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 Libav.
7  *
8  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
35 #include <math.h>
36 #include <stddef.h>
37 #include <stdio.h>
38 
39 #include "avcodec.h"
40 #include "get_bits.h"
41 #include "dsputil.h"
42 #include "bytestream.h"
43 #include "fft.h"
44 #include "fmtconvert.h"
45 
46 #include "atrac.h"
47 #include "atrac3data.h"
48 
49 #define JOINT_STEREO 0x12
50 #define STEREO 0x2
51 
52 #define SAMPLES_PER_FRAME 1024
53 #define MDCT_SIZE 512
54 
55 /* These structures are needed to store the parsed gain control data. */
56 typedef struct {
58  int levcode[8];
59  int loccode[8];
60 } gain_info;
61 
62 typedef struct {
63  gain_info gBlock[4];
64 } gain_block;
65 
66 typedef struct {
67  int pos;
68  int numCoefs;
69  float coef[8];
71 
72 typedef struct {
75  tonal_component components[64];
76  float prevFrame[SAMPLES_PER_FRAME];
78  gain_block gainBlock[2];
79 
80  DECLARE_ALIGNED(32, float, spectrum)[SAMPLES_PER_FRAME];
81  DECLARE_ALIGNED(32, float, IMDCT_buf)[SAMPLES_PER_FRAME];
82 
83  float delayBuf1[46];
84  float delayBuf2[46];
85  float delayBuf3[46];
86 } channel_unit;
87 
88 typedef struct {
92 
93  int channels;
95  int bit_rate;
99 
102  int pBs;
105 
106 
107  int matrix_coeff_index_prev[4];
108  int matrix_coeff_index_now[4];
109  int matrix_coeff_index_next[4];
110  int weighting_delay[6];
112 
113 
114  float *outSamples[2];
116  float tempBuf[1070];
118 
119 
121  int delay;
125 
128 } ATRAC3Context;
129 
132 static float gain_tab1[16];
133 static float gain_tab2[31];
135 
136 
146 static void IMLT(ATRAC3Context *q, float *pInput, float *pOutput, int odd_band)
147 {
148  int i;
149 
150  if (odd_band) {
160  for (i=0; i<128; i++)
161  FFSWAP(float, pInput[i], pInput[255-i]);
162  }
163 
164  q->mdct_ctx.imdct_calc(&q->mdct_ctx,pOutput,pInput);
165 
166  /* Perform windowing on the output. */
167  dsp.vector_fmul(pOutput, pOutput, mdct_window, MDCT_SIZE);
168 
169 }
170 
171 
180 static int decode_bytes(const uint8_t* inbuffer, uint8_t* out, int bytes){
181  int i, off;
182  uint32_t c;
183  const uint32_t* buf;
184  uint32_t* obuf = (uint32_t*) out;
185 
186  off = (intptr_t)inbuffer & 3;
187  buf = (const uint32_t*) (inbuffer - off);
188  c = av_be2ne32((0x537F6103 >> (off*8)) | (0x537F6103 << (32-(off*8))));
189  bytes += 3 + off;
190  for (i = 0; i < bytes/4; i++)
191  obuf[i] = c ^ buf[i];
192 
193  if (off)
194  av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off);
195 
196  return off;
197 }
198 
199 
200 static av_cold int init_atrac3_transforms(ATRAC3Context *q, int is_float) {
201  float enc_window[256];
202  int i;
203 
204  /* Generate the mdct window, for details see
205  * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
206  for (i=0 ; i<256; i++)
207  enc_window[i] = (sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0) * 0.5;
208 
209  if (!mdct_window[0])
210  for (i=0 ; i<256; i++) {
211  mdct_window[i] = enc_window[i]/(enc_window[i]*enc_window[i] + enc_window[255-i]*enc_window[255-i]);
212  mdct_window[511-i] = mdct_window[i];
213  }
214 
215  /* Initialize the MDCT transform. */
216  return ff_mdct_init(&q->mdct_ctx, 9, 1, is_float ? 1.0 / 32768 : 1.0);
217 }
218 
224 {
225  ATRAC3Context *q = avctx->priv_data;
226 
227  av_free(q->pUnits);
