LCOV - code coverage report
Current view: top level - libavcodec - atrac3.c (source / functions) Hit Total Coverage
Test: coverage.info Lines: 290 450 64.4 %
Date: 2017-12-17 04:34:43 Functions: 16 19 84.2 %

          Line data    Source code
       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 "get_bits.h"
      46             : #include "internal.h"
      47             : 
      48             : #include "atrac.h"
      49             : #include "atrac3data.h"
      50             : 
      51             : #define MIN_CHANNELS    1
      52             : #define MAX_CHANNELS    8
      53             : #define MAX_JS_PAIRS    8 / 2
      54             : 
      55             : #define JOINT_STEREO    0x12
      56             : #define SINGLE          0x2
      57             : 
      58             : #define SAMPLES_PER_FRAME 1024
      59             : #define MDCT_SIZE          512
      60             : 
      61             : typedef struct GainBlock {
      62             :     AtracGainInfo g_block[4];
      63             : } GainBlock;
      64             : 
      65             : typedef struct TonalComponent {
      66             :     int pos;
      67             :     int num_coefs;
      68             :     float coef[8];
      69             : } TonalComponent;
      70             : 
      71             : typedef struct ChannelUnit {
      72             :     int            bands_coded;
      73             :     int            num_components;
      74             :     float          prev_frame[SAMPLES_PER_FRAME];
      75             :     int            gc_blk_switch;
      76             :     TonalComponent components[64];
      77             :     GainBlock      gain_block[2];
      78             : 
      79             :     DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
      80             :     DECLARE_ALIGNED(32, float, imdct_buf)[SAMPLES_PER_FRAME];
      81             : 
      82             :     float          delay_buf1[46]; ///<qmf delay buffers
      83             :     float          delay_buf2[46];
      84             :     float          delay_buf3[46];
      85             : } ChannelUnit;
      86             : 
      87             : typedef struct ATRAC3Context {
      88             :     GetBitContext gb;
      89             :     //@{
      90             :     /** stream data */
      91             :     int coding_mode;
      92             : 
      93             :     ChannelUnit *units;
      94             :     //@}
      95             :     //@{
      96             :     /** joint-stereo related variables */
      97             :     int matrix_coeff_index_prev[MAX_JS_PAIRS][4];
      98             :     int matrix_coeff_index_now[MAX_JS_PAIRS][4];
      99             :     int matrix_coeff_index_next[MAX_JS_PAIRS][4];
     100             :     int weighting_delay[MAX_JS_PAIRS][6];
     101             :     //@}
     102             :     //@{
     103             :     /** data buffers */
     104             :     uint8_t *decoded_bytes_buffer;
     105             :     float temp_buf[1070];
     106             :     //@}
     107             :     //@{
     108             :     /** extradata */
     109             :     int scrambled_stream;
     110             :     //@}
     111             : 
     112             :     AtracGCContext    gainc_ctx;
     113             :     FFTContext        mdct_ctx;
     114             :     AVFloatDSPContext *fdsp;
     115             : } ATRAC3Context;
     116             : 
     117             : static DECLARE_ALIGNED(32, float, mdct_window)[MDCT_SIZE];
     118             : static VLC_TYPE atrac3_vlc_table[4096][2];
     119             : static VLC   spectral_coeff_tab[7];
     120             : 
     121             : /**
     122             :  * Regular 512 points IMDCT without overlapping, with the exception of the
     123             :  * swapping of odd bands caused by the reverse spectra of the QMF.
     124             :  *
     125             :  * @param odd_band  1 if the band is an odd band
     126             :  */
     127        2800 : static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
     128             : {
     129             :     int i;
     130             : 
     131        2800 :     if (odd_band) {
     132             :         /**
     133             :          * Reverse the odd bands before IMDCT, this is an effect of the QMF
     134             :          * transform or it gives better compression to do it this way.
     135             :          * FIXME: It should be possible to handle this in imdct_calc
     136             :          * for that to happen a modification of the prerotation step of
     137             :          * all SIMD code and C code is needed.
     138             :          * Or fix the functions before so they generate a pre reversed spectrum.
