FFmpeg
wmadec.c
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1 /*
2  * WMA compatible decoder
3  * Copyright (c) 2002 The FFmpeg Project
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * WMA compatible decoder.
25  * This decoder handles Microsoft Windows Media Audio data, versions 1 & 2.
26  * WMA v1 is identified by audio format 0x160 in Microsoft media files
27  * (ASF/AVI/WAV). WMA v2 is identified by audio format 0x161.
28  *
29  * To use this decoder, a calling application must supply the extra data
30  * bytes provided with the WMA data. These are the extra, codec-specific
31  * bytes at the end of a WAVEFORMATEX data structure. Transmit these bytes
32  * to the decoder using the extradata[_size] fields in AVCodecContext. There
33  * should be 4 extra bytes for v1 data and 6 extra bytes for v2 data.
34  */
35 
36 #include "libavutil/attributes.h"
37 #include "libavutil/ffmath.h"
38 
39 #include "avcodec.h"
40 #include "internal.h"
41 #include "wma.h"
42 
43 #define EXPVLCBITS 8
44 #define EXPMAX ((19 + EXPVLCBITS - 1) / EXPVLCBITS)
45 
46 #define HGAINVLCBITS 9
47 #define HGAINMAX ((13 + HGAINVLCBITS - 1) / HGAINVLCBITS)
48 
49 static void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len);
50 
51 #ifdef TRACE
52 static void dump_floats(WMACodecContext *s, const char *name,
53  int prec, const float *tab, int n)
54 {
55  int i;
56 
57  ff_tlog(s->avctx, "%s[%d]:\n", name, n);
58  for (i = 0; i < n; i++) {
59  if ((i & 7) == 0)
60  ff_tlog(s->avctx, "%4d: ", i);
61  ff_tlog(s->avctx, " %8.*f", prec, tab[i]);
62  if ((i & 7) == 7)
63  ff_tlog(s->avctx, "\n");
64  }
65  if ((i & 7) != 0)
66  ff_tlog(s->avctx, "\n");
67 }
68 #endif /* TRACE */
69 
71 {
72  WMACodecContext *s = avctx->priv_data;
73  int i, flags2;
74  uint8_t *extradata;
75 
76  if (!avctx->block_align) {
77  av_log(avctx, AV_LOG_ERROR, "block_align is not set\n");
78  return AVERROR(EINVAL);
79  }
80 
81  s->avctx = avctx;
82 
83  /* extract flag info */
84  flags2 = 0;
85  extradata = avctx->extradata;
86  if (avctx->codec->id == AV_CODEC_ID_WMAV1 && avctx->extradata_size >= 4)
87  flags2 = AV_RL16(extradata + 2);
88  else if (avctx->codec->id == AV_CODEC_ID_WMAV2 && avctx->extradata_size >= 6)
89  flags2 = AV_RL16(extradata + 4);
90 
91  s->use_exp_vlc = flags2 & 0x0001;
92  s->use_bit_reservoir = flags2 & 0x0002;
93  s->use_variable_block_len = flags2 & 0x0004;
94 
95  if (avctx->codec->id == AV_CODEC_ID_WMAV2 && avctx->extradata_size >= 8){
96  if (AV_RL16(extradata+4)==0xd && s->use_variable_block_len){
97  av_log(avctx, AV_LOG_WARNING, "Disabling use_variable_block_len, if this fails contact the ffmpeg developers and send us the file\n");
98  s->use_variable_block_len= 0; // this fixes issue1503
99  }
100  }
101 
102  for (i=0; i<MAX_CHANNELS; i++)
103  s->max_exponent[i] = 1.0;
104 
105  if (ff_wma_init(avctx, flags2) < 0)
106  return -1;
107 
108  /* init MDCT */
109  for (i = 0; i < s->nb_block_sizes; i++)
110  ff_mdct_init(&s->mdct_ctx[i], s->frame_len_bits - i + 1, 1, 1.0 / 32768.0);
111 
112  if (s->use_noise_coding) {
114  &ff_wma_hgain_hufftab[0][1], 2,
115  &ff_wma_hgain_hufftab[0][0], 2, 1, -18, 0, avctx);
116  }
117 
118  if (s->use_exp_vlc)
119  init_vlc(&s->exp_vlc, EXPVLCBITS, sizeof(ff_aac_scalefactor_bits), // FIXME move out of context
121  ff_aac_scalefactor_code, 4, 4, 0);
122  else
123  wma_lsp_to_curve_init(s, s->frame_len);
124 
126 
127  return 0;
128 }
129 
130 /**
131  * compute x^-0.25 with an exponent and mantissa table. We use linear
132  * interpolation to reduce the mantissa table size at a small speed
133  * expense (linear interpolation approximately doubles the number of
134  * bits of precision).
