atrac3.c
Go to the documentation of this file.
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 "libavutil/float_dsp.h"
40 #include "avcodec.h"
41 #include "bytestream.h"
42 #include "fft.h"
43 #include "fmtconvert.h"
44 #include "get_bits.h"
45 #include "internal.h"
46 
47 #include "atrac.h"
48 #include "atrac3data.h"
49 
50 #define JOINT_STEREO 0x12
51 #define STEREO 0x2
52 
53 #define SAMPLES_PER_FRAME 1024
54 #define MDCT_SIZE 512
55 
56 typedef struct GainInfo {
58  int lev_code[8];
59  int loc_code[8];
60 } GainInfo;
61 
62 typedef struct GainBlock {
64 } GainBlock;
65 
66 typedef struct TonalComponent {
67  int pos;
68  int num_coefs;
69  float coef[8];
71 
72 typedef struct ChannelUnit {
79 
82 
83  float delay_buf1[46];
84  float delay_buf2[46];
85  float delay_buf3[46];
86 } ChannelUnit;
87 
88 typedef struct ATRAC3Context {
92 
94 
97 
98 
104 
105 
107  float temp_buf[1070];
109 
110 
113 
117 } ATRAC3Context;
118 
120 static VLC_TYPE atrac3_vlc_table[4096][2];
122 static float gain_tab1[16];
123 static float gain_tab2[31];
124 
125 
126 /*
127  * Regular 512 points IMDCT without overlapping, with the exception of the
128  * swapping of odd bands caused by the reverse spectra of the QMF.
129  *
130  * @param odd_band 1 if the band is an odd band
131  */
132 static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
133 {
134  int i;
135 
136  if (odd_band) {
145  for (i = 0; i < 128; i++)
146  FFSWAP(float, input[i], input[255 - i]);
147  }
148 
149  q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
150 
151  /* Perform windowing on the output. */
152  q->fdsp.vector_fmul(output, output, mdct_window, MDCT_SIZE);
153 }
154 
155 /*
156  * indata descrambling, only used for data coming from the rm container
157  */
158 static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
159 {
160  int i, off;
161  uint32_t c;
162  const uint32_t *buf;
163  uint32_t *output = (uint32_t *)out;
164 
165  off = (intptr_t)input & 3;
166  buf = (const uint32_t *)(input - off);
167  if (off)
168  c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
169  else
170  c = av_be2ne32(0x537F6103U);
171  bytes += 3 + off;
172  for (i = 0; i < bytes / 4; i++)
173  output[i] = c ^ buf[i];
174 
175  if (off)
176  av_log_ask_for_sample(NULL, "Offset of %d not handled.\n", off);
177 
178  return off;
179 }
180 
181 static av_cold void init_atrac3_window(void)
182 {
183  int i, j;
184 
185  /* generate the mdct window, for details see
186  * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
187  for (i = 0, j = 255; i < 128; i++, j--) {
188  float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
189  float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
190  float w = 0.5 * (wi * wi + wj * wj);
191  mdct_window[i] = mdct_window[511 - i] = wi / w;
192  mdct_window[j] = mdct_window[511 - j] = wj / w;
193  }
194 }
195 
197 {
198  ATRAC3Context *q = avctx->priv_data;
199 
200  av_free(q->units);
202 
203  ff_mdct_end(&q->mdct_ctx);
204 
205  return 0;
206 }
207 
208 /*
209  * Mantissa decoding
210  *
211  * @param selector which table the output values are coded with
212  * @param coding_flag constant length coding or variable length coding
213  * @param mantissas mantissa output table
214  * @param num_codes number of values to get
215  */
216 static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
217  int coding_flag, int *mantissas,
218  int num_codes)
219 {
220  int i, code, huff_symb;
221 
222  if (selector == 1)
223  num_codes /= 2;
224 
225  if (coding_flag != 0) {
226  /* constant length coding (CLC) */
227  int num_bits = clc_length_tab[selector];
228 
229  if (selector > 1) {
230  for (i = 0; i < num_codes; i++) {
231  if (num_bits)
232  code = get_sbits(gb, num_bits);
