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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 "get_bits.h"
46 #include "internal.h"
47 
48 #include "atrac.h"
49 #include "atrac3data.h"
50 
51 #define JOINT_STEREO 0x12
52 #define STEREO 0x2
53 
54 #define SAMPLES_PER_FRAME 1024
55 #define MDCT_SIZE 512
56 
57 typedef struct GainBlock {
59 } GainBlock;
60 
61 typedef struct TonalComponent {
62  int pos;
63  int num_coefs;
64  float coef[8];
66 
67 typedef struct ChannelUnit {
74 
77 
78  float delay_buf1[46]; ///<qmf delay buffers
79  float delay_buf2[46];
80  float delay_buf3[46];
81 } ChannelUnit;
82 
83 typedef struct ATRAC3Context {
85  //@{
86  /** stream data */
88 
90  //@}
91  //@{
92  /** joint-stereo related variables */
97  //@}
98  //@{
99  /** data buffers */
101  float temp_buf[1070];
102  //@}
103  //@{
104  /** extradata */
106  //@}
107 
111 } ATRAC3Context;
112 
114 static VLC_TYPE atrac3_vlc_table[4096][2];
116 
117 /**
118  * Regular 512 points IMDCT without overlapping, with the exception of the
119  * swapping of odd bands caused by the reverse spectra of the QMF.
120  *
121  * @param odd_band 1 if the band is an odd band
122  */
123 static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
124 {
125  int i;
126 
127  if (odd_band) {
128  /**
129  * Reverse the odd bands before IMDCT, this is an effect of the QMF
130  * transform or it gives better compression to do it this way.
131  * FIXME: It should be possible to handle this in imdct_calc
132  * for that to happen a modification of the prerotation step of
133  * all SIMD code and C code is needed.
134  * Or fix the functions before so they generate a pre reversed spectrum.
135  */
136  for (i = 0; i < 128; i++)
137  FFSWAP(float, input[i], input[255 - i]);
138  }
139 
140  q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
141 
142  /* Perform windowing on the output. */
143  q->fdsp->vector_fmul(output, output, mdct_window, MDCT_SIZE);
144 }
145 
146 /*
147  * indata descrambling, only used for data coming from the rm container
148  */
149 static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
150 {
151  int i, off;
152  uint32_t c;
153  const uint32_t *buf;
154  uint32_t *output = (uint32_t *)out;
155 
156  off = (intptr_t)input & 3;
157  buf = (const uint32_t *)(input - off);
158  if (off)
159  c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
160  else
161  c = av_be2ne32(0x537F6103U);
162  bytes += 3 + off;
163  for (i = 0; i < bytes / 4; i++)
164  output[i] = c ^ buf[i];
165 
166  if (off)
167  avpriv_request_sample(NULL, "Offset of %d", off);
168 
169  return off;
170 }
171 
172 static av_cold void init_imdct_window(void)
173 {
174  int i, j;
175 
176  /* generate the mdct window, for details see
177  * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
178  for (i = 0, j = 255; i < 128; i++, j--) {
179  float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
180  float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
181  float w = 0.5 * (wi * wi + wj * wj);
182  mdct_window[i] = mdct_window[511 - i] = wi / w;
183  mdct_window[j] = mdct_window[511 - j] = wj / w;
184  }
185 }
186 
188 {
189  ATRAC3Context *q = avctx->priv_data;
190 
191  av_freep(&q->units);
193  av_freep(&q->fdsp);
194 
195  ff_mdct_end(&q->mdct_ctx);
196 
197  return 0;
198 }
199 
200 /**
201  * Mantissa decoding
202  *
203  * @param selector which table the output values are coded with
204  * @param coding_flag constant length coding or variable length coding
205  * @param mantissas mantissa output table
206  * @param num_codes number of values to get
207  */
208 static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
209  int coding_flag, int *mantissas,
210  int num_codes)
211 {
212  int i, code, huff_symb;
213 
214  if (selector == 1)
215  num_codes /= 2;
216 
217  if (coding_flag != 0) {
218  /* constant length coding (CLC) */
219  int num_bits = clc_length_tab[selector];
220 
221  if (selector > 1) {
222  for (i = 0; i < num_codes; i++) {
223  if (num_bits)
224  code = get_sbits(gb, num_bits);
225  else
226  code = 0;
227  mantissas[i] = code;
228  }
229  } else {
230  for (i = 0; i < num_codes; i++) {
231  if (num_bits)
232  code = get_bits(gb, num_bits); // num_bits is always 4 in this case