229  av_freep(&q->outSamples[0]);
230 
231  ff_mdct_end(&q->mdct_ctx);
232 
233  return 0;
234 }
235 
246 static void readQuantSpectralCoeffs (GetBitContext *gb, int selector, int codingFlag, int* mantissas, int numCodes)
247 {
248  int numBits, cnt, code, huffSymb;
249 
250  if (selector == 1)
251  numCodes /= 2;
252 
253  if (codingFlag != 0) {
254  /* constant length coding (CLC) */
255  numBits = CLCLengthTab[selector];
256 
257  if (selector > 1) {
258  for (cnt = 0; cnt < numCodes; cnt++) {
259  if (numBits)
260  code = get_sbits(gb, numBits);
261  else
262  code = 0;
263  mantissas[cnt] = code;
264  }
265  } else {
266  for (cnt = 0; cnt < numCodes; cnt++) {
267  if (numBits)
268  code = get_bits(gb, numBits); //numBits is always 4 in this case
269  else
270  code = 0;
271  mantissas[cnt*2] = seTab_0[code >> 2];
272  mantissas[cnt*2+1] = seTab_0[code & 3];
273  }
274  }
275  } else {
276  /* variable length coding (VLC) */
277  if (selector != 1) {
278  for (cnt = 0; cnt < numCodes; cnt++) {
279  huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
280  huffSymb += 1;
281  code = huffSymb >> 1;
282  if (huffSymb & 1)
283  code = -code;
284  mantissas[cnt] = code;
285  }
286  } else {
287  for (cnt = 0; cnt < numCodes; cnt++) {
288  huffSymb = get_vlc2(gb, spectral_coeff_tab[selector-1].table, spectral_coeff_tab[selector-1].bits, 3);
289  mantissas[cnt*2] = decTable1[huffSymb*2];
290  mantissas[cnt*2+1] = decTable1[huffSymb*2+1];
291  }
292  }
293  }
294 }
295 
304 static int decodeSpectrum (GetBitContext *gb, float *pOut)
305 {
306  int numSubbands, codingMode, cnt, first, last, subbWidth, *pIn;
307  int subband_vlc_index[32], SF_idxs[32];
308  int mantissas[128];
309  float SF;
310 
311  numSubbands = get_bits(gb, 5); // number of coded subbands
312  codingMode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
313 
314  /* Get the VLC selector table for the subbands, 0 means not coded. */
315  for (cnt = 0; cnt <= numSubbands; cnt++)
316  subband_vlc_index[cnt] = get_bits(gb, 3);
317 
318  /* Read the scale factor indexes from the stream. */
319  for (cnt = 0; cnt <= numSubbands; cnt++) {
320  if (subband_vlc_index[cnt] != 0)
321  SF_idxs[cnt] = get_bits(gb, 6);
322  }
323 
324  for (cnt = 0; cnt <= numSubbands; cnt++) {
325  first = subbandTab[cnt];
326  last = subbandTab[cnt+1];
327 
328  subbWidth = last - first;
329 
330  if (subband_vlc_index[cnt] != 0) {
331  /* Decode spectral coefficients for this subband. */
332  /* TODO: This can be done faster is several blocks share the
333  * same VLC selector (subband_vlc_index) */
334  readQuantSpectralCoeffs (gb, subband_vlc_index[cnt], codingMode, mantissas, subbWidth);
335 
336  /* Decode the scale factor for this subband. */
337  SF = ff_atrac_sf_table[SF_idxs[cnt]] * iMaxQuant[subband_vlc_index[cnt]];
338 
339  /* Inverse quantize the coefficients. */
340  for (pIn=mantissas ; first<last; first++, pIn++)
341  pOut[first] = *pIn * SF;
342  } else {
343  /* This subband was not coded, so zero the entire subband. */
344  memset(pOut+first, 0, subbWidth*sizeof(float));
345  }
346  }
347 
348  /* Clear the subbands that were not coded. */
349  first = subbandTab[cnt];
350  memset(pOut+first, 0, (SAMPLES_PER_FRAME - first) * sizeof(float));
351  return numSubbands;
352 }
353 
362 static int decodeTonalComponents (GetBitContext *gb, tonal_component *pComponent, int numBands)
