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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 MIN_CHANNELS 1
52 #define MAX_CHANNELS 8
53 #define MAX_JS_PAIRS 8 / 2
54 
55 #define JOINT_STEREO 0x12
56 #define SINGLE 0x2
57 
58 #define SAMPLES_PER_FRAME 1024
59 #define MDCT_SIZE 512
60 
61 typedef struct GainBlock {
63 } GainBlock;
64 
65 typedef struct TonalComponent {
66  int pos;
67  int num_coefs;
68  float coef[8];
70 
71 typedef struct ChannelUnit {
78 
81 
82  float delay_buf1[46]; ///<qmf delay buffers
83  float delay_buf2[46];
84  float delay_buf3[46];
85 } ChannelUnit;
86 
87 typedef struct ATRAC3Context {
89  //@{
90  /** stream data */
92 
94  //@}
95  //@{
96  /** joint-stereo related variables */
101  //@}
102  //@{
103  /** data buffers */
105  float temp_buf[1070];
106  //@}
107  //@{
108  /** extradata */
110  //@}
111 
115 } ATRAC3Context;
116 
118 static VLC_TYPE atrac3_vlc_table[4096][2];
120 
121 /**
122  * Regular 512 points IMDCT without overlapping, with the exception of the
123  * swapping of odd bands caused by the reverse spectra of the QMF.
124  *
125  * @param odd_band 1 if the band is an odd band
126  */
127 static void imlt(ATRAC3Context *q, float *input, float *output, int odd_band)
128 {
129  int i;
130 
131  if (odd_band) {
132  /**
133  * Reverse the odd bands before IMDCT, this is an effect of the QMF
134  * transform or it gives better compression to do it this way.
135  * FIXME: It should be possible to handle this in imdct_calc
136  * for that to happen a modification of the prerotation step of
137  * all SIMD code and C code is needed.
138  * Or fix the functions before so they generate a pre reversed spectrum.
139  */
140  for (i = 0; i < 128; i++)
141  FFSWAP(float, input[i], input[255 - i]);
142  }
143 
144  q->mdct_ctx.imdct_calc(&q->mdct_ctx, output, input);
145 
146  /* Perform windowing on the output. */
147  q->fdsp->vector_fmul(output, output, mdct_window, MDCT_SIZE);
148 }
149 
150 /*
151  * indata descrambling, only used for data coming from the rm container
152  */
153 static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
154 {
155  int i, off;
156  uint32_t c;
157  const uint32_t *buf;
158  uint32_t *output = (uint32_t *)out;
159 
160  off = (intptr_t)input & 3;
161  buf = (const uint32_t *)(input - off);
162  if (off)
163  c = av_be2ne32((0x537F6103U >> (off * 8)) | (0x537F6103U << (32 - (off * 8))));
164  else
165  c = av_be2ne32(0x537F6103U);
166  bytes += 3 + off;
167  for (i = 0; i < bytes / 4; i++)
168  output[i] = c ^ buf[i];
169 
170  if (off)
171  avpriv_request_sample(NULL, "Offset of %d", off);
172 
173  return off;
174 }
175 
176 static av_cold void init_imdct_window(void)
177 {
178  int i, j;
179 
180  /* generate the mdct window, for details see
181  * http://wiki.multimedia.cx/index.php?title=RealAudio_atrc#Windows */
182  for (i = 0, j = 255; i < 128; i++, j--) {
183  float wi = sin(((i + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
184  float wj = sin(((j + 0.5) / 256.0 - 0.5) * M_PI) + 1.0;
185  float w = 0.5 * (wi * wi + wj * wj);
186  mdct_window[i] = mdct_window[511 - i] = wi / w;
187  mdct_window[j] = mdct_window[511 - j] = wj / w;
188  }
189 }
190 
192 {
193  ATRAC3Context *q = avctx->priv_data;
194 
195  av_freep(&q->units);
197  av_freep(&q->fdsp);
198 
199  ff_mdct_end(&q->mdct_ctx);
200 
201  return 0;
202 }
203 
204 /**
205  * Mantissa decoding
206  *
207  * @param selector which table the output values are coded with
208  * @param coding_flag constant length coding or variable length coding
209  * @param mantissas mantissa output table
210  * @param num_codes number of values to get
211  */
212 static void read_quant_spectral_coeffs(GetBitContext *gb, int selector,
213  int coding_flag, int *mantissas,
214  int num_codes)
215 {
216  int i, code, huff_symb;
217 
218  if (selector == 1)
219  num_codes /= 2;
220 
221  if (coding_flag != 0) {
222  /* constant length coding (CLC) */
223  int num_bits = clc_length_tab[selector];
224 
225  if (selector > 1) {
226  for (i = 0; i < num_codes; i++) {