233  else
234  code = 0;
235  mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
236  mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
237  }
238  }
239  } else {
240  /* variable length coding (VLC) */
241  if (selector != 1) {
242  for (i = 0; i < num_codes; i++) {
243  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
244  spectral_coeff_tab[selector-1].bits, 3);
245  huff_symb += 1;
246  code = huff_symb >> 1;
247  if (huff_symb & 1)
248  code = -code;
249  mantissas[i] = code;
250  }
251  } else {
252  for (i = 0; i < num_codes; i++) {
253  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
254  spectral_coeff_tab[selector - 1].bits, 3);
255  mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
256  mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
257  }
258  }
259  }
260 }
261 
262 /**
263  * Restore the quantized band spectrum coefficients
264  *
265  * @return subband count, fix for broken specification/files
266  */
267 static int decode_spectrum(GetBitContext *gb, float *output)
268 {
269  int num_subbands, coding_mode, i, j, first, last, subband_size;
270  int subband_vlc_index[32], sf_index[32];
271  int mantissas[128];
272  float scale_factor;
273 
274  num_subbands = get_bits(gb, 5); // number of coded subbands
275  coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
276 
277  /* get the VLC selector table for the subbands, 0 means not coded */
278  for (i = 0; i <= num_subbands; i++)
279  subband_vlc_index[i] = get_bits(gb, 3);
280 
281  /* read the scale factor indexes from the stream */
282  for (i = 0; i <= num_subbands; i++) {
283  if (subband_vlc_index[i] != 0)
284  sf_index[i] = get_bits(gb, 6);
285  }
286 
287  for (i = 0; i <= num_subbands; i++) {
288  first = subband_tab[i ];
289  last = subband_tab[i + 1];
290 
291  subband_size = last - first;
292 
293  if (subband_vlc_index[i] != 0) {
294  /* decode spectral coefficients for this subband */
295  /* TODO: This can be done faster is several blocks share the
296  * same VLC selector (subband_vlc_index) */
297  read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
298  mantissas, subband_size);
299 
300  /* decode the scale factor for this subband */
301  scale_factor = ff_atrac_sf_table[sf_index[i]] *
302  inv_max_quant[subband_vlc_index[i]];
303 
304  /* inverse quantize the coefficients */
305  for (j = 0; first < last; first++, j++)
306  output[first] = mantissas[j] * scale_factor;
307  } else {
308  /* this subband was not coded, so zero the entire subband */
309  memset(output + first, 0, subband_size * sizeof(*output));
310  }
311  }
312 
313  /* clear the subbands that were not coded */
314  first = subband_tab[i];
315  memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
316  return num_subbands;
317 }
318 
319 /**
320  * Restore the quantized tonal components
321  *
322  * @param components tonal components
323  * @param num_bands number of coded bands
324  */
326  TonalComponent *components, int num_bands)
327 {
328  int i, b, c, m;
329  int nb_components, coding_mode_selector, coding_mode;
330  int band_flags[4], mantissa[8];
331  int component_count = 0;
332 
333  nb_components = get_bits(gb, 5);
334 
335  /* no tonal components */
336  if (nb_components == 0)
337  return 0;
338 
339  coding_mode_selector = get_bits(gb, 2);
340  if (coding_mode_selector == 2)
341  return AVERROR_INVALIDDATA;
342 
343  coding_mode = coding_mode_selector & 1;
344 
345  for (i = 0; i < nb_components; i++) {
346  int coded_values_per_component, quant_step_index;
347 
348  for (b = 0; b <= num_bands; b++)
349  band_flags[b] = get_bits1(gb);
350 
351  coded_values_per_component = get_bits(gb, 3);
352 
353  quant_step_index = get_bits(gb, 3);
354  if (quant_step_index <= 1)
355  return AVERROR_INVALIDDATA;
356 
357  if (coding_mode_selector == 3)
358  coding_mode = get_bits1(gb);
359 
360  for (b = 0; b < (num_bands + 1) * 4; b++) {
361  int coded_components;
362 
363  if (band_flags[b >> 2] == 0)
364  continue;
365 
366  coded_components = get_bits(gb, 3);
367 
368  for (c = 0; c < coded_components; c++) {
369  TonalComponent *cmp = &components[component_count];
370  int sf_index, coded_values, max_coded_values;
371  float scale_factor;
372 
373  sf_index = get_bits(gb, 6);
