falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
13796 Commits
32 Branches
322MB
Tree: f61d45c99e
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from 'f61d45c99e'
${ noResults }
FFmpeg/libavcodec/atrac3.c

1077 lines
32KB
Raw Blame History 

  1. /*

  2.  * Atrac 3 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 atrac3.c

  25.  * Atrac 3 compatible decoder.

  26.  * This decoder handles Sony's ATRAC3 data.

  27.  *

  28.  * Container formats used to store atrac 3 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 "avcodec.h"

  40. #include "bitstream.h"

  41. #include "dsputil.h"

  42. #include "bytestream.h"

  43. 

  44. #include "atrac3data.h"

  45. 

  46. #define JOINT_STEREO    0x12

  47. #define STEREO          0x2

  48. 

  49. 

  50. /* These structures are needed to store the parsed gain control data. */

  51. typedef struct {

  52.     int   num_gain_data;

  53.     int   levcode[8];

  54.     int   loccode[8];

  55. } gain_info;

  56. 

  57. typedef struct {

  58.     gain_info   gBlock[4];

  59. } gain_block;

  60. 

  61. typedef struct {

  62.     int     pos;

  63.     int     numCoefs;

  64.     float   coef[8];

  65. } tonal_component;

  66. 

  67. typedef struct {

  68.     int               bandsCoded;

  69.     int               numComponents;

  70.     tonal_component   components[64];

  71.     float             prevFrame[1024];

  72.     int               gcBlkSwitch;

  73.     gain_block        gainBlock[2];

  74. 

  75.     DECLARE_ALIGNED_16(float, spectrum[1024]);

  76.     DECLARE_ALIGNED_16(float, IMDCT_buf[1024]);

  77. 

  78.     float             delayBuf1[46]; ///<qmf delay buffers

  79.     float             delayBuf2[46];

  80.     float             delayBuf3[46];

  81. } channel_unit;

  82. 

  83. typedef struct {

  84.     GetBitContext       gb;

  85.     //@{

  86.     /** stream data */

  87.     int                 channels;

  88.     int                 codingMode;

  89.     int                 bit_rate;

  90.     int                 sample_rate;

  91.     int                 samples_per_channel;

  92.     int                 samples_per_frame;

  93. 

  94.     int                 bits_per_frame;

  95.     int                 bytes_per_frame;

  96.     int                 pBs;

  97.     channel_unit*       pUnits;

  98.     //@}

  99.     //@{

  100.     /** joint-stereo related variables */

  101.     int                 matrix_coeff_index_prev[4];

  102.     int                 matrix_coeff_index_now[4];

  103.     int                 matrix_coeff_index_next[4];

  104.     int                 weighting_delay[6];

  105.     //@}

  106.     //@{

  107.     /** data buffers */

  108.     float               outSamples[2048];

  109.     uint8_t*            decoded_bytes_buffer;

  110.     float               tempBuf[1070];

  111.     DECLARE_ALIGNED_16(float,mdct_tmp[512]);

  112.     //@}

  113.     //@{

  114.     /** extradata */

  115.     int                 atrac3version;

  116.     int                 delay;

  117.     int                 scrambled_stream;

  118.     int                 frame_factor;

  119.     //@}

  120. } ATRAC3Context;

  121. 

  122. static DECLARE_ALIGNED_16(float,mdct_window[512]);

  123. static float            qmf_window[48];

  124. static VLC              spectral_coeff_tab[7];

  125. static float            SFTable[64];

  126. static float            gain_tab1[16];

  127. static float            gain_tab2[31];

  128. static MDCTContext      mdct_ctx;

  129. static DSPContext       dsp;

  130. 

  131. 

  132. /* quadrature mirror synthesis filter */

  133. 

  134. /**

  135.  * Quadrature mirror synthesis filter.

  136.  *

  137.  * @param inlo      lower part of spectrum

  138.  * @param inhi      higher part of spectrum

  139.  * @param nIn       size of spectrum buffer

  140.  * @param pOut      out buffer

  141.  * @param delayBuf  delayBuf buffer

  142.  * @param temp      temp buffer

  143.  */

  144. 

  145. 

  146. static void iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)

  147. {

  148.     int   i, j;

  149.     float   *p1, *p3;

  150. 

  151.     memcpy(temp, delayBuf, 46*sizeof(float));

  152. 

  153.     p3 = temp + 46;

  154. 