363 {
364  int i,j,k,cnt;
365  int components, coding_mode_selector, coding_mode, coded_values_per_component;
366  int sfIndx, coded_values, max_coded_values, quant_step_index, coded_components;
367  int band_flags[4], mantissa[8];
368  float *pCoef;
369  float scalefactor;
370  int component_count = 0;
371 
372  components = get_bits(gb,5);
373 
374  /* no tonal components */
375  if (components == 0)
376  return 0;
377 
378  coding_mode_selector = get_bits(gb,2);
379  if (coding_mode_selector == 2)
380  return AVERROR_INVALIDDATA;
381 
382  coding_mode = coding_mode_selector & 1;
383 
384  for (i = 0; i < components; i++) {
385  for (cnt = 0; cnt <= numBands; cnt++)
386  band_flags[cnt] = get_bits1(gb);
387 
388  coded_values_per_component = get_bits(gb,3);
389 
390  quant_step_index = get_bits(gb,3);
391  if (quant_step_index <= 1)
392  return AVERROR_INVALIDDATA;
393 
394  if (coding_mode_selector == 3)
395  coding_mode = get_bits1(gb);
396 
397  for (j = 0; j < (numBands + 1) * 4; j++) {
398  if (band_flags[j >> 2] == 0)
399  continue;
400 
401  coded_components = get_bits(gb,3);
402 
403  for (k=0; k<coded_components; k++) {
404  sfIndx = get_bits(gb,6);
405  if (component_count >= 64)
406  return AVERROR_INVALIDDATA;
407  pComponent[component_count].pos = j * 64 + (get_bits(gb,6));
408  max_coded_values = SAMPLES_PER_FRAME - pComponent[component_count].pos;
409  coded_values = coded_values_per_component + 1;
410  coded_values = FFMIN(max_coded_values,coded_values);
411 
412  scalefactor = ff_atrac_sf_table[sfIndx] * iMaxQuant[quant_step_index];
413 
414  readQuantSpectralCoeffs(gb, quant_step_index, coding_mode, mantissa, coded_values);
415 
416  pComponent[component_count].numCoefs = coded_values;
417 
418  /* inverse quant */
419  pCoef = pComponent[component_count].coef;
420  for (cnt = 0; cnt < coded_values; cnt++)
421  pCoef[cnt] = mantissa[cnt] * scalefactor;
422 
423  component_count++;
424  }
425  }
426  }
427 
428  return component_count;
429 }
430 
439 static int decodeGainControl (GetBitContext *gb, gain_block *pGb, int numBands)
440 {
441  int i, cf, numData;
442  int *pLevel, *pLoc;
443 
444  gain_info *pGain = pGb->gBlock;
445 
446  for (i=0 ; i<=numBands; i++)
447  {
448  numData = get_bits(gb,3);
449  pGain[i].num_gain_data = numData;
450  pLevel = pGain[i].levcode;
451  pLoc = pGain[i].loccode;
452 
453  for (cf = 0; cf < numData; cf++){
454  pLevel[cf]= get_bits(gb,4);
455  pLoc [cf]= get_bits(gb,5);
456  if(cf && pLoc[cf] <= pLoc[cf-1])
457  return AVERROR_INVALIDDATA;
458  }
459  }
460 
461  /* Clear the unused blocks. */
462  for (; i<4 ; i++)
463  pGain[i].num_gain_data = 0;
464 
465  return 0;
466 }
467 
478 static void gainCompensateAndOverlap (float *pIn, float *pPrev, float *pOut, gain_info *pGain1, gain_info *pGain2)
479 {
480  /* gain compensation function */
481  float gain1, gain2, gain_inc;
482  int cnt, numdata, nsample, startLoc, endLoc;
483 
484 
485  if (pGain2->num_gain_data == 0)
486  gain1 = 1.0;
487  else
488  gain1 = gain_tab1[pGain2->levcode[0]];
489 
490  if (pGain1->num_gain_data == 0) {
491  for (cnt = 0; cnt < 256; cnt++)
492  pOut[cnt] = pIn[cnt] * gain1 + pPrev[cnt];
493  } else {
494  numdata = pGain1->num_gain_data;
495  pGain1->loccode[numdata] = 32;
