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atrac3.c
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1 /*
2  * ATRAC3 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
25  * ATRAC3 compatible decoder.
26  * This decoder handles Sony's ATRAC3 data.
27  *
28  * Container formats used to store ATRAC3 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 "libavutil/attributes.h"
40 #include "libavutil/float_dsp.h"
41 #include "libavutil/libm.h"
42 #include "avcodec.h"
43 #include "bytestream.h"
44 #include "fft.h"
45 #include "fmtconvert.h"
46 #include "get_bits.h"
47 #include "internal.h"
48 
49 #include "atrac.h"
50 #include "atrac3data.h"
51 
52 #define JOINT_STEREO 0x12
53 #define STEREO 0x2
54 
55 #define SAMPLES_PER_FRAME 1024
56 #define MDCT_SIZE 512
57 
58 typedef struct GainBlock {
60 } GainBlock;
61 
62 typedef struct TonalComponent {
63  int pos;
64  int num_coefs;
65  float coef[8];
67 
68 typedef struct ChannelUnit {
75 
78 
79  float delay_buf1[46]; ///<qmf delay buffers
80  float delay_buf2[46];
81  float delay_buf3[46];
82 } ChannelUnit;
83 
84 typedef struct ATRAC3Context {
86  //@{
87  /** stream data */
89 
91  //@}
92  //@{
93  /** joint-stereo related variables */
98  //@}
99  //@{
100  /** data buffers */
102  float temp_buf[1070];
103  //@}
104  //@{
105  /** extradata */
107  //@}
108 
113 } ATRAC3Context;
114 
116 static VLC_TYPE atrac3_vlc_table[4096][2];
118 
119 /**
120  * Regular 512 points IMDCT without overlapping, with the exception of the
121  * swapping of odd bands caused by the reverse spectra of the QMF.
122  *
123  * @param odd_band 1 if the band is an odd band
124  */
125 static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
126 {
127  int i;
128 
129  if (odd_band) {
130  /**
131  * Reverse the odd bands before IMDCT, this is an effect of the QMF
132  * transform or it gives better compression to do it this way.
133  * FIXME: It should be possible to handle this in imdct_calc
134  * for that to happen a modification of the prerotation step of
135  * all SIMD code and C code is needed.
136  * Or fix the functions before so they generate a pre reversed spectrum.
137  */
138  for (i = 0; i < 128; i++)
139  FFSWAP(float, input[i], input[255 - i]);
140  }
141 
142  q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
143 
144  /* Perform windowing on the output. */
145  q->fdsp->vector_fmul(output, output, mdct_window, MDCT_SIZE);
146 }
147 
148 /*
149  * indata descrambling, only used for data coming from the rm container
150  */
151 static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
152 {
153  int i, off;
154  uint32_t c;
155  const uint32_t *buf;
156  uint32_t *output = (uint32_t *)out;
157 
158  off = (intptr_t)input & 3;
159  buf = (const uint32_t *)(input - off);
160  if (off)
161  c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
162  else
163  c = av_be2ne32(0x537F6103U);
164  bytes += 3 + off;
165  for (i = 0; i < bytes / 4; i++)
166  output[i] = c ^ buf[i];
167 
168  if (off)
169  avpriv_request_sample(NULL, "Offset of %d", off);
170 
171  return off;
172 }
173 
174 static av_cold void init_imdct_window(void)
175 {
176  int i, j;
177 
178  /* generate the mdct window, for details see
179  * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
180  for (i = 0, j = 255; i < 128; i++, j--) {
181  float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
182  float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
183  float w = 0.5 * (wi * wi + wj * wj);
184  mdct_window[i] = mdct_window[511 - i] = wi / w;
185  mdct_window[j] = mdct_window[511 - j] = wj / w;
186  }
187 }
188 
190 {
191  ATRAC3Context *q = avctx->priv_data;
192 
193  av_freep(&q->units);
195  av_freep(&q->fdsp);
196 
197  ff_mdct_end(&q->mdct_ctx);
198 
199  return 0;
200 }
201 
202 /**
203  * Mantissa decoding
204  *
205  * @param selector which table the output values are coded with
206  * @param coding_flag constant length coding or variable length coding
207  * @param mantissas mantissa output table
208  * @param num_codes number of values to get
209  */
210 static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
211  int coding_flag, int *mantissas,
212  int num_codes)