227  if (num_bits)
228  code = get_sbits(gb, num_bits);
229  else
230  code = 0;
231  mantissas[i] = code;
232  }
233  } else {
234  for (i = 0; i < num_codes; i++) {
235  if (num_bits)
236  code = get_bits(gb, num_bits); // num_bits is always 4 in this case
237  else
238  code = 0;
239  mantissas[i * 2 ] = mantissa_clc_tab[code >> 2];
240  mantissas[i * 2 + 1] = mantissa_clc_tab[code & 3];
241  }
242  }
243  } else {
244  /* variable length coding (VLC) */
245  if (selector != 1) {
246  for (i = 0; i < num_codes; i++) {
247  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector-1].table,
248  spectral_coeff_tab[selector-1].bits, 3);
249  huff_symb += 1;
250  code = huff_symb >> 1;
251  if (huff_symb & 1)
252  code = -code;
253  mantissas[i] = code;
254  }
255  } else {
256  for (i = 0; i < num_codes; i++) {
257  huff_symb = get_vlc2(gb, spectral_coeff_tab[selector - 1].table,
258  spectral_coeff_tab[selector - 1].bits, 3);
259  mantissas[i * 2 ] = mantissa_vlc_tab[huff_symb * 2 ];
260  mantissas[i * 2 + 1] = mantissa_vlc_tab[huff_symb * 2 + 1];
261  }
262  }
263  }
264 }
265 
266 /**
267  * Restore the quantized band spectrum coefficients
268  *
269  * @return subband count, fix for broken specification/files
270  */
271 static int decode_spectrum(GetBitContext *gb, float *output)
272 {
273  int num_subbands, coding_mode, i, j, first, last, subband_size;
274  int subband_vlc_index[32], sf_index[32];
275  int mantissas[128];
276  float scale_factor;
277 
278  num_subbands = get_bits(gb, 5); // number of coded subbands
279  coding_mode = get_bits1(gb); // coding Mode: 0 - VLC/ 1-CLC
280 
281  /* get the VLC selector table for the subbands, 0 means not coded */
282  for (i = 0; i <= num_subbands; i++)
283  subband_vlc_index[i] = get_bits(gb, 3);
284 
285  /* read the scale factor indexes from the stream */
286  for (i = 0; i <= num_subbands; i++) {
287  if (subband_vlc_index[i] != 0)
288  sf_index[i] = get_bits(gb, 6);
289  }
290 
291  for (i = 0; i <= num_subbands; i++) {
292  first = subband_tab[i ];
293  last = subband_tab[i + 1];
294 
295  subband_size = last - first;
296 
297  if (subband_vlc_index[i] != 0) {
298  /* decode spectral coefficients for this subband */
299  /* TODO: This can be done faster is several blocks share the
300  * same VLC selector (subband_vlc_index) */
301  read_quant_spectral_coeffs(gb, subband_vlc_index[i], coding_mode,
302  mantissas, subband_size);
303 
304  /* decode the scale factor for this subband */
305  scale_factor = ff_atrac_sf_table[sf_index[i]] *
306  inv_max_quant[subband_vlc_index[i]];
307 
308  /* inverse quantize the coefficients */
309  for (j = 0; first < last; first++, j++)
310  output[first] = mantissas[j] * scale_factor;
311  } else {
312  /* this subband was not coded, so zero the entire subband */
313  memset(output + first, 0, subband_size * sizeof(*output));
314  }
315  }
316 
317  /* clear the subbands that were not coded */
318  first = subband_tab[i];
319  memset(output + first, 0, (SAMPLES_PER_FRAME - first) * sizeof(*output));
320  return num_subbands;
321 }
322 
323 /**
324  * Restore the quantized tonal components
325  *
326  * @param components tonal components
327  * @param num_bands number of coded bands
328  */
330  TonalComponent *components, int num_bands)
331 {
332  int i, b, c, m;
333  int nb_components, coding_mode_selector, coding_mode;
334  int band_flags[4], mantissa[8];
335  int component_count = 0;
336 
337  nb_components = get_bits(gb, 5);
338 
339  /* no tonal components */
340  if (nb_components == 0)
341  return 0;
342 
343  coding_mode_selector = get_bits(gb, 2);
344  if (coding_mode_selector == 2)
345  return AVERROR_INVALIDDATA;
346 
347  coding_mode = coding_mode_selector & 1;
348 
349  for (i = 0; i < nb_components; i++) {
350  int coded_values_per_component, quant_step_index;
351 
352  for (b = 0; b <= num_bands; b++)
353  band_flags[b] = get_bits1(gb);
354 
355  coded_values_per_component = get_bits(gb, 3);
356 
357  quant_step_index = get_bits(gb, 3);
358  if (quant_step_index <= 1)
359  return AVERROR_INVALIDDATA;
360 