496  pGain1->levcode[numdata] = 4;
497 
498  nsample = 0; // current sample = 0
499 
500  for (cnt = 0; cnt < numdata; cnt++) {
501  startLoc = pGain1->loccode[cnt] * 8;
502  endLoc = startLoc + 8;
503 
504  gain2 = gain_tab1[pGain1->levcode[cnt]];
505  gain_inc = gain_tab2[(pGain1->levcode[cnt+1] - pGain1->levcode[cnt])+15];
506 
507  /* interpolate */
508  for (; nsample < startLoc; nsample++)
509  pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
510 
511  /* interpolation is done over eight samples */
512  for (; nsample < endLoc; nsample++) {
513  pOut[nsample] = (pIn[nsample] * gain1 + pPrev[nsample]) * gain2;
514  gain2 *= gain_inc;
515  }
516  }
517 
518  for (; nsample < 256; nsample++)
519  pOut[nsample] = (pIn[nsample] * gain1) + pPrev[nsample];
520  }
521 
522  /* Delay for the overlapping part. */
523  memcpy(pPrev, &pIn[256], 256*sizeof(float));
524 }
525 
535 static int addTonalComponents (float *pSpectrum, int numComponents, tonal_component *pComponent)
536 {
537  int cnt, i, lastPos = -1;
538  float *pIn, *pOut;
539 
540  for (cnt = 0; cnt < numComponents; cnt++){
541  lastPos = FFMAX(pComponent[cnt].pos + pComponent[cnt].numCoefs, lastPos);
542  pIn = pComponent[cnt].coef;
543  pOut = &(pSpectrum[pComponent[cnt].pos]);
544 
545  for (i=0 ; i<pComponent[cnt].numCoefs ; i++)
546  pOut[i] += pIn[i];
547  }
548 
549  return lastPos;
550 }
551 
552 
553 #define INTERPOLATE(old,new,nsample) ((old) + (nsample)*0.125*((new)-(old)))
554 
555 static void reverseMatrixing(float *su1, float *su2, int *pPrevCode, int *pCurrCode)
556 {
557  int i, band, nsample, s1, s2;
558  float c1, c2;
559  float mc1_l, mc1_r, mc2_l, mc2_r;
560 
561  for (i=0,band = 0; band < 4*256; band+=256,i++) {
562  s1 = pPrevCode[i];
563  s2 = pCurrCode[i];
564  nsample = 0;
565 
566  if (s1 != s2) {
567  /* Selector value changed, interpolation needed. */
568  mc1_l = matrixCoeffs[s1*2];
569  mc1_r = matrixCoeffs[s1*2+1];
570  mc2_l = matrixCoeffs[s2*2];
571  mc2_r = matrixCoeffs[s2*2+1];
572 
573  /* Interpolation is done over the first eight samples. */
574  for(; nsample < 8; nsample++) {
575  c1 = su1[band+nsample];
576  c2 = su2[band+nsample];
577  c2 = c1 * INTERPOLATE(mc1_l,mc2_l,nsample) + c2 * INTERPOLATE(mc1_r,mc2_r,nsample);
578  su1[band+nsample] = c2;
579  su2[band+nsample] = c1 * 2.0 - c2;
580  }
581  }
582 
583  /* Apply the matrix without interpolation. */
584  switch (s2) {
585  case 0: /* M/S decoding */
586  for (; nsample < 256; nsample++) {
587  c1 = su1[band+nsample];
588  c2 = su2[band+nsample];
589  su1[band+nsample] = c2 * 2.0;
590  su2[band+nsample] = (c1 - c2) * 2.0;
591  }
592  break;
593 
594  case 1:
595  for (; nsample < 256; nsample++) {
596  c1 = su1[band+nsample];
597  c2 = su2[band+nsample];
598  su1[band+nsample] = (c1 + c2) * 2.0;
599  su2[band+nsample] = c2 * -2.0;
600  }
601  break;
602  case 2:
603  case 3:
604  for (; nsample < 256; nsample++) {
605  c1 = su1[band+nsample];
606  c2 = su2[band+nsample];
607  su1[band+nsample] = c1 + c2;
608  su2[band+nsample] = c1 - c2;
609  }
610  break;
611  default:
612  assert(0);
613  }
614  }
615 }
616 
617 static void getChannelWeights (int indx, int flag, float ch[2]){
618 
619  if (indx == 7) {
620  ch[0] = 1.0;
621  ch[1] = 1.0;
622  } else {
623  ch[0] = (float)(indx & 7) / 7.0;
624  ch[1] = sqrt(2 - ch[0]*ch[0]);
