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libavcodec/atrac3.c

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00001 /*
00002  * Atrac 3 compatible decoder
00003  * Copyright (c) 2006-2008 Maxim Poliakovski
00004  * Copyright (c) 2006-2008 Benjamin Larsson
00005  *
00006  * This file is part of FFmpeg.
00007  *
00008  * FFmpeg is free software; you can redistribute it and/or
00009  * modify it under the terms of the GNU Lesser General Public
00010  * License as published by the Free Software Foundation; either
00011  * version 2.1 of the License, or (at your option) any later version.
00012  *
00013  * FFmpeg is distributed in the hope that it will be useful,
00014  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00015  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00016  * Lesser General Public License for more details.
00017  *
00018  * You should have received a copy of the GNU Lesser General Public
00019  * License along with FFmpeg; if not, write to the Free Software
00020  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00021  */
00022 
00035 #include <math.h>
00036 #include <stddef.h>
00037 #include <stdio.h>
00038 
00039 #include "avcodec.h"
00040 #include "bitstream.h"
00041 #include "dsputil.h"
00042 #include "bytestream.h"
00043 
00044 #include "atrac3data.h"
00045 
00046 #define JOINT_STEREO    0x12
00047 #define STEREO          0x2
00048 
00049 
00050 /* These structures are needed to store the parsed gain control data. */
00051 typedef struct {
00052     int   num_gain_data;
00053     int   levcode[8];
00054     int   loccode[8];
00055 } gain_info;
00056 
00057 typedef struct {
00058     gain_info   gBlock[4];
00059 } gain_block;
00060 
00061 typedef struct {
00062     int     pos;
00063     int     numCoefs;
00064     float   coef[8];
00065 } tonal_component;
00066 
00067 typedef struct {
00068     int               bandsCoded;
00069     int               numComponents;
00070     tonal_component   components[64];
00071     float             prevFrame[1024];
00072     int               gcBlkSwitch;
00073     gain_block        gainBlock[2];
00074 
00075     DECLARE_ALIGNED_16(float, spectrum[1024]);
00076     DECLARE_ALIGNED_16(float, IMDCT_buf[1024]);
00077 
00078     float             delayBuf1[46]; 
00079     float             delayBuf2[46];
00080     float             delayBuf3[46];
00081 } channel_unit;
00082 
00083 typedef struct {
00084     GetBitContext       gb;
00086 
00087     int                 channels;
00088     int                 codingMode;
00089     int                 bit_rate;
00090     int                 sample_rate;
00091     int                 samples_per_channel;
00092     int                 samples_per_frame;
00093 
00094     int                 bits_per_frame;
00095     int                 bytes_per_frame;
00096     int                 pBs;
00097     channel_unit*       pUnits;
00099 
00100 
00101     int                 matrix_coeff_index_prev[4];
00102     int                 matrix_coeff_index_now[4];
00103     int                 matrix_coeff_index_next[4];
00104     int                 weighting_delay[6];
00106 
00107 
00108     float               outSamples[2048];
00109     uint8_t*            decoded_bytes_buffer;
00110     float               tempBuf[1070];
00112 
00113 
00114     int                 atrac3version;
00115     int                 delay;
00116     int                 scrambled_stream;
00117     int                 frame_factor;
00119 } ATRAC3Context;
00120 
00121 static DECLARE_ALIGNED_16(float,mdct_window[512]);
00122 static float            qmf_window[48];
00123 static VLC              spectral_coeff_tab[7];
00124 static float            SFTable[64];
00125 static float            gain_tab1[16];
00126 static float            gain_tab2[31];
00127 static MDCTContext      mdct_ctx;
00128 static DSPContext       dsp;
00129 
00130 
00131 /* quadrature mirror synthesis filter */
00132 
00145 static void iqmf (float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
00146 {
00147     int   i, j;
00148     float   *p1, *p3;
00149 
00150     memcpy(temp, delayBuf, 46*sizeof(float));
00151 
00152     p3 = temp + 46;
00153 
00154     /* loop1 */
00155     for(i=0; i<nIn; i+=2){
