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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 
196  ff_mdct_end(&q->mdct_ctx);
197 
198  return 0;
199 }
200 
201 /**
202  * Mantissa decoding
203  *
204  * @param selector which table the output values are coded with
205  * @param coding_flag constant length coding or variable length coding
206  * @param mantissas mantissa output table
207  * @param num_codes number of values to get
208  */
209 static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
210  int coding_flag, int *mantissas,
211  int num_codes)
212 {
213  int i, code, huff_symb;
214 
215  if (selector == 1)
216  num_codes /= 2;
217 
218  if (coding_flag != 0) {
219  /* constant length coding (CLC) */
220  int num_bits = clc_length_tab[selector];
221 
222  if (selector > 1) {
223  for (i = 0; i < num_codes; i++) {
224  if (num_bits)
225  code = get_sbits(gb, num_bits);
226  else
227  code = 0;
228  mantissas[i] = code;
229  }
230  } else {
231  for (i = 0; i < num_codes; i++) {
232  if (num_bits)
233  code = get_bits(gb, num_bits); // num_bits is always 4 in this case
234  else
235  code = 0;
236  mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
237  mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
238  }
239  }
240  } else {
241  /* variable length coding (VLC) */
242  if (selector != 1) {
243  for (i = 0; i < num_codes; i++) {
244  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
245  spectral_coeff_tab[selector-1].bits, 3);
246  huff_symb += 1;
247  code = huff_symb >> 1;
248  if (huff_symb & 1)
249  code = -code;
250  mantissas[i] = code;
251  }
252  } else {
253  for (i = 0; i < num_codes; i++) {
254  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
255  spectral_coeff_tab[selector - 1].bits, 3);
256  mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
257  mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
258  }
259  }
260  }
261 }
262 
263 /**
264  * Restore the quantized band spectrum coefficients
265  *
266  * @return subband count, fix for broken specification/files
267  */
268 static int decode_spectrum(GetBitContext *gb, float *output)
269 {
270  int num_subbands, coding_mode, i, j, first, last, subband_size;
271  int subband_vlc_index[32], sf_index[32];
272  int mantissas[128];
273  float scale_factor;
274 
275  num_subbands = get_bits(gb, 5); // number of coded subbands
276  coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
277 
278  /* get the VLC selector table for the subbands, 0 means not coded */
279  for (i = 0; i <= num_subbands; i++)
280  subband_vlc_index[i] = get_bits(gb, 3);
281 
282  /* read the scale factor indexes from the stream */
283  for (i = 0; i <= num_subbands; i++) {
284  if (subband_vlc_index[i] != 0)
285  sf_index[i] = get_bits(gb, 6);
286  }
287 
288  for (i = 0; i <= num_subbands; i++) {
289  first = subband_tab[i ];
290  last = subband_tab[i + 1];
291 
292  subband_size = last - first;
293 
294  if (subband_vlc_index[i] != 0) {
295  /* decode spectral coefficients for this subband */
296  /* TODO: This can be done faster is several blocks share the
297  * same VLC selector (subband_vlc_index) */
298  read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
299  mantissas, subband_size);
300 
301  /* decode the scale factor for this subband */
302  scale_factor = ff_atrac_sf_table[sf_index[i]] *
303  inv_max_quant[subband_vlc_index[i]];
304 
305  /* inverse quantize the coefficients */
306  for (j = 0; first < last; first++, j++)
307  output[first] = mantissas[j] * scale_factor;
308  } else {
309  /* this subband was not coded, so zero the entire subband */
310  memset(output + first, 0, subband_size * sizeof(*output));
311  }
312  }
313 
314  /* clear the subbands that were not coded */
315  first = subband_tab[i];
316  memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
317  return num_subbands;
318 }
319 
320 /**
321  * Restore the quantized tonal components
322  *
323  * @param components tonal components
324  * @param num_bands number of coded bands
325  */
327  TonalComponent *components, int num_bands)
328 {
329  int i, b, c, m;
330  int nb_components, coding_mode_selector, coding_mode;
331  int band_flags[4], mantissa[8];
332  int component_count = 0;
333 
334  nb_components = get_bits(gb, 5);
335 
336  /* no tonal components */
