Subversion Repositories Kolibri OS

Rev

Blame | Last modification | View Log | RSS feed

  1. /*
  2.  * ATRAC3 compatible decoder
  3.  * Copyright (c) 2006-2008 Maxim Poliakovski
  4.  * Copyright (c) 2006-2008 Benjamin Larsson
  5.  *
  6.  * This file is part of FFmpeg.
  7.  *
  8.  * FFmpeg is free software; you can redistribute it and/or
  9.  * modify it under the terms of the GNU Lesser General Public
  10.  * License as published by the Free Software Foundation; either
  11.  * version 2.1 of the License, or (at your option) any later version.
  12.  *
  13.  * FFmpeg is distributed in the hope that it will be useful,
  14.  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16.  * Lesser General Public License for more details.
  17.  *
  18.  * You should have received a copy of the GNU Lesser General Public
  19.  * License along with FFmpeg; if not, write to the Free Software
  20.  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21.  */
  22.  
  23. /**
  24.  * @file
  25.  * ATRAC3 compatible decoder.
  26.  * This decoder handles Sony's ATRAC3 data.
  27.  *
  28.  * Container formats used to store ATRAC3 data:
  29.  * RealMedia (.rm), RIFF WAV (.wav, .at3), Sony OpenMG (.oma, .aa3).
  30.  *
  31.  * To use this decoder, a calling application must supply the extradata
  32.  * bytes provided in the containers above.
  33.  */
  34.  
  35. #include <math.h>
  36. #include <stddef.h>
  37. #include <stdio.h>
  38.  
  39. #include "libavutil/attributes.h"
  40. #include "libavutil/float_dsp.h"
  41. #include "libavutil/libm.h"
  42. #include "avcodec.h"
  43. #include "bytestream.h"
  44. #include "fft.h"
  45. #include "fmtconvert.h"
  46. #include "get_bits.h"
  47. #include "internal.h"
  48.  
  49. #include "atrac.h"
  50. #include "atrac3data.h"
  51.  
  52. #define JOINT_STEREO    0x12
  53. #define STEREO          0x2
  54.  
  55. #define SAMPLES_PER_FRAME 1024
  56. #define MDCT_SIZE          512
  57.  
  58. typedef struct GainBlock {
  59.     AtracGainInfo g_block[4];
  60. } GainBlock;
  61.  
  62. typedef struct TonalComponent {
  63.     int pos;
  64.     int num_coefs;
  65.     float coef[8];
  66. } TonalComponent;
  67.  
  68. typedef struct ChannelUnit {
  69.     int            bands_coded;
  70.     int            num_components;
  71.     float          prev_frame[SAMPLES_PER_FRAME];
  72.     int            gc_blk_switch;
  73.     TonalComponent components[64];
  74.     GainBlock      gain_block[2];
  75.  
  76.     DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
  77.     DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
  78.  
  79.     float          delay_buf1[46]; ///<qmf delay buffers
  80.     float          delay_buf2[46];
  81.     float          delay_buf3[46];
  82. } ChannelUnit;
  83.  
  84. typedef struct ATRAC3Context {
  85.     GetBitContext gb;
  86.     //@{
  87.     /** stream data */
  88.     int coding_mode;
  89.  
  90.     ChannelUnit *units;
  91.     //@}
  92.     //@{
  93.     /** joint-stereo related variables */
  94.     int matrix_coeff_index_prev[4];
  95.     int matrix_coeff_index_now[4];
  96.     int matrix_coeff_index_next[4];
  97.     int weighting_delay[6];
  98.     //@}
  99.     //@{
  100.     /** data buffers */
  101.     uint8_t *decoded_bytes_buffer;
  102.     float temp_buf[1070];
  103.     //@}
  104.     //@{
  105.     /** extradata */
  106.     int scrambled_stream;
  107.     //@}
  108.  
  109.     AtracGCContext    gainc_ctx;
  110.     FFTContext        mdct_ctx;
  111.     FmtConvertContext fmt_conv;
  112.     AVFloatDSPContext fdsp;
  113. } ATRAC3Context;
  114.  
  115. static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
  116. static VLC_TYPE atrac3_vlc_table[4096][2];
  117. static VLC   spectral_coeff_tab[7];
  118.  
  119. /**
  120.  * Regular 512 points IMDCT without overlapping, with the exception of the
  121.  * swapping of odd bands caused by the reverse spectra of the QMF.