361  if (coding_mode_selector == 3)
362  coding_mode = get_bits1(gb);
363 
364  for (b = 0; b < (num_bands + 1) * 4; b++) {
365  int coded_components;
366 
367  if (band_flags[b >> 2] == 0)
368  continue;
369 
370  coded_components = get_bits(gb, 3);
371 
372  for (c = 0; c < coded_components; c++) {
373  TonalComponent *cmp = &components[component_count];
374  int sf_index, coded_values, max_coded_values;
375  float scale_factor;
376 
377  sf_index = get_bits(gb, 6);
378  if (component_count >= 64)
379  return AVERROR_INVALIDDATA;
380 
381  cmp->pos = b * 64 + get_bits(gb, 6);
382 
383  max_coded_values = SAMPLES_PER_FRAME - cmp->pos;
384  coded_values = coded_values_per_component + 1;
385  coded_values = FFMIN(max_coded_values, coded_values);
386 
387  scale_factor = ff_atrac_sf_table[sf_index] *
388  inv_max_quant[quant_step_index];
389 
390  read_quant_spectral_coeffs(gb, quant_step_index, coding_mode,
391  mantissa, coded_values);
392 
393  cmp->num_coefs = coded_values;
394 
395  /* inverse quant */
396  for (m = 0; m < coded_values; m++)
397  cmp->coef[m] = mantissa[m] * scale_factor;
398 
399  component_count++;
400  }
401  }
402  }
403 
404  return component_count;
405 }
406 
407 /**
408  * Decode gain parameters for the coded bands
409  *
410  * @param block the gainblock for the current band
411  * @param num_bands amount of coded bands
412  */
414  int num_bands)
415 {
416  int b, j;
417  int *level, *loc;
418 
419  AtracGainInfo *gain = block->g_block;
420 
421  for (b = 0; b <= num_bands; b++) {
422  gain[b].num_points = get_bits(gb, 3);
423  level = gain[b].lev_code;
424  loc = gain[b].loc_code;
425 
426  for (j = 0; j < gain[b].num_points; j++) {
427  level[j] = get_bits(gb, 4);
428  loc[j] = get_bits(gb, 5);
429  if (j && loc[j] <= loc[j - 1])
430  return AVERROR_INVALIDDATA;
431  }
432  }
433 
434  /* Clear the unused blocks. */
435  for (; b < 4 ; b++)
436  gain[b].num_points = 0;
437 
438  return 0;
439 }
440 
441 /**
442  * Combine the tonal band spectrum and regular band spectrum
443  *
444  * @param spectrum output spectrum buffer
445  * @param num_components number of tonal components
446  * @param components tonal components for this band
447  * @return position of the last tonal coefficient
448  */
449 static int add_tonal_components(float *spectrum, int num_components,
450  TonalComponent *components)
451 {
452  int i, j, last_pos = -1;
453  float *input, *output;
454 
455  for (i = 0; i < num_components; i++) {
456  last_pos = FFMAX(components[i].pos + components[i].num_coefs, last_pos);
457  input = components[i].coef;
458  output = &spectrum[components[i].pos];
459 
460  for (j = 0; j < components[i].num_coefs; j++)
461  output[j] += input[j];
462  }
463 
464  return last_pos;
465 }
466 
467 #define INTERPOLATE(old, new, nsample) \
468  ((old) + (nsample) * 0.125 * ((new) - (old)))
469 
470 static void reverse_matrixing(float *su1, float *su2, int *prev_code,
471  int *curr_code)
472 {
473  int i, nsample, band;
474  float mc1_l, mc1_r, mc2_l, mc2_r;
475 
476  for (i = 0, band = 0; band < 4 * 256; band += 256, i++) {
477  int s1 = prev_code[i];
478  int s2 = curr_code[i];
479  nsample = band;
480 
481  if (s1 != s2) {
482  /* Selector value changed, interpolation needed. */
483  mc1_l = matrix_coeffs[s1 * 2 ];
484  mc1_r = matrix_coeffs[s1 * 2 + 1];
485  mc2_l = matrix_coeffs[s2 * 2 ];
486  mc2_r = matrix_coeffs[s2 * 2 + 1];
487 
488  /* Interpolation is done over the first eight samples. */
489  for (; nsample < band + 8; nsample++) {
490  float c1 = su1[nsample];
491  float c2 = su2[nsample];
492  c2 = c1 * INTERPOLATE(mc1_l, mc2_l, nsample - band) +
493  c2 * INTERPOLATE(mc1_r, mc2_r, nsample - band);
494  su1[nsample] = c2;
495  su2[nsample] = c1 * 2.0 - c2;
496  }
497  }
498 
499  /* Apply the matrix without interpolation. */
500  switch (s2) {
501  case 0: /* M/S decoding */
502  for (; nsample < band + 256; nsample++) {
503  float c1 = su1[nsample];
504  float c2 = su2[nsample];
505  su1[nsample] = c2 * 2.0;
506  su2[nsample] = (c1 - c2) * 2.0;
507  }
508  break;