337  if (nb_components == 0)
338  return 0;
339 
340  coding_mode_selector = get_bits(gb, 2);
341  if (coding_mode_selector == 2)
342  return AVERROR_INVALIDDATA;
343 
344  coding_mode = coding_mode_selector & 1;
345 
346  for (i = 0; i < nb_components; i++) {
347  int coded_values_per_component, quant_step_index;
348 
349  for (b = 0; b <= num_bands; b++)
350  band_flags[b] = get_bits1(gb);
351 
352  coded_values_per_component = get_bits(gb, 3);
353 
354  quant_step_index = get_bits(gb, 3);
355  if (quant_step_index <= 1)
356  return AVERROR_INVALIDDATA;
357 
358  if (coding_mode_selector == 3)
359  coding_mode = get_bits1(gb);
360 
361  for (b = 0; b < (num_bands + 1) * 4; b++) {
362  int coded_components;
363 
364  if (band_flags[b >> 2] == 0)
365  continue;
366 
367  coded_components = get_bits(gb, 3);
368 
369  for (c = 0; c < coded_components; c++) {
370  TonalComponent *cmp = &components[component_count];
371  int sf_index, coded_values, max_coded_values;
372  float scale_factor;
373 
374  sf_index = get_bits(gb, 6);
375  if (component_count >= 64)
376  return AVERROR_INVALIDDATA;
377 
378  cmp->pos = b * 64 + get_bits(gb, 6);
379 
380  max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
381  coded_values = coded_values_per_component + 1;
382  coded_values = FFMIN(max_coded_values, coded_values);
383 
384  scale_factor = ff_atrac_sf_table[sf_index] *
385  inv_max_quant[quant_step_index];
386 
387  read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
388  mantissa, coded_values);
389 
390  cmp->num_coefs = coded_values;
391 
392  /* inverse quant */
393  for (m = 0; m < coded_values; m++)
394  cmp->coef[m] = mantissa[m] * scale_factor;
395 
396  component_count++;
397  }
398  }
399  }
400 
401  return component_count;
402 }
403 
404 /**
405  * Decode gain parameters for the coded bands
406  *
407  * @param block the gainblock for the current band
408  * @param num_bands amount of coded bands
409  */
411  int num_bands)
412 {
413  int b, j;
414  int *level, *loc;
415 
416  AtracGainInfo *gain = block->g_block;
417 
418  for (b = 0; b <= num_bands; b++) {
419  gain[b].num_points = get_bits(gb, 3);
420  level = gain[b].lev_code;
421  loc = gain[b].loc_code;
422 
423  for (j = 0; j < gain[b].num_points; j++) {
424  level[j] = get_bits(gb, 4);
425  loc[j] = get_bits(gb, 5);
426  if (j && loc[j] <= loc[j - 1])
427  return AVERROR_INVALIDDATA;
428  }
429  }
430 
431  /* Clear the unused blocks. */
432  for (; b < 4 ; b++)
433  gain[b].num_points = 0;
434 
435  return 0;
436 }
437 
438 /**
439  * Combine the tonal band spectrum and regular band spectrum
440  *
441  * @param spectrum output spectrum buffer
442  * @param num_components number of tonal components
443  * @param components tonal components for this band
444  * @return position of the last tonal coefficient
445  */
446 static int add_tonal_components(float *spectrum, int num_components,
447  TonalComponent *components)
448 {
449  int i, j, last_pos = -1;
450  float *input, *output;
451 
452  for (i = 0; i < num_components; i++) {
453  last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
454  input = components[i].coef;
455  output = &spectrum[components[i].pos];
456 
457  for (j = 0; j < components[i].num_coefs; j++)
458  output[j] += input[j];
459  }
460 
461  return last_pos;
462 }
463 
464 #define INTERPOLATE(old, new, nsample) \
465  ((old) + (nsample) * 0.125 * ((new) - (old)))
466 
467 static void reverse_matrixing(float *su1, float *su2, int *prev_code,
468  int *curr_code)
469 {
470  int i, nsample, band;
471  float mc1_l, mc1_r, mc2_l, mc2_r;
472 
473  for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
474  int s1 = prev_code[i];
475  int s2 = curr_code[i];
476  nsample = band;
477 
478  if (s1 != s2) {
479  /* Selector value changed, interpolation needed. */
480  mc1_l = matrix_coeffs[s1 * 2 ];
481  mc1_r = matrix_coeffs[s1 * 2 + 1];
482  mc2_l = matrix_coeffs[s2 * 2 ];
483  mc2_r = matrix_coeffs[s2 * 2 + 1];
484 
485  /* Interpolation is done over the first eight samples. */
486  for (; nsample < band + 8; nsample++) {
487  float c1 = su1[nsample];
488  float c2 = su2[nsample];
489  c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
490  c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
491  su1[nsample] = c2;