135  */
136 static inline float pow_m1_4(WMACodecContext *s, float x)
137 {
138  union {
139  float f;
140  unsigned int v;
141  } u, t;
142  unsigned int e, m;
143  float a, b;
144 
145  u.f = x;
146  e = u.v >> 23;
147  m = (u.v >> (23 - LSP_POW_BITS)) & ((1 << LSP_POW_BITS) - 1);
148  /* build interpolation scale: 1 <= t < 2. */
149  t.v = ((u.v << LSP_POW_BITS) & ((1 << 23) - 1)) | (127 << 23);
150  a = s->lsp_pow_m_table1[m];
151  b = s->lsp_pow_m_table2[m];
152  return s->lsp_pow_e_table[e] * (a + b * t.f);
153 }
154 
155 static av_cold void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len)
156 {
157  float wdel, a, b;
158  int i, e, m;
159 
160  wdel = M_PI / frame_len;
161  for (i = 0; i < frame_len; i++)
162  s->lsp_cos_table[i] = 2.0f * cos(wdel * i);
163 
164  /* tables for x^-0.25 computation */
165  for (i = 0; i < 256; i++) {
166  e = i - 126;
167  s->lsp_pow_e_table[i] = exp2f(e * -0.25);
168  }
169 
170  /* NOTE: these two tables are needed to avoid two operations in
171  * pow_m1_4 */
172  b = 1.0;
173  for (i = (1 << LSP_POW_BITS) - 1; i >= 0; i--) {
174  m = (1 << LSP_POW_BITS) + i;
175  a = (float) m * (0.5 / (1 << LSP_POW_BITS));
176  a = 1/sqrt(sqrt(a));
177  s->lsp_pow_m_table1[i] = 2 * a - b;
178  s->lsp_pow_m_table2[i] = b - a;
179  b = a;
180  }
181 }
182 
183 /**
184  * NOTE: We use the same code as Vorbis here
185  * @todo optimize it further with SSE/3Dnow
186  */
187 static void wma_lsp_to_curve(WMACodecContext *s, float *out, float *val_max_ptr,
188  int n, float *lsp)
189 {
190  int i, j;
191  float p, q, w, v, val_max;
192 
193  val_max = 0;
194  for (i = 0; i < n; i++) {
195  p = 0.5f;
196  q = 0.5f;
197  w = s->lsp_cos_table[i];
198  for (j = 1; j < NB_LSP_COEFS; j += 2) {
199  q *= w - lsp[j - 1];
200  p *= w - lsp[j];
201  }
202  p *= p * (2.0f - w);
203  q *= q * (2.0f + w);
204  v = p + q;
205  v = pow_m1_4(s, v);
206  if (v > val_max)
207  val_max = v;
208  out[i] = v;
209  }
210  *val_max_ptr = val_max;
211 }
212 
213 /**
214  * decode exponents coded with LSP coefficients (same idea as Vorbis)
215  */
216 static void decode_exp_lsp(WMACodecContext *s, int ch)
217 {
218  float lsp_coefs[NB_LSP_COEFS];
219  int val, i;
220 
221  for (i = 0; i < NB_LSP_COEFS; i++) {
222  if (i == 0 || i >= 8)
223  val = get_bits(&s->gb, 3);
224  else
225  val = get_bits(&s->gb, 4);
226  lsp_coefs[i] = ff_wma_lsp_codebook[i][val];
227  }
228 
229  wma_lsp_to_curve(s, s->exponents[ch], &s->max_exponent[ch],
230  s->block_len, lsp_coefs);
231 }
232 
233 /** pow(10, i / 16.0) for i in -60..95 */
234 static const float pow_tab[] = {
235  1.7782794100389e-04, 2.0535250264571e-04,