  122.  *
  123.  * @param odd_band  1 if the band is an odd band
  124.  */
  125. static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
  126. {
  127.     int i;
  128.  
  129.     if (odd_band) {
  130.         /**
  131.          * Reverse the odd bands before IMDCT, this is an effect of the QMF
  132.          * transform or it gives better compression to do it this way.
  133.          * FIXME: It should be possible to handle this in imdct_calc
  134.          * for that to happen a modification of the prerotation step of
  135.          * all SIMD code and C code is needed.
  136.          * Or fix the functions before so they generate a pre reversed spectrum.
  137.          */
  138.         for (i = 0; i < 128; i++)
  139.             FFSWAP(float, input[i], input[255 - i]);
  140.     }
  141.  
  142.     q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
  143.  
  144.     /* Perform windowing on the output. */
  145.     q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE);
  146. }
  147.  
  148. /*
  149.  * indata descrambling, only used for data coming from the rm container
  150.  */
  151. static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
  152. {
  153.     int i, off;
  154.     uint32_t c;
  155.     const uint32_t *buf;
  156.     uint32_t *output = (uint32_t *)out;
  157.  
  158.     off = (intptr_t)input & 3;
  159.     buf = (const uint32_t *)(input - off);
  160.     if (off)
  161.         c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
  162.     else
  163.         c = av_be2ne32(0x537F6103U);
  164.     bytes += 3 + off;
  165.     for (i = 0; i < bytes / 4; i++)
  166.         output[i] = c ^ buf[i];
  167.  
  168.     if (off)
  169.         avpriv_request_sample(NULL, "Offset of %d", off);
  170.  
  171.     return off;
  172. }
  173.  
  174. static av_cold void init_imdct_window(void)
  175. {
  176.     int i, j;
  177.  
  178.     /* generate the mdct window, for details see
  179.      * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
  180.     for (i = 0, j = 255; i < 128; i++, j--) {
  181.         float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
  182.         float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
  183.         float w  = 0.5 * (wi * wi + wj * wj);
  184.         mdct_window[i] = mdct_window[511 - i] = wi / w;
  185.         mdct_window[j] = mdct_window[511 - j] = wj / w;
  186.     }
  187. }
  188.  
  189. static av_cold int atrac3_decode_close(AVCodecContext *avctx)
  190. {
  191.     ATRAC3Context *q = avctx->priv_data;
  192.  
  193.     av_free(q->units);
  194.     av_free(q->decoded_bytes_buffer);
  195.  
  196.     ff_mdct_end(&q->mdct_ctx);
  197.  
  198.     return 0;
  199. }
  200.  
  201. /**
  202.  * Mantissa decoding
  203.  *
  204.  * @param selector     which table the output values are coded with
  205.  * @param coding_flag  constant length coding or variable length coding
  206.  * @param mantissas    mantissa output table
  207.  * @param num_codes    number of values to get
  208.  */
  209. static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
  210.                                        int coding_flag, int *mantissas,
  211.                                        int num_codes)
  212. {
  213.     int i, code, huff_symb;
  214.  
  215.     if (selector == 1)
  216.         num_codes /= 2;
  217.  
  218.     if (coding_flag != 0) {
  219.         /* constant length coding (CLC) */
  220.         int num_bits = clc_length_tab[selector];
  221.  
  222.         if (selector > 1) {
  223.             for (i = 0; i < num_codes; i++) {
  224.                 if (num_bits)
  225.                     code = get_sbits(gb, num_bits);
  226.                 else
  227.                     code = 0;
  228.                 mantissas[i] = code;
  229.             }
  230.         } else {
  231.             for (i = 0; i < num_codes; i++) {
  232.                 if (num_bits)
  233.                     code = get_bits(gb, num_bits); // num_bits is always 4 in this case
  234.                 else
  235.                     code = 0;
  236.                 mantissas[i * 2    ] = mantissa_clc_tab[code >> 2];
  237.                 mantissas[i * 2 + 1] = mantissa_clc_tab[code &  3];
  238.             }
  239.         }
  240.     } else {
  241.         /* variable length coding (VLC) */
  242.         if (selector != 1) {
  243.             for (i = 0; i < num_codes; i++) {
  244.                 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
  245.                                      spectral_coeff_tab[selector-1].bits, 3);
  246.                 huff_symb += 1;
  247.                 code = huff_symb >> 1;
  248.                 if (huff_symb & 1)
  249.                     code = -code;
  250.                 mantissas[i] = code;
  251.             }
  252.         } else {
  253.             for (i = 0; i < num_codes; i++) {
  254.                 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
  255.                                      spectral_coeff_tab[selector - 1].bits, 3);
  256.                 mantissas[i * 2    ] = mantissa_vlc_tab[huff_symb * 2    ];
  257.                 mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
  258.             }
  259.         }
  260.     }
  261. }
  262.  