509  case 1:
510  for (; nsample < band + 256; nsample++) {
511  float c1 = su1[nsample];
512  float c2 = su2[nsample];
513  su1[nsample] = (c1 + c2) * 2.0;
514  su2[nsample] = c2 * -2.0;
515  }
516  break;
517  case 2:
518  case 3:
519  for (; nsample < band + 256; nsample++) {
520  float c1 = su1[nsample];
521  float c2 = su2[nsample];
522  su1[nsample] = c1 + c2;
523  su2[nsample] = c1 - c2;
524  }
525  break;
526  default:
527  av_assert1(0);
528  }
529  }
530 }
531 
532 static void get_channel_weights(int index, int flag, float ch[2])
533 {
534  if (index == 7) {
535  ch[0] = 1.0;
536  ch[1] = 1.0;
537  } else {
538  ch[0] = (index & 7) / 7.0;
539  ch[1] = sqrt(2 - ch[0] * ch[0]);
540  if (flag)
541  FFSWAP(float, ch[0], ch[1]);
542  }
543 }
544 
545 static void channel_weighting(float *su1, float *su2, int *p3)
546 {
547  int band, nsample;
548  /* w[x][y] y=0 is left y=1 is right */
549  float w[2][2];
550 
551  if (p3[1] != 7 || p3[3] != 7) {
552  get_channel_weights(p3[1], p3[0], w[0]);
553  get_channel_weights(p3[3], p3[2], w[1]);
554 
555  for (band = 256; band < 4 * 256; band += 256) {
556  for (nsample = band; nsample < band + 8; nsample++) {
557  su1[nsample] *= INTERPOLATE(w[0][0], w[0][1], nsample - band);
558  su2[nsample] *= INTERPOLATE(w[1][0], w[1][1], nsample - band);
559  }
560  for(; nsample < band + 256; nsample++) {
561  su1[nsample] *= w[1][0];
562  su2[nsample] *= w[1][1];
563  }
564  }
565  }
566 }
567 
568 /**
569  * Decode a Sound Unit
570  *
571  * @param snd the channel unit to be used
572  * @param output the decoded samples before IQMF in float representation
573  * @param channel_num channel number
574  * @param coding_mode the coding mode (JOINT_STEREO or single channels)
575  */
577  ChannelUnit *snd, float *output,
578  int channel_num, int coding_mode)
579 {
580  int band, ret, num_subbands, last_tonal, num_bands;
581  GainBlock *gain1 = &snd->gain_block[ snd->gc_blk_switch];
582  GainBlock *gain2 = &snd->gain_block[1 - snd->gc_blk_switch];
583 
584  if (coding_mode == JOINT_STEREO && (channel_num % 2) == 1) {
585  if (get_bits(gb, 2) != 3) {
586  av_log(NULL,AV_LOG_ERROR,"JS mono Sound Unit id != 3.\n");
587  return AVERROR_INVALIDDATA;
588  }
589  } else {
590  if (get_bits(gb, 6) != 0x28) {
591  av_log(NULL,AV_LOG_ERROR,"Sound Unit id != 0x28.\n");
592  return AVERROR_INVALIDDATA;
593  }
594  }
595 
596  /* number of coded QMF bands */
597  snd->bands_coded = get_bits(gb, 2);
598 
599  ret = decode_gain_control(gb, gain2, snd->bands_coded);
600  if (ret)
601  return ret;
602 
604  snd->bands_coded);
605  if (snd->num_components < 0)
606  return snd->num_components;
607 
608  num_subbands = decode_spectrum(gb, snd->spectrum);
609 
610  /* Merge the decoded spectrum and tonal components. */
611  last_tonal = add_tonal_components(snd->spectrum, snd->num_components,
612  snd->components);
613 
614 
615  /* calculate number of used MLT/QMF bands according to the amount of coded
616  spectral lines */
617  num_bands = (subband_tab[num_subbands] - 1) >> 8;
618  if (last_tonal >= 0)
619  num_bands = FFMAX((last_tonal + 256) >> 8, num_bands);
620 
621 
622  /* Reconstruct time domain samples. */
623  for (band = 0; band < 4; band++) {
624  /* Perform the IMDCT step without overlapping. */
625  if (band <= num_bands)
626  imlt(q, &snd->spectrum[band * 256], snd->imdct_buf, band & 1);
627  else
628  memset(snd->imdct_buf, 0, 512 * sizeof(*snd->imdct_buf));
629 
630  /* gain compensation and overlapping */
632  &snd->prev_frame[band * 256],
633  &gain1->g_block[band], &gain2->g_block[band],
634  256, &output[band * 256]);
635  }
636 
637  /* Swap the gain control buffers for the next frame. */
638  snd->gc_blk_switch ^= 1;
639 
640  return 0;
641 }
642 
643 static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
644  float **out_samples)
645 {
646  ATRAC3Context *q = avctx->priv_data;
647  int ret, i, ch;
648  uint8_t *ptr1;
649 
650  if (q->coding_mode == JOINT_STEREO) {
651  /* channel coupling mode */
652 
653  /* Decode sound unit pairs (channels are expected to be even).