236  2.3713737056617e-04, 2.7384196342644e-04,
237  3.1622776601684e-04, 3.6517412725484e-04,
238  4.2169650342858e-04, 4.8696752516586e-04,
239  5.6234132519035e-04, 6.4938163157621e-04,
240  7.4989420933246e-04, 8.6596432336006e-04,
241  1.0000000000000e-03, 1.1547819846895e-03,
242  1.3335214321633e-03, 1.5399265260595e-03,
243  1.7782794100389e-03, 2.0535250264571e-03,
244  2.3713737056617e-03, 2.7384196342644e-03,
245  3.1622776601684e-03, 3.6517412725484e-03,
246  4.2169650342858e-03, 4.8696752516586e-03,
247  5.6234132519035e-03, 6.4938163157621e-03,
248  7.4989420933246e-03, 8.6596432336006e-03,
249  1.0000000000000e-02, 1.1547819846895e-02,
250  1.3335214321633e-02, 1.5399265260595e-02,
251  1.7782794100389e-02, 2.0535250264571e-02,
252  2.3713737056617e-02, 2.7384196342644e-02,
253  3.1622776601684e-02, 3.6517412725484e-02,
254  4.2169650342858e-02, 4.8696752516586e-02,
255  5.6234132519035e-02, 6.4938163157621e-02,
256  7.4989420933246e-02, 8.6596432336007e-02,
257  1.0000000000000e-01, 1.1547819846895e-01,
258  1.3335214321633e-01, 1.5399265260595e-01,
259  1.7782794100389e-01, 2.0535250264571e-01,
260  2.3713737056617e-01, 2.7384196342644e-01,
261  3.1622776601684e-01, 3.6517412725484e-01,
262  4.2169650342858e-01, 4.8696752516586e-01,
263  5.6234132519035e-01, 6.4938163157621e-01,
264  7.4989420933246e-01, 8.6596432336007e-01,
265  1.0000000000000e+00, 1.1547819846895e+00,
266  1.3335214321633e+00, 1.5399265260595e+00,
267  1.7782794100389e+00, 2.0535250264571e+00,
268  2.3713737056617e+00, 2.7384196342644e+00,
269  3.1622776601684e+00, 3.6517412725484e+00,
270  4.2169650342858e+00, 4.8696752516586e+00,
271  5.6234132519035e+00, 6.4938163157621e+00,
272  7.4989420933246e+00, 8.6596432336007e+00,
273  1.0000000000000e+01, 1.1547819846895e+01,
274  1.3335214321633e+01, 1.5399265260595e+01,
275  1.7782794100389e+01, 2.0535250264571e+01,
276  2.3713737056617e+01, 2.7384196342644e+01,
277  3.1622776601684e+01, 3.6517412725484e+01,
278  4.2169650342858e+01, 4.8696752516586e+01,
279  5.6234132519035e+01, 6.4938163157621e+01,
280  7.4989420933246e+01, 8.6596432336007e+01,
281  1.0000000000000e+02, 1.1547819846895e+02,
282  1.3335214321633e+02, 1.5399265260595e+02,
283  1.7782794100389e+02, 2.0535250264571e+02,
284  2.3713737056617e+02, 2.7384196342644e+02,
285  3.1622776601684e+02, 3.6517412725484e+02,
286  4.2169650342858e+02, 4.8696752516586e+02,
287  5.6234132519035e+02, 6.4938163157621e+02,
288  7.4989420933246e+02, 8.6596432336007e+02,
289  1.0000000000000e+03, 1.1547819846895e+03,
290  1.3335214321633e+03, 1.5399265260595e+03,
291  1.7782794100389e+03, 2.0535250264571e+03,
292  2.3713737056617e+03, 2.7384196342644e+03,