  263. /**
  264.  * Restore the quantized band spectrum coefficients
  265.  *
  266.  * @return subband count, fix for broken specification/files
  267.  */
  268. static int decode_spectrum(GetBitContext *gb, float *output)
  269. {
  270.     int num_subbands, coding_mode, i, j, first, last, subband_size;
  271.     int subband_vlc_index[32], sf_index[32];
  272.     int mantissas[128];
  273.     float scale_factor;
  274.  
  275.     num_subbands = get_bits(gb, 5);  // number of coded subbands
  276.     coding_mode  = get_bits1(gb);    // coding Mode: 0 - VLC/ 1-CLC
  277.  
  278.     /* get the VLC selector table for the subbands, 0 means not coded */
  279.     for (i = 0; i <= num_subbands; i++)
  280.         subband_vlc_index[i] = get_bits(gb, 3);
  281.  
  282.     /* read the scale factor indexes from the stream */
  283.     for (i = 0; i <= num_subbands; i++) {
  284.         if (subband_vlc_index[i] != 0)
  285.             sf_index[i] = get_bits(gb, 6);
  286.     }
  287.  
  288.     for (i = 0; i <= num_subbands; i++) {
  289.         first = subband_tab[i    ];
  290.         last  = subband_tab[i + 1];
  291.  
  292.         subband_size = last - first;
  293.  
  294.         if (subband_vlc_index[i] != 0) {
  295.             /* decode spectral coefficients for this subband */
  296.             /* TODO: This can be done faster is several blocks share the
  297.              * same VLC selector (subband_vlc_index) */
  298.             read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
  299.                                        mantissas, subband_size);
  300.  
  301.             /* decode the scale factor for this subband */
  302.             scale_factor = ff_atrac_sf_table[sf_index[i]] *
  303.                            inv_max_quant[subband_vlc_index[i]];
  304.  
  305.             /* inverse quantize the coefficients */
  306.             for (j = 0; first < last; first++, j++)
  307.                 output[first] = mantissas[j] * scale_factor;
  308.         } else {
  309.             /* this subband was not coded, so zero the entire subband */
  310.             memset(output + first, 0, subband_size * sizeof(*output));
  311.         }
  312.     }
  313.  
  314.     /* clear the subbands that were not coded */
  315.     first = subband_tab[i];
  316.     memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
  317.     return num_subbands;
  318. }
  319.  
  320. /**
  321.  * Restore the quantized tonal components
  322.  *
  323.  * @param components tonal components
  324.  * @param num_bands  number of coded bands
  325.  */
  326. static int decode_tonal_components(GetBitContext *gb,
  327.                                    TonalComponent *components, int num_bands)
  328. {
  329.     int i, b, c, m;
  330.     int nb_components, coding_mode_selector, coding_mode;
  331.     int band_flags[4], mantissa[8];
  332.     int component_count = 0;
  333.  
  334.     nb_components = get_bits(gb, 5);
  335.  
  336.     /* no tonal components */
  337.     if (nb_components == 0)
  338.         return 0;
  339.  
  340.     coding_mode_selector = get_bits(gb, 2);
  341.     if (coding_mode_selector == 2)
  342.         return AVERROR_INVALIDDATA;
  343.  
  344.     coding_mode = coding_mode_selector & 1;
  345.  
  346.     for (i = 0; i < nb_components; i++) {
  347.         int coded_values_per_component, quant_step_index;
  348.  
  349.         for (b = 0; b <= num_bands; b++)
  350.             band_flags[b] = get_bits1(gb);
  351.  
  352.         coded_values_per_component = get_bits(gb, 3);
  353.  
  354.         quant_step_index = get_bits(gb, 3);
  355.         if (quant_step_index <= 1)
  356.             return AVERROR_INVALIDDATA;
  357.  