00156         p3[2*i+0] = inlo[i  ] + inhi[i  ];
00157         p3[2*i+1] = inlo[i  ] - inhi[i  ];
00158         p3[2*i+2] = inlo[i+1] + inhi[i+1];
00159         p3[2*i+3] = inlo[i+1] - inhi[i+1];
00160     }
00161 
00162     /* loop2 */
00163     p1 = temp;
00164     for (j = nIn; j != 0; j--) {
00165         float s1 = 0.0;
00166         float s2 = 0.0;
00167 
00168         for (i = 0; i < 48; i += 2) {
00169             s1 += p1[i] * qmf_window[i];
00170             s2 += p1[i+1] * qmf_window[i+1];
00171         }
00172 
00173         pOut[0] = s2;
00174         pOut[1] = s1;
00175 
00176         p1 += 2;
00177         pOut += 2;
00178     }
00179 
00180     /* Update the delay buffer. */
00181     memcpy(delayBuf, temp + nIn*2, 46*sizeof(float));
00182 }
00183 
00193 static void IMLT(float *pInput, float *pOutput, int odd_band)
00194 {
00195     int     i;
00196 
00197     if (odd_band) {
00207         for (i=0; i<128; i++)
00208             FFSWAP(float, pInput[i], pInput[255-i]);
00209     }
00210 
00211     ff_imdct_calc(&mdct_ctx,pOutput,pInput);
00212 
00213     /* Perform windowing on the output. */
00214     dsp.vector_fmul(pOutput,mdct_window,512);
00215 
00216 }
00217 
00218 
00227 static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
00228     int i, off;
00229     uint32_t c;
00230     const uint32_t* buf;
00231     uint32_t* obuf = (uint32_t*) out;
00232 
00233     off = (int)((long)inbuffer & 3);
00234     buf = (const uint32_t*) (inbuffer - off);
00235     c = be2me_32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
00236     bytes += 3 + off;
00237     for (i = 0; i < bytes/4; i++)
00238         obuf[i] = c ^ buf[i];
00239 
00240     if (off)
00241         av_log(NULL,AV_LOG_DEBUG,"Offset of %d not handled, post sample on ffmpeg-dev.\n",off);
00242 
00243     return off;
00244 }
00245 
00246 
00247 static av_cold void init_atrac3_transforms(ATRAC3Context *q) {
00248     float enc_window[256];
00249     float s;
00250     int i;
00251 
00252     /* Generate the mdct window, for details see
00253      * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
00254     for (i=0 ; i<256; i++)
00255         enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
00256 
00257     if (!mdct_window[0])
00258         for (i=0 ; i<256; i++) {
00259             mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
00260             mdct_window[511-i] = mdct_window[i];
00261         }
00262 
00263     /* Generate the QMF window. */
00264     for (i=0 ; i<24; i++) {
00265         s = qmf_48tap_half[i] * 2.0;
00266         qmf_window[i] = s;
00267         qmf_window[47 - i] = s;
00268     }
00269 
00270     /* Initialize the MDCT transform. */
00271     ff_mdct_init(&mdct_ctx, 9, 1);
00272 }
00273 
00278 static av_cold int atrac3_decode_close(AVCodecContext *avctx)
00279 {
00280     ATRAC3Context *q = avctx->priv_data;
00281 
00282     av_free(q->pUnits);
00283     av_free(q->decoded_bytes_buffer);
00284 
00285     return 0;
00286 }
00287 
00298 static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
00299 {
00300     int   numBits, cnt, code, huffSymb;
00301 
00302     if (selector == 1)
00303         numCodes /= 2;
00304 
00305     if (codingFlag != 0) {
00306         /* constant length coding (CLC) */
00307         numBits = CLCLengthTab[selector];
00308 
00309         if (selector > 1) {
00310             for (cnt = 0; cnt < numCodes; cnt++) {
00311                 if (numBits)
00312                     code = get_sbits(gb, numBits);
00313                 else
00314                     code = 0;
00315                 mantissas[cnt] = code;
00316             }
00317         } else {
00318             for (cnt = 0; cnt < numCodes; cnt++) {
00319                 if (numBits)
00320                     code = get_bits(gb, numBits); //numBits is always 4 in this case
00321                 else
00322                     code = 0;
00323                 mantissas[cnt*2] = seTab_0[code >> 2];
00324                 mantissas[cnt*2+1] = seTab_0[code & 3];
00325             }
00326         }
00327     } else {
00328         /* variable length coding (VLC) */
00329         if (selector != 1) {
00330             for (cnt = 0; cnt < numCodes; cnt++) {