654  * Multichannel joint stereo interleaves pairs (6ch: 2ch + 2ch + 2ch) */
655  const uint8_t *js_databuf;
656  int js_pair, js_block_align;
657 
658  js_block_align = (avctx->block_align / avctx->channels) * 2; /* block pair */
659 
660  for (ch = 0; ch < avctx->channels; ch = ch + 2) {
661  js_pair = ch/2;
662  js_databuf = databuf + js_pair * js_block_align; /* align to current pair */
663 
664  /* Set the bitstream reader at the start of first channel sound unit. */
665  init_get_bits(&q->gb,
666  js_databuf, js_block_align * 8);
667 
668  /* decode Sound Unit 1 */
669  ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch],
670  out_samples[ch], ch, JOINT_STEREO);
671  if (ret != 0)
672  return ret;
673 
674  /* Framedata of the su2 in the joint-stereo mode is encoded in
675  * reverse byte order so we need to swap it first. */
676  if (js_databuf == q->decoded_bytes_buffer) {
677  uint8_t *ptr2 = q->decoded_bytes_buffer + js_block_align - 1;
678  ptr1 = q->decoded_bytes_buffer;
679  for (i = 0; i < js_block_align / 2; i++, ptr1++, ptr2--)
680  FFSWAP(uint8_t, *ptr1, *ptr2);
681  } else {
682  const uint8_t *ptr2 = js_databuf + js_block_align - 1;
683  for (i = 0; i < js_block_align; i++)
684  q->decoded_bytes_buffer[i] = *ptr2--;
685  }
686 
687  /* Skip the sync codes (0xF8). */
688  ptr1 = q->decoded_bytes_buffer;
689  for (i = 4; *ptr1 == 0xF8; i++, ptr1++) {
690  if (i >= js_block_align)
691  return AVERROR_INVALIDDATA;
692  }
693 
694 
695  /* set the bitstream reader at the start of the second Sound Unit */
696  ret = init_get_bits8(&q->gb,
697  ptr1, q->decoded_bytes_buffer + js_block_align - ptr1);
698  if (ret < 0)
699  return ret;
700 
701  /* Fill the Weighting coeffs delay buffer */
702  memmove(q->weighting_delay[js_pair], &q->weighting_delay[js_pair][2],
703  4 * sizeof(*q->weighting_delay[js_pair]));
704  q->weighting_delay[js_pair][4] = get_bits1(&q->gb);
705  q->weighting_delay[js_pair][5] = get_bits(&q->gb, 3);
706 
707  for (i = 0; i < 4; i++) {
708  q->matrix_coeff_index_prev[js_pair][i] = q->matrix_coeff_index_now[js_pair][i];
709  q->matrix_coeff_index_now[js_pair][i] = q->matrix_coeff_index_next[js_pair][i];
710  q->matrix_coeff_index_next[js_pair][i] = get_bits(&q->gb, 2);
711  }
712 
713  /* Decode Sound Unit 2. */
714  ret = decode_channel_sound_unit(q, &q->gb, &q->units[ch+1],
715  out_samples[ch+1], ch+1, JOINT_STEREO);
716  if (ret != 0)
717  return ret;
718 
719  /* Reconstruct the channel coefficients. */
720  reverse_matrixing(out_samples[ch], out_samples[ch+1],
721  q->matrix_coeff_index_prev[js_pair],
722  q->matrix_coeff_index_now[js_pair]);
723 
724  channel_weighting(out_samples[ch], out_samples[ch+1], q->weighting_delay[js_pair]);
725  }
726  } else {
727  /* single channels */
728  /* Decode the channel sound units. */
729  for (i = 0; i < avctx->channels; i++) {
730  /* Set the bitstream reader at the start of a channel sound unit. */
731  init_get_bits(&q->gb,
732  databuf + i * avctx->block_align / avctx->channels,
733  avctx->block_align * 8 / avctx->channels);
734 
735  ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
736  out_samples[i], i, q->coding_mode);
737  if (ret != 0)
738  return ret;
739  }
740  }
741 
742  /* Apply the iQMF synthesis filter. */
743  for (i = 0; i < avctx->channels; i++) {
744  float *p1 = out_samples[i];
745  float *p2 = p1 + 256;
746  float *p3 = p2 + 256;
747  float *p4 = p3 + 256;
748  ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
749  ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
750  ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
751  }
752 
753  return 0;
754 }
755 
756 static int al_decode_frame(AVCodecContext *avctx, const uint8_t *databuf,
757  int size, float **out_samples)
758 {
759  ATRAC3Context *q = avctx->priv_data;
760  int ret, i;
761 
762  /* Set the bitstream reader at the start of a channel sound unit. */
763  init_get_bits(&q->gb, databuf, size * 8);
764  /* single channels */
765  /* Decode the channel sound units. */
766  for (i = 0; i < avctx->channels; i++) {
767  ret = decode_channel_sound_unit(q, &q->gb, &q->units[i],
768  out_samples[i], i, q->coding_mode);
769  if (ret != 0)
770  return ret;
771  while (i < avctx->channels && get_bits_left(&q->gb) > 6 && show_bits(&q->gb, 6) != 0x28) {
772  skip_bits(&q->gb, 1);
773  }
774  }
775 
776  /* Apply the iQMF synthesis filter. */
777  for (i = 0; i < avctx->channels; i++) {
778  float *p1 = out_samples[i];
779  float *p2 = p1 + 256;
780  float *p3 = p2 + 256;