  358.         if (coding_mode_selector == 3)
  359.             coding_mode = get_bits1(gb);
  360.  
  361.         for (b = 0; b < (num_bands + 1) * 4; b++) {
  362.             int coded_components;
  363.  
  364.             if (band_flags[b >> 2] == 0)
  365.                 continue;
  366.  
  367.             coded_components = get_bits(gb, 3);
  368.  
  369.             for (c = 0; c < coded_components; c++) {
  370.                 TonalComponent *cmp = &components[component_count];
  371.                 int sf_index, coded_values, max_coded_values;
  372.                 float scale_factor;
  373.  
  374.                 sf_index = get_bits(gb, 6);
  375.                 if (component_count >= 64)
  376.                     return AVERROR_INVALIDDATA;
  377.  
  378.                 cmp->pos = b * 64 + get_bits(gb, 6);
  379.  
  380.                 max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
  381.                 coded_values     = coded_values_per_component + 1;
  382.                 coded_values     = FFMIN(max_coded_values, coded_values);
  383.  
  384.                 scale_factor = ff_atrac_sf_table[sf_index] *
  385.                                inv_max_quant[quant_step_index];
  386.  
  387.                 read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
  388.                                            mantissa, coded_values);
  389.  
  390.                 cmp->num_coefs = coded_values;
  391.  
  392.                 /* inverse quant */
  393.                 for (m = 0; m < coded_values; m++)
  394.                     cmp->coef[m] = mantissa[m] * scale_factor;
  395.  
  396.                 component_count++;
  397.             }
  398.         }
  399.     }
  400.  
  401.     return component_count;
  402. }
  403.  
  404. /**
  405.  * Decode gain parameters for the coded bands
  406.  *
  407.  * @param block      the gainblock for the current band
  408.  * @param num_bands  amount of coded bands
  409.  */
  410. static int decode_gain_control(GetBitContext *gb, GainBlock *block,
  411.                                int num_bands)
  412. {
  413.     int b, j;
  414.     int *level, *loc;
  415.  
  416.     AtracGainInfo *gain = block->g_block;
  417.  
  418.     for (b = 0; b <= num_bands; b++) {
  419.         gain[b].num_points = get_bits(gb, 3);
  420.         level              = gain[b].lev_code;
  421.         loc                = gain[b].loc_code;
  422.  
  423.         for (j = 0; j < gain[b].num_points; j++) {
  424.             level[j] = get_bits(gb, 4);
  425.             loc[j]   = get_bits(gb, 5);
  426.             if (j && loc[j] <= loc[j - 1])
  427.                 return AVERROR_INVALIDDATA;
  428.         }
  429.     }
  430.  
  431.     /* Clear the unused blocks. */
  432.     for (; b < 4 ; b++)
  433.         gain[b].num_points = 0;
  434.  
  435.     return 0;
  436. }
  437.  
  438. /**
  439.  * Combine the tonal band spectrum and regular band spectrum
  440.  *
  441.  * @param spectrum        output spectrum buffer
  442.  * @param num_components  number of tonal components
  443.  * @param components      tonal components for this band
  444.  * @return                position of the last tonal coefficient
  445.  */
  446. static int add_tonal_components(float *spectrum, int num_components,
  447.                                 TonalComponent *components)
  448. {
  449.     int i, j, last_pos = -1;
  450.     float *input, *output;
  451.  
  452.     for (i = 0; i < num_components; i++) {
  453.         last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
  454.         input    = components[i].coef;
  455.         output   = &spectrum[components[i].pos];
  456.  
  457.         for (j = 0; j < components[i].num_coefs; j++)
  458.             output[j] += input[j];
  459.     }
  460.  
  461.     return last_pos;
  462. }
  463.  
  464. #define INTERPOLATE(old, new, nsample) \
  465.     ((old) + (nsample) * 0.125 * ((new) - (old)))
  466.  
  467. static void reverse_matrixing(float *su1, float *su2, int *prev_code,
  468.                               int *curr_code)
  469. {
  470.     int i, nsample, band;
  471.     float mc1_l, mc1_r, mc2_l, mc2_r;
  472.  
  473.     for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
  474.         int s1 = prev_code[i];
  475.         int s2 = curr_code[i];
  476.         nsample = band;
  477.  