  155.     /* loop1 */

  156.     for(i=0; i<nIn; i+=2){

  157.         p3[2*i+0] = inlo[i  ] + inhi[i  ];

  158.         p3[2*i+1] = inlo[i  ] - inhi[i  ];

  159.         p3[2*i+2] = inlo[i+1] + inhi[i+1];

  160.         p3[2*i+3] = inlo[i+1] - inhi[i+1];

  161.     }

  162. 

  163.     /* loop2 */

  164.     p1 = temp;

  165.     for (j = nIn; j != 0; j--) {

  166.         float s1 = 0.0;

  167.         float s2 = 0.0;

  168. 

  169.         for (i = 0; i < 48; i += 2) {

  170.             s1 += p1[i] * qmf_window[i];

  171.             s2 += p1[i+1] * qmf_window[i+1];

  172.         }

  173. 

  174.         pOut[0] = s2;

  175.         pOut[1] = s1;

  176. 

  177.         p1 += 2;

  178.         pOut += 2;

  179.     }

  180. 

  181.     /* Update the delay buffer. */

  182.     memcpy(delayBuf, temp + nIn*2, 46*sizeof(float));

  183. }

  184. 

  185. /**

  186.  * Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands

  187.  * caused by the reverse spectra of the QMF.

  188.  *

  189.  * @param pInput    float input

  190.  * @param pOutput   float output

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

  192.  * @param mdct_tmp  aligned temporary buffer for the mdct

  193.  */

  194. 

  195. static void IMLT(float *pInput, float *pOutput, int odd_band, float* mdct_tmp)

  196. {

  197.     int     i;

  198. 

  199.     if (odd_band) {

  200.         /**

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

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

  203.         * FIXME: It should be possible to handle this in ff_imdct_calc

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

  205.         * all SIMD code and C code is needed.

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

  207.         */

  208. 

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

  210.             FFSWAP(float, pInput[i], pInput[255-i]);

  211.     }

  212. 

  213.     mdct_ctx.fft.imdct_calc(&mdct_ctx,pOutput,pInput,mdct_tmp);

  214. 

  215.     /* Perform windowing on the output. */

  216.     dsp.vector_fmul(pOutput,mdct_window,512);

  217. 

  218. }

  219. 

  220. 

  221. /**

  222.  * Atrac 3 indata descrambling, only used for data coming from the rm container

  223.  *

  224.  * @param in        pointer to 8 bit array of indata

  225.  * @param bits      amount of bits

  226.  * @param out       pointer to 8 bit array of outdata

  227.  */

  228. 

  229. static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){

  230.     int i, off;

  231.     uint32_t c;

  232.     const uint32_t* buf;

  233.     uint32_t* obuf = (uint32_t*) out;

  234. 

  235.     off = (int)((long)inbuffer & 3);

  236.     buf = (const uint32_t*) (inbuffer - off);

  237.     c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));

  238.     bytes += 3 + off;

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

  240.         obuf[i] = c ^ buf[i];

  241. 

  242.     if (off)

  243.         av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);

  244. 

  245.     return off;

  246. }

  247. 

  248. 

  249. static void init_atrac3_transforms(ATRAC3Context *q) {

  250.     float enc_window[256];

  251.     float s;

  252.     int i;

  253. 

  254.     /* Generate the mdct window, for details see

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

  256.     for (i=0 ; i<256; i++)

  257.         enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;

  258. 

  259.     if (!mdct_window[0])

  260.         for (i=0 ; i<256; i++) {

  261.             mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);

  262.             mdct_window[511-i] = mdct_window[i];

  263.         }

  264. 

  265.     /* Generate the QMF window. */

  266.     for (i=0 ; i<24; i++) {

  267.         s = qmf_48tap_half[i] * 2.0;

  268.         qmf_window[i] = s;

  269.         qmf_window[47 - i] = s;

  270.     }

  271. 

  272.     /* Initialize the MDCT transform. */

  273.     ff_mdct_init(&mdct_ctx, 9, 1);

  274. }

  275. 

  276. /**

  277.  * Atrac3 uninit, free all allocated memory

  278.  */

  279. 

  280. static int atrac3_decode_close(AVCodecContext *avctx)

  281. {

  282.     ATRAC3Context *q = avctx->priv_data;

  283. 

  284.     av_free(q->pUnits);

  285.     av_free(q->decoded_bytes_buffer);

  286. 

  287.     return 0;

  288. }

  289. 

  290. /**

  291. / * Mantissa decoding

  292.  *

  293.  * @param gb            the GetBit context

  294.  * @param selector      what table is the output values coded with

  295.  * @param codingFlag    constant length coding or variable length coding

  296.  * @param mantissas     mantissa output table

  297.  * @param numCodes      amount of values to get

  298.  */

  299. 

