float c1 = su1[nsample];

                float c2 = su2[nsample];

                su1[nsample] =  c2       * 2.0;

                su2[nsample] = (c1 - c2) * 2.0;

            }

            break;

        case 1:

            for (; nsample < band + 256; nsample++) {

                float c1 = su1[nsample];

                float c2 = su2[nsample];

                su1[nsample] = (c1 + c2) *  2.0;

                su2[nsample] =  c2       * -2.0;

            }

            break;

        case 2:

        case 3:

            for (; nsample < band + 256; nsample++) {

                float c1 = su1[nsample];

                float c2 = su2[nsample];

                su1[nsample] = c1 + c2;

                su2[nsample] = c1 - c2;

            }

            break;

        default:

            av_assert1(0);

        }

    }

}

static void get_channel_weights(int index, int flag, float ch[2])

{

    if (index == 7) {

        ch[0] = 1.0;

        ch[1] = 1.0;

    } else {

        ch[0] = (index & 7) / 7.0;

        ch[1] = sqrt(2 - ch[0] * ch[0]);

        if (flag)

            FFSWAP(float, ch[0], ch[1]);

    }

}

static void channel_weighting(float *su1, float *su2, int *p3)

{

    int band, nsample;

    /* w[x][y] y=0 is left y=1 is right */

    float w[2][2];

    if (p3[1] != 7 || p3[3] != 7) {

        get_channel_weights(p3[1], p3[0], w[0]);

        get_channel_weights(p3[3], p3[2], w[1]);

        for (band = 256; band < 4 * 256; band += 256) {

            for (nsample = band; nsample < band + 8; nsample++) {

                su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);

                su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);

            }

            for(; nsample < band + 256; nsample++) {

                su1[nsample] *= w[1][0];

                su2[nsample] *= w[1][1];

            }

        }

    }

}

/**

 * Decode a Sound Unit

 *

 * @param snd           the channel unit to be used

 * @param output        the decoded samples before IQMF in float representation

 * @param channel_num   channel number

 * @param coding_mode   the coding mode (JOINT_STEREO or regular stereo/mono)

 */

static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,

                                     ChannelUnit *snd, float *output,

                                     int channel_num, int coding_mode)

{

    int band, ret, num_subbands, last_tonal, num_bands;

    GainBlock *gain1 = &snd->gain_block[    snd->gc_blk_switch];

    GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];

    if (coding_mode == JOINT_STEREO && channel_num == 1) {

        if (get_bits(gb, 2) != 3) {

            av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");

            return AVERROR_INVALIDDATA;

        }

    } else {

        if (get_bits(gb, 6) != 0x28) {

            av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");

            return AVERROR_INVALIDDATA;

        }

    }

    /* number of coded QMF bands */

    snd->bands_coded = get_bits(gb, 2);

    ret = decode_gain_control(gb, gain2, snd->bands_coded);

    if (ret)

        return ret;

    snd->num_components = decode_tonal_components(gb, snd->components,

                                                  snd->bands_coded);

    if (snd->num_components < 0)

        return snd->num_components;

    num_subbands = decode_spectrum(gb, snd->spectrum);

    /* Merge the decoded spectrum and tonal components. */

    last_tonal = add_tonal_components(snd->spectrum, snd->num_components,

                                      snd->components);

    /* calculate number of used MLT/QMF bands according to the amount of coded

       spectral lines */

    num_bands = (subband_tab[num_subbands] - 1) >> 8;

    if (last_tonal >= 0)

        num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);

    /* Reconstruct time domain samples. */

    for (band = 0; band < 4; band++) {

        /* Perform the IMDCT step without overlapping. */

        if (band <= num_bands)

            imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);

        else

            memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));

        /* gain compensation and overlapping */

        ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,

                                   &snd->prev_frame[band * 256],

                                   &gain1->g_block[band], &gain2->g_block[band],

                                   256, &output[band * 256]);

    }

    /* Swap the gain control buffers for the next frame. */

    snd->gc_blk_switch ^= 1;

    return 0;

}

static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,

                        float **out_samples)

{

    ATRAC3Context *q = avctx->priv_data;

    int ret, i;

    uint8_t *ptr1;

    if (q->coding_mode == JOINT_STEREO) {

        /* channel coupling mode */

        /* decode Sound Unit 1 */

        init_get_bits(&q->gb, databuf, avctx->block_align * 8);

        ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,

                                        JOINT_STEREO);

        if (ret != 0)

            return ret;

        /* Framedata of the su2 in the joint-stereo mode is encoded in

         * reverse byte order so we need to swap it first. */

        if (databuf == q->decoded_bytes_buffer) {

            uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;

            ptr1          = q->decoded_bytes_buffer;

            for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)

                FFSWAP(uint8_t, *ptr1, *ptr2);

        } else {

            const uint8_t *ptr2 = databuf + avctx->block_align - 1;

            for (i = 0; i < avctx->block_align; i++)

                q->decoded_bytes_buffer[i] = *ptr2--;

        }

        /* Skip the sync codes (0xF8). */

        ptr1 = q->decoded_bytes_buffer;

        for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {

            if (i >= avctx->block_align)

                return AVERROR_INVALIDDATA;

        }

        /* set the bitstream reader at the start of the second Sound Unit*/

        init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1);

        /* Fill the Weighting coeffs delay buffer */

        memmove(q->weighting_delay, &q->weighting_delay[2],

                4 * sizeof(*q->weighting_delay));

        q->weighting_delay[4] = get_bits1(&q->gb);

        q->weighting_delay[5] = get_bits(&q->gb, 3);

        for (i = 0; i < 4; i++) {

            q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];

            q->matrix_coeff_index_now[i]  = q->matrix_coeff_index_next[i];

            q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);

        }

        /* Decode Sound Unit 2. */

        ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],

                                        out_samples[1], 1, JOINT_STEREO);

        if (ret != 0)

            return ret;

        /* Reconstruct the channel coefficients. */

        reverse_matrixing(out_samples[0], out_samples[1],

                          q->matrix_coeff_index_prev,

                          q->matrix_coeff_index_now);

        channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);

    } else {

        /* normal stereo mode or mono */

        /* Decode the channel sound units. */

        for (i = 0; i < avctx->channels; i++) {

            /* Set the bitstream reader at the start of a channel sound unit. */

            init_get_bits(&q->gb,

                          databuf + i * avctx->block_align / avctx->channels,

                          avctx->block_align * 8 / avctx->channels);

            ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],

                                            out_samples[i], i, q->coding_mode);

            if (ret != 0)

                return ret;

        }

    }

    /* Apply the iQMF synthesis filter. */

    for (i = 0; i < avctx->channels; i++) {

        float *p1 = out_samples[i];

        float *p2 = p1 + 256;

        float *p3 = p2 + 256;

        float *p4 = p3 + 256;

        ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);

        ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);

        ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);

    }

    return 0;

}

static int atrac3_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;

    ATRAC3Context *q = avctx->priv_data;

    int ret;

    const uint8_t *databuf;

    if (buf_size < avctx->block_align) {

        av_log(avctx, AV_LOG_ERROR,

               "Frame too small (%d bytes). Truncated file?\n", buf_size);

        return AVERROR_INVALIDDATA;

    }

    /* get output buffer */

    frame->nb_samples = SAMPLES_PER_FRAME;

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

        return ret;

    /* Check if we need to descramble and what buffer to pass on. */

    if (q->scrambled_stream) {

        decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);

        databuf = q->decoded_bytes_buffer;

    } else {

        databuf = buf;

    }

    ret = decode_frame(avctx, databuf, (float **)frame->extended_data);

    if (ret) {

        av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");

        return ret;

    }

    *got_frame_ptr = 1;

    return avctx->block_align;

}

static av_cold void atrac3_init_static_data(void)

{

    int i;

    init_imdct_window();

    ff_atrac_generate_tables();

    /* Initialize the VLC tables. */

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

        spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];

        spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -

                                                atrac3_vlc_offs[i    ];

        init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],

                 huff_bits[i],  1, 1,

                 huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);

    }

}

static av_cold int atrac3_decode_init(AVCodecContext *avctx)

{

    static int static_init_done;

    int i, ret;

    int version, delay, samples_per_frame, frame_factor;

    const uint8_t *edata_ptr = avctx->extradata;

    ATRAC3Context *q = avctx->priv_data;

    if (avctx->channels <= 0 || avctx->channels > 2) {

        av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");

        return AVERROR(EINVAL);

    }

    if (!static_init_done)

        atrac3_init_static_data();

    static_init_done = 1;