213 {
214  int i, code, huff_symb;
215 
216  if (selector == 1)
217  num_codes /= 2;
218 
219  if (coding_flag != 0) {
220  /* constant length coding (CLC) */
221  int num_bits = clc_length_tab[selector];
222 
223  if (selector > 1) {
224  for (i = 0; i < num_codes; i++) {
225  if (num_bits)
226  code = get_sbits(gb, num_bits);
227  else
228  code = 0;
229  mantissas[i] = code;
230  }
231  } else {
232  for (i = 0; i < num_codes; i++) {
233  if (num_bits)
234  code = get_bits(gb, num_bits); // num_bits is always 4 in this case
235  else
236  code = 0;
237  mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
238  mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
239  }
240  }
241  } else {
242  /* variable length coding (VLC) */
243  if (selector != 1) {
244  for (i = 0; i < num_codes; i++) {
245  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
246  spectral_coeff_tab[selector-1].bits, 3);
247  huff_symb += 1;
248  code = huff_symb >> 1;
249  if (huff_symb & 1)
250  code = -code;
251  mantissas[i] = code;
252  }
253  } else {
254  for (i = 0; i < num_codes; i++) {
255  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
256  spectral_coeff_tab[selector - 1].bits, 3);
257  mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
258  mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
259  }
260  }
261  }
262 }
263 
264 /**
265  * Restore the quantized band spectrum coefficients
266  *
267  * @return subband count, fix for broken specification/files
268  */
269 static int decode_spectrum(GetBitContext *gb, float *output)
270 {
271  int num_subbands, coding_mode, i, j, first, last, subband_size;
272  int subband_vlc_index[32], sf_index[32];
273  int mantissas[128];
274  float scale_factor;
275 
276  num_subbands = get_bits(gb, 5); // number of coded subbands
277  coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
278 
279  /* get the VLC selector table for the subbands, 0 means not coded */
280  for (i = 0; i <= num_subbands; i++)
281  subband_vlc_index[i] = get_bits(gb, 3);
282 
283  /* read the scale factor indexes from the stream */
284  for (i = 0; i <= num_subbands; i++) {
285  if (subband_vlc_index[i] != 0)
286  sf_index[i] = get_bits(gb, 6);
287  }
288 
289  for (i = 0; i <= num_subbands; i++) {
290  first = subband_tab[i ];
291  last = subband_tab[i + 1];
292 
293  subband_size = last - first;
294 
295  if (subband_vlc_index[i] != 0) {
296  /* decode spectral coefficients for this subband */
297  /* TODO: This can be done faster is several blocks share the
298  * same VLC selector (subband_vlc_index) */
299  read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
300  mantissas, subband_size);
301 
302  /* decode the scale factor for this subband */
303  scale_factor = ff_atrac_sf_table[sf_index[i]] *
304  inv_max_quant[subband_vlc_index[i]];
305 
306  /* inverse quantize the coefficients */
307  for (j = 0; first < last; first++, j++)
308  output[first] = mantissas[j] * scale_factor;
309  } else {
310  /* this subband was not coded, so zero the entire subband */
311  memset(output + first, 0, subband_size * sizeof(*output));
312  }
313  }
314 
315  /* clear the subbands that were not coded */
316  first = subband_tab[i];
317  memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
318  return num_subbands;
319 }
320 
321 /**
322  * Restore the quantized tonal components
323  *
324  * @param components tonal components
325  * @param num_bands number of coded bands
326  */
328  TonalComponent *components, int num_bands)
329 {
330  int i, b, c, m;
331  int nb_components, coding_mode_selector, coding_mode;
332  int band_flags[4], mantissa[8];
333  int component_count = 0;
334 
335  nb_components = get_bits(gb, 5);
336 
337  /* no tonal components */
338  if (nb_components == 0)
339  return 0;
340 
341  coding_mode_selector = get_bits(gb, 2);
342  if (coding_mode_selector == 2)
343  return AVERROR_INVALIDDATA;
344 
345  coding_mode = coding_mode_selector & 1;
346 
347  for (i = 0; i < nb_components; i++) {
348  int coded_values_per_component, quant_step_index;
349 
350  for (b = 0; b <= num_bands; b++)
351  band_flags[b] = get_bits1(gb);
352 
353  coded_values_per_component = get_bits(gb, 3);