492  su2[nsample] = c1 * 2.0 - c2;
493  }
494  }
495 
496  /* Apply the matrix without interpolation. */
497  switch (s2) {
498  case 0: /* M/S decoding */
499  for (; nsample < band + 256; nsample++) {
500  float c1 = su1[nsample];
501  float c2 = su2[nsample];
502  su1[nsample] = c2 * 2.0;
503  su2[nsample] = (c1 - c2) * 2.0;
504  }
505  break;
506  case 1:
507  for (; nsample < band + 256; nsample++) {
508  float c1 = su1[nsample];
509  float c2 = su2[nsample];
510  su1[nsample] = (c1 + c2) * 2.0;
511  su2[nsample] = c2 * -2.0;
512  }
513  break;
514  case 2:
515  case 3:
516  for (; nsample < band + 256; nsample++) {
517  float c1 = su1[nsample];
518  float c2 = su2[nsample];
519  su1[nsample] = c1 + c2;
520  su2[nsample] = c1 - c2;
521  }
522  break;
523  default:
524  av_assert1(0);
525  }
526  }
527 }
528 
529 static void get_channel_weights(int index, int flag, float ch[2])
530 {
531  if (index == 7) {
532  ch[0] = 1.0;
533  ch[1] = 1.0;
534  } else {
535  ch[0] = (index & 7) / 7.0;
536  ch[1] = sqrt(2 - ch[0] * ch[0]);
537  if (flag)
538  FFSWAP(float, ch[0], ch[1]);
539  }
540 }
541 
542 static void channel_weighting(float *su1, float *su2, int *p3)
543 {
544  int band, nsample;
545  /* w[x][y] y=0 is left y=1 is right */
546  float w[2][2];
547 
548  if (p3[1] != 7 || p3[3] != 7) {
549  get_channel_weights(p3[1], p3[0], w[0]);
550  get_channel_weights(p3[3], p3[2], w[1]);
551 
552  for (band = 256; band < 4 * 256; band += 256) {
553  for (nsample = band; nsample < band + 8; nsample++) {
554  su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
555  su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
556  }
557  for(; nsample < band + 256; nsample++) {
558  su1[nsample] *= w[1][0];
559  su2[nsample] *= w[1][1];
560  }
561  }
562  }
563 }
564 
565 /**
566  * Decode a Sound Unit
567  *
568  * @param snd the channel unit to be used
569  * @param output the decoded samples before IQMF in float representation
570  * @param channel_num channel number
571  * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)
572  */
574  ChannelUnit *snd, float *output,
575  int channel_num, int coding_mode)
576 {
577  int band, ret, num_subbands, last_tonal, num_bands;
578  GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
579  GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
580 
581  if (coding_mode == JOINT_STEREO && channel_num == 1) {
582  if (get_bits(gb, 2) != 3) {
583  av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
584  return AVERROR_INVALIDDATA;
585  }
586  } else {
587  if (get_bits(gb, 6) != 0x28) {
588  av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
589  return AVERROR_INVALIDDATA;
590  }
591  }
592 
593  /* number of coded QMF bands */
594  snd->bands_coded = get_bits(gb, 2);
595 
596  ret = decode_gain_control(gb, gain2, snd->bands_coded);
597  if (ret)
598  return ret;
599 
601  snd->bands_coded);
602  if (snd->num_components < 0)
603  return snd->num_components;
604 
605  num_subbands = decode_spectrum(gb, snd->spectrum);
606 
607  /* Merge the decoded spectrum and tonal components. */
608  last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
609  snd->components);
610 
611 
612  /* calculate number of used MLT/QMF bands according to the amount of coded
613  spectral lines */
614  num_bands = (subband_tab[num_subbands] - 1) >> 8;
615  if (last_tonal >= 0)
616  num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
617 
618 
619  /* Reconstruct time domain samples. */
620  for (band = 0; band < 4; band++) {
621  /* Perform the IMDCT step without overlapping. */
622  if (band <= num_bands)
623  imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
624  else
625  memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
626 
627  /* gain compensation and overlapping */
629  &snd->prev_frame[band * 256],
630  &gain1->g_block[band], &gain2->g_block[band],
631  256, &output[band * 256]);
632  }
633 
634  /* Swap the gain control buffers for the next frame. */
635  snd->gc_blk_switch ^= 1;
636 
637  return 0;
638 }
639 
640 static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
641  float **out_samples)
642 {
643  ATRAC3Context *q = avctx->priv_data;
644  int ret, i;
645  uint8_t *ptr1;
646 
647  if (q->coding_mode == JOINT_STEREO) {
648  /* channel coupling mode */
649  /* decode Sound Unit 1 */
