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FFmpeg/libavcodec/atrac3.c

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  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 libavcodec/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.     //@}

  112.     //@{

  113.     /** extradata */

  114.     int                 atrac3version;

  115.     int                 delay;

  116.     int                 scrambled_stream;

  117.     int                 frame_factor;

  118.     //@}

  119. } ATRAC3Context;

  120. 

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

  122. static float            qmf_window[48];

  123. static VLC              spectral_coeff_tab[7];

  124. static float            SFTable[64];

  125. static float            gain_tab1[16];

  126. static float            gain_tab2[31];

  127. static MDCTContext      mdct_ctx;

  128. static DSPContext       dsp;

  129. 

  130. 

  131. /* quadrature mirror synthesis filter */

  132. 

  133. /**

  134.  * Quadrature mirror synthesis filter.

  135.  *

  136.  * @param inlo      lower part of spectrum

  137.  * @param inhi      higher part of spectrum

  138.  * @param nIn       size of spectrum buffer

  139.  * @param pOut      out buffer

  140.  * @param delayBuf  delayBuf buffer

  141.  * @param temp      temp buffer

  142.  */

  143. 

  144. 

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

  146. {

  147.     int   i, j;

  148.     float   *p1, *p3;

  149. 

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

  151. 

  152.     p3 = temp + 46;

  153. 

  154.     /* loop1 */

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

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

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

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

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

  160.     }

  161. 

  162.     /* loop2 */

  163.     p1 = temp;

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

  165.         float s1 = 0.0;

  166.         float s2 = 0.0;

  167. 

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

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

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

  171.         }

  172. 

  173.         pOut[0] = s2;

  174.         pOut[1] = s1;

  175. 

  176.         p1 += 2;

  177.         pOut += 2;

  178.     }

  179. 

  180.     /* Update the delay buffer. */

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

  182. }

  183. 

  184. /**

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

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

  187.  *

  188.  * @param pInput    float input

  189.  * @param pOutput   float output

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

  191.  */

  192. 

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

  194. {

  195.     int     i;

  196. 

  197.     if (odd_band) {

  198.         /**

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

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

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

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

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

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

  205.         */

  206. 

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

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

  209.     }

  210. 

  211.     ff_imdct_calc(&mdct_ctx,pOutput,pInput);

  212. 

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

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

  215. 

  216. }

  217. 

  218. 

  219. /**

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

  221.  *

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

  223.  * @param bits      amount of bits

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

  225.  */

  226. 

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

  228.     int i, off;

  229.     uint32_t c;

  230.     const uint32_t* buf;

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

  232. 

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

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

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

  236.     bytes += 3 + off;

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

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

  239. 

  240.     if (off)

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

  242. 

  243.     return off;

  244. }

  245. 

  246. 

  247. static void init_atrac3_transforms(ATRAC3Context *q) {

  248.     float enc_window[256];

  249.     float s;

  250.     int i;

  251. 

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

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

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

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

  256. 

  257.     if (!mdct_window[0])

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

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

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

  261.         }

  262. 

  263.     /* Generate the QMF window. */

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

  265.         s = qmf_48tap_half[i] * 2.0;

  266.         qmf_window[i] = s;

  267.         qmf_window[47 - i] = s;

  268.     }

  269. 

  270.     /* Initialize the MDCT transform. */

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

  272. }

  273. 

  274. /**

  275.  * Atrac3 uninit, free all allocated memory

  276.  */

  277. 

  278. static int atrac3_decode_close(AVCodecContext *avctx)

  279. {

  280.     ATRAC3Context *q = avctx->priv_data;

  281. 

  282.     av_free(q->pUnits);

  283.     av_free(q->decoded_bytes_buffer);

  284. 

  285.     return 0;

  286. }

  287. 

  288. /**

  289. / * Mantissa decoding

  290.  *

  291.  * @param gb            the GetBit context

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

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

  294.  * @param mantissas     mantissa output table

  295.  * @param numCodes      amount of values to get

  296.  */

  297. 

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

  299. {

  300.     int   numBits, cnt, code, huffSymb;

  301. 