233  else
234  code = 0;
235  mantissas[i] = code;
236  }
237  } else {
238  for (i = 0; i < num_codes; i++) {
239  if (num_bits)
240  code = get_bits(gb, num_bits); // num_bits is always 4 in this case
241  else
242  code = 0;
243  mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
244  mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
245  }
246  }
247  } else {
248  /* variable length coding (VLC) */
249  if (selector != 1) {
250  for (i = 0; i < num_codes; i++) {
251  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
252  spectral_coeff_tab[selector-1].bits, 3);
253  huff_symb += 1;
254  code = huff_symb >> 1;
255  if (huff_symb & 1)
256  code = -code;
257  mantissas[i] = code;
258  }
259  } else {
260  for (i = 0; i < num_codes; i++) {
261  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
262  spectral_coeff_tab[selector - 1].bits, 3);
263  mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
264  mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
265  }
266  }
267  }
268 }
269 
270 /*
271  * Restore the quantized band spectrum coefficients
272  *
273  * @return subband count, fix for broken specification/files
274  */
275 static int decode_spectrum(GetBitContext *gb, float *output)
276 {
277  int num_subbands, coding_mode, i, j, first, last, subband_size;
278  int subband_vlc_index[32], sf_index[32];
279  int mantissas[128];
280  float scale_factor;
281 
282  num_subbands = get_bits(gb, 5); // number of coded subbands
283  coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
284 
285  /* get the VLC selector table for the subbands, 0 means not coded */
286  for (i = 0; i <= num_subbands; i++)
287  subband_vlc_index[i] = get_bits(gb, 3);
288 
289  /* read the scale factor indexes from the stream */
290  for (i = 0; i <= num_subbands; i++) {
291  if (subband_vlc_index[i] != 0)
292  sf_index[i] = get_bits(gb, 6);
293  }
294 
295  for (i = 0; i <= num_subbands; i++) {
296  first = subband_tab[i ];
297  last = subband_tab[i + 1];
298 
299  subband_size = last - first;
300 
301  if (subband_vlc_index[i] != 0) {
302  /* decode spectral coefficients for this subband */
303  /* TODO: This can be done faster is several blocks share the
304  * same VLC selector (subband_vlc_index) */
305  read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
306  mantissas, subband_size);
307 
308  /* decode the scale factor for this subband */
309  scale_factor = ff_atrac_sf_table[sf_index[i]] *
310  inv_max_quant[subband_vlc_index[i]];
311 
312  /* inverse quantize the coefficients */
313  for (j = 0; first < last; first++, j++)
314  output[first] = mantissas[j] * scale_factor;
315  } else {
316  /* this subband was not coded, so zero the entire subband */
317  memset(output + first, 0, subband_size * sizeof(*output));
318  }
319  }
320 
321  /* clear the subbands that were not coded */
322  first = subband_tab[i];
323  memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
324  return num_subbands;
325 }
326 
327 /*
328  * Restore the quantized tonal components
329  *
330  * @param components tonal components
331  * @param num_bands number of coded bands
332  */
334  TonalComponent *components, int num_bands)
335 {
336  int i, b, c, m;
337  int nb_components, coding_mode_selector, coding_mode;
338  int band_flags[4], mantissa[8];
339  int component_count = 0;
340 
341  nb_components = get_bits(gb, 5);
342 
343  /* no tonal components */
344  if (nb_components == 0)
345  return 0;
346 
347  coding_mode_selector = get_bits(gb, 2);
348  if (coding_mode_selector == 2)
349  return AVERROR_INVALIDDATA;
350 
351  coding_mode = coding_mode_selector & 1;
352 
353  for (i = 0; i < nb_components; i++) {
354  int coded_values_per_component, quant_step_index;
355 