  300. static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)

  301. {

  302.     int   numBits, cnt, code, huffSymb;

  303. 

  304.     if (selector == 1)

  305.         numCodes /= 2;

  306. 

  307.     if (codingFlag != 0) {

  308.         /* constant length coding (CLC) */

  309.         //FIXME we don't have any samples coded in CLC mode

  310.         numBits = CLCLengthTab[selector];

  311. 

  312.         if (selector > 1) {

  313.             for (cnt = 0; cnt < numCodes; cnt++) {

  314.                 if (numBits)

  315.                     code = get_sbits(gb, numBits);

  316.                 else

  317.                     code = 0;

  318.                 mantissas[cnt] = code;

  319.             }

  320.         } else {

  321.             for (cnt = 0; cnt < numCodes; cnt++) {

  322.                 if (numBits)

  323.                     code = get_bits(gb, numBits); //numBits is always 4 in this case

  324.                 else

  325.                     code = 0;

  326.                 mantissas[cnt*2] = seTab_0[code >> 2];

  327.                 mantissas[cnt*2+1] = seTab_0[code & 3];

  328.             }

  329.         }

  330.     } else {

  331.         /* variable length coding (VLC) */

  332.         if (selector != 1) {

  333.             for (cnt = 0; cnt < numCodes; cnt++) {

  334.                 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);

  335.                 huffSymb += 1;

  336.                 code = huffSymb >> 1;

  337.                 if (huffSymb & 1)

  338.                     code = -code;

  339.                 mantissas[cnt] = code;

  340.             }

  341.         } else {

  342.             for (cnt = 0; cnt < numCodes; cnt++) {

  343.                 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);

  344.                 mantissas[cnt*2] = decTable1[huffSymb*2];

  345.                 mantissas[cnt*2+1] = decTable1[huffSymb*2+1];

  346.             }

  347.         }

  348.     }

  349. }

  350. 

  351. /**

  352.  * Restore the quantized band spectrum coefficients

  353.  *

  354.  * @param gb            the GetBit context

  355.  * @param pOut          decoded band spectrum

  356.  * @return outSubbands   subband counter, fix for broken specification/files

  357.  */

  358. 

  359. static int decodeSpectrum (GetBitContext *gb, float *pOut)

  360. {

  361.     int   numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;

  362.     int   subband_vlc_index[32], SF_idxs[32];

  363.     int   mantissas[128];

  364.     float SF;

  365. 

  366.     numSubbands = get_bits(gb, 5); // number of coded subbands

  367.     codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC

  368. 

  369.     /* Get the VLC selector table for the subbands, 0 means not coded. */

  370.     for (cnt = 0; cnt <= numSubbands; cnt++)

  371.         subband_vlc_index[cnt] = get_bits(gb, 3);

  372. 

  373.     /* Read the scale factor indexes from the stream. */

  374.     for (cnt = 0; cnt <= numSubbands; cnt++) {

  375.         if (subband_vlc_index[cnt] != 0)

  376.             SF_idxs[cnt] = get_bits(gb, 6);

  377.     }

  378. 

  379.     for (cnt = 0; cnt <= numSubbands; cnt++) {

  380.         first = subbandTab[cnt];

  381.         last = subbandTab[cnt+1];

  382. 

  383.         subbWidth = last - first;

  384. 

  385.         if (subband_vlc_index[cnt] != 0) {

  386.             /* Decode spectral coefficients for this subband. */

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

  388.              * same VLC selector (subband_vlc_index) */

  389.             readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);

  390. 

  391.             /* Decode the scale factor for this subband. */

  392.             SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];

  393. 

  394.             /* Inverse quantize the coefficients. */

  395.             for (pIn=mantissas ; first<last; first++, pIn++)

  396.                 pOut[first] = *pIn * SF;

  397.         } else {

  398.             /* This subband was not coded, so zero the entire subband. */

  399.             memset(pOut+first, 0, subbWidth*sizeof(float));

  400.         }

  401.     }

  402. 

  403.     /* Clear the subbands that were not coded. */

  404.     first = subbandTab[cnt];

  405.     memset(pOut+first, 0, (1024 - first) * sizeof(float));

  406.     return numSubbands;

  407. }

  408. 

  409. /**

  410.  * Restore the quantized tonal components

  411.  *

  412.  * @param gb            the GetBit context

  413.  * @param pComponent    tone component

  414.  * @param numBands      amount of coded bands

  415.  */

  416. 