  302.     if (selector == 1)

  303.         numCodes /= 2;

  304. 

  305.     if (codingFlag != 0) {

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

  307.         numBits = CLCLengthTab[selector];

  308. 

  309.         if (selector > 1) {

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

  311.                 if (numBits)

  312.                     code = get_sbits(gb, numBits);

  313.                 else

  314.                     code = 0;

  315.                 mantissas[cnt] = code;

  316.             }

  317.         } else {

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

  319.                 if (numBits)

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

  321.                 else

  322.                     code = 0;

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

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

  325.             }

  326.         }

  327.     } else {

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

  329.         if (selector != 1) {

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

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

  332.                 huffSymb += 1;

  333.                 code = huffSymb >> 1;

  334.                 if (huffSymb & 1)

  335.                     code = -code;

  336.                 mantissas[cnt] = code;

  337.             }

  338.         } else {

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

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

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

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

  343.             }

  344.         }

  345.     }

  346. }

  347. 

  348. /**

  349.  * Restore the quantized band spectrum coefficients

  350.  *

  351.  * @param gb            the GetBit context

  352.  * @param pOut          decoded band spectrum

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

  354.  */

  355. 

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

  357. {

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

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

  360.     int   mantissas[128];

  361.     float SF;

  362. 

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

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

  365. 

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

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

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

  369. 

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

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

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

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

  374.     }

  375. 

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

  377.         first = subbandTab[cnt];

  378.         last = subbandTab[cnt+1];

  379. 

  380.         subbWidth = last - first;

  381. 

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

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

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

  385.              * same VLC selector (subband_vlc_index) */

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

  387. 

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

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

  390. 

  391.             /* Inverse quantize the coefficients. */

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

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

  394.         } else {

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

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

  397.         }

  398.     }

  399. 

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

  401.     first = subbandTab[cnt];

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

  403.     return numSubbands;

  404. }

  405. 

  406. /**

  407.  * Restore the quantized tonal components

  408.  *

  409.  * @param gb            the GetBit context

  410.  * @param pComponent    tone component

  411.  * @param numBands      amount of coded bands

  412.  */

  413. 

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

  415. {

  416.     int i,j,k,cnt;

  417.     int   components, coding_mode_selector, coding_mode, coded_values_per_component;

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

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

  420.     float  *pCoef;

  421.     float  scalefactor;

  422.     int   component_count = 0;

  423. 

  424.     components = get_bits(gb,5);

  425. 

  426.     /* no tonal components */

  427.     if (components == 0)

  428.         return 0;

  429. 

  430.     coding_mode_selector = get_bits(gb,2);

  431.     if (coding_mode_selector == 2)

  432.         return -1;

  433. 

  434.     coding_mode = coding_mode_selector & 1;

  435. 

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

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

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

  439. 

  440.         coded_values_per_component = get_bits(gb,3);

  441. 

  442.         quant_step_index = get_bits(gb,3);

  443.         if (quant_step_index <= 1)

  444.             return -1;

  445. 

  446.         if (coding_mode_selector == 3)

  447.             coding_mode = get_bits1(gb);

  448. 

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

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

  451.                 continue;

  452. 

  453.             coded_components = get_bits(gb,3);

  454. 

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

  456.                 sfIndx = get_bits(gb,6);

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

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

  459.                 coded_values = coded_values_per_component + 1;

  460.                 coded_values = FFMIN(max_coded_values,coded_values);

  461. 

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

  463. 

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

  465. 

  466.                 pComponent[component_count].numCoefs = coded_values;

  467. 

  468.                 /* inverse quant */

  469.                 pCoef = pComponent[component_count].coef;

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

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

  472. 

  473.                 component_count++;

  474.             }

  475.         }

  476.     }

  477. 

  478.     return component_count;

  479. }

  480. 

  481. /**

  482.  * Decode gain parameters for the coded bands

  483.  *

  484.  * @param gb            the GetBit context

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

  486.  * @param numBands      amount of coded bands

  487.  */

  488. 

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

  490. {

  491.     int   i, cf, numData;

  492.     int   *pLevel, *pLoc;

  493. 