356  for (b = 0; b <= num_bands; b++)
357  band_flags[b] = get_bits1(gb);
358 
359  coded_values_per_component = get_bits(gb, 3);
360 
361  quant_step_index = get_bits(gb, 3);
362  if (quant_step_index <= 1)
363  return AVERROR_INVALIDDATA;
364 
365  if (coding_mode_selector == 3)
366  coding_mode = get_bits1(gb);
367 
368  for (b = 0; b < (num_bands + 1) * 4; b++) {
369  int coded_components;
370 
371  if (band_flags[b >> 2] == 0)
372  continue;
373 
374  coded_components = get_bits(gb, 3);
375 
376  for (c = 0; c < coded_components; c++) {
377  TonalComponent *cmp = &components[component_count];
378  int sf_index, coded_values, max_coded_values;
379  float scale_factor;
380 
381  sf_index = get_bits(gb, 6);
382  if (component_count >= 64)
383  return AVERROR_INVALIDDATA;
384 
385  cmp->pos = b * 64 + get_bits(gb, 6);
386 
387  max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
388  coded_values = coded_values_per_component + 1;
389  coded_values = FFMIN(max_coded_values, coded_values);
390 
391  scale_factor = ff_atrac_sf_table[sf_index] *
392  inv_max_quant[quant_step_index];
393 
394  read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
395  mantissa, coded_values);
396 
397  cmp->num_coefs = coded_values;
398 
399  /* inverse quant */
400  for (m = 0; m < coded_values; m++)
401  cmp->coef[m] = mantissa[m] * scale_factor;
402 
403  component_count++;
404  }
405  }
406  }
407 
408  return component_count;
409 }
410 
411 /*
412  * Decode gain parameters for the coded bands
413  *
414  * @param block the gainblock for the current band
415  * @param num_bands amount of coded bands
416  */
418  int num_bands)
419 {
420  int i, cf, num_data;
421  int *level, *loc;
422 
423  GainInfo *gain = block->g_block;
424 
425  for (i = 0; i <= num_bands; i++) {
426  num_data = get_bits(gb, 3);
427  gain[i].num_gain_data = num_data;
428  level = gain[i].lev_code;
429  loc = gain[i].loc_code;
430 
431  for (cf = 0; cf < gain[i].num_gain_data; cf++) {
432  level[cf] = get_bits(gb, 4);
433  loc [cf] = get_bits(gb, 5);
434  if (cf && loc[cf] <= loc[cf - 1])
435  return AVERROR_INVALIDDATA;
436  }
437  }
438 
439  /* Clear the unused blocks. */
440  for (; i < 4 ; i++)
441  gain[i].num_gain_data = 0;
442 
443  return 0;
444 }
445 
446 /*
447  * Apply gain parameters and perform the MDCT overlapping part
448  *
449  * @param input input buffer
450  * @param prev previous buffer to perform overlap against
451  * @param output output buffer
452  * @param gain1 current band gain info
453  * @param gain2 next band gain info
454  */
455 static void gain_compensate_and_overlap(float *input, float *prev,
456  float *output, GainInfo *gain1,
457  GainInfo *gain2)
458 {
459  float g1, g2, gain_inc;
460  int i, j, num_data, start_loc, end_loc;
461 
462 
463  if (gain2->num_gain_data == 0)
464  g1 = 1.0;
465  else
466  g1 = gain_tab1[gain2->lev_code[0]];
467 
468  if (gain1->num_gain_data == 0) {
469  for (i = 0; i < 256; i++)
470  output[i] = input[i] * g1 + prev[i];
471  } else {
472  num_data = gain1->num_gain_data;
473  gain1->loc_code[num_data] = 32;
474  gain1->lev_code[num_data] = 4;
475 
476  for (i = 0, j = 0; i < num_data; i++) {
477  start_loc = gain1->loc_code[i] * 8;
478  end_loc = start_loc + 8;
479 
480  g2 = gain_tab1[gain1->lev_code[i]];
481  gain_inc = gain_tab2[gain1->lev_code[i + 1] -
482  gain1->lev_code[i ] + 15];
483 
484  /* interpolate */
485  for (; j < start_loc; j++)
486  output[j] = (input[j] * g1 + prev[j]) * g2;
487 
488  /* interpolation is done over eight samples */
489  for (; j < end_loc; j++) {
490  output[j] = (input[j] * g1 + prev[j]) * g2;
491  g2 *= gain_inc;
492  }
493  }
494 
495  for (; j < 256; j++)
496  output[j] = input[j] * g1 + prev[j];
497  }
498 
499  /* Delay for the overlapping part. */