354 
355  quant_step_index = get_bits(gb, 3);
356  if (quant_step_index <= 1)
357  return AVERROR_INVALIDDATA;
358 
359  if (coding_mode_selector == 3)
360  coding_mode = get_bits1(gb);
361 
362  for (b = 0; b < (num_bands + 1) * 4; b++) {
363  int coded_components;
364 
365  if (band_flags[b >> 2] == 0)
366  continue;
367 
368  coded_components = get_bits(gb, 3);
369 
370  for (c = 0; c < coded_components; c++) {
371  TonalComponent *cmp = &components[component_count];
372  int sf_index, coded_values, max_coded_values;
373  float scale_factor;
374 
375  sf_index = get_bits(gb, 6);
376  if (component_count >= 64)
377  return AVERROR_INVALIDDATA;
378 
379  cmp->pos = b * 64 + get_bits(gb, 6);
380 
381  max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
382  coded_values = coded_values_per_component + 1;
383  coded_values = FFMIN(max_coded_values, coded_values);
384 
385  scale_factor = ff_atrac_sf_table[sf_index] *
386  inv_max_quant[quant_step_index];
387 
388  read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
389  mantissa, coded_values);
390 
391  cmp->num_coefs = coded_values;
392 
393  /* inverse quant */
394  for (m = 0; m < coded_values; m++)
395  cmp->coef[m] = mantissa[m] * scale_factor;
396 
397  component_count++;
398  }
399  }
400  }
401 
402  return component_count;
403 }
404 
405 /**
406  * Decode gain parameters for the coded bands
407  *
408  * @param block the gainblock for the current band
409  * @param num_bands amount of coded bands
410  */
412  int num_bands)
413 {
414  int b, j;
415  int *level, *loc;
416 
417  AtracGainInfo *gain = block->g_block;
418 
419  for (b = 0; b <= num_bands; b++) {
420  gain[b].num_points = get_bits(gb, 3);
421  level = gain[b].lev_code;
422  loc = gain[b].loc_code;
423 
424  for (j = 0; j < gain[b].num_points; j++) {
425  level[j] = get_bits(gb, 4);
426  loc[j] = get_bits(gb, 5);
427  if (j && loc[j] <= loc[j - 1])
428  return AVERROR_INVALIDDATA;
429  }
430  }
431 
432  /* Clear the unused blocks. */
433  for (; b < 4 ; b++)
434  gain[b].num_points = 0;
435 
436  return 0;
437 }
438 
439 /**
440  * Combine the tonal band spectrum and regular band spectrum
441  *
442  * @param spectrum output spectrum buffer
443  * @param num_components number of tonal components
444  * @param components tonal components for this band
445  * @return position of the last tonal coefficient
446  */
447 static int add_tonal_components(float *spectrum, int num_components,
448  TonalComponent *components)
449 {
450  int i, j, last_pos = -1;
451  float *input, *output;
452 
453  for (i = 0; i < num_components; i++) {
454  last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
455  input = components[i].coef;
456  output = &spectrum[components[i].pos];
457 
458  for (j = 0; j < components[i].num_coefs; j++)
459  output[j] += input[j];
460  }
461 
462  return last_pos;
463 }
464 
465 #define INTERPOLATE(old, new, nsample) \
466  ((old) + (nsample) * 0.125 * ((new) - (old)))
467 
468 static void reverse_matrixing(float *su1, float *su2, int *prev_code,
469  int *curr_code)
470 {
471  int i, nsample, band;
472  float mc1_l, mc1_r, mc2_l, mc2_r;
473 
474  for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
475  int s1 = prev_code[i];
476  int s2 = curr_code[i];
477  nsample = band;
478 
479  if (s1 != s2) {
480  /* Selector value changed, interpolation needed. */
481  mc1_l = matrix_coeffs[s1 * 2 ];
482  mc1_r = matrix_coeffs[s1 * 2 + 1];
483  mc2_l = matrix_coeffs[s2 * 2 ];
484  mc2_r = matrix_coeffs[s2 * 2 + 1];
485 
486  /* Interpolation is done over the first eight samples. */
487  for (; nsample < band + 8; nsample++) {
488  float c1 = su1[nsample];
489  float c2 = su2[nsample];
490  c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
491  c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
492  su1[nsample] = c2;
493  su2[nsample] = c1 * 2.0 - c2;
494  }
495  }
496 
497  /* Apply the matrix without interpolation. */
498  switch (s2) {
499  case 0: /* M/S decoding */
500  for (; nsample < band + 256; nsample++) {
501  float c1 = su1[nsample];
502  float c2 = su2[nsample];
503  su1[nsample] = c2 * 2.0;
504  su2[nsample] = (c1 - c2) * 2.0;
505  }
506  break;