00331                 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
00332                 huffSymb += 1;
00333                 code = huffSymb >> 1;
00334                 if (huffSymb & 1)
00335                     code = -code;
00336                 mantissas[cnt] = code;
00337             }
00338         } else {
00339             for (cnt = 0; cnt < numCodes; cnt++) {
00340                 huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
00341                 mantissas[cnt*2] = decTable1[huffSymb*2];
00342                 mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
00343             }
00344         }
00345     }
00346 }
00347 
00356 static int decodeSpectrum (GetBitContext *gb, float *pOut)
00357 {
00358     int   numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
00359     int   subband_vlc_index[32], SF_idxs[32];
00360     int   mantissas[128];
00361     float SF;
00362 
00363     numSubbands = get_bits(gb, 5); // number of coded subbands
00364     codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
00365 
00366     /* Get the VLC selector table for the subbands, 0 means not coded. */
00367     for (cnt = 0; cnt <= numSubbands; cnt++)
00368         subband_vlc_index[cnt] = get_bits(gb, 3);
00369 
00370     /* Read the scale factor indexes from the stream. */
00371     for (cnt = 0; cnt <= numSubbands; cnt++) {
00372         if (subband_vlc_index[cnt] != 0)
00373             SF_idxs[cnt] = get_bits(gb, 6);
00374     }
00375 
00376     for (cnt = 0; cnt <= numSubbands; cnt++) {
00377         first = subbandTab[cnt];
00378         last = subbandTab[cnt+1];
00379 
00380         subbWidth = last - first;
00381 
00382         if (subband_vlc_index[cnt] != 0) {
00383             /* Decode spectral coefficients for this subband. */
00384             /* TODO: This can be done faster is several blocks share the
00385              * same VLC selector (subband_vlc_index) */
00386             readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
00387 
00388             /* Decode the scale factor for this subband. */
00389             SF = SFTable[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
00390 
00391             /* Inverse quantize the coefficients. */
00392             for (pIn=mantissas ; first<last; first++, pIn++)
00393                 pOut[first] = *pIn * SF;
00394         } else {
00395             /* This subband was not coded, so zero the entire subband. */
00396             memset(pOut+first, 0, subbWidth*sizeof(float));
00397         }
00398     }
00399 
00400     /* Clear the subbands that were not coded. */
00401     first = subbandTab[cnt];
00402     memset(pOut+first, 0, (1024 - first) * sizeof(float));
00403     return numSubbands;
00404 }
00405 
00414 static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
00415 {
00416     int i,j,k,cnt;
00417     int   components, coding_mode_selector, coding_mode, coded_values_per_component;
00418     int   sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
00419     int   band_flags[4], mantissa[8];
00420     float  *pCoef;
00421     float  scalefactor;
00422     int   component_count = 0;
00423 
00424     components = get_bits(gb,5);
00425 
00426     /* no tonal components */
00427     if (components == 0)
00428         return 0;
00429 
00430     coding_mode_selector = get_bits(gb,2);
00431     if (coding_mode_selector == 2)
00432         return -1;
00433 
00434     coding_mode = coding_mode_selector & 1;
00435 
00436     for (i = 0; i < components; i++) {
00437         for (cnt = 0; cnt <= numBands; cnt++)
00438             band_flags[cnt] = get_bits1(gb);
00439 
00440         coded_values_per_component = get_bits(gb,3);
00441 
00442         quant_step_index = get_bits(gb,3);
00443         if (quant_step_index <= 1)
00444             return -1;
00445 
00446         if (coding_mode_selector == 3)
00447             coding_mode = get_bits1(gb);
00448 
00449         for (j = 0; j < (numBands + 1) * 4; j++) {
00450             if (band_flags[j >> 2] == 0)
00451                 continue;
00452 
00453             coded_components = get_bits(gb,3);
00454 
00455             for (k=0; k<coded_components; k++) {
00456                 sfIndx = get_bits(gb,6);
00457                 if (component_count >= 64)