     139             :          */
     140      109392 :         for (i = 0; i < 128; i++)
     141      108544 :             FFSWAP(float, input[i], input[255 - i]);
     142             :     }
     143             : 
     144        2800 :     q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
     145             : 
     146             :     /* Perform windowing on the output. */
     147        2800 :     q->fdsp->vector_fmul(output, output, mdct_window, MDCT_SIZE);
     148        2800 : }
     149             : 
     150             : /*
     151             :  * indata descrambling, only used for data coming from the rm container
     152             :  */
     153           0 : static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
     154             : {
     155             :     int i, off;
     156             :     uint32_t c;
     157             :     const uint32_t *buf;
     158           0 :     uint32_t *output = (uint32_t *)out;
     159             : 
     160           0 :     off = (intptr_t)input & 3;
     161           0 :     buf = (const uint32_t *)(input - off);
     162           0 :     if (off)
     163           0 :         c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
     164             :     else
     165           0 :         c = av_be2ne32(0x537F6103U);
     166           0 :     bytes += 3 + off;
     167           0 :     for (i = 0; i < bytes / 4; i++)
     168           0 :         output[i] = c ^ buf[i];
     169             : 
     170           0 :     if (off)
     171           0 :         avpriv_request_sample(NULL, "Offset of %d", off);
     172             : 
     173           0 :     return off;
     174             : }
     175             : 
     176           4 : static av_cold void init_imdct_window(void)
     177             : {
     178             :     int i, j;
     179             : 
     180             :     /* generate the mdct window, for details see
     181             :      * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
     182         516 :     for (i = 0, j = 255; i < 128; i++, j--) {
     183         512 :         float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
     184         512 :         float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
     185         512 :         float w  = 0.5 * (wi * wi + wj * wj);
     186         512 :         mdct_window[i] = mdct_window[511 - i] = wi / w;
     187         512 :         mdct_window[j] = mdct_window[511 - j] = wj / w;
     188             :     }
     189           4 : }
     190             : 
     191           7 : static av_cold int atrac3_decode_close(AVCodecContext *avctx)
     192             : {
     193           7 :     ATRAC3Context *q = avctx->priv_data;
     194             : 
     195           7 :     av_freep(&q->units);
     196           7 :     av_freep(&q->decoded_bytes_buffer);
     197           7 :     av_freep(&q->fdsp);
     198             : 
     199           7 :     ff_mdct_end(&q->mdct_ctx);
     200             : 
     201           7 :     return 0;
     202             : }
     203             : 
     204             : /**
     205             :  * Mantissa decoding
     206             :  *
     207             :  * @param selector     which table the output values are coded with
     208             :  * @param coding_flag  constant length coding or variable length coding
     209             :  * @param mantissas    mantissa output table
     210             :  * @param num_codes    number of values to get
     211             :  */
     212       28209 : static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
     213             :                                        int coding_flag, int *mantissas,
     214             :                                        int num_codes)
     215             : {
     216             :     int i, code, huff_symb;
     217             : 
     218       28209 :     if (selector == 1)
     219       15517 :         num_codes /= 2;
     220             : 
     221       28209 :     if (coding_flag != 0) {
     222             :         /* constant length coding (CLC) */
     223           0 :         int num_bits = clc_length_tab[selector];
     224             : 
     225           0 :         if (selector > 1) {
     226           0 :             for (i = 0; i < num_codes; i++) {
     227           0 :                 if (num_bits)
     228           0 :                     code = get_sbits(gb, num_bits);
     229             :                 else
     230           0 :                     code = 0;
     231           0 :                 mantissas[i] = code;
     232             :             }
     233             :         } else {
     234           0 :             for (i = 0; i < num_codes; i++) {
     235           0 :                 if (num_bits)
     236           0 :                     code = get_bits(gb, num_bits); // num_bits is always 4 in this case
     237             :                 else
     238           0 :                     code = 0;
     239           0 :                 mantissas[i * 2    ] = mantissa_clc_tab[code >> 2];
     240           0 :                 mantissas[i * 2 + 1] = mantissa_clc_tab[code &  3];
     241             :             }
     242             :         }
     243             :     } else {
     244             :         /* variable length coding (VLC) */
     245       28209 :         if (selector != 1) {
     246      192516 :             for (i = 0; i < num_codes; i++) {
     247      179824 :                 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
     248      179824 :                                      spectral_coeff_tab[selector-1].bits, 3);
     249      179824 :                 huff_symb += 1;
     250      179824 :                 code = huff_symb >> 1;
     251      179824 :                 if (huff_symb & 1)
     252      139733 :                     code = -code;
     253      179824 :                 mantissas[i] = code;
     254             :             }
     255             :         } else {
     256      250405 :             for (i = 0; i < num_codes; i++) {
     257      234888 :                 huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
     258      234888 :                                      spectral_coeff_tab[selector - 1].bits, 3);
     259      234888 :                 mantissas[i * 2    ] = mantissa_vlc_tab[huff_symb * 2    ];
     260      234888 :                 mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
     261             :             }
     262             :         }
     263             :     }
     264       28209 : }
     265             : 
     266             : /**
     267             :  * Restore the quantized band spectrum coefficients
     268             :  *
     269             :  * @return subband count, fix for broken specification/files
     270             :  */
     271        1108 : static int decode_spectrum(GetBitContext *gb, float *output)
     272             : {
     273             :     int num_subbands, coding_mode, i, j, first, last, subband_size;
     274             :     int subband_vlc_index[32], sf_index[32];
     275             :     int mantissas[128];
     276             :     float scale_factor;
     277             : 