  478.         if (s1 != s2) {
  479.             /* Selector value changed, interpolation needed. */
  480.             mc1_l = matrix_coeffs[s1 * 2    ];
  481.             mc1_r = matrix_coeffs[s1 * 2 + 1];
  482.             mc2_l = matrix_coeffs[s2 * 2    ];
  483.             mc2_r = matrix_coeffs[s2 * 2 + 1];
  484.  
  485.             /* Interpolation is done over the first eight samples. */
  486.             for (; nsample < band + 8; nsample++) {
  487.                 float c1 = su1[nsample];
  488.                 float c2 = su2[nsample];
  489.                 c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
  490.                      c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
  491.                 su1[nsample] = c2;
  492.                 su2[nsample] = c1 * 2.0 - c2;
  493.             }
  494.         }
  495.  
  496.         /* Apply the matrix without interpolation. */
  497.         switch (s2) {
  498.         case 0:     /* M/S decoding */
  499.             for (; nsample < band + 256; nsample++) {
  500.                 float c1 = su1[nsample];
  501.                 float c2 = su2[nsample];
  502.                 su1[nsample] =  c2       * 2.0;
  503.                 su2[nsample] = (c1 - c2) * 2.0;
  504.             }
  505.             break;
  506.         case 1:
  507.             for (; nsample < band + 256; nsample++) {
  508.                 float c1 = su1[nsample];
  509.                 float c2 = su2[nsample];
  510.                 su1[nsample] = (c1 + c2) *  2.0;
  511.                 su2[nsample] =  c2       * -2.0;
  512.             }
  513.             break;
  514.         case 2:
  515.         case 3:
  516.             for (; nsample < band + 256; nsample++) {
  517.                 float c1 = su1[nsample];
  518.                 float c2 = su2[nsample];
  519.                 su1[nsample] = c1 + c2;
  520.                 su2[nsample] = c1 - c2;
  521.             }
  522.             break;
  523.         default:
  524.             av_assert1(0);
  525.         }
  526.     }
  527. }
  528.  
  529. static void get_channel_weights(int index, int flag, float ch[2])
  530. {
  531.     if (index == 7) {
  532.         ch[0] = 1.0;
  533.         ch[1] = 1.0;
  534.     } else {
  535.         ch[0] = (index & 7) / 7.0;
  536.         ch[1] = sqrt(2 - ch[0] * ch[0]);
  537.         if (flag)
  538.             FFSWAP(float, ch[0], ch[1]);
  539.     }
  540. }
  541.  
  542. static void channel_weighting(float *su1, float *su2, int *p3)
  543. {
  544.     int band, nsample;
  545.     /* w[x][y] y=0 is left y=1 is right */
  546.     float w[2][2];
  547.  
  548.     if (p3[1] != 7 || p3[3] != 7) {
  549.         get_channel_weights(p3[1], p3[0], w[0]);
  550.         get_channel_weights(p3[3], p3[2], w[1]);
  551.  
  552.         for (band = 256; band < 4 * 256; band += 256) {
  553.             for (nsample = band; nsample < band + 8; nsample++) {
  554.                 su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
  555.                 su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
  556.             }
  557.             for(; nsample < band + 256; nsample++) {
  558.                 su1[nsample] *= w[1][0];
  559.                 su2[nsample] *= w[1][1];
  560.             }
  561.         }
  562.     }
  563. }
  564.  
  565. /**
  566.  * Decode a Sound Unit
  567.  *
  568.  * @param snd           the channel unit to be used
  569.  * @param output        the decoded samples before IQMF in float representation
  570.  * @param channel_num   channel number
  571.  * @param coding_mode   the coding mode (JOINT_STEREO or regular stereo/mono)
  572.  */
  573. static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
  574.                                      ChannelUnit *snd, float *output,
  575.                                      int channel_num, int coding_mode)
  576. {
  577.     int band, ret, num_subbands, last_tonal, num_bands;
  578.     GainBlock *gain1 = &snd->gain_block[    snd->gc_blk_switch];
  579.     GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
  580.  
  581.     if (coding_mode == JOINT_STEREO && channel_num == 1) {
  582.         if (get_bits(gb, 2) != 3) {
  583.             av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
  584.             return AVERROR_INVALIDDATA;
  585.         }
  586.     } else {
  587.         if (get_bits(gb, 6) != 0x28) {
  588.             av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
  589.             return AVERROR_INVALIDDATA;
  590.         }
  591.     }
  592.  
  593.     /* number of coded QMF bands */
  594.     snd->bands_coded = get_bits(gb, 2);
  595.  