293  3.1622776601684e+03, 3.6517412725484e+03,
294  4.2169650342858e+03, 4.8696752516586e+03,
295  5.6234132519035e+03, 6.4938163157621e+03,
296  7.4989420933246e+03, 8.6596432336007e+03,
297  1.0000000000000e+04, 1.1547819846895e+04,
298  1.3335214321633e+04, 1.5399265260595e+04,
299  1.7782794100389e+04, 2.0535250264571e+04,
300  2.3713737056617e+04, 2.7384196342644e+04,
301  3.1622776601684e+04, 3.6517412725484e+04,
302  4.2169650342858e+04, 4.8696752516586e+04,
303  5.6234132519035e+04, 6.4938163157621e+04,
304  7.4989420933246e+04, 8.6596432336007e+04,
305  1.0000000000000e+05, 1.1547819846895e+05,
306  1.3335214321633e+05, 1.5399265260595e+05,
307  1.7782794100389e+05, 2.0535250264571e+05,
308  2.3713737056617e+05, 2.7384196342644e+05,
309  3.1622776601684e+05, 3.6517412725484e+05,
310  4.2169650342858e+05, 4.8696752516586e+05,
311  5.6234132519035e+05, 6.4938163157621e+05,
312  7.4989420933246e+05, 8.6596432336007e+05,
313 };
314 
315 /**
316  * decode exponents coded with VLC codes
317  */
318 static int decode_exp_vlc(WMACodecContext *s, int ch)
319 {
320  int last_exp, n, code;
321  const uint16_t *ptr;
322  float v, max_scale;
323  uint32_t *q, *q_end, iv;
324  const float *ptab = pow_tab + 60;
325  const uint32_t *iptab = (const uint32_t *) ptab;
326 
327  ptr = s->exponent_bands[s->frame_len_bits - s->block_len_bits];
328  q = (uint32_t *) s->exponents[ch];
329  q_end = q + s->block_len;
330  max_scale = 0;
331  if (s->version == 1) {
332  last_exp = get_bits(&s->gb, 5) + 10;
333  v = ptab[last_exp];
334  iv = iptab[last_exp];
335  max_scale = v;
336  n = *ptr++;
337  switch (n & 3) do {
338  case 0: *q++ = iv;
339  case 3: *q++ = iv;
340  case 2: *q++ = iv;
341  case 1: *q++ = iv;
342  } while ((n -= 4) > 0);
343  } else
344  last_exp = 36;
345 
346  while (q < q_end) {
347  code = get_vlc2(&s->gb, s->exp_vlc.table, EXPVLCBITS, EXPMAX);
348  /* NOTE: this offset is the same as MPEG-4 AAC! */
349  last_exp += code - 60;
350  if ((unsigned) last_exp + 60 >= FF_ARRAY_ELEMS(pow_tab)) {
351  av_log(s->avctx, AV_LOG_ERROR, "Exponent out of range: %d\n",
352  last_exp);
353  return -1;
354  }
355  v = ptab[last_exp];
356  iv = iptab[last_exp];
357  if (v > max_scale)
358  max_scale = v;
359  n = *ptr++;
360  switch (n & 3) do {
361  case 0: *q++ = iv;
362  case 3: *q++ = iv;
363  case 2: *q++ = iv;
364  case 1: *q++ = iv;
365  } while ((n -= 4) > 0);
366  }
367  s->max_exponent[ch] = max_scale;
368  return 0;
369 }
370 
371 /**
372  * Apply MDCT window and add into output.
373  *
374  * We ensure that when the windows overlap their squared sum
375  * is always 1 (MDCT reconstruction rule).