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

    if (avctx->extradata_size == 14) {

        /* Parse the extradata, WAV format */

        av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",

               bytestream_get_le16(&edata_ptr));  // Unknown value always 1

        edata_ptr += 4;                             // samples per channel

        q->coding_mode = bytestream_get_le16(&edata_ptr);

        av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",

               bytestream_get_le16(&edata_ptr));  //Dupe of coding mode

        frame_factor = bytestream_get_le16(&edata_ptr);  // Unknown always 1

        av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",

               bytestream_get_le16(&edata_ptr));  // Unknown always 0

        /* setup */

        samples_per_frame    = SAMPLES_PER_FRAME * avctx->channels;

        version              = 4;

        delay                = 0x88E;

        q->coding_mode       = q->coding_mode ? JOINT_STEREO : STEREO;

        q->scrambled_stream  = 0;

        if (avctx->block_align !=  96 * avctx->channels * frame_factor &&

            avctx->block_align != 152 * avctx->channels * frame_factor &&

            avctx->block_align != 192 * avctx->channels * frame_factor) {

            av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "

                   "configuration %d/%d/%d\n", avctx->block_align,

                   avctx->channels, frame_factor);

            return AVERROR_INVALIDDATA;

        }

    } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {

        /* Parse the extradata, RM format. */

        version                = bytestream_get_be32(&edata_ptr);

        samples_per_frame      = bytestream_get_be16(&edata_ptr);

        delay                  = bytestream_get_be16(&edata_ptr);

        q->coding_mode         = bytestream_get_be16(&edata_ptr);

        q->scrambled_stream    = 1;

    } else {

        av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",

               avctx->extradata_size);

        return AVERROR(EINVAL);

    }

    /* Check the extradata */

    if (version != 4) {

        av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);

        return AVERROR_INVALIDDATA;

    }

    if (samples_per_frame != SAMPLES_PER_FRAME &&

        samples_per_frame != SAMPLES_PER_FRAME * 2) {

        av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",

               samples_per_frame);

        return AVERROR_INVALIDDATA;

    }

    if (delay != 0x88E) {

        av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",

               delay);

        return AVERROR_INVALIDDATA;

    }

    if (q->coding_mode == STEREO)

        av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");

    else if (q->coding_mode == JOINT_STEREO) {

        if (avctx->channels != 2) {

            av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");

            return AVERROR_INVALIDDATA;

        }

        av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");

    } else {

        av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",

               q->coding_mode);

        return AVERROR_INVALIDDATA;

    }

    if (avctx->block_align >= UINT_MAX / 2)

        return AVERROR(EINVAL);

    q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +

                                         FF_INPUT_BUFFER_PADDING_SIZE);

    if (q->decoded_bytes_buffer == NULL)

        return AVERROR(ENOMEM);

    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;

    /* initialize the MDCT transform */

    if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {

        av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");

        av_freep(&q->decoded_bytes_buffer);

        return ret;

    }

    /* init the joint-stereo decoding data */

    q->weighting_delay[0] = 0;

    q->weighting_delay[1] = 7;

    q->weighting_delay[2] = 0;

    q->weighting_delay[3] = 7;

    q->weighting_delay[4] = 0;

    q->weighting_delay[5] = 7;

    for (i = 0; i < 4; i++) {

        q->matrix_coeff_index_prev[i] = 3;

        q->matrix_coeff_index_now[i]  = 3;

        q->matrix_coeff_index_next[i] = 3;

    }

    ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);

    avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);

    ff_fmt_convert_init(&q->fmt_conv, avctx);

    q->units = av_mallocz(sizeof(*q->units) * avctx->channels);

    if (!q->units) {

        atrac3_decode_close(avctx);

        return AVERROR(ENOMEM);

    }

    return 0;

}

AVCodec ff_atrac3_decoder = {

    .name             = "atrac3",

    .long_name        = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),

    .type             = AVMEDIA_TYPE_AUDIO,

    .id               = AV_CODEC_ID_ATRAC3,

    .priv_data_size   = sizeof(ATRAC3Context),

    .init             = atrac3_decode_init,

    .close            = atrac3_decode_close,

    .decode           = atrac3_decode_frame,

    .capabilities     = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,

    .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,

                                                        AV_SAMPLE_FMT_NONE },

};