374  if (component_count >= 64)
375  return AVERROR_INVALIDDATA;
376 
377  cmp->pos = b * 64 + get_bits(gb, 6);
378 
379  max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
380  coded_values = coded_values_per_component + 1;
381  coded_values = FFMIN(max_coded_values, coded_values);
382 
383  scale_factor = ff_atrac_sf_table[sf_index] *
384  inv_max_quant[quant_step_index];
385 
386  read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
387  mantissa, coded_values);
388 
389  cmp->num_coefs = coded_values;
390 
391  /* inverse quant */
392  for (m = 0; m < coded_values; m++)
393  cmp->coef[m] = mantissa[m] * scale_factor;
394 
395  component_count++;
396  }
397  }
398  }
399 
400  return component_count;
401 }
402 
403 /**
404  * Decode gain parameters for the coded bands
405  *
406  * @param block the gainblock for the current band
407  * @param num_bands amount of coded bands
408  */
410  int num_bands)
411 {
412  int b, j;
413  int *level, *loc;
414 
415  AtracGainInfo *gain = block->g_block;
416 
417  for (b = 0; b <= num_bands; b++) {
418  gain[b].num_points = get_bits(gb, 3);
419  level = gain[b].lev_code;
420  loc = gain[b].loc_code;
421 
422  for (j = 0; j < gain[b].num_points; j++) {
423  level[j] = get_bits(gb, 4);
424  loc[j] = get_bits(gb, 5);
425  if (j && loc[j] <= loc[j - 1])
426  return AVERROR_INVALIDDATA;
427  }
428  }
429 
430  /* Clear the unused blocks. */
431  for (; b < 4 ; b++)
432  gain[b].num_points = 0;
433 
434  return 0;
435 }
436 
437 /**
438  * Combine the tonal band spectrum and regular band spectrum
439  *
440  * @param spectrum output spectrum buffer
441  * @param num_components number of tonal components
442  * @param components tonal components for this band
443  * @return position of the last tonal coefficient
444  */
445 static int add_tonal_components(float *spectrum, int num_components,
446  TonalComponent *components)
447 {
448  int i, j, last_pos = -1;
449  float *input, *output;
450 
451  for (i = 0; i < num_components; i++) {
452  last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
453  input = components[i].coef;
454  output = &spectrum[components[i].pos];
455 
456  for (j = 0; j < components[i].num_coefs; j++)
457  output[j] += input[j];
458  }
459 
460  return last_pos;
461 }
462 
463 #define INTERPOLATE(old, new, nsample) \
464  ((old) + (nsample) * 0.125 * ((new) - (old)))
465 
466 static void reverse_matrixing(float *su1, float *su2, int *prev_code,
467  int *curr_code)
468 {
469  int i, nsample, band;
470  float mc1_l, mc1_r, mc2_l, mc2_r;
471 
472  for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
473  int s1 = prev_code[i];
474  int s2 = curr_code[i];
475  nsample = band;
476 
477  if (s1 != s2) {
478  /* Selector value changed, interpolation needed. */
479  mc1_l = matrix_coeffs[s1 * 2 ];
480  mc1_r = matrix_coeffs[s1 * 2 + 1];
481  mc2_l = matrix_coeffs[s2 * 2 ];
482  mc2_r = matrix_coeffs[s2 * 2 + 1];
483 
484  /* Interpolation is done over the first eight samples. */
485  for (; nsample < band + 8; nsample++) {
486  float c1 = su1[nsample];
487  float c2 = su2[nsample];
488  c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
489  c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
490  su1[nsample] = c2;
491  su2[nsample] = c1 * 2.0 - c2;
492  }
493  }
494 
495  /* Apply the matrix without interpolation. */
496  switch (s2) {
497  case 0: /* M/S decoding */
498  for (; nsample < band + 256; nsample++) {
499  float c1 = su1[nsample];
500  float c2 = su2[nsample];
501  su1[nsample] = c2 * 2.0;
502  su2[nsample] = (c1 - c2) * 2.0;
503  }
504  break;
505  case 1:
506  for (; nsample < band + 256; nsample++) {
507  float c1 = su1[nsample];
508  float c2 = su2[nsample];
509  su1[nsample] = (c1 + c2) * 2.0;
510  su2[nsample] = c2 * -2.0;
511  }
512  break;
513  case 2:
514  case 3:
515  for (; nsample < band + 256; nsample++) {
516  float c1 = su1[nsample];
517  float c2 = su2[nsample];
518  su1[nsample] = c1 + c2;
519  su2[nsample] = c1 - c2;
520  }
521  break;
522  default:
523  av_assert1(0);
524  }
525  }
526 }
527 
528 static void get_channel_weights(int index, int flag, float ch[2])
529 {
530  if (index == 7) {
531  ch[0] = 1.0;
532  ch[1] = 1.0;
533  } else {
534  ch[0] = (index & 7) / 7.0;