376  */
377 static void wma_window(WMACodecContext *s, float *out)
378 {
379  float *in = s->output;
380  int block_len, bsize, n;
381 
382  /* left part */
383  if (s->block_len_bits <= s->prev_block_len_bits) {
384  block_len = s->block_len;
385  bsize = s->frame_len_bits - s->block_len_bits;
386 
387  s->fdsp->vector_fmul_add(out, in, s->windows[bsize],
388  out, block_len);
389  } else {
390  block_len = 1 << s->prev_block_len_bits;
391  n = (s->block_len - block_len) / 2;
392  bsize = s->frame_len_bits - s->prev_block_len_bits;
393 
394  s->fdsp->vector_fmul_add(out + n, in + n, s->windows[bsize],
395  out + n, block_len);
396 
397  memcpy(out + n + block_len, in + n + block_len, n * sizeof(float));
398  }
399 
400  out += s->block_len;
401  in += s->block_len;
402 
403  /* right part */
404  if (s->block_len_bits <= s->next_block_len_bits) {
405  block_len = s->block_len;
406  bsize = s->frame_len_bits - s->block_len_bits;
407 
408  s->fdsp->vector_fmul_reverse(out, in, s->windows[bsize], block_len);
409  } else {
410  block_len = 1 << s->next_block_len_bits;
411  n = (s->block_len - block_len) / 2;
412  bsize = s->frame_len_bits - s->next_block_len_bits;
413 
414  memcpy(out, in, n * sizeof(float));
415 
416  s->fdsp->vector_fmul_reverse(out + n, in + n, s->windows[bsize],
417  block_len);
418 
419  memset(out + n + block_len, 0, n * sizeof(float));
420  }
421 }
422 
423 /**
424  * @return 0 if OK. 1 if last block of frame. return -1 if
425  * unrecoverable error.
426  */
428 {
429  int n, v, a, ch, bsize;
430  int coef_nb_bits, total_gain;
431  int nb_coefs[MAX_CHANNELS];
432  float mdct_norm;
433  FFTContext *mdct;
434 
435 #ifdef TRACE
436  ff_tlog(s->avctx, "***decode_block: %d:%d\n",
437  s->frame_count - 1, s->block_num);
438 #endif /* TRACE */
439 
440  /* compute current block length */
441  if (s->use_variable_block_len) {
442  n = av_log2(s->nb_block_sizes - 1) + 1;
443 
444  if (s->reset_block_lengths) {
445  s->reset_block_lengths = 0;
446  v = get_bits(&s->gb, n);
447  if (v >= s->nb_block_sizes) {
448  av_log(s->avctx, AV_LOG_ERROR,
449  "prev_block_len_bits %d out of range\n",
450  s->frame_len_bits - v);
451  return -1;
452  }
453  s->prev_block_len_bits = s->frame_len_bits - v;
454  v = get_bits(&s->gb, n);
455  if (v >= s->nb_block_sizes) {
456  av_log(s->avctx, AV_LOG_ERROR,
457  "block_len_bits %d out of range\n",
458  s->frame_len_bits - v);
459  return -1;
460  }
461  s->block_len_bits = s->frame_len_bits - v;
462  } else {
463  /* update block lengths */
464  s->prev_block_len_bits = s->block_len_bits;
465  s->block_len_bits = s->next_block_len_bits;
466  }
467  v = get_bits(&s->gb, n);
468  if (v >= s->nb_block_sizes) {
469  av_log(s->avctx, AV_LOG_ERROR,
470  "next_block_len_bits %d out of range\n",
471  s->frame_len_bits - v);
472  return -1;
473  }
474  s->next_block_len_bits = s->frame_len_bits - v;
475  } else {
476  /* fixed block len */
477  s->next_block_len_bits = s->frame_len_bits;
478  s->prev_block_len_bits = s->frame_len_bits;
479  s->block_len_bits = s->frame_len_bits;
480  }
481 
482  if (s->frame_len_bits - s->block_len_bits >= s->nb_block_sizes){
483  av_log(s->avctx, AV_LOG_ERROR, "block_len_bits not initialized to a valid value\n");
484  return -1;
485  }