  417. static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)

  418. {

  419.     int i,j,k,cnt;

  420.     int   components, coding_mode_selector, coding_mode, coded_values_per_component;

  421.     int   sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;

  422.     int   band_flags[4], mantissa[8];

  423.     float  *pCoef;

  424.     float  scalefactor;

  425.     int   component_count = 0;

  426. 

  427.     components = get_bits(gb,5);

  428. 

  429.     /* no tonal components */

  430.     if (components == 0)

  431.         return 0;

  432. 

  433.     coding_mode_selector = get_bits(gb,2);

  434.     if (coding_mode_selector == 2)

  435.         return -1;

  436. 

  437.     coding_mode = coding_mode_selector & 1;

  438. 

  439.     for (i = 0; i < components; i++) {

  440.         for (cnt = 0; cnt <= numBands; cnt++)

  441.             band_flags[cnt] = get_bits1(gb);

  442. 

  443.         coded_values_per_component = get_bits(gb,3);

  444. 

  445.         quant_step_index = get_bits(gb,3);

  446.         if (quant_step_index <= 1)

  447.             return -1;

  448. 

  449.         if (coding_mode_selector == 3)

  450.             coding_mode = get_bits1(gb);

  451. 

  452.         for (j = 0; j < (numBands + 1) * 4; j++) {

  453.             if (band_flags[j >> 2] == 0)

  454.                 continue;

  455. 

  456.             coded_components = get_bits(gb,3);

  457. 

  458.             for (k=0; k<coded_components; k++) {

  459.                 sfIndx = get_bits(gb,6);

  460.                 pComponent[component_count].pos = j * 64 + (get_bits(gb,6));

  461.                 max_coded_values = 1024 - pComponent[component_count].pos;

  462.                 coded_values = coded_values_per_component + 1;

  463.                 coded_values = FFMIN(max_coded_values,coded_values);

  464. 

  465.                 scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index];

  466. 

  467.                 readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);

  468. 

  469.                 pComponent[component_count].numCoefs = coded_values;

  470. 

  471.                 /* inverse quant */

  472.                 pCoef = pComponent[component_count].coef;

  473.                 for (cnt = 0; cnt < coded_values; cnt++)

  474.                     pCoef[cnt] = mantissa[cnt] * scalefactor;

  475. 

  476.                 component_count++;

  477.             }

  478.         }

  479.     }

  480. 

  481.     return component_count;

  482. }

  483. 

  484. /**

  485.  * Decode gain parameters for the coded bands

  486.  *

  487.  * @param gb            the GetBit context

  488.  * @param pGb           the gainblock for the current band

  489.  * @param numBands      amount of coded bands

  490.  */

  491. 

  492. static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)

  493. {

  494.     int   i, cf, numData;

  495.     int   *pLevel, *pLoc;

  496. 

  497.     gain_info   *pGain = pGb->gBlock;

  498. 

  499.     for (i=0 ; i<=numBands; i++)

  500.     {

  501.         numData = get_bits(gb,3);

  502.         pGain[i].num_gain_data = numData;

  503.         pLevel = pGain[i].levcode;

  504.         pLoc = pGain[i].loccode;

  505. 

  506.         for (cf = 0; cf < numData; cf++){

  507.             pLevel[cf]= get_bits(gb,4);

  508.             pLoc  [cf]= get_bits(gb,5);

  509.             if(cf && pLoc[cf] <= pLoc[cf-1])

  510.                 return -1;

  511.         }

  512.     }

  513. 

  514.     /* Clear the unused blocks. */

  515.     for (; i<4 ; i++)

  516.         pGain[i].num_gain_data = 0;

  517. 

  518.     return 0;

  519. }

  520. 

  521. /**

  522.  * Apply gain parameters and perform the MDCT overlapping part

  523.  *

  524.  * @param pIn           input float buffer

  525.  * @param pPrev         previous float buffer to perform overlap against

  526.  * @param pOut          output float buffer

  527.  * @param pGain1        current band gain info

  528.  * @param pGain2        next band gain info

  529.  */

  530. 

  531. static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)

  532. {

  533.     /* gain compensation function */

  534.     float  gain1, gain2, gain_inc;

  535.     int   cnt, numdata, nsample, startLoc, endLoc;

  536. 

  537. 

  538.     if (pGain2->num_gain_data == 0)

  539.         gain1 = 1.0;

  540.     else

  541.         gain1 = gain_tab1[pGain2->levcode[0]];

  542. 