625  if(flag)
626  FFSWAP(float, ch[0], ch[1]);
627  }
628 }
629 
630 static void channelWeighting (float *su1, float *su2, int *p3)
631 {
632  int band, nsample;
633  /* w[x][y] y=0 is left y=1 is right */
634  float w[2][2];
635 
636  if (p3[1] != 7 || p3[3] != 7){
637  getChannelWeights(p3[1], p3[0], w[0]);
638  getChannelWeights(p3[3], p3[2], w[1]);
639 
640  for(band = 1; band < 4; band++) {
641  /* scale the channels by the weights */
642  for(nsample = 0; nsample < 8; nsample++) {
643  su1[band*256+nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample);
644  su2[band*256+nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample);
645  }
646 
647  for(; nsample < 256; nsample++) {
648  su1[band*256+nsample] *= w[1][0];
649  su2[band*256+nsample] *= w[1][1];
650  }
651  }
652  }
653 }
654 
655 
667 static int decodeChannelSoundUnit (ATRAC3Context *q, GetBitContext *gb, channel_unit *pSnd, float *pOut, int channelNum, int codingMode)
668 {
669  int band, result=0, numSubbands, lastTonal, numBands;
670 
671  if (codingMode == JOINT_STEREO && channelNum == 1) {
672  if (get_bits(gb,2) != 3) {
673  av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
674  return AVERROR_INVALIDDATA;
675  }
676  } else {
677  if (get_bits(gb,6) != 0x28) {
678  av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
679  return AVERROR_INVALIDDATA;
680  }
681  }
682 
683  /* number of coded QMF bands */
684  pSnd->bandsCoded = get_bits(gb,2);
685 
686  result = decodeGainControl (gb, &(pSnd->gainBlock[pSnd->gcBlkSwitch]), pSnd->bandsCoded);
687  if (result) return result;
688 
689  pSnd->numComponents = decodeTonalComponents (gb, pSnd->components, pSnd->bandsCoded);
690  if (pSnd->numComponents == -1) return -1;
691 
692  numSubbands = decodeSpectrum (gb, pSnd->spectrum);
693 
694  /* Merge the decoded spectrum and tonal components. */
695  lastTonal = addTonalComponents (pSnd->spectrum, pSnd->numComponents, pSnd->components);
696 
697 
698  /* calculate number of used MLT/QMF bands according to the amount of coded spectral lines */
699  numBands = (subbandTab[numSubbands] - 1) >> 8;
700  if (lastTonal >= 0)
701  numBands = FFMAX((lastTonal + 256) >> 8, numBands);
702 
703 
704  /* Reconstruct time domain samples. */
705  for (band=0; band<4; band++) {
706  /* Perform the IMDCT step without overlapping. */
707  if (band <= numBands) {
708  IMLT(q, &(pSnd->spectrum[band*256]), pSnd->IMDCT_buf, band&1);
709  } else
710  memset(pSnd->IMDCT_buf, 0, 512 * sizeof(float));
711 
712  /* gain compensation and overlapping */
713  gainCompensateAndOverlap(pSnd->IMDCT_buf, &pSnd->prevFrame[band * 256],
714  &pOut[band * 256],
715  &pSnd->gainBlock[1 - pSnd->gcBlkSwitch].gBlock[band],
716  &pSnd->gainBlock[ pSnd->gcBlkSwitch].gBlock[band]);
717  }
718 
719  /* Swap the gain control buffers for the next frame. */
720  pSnd->gcBlkSwitch ^= 1;
721 
722  return 0;
723 }
724 
732 static int decodeFrame(ATRAC3Context *q, const uint8_t* databuf,
733  float **out_samples)
734 {
735  int result, i;
736  float *p1, *p2, *p3, *p4;
737  uint8_t *ptr1;
738 
739  if (q->codingMode == JOINT_STEREO) {
740 
741  /* channel coupling mode */
742  /* decode Sound Unit 1 */
743  init_get_bits(&q->gb,databuf,q->bits_per_frame);
744 
745  result = decodeChannelSoundUnit(q,&q->gb, q->pUnits, out_samples[0], 0, JOINT_STEREO);