  596.     ret = decode_gain_control(gb, gain2, snd->bands_coded);
  597.     if (ret)
  598.         return ret;
  599.  
  600.     snd->num_components = decode_tonal_components(gb, snd->components,
  601.                                                   snd->bands_coded);
  602.     if (snd->num_components < 0)
  603.         return snd->num_components;
  604.  
  605.     num_subbands = decode_spectrum(gb, snd->spectrum);
  606.  
  607.     /* Merge the decoded spectrum and tonal components. */
  608.     last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
  609.                                       snd->components);
  610.  
  611.  
  612.     /* calculate number of used MLT/QMF bands according to the amount of coded
  613.        spectral lines */
  614.     num_bands = (subband_tab[num_subbands] - 1) >> 8;
  615.     if (last_tonal >= 0)
  616.         num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
  617.  
  618.  
  619.     /* Reconstruct time domain samples. */
  620.     for (band = 0; band < 4; band++) {
  621.         /* Perform the IMDCT step without overlapping. */
  622.         if (band <= num_bands)
  623.             imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
  624.         else
  625.             memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
  626.  
  627.         /* gain compensation and overlapping */
  628.         ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
  629.                                    &snd->prev_frame[band * 256],
  630.                                    &gain1->g_block[band], &gain2->g_block[band],
  631.                                    256, &output[band * 256]);
  632.     }
  633.  
  634.     /* Swap the gain control buffers for the next frame. */
  635.     snd->gc_blk_switch ^= 1;
  636.  
  637.     return 0;
  638. }
  639.  
  640. static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
  641.                         float **out_samples)
  642. {
  643.     ATRAC3Context *q = avctx->priv_data;
  644.     int ret, i;
  645.     uint8_t *ptr1;
  646.  
  647.     if (q->coding_mode == JOINT_STEREO) {
  648.         /* channel coupling mode */
  649.         /* decode Sound Unit 1 */
  650.         init_get_bits(&q->gb, databuf, avctx->block_align * 8);
  651.  
  652.         ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
  653.                                         JOINT_STEREO);
  654.         if (ret != 0)
  655.             return ret;
  656.  
  657.         /* Framedata of the su2 in the joint-stereo mode is encoded in
  658.          * reverse byte order so we need to swap it first. */
  659.         if (databuf == q->decoded_bytes_buffer) {
  660.             uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
  661.             ptr1          = q->decoded_bytes_buffer;
  662.             for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
  663.                 FFSWAP(uint8_t, *ptr1, *ptr2);
  664.         } else {
  665.             const uint8_t *ptr2 = databuf + avctx->block_align - 1;
  666.             for (i = 0; i < avctx->block_align; i++)
  667.                 q->decoded_bytes_buffer[i] = *ptr2--;
  668.         }
  669.  
  670.         /* Skip the sync codes (0xF8). */
  671.         ptr1 = q->decoded_bytes_buffer;
  672.         for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
  673.             if (i >= avctx->block_align)
  674.                 return AVERROR_INVALIDDATA;
  675.         }
  676.  
  677.  
  678.         /* set the bitstream reader at the start of the second Sound Unit*/
  679.         init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1);
  680.  
  681.         /* Fill the Weighting coeffs delay buffer */
  682.         memmove(q->weighting_delay, &q->weighting_delay[2],
  683.                 4 * sizeof(*q->weighting_delay));
  684.         q->weighting_delay[4] = get_bits1(&q->gb);
  685.         q->weighting_delay[5] = get_bits(&q->gb, 3);
  686.  
  687.         for (i = 0; i < 4; i++) {
  688.             q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
  689.             q->matrix_coeff_index_now[i]  = q->matrix_coeff_index_next[i];
  690.             q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
  691.         }
  692.  
  693.         /* Decode Sound Unit 2. */
  694.         ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
  695.                                         out_samples[1], 1, JOINT_STEREO);
  696.         if (ret != 0)
  697.             return ret;
  698.  
  699.         /* Reconstruct the channel coefficients. */
  700.         reverse_matrixing(out_samples[0], out_samples[1],
  701.                           q->matrix_coeff_index_prev,
  702.                           q->matrix_coeff_index_now);
  703.  