     278        1108 :     num_subbands = get_bits(gb, 5);  // number of coded subbands
     279        1108 :     coding_mode  = get_bits1(gb);    // coding Mode: 0 - VLC/ 1-CLC
     280             : 
     281             :     /* get the VLC selector table for the subbands, 0 means not coded */
     282       29317 :     for (i = 0; i <= num_subbands; i++)
     283       28209 :         subband_vlc_index[i] = get_bits(gb, 3);
     284             : 
     285             :     /* read the scale factor indexes from the stream */
     286       29317 :     for (i = 0; i <= num_subbands; i++) {
     287       28209 :         if (subband_vlc_index[i] != 0)
     288       28209 :             sf_index[i] = get_bits(gb, 6);
     289             :     }
     290             : 
     291       29317 :     for (i = 0; i <= num_subbands; i++) {
     292       28209 :         first = subband_tab[i    ];
     293       28209 :         last  = subband_tab[i + 1];
     294             : 
     295       28209 :         subband_size = last - first;
     296             : 
     297       28209 :         if (subband_vlc_index[i] != 0) {
     298             :             /* decode spectral coefficients for this subband */
     299             :             /* TODO: This can be done faster is several blocks share the
     300             :              * same VLC selector (subband_vlc_index) */
     301       28209 :             read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
     302             :                                        mantissas, subband_size);
     303             : 
     304             :             /* decode the scale factor for this subband */
     305       56418 :             scale_factor = ff_atrac_sf_table[sf_index[i]] *
     306       28209 :                            inv_max_quant[subband_vlc_index[i]];
     307             : 
     308             :             /* inverse quantize the coefficients */
     309      677809 :             for (j = 0; first < last; first++, j++)
     310      649600 :                 output[first] = mantissas[j] * scale_factor;
     311             :         } else {
     312             :             /* this subband was not coded, so zero the entire subband */
     313           0 :             memset(output + first, 0, subband_size * sizeof(*output));
     314             :         }
     315             :     }
     316             : 
     317             :     /* clear the subbands that were not coded */
     318        1108 :     first = subband_tab[i];
     319        1108 :     memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
     320        1108 :     return num_subbands;
     321             : }
     322             : 
     323             : /**
     324             :  * Restore the quantized tonal components
     325             :  *
     326             :  * @param components tonal components
     327             :  * @param num_bands  number of coded bands
     328             :  */
     329        1108 : static int decode_tonal_components(GetBitContext *gb,
     330             :                                    TonalComponent *components, int num_bands)
     331             : {
     332             :     int i, b, c, m;
     333             :     int nb_components, coding_mode_selector, coding_mode;
     334             :     int band_flags[4], mantissa[8];
     335        1108 :     int component_count = 0;
     336             : 
     337        1108 :     nb_components = get_bits(gb, 5);
     338             : 
     339             :     /* no tonal components */
     340        1108 :     if (nb_components == 0)
     341        1108 :         return 0;
     342             : 
     343           0 :     coding_mode_selector = get_bits(gb, 2);
     344           0 :     if (coding_mode_selector == 2)
     345           0 :         return AVERROR_INVALIDDATA;
     346             : 
     347           0 :     coding_mode = coding_mode_selector & 1;
     348             : 
     349           0 :     for (i = 0; i < nb_components; i++) {
     350             :         int coded_values_per_component, quant_step_index;
     351             : 
     352           0 :         for (b = 0; b <= num_bands; b++)
     353           0 :             band_flags[b] = get_bits1(gb);
     354             : 
     355           0 :         coded_values_per_component = get_bits(gb, 3);
     356             : 
     357           0 :         quant_step_index = get_bits(gb, 3);
     358           0 :         if (quant_step_index <= 1)
     359           0 :             return AVERROR_INVALIDDATA;
     360             : 
     361           0 :         if (coding_mode_selector == 3)
     362           0 :             coding_mode = get_bits1(gb);
     363             : 
     364           0 :         for (b = 0; b < (num_bands + 1) * 4; b++) {
     365             :             int coded_components;
     366             : 
     367           0 :             if (band_flags[b >> 2] == 0)
     368           0 :                 continue;
     369             : 
     370           0 :             coded_components = get_bits(gb, 3);
     371             : 
     372           0 :             for (c = 0; c < coded_components; c++) {
     373           0 :                 TonalComponent *cmp = &components[component_count];
     374             :                 int sf_index, coded_values, max_coded_values;
     375             :                 float scale_factor;
     376             : 
     377           0 :                 sf_index = get_bits(gb, 6);
     378           0 :                 if (component_count >= 64)
     379           0 :                     return AVERROR_INVALIDDATA;
     380             : 
     381           0 :                 cmp->pos = b * 64 + get_bits(gb, 6);
     382             : 
     383           0 :                 max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
     384           0 :                 coded_values     = coded_values_per_component + 1;
     385           0 :                 coded_values     = FFMIN(max_coded_values, coded_values);
     386             : 
     387           0 :                 scale_factor = ff_atrac_sf_table[sf_index] *
     388           0 :                                inv_max_quant[quant_step_index];
     389             : 
     390           0 :                 read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
     391             :                                            mantissa, coded_values);
     392             : 
     393           0 :                 cmp->num_coefs = coded_values;
     394             : 
     395             :                 /* inverse quant */
     396           0 :                 for (m = 0; m < coded_values; m++)
     397           0 :                     cmp->coef[m] = mantissa[m] * scale_factor;
     398             : 
     399           0 :                 component_count++;
     400             :             }
     401             :         }
     402             :     }
     403             : 
     404           0 :     return component_count;
     405             : }
     406             : 
     407             : /**
     408             :  * Decode gain parameters for the coded bands
     409             :  *
     410             :  * @param block      the gainblock for the current band