486 
487  /* now check if the block length is coherent with the frame length */
488  s->block_len = 1 << s->block_len_bits;
489  if ((s->block_pos + s->block_len) > s->frame_len) {
490  av_log(s->avctx, AV_LOG_ERROR, "frame_len overflow\n");
491  return -1;
492  }
493 
494  if (s->avctx->channels == 2)
495  s->ms_stereo = get_bits1(&s->gb);
496  v = 0;
497  for (ch = 0; ch < s->avctx->channels; ch++) {
498  a = get_bits1(&s->gb);
499  s->channel_coded[ch] = a;
500  v |= a;
501  }
502 
503  bsize = s->frame_len_bits - s->block_len_bits;
504 
505  /* if no channel coded, no need to go further */
506  /* XXX: fix potential framing problems */
507  if (!v)
508  goto next;
509 
510  /* read total gain and extract corresponding number of bits for
511  * coef escape coding */
512  total_gain = 1;
513  for (;;) {
514  if (get_bits_left(&s->gb) < 7) {
515  av_log(s->avctx, AV_LOG_ERROR, "total_gain overread\n");
516  return AVERROR_INVALIDDATA;
517  }
518  a = get_bits(&s->gb, 7);
519  total_gain += a;
520  if (a != 127)
521  break;
522  }
523 
524  coef_nb_bits = ff_wma_total_gain_to_bits(total_gain);
525 
526  /* compute number of coefficients */
527  n = s->coefs_end[bsize] - s->coefs_start;
528  for (ch = 0; ch < s->avctx->channels; ch++)
529  nb_coefs[ch] = n;
530 
531  /* complex coding */
532  if (s->use_noise_coding) {
533  for (ch = 0; ch < s->avctx->channels; ch++) {
534  if (s->channel_coded[ch]) {
535  int i, n, a;
536  n = s->exponent_high_sizes[bsize];
537  for (i = 0; i < n; i++) {
538  a = get_bits1(&s->gb);
539  s->high_band_coded[ch][i] = a;
540  /* if noise coding, the coefficients are not transmitted */
541  if (a)
542  nb_coefs[ch] -= s->exponent_high_bands[bsize][i];
543  }
544  }
545  }
546  for (ch = 0; ch < s->avctx->channels; ch++) {
547  if (s->channel_coded[ch]) {
548  int i, n, val;
549 
550  n = s->exponent_high_sizes[bsize];
551  val = (int) 0x80000000;
552  for (i = 0; i < n; i++) {
553  if (s->high_band_coded[ch][i]) {
554  if (val == (int) 0x80000000) {
555  val = get_bits(&s->gb, 7) - 19;
556  } else {
557  val += get_vlc2(&s->gb, s->hgain_vlc.table,
559  }
560  s->high_band_values[ch][i] = val;
561  }
562  }
563  }
564  }
565  }
566 
567  /* exponents can be reused in short blocks. */
568  if ((s->block_len_bits == s->frame_len_bits) || get_bits1(&s->gb)) {
569  for (ch = 0; ch < s->avctx->channels; ch++) {
570  if (s->channel_coded[ch]) {
571  if (s->use_exp_vlc) {
572  if (decode_exp_vlc(s, ch) < 0)
573  return -1;
574  } else {
575  decode_exp_lsp(s, ch);
576  }
577  s->exponents_bsize[ch] = bsize;
578  s->exponents_initialized[ch] = 1;
579  }
580  }
581  }
582 
583  for (ch = 0; ch < s->avctx->channels; ch++) {
584  if (s->channel_coded[ch] && !s->exponents_initialized[ch])
585  return AVERROR_INVALIDDATA;
586  }
587 
588  /* parse spectral coefficients : just RLE encoding */
589  for (ch = 0; ch < s->avctx->channels; ch++) {
590  if (s->channel_coded[ch]) {
591  int tindex;
592  WMACoef *ptr = &s->coefs1[ch][0];
593  int ret;
594 
595  /* special VLC tables are used for ms stereo because
596  * there is potentially less energy there */
597  tindex = (ch == 1 && s->ms_stereo);
598  memset(ptr, 0, s->block_len * sizeof(WMACoef));
599  ret = ff_wma_run_level_decode(s->avctx, &s->gb, &s->coef_vlc[tindex],