535  ch[1] = sqrt(2 - ch[0] * ch[0]);
536  if (flag)
537  FFSWAP(float, ch[0], ch[1]);
538  }
539 }
540 
541 static void channel_weighting(float *su1, float *su2, int *p3)
542 {
543  int band, nsample;
544  /* w[x][y] y=0 is left y=1 is right */
545  float w[2][2];
546 
547  if (p3[1] != 7 || p3[3] != 7) {
548  get_channel_weights(p3[1], p3[0], w[0]);
549  get_channel_weights(p3[3], p3[2], w[1]);
550 
551  for (band = 256; band < 4 * 256; band += 256) {
552  for (nsample = band; nsample < band + 8; nsample++) {
553  su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
554  su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
555  }
556  for(; nsample < band + 256; nsample++) {
557  su1[nsample] *= w[1][0];
558  su2[nsample] *= w[1][1];
559  }
560  }
561  }
562 }
563 
564 /**
565  * Decode a Sound Unit
566  *
567  * @param snd the channel unit to be used
568  * @param output the decoded samples before IQMF in float representation
569  * @param channel_num channel number
570  * @param coding_mode the coding mode (JOINT_STEREO or regular stereo/mono)
571  */
573  ChannelUnit *snd, float *output,
574  int channel_num, int coding_mode)
575 {
576  int band, ret, num_subbands, last_tonal, num_bands;
577  GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
578  GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
579 
580  if (coding_mode == JOINT_STEREO && channel_num == 1) {
581  if (get_bits(gb, 2) != 3) {
582  av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
583  return AVERROR_INVALIDDATA;
584  }
585  } else {
586  if (get_bits(gb, 6) != 0x28) {
587  av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
588  return AVERROR_INVALIDDATA;
589  }
590  }
591 
592  /* number of coded QMF bands */
593  snd->bands_coded = get_bits(gb, 2);
594 
595  ret = decode_gain_control(gb, gain2, snd->bands_coded);
596  if (ret)
597  return ret;
598 
600  snd->bands_coded);
601  if (snd->num_components < 0)
602  return snd->num_components;
603 
604  num_subbands = decode_spectrum(gb, snd->spectrum);
605 
606  /* Merge the decoded spectrum and tonal components. */
607  last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
608  snd->components);
609 
610 
611  /* calculate number of used MLT/QMF bands according to the amount of coded
612  spectral lines */
613  num_bands = (subband_tab[num_subbands] - 1) >> 8;
614  if (last_tonal >= 0)
615  num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
616 
617 
618  /* Reconstruct time domain samples. */
619  for (band = 0; band < 4; band++) {
620  /* Perform the IMDCT step without overlapping. */
621  if (band <= num_bands)
622  imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
623  else
624  memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
625 
626  /* gain compensation and overlapping */
628  &snd->prev_frame[band * 256],
629  &gain1->g_block[band], &gain2->g_block[band],
630  256, &output[band * 256]);
631  }
632 
633  /* Swap the gain control buffers for the next frame. */
634  snd->gc_blk_switch ^= 1;
635 
636  return 0;
637 }
638 
639 static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
640  float **out_samples)
641 {
642  ATRAC3Context *q = avctx->priv_data;
643  int ret, i;
644  uint8_t *ptr1;
645 
646  if (q->coding_mode == JOINT_STEREO) {
647  /* channel coupling mode */
648  /* decode Sound Unit 1 */
649  init_get_bits(&q->gb, databuf, avctx->block_align * 8);
650 
651  ret = decode_channel_sound_unit(q, &q->gb, q->units, out_samples[0], 0,
652  JOINT_STEREO);
653  if (ret != 0)
654  return ret;
655 
656  /* Framedata of the su2 in the joint-stereo mode is encoded in
657  * reverse byte order so we need to swap it first. */
658  if (databuf == q->decoded_bytes_buffer) {
659  uint8_t *ptr2 = q->decoded_bytes_buffer + avctx->block_align - 1;
660  ptr1 = q->decoded_bytes_buffer;
661  for (i = 0; i < avctx->block_align / 2; i++, ptr1++, ptr2--)
662  FFSWAP(uint8_t, *ptr1, *ptr2);
663  } else {
664  const uint8_t *ptr2 = databuf + avctx->block_align - 1;
665  for (i = 0; i < avctx->block_align; i++)
666  q->decoded_bytes_buffer[i] = *ptr2--;
667  }
668 
669  /* Skip the sync codes (0xF8). */
670  ptr1 = q->decoded_bytes_buffer;
671  for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