500  memcpy(prev, &input[256], 256 * sizeof(*prev));
501 }
502 
503 /*
504  * Combine the tonal band spectrum and regular band spectrum
505  *
506  * @param spectrum output spectrum buffer
507  * @param num_components number of tonal components
508  * @param components tonal components for this band
509  * @return position of the last tonal coefficient
510  */
511 static int add_tonal_components(float *spectrum, int num_components,
512  TonalComponent *components)
513 {
514  int i, j, last_pos = -1;
515  float *input, *output;
516 
517  for (i = 0; i < num_components; i++) {
518  last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
519  input = components[i].coef;
520  output = &spectrum[components[i].pos];
521 
522  for (j = 0; j < components[i].num_coefs; j++)
523  output[j] += input[j];
524  }
525 
526  return last_pos;
527 }
528 
529 #define INTERPOLATE(old, new, nsample) \
530  ((old) + (nsample) * 0.125 * ((new) - (old)))
531 
532 static void reverse_matrixing(float *su1, float *su2, int *prev_code,
533  int *curr_code)
534 {
535  int i, nsample, band;
536  float mc1_l, mc1_r, mc2_l, mc2_r;
537 
538  for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
539  int s1 = prev_code[i];
540  int s2 = curr_code[i];
541  nsample = band;
542 
543  if (s1 != s2) {
544  /* Selector value changed, interpolation needed. */
545  mc1_l = matrix_coeffs[s1 * 2 ];
546  mc1_r = matrix_coeffs[s1 * 2 + 1];
547  mc2_l = matrix_coeffs[s2 * 2 ];
548  mc2_r = matrix_coeffs[s2 * 2 + 1];
549 
550  /* Interpolation is done over the first eight samples. */
551  for (; nsample < band + 8; nsample++) {
552  float c1 = su1[nsample];
553  float c2 = su2[nsample];
554  c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
555  c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
556  su1[nsample] = c2;
557  su2[nsample] = c1 * 2.0 - c2;
558  }
559  }
560 
561  /* Apply the matrix without interpolation. */
562  switch (s2) {
563  case 0: /* M/S decoding */
564  for (; nsample < band + 256; nsample++) {
565  float c1 = su1[nsample];
566  float c2 = su2[nsample];
567  su1[nsample] = c2 * 2.0;
568  su2[nsample] = (c1 - c2) * 2.0;
569  }
570  break;
571  case 1:
572  for (; nsample < band + 256; nsample++) {
573  float c1 = su1[nsample];
574  float c2 = su2[nsample];
575  su1[nsample] = (c1 + c2) * 2.0;
576  su2[nsample] = c2 * -2.0;
577  }
578  break;
579  case 2:
580  case 3:
581  for (; nsample < band + 256; nsample++) {
582  float c1 = su1[nsample];
583  float c2 = su2[nsample];
584  su1[nsample] = c1 + c2;
585  su2[nsample] = c1 - c2;
586  }
587  break;
588  default:
589  assert(0);
590  }
591  }
592 }
593 
594 static void get_channel_weights(int index, int flag, float ch[2])
595 {
596  if (index == 7) {
597  ch[0] = 1.0;
598  ch[1] = 1.0;
599  } else {
600  ch[0] = (index & 7) / 7.0;
601  ch[1] = sqrt(2 - ch[0] * ch[0]);
602  if (flag)
603  FFSWAP(float, ch[0], ch[1]);
604  }
605 }
606 
607 static void channel_weighting(float *su1, float *su2, int *p3)
608 {
609  int band, nsample;
610  /* w[x][y] y=0 is left y=1 is right */
611  float w[2][2];
612 
613  if (p3[1] != 7 || p3[3] != 7) {
614  get_channel_weights(p3[1], p3[0], w[0]);
615  get_channel_weights(p3[3], p3[2], w[1]);
616 
617  for (band = 256; band < 4 * 256; band += 256) {
618  for (nsample = band; nsample < band + 8; nsample++) {
619  su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
620  su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
621  }
622  for(; nsample < band + 256; nsample++) {
623  su1[nsample] *= w[1][0];
624  su2[nsample] *= w[1][1];
625  }
626  }
627  }
628 }
629 
630 /*
631  * Decode a Sound Unit
632  *
633  * @param snd the channel unit to be used
634  * @param output the decoded samples before IQMF in float representation