746  if (result != 0)
747  return result;
748 
749  /* Framedata of the su2 in the joint-stereo mode is encoded in
750  * reverse byte order so we need to swap it first. */
751  if (databuf == q->decoded_bytes_buffer) {
752  uint8_t *ptr2 = q->decoded_bytes_buffer+q->bytes_per_frame-1;
753  ptr1 = q->decoded_bytes_buffer;
754  for (i = 0; i < (q->bytes_per_frame/2); i++, ptr1++, ptr2--) {
755  FFSWAP(uint8_t,*ptr1,*ptr2);
756  }
757  } else {
758  const uint8_t *ptr2 = databuf+q->bytes_per_frame-1;
759  for (i = 0; i < q->bytes_per_frame; i++)
760  q->decoded_bytes_buffer[i] = *ptr2--;
761  }
762 
763  /* Skip the sync codes (0xF8). */
764  ptr1 = q->decoded_bytes_buffer;
765  for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
766  if (i >= q->bytes_per_frame)
767  return AVERROR_INVALIDDATA;
768  }
769 
770 
771  /* set the bitstream reader at the start of the second Sound Unit*/
772  init_get_bits(&q->gb,ptr1,q->bits_per_frame);
773 
774  /* Fill the Weighting coeffs delay buffer */
775  memmove(q->weighting_delay,&(q->weighting_delay[2]),4*sizeof(int));
776  q->weighting_delay[4] = get_bits1(&q->gb);
777  q->weighting_delay[5] = get_bits(&q->gb,3);
778 
779  for (i = 0; i < 4; i++) {
782  q->matrix_coeff_index_next[i] = get_bits(&q->gb,2);
783  }
784 
785  /* Decode Sound Unit 2. */
786  result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[1], out_samples[1], 1, JOINT_STEREO);
787  if (result != 0)
788  return result;
789 
790  /* Reconstruct the channel coefficients. */
791  reverseMatrixing(out_samples[0], out_samples[1], q->matrix_coeff_index_prev, q->matrix_coeff_index_now);
792 
793  channelWeighting(out_samples[0], out_samples[1], q->weighting_delay);
794 
795  } else {
796  /* normal stereo mode or mono */
797  /* Decode the channel sound units. */
798  for (i=0 ; i<q->channels ; i++) {
799 
800  /* Set the bitstream reader at the start of a channel sound unit. */
801  init_get_bits(&q->gb,
802  databuf + i * q->bytes_per_frame / q->channels,
803  q->bits_per_frame / q->channels);
804 
805  result = decodeChannelSoundUnit(q,&q->gb, &q->pUnits[i], out_samples[i], i, q->codingMode);
806  if (result != 0)
807  return result;
808  }
809  }
810 
811  /* Apply the iQMF synthesis filter. */
812  for (i=0 ; i<q->channels ; i++) {
813  p1 = out_samples[i];
814  p2= p1+256;
815  p3= p2+256;
816  p4= p3+256;
817  atrac_iqmf (p1, p2, 256, p1, q->pUnits[i].delayBuf1, q->tempBuf);
818  atrac_iqmf (p4, p3, 256, p3, q->pUnits[i].delayBuf2, q->tempBuf);
819  atrac_iqmf (p1, p3, 512, p1, q->pUnits[i].delayBuf3, q->tempBuf);
820  }
821 
822  return 0;
823 }
824 
825 
832 static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
833  int *got_frame_ptr, AVPacket *avpkt)
834 {
835  const uint8_t *buf = avpkt->data;
836  int buf_size = avpkt->size;
837  ATRAC3Context *q = avctx->priv_data;
838  int result;
839  const uint8_t* databuf;
840  float *samples_flt;
841  int16_t *samples_s16;
842 
843  if (buf_size < avctx->block_align) {
844  av_log(avctx, AV_LOG_ERROR,
845  "Frame too small (%d bytes). Truncated file?\n", buf_size);
846  return AVERROR_INVALIDDATA;
847  }
848 
849  /* get output buffer */
851  if ((result = avctx->get_buffer(avctx, &q->frame)) < 0) {
852  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
853  return result;
854  }