672  if (i >= avctx->block_align)
673  return AVERROR_INVALIDDATA;
674  }
675 
676 
677  /* set the bitstream reader at the start of the second Sound Unit*/
678  init_get_bits8(&q->gb, ptr1, q->decoded_bytes_buffer + avctx->block_align - ptr1);
679 
680  /* Fill the Weighting coeffs delay buffer */
681  memmove(q->weighting_delay, &q->weighting_delay[2],
682  4 * sizeof(*q->weighting_delay));
683  q->weighting_delay[4] = get_bits1(&q->gb);
684  q->weighting_delay[5] = get_bits(&q->gb, 3);
685 
686  for (i = 0; i < 4; i++) {
689  q->matrix_coeff_index_next[i] = get_bits(&q->gb, 2);
690  }
691 
692  /* Decode Sound Unit 2. */
693  ret = decode_channel_sound_unit(q, &q->gb, &q->units[1],
694  out_samples[1], 1, JOINT_STEREO);
695  if (ret != 0)
696  return ret;
697 
698  /* Reconstruct the channel coefficients. */
699  reverse_matrixing(out_samples[0], out_samples[1],
702 
703  channel_weighting(out_samples[0], out_samples[1], q->weighting_delay);
704  } else {
705  /* normal stereo mode or mono */
706  /* Decode the channel sound units. */
707  for (i = 0; i < avctx->channels; i++) {
708  /* Set the bitstream reader at the start of a channel sound unit. */
709  init_get_bits(&q->gb,
710  databuf + i * avctx->block_align / avctx->channels,
711  avctx->block_align * 8 / avctx->channels);
712 
713  ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
714  out_samples[i], i, q->coding_mode);
715  if (ret != 0)
716  return ret;
717  }
718  }
719 
720  /* Apply the iQMF synthesis filter. */
721  for (i = 0; i < avctx->channels; i++) {
722  float *p1 = out_samples[i];
723  float *p2 = p1 + 256;
724  float *p3 = p2 + 256;
725  float *p4 = p3 + 256;
726  ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
727  ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
728  ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
729  }
730 
731  return 0;
732 }
733 
734 static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
735  int *got_frame_ptr, AVPacket *avpkt)
736 {
737  AVFrame *frame = data;
738  const uint8_t *buf = avpkt->data;
739  int buf_size = avpkt->size;
740  ATRAC3Context *q = avctx->priv_data;
741  int ret;
742  const uint8_t *databuf;
743 
744  if (buf_size < avctx->block_align) {
745  av_log(avctx, AV_LOG_ERROR,
746  "Frame too small (%d bytes). Truncated file?\n", buf_size);
747  return AVERROR_INVALIDDATA;
748  }
749 
750  /* get output buffer */
751  frame->nb_samples = SAMPLES_PER_FRAME;
752  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
753  return ret;
754 
755  /* Check if we need to descramble and what buffer to pass on. */
756  if (q->scrambled_stream) {
758  databuf = q->decoded_bytes_buffer;
759  } else {
760  databuf = buf;
761  }
762 
763  ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
764  if (ret) {
765  av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
766  return ret;
767  }
768 
769  *got_frame_ptr = 1;
770 
771  return avctx->block_align;
772 }
773 
775 {
776  int i;
777 
780 
781  /* Initialize the VLC tables. */
782  for (i = 0; i < 7; i++) {
783  spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
784  spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
785  atrac3_vlc_offs[i ];
786  init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
787  huff_bits[i], 1, 1,
789  }
790 }
791 
793 {
794  static int static_init_done;
795  int i, ret;
796  int version, delay, samples_per_frame, frame_factor;
797  const uint8_t *edata_ptr = avctx->extradata;
798  ATRAC3Context *q = avctx->priv_data;
799 
800  if (avctx->channels <= 0 || avctx->channels > 2) {
801  av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
802  return AVERROR(EINVAL);
803  }
804 
805  if (!static_init_done)
807  static_init_done = 1;
808 
809  /* Take care of the codec-specific extradata. */
810  if (avctx->extradata_size == 14) {
811  /* Parse the extradata, WAV format */
812  av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
813  bytestream_get_le16(&edata_ptr)); // Unknown value always 1
814  edata_ptr += 4; // samples per channel
815  q->coding_mode = bytestream_get_le16(&edata_ptr);
816  av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
817  bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
818  frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