635  * @param channel_num channel number
636  * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)
637  */
639  ChannelUnit *snd, float *output,
640  int channel_num, int coding_mode)
641 {
642  int band, ret, num_subbands, last_tonal, num_bands;
643  GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
644  GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
645 
646  if (coding_mode == JOINT_STEREO && channel_num == 1) {
647  if (get_bits(gb, 2) != 3) {
648  av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
649  return AVERROR_INVALIDDATA;
650  }
651  } else {
652  if (get_bits(gb, 6) != 0x28) {
653  av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
654  return AVERROR_INVALIDDATA;
655  }
656  }
657 
658  /* number of coded QMF bands */
659  snd->bands_coded = get_bits(gb, 2);
660 
661  ret = decode_gain_control(gb, gain2, snd->bands_coded);
662  if (ret)
663  return ret;
664 
666  snd->bands_coded);
667  if (snd->num_components < 0)
668  return snd->num_components;
669 
670  num_subbands = decode_spectrum(gb, snd->spectrum);
671 
672  /* Merge the decoded spectrum and tonal components. */
673  last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
674  snd->components);
675 
676 
677  /* calculate number of used MLT/QMF bands according to the amount of coded
678  spectral lines */
679  num_bands = (subband_tab[num_subbands] - 1) >> 8;
680  if (last_tonal >= 0)
681  num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
682 
683 
684  /* Reconstruct time domain samples. */
685  for (band = 0; band < 4; band++) {
686  /* Perform the IMDCT step without overlapping. */
687  if (band <= num_bands)
688  imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
689  else
690  memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
691 
692  /* gain compensation and overlapping */
694  &snd->prev_frame[band * 256],
695  &output[band * 256],
696  &gain1->g_block[band],
697  &gain2->g_block[band]);
698  }
699 
700  /* Swap the gain control buffers for the next frame. */
701  snd->gc_blk_switch ^= 1;
702 
703  return 0;
704 }
705 
706 static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
707  float **out_samples)
708 {
709  ATRAC3Context *q = avctx->priv_data;
710  int ret, i;
711  uint8_t *ptr1;
712 
713  if (q->coding_mode == JOINT_STEREO) {
714  /* channel coupling mode */
715  /* decode Sound Unit 1 */
716  init_get_bits(&q->gb, databuf, avctx->block_align * 8);
717 
718  ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
719  JOINT_STEREO);
720  if (ret != 0)
721  return ret;
722 
723  /* Framedata of the su2 in the joint-stereo mode is encoded in
724  * reverse byte order so we need to swap it first. */
725  if (databuf == q->decoded_bytes_buffer) {
726  uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
727  ptr1 = q->decoded_bytes_buffer;
728  for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
729  FFSWAP(uint8_t, *ptr1, *ptr2);
730  } else {
731  const uint8_t *ptr2 = databuf + avctx->block_align - 1;
732  for (i = 0; i < avctx->block_align; i++)
733  q->decoded_bytes_buffer[i] = *ptr2--;
734  }
735 
736  /* Skip the sync codes (0xF8). */
737  ptr1 = q->decoded_bytes_buffer;
738  for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
739  if (i >= avctx->block_align)
740  return AVERROR_INVALIDDATA;
741  }
742 
743 
744  /* set the bitstream reader at the start of the second Sound Unit*/
745  init_get_bits(&q->gb, ptr1, (avctx->block_align - i) * 8);
746 
747  /* Fill the Weighting coeffs delay buffer */
748  memmove(q->weighting_delay, &q->weighting_delay[2],
749  4 * sizeof(*q->weighting_delay));
750  q->weighting_delay[4] = get_bits1(&q->gb);
751  q->weighting_delay[5] = get_bits(&q->gb, 3);
752 