  494.     gain_info   *pGain = pGb->gBlock;

  495. 

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

  497.     {

  498.         numData = get_bits(gb,3);

  499.         pGain[i].num_gain_data = numData;

  500.         pLevel = pGain[i].levcode;

  501.         pLoc = pGain[i].loccode;

  502. 

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

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

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

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

  507.                 return -1;

  508.         }

  509.     }

  510. 

  511.     /* Clear the unused blocks. */

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

  513.         pGain[i].num_gain_data = 0;

  514. 

  515.     return 0;

  516. }

  517. 

  518. /**

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

  520.  *

  521.  * @param pIn           input float buffer

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

  523.  * @param pOut          output float buffer

  524.  * @param pGain1        current band gain info

  525.  * @param pGain2        next band gain info

  526.  */

  527. 

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

  529. {

  530.     /* gain compensation function */

  531.     float  gain1, gain2, gain_inc;

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

  533. 

  534. 

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

  536.         gain1 = 1.0;

  537.     else

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

  539. 

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

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

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

  543.     } else {

  544.         numdata = pGain1->num_gain_data;

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

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

  547. 

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

  549. 

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

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

  552.             endLoc = startLoc + 8;

  553. 

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

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

  556. 

  557.             /* interpolate */

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

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

  560. 

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

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

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

  564.                 gain2 *= gain_inc;

  565.             }

  566.         }

  567. 

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

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

  570.     }

  571. 

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

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

  574. }

  575. 

  576. /**

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

  578.  * Return position of the last tonal coefficient

  579.  *

  580.  * @param pSpectrum     output spectrum buffer

  581.  * @param numComponents amount of tonal components

  582.  * @param pComponent    tonal components for this band

  583.  */

  584. 

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

  586. {

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

  588.     float   *pIn, *pOut;

  589. 

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

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

  592.         pIn = pComponent[cnt].coef;

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

  594. 

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

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

  597.     }

  598. 

  599.     return lastPos;

  600. }

  601. 

  602. 

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

  604. 

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

  606. {

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

  608.     float    c1, c2;

  609.     float    mc1_l, mc1_r, mc2_l, mc2_r;

  610. 

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

  612.         s1 = pPrevCode[i];

  613.         s2 = pCurrCode[i];

  614.         nsample = 0;

  615. 

  616.         if (s1 != s2) {

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

  618.             mc1_l = matrixCoeffs[s1*2];

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

  620.             mc2_l = matrixCoeffs[s2*2];

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

  622. 

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

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

  625.                 c1 = su1[band+nsample];

  626.                 c2 = su2[band+nsample];

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

  628.                 su1[band+nsample] = c2;

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

  630.             }

  631.         }

  632. 

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

  634.         switch (s2) {

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

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

  637.                     c1 = su1[band+nsample];

  638.                     c2 = su2[band+nsample];

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

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

  641.                 }

  642.                 break;

  643. 

  644.             case 1:

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

  646.                     c1 = su1[band+nsample];

  647.                     c2 = su2[band+nsample];

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

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

  650.                 }

  651.                 break;

  652.             case 2:

  653.             case 3:

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

  655.                     c1 = su1[band+nsample];

  656.                     c2 = su2[band+nsample];

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

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

  659.                 }

  660.                 break;

  661.             default:

  662.                 assert(0);

  663.         }

  664.     }

  665. }

  666. 

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

  668. 

  669.     if (indx == 7) {

  670.         ch[0] = 1.0;

  671.         ch[1] = 1.0;

  672.     } else {

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

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

  675.         if(flag)

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

  677.     }

  678. }

  679. 

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

  681. {

  682.     int   band, nsample;

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

  684.     float w[2][2];

  685. 

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

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

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

  689. 

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

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

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

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

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

  695.             }

  696. 

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

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

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

  700.             }

  701.         }

  702.     }

  703. }

  704. 

  705. 

  706. /**

  707.  * Decode a Sound Unit

  708.  *

  709.  * @param gb            the GetBit context

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

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

  712.  * @param channelNum    channel number

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

  714.  */

  715. 

  716. 