00458                     return AVERROR_INVALIDDATA;
00459                 pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
00460                 max_coded_values = 1024 - pComponent[component_count].pos;
00461                 coded_values = coded_values_per_component + 1;
00462                 coded_values = FFMIN(max_coded_values,coded_values);
00463 
00464                 scalefactor = SFTable[sfIndx] * iMaxQuant[quant_step_index];
00465 
00466                 readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
00467 
00468                 pComponent[component_count].numCoefs = coded_values;
00469 
00470                 /* inverse quant */
00471                 pCoef = pComponent[component_count].coef;
00472                 for (cnt = 0; cnt < coded_values; cnt++)
00473                     pCoef[cnt] = mantissa[cnt] * scalefactor;
00474 
00475                 component_count++;
00476             }
00477         }
00478     }
00479 
00480     return component_count;
00481 }
00482 
00491 static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
00492 {
00493     int   i, cf, numData;
00494     int   *pLevel, *pLoc;
00495 
00496     gain_info   *pGain = pGb->gBlock;
00497 
00498     for (i=0 ; i<=numBands; i++)
00499     {
00500         numData = get_bits(gb,3);
00501         pGain[i].num_gain_data = numData;
00502         pLevel = pGain[i].levcode;
00503         pLoc = pGain[i].loccode;
00504 
00505         for (cf = 0; cf < numData; cf++){
00506             pLevel[cf]= get_bits(gb,4);
00507             pLoc  [cf]= get_bits(gb,5);
00508             if(cf && pLoc[cf] <= pLoc[cf-1])
00509                 return -1;
00510         }
00511     }
00512 
00513     /* Clear the unused blocks. */
00514     for (; i<4 ; i++)
00515         pGain[i].num_gain_data = 0;
00516 
00517     return 0;
00518 }
00519 
00530 static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
00531 {
00532     /* gain compensation function */
00533     float  gain1, gain2, gain_inc;
00534     int   cnt, numdata, nsample, startLoc, endLoc;
00535 
00536 
00537     if (pGain2->num_gain_data == 0)
00538         gain1 = 1.0;
00539     else
00540         gain1 = gain_tab1[pGain2->levcode[0]];
00541 
00542     if (pGain1->num_gain_data == 0) {
00543         for (cnt = 0; cnt < 256; cnt++)
00544             pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
00545     } else {
00546         numdata = pGain1->num_gain_data;
00547         pGain1->loccode[numdata] = 32;
00548         pGain1->levcode[numdata] = 4;
00549 
00550         nsample = 0; // current sample = 0
00551 
00552         for (cnt = 0; cnt < numdata; cnt++) {
00553             startLoc = pGain1->loccode[cnt] * 8;
00554             endLoc = startLoc + 8;
00555 
00556             gain2 = gain_tab1[pGain1->levcode[cnt]];
00557             gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
00558 
00559             /* interpolate */
00560             for (; nsample < startLoc; nsample++)
00561                 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
00562 
00563             /* interpolation is done over eight samples */
00564             for (; nsample < endLoc; nsample++) {
00565                 pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
00566                 gain2 *= gain_inc;
00567             }
00568         }
00569 
00570         for (; nsample < 256; nsample++)
00571             pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
00572     }
00573 
00574     /* Delay for the overlapping part. */
00575     memcpy(pPrev, &pIn[256], 256*sizeof(float));
00576 }
00577 
00587 static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
00588 {
00589     int   cnt, i, lastPos = -1;
00590     float   *pIn, *pOut;
00591 
00592     for (cnt = 0; cnt < numComponents; cnt++){
00593         lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);
00594         pIn = pComponent[cnt].coef;
00595         pOut = &(pSpectrum[pComponent[cnt].pos]);
00596 
00597         for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
00598             pOut[i] += pIn[i];
00599     }
00600 
00601     return lastPos;
00602 }
00603 
00604 
00605 #define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
00606 