753  for (i = 0; i < 4; i++) {
756  q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
757  }
758 
759  /* Decode Sound Unit 2. */
760  ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
761  out_samples[1], 1, JOINT_STEREO);
762  if (ret != 0)
763  return ret;
764 
765  /* Reconstruct the channel coefficients. */
766  reverse_matrixing(out_samples[0], out_samples[1],
769 
770  channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
771  } else {
772  /* normal stereo mode or mono */
773  /* Decode the channel sound units. */
774  for (i = 0; i < avctx->channels; i++) {
775  /* Set the bitstream reader at the start of a channel sound unit. */
776  init_get_bits(&q->gb,
777  databuf + i * avctx->block_align / avctx->channels,
778  avctx->block_align * 8 / avctx->channels);
779 
780  ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
781  out_samples[i], i, q->coding_mode);
782  if (ret != 0)
783  return ret;
784  }
785  }
786 
787  /* Apply the iQMF synthesis filter. */
788  for (i = 0; i < avctx->channels; i++) {
789  float *p1 = out_samples[i];
790  float *p2 = p1 + 256;
791  float *p3 = p2 + 256;
792  float *p4 = p3 + 256;
793  ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
794  ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
795  ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
796  }
797 
798  return 0;
799 }
800 
801 static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
802  int *got_frame_ptr, AVPacket *avpkt)
803 {
804  const uint8_t *buf = avpkt->data;
805  int buf_size = avpkt->size;
806  ATRAC3Context *q = avctx->priv_data;
807  int ret;
808  const uint8_t *databuf;
809 
810  if (buf_size < avctx->block_align) {
811  av_log(avctx, AV_LOG_ERROR,
812  "Frame too small (%d bytes). Truncated file?\n", buf_size);
813  return AVERROR_INVALIDDATA;
814  }
815 
816  /* get output buffer */
818  if ((ret = ff_get_buffer(avctx, &q->frame)) < 0) {
819  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
820  return ret;
821  }
822 
823  /* Check if we need to descramble and what buffer to pass on. */
824  if (q->scrambled_stream) {
826  databuf = q->decoded_bytes_buffer;
827  } else {
828  databuf = buf;
829  }
830 
831  ret = decode_frame(avctx, databuf, (float **)q->frame.extended_data);
832  if (ret) {
833  av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
834  return ret;
835  }
836 
837  *got_frame_ptr = 1;
838  *(AVFrame *)data = q->frame;
839 
840  return avctx->block_align;
841 }
842 
843 static void atrac3_init_static_data(AVCodec *codec)
844 {
845  int i;
846 
849 
850  /* Initialize the VLC tables. */
851  for (i = 0; i < 7; i++) {
852  spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
853  spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
854  atrac3_vlc_offs[i ];
855  init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
856  huff_bits[i], 1, 1,
858  }
859 
860  /* Generate gain tables */
861  for (i = 0; i < 16; i++)
862  gain_tab1[i] = powf(2.0, (4 - i));
863 
864  for (i = -15; i < 16; i++)
865  gain_tab2[i + 15] = powf(2.0, i * -0.125);
866 }
867 
869 {
870  int i, ret;
871  int version, delay, samples_per_frame, frame_factor;
872  const uint8_t *edata_ptr = avctx->extradata;
873  ATRAC3Context *q = avctx->priv_data;
874 
875  if (avctx->channels <= 0 || avctx->channels > 2) {
876  av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
877  return AVERROR(EINVAL);
878  }
879 
880  /* Take care of the codec-specific extradata. */
881  if (avctx->extradata_size == 14) {
882  /* Parse the extradata, WAV format */
883  av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
884  bytestream_get_le16(&edata_ptr)); // Unknown value always 1
885  edata_ptr += 4; // samples per channel
886  q->coding_mode = bytestream_get_le16(&edata_ptr);