819  av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
820  bytestream_get_le16(&edata_ptr)); // Unknown always 0
821 
822  /* setup */
823  samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
824  version = 4;
825  delay = 0x88E;
827  q->scrambled_stream = 0;
828 
829  if (avctx->block_align != 96 * avctx->channels * frame_factor &&
830  avctx->block_align != 152 * avctx->channels * frame_factor &&
831  avctx->block_align != 192 * avctx->channels * frame_factor) {
832  av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
833  "configuration %d/%d/%d\n", avctx->block_align,
834  avctx->channels, frame_factor);
835  return AVERROR_INVALIDDATA;
836  }
837  } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
838  /* Parse the extradata, RM format. */
839  version = bytestream_get_be32(&edata_ptr);
840  samples_per_frame = bytestream_get_be16(&edata_ptr);
841  delay = bytestream_get_be16(&edata_ptr);
842  q->coding_mode = bytestream_get_be16(&edata_ptr);
843  q->scrambled_stream = 1;
844 
845  } else {
846  av_log(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n",
847  avctx->extradata_size);
848  return AVERROR(EINVAL);
849  }
850 
851  /* Check the extradata */
852 
853  if (version != 4) {
854  av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
855  return AVERROR_INVALIDDATA;
856  }
857 
858  if (samples_per_frame != SAMPLES_PER_FRAME &&
859  samples_per_frame != SAMPLES_PER_FRAME * 2) {
860  av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
861  samples_per_frame);
862  return AVERROR_INVALIDDATA;
863  }
864 
865  if (delay != 0x88E) {
866  av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
867  delay);
868  return AVERROR_INVALIDDATA;
869  }
870 
871  if (q->coding_mode == STEREO)
872  av_log(avctx, AV_LOG_DEBUG, "Normal stereo detected.\n");
873  else if (q->coding_mode == JOINT_STEREO) {
874  if (avctx->channels != 2) {
875  av_log(avctx, AV_LOG_ERROR, "Invalid coding mode\n");
876  return AVERROR_INVALIDDATA;
877  }
878  av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
879  } else {
880  av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
881  q->coding_mode);
882  return AVERROR_INVALIDDATA;
883  }
884 
885  if (avctx->block_align >= UINT_MAX / 2)
886  return AVERROR(EINVAL);
887 
890  if (!q->decoded_bytes_buffer)
891  return AVERROR(ENOMEM);
892 
894 
895  /* initialize the MDCT transform */
896  if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
897  av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
899  return ret;
900  }
901 
902  /* init the joint-stereo decoding data */
903  q->weighting_delay[0] = 0;
904  q->weighting_delay[1] = 7;
905  q->weighting_delay[2] = 0;
906  q->weighting_delay[3] = 7;
907  q->weighting_delay[4] = 0;
908  q->weighting_delay[5] = 7;
909 
910  for (i = 0; i < 4; i++) {
911  q->matrix_coeff_index_prev[i] = 3;
912  q->matrix_coeff_index_now[i] = 3;
913  q->matrix_coeff_index_next[i] = 3;
914  }
915 
918 
919  q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
920  if (!q->units || !q->fdsp) {
921  atrac3_decode_close(avctx);
922  return AVERROR(ENOMEM);
923  }
924 
925  return 0;
926 }
927 
929  .name = "atrac3",
930  .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
931  .type = AVMEDIA_TYPE_AUDIO,
932  .id = AV_CODEC_ID_ATRAC3,
933  .priv_data_size = sizeof(ATRAC3Context),
935  .close = atrac3_decode_close,
937  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
938  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
940 };
float, planar
Definition: samplefmt.h:69
static const uint16_t atrac3_vlc_offs[9]
Definition: atrac3data.h:106
#define NULL
Definition: coverity.c:32
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
static void reverse_matrixing(float *su1, float *su2, int *prev_code, int *curr_code)
Definition: atrac3.c:466
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
int matrix_coeff_index_next[4]
Definition: atrac3.c:95
#define JOINT_STEREO
Definition: atrac3.c:51
uint8_t * decoded_bytes_buffer
data buffers
Definition: atrac3.c:100
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:247
static int add_tonal_components(float *spectrum, int num_components, TonalComponent *components)
Combine the tonal band spectrum and regular band spectrum.