855  samples_flt = (float *)q->frame.data[0];
856  samples_s16 = (int16_t *)q->frame.data[0];
857 
858  /* Check if we need to descramble and what buffer to pass on. */
859  if (q->scrambled_stream) {
861  databuf = q->decoded_bytes_buffer;
862  } else {
863  databuf = buf;
864  }
865 
866  if (q->channels == 1 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT)
867  result = decodeFrame(q, databuf, &samples_flt);
868  else
869  result = decodeFrame(q, databuf, q->outSamples);
870 
871  if (result != 0) {
872  av_log(NULL,AV_LOG_ERROR,"Frame decoding error!\n");
873  return result;
874  }
875 
876  /* interleave */
877  if (q->channels == 2 && avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
878  q->fmt_conv.float_interleave(samples_flt,
879  (const float **)q->outSamples,
880  SAMPLES_PER_FRAME, 2);
881  } else if (avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
882  q->fmt_conv.float_to_int16_interleave(samples_s16,
883  (const float **)q->outSamples,
885  }
886 
887  *got_frame_ptr = 1;
888  *(AVFrame *)data = q->frame;
889 
890  return avctx->block_align;
891 }
892 
893 
901 {
902  int i, ret;
903  const uint8_t *edata_ptr = avctx->extradata;
904  ATRAC3Context *q = avctx->priv_data;
905  static VLC_TYPE atrac3_vlc_table[4096][2];
906  static int vlcs_initialized = 0;
907 
908  /* Take data from the AVCodecContext (RM container). */
909  q->sample_rate = avctx->sample_rate;
910  q->channels = avctx->channels;
911  q->bit_rate = avctx->bit_rate;
912  q->bits_per_frame = avctx->block_align * 8;
913  q->bytes_per_frame = avctx->block_align;
914 
915  /* Take care of the codec-specific extradata. */
916  if (avctx->extradata_size == 14) {
917  /* Parse the extradata, WAV format */
918  av_log(avctx,AV_LOG_DEBUG,"[0-1] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown value always 1
919  q->samples_per_channel = bytestream_get_le32(&edata_ptr);
920  q->codingMode = bytestream_get_le16(&edata_ptr);
921  av_log(avctx,AV_LOG_DEBUG,"[8-9] %d\n",bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
922  q->frame_factor = bytestream_get_le16(&edata_ptr); //Unknown always 1
923  av_log(avctx,AV_LOG_DEBUG,"[12-13] %d\n",bytestream_get_le16(&edata_ptr)); //Unknown always 0
924 
925  /* setup */
927  q->atrac3version = 4;
928  q->delay = 0x88E;
929  if (q->codingMode)
931  else
932  q->codingMode = STEREO;
933 
934  q->scrambled_stream = 0;
935 
936  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)) {
937  } else {
938  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);
939  return AVERROR_INVALIDDATA;
940  }
941 
942  } else if (avctx->extradata_size == 10) {
943  /* Parse the extradata, RM format. */
944  q->atrac3version = bytestream_get_be32(&edata_ptr);
945  q->samples_per_frame = bytestream_get_be16(&edata_ptr);
946  q->delay = bytestream_get_be16(&edata_ptr);
947  q->codingMode = bytestream_get_be16(&edata_ptr);
948 
950  q->scrambled_stream = 1;
951 
952  } else {
953  av_log(NULL,AV_LOG_ERROR,"Unknown extradata size %d.\n",avctx->extradata_size);
954  }
955  /* Check the extradata. */
956 
957  if (q->atrac3version != 4) {
958  av_log(avctx,AV_LOG_ERROR,"Version %d != 4.\n",q->atrac3version);
959  return AVERROR_INVALIDDATA;
960  }
961 
963  av_log(avctx,AV_LOG_ERROR,"Unknown amount of samples per frame %d.\n",q->samples_per_frame);
964  return AVERROR_INVALIDDATA;