781  float *p4 = p3 + 256;
782  ff_atrac_iqmf(p1, p2, 256, p1, q->units[i].delay_buf1, q->temp_buf);
783  ff_atrac_iqmf(p4, p3, 256, p3, q->units[i].delay_buf2, q->temp_buf);
784  ff_atrac_iqmf(p1, p3, 512, p1, q->units[i].delay_buf3, q->temp_buf);
785  }
786 
787  return 0;
788 }
789 
790 static int atrac3_decode_frame(AVCodecContext *avctx, void *data,
791  int *got_frame_ptr, AVPacket *avpkt)
792 {
793  AVFrame *frame = data;
794  const uint8_t *buf = avpkt->data;
795  int buf_size = avpkt->size;
796  ATRAC3Context *q = avctx->priv_data;
797  int ret;
798  const uint8_t *databuf;
799 
800  if (buf_size < avctx->block_align) {
801  av_log(avctx, AV_LOG_ERROR,
802  "Frame too small (%d bytes). Truncated file?\n", buf_size);
803  return AVERROR_INVALIDDATA;
804  }
805 
806  /* get output buffer */
807  frame->nb_samples = SAMPLES_PER_FRAME;
808  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
809  return ret;
810 
811  /* Check if we need to descramble and what buffer to pass on. */
812  if (q->scrambled_stream) {
814  databuf = q->decoded_bytes_buffer;
815  } else {
816  databuf = buf;
817  }
818 
819  ret = decode_frame(avctx, databuf, (float **)frame->extended_data);
820  if (ret) {
821  av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
822  return ret;
823  }
824 
825  *got_frame_ptr = 1;
826 
827  return avctx->block_align;
828 }
829 
830 static int atrac3al_decode_frame(AVCodecContext *avctx, void *data,
831  int *got_frame_ptr, AVPacket *avpkt)
832 {
833  AVFrame *frame = data;
834  int ret;
835 
836  frame->nb_samples = SAMPLES_PER_FRAME;
837  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
838  return ret;
839 
840  ret = al_decode_frame(avctx, avpkt->data, avpkt->size,
841  (float **)frame->extended_data);
842  if (ret) {
843  av_log(avctx, AV_LOG_ERROR, "Frame decoding error!\n");
844  return ret;
845  }
846 
847  *got_frame_ptr = 1;
848 
849  return avpkt->size;
850 }
851 
853 {
854  int i;
855 
858 
859  /* Initialize the VLC tables. */
860  for (i = 0; i < 7; i++) {
861  spectral_coeff_tab[i].table = &atrac3_vlc_table[atrac3_vlc_offs[i]];
862  spectral_coeff_tab[i].table_allocated = atrac3_vlc_offs[i + 1] -
863  atrac3_vlc_offs[i ];
864  init_vlc(&spectral_coeff_tab[i], 9, huff_tab_sizes[i],
865  huff_bits[i], 1, 1,
867  }
868 }
869 
871 {
872  static int static_init_done;
873  int i, js_pair, ret;
874  int version, delay, samples_per_frame, frame_factor;
875  const uint8_t *edata_ptr = avctx->extradata;
876  ATRAC3Context *q = avctx->priv_data;
877 
878  if (avctx->channels < MIN_CHANNELS || avctx->channels > MAX_CHANNELS) {
879  av_log(avctx, AV_LOG_ERROR, "Channel configuration error!\n");
880  return AVERROR(EINVAL);
881  }
882 
883  if (!static_init_done)
885  static_init_done = 1;
886 
887  /* Take care of the codec-specific extradata. */
888  if (avctx->codec_id == AV_CODEC_ID_ATRAC3AL) {
889  version = 4;
890  samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
891  delay = 0x88E;
892  q->coding_mode = SINGLE;
893  } else if (avctx->extradata_size == 14) {
894  /* Parse the extradata, WAV format */
895  av_log(avctx, AV_LOG_DEBUG, "[0-1] %d\n",
896  bytestream_get_le16(&edata_ptr)); // Unknown value always 1
897  edata_ptr += 4; // samples per channel
898  q->coding_mode = bytestream_get_le16(&edata_ptr);
899  av_log(avctx, AV_LOG_DEBUG,"[8-9] %d\n",
900  bytestream_get_le16(&edata_ptr)); //Dupe of coding mode
901  frame_factor = bytestream_get_le16(&edata_ptr); // Unknown always 1
902  av_log(avctx, AV_LOG_DEBUG,"[12-13] %d\n",
903  bytestream_get_le16(&edata_ptr)); // Unknown always 0
904 
905  /* setup */
906  samples_per_frame = SAMPLES_PER_FRAME * avctx->channels;
907  version = 4;
908  delay = 0x88E;
910  q->scrambled_stream = 0;
911 
912  if (avctx->block_align != 96 * avctx->channels * frame_factor &&
913  avctx->block_align != 152 * avctx->channels * frame_factor &&
914  avctx->block_align != 192 * avctx->channels * frame_factor) {
915  av_log(avctx, AV_LOG_ERROR, "Unknown frame/channel/frame_factor "
916  "configuration %d/%d/%d\n", avctx->block_align,
917  avctx->channels, frame_factor);
918  return AVERROR_INVALIDDATA;
919  }
920  } else if (avctx->extradata_size == 12 || avctx->extradata_size == 10) {
921  /* Parse the extradata, RM format. */
922  version = bytestream_get_be32(&edata_ptr);
923  samples_per_frame = bytestream_get_be16(&edata_ptr);
924  delay = bytestream_get_be16(&edata_ptr);
925  q->coding_mode = bytestream_get_be16(&edata_ptr);
926  q->scrambled_stream = 1;
927 