     411             :  * @param num_bands  amount of coded bands
     412             :  */
     413        1108 : static int decode_gain_control(GetBitContext *gb, GainBlock *block,
     414             :                                int num_bands)
     415             : {
     416             :     int b, j;
     417             :     int *level, *loc;
     418             : 
     419        1108 :     AtracGainInfo *gain = block->g_block;
     420             : 
     421        3912 :     for (b = 0; b <= num_bands; b++) {
     422        2804 :         gain[b].num_points = get_bits(gb, 3);
     423        2804 :         level              = gain[b].lev_code;
     424        2804 :         loc                = gain[b].loc_code;
     425             : 
     426        3678 :         for (j = 0; j < gain[b].num_points; j++) {
     427         874 :             level[j] = get_bits(gb, 4);
     428         874 :             loc[j]   = get_bits(gb, 5);
     429         874 :             if (j && loc[j] <= loc[j - 1])
     430           0 :                 return AVERROR_INVALIDDATA;
     431             :         }
     432             :     }
     433             : 
     434             :     /* Clear the unused blocks. */
     435        2736 :     for (; b < 4 ; b++)
     436        1628 :         gain[b].num_points = 0;
     437             : 
     438        1108 :     return 0;
     439             : }
     440             : 
     441             : /**
     442             :  * Combine the tonal band spectrum and regular band spectrum
     443             :  *
     444             :  * @param spectrum        output spectrum buffer
     445             :  * @param num_components  number of tonal components
     446             :  * @param components      tonal components for this band
     447             :  * @return                position of the last tonal coefficient
     448             :  */
     449        1108 : static int add_tonal_components(float *spectrum, int num_components,
     450             :                                 TonalComponent *components)
     451             : {
     452        1108 :     int i, j, last_pos = -1;
     453             :     float *input, *output;
     454             : 
     455        1108 :     for (i = 0; i < num_components; i++) {
     456           0 :         last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
     457           0 :         input    = components[i].coef;
     458           0 :         output   = &spectrum[components[i].pos];
     459             : 
     460           0 :         for (j = 0; j < components[i].num_coefs; j++)
     461           0 :             output[j] += input[j];
     462             :     }
     463             : 
     464        1108 :     return last_pos;
     465             : }
     466             : 
     467             : #define INTERPOLATE(old, new, nsample) \
     468             :     ((old) + (nsample) * 0.125 * ((new) - (old)))
     469             : 
     470         260 : static void reverse_matrixing(float *su1, float *su2, int *prev_code,
     471             :                               int *curr_code)
     472             : {
     473             :     int i, nsample, band;
     474             :     float mc1_l, mc1_r, mc2_l, mc2_r;
     475             : 
     476        1300 :     for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
     477        1040 :         int s1 = prev_code[i];
     478        1040 :         int s2 = curr_code[i];
     479        1040 :         nsample = band;
     480             : 
     481        1040 :         if (s1 != s2) {
     482             :             /* Selector value changed, interpolation needed. */
     483          38 :             mc1_l = matrix_coeffs[s1 * 2    ];
     484          38 :             mc1_r = matrix_coeffs[s1 * 2 + 1];
     485          38 :             mc2_l = matrix_coeffs[s2 * 2    ];
     486          38 :             mc2_r = matrix_coeffs[s2 * 2 + 1];
     487             : 
     488             :             /* Interpolation is done over the first eight samples. */
     489         342 :             for (; nsample < band + 8; nsample++) {
     490         304 :                 float c1 = su1[nsample];
     491         304 :                 float c2 = su2[nsample];
     492         608 :                 c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
     493         304 :                      c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
     494         304 :                 su1[nsample] = c2;
     495         304 :                 su2[nsample] = c1 * 2.0 - c2;
     496             :             }
     497             :         }
     498             : 
     499             :         /* Apply the matrix without interpolation. */
     500        1040 :         switch (s2) {
     501          45 :         case 0:     /* M/S decoding */
     502       11413 :             for (; nsample < band + 256; nsample++) {
     503       11368 :                 float c1 = su1[nsample];
     504       11368 :                 float c2 = su2[nsample];
     505       11368 :                 su1[nsample] =  c2       * 2.0;
     506       11368 :                 su2[nsample] = (c1 - c2) * 2.0;
     507             :             }
     508          45 :             break;
     509           0 :         case 1:
     510           0 :             for (; nsample < band + 256; nsample++) {
     511           0 :                 float c1 = su1[nsample];
     512           0 :                 float c2 = su2[nsample];
     513           0 :                 su1[nsample] = (c1 + c2) *  2.0;
     514           0 :                 su2[nsample] =  c2       * -2.0;
     515             :             }
     516           0 :             break;
     517         995 :         case 2:
     518             :         case 3:
     519      255563 :             for (; nsample < band + 256; nsample++) {
     520      254568 :                 float c1 = su1[nsample];
     521      254568 :                 float c2 = su2[nsample];
     522      254568 :                 su1[nsample] = c1 + c2;
     523      254568 :                 su2[nsample] = c1 - c2;
     524             :             }
     525         995 :             break;
     526        1040 :         default:
     527             :             av_assert1(0);
     528             :         }
     529             :     }
     530         260 : }
     531             : 
     532         492 : static void get_channel_weights(int index, int flag, float ch[2])
     533             : {
     534         492 :     if (index == 7) {
     535          53 :         ch[0] = 1.0;
     536          53 :         ch[1] = 1.0;
     537             :     } else {
     538         439 :         ch[0] = (index & 7) / 7.0;
     539         439 :         ch[1] = sqrt(2 - ch[0] * ch[0]);
     540         439 :         if (flag)
     541         115 :             FFSWAP(float, ch[0], ch[1]);
     542             :     }
     543         492 : }
     544             : 
     545         260 : static void channel_weighting(float *su1, float *su2, int *p3)
     546             : {
     547             :     int band, nsample;
     548             :     /* w[x][y] y=0 is left y=1 is right */