965  }
966 
967  if (q->delay != 0x88E) {
968  av_log(avctx,AV_LOG_ERROR,"Unknown amount of delay %x != 0x88E.\n",q->delay);
969  return AVERROR_INVALIDDATA;
970  }
971 
972  if (q->codingMode == STEREO) {
973  av_log(avctx,AV_LOG_DEBUG,"Normal stereo detected.\n");
974  } else if (q->codingMode == JOINT_STEREO) {
975  av_log(avctx,AV_LOG_DEBUG,"Joint stereo detected.\n");
976  } else {
977  av_log(avctx,AV_LOG_ERROR,"Unknown channel coding mode %x!\n",q->codingMode);
978  return AVERROR_INVALIDDATA;
979  }
980 
981  if (avctx->channels <= 0 || avctx->channels > 2 /*|| ((avctx->channels * 1024) != q->samples_per_frame)*/) {
982  av_log(avctx,AV_LOG_ERROR,"Channel configuration error!\n");
983  return AVERROR(EINVAL);
984  }
985 
986 
987  if(avctx->block_align >= UINT_MAX/2)
988  return AVERROR(EINVAL);
989 
990  /* Pad the data buffer with FF_INPUT_BUFFER_PADDING_SIZE,
991  * this is for the bitstream reader. */
993  return AVERROR(ENOMEM);
994 
995 
996  /* Initialize the VLC tables. */
997  if (!vlcs_initialized) {
998  for (i=0 ; i<7 ; i++) {
999  spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
1000  spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] - atrac3_vlc_offs[i];
1001  init_vlc (&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
1002  huff_bits[i], 1, 1,
1004  }
1005  vlcs_initialized = 1;
1006  }
1007 
1008  if (avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT)
1009  avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
1010  else
1011  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1012 
1013  if ((ret = init_atrac3_transforms(q, avctx->sample_fmt == AV_SAMPLE_FMT_FLT))) {
1014  av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
1016  return ret;
1017  }
1018 
1020 
1021  /* Generate gain tables. */
1022  for (i=0 ; i<16 ; i++)
1023  gain_tab1[i] = powf (2.0, (4 - i));
1024 
1025  for (i=-15 ; i<16 ; i++)
1026  gain_tab2[i+15] = powf (2.0, i * -0.125);
1027 
1028  /* init the joint-stereo decoding data */
1029  q->weighting_delay[0] = 0;
1030  q->weighting_delay[1] = 7;
1031  q->weighting_delay[2] = 0;
1032  q->weighting_delay[3] = 7;
1033  q->weighting_delay[4] = 0;
1034  q->weighting_delay[5] = 7;
1035 
1036  for (i=0; i<4; i++) {
1037  q->matrix_coeff_index_prev[i] = 3;
1038  q->matrix_coeff_index_now[i] = 3;
1039  q->matrix_coeff_index_next[i] = 3;
1040  }
1041 
1042  dsputil_init(&dsp, avctx);
1043  ff_fmt_convert_init(&q->fmt_conv, avctx);
1044 
1045  q->pUnits = av_mallocz(sizeof(channel_unit)*q->channels);
1046  if (!q->pUnits) {
1047  atrac3_decode_close(avctx);
1048  return AVERROR(ENOMEM);
1049  }
1050 
1051  if (avctx->channels > 1 || avctx->sample_fmt == AV_SAMPLE_FMT_S16) {
1052  q->outSamples[0] = av_mallocz(SAMPLES_PER_FRAME * avctx->channels * sizeof(*q->outSamples[0]));
1053  q->outSamples[1] = q->outSamples[0] + SAMPLES_PER_FRAME;
1054  if (!q->outSamples[0]) {
1055  atrac3_decode_close(avctx);
1056  return AVERROR(ENOMEM);
1057  }
1058  }
1059 
1061  avctx->coded_frame = &q->frame;
1062 
1063  return 0;
1064 }
1065 
1066 
1068 {
1069  .name = "atrac3",
1070  .type = AVMEDIA_TYPE_AUDIO,
1071  .id = CODEC_ID_ATRAC3,
1072  .priv_data_size = sizeof(ATRAC3Context),
1076  .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
1077  .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
1078 };