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

  718. {

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

  720. 

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

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

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

  724.             return -1;

  725.         }

  726.     } else {

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

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

  729.             return -1;

  730.         }

  731.     }

  732. 

  733.     /* number of coded QMF bands */

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

  735. 

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

  737.     if (result) return result;

  738. 

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

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

  741. 

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

  743. 

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

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

  746. 

  747. 

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

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

  750.     if (lastTonal >= 0)

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

  752. 

  753. 

  754.     /* Reconstruct time domain samples. */

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

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

  757.         if (band <= numBands) {

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

  759.         } else

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

  761. 

  762.         /* gain compensation and overlapping */

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

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

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

  766.     }

  767. 

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

  769.     pSnd->gcBlkSwitch ^= 1;

  770. 

  771.     return 0;

  772. }

  773. 

  774. /**

  775.  * Frame handling

  776.  *

  777.  * @param q             Atrac3 private context

  778.  * @param databuf       the input data

  779.  */

  780. 

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

  782. {

  783.     int   result, i;

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

  785.     uint8_t *ptr1;

  786. 

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

  788. 

  789.         /* channel coupling mode */

  790.         /* decode Sound Unit 1 */

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

  792. 

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

  794.         if (result != 0)

  795.             return (result);

  796. 

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

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

  799.         if (databuf == q->decoded_bytes_buffer) {

  800.             uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1;

  801.             ptr1 = q->decoded_bytes_buffer;

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

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

  804.             }

  805.         } else {

  806.             const uint8_t *ptr2 = databuf+q->bytes_per_frame-1;

  807.             for (i = 0; i < q->bytes_per_frame; i++)

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

  809.         }

  810. 

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

  812.         ptr1 = q->decoded_bytes_buffer;

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

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

  815.                 return -1;

  816.         }

  817. 

  818. 

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

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

  821. 

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

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

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

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

  826. 

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

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

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

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

  831.         }

  832. 

  833.         /* Decode Sound Unit 2. */

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

  835.         if (result != 0)

  836.             return (result);

  837. 

  838.         /* Reconstruct the channel coefficients. */

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

  840. 

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

  842. 

  843.     } else {

  844.         /* normal stereo mode or mono */

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

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

  847. 

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

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

  850. 

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

  852.             if (result != 0)

  853.                 return (result);

  854.         }

  855.     }

  856. 

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

  858.     p1= q->outSamples;

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

  860.         p2= p1+256;

  861.         p3= p2+256;

  862.         p4= p3+256;

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

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

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

  866.         p1 +=1024;

  867.     }

  868. 

  869.     return 0;

  870. }

  871. 

  872. 

  873. /**

  874.  * Atrac frame decoding

  875.  *

  876.  * @param avctx     pointer to the AVCodecContext

  877.  */

  878. 

  879. static int atrac3_decode_frame(AVCodecContext *avctx,

  880.             void *data, int *data_size,

  881.             const uint8_t *buf, int buf_size) {

  882.     ATRAC3Context *q = avctx->priv_data;

  883.     int result = 0, i;

  884.     const uint8_t* databuf;

  885.     int16_t* samples = data;

  886. 

  887.     if (buf_size < avctx->block_align)

  888.         return buf_size;

  889. 

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

  891.     if (q->scrambled_stream) {

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

  893.         databuf = q->decoded_bytes_buffer;

  894.     } else {

  895.         databuf = buf;

  896.     }

  897. 

  898.     result = decodeFrame(q, databuf);

  899. 

  900.     if (result != 0) {

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

  902.         return -1;

  903.     }

  904. 

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

  906.         /* mono */

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

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

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

  910.     } else {

  911.         /* stereo */

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

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

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

  915.         }

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

  917.     }

  918. 

  919.     return avctx->block_align;

  920. }

  921. 

  922. 

  923. /**

  924.  * Atrac3 initialization

  925.  *

  926.  * @param avctx     pointer to the AVCodecContext

  927.  */

  928. 

  929. static int atrac3_decode_init(AVCodecContext *avctx)

  930. {

  931.     int i;

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

  933.     ATRAC3Context *q = avctx->priv_data;

  934. 