     549             :     float w[2][2];
     550             : 
     551         260 :     if (p3[1] != 7 || p3[3] != 7) {
     552         246 :         get_channel_weights(p3[1], p3[0], w[0]);
     553         246 :         get_channel_weights(p3[3], p3[2], w[1]);
     554             : 
     555         984 :         for (band = 256; band < 4 * 256; band += 256) {
     556        6642 :             for (nsample = band; nsample < band + 8; nsample++) {
     557        5904 :                 su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
     558        5904 :                 su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
     559             :             }
     560      183762 :             for(; nsample < band + 256; nsample++) {
     561      183024 :                 su1[nsample] *= w[1][0];
     562      183024 :                 su2[nsample] *= w[1][1];
     563             :             }
     564             :         }
     565             :     }
     566         260 : }
     567             : 
     568             : /**
     569             :  * Decode a Sound Unit
     570             :  *
     571             :  * @param snd           the channel unit to be used
     572             :  * @param output        the decoded samples before IQMF in float representation
     573             :  * @param channel_num   channel number
     574             :  * @param coding_mode   the coding mode (JOINT_STEREO or single channels)
     575             :  */
     576        1108 : static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb,
     577             :                                      ChannelUnit *snd, float *output,
     578             :                                      int channel_num, int coding_mode)
     579             : {
     580             :     int band, ret, num_subbands, last_tonal, num_bands;
     581        1108 :     GainBlock *gain1 = &snd->gain_block[    snd->gc_blk_switch];
     582        1108 :     GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
     583             : 
     584        1108 :     if (coding_mode == JOINT_STEREO && (channel_num % 2) == 1) {
     585         520 :         if (get_bits(gb, 2) != 3) {
     586           0 :             av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
     587           0 :             return AVERROR_INVALIDDATA;
     588             :         }
     589             :     } else {
     590         848 :         if (get_bits(gb, 6) != 0x28) {
     591           0 :             av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
     592           0 :             return AVERROR_INVALIDDATA;
     593             :         }
     594             :     }
     595             : 
     596             :     /* number of coded QMF bands */
     597        1108 :     snd->bands_coded = get_bits(gb, 2);
     598             : 
     599        1108 :     ret = decode_gain_control(gb, gain2, snd->bands_coded);
     600        1108 :     if (ret)
     601           0 :         return ret;
     602             : 
     603        1108 :     snd->num_components = decode_tonal_components(gb, snd->components,
     604             :                                                   snd->bands_coded);
     605        1108 :     if (snd->num_components < 0)
     606           0 :         return snd->num_components;
     607             : 
     608        1108 :     num_subbands = decode_spectrum(gb, snd->spectrum);
     609             : 
     610             :     /* Merge the decoded spectrum and tonal components. */
     611        1108 :     last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
     612        1108 :                                       snd->components);
     613             : 
     614             : 
     615             :     /* calculate number of used MLT/QMF bands according to the amount of coded
     616             :        spectral lines */
     617        1108 :     num_bands = (subband_tab[num_subbands] - 1) >> 8;
     618        1108 :     if (last_tonal >= 0)
     619           0 :         num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
     620             : 
     621             : 
     622             :     /* Reconstruct time domain samples. */
     623        5540 :     for (band = 0; band < 4; band++) {
     624             :         /* Perform the IMDCT step without overlapping. */
     625        4432 :         if (band <= num_bands)
     626        2800 :             imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
     627             :         else
     628        1632 :             memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
     629             : 
     630             :         /* gain compensation and overlapping */
     631        8864 :         ff_atrac_gain_compensation(&q->gainc_ctx, snd->imdct_buf,
     632        4432 :                                    &snd->prev_frame[band * 256],
     633             :                                    &gain1->g_block[band], &gain2->g_block[band],
     634             :                                    256, &output[band * 256]);
     635             :     }
     636             : 
     637             :     /* Swap the gain control buffers for the next frame. */
     638        1108 :     snd->gc_blk_switch ^= 1;
     639             : 
     640        1108 :     return 0;
     641             : }
     642             : 
     643         554 : static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
     644             :                         float **out_samples)
     645             : {
     646         554 :     ATRAC3Context *q = avctx->priv_data;
     647             :     int ret, i, ch;
     648             :     uint8_t *ptr1;
     649             : 
     650         554 :     if (q->coding_mode == JOINT_STEREO) {
     651             :         /* channel coupling mode */
     652             : 
     653             :         /* Decode sound unit pairs (channels are expected to be even).
     654             :          * Multichannel joint stereo interleaves pairs (6ch: 2ch + 2ch + 2ch) */
     655             :         const uint8_t *js_databuf;
     656             :         int js_pair, js_block_align;
     657             : 
     658         260 :         js_block_align = (avctx->block_align / avctx->channels) * 2; /* block pair */
     659             : 
     660         520 :         for (ch = 0; ch < avctx->channels; ch = ch + 2) {
     661         260 :             js_pair = ch/2;
     662         260 :             js_databuf = databuf + js_pair * js_block_align; /* align to current pair */
     663             : 
     664             :             /* Set the bitstream reader at the start of first channel sound unit. */
     665         260 :             init_get_bits(&q->gb,
     666             :                           js_databuf, js_block_align * 8);
     667             : 
     668             :             /* decode Sound Unit 1 */
     669         260 :             ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch],
     670         260 :                                             out_samples[ch], ch, JOINT_STEREO);
     671         260 :             if (ret != 0)
     672           0 :                 return ret;
     673             : 