00607 static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
00608 {
00609     int    i, band, nsample, s1, s2;
00610     float    c1, c2;
00611     float    mc1_l, mc1_r, mc2_l, mc2_r;
00612 
00613     for (i=0,band = 0; band < 4*256; band+=256,i++) {
00614         s1 = pPrevCode[i];
00615         s2 = pCurrCode[i];
00616         nsample = 0;
00617 
00618         if (s1 != s2) {
00619             /* Selector value changed, interpolation needed. */
00620             mc1_l = matrixCoeffs[s1*2];
00621             mc1_r = matrixCoeffs[s1*2+1];
00622             mc2_l = matrixCoeffs[s2*2];
00623             mc2_r = matrixCoeffs[s2*2+1];
00624 
00625             /* Interpolation is done over the first eight samples. */
00626             for(; nsample < 8; nsample++) {
00627                 c1 = su1[band+nsample];
00628                 c2 = su2[band+nsample];
00629                 c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
00630                 su1[band+nsample] = c2;
00631                 su2[band+nsample] = c1 * 2.0 - c2;
00632             }
00633         }
00634 
00635         /* Apply the matrix without interpolation. */
00636         switch (s2) {
00637             case 0:     /* M/S decoding */
00638                 for (; nsample < 256; nsample++) {
00639                     c1 = su1[band+nsample];
00640                     c2 = su2[band+nsample];
00641                     su1[band+nsample] = c2 * 2.0;
00642                     su2[band+nsample] = (c1 - c2) * 2.0;
00643                 }
00644                 break;
00645 
00646             case 1:
00647                 for (; nsample < 256; nsample++) {
00648                     c1 = su1[band+nsample];
00649                     c2 = su2[band+nsample];
00650                     su1[band+nsample] = (c1 + c2) * 2.0;
00651                     su2[band+nsample] = c2 * -2.0;
00652                 }
00653                 break;
00654             case 2:
00655             case 3:
00656                 for (; nsample < 256; nsample++) {
00657                     c1 = su1[band+nsample];
00658                     c2 = su2[band+nsample];
00659                     su1[band+nsample] = c1 + c2;
00660                     su2[band+nsample] = c1 - c2;
00661                 }
00662                 break;
00663             default:
00664                 assert(0);
00665         }
00666     }
00667 }
00668 
00669 static void getChannelWeights (int indx, int flag, float ch[2]){
00670 
00671     if (indx == 7) {
00672         ch[0] = 1.0;
00673         ch[1] = 1.0;
00674     } else {
00675         ch[0] = (float)(indx & 7) / 7.0;
00676         ch[1] = sqrt(2 - ch[0]*ch[0]);
00677         if(flag)
00678             FFSWAP(float, ch[0], ch[1]);
00679     }
00680 }
00681 
00682 static void channelWeighting (float *su1, float *su2, int *p3)
00683 {
00684     int   band, nsample;
00685     /* w[x][y] y=0 is left y=1 is right */
00686     float w[2][2];
00687 
00688     if (p3[1] != 7 || p3[3] != 7){
00689         getChannelWeights(p3[1], p3[0], w[0]);
00690         getChannelWeights(p3[3], p3[2], w[1]);
00691 
00692         for(band = 1; band < 4; band++) {
00693             /* scale the channels by the weights */
00694             for(nsample = 0; nsample < 8; nsample++) {
00695                 su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
00696                 su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
00697             }
00698 
00699             for(; nsample < 256; nsample++) {
00700                 su1[band*256+nsample] *= w[1][0];
00701                 su2[band*256+nsample] *= w[1][1];
00702             }
00703         }
00704     }
00705 }
00706 
00707 
00719 static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
00720 {
00721     int   band, result=0, numSubbands, lastTonal, numBands;
00722 
00723     if (codingMode == JOINT_STEREO && channelNum == 1) {
00724         if (get_bits(gb,2) != 3) {
00725             av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
00726             return -1;
00727         }
00728     } else {
00729         if (get_bits(gb,6) != 0x28) {
00730             av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
00731             return -1;
00732         }
00733     }
00734 