Definition: atrac3.c:445
static const uint8_t clc_length_tab[8]
Definition: atrac3data.h:112
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
static const uint8_t *const huff_codes[7]
Definition: atrac3data.h:98
#define MDCT_SIZE
Definition: atrac3.c:55
AtracGCContext gainc_ctx
Definition: atrac3.c:108
#define SAMPLES_PER_FRAME
Definition: atrac3.c:54
TonalComponent components[64]
Definition: atrac3.c:72
int size
Definition: avcodec.h:1591
const char * b
Definition: vf_curves.c:113
int flag
Definition: cpu.c:34
void ff_atrac_iqmf(float *inlo, float *inhi, unsigned int nIn, float *pOut, float *delayBuf, float *temp)
Quadrature mirror synthesis filter.
Definition: atrac.c:127
float coef[8]
Definition: atrac3.c:64
int version
Definition: avisynth_c.h:766
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:252
FFTContext mdct_ctx
Definition: atrac3.c:109
static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
Regular 512 points IMDCT without overlapping, with the exception of the swapping of odd bands caused ...
Definition: atrac3.c:123
#define av_be2ne32(x)
Definition: bswap.h:93
AVCodec.
Definition: avcodec.h:3573
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2458
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:232
GetBitContext gb
Definition: atrac3.c:84
Macro definitions for various function/variable attributes.
int lev_code[7]
level at corresponding control point
Definition: atrac.h:37
static int16_t block[64]
Definition: dct.c:113
float ff_atrac_sf_table[64]
Definition: atrac.c:36
static const uint8_t *const huff_bits[7]
Definition: atrac3data.h:102
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
uint8_t bits
Definition: crc.c:296
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2429
uint8_t
GainBlock gain_block[2]
Definition: atrac3.c:73
#define av_cold
Definition: attributes.h:82
static void channel_weighting(float *su1, float *su2, int *p3)
Definition: atrac3.c:541
static float mdct_window[MDCT_SIZE]
Definition: atrac3.c:113
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1774
static AVFrame * frame
static const int8_t mantissa_clc_tab[4]
Definition: atrac3data.h:114
static const float inv_max_quant[8]
Definition: atrac3data.h:123
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:101
uint8_t * data
Definition: avcodec.h:1590
ATRAC common header.
static const uint64_t c1
Definition: murmur3.c:49
int num_components
Definition: atrac3.c:69
bitstream reader API header.
int loc_code[7]
location of gain control points
Definition: atrac.h:38
static int atrac3_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: atrac3.c:734
#define FFALIGN(x, a)
Definition: macros.h:48
#define av_log(a,...)
static int decode_spectrum(GetBitContext *gb, float *output)
Restore the quantized band spectrum coefficients.
Definition: atrac3.c:267
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define s2
Definition: regdef.h:39
#define init_vlc(vlc, nb_bits, nb_codes,bits, bits_wrap, bits_size,codes, codes_wrap, codes_size,flags)
Definition: vlc.h:38
void(* vector_fmul)(float *dst, const float *src0, const float *src1, int len)
Calculate the entry wise product of two vectors of floats and store the result in a vector of floats...
Definition: float_dsp.h:38
#define AVERROR(e)
Definition: error.h:43
static const struct endianess table[]
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
int bands_coded
Definition: atrac3.c:68
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
AVFloatDSPContext * fdsp
Definition: atrac3.c:110
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1754
float temp_buf[1070]
Definition: atrac3.c:101
const char * name
Name of the codec implementation.
Definition: avcodec.h:3580
#define ff_mdct_init
Definition: fft.h:169
#define FFMAX(a, b)
Definition: common.h:94
Gain compensation context structure.
Definition: atrac.h:44
#define STEREO
Definition: atrac3.c:52
static void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.h:226
av_cold void ff_atrac_init_gain_compensation(AtracGCContext *gctx, int id2exp_offset, int loc_scale)
Initialize gain compensation context.
Definition: atrac.c:66
Definition: vlc.h:26
void(* imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:107
float spectrum[SAMPLES_PER_FRAME]
Definition: atrac3.c:75
static const uint16_t subband_tab[33]
Definition: atrac3data.h:128
int matrix_coeff_index_now[4]
Definition: atrac3.c:94
static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf, float **out_samples)
Definition: atrac3.c:639
Definition: fft.h:88
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:885
#define av_assert1(cond)
assert() equivalent, that does not lie in speed critical code.