  543.     if (pGain1->num_gain_data == 0) {

  544.         for (cnt = 0; cnt < 256; cnt++)

  545.             pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];

  546.     } else {

  547.         numdata = pGain1->num_gain_data;

  548.         pGain1->loccode[numdata] = 32;

  549.         pGain1->levcode[numdata] = 4;

  550. 

  551.         nsample = 0; // current sample = 0

  552. 

  553.         for (cnt = 0; cnt < numdata; cnt++) {

  554.             startLoc = pGain1->loccode[cnt] * 8;

  555.             endLoc = startLoc + 8;

  556. 

  557.             gain2 = gain_tab1[pGain1->levcode[cnt]];

  558.             gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];

  559. 

  560.             /* interpolate */

  561.             for (; nsample < startLoc; nsample++)

  562.                 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;

  563. 

  564.             /* interpolation is done over eight samples */

  565.             for (; nsample < endLoc; nsample++) {

  566.                 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;

  567.                 gain2 *= gain_inc;

  568.             }

  569.         }

  570. 

  571.         for (; nsample < 256; nsample++)

  572.             pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];

  573.     }

  574. 

  575.     /* Delay for the overlapping part. */

  576.     memcpy(pPrev, &pIn[256], 256*sizeof(float));

  577. }

  578. 

  579. /**

  580.  * Combine the tonal band spectrum and regular band spectrum

  581.  * Return position of the last tonal coefficient

  582.  *

  583.  * @param pSpectrum     output spectrum buffer

  584.  * @param numComponents amount of tonal components

  585.  * @param pComponent    tonal components for this band

  586.  */

  587. 

  588. static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)

  589. {

  590.     int   cnt, i, lastPos = -1;

  591.     float   *pIn, *pOut;

  592. 

  593.     for (cnt = 0; cnt < numComponents; cnt++){

  594.         lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);

  595.         pIn = pComponent[cnt].coef;

  596.         pOut = &(pSpectrum[pComponent[cnt].pos]);

  597. 

  598.         for (i=0 ; i<pComponent[cnt].numCoefs ; i++)

  599.             pOut[i] += pIn[i];

  600.     }

  601. 

  602.     return lastPos;

  603. }

  604. 

  605. 

  606. #define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))

  607. 

  608. static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)

  609. {

  610.     int    i, band, nsample, s1, s2;

  611.     float    c1, c2;

  612.     float    mc1_l, mc1_r, mc2_l, mc2_r;

  613. 

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

  615.         s1 = pPrevCode[i];

  616.         s2 = pCurrCode[i];

  617.         nsample = 0;

  618. 

  619.         if (s1 != s2) {

  620.             /* Selector value changed, interpolation needed. */

  621.             mc1_l = matrixCoeffs[s1*2];

  622.             mc1_r = matrixCoeffs[s1*2+1];

  623.             mc2_l = matrixCoeffs[s2*2];

  624.             mc2_r = matrixCoeffs[s2*2+1];

  625. 

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

  627.             for(; nsample < 8; nsample++) {

  628.                 c1 = su1[band+nsample];

  629.                 c2 = su2[band+nsample];

  630.                 c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);

  631.                 su1[band+nsample] = c2;

  632.                 su2[band+nsample] = c1 * 2.0 - c2;

  633.             }

  634.         }

  635. 

  636.         /* Apply the matrix without interpolation. */

  637.         switch (s2) {

  638.             case 0:     /* M/S decoding */

  639.                 for (; nsample < 256; nsample++) {

  640.                     c1 = su1[band+nsample];

  641.                     c2 = su2[band+nsample];

  642.                     su1[band+nsample] = c2 * 2.0;

  643.                     su2[band+nsample] = (c1 - c2) * 2.0;

  644.                 }

  645.                 break;

  646. 

  647.             case 1:

  648.                 for (; nsample < 256; nsample++) {

  649.                     c1 = su1[band+nsample];

  650.                     c2 = su2[band+nsample];

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

  652.                     su2[band+nsample] = c2 * -2.0;

  653.                 }

  654.                 break;

  655.             case 2:

  656.             case 3:

  657.                 for (; nsample < 256; nsample++) {

  658.                     c1 = su1[band+nsample];

  659.                     c2 = su2[band+nsample];

  660.                     su1[band+nsample] = c1 + c2;

  661.                     su2[band+nsample] = c1 - c2;

  662.                 }

  663.                 break;

  664.             default:

  665.                 assert(0);

  666.         }

  667.     }

  668. }

  669. 