00735     /* number of coded QMF bands */
00736     pSnd->bandsCoded = get_bits(gb,2);
00737 
00738     result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
00739     if (result) return result;
00740 
00741     pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
00742     if (pSnd->numComponents == -1) return -1;
00743 
00744     numSubbands = decodeSpectrum (gb, pSnd->spectrum);
00745 
00746     /* Merge the decoded spectrum and tonal components. */
00747     lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
00748 
00749 
00750     /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
00751     numBands = (subbandTab[numSubbands] - 1) >> 8;
00752     if (lastTonal >= 0)
00753         numBands = FFMAX((lastTonal + 256) >> 8, numBands);
00754 
00755 
00756     /* Reconstruct time domain samples. */
00757     for (band=0; band<4; band++) {
00758         /* Perform the IMDCT step without overlapping. */
00759         if (band <= numBands) {
00760             IMLT(&(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
00761         } else
00762             memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
00763 
00764         /* gain compensation and overlapping */
00765         gainCompensateAndOverlap (pSnd->IMDCT_buf, &(pSnd->prevFrame[band*256]), &(pOut[band*256]),
00766                                     &((pSnd->gainBlock[1 - (pSnd->gcBlkSwitch)]).gBlock[band]),
00767                                     &((pSnd->gainBlock[pSnd->gcBlkSwitch]).gBlock[band]));
00768     }
00769 
00770     /* Swap the gain control buffers for the next frame. */
00771     pSnd->gcBlkSwitch ^= 1;
00772 
00773     return 0;
00774 }
00775 
00783 static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf)
00784 {
00785     int   result, i;
00786     float   *p1, *p2, *p3, *p4;
00787     uint8_t *ptr1;
00788 
00789     if (q->codingMode == JOINT_STEREO) {
00790 
00791         /* channel coupling mode */
00792         /* decode Sound Unit 1 */
00793         init_get_bits(&q->gb,databuf,q->bits_per_frame);
00794 
00795         result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, q->outSamples, 0, JOINT_STEREO);
00796         if (result != 0)
00797             return (result);
00798 
00799         /* Framedata of the su2 in the joint-stereo mode is encoded in
00800          * reverse byte order so we need to swap it first. */
00801         if (databuf == q->decoded_bytes_buffer) {
00802             uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1;
00803             ptr1 = q->decoded_bytes_buffer;
00804             for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
00805                 FFSWAP(uint8_t,*ptr1,*ptr2);
00806             }
00807         } else {
00808             const uint8_t *ptr2 = databuf+q->bytes_per_frame-1;
00809             for (i = 0; i < q->bytes_per_frame; i++)
00810                 q->decoded_bytes_buffer[i] = *ptr2--;
00811         }
00812 
00813         /* Skip the sync codes (0xF8). */
00814         ptr1 = q->decoded_bytes_buffer;
00815         for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
00816             if (i >= q->bytes_per_frame)
00817                 return -1;
00818         }
00819 
00820 
00821         /* set the bitstream reader at the start of the second Sound Unit*/
00822         init_get_bits(&q->gb,ptr1,q->bits_per_frame);
00823 
00824         /* Fill the Weighting coeffs delay buffer */
00825         memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
00826         q->weighting_delay[4] = get_bits1(&q->gb);
00827         q->weighting_delay[5] = get_bits(&q->gb,3);
00828 
00829         for (i = 0; i < 4; i++) {
00830             q->matrix_coeff_index_prev[i] = q->matrix_coeff_index_now[i];
00831             q->matrix_coeff_index_now[i] = q->matrix_coeff_index_next[i];
00832             q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
00833         }
00834 
00835         /* Decode Sound Unit 2. */
00836         result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], &q->outSamples[1024], 1, JOINT_STEREO);
00837         if (result != 0)
00838             return (result);
00839 
00840         /* Reconstruct the channel coefficients. */
00841         reverseMatrixing(q->outSamples, &q->outSamples[1024], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