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

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

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

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

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

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

  941. 

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

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

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

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

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

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

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

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

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

  951. 

  952.         /* setup */

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

  954.         q->atrac3version = 4;

  955.         q->delay = 0x88E;

  956.         if (q->codingMode)

  957.             q->codingMode = JOINT_STEREO;

  958.         else

  959.             q->codingMode = STEREO;

  960. 

  961.         q->scrambled_stream = 0;

  962. 

  963.         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)) {

  964.         } else {

  965.             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);

  966.             return -1;

  967.         }

  968. 

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

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

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

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

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

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

  975. 

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

  977.         q->scrambled_stream = 1;

  978. 

  979.     } else {

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

  981.     }

  982.     /* Check the extradata. */

  983. 

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

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

  986.         return -1;

  987.     }

  988. 

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

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

  991.         return -1;

  992.     }

  993. 

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

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

  996.         return -1;

  997.     }

  998. 

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

  1000.         av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");

  1001.     } else if (q->codingMode == JOINT_STEREO) {

  1002.         av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");

  1003.     } else {

  1004.         av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);

  1005.         return -1;

  1006.     }

  1007. 

  1008.     if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {

  1009.         av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");

  1010.         return -1;

  1011.     }

  1012. 

  1013. 

  1014.     if(avctx->block_align >= UINT_MAX/2)

  1015.         return -1;

  1016. 

  1017.     /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,

  1018.      * this is for the bitstream reader. */

  1019.     if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)))  == NULL)

  1020.         return AVERROR(ENOMEM);

  1021. 

  1022. 

  1023.     /* Initialize the VLC tables. */

  1024.     for (i=0 ; i<7 ; i++) {

  1025.         init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],

  1026.             huff_bits[i], 1, 1,

  1027.             huff_codes[i], 1, 1, INIT_VLC_USE_STATIC);

  1028.     }

  1029. 

  1030.     init_atrac3_transforms(q);

  1031. 

  1032.     /* Generate the scale factors. */

  1033.     for (i=0 ; i<64 ; i++)

  1034.         SFTable[i] = pow(2.0, (i - 15) / 3.0);

  1035. 

  1036.     /* Generate gain tables. */

  1037.     for (i=0 ; i<16 ; i++)

  1038.         gain_tab1[i] = powf (2.0, (4 - i));

  1039. 

  1040.     for (i=-15 ; i<16 ; i++)

  1041.         gain_tab2[i+15] = powf (2.0, i * -0.125);

  1042. 

  1043.     /* init the joint-stereo decoding data */

  1044.     q->weighting_delay[0] = 0;

  1045.     q->weighting_delay[1] = 7;

  1046.     q->weighting_delay[2] = 0;

  1047.     q->weighting_delay[3] = 7;

  1048.     q->weighting_delay[4] = 0;

  1049.     q->weighting_delay[5] = 7;

  1050. 

  1051.     for (i=0; i<4; i++) {

  1052.         q->matrix_coeff_index_prev[i] = 3;

  1053.         q->matrix_coeff_index_now[i] = 3;

  1054.         q->matrix_coeff_index_next[i] = 3;

  1055.     }

  1056. 

  1057.     dsputil_init(&dsp, avctx);

  1058. 

  1059.     q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);

  1060.     if (!q->pUnits) {

  1061.         av_free(q->decoded_bytes_buffer);

  1062.         return AVERROR(ENOMEM);

  1063.     }

  1064. 

  1065.     avctx->sample_fmt = SAMPLE_FMT_S16;

  1066.     return 0;

  1067. }

  1068. 

  1069. 

  1070. AVCodec atrac3_decoder =

  1071. {

  1072.     .name = "atrac3",

  1073.     .type = CODEC_TYPE_AUDIO,

  1074.     .id = CODEC_ID_ATRAC3,

  1075.     .priv_data_size = sizeof(ATRAC3Context),

  1076.     .init = atrac3_decode_init,

  1077.     .close = atrac3_decode_close,

  1078.     .decode = atrac3_decode_frame,

  1079.     .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),

  1080. };