928  } else {
929  av_log(avctx, AV_LOG_ERROR, "Unknown extradata size %d.\n",
930  avctx->extradata_size);
931  return AVERROR(EINVAL);
932  }
933 
934  /* Check the extradata */
935 
936  if (version != 4) {
937  av_log(avctx, AV_LOG_ERROR, "Version %d != 4.\n", version);
938  return AVERROR_INVALIDDATA;
939  }
940 
941  if (samples_per_frame != SAMPLES_PER_FRAME * avctx->channels) {
942  av_log(avctx, AV_LOG_ERROR, "Unknown amount of samples per frame %d.\n",
943  samples_per_frame);
944  return AVERROR_INVALIDDATA;
945  }
946 
947  if (delay != 0x88E) {
948  av_log(avctx, AV_LOG_ERROR, "Unknown amount of delay %x != 0x88E.\n",
949  delay);
950  return AVERROR_INVALIDDATA;
951  }
952 
953  if (q->coding_mode == SINGLE)
954  av_log(avctx, AV_LOG_DEBUG, "Single channels detected.\n");
955  else if (q->coding_mode == JOINT_STEREO) {
956  if (avctx->channels % 2 == 1) { /* Joint stereo channels must be even */
957  av_log(avctx, AV_LOG_ERROR, "Invalid joint stereo channel configuration.\n");
958  return AVERROR_INVALIDDATA;
959  }
960  av_log(avctx, AV_LOG_DEBUG, "Joint stereo detected.\n");
961  } else {
962  av_log(avctx, AV_LOG_ERROR, "Unknown channel coding mode %x!\n",
963  q->coding_mode);
964  return AVERROR_INVALIDDATA;
965  }
966 
967  if (avctx->block_align >= UINT_MAX / 2)
968  return AVERROR(EINVAL);
969 
972  if (!q->decoded_bytes_buffer)
973  return AVERROR(ENOMEM);
974 
976 
977  /* initialize the MDCT transform */
978  if ((ret = ff_mdct_init(&q->mdct_ctx, 9, 1, 1.0 / 32768)) < 0) {
979  av_log(avctx, AV_LOG_ERROR, "Error initializing MDCT\n");
981  return ret;
982  }
983 
984  /* init the joint-stereo decoding data */
985  for (js_pair = 0; js_pair < MAX_JS_PAIRS; js_pair++) {
986  q->weighting_delay[js_pair][0] = 0;
987  q->weighting_delay[js_pair][1] = 7;
988  q->weighting_delay[js_pair][2] = 0;
989  q->weighting_delay[js_pair][3] = 7;
990  q->weighting_delay[js_pair][4] = 0;
991  q->weighting_delay[js_pair][5] = 7;
992 
993  for (i = 0; i < 4; i++) {
994  q->matrix_coeff_index_prev[js_pair][i] = 3;
995  q->matrix_coeff_index_now[js_pair][i] = 3;
996  q->matrix_coeff_index_next[js_pair][i] = 3;
997  }
998  }
999 
1002 
1003  q->units = av_mallocz_array(avctx->channels, sizeof(*q->units));
1004  if (!q->units || !q->fdsp) {
1005  atrac3_decode_close(avctx);
1006  return AVERROR(ENOMEM);
1007  }
1008 
1009  return 0;
1010 }
1011 
1013  .name = "atrac3",
1014  .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 (Adaptive TRansform Acoustic Coding 3)"),
1015  .type = AVMEDIA_TYPE_AUDIO,
1016  .id = AV_CODEC_ID_ATRAC3,
1017  .priv_data_size = sizeof(ATRAC3Context),
1019  .close = atrac3_decode_close,
1021  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1022  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1024 };
1025 
1027  .name = "atrac3al",
1028  .long_name = NULL_IF_CONFIG_SMALL("ATRAC3 AL (Adaptive TRansform Acoustic Coding 3 Advanced Lossless)"),
1029  .type = AVMEDIA_TYPE_AUDIO,
1030  .id = AV_CODEC_ID_ATRAC3AL,
1031  .priv_data_size = sizeof(ATRAC3Context),
1033  .close = atrac3_decode_close,
1035  .capabilities = AV_CODEC_CAP_SUBFRAMES | AV_CODEC_CAP_DR1,
1036  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1038 };
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:194
static void reverse_matrixing(float *su1, float *su2, int *prev_code, int *curr_code)
Definition: atrac3.c:470
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
#define JOINT_STEREO
Definition: atrac3.c:55
uint8_t * decoded_bytes_buffer
data buffers
Definition: atrac3.c:104
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:261
static int add_tonal_components(float *spectrum, int num_components, TonalComponent *components)
Combine the tonal band spectrum and regular band spectrum.
Definition: atrac3.c:449
static const uint8_t clc_length_tab[8]
Definition: atrac3data.h:112
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
#define SINGLE
Definition: atrac3.c:56
static const uint8_t *const huff_codes[7]
Definition: atrac3data.h:98
#define MDCT_SIZE
Definition: atrac3.c:59
AtracGCContext gainc_ctx
Definition: atrac3.c:112
#define SAMPLES_PER_FRAME
Definition: atrac3.c:58
TonalComponent components[64]
Definition: atrac3.c:76
int size
Definition: avcodec.h:1669
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:68
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:222
FFTContext mdct_ctx
Definition: atrac3.c:113
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:127
#define av_be2ne32(x)
Definition: bswap.h:93
AVCodec.