507  case 1:
508  for (; nsample < band + 256; nsample++) {
509  float c1 = su1[nsample];
510  float c2 = su2[nsample];
511  su1[nsample] = (c1 + c2) * 2.0;
512  su2[nsample] = c2 * -2.0;
513  }
514  break;
515  case 2:
516  case 3:
517  for (; nsample < band + 256; nsample++) {
518  float c1 = su1[nsample];
519  float c2 = su2[nsample];
520  su1[nsample] = c1 + c2;
521  su2[nsample] = c1 - c2;
522  }
523  break;
524  default:
525  av_assert1(0);
526  }
527  }
528 }
529 
530 static void get_channel_weights(int index, int flag, float ch[2])
531 {
532  if (index == 7) {
533  ch[0] = 1.0;
534  ch[1] = 1.0;
535  } else {
536  ch[0] = (index & 7) / 7.0;
537  ch[1] = sqrt(2 - ch[0] * ch[0]);
538  if (flag)
539  FFSWAP(float, ch[0], ch[1]);
540  }
541 }
542 
543 static void channel_weighting(float *su1, float *su2, int *p3)
544 {
545  int band, nsample;
546  /* w[x][y] y=0 is left y=1 is right */
547  float w[2][2];
548 
549  if (p3[1] != 7 || p3[3] != 7) {
550  get_channel_weights(p3[1], p3[0], w[0]);
551  get_channel_weights(p3[3], p3[2], w[1]);
552 
553  for (band = 256; band < 4 * 256; band += 256) {
554  for (nsample = band; nsample < band + 8; nsample++) {
555  su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
556  su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
557  }
558  for(; nsample < band + 256; nsample++) {
559  su1[nsample] *= w[1][0];
560  su2[nsample] *= w[1][1];
561  }
562  }
563  }
564 }
565 
566 /**
567  * Decode a Sound Unit
568  *
569  * @param snd the channel unit to be used
570  * @param output the decoded samples before IQMF in float representation
571  * @param channel_num channel number
572  * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)
573  */
575  ChannelUnit *snd, float *output,
576  int channel_num, int coding_mode)
577 {
578  int band, ret, num_subbands, last_tonal, num_bands;
579  GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
580  GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
581 
582  if (coding_mode == JOINT_STEREO && channel_num == 1) {
583  if (get_bits(gb, 2) != 3) {
584  av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
585  return AVERROR_INVALIDDATA;
586  }
587  } else {
588  if (get_bits(gb, 6) != 0x28) {
589  av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
590  return AVERROR_INVALIDDATA;
591  }
592  }
593 
594  /* number of coded QMF bands */
595  snd->bands_coded = get_bits(gb, 2);
596 
597  ret = decode_gain_control(gb, gain2, snd->bands_coded);
598  if (ret)
599  return ret;
600 
602  snd->bands_coded);
603  if (snd->num_components < 0)
604  return snd->num_components;
605 
606  num_subbands = decode_spectrum(gb, snd->spectrum);
607 
608  /* Merge the decoded spectrum and tonal components. */
609  last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
610  snd->components);
611 
612 
613  /* calculate number of used MLT/QMF bands according to the amount of coded
614  spectral lines */
615  num_bands = (subband_tab[num_subbands] - 1) >> 8;
616  if (last_tonal >= 0)
617  num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
618 
619 
620  /* Reconstruct time domain samples. */
621  for (band = 0; band < 4; band++) {
622  /* Perform the IMDCT step without overlapping. */
623  if (band <= num_bands)
624  imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
625  else
626  memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
627 
628  /* gain compensation and overlapping */
630  &snd->prev_frame[band * 256],
631  &gain1->g_block[band], &gain2->g_block[band],
632  256, &output[band * 256]);
633  }
634 
635  /* Swap the gain control buffers for the next frame. */
636  snd->gc_blk_switch ^= 1;
637 
638  return 0;
639 }
640 
641 static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
642  float **out_samples)
643 {
644  ATRAC3Context *q = avctx->priv_data;
645  int ret, i;
646  uint8_t *ptr1;
647 
648  if (q->coding_mode == JOINT_STEREO) {
649  /* channel coupling mode */
650  /* decode Sound Unit 1 */
651  init_get_bits(&q->gb, databuf, avctx->block_align * 8);
652 
653  ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
654  JOINT_STEREO);
655  if (ret != 0)
656  return ret;
657 
658  /* Framedata of the su2 in the joint-stereo mode is encoded in