     674             :             /* Framedata of the su2 in the joint-stereo mode is encoded in
     675             :              * reverse byte order so we need to swap it first. */
     676         260 :             if (js_databuf == q->decoded_bytes_buffer) {
     677           0 :                 uint8_t *ptr2 = q->decoded_bytes_buffer + js_block_align - 1;
     678           0 :                 ptr1          = q->decoded_bytes_buffer;
     679           0 :                 for (i = 0; i < js_block_align / 2; i++, ptr1++, ptr2--)
     680           0 :                     FFSWAP(uint8_t, *ptr1, *ptr2);
     681             :             } else {
     682         260 :                 const uint8_t *ptr2 = js_databuf + js_block_align - 1;
     683       50180 :                 for (i = 0; i < js_block_align; i++)
     684       49920 :                     q->decoded_bytes_buffer[i] = *ptr2--;
     685             :             }
     686             : 
     687             :             /* Skip the sync codes (0xF8). */
     688         260 :             ptr1 = q->decoded_bytes_buffer;
     689         260 :             for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
     690           0 :                 if (i >= js_block_align)
     691           0 :                     return AVERROR_INVALIDDATA;
     692             :             }
     693             : 
     694             : 
     695             :             /* set the bitstream reader at the start of the second Sound Unit */
     696         260 :             ret = init_get_bits8(&q->gb,
     697         260 :                            ptr1, q->decoded_bytes_buffer + js_block_align - ptr1);
     698         260 :             if (ret < 0)
     699           0 :                 return ret;
     700             : 
     701             :             /* Fill the Weighting coeffs delay buffer */
     702         260 :             memmove(q->weighting_delay[js_pair], &q->weighting_delay[js_pair][2],
     703             :                     4 * sizeof(*q->weighting_delay[js_pair]));
     704         260 :             q->weighting_delay[js_pair][4] = get_bits1(&q->gb);
     705         260 :             q->weighting_delay[js_pair][5] = get_bits(&q->gb, 3);
     706             : 
     707        1300 :             for (i = 0; i < 4; i++) {
     708        1040 :                 q->matrix_coeff_index_prev[js_pair][i] = q->matrix_coeff_index_now[js_pair][i];
     709        1040 :                 q->matrix_coeff_index_now[js_pair][i]  = q->matrix_coeff_index_next[js_pair][i];
     710        1040 :                 q->matrix_coeff_index_next[js_pair][i] = get_bits(&q->gb, 2);
     711             :             }
     712             : 
     713             :             /* Decode Sound Unit 2. */
     714         520 :             ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch+1],
     715         260 :                                             out_samples[ch+1], ch+1, JOINT_STEREO);
     716         260 :             if (ret != 0)
     717           0 :                 return ret;
     718             : 
     719             :             /* Reconstruct the channel coefficients. */
     720         260 :             reverse_matrixing(out_samples[ch], out_samples[ch+1],
     721         260 :                               q->matrix_coeff_index_prev[js_pair],
     722         260 :                               q->matrix_coeff_index_now[js_pair]);
     723             : 
     724         260 :             channel_weighting(out_samples[ch], out_samples[ch+1], q->weighting_delay[js_pair]);
     725             :         }
     726             :     } else {
     727             :         /* single channels */
     728             :         /* Decode the channel sound units. */
     729         882 :         for (i = 0; i < avctx->channels; i++) {
     730             :             /* Set the bitstream reader at the start of a channel sound unit. */
     731        1176 :             init_get_bits(&q->gb,
     732         588 :                           databuf + i * avctx->block_align / avctx->channels,
     733         588 :                           avctx->block_align * 8 / avctx->channels);
     734             : 
     735        1176 :             ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
     736         588 :                                             out_samples[i], i, q->coding_mode);
     737         588 :             if (ret != 0)
     738           0 :                 return ret;
     739             :         }
     740             :     }
     741             : 
     742             :     /* Apply the iQMF synthesis filter. */
     743        1662 :     for (i = 0; i < avctx->channels; i++) {
     744        1108 :         float *p1 = out_samples[i];
     745        1108 :         float *p2 = p1 + 256;
     746        1108 :         float *p3 = p2 + 256;
     747        1108 :         float *p4 = p3 + 256;
     748        1108 :         ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
     749        1108 :         ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
     750        1108 :         ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
     751             :     }
     752             : 
     753         554 :     return 0;
     754             : }
     755             : 
     756           0 : static int al_decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
     757             :                            int size, float **out_samples)
     758             : {
     759           0 :     ATRAC3Context *q = avctx->priv_data;
     760             :     int ret, i;
     761             : 
     762             :     /* Set the bitstream reader at the start of a channel sound unit. */
     763           0 :     init_get_bits(&q->gb, databuf, size * 8);
     764             :     /* single channels */
     765             :     /* Decode the channel sound units. */
     766           0 :     for (i = 0; i < avctx->channels; i++) {
     767           0 :         ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
     768           0 :                                         out_samples[i], i, q->coding_mode);
     769           0 :         if (ret != 0)
     770           0 :             return ret;
     771           0 :         while (i < avctx->channels && get_bits_left(&q->gb) > 6 && show_bits(&q->gb, 6) != 0x28) {
     772           0 :             skip_bits(&q->gb, 1);
     773             :         }
     774             :     }
     775             : 
     776             :     /* Apply the iQMF synthesis filter. */
     777           0 :     for (i = 0; i < avctx->channels; i++) {
     778           0 :         float *p1 = out_samples[i];
     779           0 :         float *p2 = p1 + 256;
     780           0 :         float *p3 = p2 + 256;
     781           0 :         float *p4 = p3 + 256;
     782           0 :         ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
     783           0 :         ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
     784           0 :         ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
     785             :     }
     786             : 
     787           0 :     return 0;
     788             : }
     789             : 
     790         557 : static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