  704.         channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
  705.     } else {
  706.         /* normal stereo mode or mono */
  707.         /* Decode the channel sound units. */
  708.         for (i = 0; i < avctx->channels; i++) {
  709.             /* Set the bitstream reader at the start of a channel sound unit. */
  710.             init_get_bits(&q->gb,
  711.                           databuf + i * avctx->block_align / avctx->channels,
  712.                           avctx->block_align * 8 / avctx->channels);
  713.  
  714.             ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
  715.                                             out_samples[i], i, q->coding_mode);
  716.             if (ret != 0)
  717.                 return ret;
  718.         }
  719.     }
  720.  
  721.     /* Apply the iQMF synthesis filter. */
  722.     for (i = 0; i < avctx->channels; i++) {
  723.         float *p1 = out_samples[i];
  724.         float *p2 = p1 + 256;
  725.         float *p3 = p2 + 256;
  726.         float *p4 = p3 + 256;
  727.         ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
  728.         ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
  729.         ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
  730.     }
  731.  
  732.     return 0;
  733. }
  734.  
  735. static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
  736.                                int *got_frame_ptr, AVPacket *avpkt)
  737. {
  738.     AVFrame *frame     = data;
  739.     const uint8_t *buf = avpkt->data;
  740.     int buf_size = avpkt->size;
  741.     ATRAC3Context *q = avctx->priv_data;
  742.     int ret;
  743.     const uint8_t *databuf;
  744.  
  745.     if (buf_size < avctx->block_align) {
  746.         av_log(avctx, AV_LOG_ERROR,
  747.                "Frame too small (%d bytes). Truncated file?\n", buf_size);
  748.         return AVERROR_INVALIDDATA;
  749.     }
  750.  
  751.     /* get output buffer */
  752.     frame->nb_samples = SAMPLES_PER_FRAME;
  753.     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
  754.         return ret;
  755.  
  756.     /* Check if we need to descramble and what buffer to pass on. */
  757.     if (q->scrambled_stream) {
  758.         decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
  759.         databuf = q->decoded_bytes_buffer;
  760.     } else {
  761.         databuf = buf;
  762.     }
  763.  
  764.     ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
  765.     if (ret) {
  766.         av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
  767.         return ret;
  768.     }
  769.  
  770.     *got_frame_ptr = 1;
  771.  
  772.     return avctx->block_align;
  773. }
  774.  
  775. static av_cold void atrac3_init_static_data(void)
  776. {
  777.     int i;
  778.  
  779.     init_imdct_window();
  780.     ff_atrac_generate_tables();
  781.  
  782.     /* Initialize the VLC tables. */
  783.     for (i = 0; i < 7; i++) {
  784.         spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
  785.         spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
  786.                                                 atrac3_vlc_offs[i    ];
  787.         init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
  788.                  huff_bits[i],  1, 1,
  789.                  huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
  790.     }
  791. }
  792.  
  793. static av_cold int atrac3_decode_init(AVCodecContext *avctx)
  794. {
  795.     static int static_init_done;
  796.     int i, ret;
  797.     int version, delay, samples_per_frame, frame_factor;
  798.     const uint8_t *edata_ptr = avctx->extradata;
  799.     ATRAC3Context *q = avctx->priv_data;
  800.  
  801.     if (avctx->channels <= 0 || avctx->channels > 2) {
  802.         av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
  803.         return AVERROR(EINVAL);
  804.     }
  805.  
  806.     if (!static_init_done)
  807.         atrac3_init_static_data();
  808.     static_init_done = 1;
  809.  
  810.     /* Take care of the codec-specific extradata. */
  811.     if (avctx->extradata_size == 14) {
  812.         /* Parse the extradata, WAV format */
  813.         av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
  814.                bytestream_get_le16(&edata_ptr));  // Unknown value always 1
  815.         edata_ptr += 4;                             // samples per channel
  816.         q->coding_mode = bytestream_get_le16(&edata_ptr);
  817.         av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
  818.                bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
  819.         frame_factor = bytestream_get_le16(&edata_ptr);  // Unknown always 1
  820.         av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
  821.                bytestream_get_le16(&edata_ptr));  // Unknown always 0
  822.  
  823.         /* setup */
  824.         samples_per_frame    = SAMPLES_PER_FRAME * avctx->channels;
  825.         version              = 4;
  826.         delay                = 0x88E;
  827.         q->coding_mode       = q->coding_mode ? JOINT_STEREO : STEREO;
  828.         q->scrambled_stream  = 0;
  829.  