650  init_get_bits(&q->gb, databuf, avctx->block_align * 8);
651 
652  ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
653  JOINT_STEREO);
654  if (ret != 0)
655  return ret;
656 
657  /* Framedata of the su2 in the joint-stereo mode is encoded in
658  * reverse byte order so we need to swap it first. */
659  if (databuf == q->decoded_bytes_buffer) {
660  uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
661  ptr1 = q->decoded_bytes_buffer;
662  for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
663  FFSWAP(uint8_t, *ptr1, *ptr2);
664  } else {
665  const uint8_t *ptr2 = databuf + avctx->block_align - 1;
666  for (i = 0; i < avctx->block_align; i++)
667  q->decoded_bytes_buffer[i] = *ptr2--;
668  }
669 
670  /* Skip the sync codes (0xF8). */
671  ptr1 = q->decoded_bytes_buffer;
672  for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
673  if (i >= avctx->block_align)
674  return AVERROR_INVALIDDATA;
675  }
676 
677 
678  /* set the bitstream reader at the start of the second Sound Unit*/
679  init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1);
680 
681  /* Fill the Weighting coeffs delay buffer */
682  memmove(q->weighting_delay, &q->weighting_delay[2],
683  4 * sizeof(*q->weighting_delay));
684  q->weighting_delay[4] = get_bits1(&q->gb);
685  q->weighting_delay[5] = get_bits(&q->gb, 3);
686 
687  for (i = 0; i < 4; i++) {
690  q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
691  }
692 
693  /* Decode Sound Unit 2. */
694  ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
695  out_samples[1], 1, JOINT_STEREO);
696  if (ret != 0)
697  return ret;
698 
699  /* Reconstruct the channel coefficients. */
700  reverse_matrixing(out_samples[0], out_samples[1],
703 
704  channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
705  } else {
706  /* normal stereo mode or mono */
707  /* Decode the channel sound units. */
708  for (i = 0; i < avctx->channels; i++) {
709  /* Set the bitstream reader at the start of a channel sound unit. */
710  init_get_bits(&q->gb,
711  databuf + i * avctx->block_align / avctx->channels,
712  avctx->block_align * 8 / avctx->channels);
713 
714  ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
715  out_samples[i], i, q->coding_mode);
716  if (ret != 0)
717  return ret;
718  }
719  }
720 
721  /* Apply the iQMF synthesis filter. */
722  for (i = 0; i < avctx->channels; i++) {
723  float *p1 = out_samples[i];
724  float *p2 = p1 + 256;
725  float *p3 = p2 + 256;
726  float *p4 = p3 + 256;
727  ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
728  ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
729  ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
730  }
731 
732  return 0;
733 }
734 
735 static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
736  int *got_frame_ptr, AVPacket *avpkt)
737 {
738  AVFrame *frame = data;
739  const uint8_t *buf = avpkt->data;
740  int buf_size = avpkt->size;
741  ATRAC3Context *q = avctx->priv_data;
742  int ret;
743  const uint8_t *databuf;
744 
745  if (buf_size < avctx->block_align) {
746  av_log(avctx, AV_LOG_ERROR,
747  "Frame too small (%d bytes). Truncated file?\n", buf_size);
748  return AVERROR_INVALIDDATA;
749  }
750 
751  /* get output buffer */
752  frame->nb_samples = SAMPLES_PER_FRAME;
753  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
754  return ret;
755 
756  /* Check if we need to descramble and what buffer to pass on. */
757  if (q->scrambled_stream) {
759  databuf = q->decoded_bytes_buffer;
760  } else {
761  databuf = buf;
762  }
763 
764  ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
765  if (ret) {
766  av_log(NULL, AV_LOG_ERROR, "Frame decoding error!\n");
767  return ret;
768  }
769 
770  *got_frame_ptr = 1;
771 
772  return avctx->block_align;
773 }
774 
776 {
777  int i;
778 
781 
782  /* Initialize the VLC tables. */
783  for (i = 0; i < 7; i++) {
784  spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
785  spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
786  atrac3_vlc_offs[i ];
787  init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
788  huff_bits[i], 1, 1,
790  }
791 }
792 
794 {
795  static int static_init_done;
796  int i, ret;
797  int version, delay, samples_per_frame, frame_factor;
798  const uint8_t *edata_ptr = avctx->extradata;