659  * reverse byte order so we need to swap it first. */
660  if (databuf == q->decoded_bytes_buffer) {
661  uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
662  ptr1 = q->decoded_bytes_buffer;
663  for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
664  FFSWAP(uint8_t, *ptr1, *ptr2);
665  } else {
666  const uint8_t *ptr2 = databuf + avctx->block_align - 1;
667  for (i = 0; i < avctx->block_align; i++)
668  q->decoded_bytes_buffer[i] = *ptr2--;
669  }
670 
671  /* Skip the sync codes (0xF8). */
672  ptr1 = q->decoded_bytes_buffer;
673  for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
674  if (i >= avctx->block_align)
675  return AVERROR_INVALIDDATA;
676  }
677 
678 
679  /* set the bitstream reader at the start of the second Sound Unit*/
680  init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1);
681 
682  /* Fill the Weighting coeffs delay buffer */
683  memmove(q->weighting_delay, &q->weighting_delay[2],
684  4 * sizeof(*q->weighting_delay));
685  q->weighting_delay[4] = get_bits1(&q->gb);
686  q->weighting_delay[5] = get_bits(&q->gb, 3);
687 
688  for (i = 0; i < 4; i++) {
691  q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
692  }
693 
694  /* Decode Sound Unit 2. */
695  ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
696  out_samples[1], 1, JOINT_STEREO);
697  if (ret != 0)
698  return ret;
699 
700  /* Reconstruct the channel coefficients. */
701  reverse_matrixing(out_samples[0], out_samples[1],
704 
705  channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
706  } else {
707  /* normal stereo mode or mono */
708  /* Decode the channel sound units. */
709  for (i = 0; i < avctx->channels; i++) {
710  /* Set the bitstream reader at the start of a channel sound unit. */
711  init_get_bits(&q->gb,
712  databuf + i * avctx->block_align / avctx->channels,
713  avctx->block_align * 8 / avctx->channels);
714 
715  ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
716  out_samples[i], i, q->coding_mode);
717  if (ret != 0)
718  return ret;
719  }
720  }
721 
722  /* Apply the iQMF synthesis filter. */
723  for (i = 0; i < avctx->channels; i++) {
724  float *p1 = out_samples[i];
725  float *p2 = p1 + 256;
726  float *p3 = p2 + 256;
727  float *p4 = p3 + 256;
728  ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
729  ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
730  ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
731  }
732 
733  return 0;
734 }
735 
736 static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
737  int *got_frame_ptr, AVPacket *avpkt)
738 {
739  AVFrame *frame = data;
740  const uint8_t *buf = avpkt->data;
741  int buf_size = avpkt->size;
742  ATRAC3Context *q = avctx->priv_data;
743  int ret;
744  const uint8_t *databuf;
745 
746  if (buf_size < avctx->block_align) {
747  av_log(avctx, AV_LOG_ERROR,
748  "Frame too small (%d bytes). Truncated file?\n", buf_size);
749  return AVERROR_INVALIDDATA;
750  }
751 
752  /* get output buffer */
753  frame->nb_samples = SAMPLES_PER_FRAME;
754  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
755  return ret;
756 
757  /* Check if we need to descramble and what buffer to pass on. */
758  if (q->scrambled_stream) {
760  databuf = q->decoded_bytes_buffer;
761  } else {
762  databuf = buf;
763  }
764 
765  ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
766  if (ret) {
767  av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
768  return ret;
769  }
770 
771  *got_frame_ptr = 1;
772 
773  return avctx->block_align;
774 }
775 
777 {
778  int i;
779 
782 
783  /* Initialize the VLC tables. */
784  for (i = 0; i < 7; i++) {
785  spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
786  spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
787  atrac3_vlc_offs[i ];
788  init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
789  huff_bits[i], 1, 1,
791  }
792 }
793 
795 {
796  static int static_init_done;
797  int i, ret;
798  int version, delay, samples_per_frame, frame_factor;
799  const uint8_t *edata_ptr = avctx->extradata;
800  ATRAC3Context *q = avctx->priv_data;
801 
802  if (avctx->channels <= 0 || avctx->channels > 2) {