     791             :                                int *got_frame_ptr, AVPacket *avpkt)
     792             : {
     793         557 :     AVFrame *frame     = data;
     794         557 :     const uint8_t *buf = avpkt->data;
     795         557 :     int buf_size = avpkt->size;
     796         557 :     ATRAC3Context *q = avctx->priv_data;
     797             :     int ret;
     798             :     const uint8_t *databuf;
     799             : 
     800         557 :     if (buf_size < avctx->block_align) {
     801           3 :         av_log(avctx, AV_LOG_ERROR,
     802             :                "Frame too small (%d bytes). Truncated file?\n", buf_size);
     803           3 :         return AVERROR_INVALIDDATA;
     804             :     }
     805             : 
     806             :     /* get output buffer */
     807         554 :     frame->nb_samples = SAMPLES_PER_FRAME;
     808         554 :     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
     809           0 :         return ret;
     810             : 
     811             :     /* Check if we need to descramble and what buffer to pass on. */
     812         554 :     if (q->scrambled_stream) {
     813           0 :         decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
     814           0 :         databuf = q->decoded_bytes_buffer;
     815             :     } else {
     816         554 :         databuf = buf;
     817             :     }
     818             : 
     819         554 :     ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
     820         554 :     if (ret) {
     821           0 :         av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
     822           0 :         return ret;
     823             :     }
     824             : 
     825         554 :     *got_frame_ptr = 1;
     826             : 
     827         554 :     return avctx->block_align;
     828             : }
     829             : 
     830           0 : static int atrac3al_decode_frame(AVCodecContext *avctx, void *data,
     831             :                                  int *got_frame_ptr, AVPacket *avpkt)
     832             : {
     833           0 :     AVFrame *frame = data;
     834             :     int ret;
     835             : 
     836           0 :     frame->nb_samples = SAMPLES_PER_FRAME;
     837           0 :     if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
     838           0 :         return ret;
     839             : 
     840           0 :     ret = al_decode_frame(avctx, avpkt->data, avpkt->size,
     841           0 :                           (float **)frame->extended_data);
     842           0 :     if (ret) {
     843           0 :         av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
     844           0 :         return ret;
     845             :     }
     846             : 
     847           0 :     *got_frame_ptr = 1;
     848             : 
     849           0 :     return avpkt->size;
     850             : }
     851             : 
     852           4 : static av_cold void atrac3_init_static_data(void)
     853             : {
     854             :     int i;
     855             : 
     856           4 :     init_imdct_window();
     857           4 :     ff_atrac_generate_tables();
     858             : 
     859             :     /* Initialize the VLC tables. */
     860          32 :     for (i = 0; i < 7; i++) {
     861          28 :         spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
     862          56 :         spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
     863          28 :                                                 atrac3_vlc_offs[i    ];
     864          28 :         init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
     865             :                  huff_bits[i],  1, 1,
     866             :                  huff_codes[i], 1, 1, INIT_VLC_USE_NEW_STATIC);
     867             :     }
     868           4 : }
     869             : 
     870           7 : static av_cold int atrac3_decode_init(AVCodecContext *avctx)
     871             : {
     872             :     static int static_init_done;
     873             :     int i, js_pair, ret;
     874             :     int version, delay, samples_per_frame, frame_factor;
     875           7 :     const uint8_t *edata_ptr = avctx->extradata;
     876           7 :     ATRAC3Context *q = avctx->priv_data;
     877             : 
     878           7 :     if (avctx->channels < MIN_CHANNELS || avctx->channels > MAX_CHANNELS) {
     879           0 :         av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
     880           0 :         return AVERROR(EINVAL);
     881             :     }
     882             : 
     883           7 :     if (!static_init_done)
     884           4 :         atrac3_init_static_data();
     885           7 :     static_init_done = 1;
     886             : 
     887             :     /* Take care of the codec-specific extradata. */
     888           7 :     if (avctx->codec_id == AV_CODEC_ID_ATRAC3AL) {
     889           0 :         version           = 4;
     890           0 :         samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
     891           0 :         delay             = 0x88E;
     892           0 :         q->coding_mode    = SINGLE;
     893           7 :     } else if (avctx->extradata_size == 14) {
     894             :         /* Parse the extradata, WAV format */
     895           7 :         av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
     896             :                bytestream_get_le16(&edata_ptr));  // Unknown value always 1
     897           7 :         edata_ptr += 4;                             // samples per channel
     898           7 :         q->coding_mode = bytestream_get_le16(&edata_ptr);
     899           7 :         av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
     900             :                bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
     901           7 :         frame_factor = bytestream_get_le16(&edata_ptr);  // Unknown always 1
     902           7 :         av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
     903             :                bytestream_get_le16(&edata_ptr));  // Unknown always 0
     904             : 
     905             :         /* setup */
     906           7 :         samples_per_frame    = SAMPLES_PER_FRAME * avctx->channels;
     907           7 :         version              = 4;
     908           7 :         delay                = 0x88E;
     909           7 :         q->coding_mode       = q->coding_mode ? JOINT_STEREO : SINGLE;
     910           7 :         q->scrambled_stream  = 0;
     911             : 
     912          12 :         if (avctx->block_align !=  96 * avctx->channels * frame_factor &&
     913           8 :             avctx->block_align != 152 * avctx->channels * frame_factor &&
     914           3 :             avctx->block_align != 192 * avctx->channels * frame_factor) {
     915           0 :             av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
     916             :                    "configuration %d/%d/%d\n", avctx->block_align,
     917             :                    avctx->channels, frame_factor);