799  ATRAC3Context *q = avctx->priv_data;
800 
801  if (avctx->channels <= 0 || avctx->channels > 2) {
802  av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
803  return AVERROR(EINVAL);
804  }
805 
806  if (!static_init_done)
808  static_init_done = 1;
809 
810  /* Take care of the codec-specific extradata. */
811  if (avctx->extradata_size == 14) {
812  /* Parse the extradata, WAV format */
813  av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
814  bytestream_get_le16(&edata_ptr)); // Unknown value always 1
815  edata_ptr += 4; // samples per channel
816  q->coding_mode = bytestream_get_le16(&edata_ptr);
817  av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
818  bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
819  frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
820  av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
821  bytestream_get_le16(&edata_ptr)); // Unknown always 0
822 
823  /* setup */
824  samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
825  version = 4;
826  delay = 0x88E;
828  q->scrambled_stream = 0;
829 
830  if (avctx->block_align != 96 * avctx->channels * frame_factor &&
831  avctx->block_align != 152 * avctx->channels * frame_factor &&
832  avctx->block_align != 192 * avctx->channels * frame_factor) {
833  av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
834  "configuration %d/%d/%d\n", avctx->block_align,
835  avctx->channels, frame_factor);
836  return AVERROR_INVALIDDATA;
837  }
838  } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
839  /* Parse the extradata, RM format. */
840  version = bytestream_get_be32(&edata_ptr);
841  samples_per_frame = bytestream_get_be16(&edata_ptr);
842  delay = bytestream_get_be16(&edata_ptr);
843  q->coding_mode = bytestream_get_be16(&edata_ptr);
844  q->scrambled_stream = 1;
845 
846  } else {
847  av_log(NULL, AV_LOG_ERROR, "Unknown extradata size %d.\n",
848  avctx->extradata_size);
849  return AVERROR(EINVAL);
850  }
851 
852  /* Check the extradata */
853 
854  if (version != 4) {
855  av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
856  return AVERROR_INVALIDDATA;
857  }
858 
859  if (samples_per_frame != SAMPLES_PER_FRAME &&
860  samples_per_frame != SAMPLES_PER_FRAME * 2) {
861  av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
862  samples_per_frame);
863  return AVERROR_INVALIDDATA;
864  }
865 
866  if (delay != 0x88E) {
867  av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
868  delay);
869  return AVERROR_INVALIDDATA;
870  }
871 
872  if (q->coding_mode == STEREO)
873  av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
874  else if (q->coding_mode == JOINT_STEREO) {
875  if (avctx->channels != 2) {
876  av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");
877  return AVERROR_INVALIDDATA;
878  }
879  av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
880  } else {
881  av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
882  q->coding_mode);
883  return AVERROR_INVALIDDATA;
884  }
885 
886  if (avctx->block_align >= UINT_MAX / 2)
887  return AVERROR(EINVAL);
888 
891  if (!q->decoded_bytes_buffer)
892  return AVERROR(ENOMEM);
893 
895 
896  /* initialize the MDCT transform */
897  if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
898  av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
900  return ret;
901  }
902 
903  /* init the joint-stereo decoding data */
904  q->weighting_delay[0] = 0;
905  q->weighting_delay[1] = 7;
906  q->weighting_delay[2] = 0;
907  q->weighting_delay[3] = 7;
908  q->weighting_delay[4] = 0;
909  q->weighting_delay[5] = 7;
910 
911  for (i = 0; i < 4; i++) {
912  q->matrix_coeff_index_prev[i] = 3;
913  q->matrix_coeff_index_now[i] = 3;
914  q->matrix_coeff_index_next[i] = 3;
915  }
916 
919  ff_fmt_convert_init(&q->fmt_conv, avctx);
920 
921  q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
922  if (!q->units) {
923  atrac3_decode_close(avctx);
924  return AVERROR(ENOMEM);
925  }
926 
927  return 0;
928 }
929 
931  .name = "atrac3",
932  .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
933  .type = AVMEDIA_TYPE_AUDIO,
934  .id = AV_CODEC_ID_ATRAC3,
935  .priv_data_size = sizeof(ATRAC3Context),
939  .capabilities = CODEC_CAP_SUBFRAMES | CODEC_CAP_DR1,
940  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
942 };