Definition: avassert.h:53
static void get_channel_weights(int index, int flag, float ch[2])
Definition: atrac3.c:528
#define FFMIN(a, b)
Definition: common.h:96
static const int8_t mantissa_vlc_tab[18]
Definition: atrac3data.h:116
float prev_frame[SAMPLES_PER_FRAME]
Definition: atrac3.c:70
int num_coefs
Definition: atrac3.c:63
int gc_blk_switch
Definition: atrac3.c:71
float imdct_buf[SAMPLES_PER_FRAME]
Definition: atrac3.c:76
static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
Definition: atrac3.c:149
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:535
static av_always_inline int cmp(MpegEncContext *s, const int x, const int y, const int subx, const int suby, const int size, const int h, int ref_index, int src_index, me_cmp_func cmp_func, me_cmp_func chroma_cmp_func, const int flags)
compares a block (either a full macroblock or a partition thereof) against a proposed motion-compensa...
Definition: motion_est.c:260
static VLC_TYPE atrac3_vlc_table[4096][2]
Definition: atrac3.c:114
int weighting_delay[6]
Definition: atrac3.c:96
int table_allocated
Definition: vlc.h:29
static int decode_gain_control(GetBitContext *gb, GainBlock *block, int num_bands)
Decode gain parameters for the coded bands.
Definition: atrac3.c:409
Gain control parameters for one subband.
Definition: atrac.h:35
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
static void read_quant_spectral_coeffs(GetBitContext *gb, int selector, int coding_flag, int *mantissas, int num_codes)
Mantissa decoding.
Definition: atrac3.c:208
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:437
main external API structure.
Definition: avcodec.h:1659
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:928
int coding_mode
stream data
Definition: atrac3.c:87
void * buf
Definition: avisynth_c.h:690
int extradata_size
Definition: avcodec.h:1775
Replacements for frequently missing libm functions.
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:299
static av_cold int atrac3_decode_close(AVCodecContext *avctx)
Definition: atrac3.c:187
int index
Definition: gxfenc.c:89
#define AV_CODEC_CAP_SUBFRAMES
Codec can output multiple frames per AVPacket Normally demuxers return one frame at a time...
Definition: avcodec.h:1008
AtracGainInfo g_block[4]
Definition: atrac3.c:58
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:406
#define s1
Definition: regdef.h:38
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:119
ChannelUnit * units
Definition: atrac3.c:89
static av_cold void init_imdct_window(void)
Definition: atrac3.c:172
float delay_buf2[46]
Definition: atrac3.c:79
int num_points
number of gain control points
Definition: atrac.h:36
uint8_t level
Definition: svq3.c:206
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
Definition: atrac3.c:792
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
Definition: ccaption_dec.c:752
float delay_buf3[46]
Definition: atrac3.c:80
common internal api header.
int scrambled_stream
extradata
Definition: atrac3.c:105
#define ff_mdct_end
Definition: fft.h:170
static double c[64]
#define INIT_VLC_USE_NEW_STATIC
Definition: vlc.h:55
static const uint64_t c2
Definition: murmur3.c:50
static VLC spectral_coeff_tab[7]
Definition: atrac3.c:115
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_YASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
float delay_buf1[46]
qmf delay buffers
Definition: atrac3.c:78
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:733
AVCodec ff_atrac3_decoder
Definition: atrac3.c:928
void * priv_data
Definition: avcodec.h:1701
static const float matrix_coeffs[8]
Definition: atrac3data.h:137
int channels
number of audio channels
Definition: avcodec.h:2422
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
#define INTERPOLATE(old, new, nsample)
Definition: atrac3.c:463
void ff_atrac_gain_compensation(AtracGCContext *gctx, float *in, float *prev, AtracGainInfo *gc_now, AtracGainInfo *gc_next, int num_samples, float *out)
Apply gain compensation and perform the MDCT overlapping part.
Definition: atrac.c:84
av_cold void ff_atrac_generate_tables(void)
Generate common tables.
Definition: atrac.c:48
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
static av_cold void atrac3_init_static_data(void)
Definition: atrac3.c:774
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
int matrix_coeff_index_prev[4]
joint-stereo related variables
Definition: atrac3.c:93
#define VLC_TYPE
Definition: vlc.h:24
static const uint8_t huff_tab_sizes[7]
Definition: atrac3data.h:94
#define FFSWAP(type, a, b)
Definition: common.h:99
ATRAC3 AKA RealAudio 8 compatible decoder data.
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:231
This structure stores compressed data.
Definition: avcodec.h:1567
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:241
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:958
static int decode_tonal_components(GetBitContext *gb, TonalComponent *components, int num_bands)
Restore the quantized tonal components.
Definition: atrac3.c:325
static int decode_channel_sound_unit(ATRAC3Context *q, GetBitContext *gb, ChannelUnit *snd, float *output, int channel_num, int coding_mode)
Decode a Sound Unit.
Definition: atrac3.c:572