00842 
00843         channelWeighting(q->outSamples, &q->outSamples[1024], q->weighting_delay);
00844 
00845     } else {
00846         /* normal stereo mode or mono */
00847         /* Decode the channel sound units. */
00848         for (i=0 ; i<q->channels ; i++) {
00849 
00850             /* Set the bitstream reader at the start of a channel sound unit. */
00851             init_get_bits(&q->gb, databuf+((i*q->bytes_per_frame)/q->channels), (q->bits_per_frame)/q->channels);
00852 
00853             result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], &q->outSamples[i*1024], i, q->codingMode);
00854             if (result != 0)
00855                 return (result);
00856         }
00857     }
00858 
00859     /* Apply the iQMF synthesis filter. */
00860     p1= q->outSamples;
00861     for (i=0 ; i<q->channels ; i++) {
00862         p2= p1+256;
00863         p3= p2+256;
00864         p4= p3+256;
00865         iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
00866         iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
00867         iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
00868         p1 +=1024;
00869     }
00870 
00871     return 0;
00872 }
00873 
00874 
00881 static int atrac3_decode_frame(AVCodecContext *avctx,
00882             void *data, int *data_size,
00883             const uint8_t *buf, int buf_size) {
00884     ATRAC3Context *q = avctx->priv_data;
00885     int result = 0, i;
00886     const uint8_t* databuf;
00887     int16_t* samples = data;
00888 
00889     if (buf_size < avctx->block_align)
00890         return buf_size;
00891 
00892     /* Check if we need to descramble and what buffer to pass on. */
00893     if (q->scrambled_stream) {
00894         decode_bytes(buf, q->decoded_bytes_buffer, avctx->block_align);
00895         databuf = q->decoded_bytes_buffer;
00896     } else {
00897         databuf = buf;
00898     }
00899 
00900     result = decodeFrame(q, databuf);
00901 
00902     if (result != 0) {
00903         av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
00904         return -1;
00905     }
00906 
00907     if (q->channels == 1) {
00908         /* mono */
00909         for (i = 0; i<1024; i++)
00910             samples[i] = av_clip_int16(round(q->outSamples[i]));
00911         *data_size = 1024 * sizeof(int16_t);
00912     } else {
00913         /* stereo */
00914         for (i = 0; i < 1024; i++) {
00915             samples[i*2] = av_clip_int16(round(q->outSamples[i]));
00916             samples[i*2+1] = av_clip_int16(round(q->outSamples[1024+i]));
00917         }
00918         *data_size = 2048 * sizeof(int16_t);
00919     }
00920 
00921     return avctx->block_align;
00922 }
00923 
00924 
00931 static av_cold int atrac3_decode_init(AVCodecContext *avctx)
00932 {
00933     int i;
00934     const uint8_t *edata_ptr = avctx->extradata;
00935     ATRAC3Context *q = avctx->priv_data;
00936 
00937     /* Take data from the AVCodecContext (RM container). */
00938     q->sample_rate = avctx->sample_rate;
00939     q->channels = avctx->channels;
00940     q->bit_rate = avctx->bit_rate;
00941     q->bits_per_frame = avctx->block_align * 8;
00942     q->bytes_per_frame = avctx->block_align;
00943 
00944     /* Take care of the codec-specific extradata. */
00945     if (avctx->extradata_size == 14) {
00946         /* Parse the extradata, WAV format */
00947         av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown value always 1
00948         q->samples_per_channel = bytestream_get_le32(&edata_ptr);
00949         q->codingMode = bytestream_get_le16(&edata_ptr);
00950         av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr));  //Dupe of coding mode
00951         q->frame_factor = bytestream_get_le16(&edata_ptr);  //Unknown always 1
00952         av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr));  //Unknown always 0
00953 
00954         /* setup */
00955         q->samples_per_frame = 1024 * q->channels;
00956         q->atrac3version = 4;
00957         q->delay = 0x88E;
00958         if (q->codingMode)
00959             q->codingMode = JOINT_STEREO;
00960         else
00961             q->codingMode = STEREO;
00962 