     918           0 :             return AVERROR_INVALIDDATA;
     919             :         }
     920           0 :     } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
     921             :         /* Parse the extradata, RM format. */
     922           0 :         version                = bytestream_get_be32(&edata_ptr);
     923           0 :         samples_per_frame      = bytestream_get_be16(&edata_ptr);
     924           0 :         delay                  = bytestream_get_be16(&edata_ptr);
     925           0 :         q->coding_mode         = bytestream_get_be16(&edata_ptr);
     926           0 :         q->scrambled_stream    = 1;
     927             : 
     928             :     } else {
     929           0 :         av_log(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n",
     930             :                avctx->extradata_size);
     931           0 :         return AVERROR(EINVAL);
     932             :     }
     933             : 
     934             :     /* Check the extradata */
     935             : 
     936           7 :     if (version != 4) {
     937           0 :         av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
     938           0 :         return AVERROR_INVALIDDATA;
     939             :     }
     940             : 
     941           7 :     if (samples_per_frame != SAMPLES_PER_FRAME * avctx->channels) {
     942           0 :         av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
     943             :                samples_per_frame);
     944           0 :         return AVERROR_INVALIDDATA;
     945             :     }
     946             : 
     947           7 :     if (delay != 0x88E) {
     948           0 :         av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
     949             :                delay);
     950           0 :         return AVERROR_INVALIDDATA;
     951             :     }
     952             : 
     953           7 :     if (q->coding_mode == SINGLE)
     954           5 :         av_log(avctx, AV_LOG_DEBUG, "Single channels detected.\n");
     955           2 :     else if (q->coding_mode == JOINT_STEREO) {
     956           2 :         if (avctx->channels % 2 == 1) { /* Joint stereo channels must be even */
     957           0 :             av_log(avctx, AV_LOG_ERROR, "Invalid joint stereo channel configuration.\n");
     958           0 :             return AVERROR_INVALIDDATA;
     959             :         }
     960           2 :         av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
     961             :     } else {
     962           0 :         av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
     963             :                q->coding_mode);
     964           0 :         return AVERROR_INVALIDDATA;
     965             :     }
     966             : 
     967           7 :     if (avctx->block_align >= UINT_MAX / 2)
     968           0 :         return AVERROR(EINVAL);
     969             : 
     970           7 :     q->decoded_bytes_buffer = av_mallocz(FFALIGN(avctx->block_align, 4) +
     971             :                                          AV_INPUT_BUFFER_PADDING_SIZE);
     972           7 :     if (!q->decoded_bytes_buffer)
     973           0 :         return AVERROR(ENOMEM);
     974             : 
     975           7 :     avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
     976             : 
     977             :     /* initialize the MDCT transform */
     978           7 :     if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
     979           0 :         av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
     980           0 :         av_freep(&q->decoded_bytes_buffer);
     981           0 :         return ret;
     982             :     }
     983             : 
     984             :     /* init the joint-stereo decoding data */
     985          35 :     for (js_pair = 0; js_pair < MAX_JS_PAIRS; js_pair++) {
     986          28 :         q->weighting_delay[js_pair][0] = 0;
     987          28 :         q->weighting_delay[js_pair][1] = 7;
     988          28 :         q->weighting_delay[js_pair][2] = 0;
     989          28 :         q->weighting_delay[js_pair][3] = 7;
     990          28 :         q->weighting_delay[js_pair][4] = 0;
     991          28 :         q->weighting_delay[js_pair][5] = 7;
     992             : 
     993         140 :         for (i = 0; i < 4; i++) {
     994         112 :             q->matrix_coeff_index_prev[js_pair][i] = 3;
     995         112 :             q->matrix_coeff_index_now[js_pair][i]  = 3;
     996         112 :             q->matrix_coeff_index_next[js_pair][i] = 3;
     997             :         }
     998             :     }
     999             : 
    1000           7 :     ff_atrac_init_gain_compensation(&q->gainc_ctx, 4, 3);
    1001           7 :     q->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
    1002             : 
    1003           7 :     q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
    1004           7 :     if (!q->units || !q->fdsp) {
    1005           0 :         atrac3_decode_close(avctx);
    1006           0 :         return AVERROR(ENOMEM);
    1007             :     }
    1008             : 
    1009           7 :     return 0;
    1010             : }
    1011             : 
    1012             : AVCodec ff_atrac3_decoder = {
    1013             :     .name             = "atrac3",
    1014             :     .long_name        = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
    1015             :     .type             = AVMEDIA_TYPE_AUDIO,
    1016             :     .id               = AV_CODEC_ID_ATRAC3,
    1017             :     .priv_data_size   = sizeof(ATRAC3Context),
    1018             :     .init             = atrac3_decode_init,
    1019             :     .close            = atrac3_decode_close,
    1020             :     .decode           = atrac3_decode_frame,
    1021             :     .capabilities     = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
    1022             :     .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
    1023             :                                                         AV_SAMPLE_FMT_NONE },
    1024             : };
    1025             : 
    1026             : AVCodec ff_atrac3al_decoder = {
    1027             :     .name             = "atrac3al",
    1028             :     .long_name        = NULL_IF_CONFIG_SMALL("ATRAC3 AL (Adaptive TRansform Acoustic Coding 3 Advanced Lossless)"),
    1029             :     .type             = AVMEDIA_TYPE_AUDIO,
    1030             :     .id               = AV_CODEC_ID_ATRAC3AL,
    1031             :     .priv_data_size   = sizeof(ATRAC3Context),
    1032             :     .init             = atrac3_decode_init,
    1033             :     .close            = atrac3_decode_close,
    1034             :     .decode           = atrac3al_decode_frame,
    1035             :     .capabilities     = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
    1036             :     .sample_fmts      = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
    1037             :                                                         AV_SAMPLE_FMT_NONE },
    1038             : };

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