Definition: avcodec.h:3719
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2542
static int get_sbits(GetBitContext *s, int n)
Definition: get_bits.h:246
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
GetBitContext gb
Definition: atrac3.c:88
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:115
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:2513
uint8_t
GainBlock gain_block[2]
Definition: atrac3.c:77
#define av_cold
Definition: attributes.h:82
static void channel_weighting(float *su1, float *su2, int *p3)
Definition: atrac3.c:545
static float mdct_window[MDCT_SIZE]
Definition: atrac3.c:117
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1858
static AVFrame * frame
static const int8_t mantissa_clc_tab[4]
Definition: atrac3data.h:114
int matrix_coeff_index_now[MAX_JS_PAIRS][4]
Definition: atrac3.c:98
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:104
uint8_t * data
Definition: avcodec.h:1668
ATRAC common header.
static const uint64_t c1
Definition: murmur3.c:49
int num_components
Definition: atrac3.c:73
bitstream reader API header.
ptrdiff_t size
Definition: opengl_enc.c:101
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:790
#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:271
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:587
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
#define s2
Definition: regdef.h:39
#define MAX_JS_PAIRS
Definition: atrac3.c:53
#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:179
int bands_coded
Definition: atrac3.c:72
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
int matrix_coeff_index_next[MAX_JS_PAIRS][4]
Definition: atrac3.c:99
AVFloatDSPContext * fdsp
Definition: atrac3.c:114
int flags
AV_CODEC_FLAG_*.
Definition: avcodec.h:1838
float temp_buf[1070]
Definition: atrac3.c:105
const char * name
Name of the codec implementation.
Definition: avcodec.h:3726
#define ff_mdct_init
Definition: fft.h:169
#define FFMAX(a, b)
Definition: common.h:94
Gain compensation context structure.
Definition: atrac.h:44
static void * av_mallocz_array(size_t nmemb, size_t size)
Definition: mem.h:229
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:79
AVCodec ff_atrac3al_decoder
Definition: atrac3.c:1026
static const uint16_t subband_tab[33]
Definition: atrac3data.h:128
static int decode_frame(AVCodecContext *avctx, const uint8_t *databuf, float **out_samples)
Definition: atrac3.c:643
Definition: fft.h:88
#define AV_CODEC_FLAG_BITEXACT
Use only bitexact stuff (except (I)DCT).
Definition: avcodec.h:925
#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:532
#define FFMIN(a, b)
Definition: common.h:96
#define MIN_CHANNELS
Definition: atrac3.c:51
static const int8_t mantissa_vlc_tab[18]
Definition: atrac3data.h:116
float prev_frame[SAMPLES_PER_FRAME]
Definition: atrac3.c:74
int num_coefs
Definition: atrac3.c:67
int gc_blk_switch
Definition: atrac3.c:75
float imdct_buf[SAMPLES_PER_FRAME]
Definition: atrac3.c:80
static int decode_bytes(const uint8_t *input, uint8_t *out, int bytes)
Definition: atrac3.c:153
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
Definition: get_bits.h:296
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:554
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:118
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:413
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:212
enum AVCodecID codec_id
Definition: avcodec.h:1760
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:456
main external API structure.
Definition: avcodec.h:1743
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1744
int coding_mode
stream data
Definition: atrac3.c:91
void * buf
Definition: avisynth_c.h:690
int extradata_size
Definition: avcodec.h:1859
Replacements for frequently missing libm functions.
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:313
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:306
static av_cold int atrac3_decode_close(AVCodecContext *avctx)
Definition: atrac3.c:191
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:1048
AtracGainInfo g_block[4]
Definition: atrac3.c:62
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:425
#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:127
ChannelUnit * units
Definition: atrac3.c:93
static av_cold void init_imdct_window(void)
Definition: atrac3.c:176
float delay_buf2[46]
Definition: atrac3.c:83
int num_points
number of gain control points
Definition: atrac.h:36
uint8_t level
Definition: svq3.c:207
static av_cold int atrac3_decode_init(AVCodecContext *avctx)
Definition: atrac3.c:870
float delay_buf3[46]
Definition: atrac3.c:84
static int atrac3al_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: atrac3.c:830
common internal api header.
int scrambled_stream
extradata
Definition: atrac3.c:109
#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:119
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
static int al_decode_frame(AVCodecContext *avctx, const uint8_t *databuf, int size, float **out_samples)
Definition: atrac3.c:756
float delay_buf1[46]
qmf delay buffers
Definition: atrac3.c:82
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:773
AVCodec ff_atrac3_decoder
Definition: atrac3.c:1012
void * priv_data
Definition: avcodec.h:1785
#define MAX_CHANNELS
Definition: atrac3.c:52
static const float matrix_coeffs[8]
Definition: atrac3data.h:137
int channels
number of audio channels
Definition: avcodec.h:2506
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
#define INTERPOLATE(old, new, nsample)
Definition: atrac3.c:467
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:852
#define av_freep(p)
int weighting_delay[MAX_JS_PAIRS][6]
Definition: atrac3.c:100
#define M_PI
Definition: mathematics.h:52
#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
int matrix_coeff_index_prev[MAX_JS_PAIRS][4]
joint-stereo related variables
Definition: atrac3.c:97
ATRAC3 AKA RealAudio 8 compatible decoder data.
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:241
This structure stores compressed data.
Definition: avcodec.h:1645
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:260
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:998
static int decode_tonal_components(GetBitContext *gb, TonalComponent *components, int num_bands)
Restore the quantized tonal components.
Definition: atrac3.c:329
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:576