  670. static void getChannelWeights (int indx, int flag, float ch[2]){

  671. 

  672.     if (indx == 7) {

  673.         ch[0] = 1.0;

  674.         ch[1] = 1.0;

  675.     } else {

  676.         ch[0] = (float)(indx & 7) / 7.0;

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

  678.         if(flag)

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

  680.     }

  681. }

  682. 

  683. static void channelWeighting (float *su1, float *su2, int *p3)

  684. {

  685.     int   band, nsample;

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

  687.     float w[2][2];

  688. 

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

  690.         getChannelWeights(p3[1], p3[0], w[0]);

  691.         getChannelWeights(p3[3], p3[2], w[1]);

  692. 

  693.         for(band = 1; band < 4; band++) {

  694.             /* scale the channels by the weights */

  695.             for(nsample = 0; nsample < 8; nsample++) {

  696.                 su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);

  697.                 su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);

  698.             }

  699. 

  700.             for(; nsample < 256; nsample++) {

  701.                 su1[band*256+nsample] *= w[1][0];

  702.                 su2[band*256+nsample] *= w[1][1];

  703.             }

  704.         }

  705.     }

  706. }

  707. 

  708. 

  709. /**

  710.  * Decode a Sound Unit

  711.  *

  712.  * @param gb            the GetBit context

  713.  * @param pSnd          the channel unit to be used

  714.  * @param pOut          the decoded samples before IQMF in float representation

  715.  * @param channelNum    channel number

  716.  * @param codingMode    the coding mode (JOINT_STEREO or regular stereo/mono)

  717.  */

  718. 

  719. 

  720. static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)

  721. {

  722.     int   band, result=0, numSubbands, lastTonal, numBands;

  723. 

  724.     if (codingMode == JOINT_STEREO && channelNum == 1) {

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

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

  727.             return -1;

  728.         }

  729.     } else {

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

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

  732.             return -1;

  733.         }

  734.     }

  735. 

  736.     /* number of coded QMF bands */

  737.     pSnd->bandsCoded = get_bits(gb,2);

  738. 

  739.     result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);

  740.     if (result) return result;

  741. 

  742.     pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);

  743.     if (pSnd->numComponents == -1) return -1;

  744. 

  745.     numSubbands = decodeSpectrum (gb, pSnd->spectrum);

  746. 

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

  748.     lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);

  749. 

  750. 

  751.     /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */

  752.     numBands = (subbandTab[numSubbands] - 1) >> 8;

  753.     if (lastTonal >= 0)

  754.         numBands = FFMAX((lastTonal + 256) >> 8, numBands);

  755. 

  756. 

  757.     /* Reconstruct time domain samples. */

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

  759.         /* Perform the IMDCT step without overlapping. */

  760.         if (band <= numBands) {

  761.             IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1,q->mdct_tmp);

  762.         } else

  763.             memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));

  764. 

  765.         /* gain compensation and overlapping */

  766.         gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),

  767.                                     &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),

  768.                                     &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));

  769.     }

  770. 

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

  772.     pSnd->gcBlkSwitch ^= 1;

  773. 

  774.     return 0;

  775. }

  776. 

  777. /**

  778.  * Frame handling

  779.  *

  780.  * @param q             Atrac3 private context

  781.  * @param databuf       the input data

  782.  */

  783. 

  784. static int decodeFrame(ATRAC3Context *q, uint8_t* databuf)

  785. {

  786.     int   result, i;

  787.     float   *p1, *p2, *p3, *p4;

  788.     uint8_t    *ptr1, *ptr2;

  789. 

  790.     if (q->codingMode == JOINT_STEREO) {

  791. 

  792.         /* channel coupling mode */

  793.         /* decode Sound Unit 1 */

  794.         init_get_bits(&q->gb,databuf,q->bits_per_frame);

  795. 

  796.         result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);

  797.         if (result != 0)

  798.             return (result);

  799. 

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

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

  802.         ptr1 = databuf;

  803.         ptr2 = databuf+q->bytes_per_frame-1;

  804.         for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {

  805.             FFSWAP(uint8_t,*ptr1,*ptr2);

  806.         }

  807. 

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

  809.         ptr1 = databuf;

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

  811.             if (i >= q->bytes_per_frame)

  812.                 return -1;

  813.         }

  814. 

  815. 

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

  817.         init_get_bits(&q->gb,ptr1,q->bits_per_frame);

  818. 

  819.         /* Fill the Weighting coeffs delay buffer */

  820.         memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));

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

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

  823. 