00963         q->scrambled_stream = 0;
00964 
00965         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)) {
00966         } else {
00967             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);
00968             return -1;
00969         }
00970 
00971     } else if (avctx->extradata_size == 10) {
00972         /* Parse the extradata, RM format. */
00973         q->atrac3version = bytestream_get_be32(&edata_ptr);
00974         q->samples_per_frame = bytestream_get_be16(&edata_ptr);
00975         q->delay = bytestream_get_be16(&edata_ptr);
00976         q->codingMode = bytestream_get_be16(&edata_ptr);
00977 
00978         q->samples_per_channel = q->samples_per_frame / q->channels;
00979         q->scrambled_stream = 1;
00980 
00981     } else {
00982         av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
00983     }
00984     /* Check the extradata. */
00985 
00986     if (q->atrac3version != 4) {
00987         av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
00988         return -1;
00989     }
00990 
00991     if (q->samples_per_frame != 1024 && q->samples_per_frame != 2048) {
00992         av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
00993         return -1;
00994     }
00995 
00996     if (q->delay != 0x88E) {
00997         av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
00998         return -1;
00999     }
01000 
01001     if (q->codingMode == STEREO) {
01002         av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
01003     } else if (q->codingMode == JOINT_STEREO) {
01004         av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
01005     } else {
01006         av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
01007         return -1;
01008     }
01009 
01010     if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
01011         av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
01012         return -1;
01013     }
01014 
01015 
01016     if(avctx->block_align >= UINT_MAX/2)
01017         return -1;
01018 
01019     /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
01020      * this is for the bitstream reader. */
01021     if ((q->decoded_bytes_buffer = av_mallocz((avctx->block_align+(4-avctx->block_align%4) + FF_INPUT_BUFFER_PADDING_SIZE)))  == NULL)
01022         return AVERROR(ENOMEM);
01023 
01024 
01025     /* Initialize the VLC tables. */
01026     for (i=0 ; i<7 ; i++) {
01027         init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
01028             huff_bits[i], 1, 1,
01029             huff_codes[i], 1, 1, INIT_VLC_USE_STATIC);
01030     }
01031 
01032     init_atrac3_transforms(q);
01033 
01034     /* Generate the scale factors. */
01035     for (i=0 ; i<64 ; i++)
01036         SFTable[i] = pow(2.0, (i - 15) / 3.0);
01037 
01038     /* Generate gain tables. */
01039     for (i=0 ; i<16 ; i++)
01040         gain_tab1[i] = powf (2.0, (4 - i));
01041 
01042     for (i=-15 ; i<16 ; i++)
01043         gain_tab2[i+15] = powf (2.0, i * -0.125);
01044 
01045     /* init the joint-stereo decoding data */
01046     q->weighting_delay[0] = 0;
01047     q->weighting_delay[1] = 7;
01048     q->weighting_delay[2] = 0;
01049     q->weighting_delay[3] = 7;
01050     q->weighting_delay[4] = 0;
01051     q->weighting_delay[5] = 7;
01052 
01053     for (i=0; i<4; i++) {
01054         q->matrix_coeff_index_prev[i] = 3;
01055         q->matrix_coeff_index_now[i] = 3;
01056         q->matrix_coeff_index_next[i] = 3;
01057     }
01058 
01059     dsputil_init(&dsp, avctx);
01060 
01061     q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
01062     if (!q->pUnits) {
01063         av_free(q->decoded_bytes_buffer);
01064         return AVERROR(ENOMEM);
01065     }
01066 
01067     avctx->sample_fmt = SAMPLE_FMT_S16;
01068     return 0;
01069 }
01070 
01071 
01072 AVCodec atrac3_decoder =
01073 {
01074     .name = "atrac3",
01075     .type = CODEC_TYPE_AUDIO,
01076     .id = CODEC_ID_ATRAC3,
01077     .priv_data_size = sizeof(ATRAC3Context),
01078     .init = atrac3_decode_init,
01079     .close = atrac3_decode_close,
01080     .decode = atrac3_decode_frame,
01081     .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
01082 };

Generated on Sat Feb 16 2013 09:23:11 for ffmpeg by  doxygen 1.7.1