803  av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
804  return AVERROR(EINVAL);
805  }
806 
807  if (!static_init_done)
809  static_init_done = 1;
810 
811  /* Take care of the codec-specific extradata. */
812  if (avctx->extradata_size == 14) {
813  /* Parse the extradata, WAV format */
814  av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
815  bytestream_get_le16(&edata_ptr)); // Unknown value always 1
816  edata_ptr += 4; // samples per channel
817  q->coding_mode = bytestream_get_le16(&edata_ptr);
818  av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
819  bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
820  frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
821  av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
822  bytestream_get_le16(&edata_ptr)); // Unknown always 0
823 
824  /* setup */
825  samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
826  version = 4;
827  delay = 0x88E;
829  q->scrambled_stream = 0;
830 
831  if (avctx->block_align != 96 * avctx->channels * frame_factor &&
832  avctx->block_align != 152 * avctx->channels * frame_factor &&
833  avctx->block_align != 192 * avctx->channels * frame_factor) {
834  av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
835  "configuration %d/%d/%d\n", avctx->block_align,
836  avctx->channels, frame_factor);
837  return AVERROR_INVALIDDATA;
838  }
839  } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
840  /* Parse the extradata, RM format. */
841  version = bytestream_get_be32(&edata_ptr);
842  samples_per_frame = bytestream_get_be16(&edata_ptr);
843  delay = bytestream_get_be16(&edata_ptr);
844  q->coding_mode = bytestream_get_be16(&edata_ptr);
845  q->scrambled_stream = 1;
846 
847  } else {
848  av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
849  avctx->extradata_size);
850  return AVERROR(EINVAL);
851  }
852 
853  /* Check the extradata */
854 
855  if (version != 4) {
856  av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
857  return AVERROR_INVALIDDATA;
858  }
859 
860  if (samples_per_frame != SAMPLES_PER_FRAME &&
861  samples_per_frame != SAMPLES_PER_FRAME * 2) {
862  av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
863  samples_per_frame);
864  return AVERROR_INVALIDDATA;
865  }
866 
867  if (delay != 0x88E) {
868  av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
869  delay);
870  return AVERROR_INVALIDDATA;
871  }
872 
873  if (q->coding_mode == STEREO)
874  av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
875  else if (q->coding_mode == JOINT_STEREO) {
876  if (avctx->channels != 2) {
877  av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");
878  return AVERROR_INVALIDDATA;
879  }
880  av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
881  } else {
882  av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
883  q->coding_mode);
884  return AVERROR_INVALIDDATA;
885  }
886 
887  if (avctx->block_align >= UINT_MAX / 2)
888  return AVERROR(EINVAL);
889 
892  if (!q->decoded_bytes_buffer)
893  return AVERROR(ENOMEM);
894 
896 
897  /* initialize the MDCT transform */
898  if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
899  av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
901  return ret;
902  }
903 
904  /* init the joint-stereo decoding data */
905  q->weighting_delay[0] = 0;
906  q->weighting_delay[1] = 7;
907  q->weighting_delay[2] = 0;
908  q->weighting_delay[3] = 7;
909  q->weighting_delay[4] = 0;
910  q->weighting_delay[5] = 7;
911 
912  for (i = 0; i < 4; i++) {
913  q->matrix_coeff_index_prev[i] = 3;
914  q->matrix_coeff_index_now[i] = 3;
915  q->matrix_coeff_index_next[i] = 3;
916  }
917 
920  ff_fmt_convert_init(&q->fmt_conv, avctx);
921 
922  q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
923  if (!q->units || !q->fdsp) {
924  atrac3_decode_close(avctx);
925  return AVERROR(ENOMEM);
926  }
927 
928  return 0;
929 }
930 
932  .name = "atrac3",
933  .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
934  .type = AVMEDIA_TYPE_AUDIO,
935  .id = AV_CODEC_ID_ATRAC3,
936  .priv_data_size = sizeof(ATRAC3Context),
940  .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
941  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
943 };