  830.         if (avctx->block_align !=  96 * avctx->channels * frame_factor &&
  831.             avctx->block_align != 152 * avctx->channels * frame_factor &&
  832.             avctx->block_align != 192 * avctx->channels * frame_factor) {
  833.             av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
  834.                    "configuration %d/%d/%d\n", avctx->block_align,
  835.                    avctx->channels, frame_factor);
  836.             return AVERROR_INVALIDDATA;
  837.         }
  838.     } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
  839.         /* Parse the extradata, RM format. */
  840.         version                = bytestream_get_be32(&edata_ptr);
  841.         samples_per_frame      = bytestream_get_be16(&edata_ptr);
  842.         delay                  = bytestream_get_be16(&edata_ptr);
  843.         q->coding_mode         = bytestream_get_be16(&edata_ptr);
  844.         q->scrambled_stream    = 1;
  845.  
  846.     } else {
  847.         av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
  848.                avctx->extradata_size);
  849.         return AVERROR(EINVAL);
  850.     }
  851.  
  852.     /* Check the extradata */
  853.  
  854.     if (version != 4) {
  855.         av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
  856.         return AVERROR_INVALIDDATA;
  857.     }
  858.  
  859.     if (samples_per_frame != SAMPLES_PER_FRAME &&
  860.         samples_per_frame != SAMPLES_PER_FRAME * 2) {
  861.         av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
  862.                samples_per_frame);
  863.         return AVERROR_INVALIDDATA;
  864.     }
  865.  
  866.     if (delay != 0x88E) {
  867.         av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
  868.                delay);
  869.         return AVERROR_INVALIDDATA;
  870.     }
  871.  
  872.     if (q->coding_mode == STEREO)
  873.         av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
  874.     else if (q->coding_mode == JOINT_STEREO) {
  875.         if (avctx->channels != 2) {
  876.             av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");
  877.             return AVERROR_INVALIDDATA;
  878.         }
  879.         av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
  880.     } else {
  881.         av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
  882.                q->coding_mode);
  883.         return AVERROR_INVALIDDATA;
  884.     }
  885.  
  886.     if (avctx->block_align >= UINT_MAX / 2)
  887.         return AVERROR(EINVAL);
  888.  
  889.     q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
  890.                                          FF_INPUT_BUFFER_PADDING_SIZE);
  891.     if (q->decoded_bytes_buffer == NULL)
  892.         return AVERROR(ENOMEM);
  893.  
  894.     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
  895.  
  896.     /* initialize the MDCT transform */
  897.     if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
  898.         av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
  899.         av_freep(&q->decoded_bytes_buffer);
  900.         return ret;
  901.     }
  902.  
  903.     /* init the joint-stereo decoding data */
  904.     q->weighting_delay[0] = 0;
  905.     q->weighting_delay[1] = 7;
  906.     q->weighting_delay[2] = 0;
  907.     q->weighting_delay[3] = 7;
  908.     q->weighting_delay[4] = 0;
  909.     q->weighting_delay[5] = 7;
  910.  
  911.     for (i = 0; i < 4; i++) {
  912.         q->matrix_coeff_index_prev[i] = 3;
  913.         q->matrix_coeff_index_now[i]  = 3;
  914.         q->matrix_coeff_index_next[i] = 3;
  915.     }
  916.  
  917.     ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
  918.     avpriv_float_dsp_init(&q->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
  919.     ff_fmt_convert_init(&q->fmt_conv, avctx);
  920.  
  921.     q->units = av_mallocz(sizeof(*q->units) * avctx->channels);
  922.     if (!q->units) {
  923.         atrac3_decode_close(avctx);
  924.         return AVERROR(ENOMEM);
  925.     }
  926.  
  927.     return 0;
  928. }
  929.  
  930. AVCodec ff_atrac3_decoder = {
  931.     .name             = "atrac3",
  932.     .long_name        = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
  933.     .type             = AVMEDIA_TYPE_AUDIO,
  934.     .id               = AV_CODEC_ID_ATRAC3,
  935.     .priv_data_size   = sizeof(ATRAC3Context),
  936.     .init             = atrac3_decode_init,
  937.     .close            = atrac3_decode_close,
  938.     .decode           = atrac3_decode_frame,
  939.     .capabilities     = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
  940.     .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
  941.                                                         AV_SAMPLE_FMT_NONE },
  942. };
  943.