887  av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
888  bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
889  frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
890  av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
891  bytestream_get_le16(&edata_ptr)); // Unknown always 0
892 
893  /* setup */
894  samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
895  version = 4;
896  delay = 0x88E;
898  q->scrambled_stream = 0;
899 
900  if (avctx->block_align != 96 * avctx->channels * frame_factor &&
901  avctx->block_align != 152 * avctx->channels * frame_factor &&
902  avctx->block_align != 192 * avctx->channels * frame_factor) {
903  av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
904  "configuration %d/%d/%d\n", avctx->block_align,
905  avctx->channels, frame_factor);
906  return AVERROR_INVALIDDATA;
907  }
908  } else if (avctx->extradata_size == 10) {
909  /* Parse the extradata, RM format. */
910  version = bytestream_get_be32(&edata_ptr);
911  samples_per_frame = bytestream_get_be16(&edata_ptr);
912  delay = bytestream_get_be16(&edata_ptr);
913  q->coding_mode = bytestream_get_be16(&edata_ptr);
914  q->scrambled_stream = 1;
915 
916  } else {
917  av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
918  avctx->extradata_size);
919  return AVERROR(EINVAL);
920  }
921 
922  /* Check the extradata */
923 
924  if (version != 4) {
925  av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
926  return AVERROR_INVALIDDATA;
927  }
928 
929  if (samples_per_frame != SAMPLES_PER_FRAME &&
930  samples_per_frame != SAMPLES_PER_FRAME * 2) {
931  av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
932  samples_per_frame);
933  return AVERROR_INVALIDDATA;
934  }
935 
936  if (delay != 0x88E) {
937  av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
938  delay);
939  return AVERROR_INVALIDDATA;
940  }
941 
942  if (q->coding_mode == STEREO)
943  av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
944  else if (q->coding_mode == JOINT_STEREO) {
945  if (avctx->channels != 2)
946  return AVERROR_INVALIDDATA;
947  av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
948  } else {
949  av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
950  q->coding_mode);
951  return AVERROR_INVALIDDATA;
952  }
953 
954  if (avctx->block_align >= UINT_MAX / 2)
955  return AVERROR(EINVAL);
956 
959  if (q->decoded_bytes_buffer == NULL)
960  return AVERROR(ENOMEM);
961 
963 
964  /* initialize the MDCT transform */
965  if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
966  av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
968  return ret;
969  }
970 
971  /* init the joint-stereo decoding data */
972  q->weighting_delay[0] = 0;
973  q->weighting_delay[1] = 7;
974  q->weighting_delay[2] = 0;
975  q->weighting_delay[3] = 7;
976  q->weighting_delay[4] = 0;
977  q->weighting_delay[5] = 7;
978 
979  for (i = 0; i < 4; i++) {
980  q->matrix_coeff_index_prev[i] = 3;
981  q->matrix_coeff_index_now[i] = 3;
982  q->matrix_coeff_index_next[i] = 3;
983  }
984 
986  ff_fmt_convert_init(&q->fmt_conv, avctx);
987 
988  q->units = av_mallocz(sizeof(*q->units) * avctx->channels);
989  if (!q->units) {
990  atrac3_decode_close(avctx);
991  return AVERROR(ENOMEM);
992  }
993 
995  avctx->coded_frame = &q->frame;
996 
997  return 0;
998 }
999 
1001  .name = "atrac3",
1002  .type = AVMEDIA_TYPE_AUDIO,
1003  .id = AV_CODEC_ID_ATRAC3,
1004  .priv_data_size = sizeof(ATRAC3Context),
1006  .init_static_data = atrac3_init_static_data,
1009  .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
1010  .long_name = NULL_IF_CONFIG_SMALL("Atrac 3 (Adaptive TRansform Acoustic Coding 3)"),
1011  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1013 };