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

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

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

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

  828.         }

  829. 

  830.         /* Decode Sound Unit 2. */

  831.         result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);

  832.         if (result != 0)

  833.             return (result);

  834. 

  835.         /* Reconstruct the channel coefficients. */

  836.         reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);

  837. 

  838.         channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);

  839. 

  840.     } else {

  841.         /* normal stereo mode or mono */

  842.         /* Decode the channel sound units. */

  843.         for (i=0 ; i<q->channels ; i++) {

  844. 

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

  846.             init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);

  847. 

  848.             result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);

  849.             if (result != 0)

  850.                 return (result);

  851.         }

  852.     }

  853. 

  854.     /* Apply the iQMF synthesis filter. */

  855.     p1= q->outSamples;

  856.     for (i=0 ; i<q->channels ; i++) {

  857.         p2= p1+256;

  858.         p3= p2+256;

  859.         p4= p3+256;

  860.         iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);

  861.         iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);

  862.         iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);

  863.         p1 +=1024;

  864.     }

  865. 

  866.     return 0;

  867. }

  868. 

  869. 

  870. /**

  871.  * Atrac frame decoding

  872.  *

  873.  * @param avctx     pointer to the AVCodecContext

  874.  */

  875. 

  876. static int atrac3_decode_frame(AVCodecContext *avctx,

  877.             void *data, int *data_size,

  878.             const uint8_t *buf, int buf_size) {

  879.     ATRAC3Context *q = avctx->priv_data;

  880.     int result = 0, i;

  881.     uint8_t* databuf;

  882.     int16_t* samples = data;

  883. 

  884.     if (buf_size < avctx->block_align)

  885.         return buf_size;

  886. 

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

  888.     if (q->scrambled_stream) {

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

  890.         databuf = q->decoded_bytes_buffer;

  891.     } else {

  892.         databuf = buf;

  893.     }

  894. 

  895.     result = decodeFrame(q, databuf);

  896. 

  897.     if (result != 0) {

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

  899.         return -1;

  900.     }

  901. 

  902.     if (q->channels == 1) {

  903.         /* mono */

  904.         for (i = 0; i<1024; i++)

  905.             samples[i] = av_clip_int16(round(q->outSamples[i]));

  906.         *data_size = 1024 * sizeof(int16_t);

  907.     } else {

  908.         /* stereo */

  909.         for (i = 0; i < 1024; i++) {

  910.             samples[i*2] = av_clip_int16(round(q->outSamples[i]));

  911.             samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));

  912.         }

  913.         *data_size = 2048 * sizeof(int16_t);

  914.     }

  915. 

  916.     return avctx->block_align;

  917. }

  918. 

  919. 

  920. /**

  921.  * Atrac3 initialization

  922.  *

  923.  * @param avctx     pointer to the AVCodecContext

  924.  */

  925. 

  926. static int atrac3_decode_init(AVCodecContext *avctx)

  927. {

  928.     int i;

  929.     const uint8_t *edata_ptr = avctx->extradata;

  930.     ATRAC3Context *q = avctx->priv_data;

  931. 

  932.     /* Take data from the AVCodecContext (RM container). */

  933.     q->sample_rate = avctx->sample_rate;

  934.     q->channels = avctx->channels;

  935.     q->bit_rate = avctx->bit_rate;

  936.     q->bits_per_frame = avctx->block_align * 8;

  937.     q->bytes_per_frame = avctx->block_align;

  938. 

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

  940.     if (avctx->extradata_size == 14) {

  941.         /* Parse the extradata, WAV format */

  942.         av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown value always 1

  943.         q->samples_per_channel = bytestream_get_le32(&edata_ptr);

  944.         q->codingMode = bytestream_get_le16(&edata_ptr);

  945.         av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr));  //Dupe of coding mode

  946.         q->frame_factor = bytestream_get_le16(&edata_ptr);  //Unknown always 1

  947.         av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown always 0

  948. 

  949.         /* setup */

  950.         q->samples_per_frame = 1024 * q->channels;

  951.         q->atrac3version = 4;

  952.         q->delay = 0x88E;

  953.         if (q->codingMode)

  954.             q->codingMode = JOINT_STEREO;

  955.         else

  956.             q->codingMode = STEREO;

  957. 

  958.         q->scrambled_stream = 0;

  959. 

  960.         if ((q->bytes_per_frame == 96*q->channels*q->frame_factor) || (q->bytes_per_frame == 152*q->channels*q->frame_factor) || (q->bytes_per_frame == 192*q->channels*q->frame_factor)) {

  961.         } else {

  962.             av_log(avctx,AV_LOG_ERROR,"Unknown frame/channel/frame_factor configuration %d/%d/%d\n", q->bytes_per_frame, q->channels, q->frame_factor);

  963.             return -1;

  964.         }

  965. 

  966.     } else if (avctx->extradata_size == 10) {

  967.         /* Parse the extradata, RM format. */

  968.         q->atrac3version = bytestream_get_be32(&edata_ptr);

  969.         q->samples_per_frame = bytestream_get_be16(&edata_ptr);

  970.         q->delay = bytestream_get_be16(&edata_ptr);

  971.         q->codingMode = bytestream_get_be16(&edata_ptr);

  972. 

  973.         q->samples_per_channel = q->samples_per_frame / q->channels;

  974.         q->scrambled_stream = 1;

  975. 

  976.     } else {

  977.         av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);

  978.     }

  979.     /* Check the extradata. */

  980. 

  981.     if (q->atrac3version != 4) {

  982.         av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);

  983.         return -1;

  984.     }

  985. 

  986.     if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {

  987.         av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);

  988.         return -1;

  989.     }

  990. 

  991.     if (q->delay != 0x88E) {

  992.         av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);

  993.         return -1;

  994.     }

  995. 

  996.     if (q->codingMode == STEREO) {

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

  998.     } else if (q->codingMode == JOINT_STEREO) {

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

  1000.     } else {

  1001.         av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);

  1002.         return -1;

  1003.     }

  1004. 

  1005.     if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {

  1006.         av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");

  1007.         return -1;

  1008.     }

  1009. 

  1010. 

  1011.     if(avctx->block_align >= UINT_MAX/2)

  1012.         return -1;

  1013. 

  1014.     /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,

  1015.      * this is for the bitstream reader. */

  1016.     if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)))  == NULL)

  1017.         return AVERROR(ENOMEM);

  1018. 

  1019. 

  1020.     /* Initialize the VLC tables. */

  1021.     for (i=0 ; i<7 ; i++) {

  1022.         init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],

  1023.             huff_bits[i], 1, 1,

  1024.             huff_codes[i], 1, 1, INIT_VLC_USE_STATIC);

  1025.     }

  1026. 

  1027.     init_atrac3_transforms(q);

  1028. 

  1029.     /* Generate the scale factors. */

  1030.     for (i=0 ; i<64 ; i++)

  1031.         SFTable[i] = pow(2.0, (i - 15) / 3.0);

  1032. 

  1033.     /* Generate gain tables. */

  1034.     for (i=0 ; i<16 ; i++)

  1035.         gain_tab1[i] = powf (2.0, (4 - i));

  1036. 

  1037.     for (i=-15 ; i<16 ; i++)

  1038.         gain_tab2[i+15] = powf (2.0, i * -0.125);

  1039. 

  1040.     /* init the joint-stereo decoding data */

  1041.     q->weighting_delay[0] = 0;

  1042.     q->weighting_delay[1] = 7;

  1043.     q->weighting_delay[2] = 0;

  1044.     q->weighting_delay[3] = 7;

  1045.     q->weighting_delay[4] = 0;

  1046.     q->weighting_delay[5] = 7;

  1047. 

  1048.     for (i=0; i<4; i++) {

  1049.         q->matrix_coeff_index_prev[i] = 3;

  1050.         q->matrix_coeff_index_now[i] = 3;

  1051.         q->matrix_coeff_index_next[i] = 3;

  1052.     }

  1053. 

  1054.     dsputil_init(&dsp, avctx);

  1055. 

  1056.     q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);

  1057.     if (!q->pUnits) {

  1058.         av_free(q->decoded_bytes_buffer);

  1059.         return AVERROR(ENOMEM);

  1060.     }

  1061. 

  1062.     return 0;

  1063. }

  1064. 

  1065. 

  1066. AVCodec atrac3_decoder =

  1067. {

  1068.     .name = "atrac3",

  1069.     .type = CODEC_TYPE_AUDIO,

  1070.     .id = CODEC_ID_ATRAC3,

  1071.     .priv_data_size = sizeof(ATRAC3Context),

  1072.     .init = atrac3_decode_init,

  1073.     .close = atrac3_decode_close,

  1074.     .decode = atrac3_decode_frame,

  1075.     .long_name = "Atrac 3 (Adaptive TRansform Acoustic Coding 3)",

  1076. };