600  s->level_table[tindex], s->run_table[tindex],
601  0, ptr, 0, nb_coefs[ch],
602  s->block_len, s->frame_len_bits, coef_nb_bits);
603  if (ret < 0)
604  return ret;
605  }
606  if (s->version == 1 && s->avctx->channels >= 2)
607  align_get_bits(&s->gb);
608  }
609 
610  /* normalize */
611  {
612  int n4 = s->block_len / 2;
613  mdct_norm = 1.0 / (float) n4;
614  if (s->version == 1)
615  mdct_norm *= sqrt(n4);
616  }
617 
618  /* finally compute the MDCT coefficients */
619  for (ch = 0; ch < s->avctx->channels; ch++) {
620  if (s->channel_coded[ch]) {
621  WMACoef *coefs1;
622  float *coefs, *exponents, mult, mult1, noise;
623  int i, j, n, n1, last_high_band, esize;
624  float exp_power[HIGH_BAND_MAX_SIZE];
625 
626  coefs1 = s->coefs1[ch];
627  exponents = s->exponents[ch];
628  esize = s->exponents_bsize[ch];
629  mult = ff_exp10(total_gain * 0.05) / s->max_exponent[ch];
630  mult *= mdct_norm;
631  coefs = s->coefs[ch];
632  if (s->use_noise_coding) {
633  mult1 = mult;
634  /* very low freqs : noise */
635  for (i = 0; i < s->coefs_start; i++) {
636  *coefs++ = s->noise_table[s->noise_index] *
637  exponents[i << bsize >> esize] * mult1;
638  s->noise_index = (s->noise_index + 1) &
639  (NOISE_TAB_SIZE - 1);
640  }
641 
642  n1 = s->exponent_high_sizes[bsize];
643 
644  /* compute power of high bands */
645  exponents = s->exponents[ch] +
646  (s->high_band_start[bsize] << bsize >> esize);
647  last_high_band = 0; /* avoid warning */
648  for (j = 0; j < n1; j++) {
649  n = s->exponent_high_bands[s->frame_len_bits -
650  s->block_len_bits][j];
651  if (s->high_band_coded[ch][j]) {
652  float e2, v;
653  e2 = 0;
654  for (i = 0; i < n; i++) {
655  v = exponents[i << bsize >> esize];
656  e2 += v * v;
657  }
658  exp_power[j] = e2 / n;
659  last_high_band = j;
660  ff_tlog(s->avctx, "%d: power=%f (%d)\n", j, exp_power[j], n);
661  }
662  exponents += n << bsize >> esize;
663  }
664 
665  /* main freqs and high freqs */
666  exponents = s->exponents[ch] + (s->coefs_start << bsize >> esize);
667  for (j = -1; j < n1; j++) {
668  if (j < 0)
669  n = s->high_band_start[bsize] - s->coefs_start;
670  else
671  n = s->exponent_high_bands[s->frame_len_bits -
672  s->block_len_bits][j];
673  if (j >= 0 && s->high_band_coded[ch][j]) {
674  /* use noise with specified power */
675  mult1 = sqrt(exp_power[j] / exp_power[last_high_band]);
676  /* XXX: use a table */
677  mult1 = mult1 * ff_exp10(s->high_band_values[ch][j] * 0.05);
678  mult1 = mult1 / (s->max_exponent[ch] * s->noise_mult);
679  mult1 *= mdct_norm;
680  for (i = 0; i < n; i++) {
681  noise = s->noise_table[s->noise_index];
682  s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
683  *coefs++ = noise * exponents[i << bsize >> esize] * mult1;
684  }
685  exponents += n << bsize >> esize;
686  } else {
687  /* coded values + small noise */
688  for (i = 0; i < n; i++) {
689  noise = s->noise_table[s->noise_index];
690  s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
691  *coefs++ = ((*coefs1++) + noise) *
692  exponents[i << bsize >> esize] * mult;
693  }
694  exponents += n << bsize >> esize;
695  }
696  }
697 
698  /* very high freqs : noise */
699  n = s->block_len - s->coefs_end[bsize];
700  mult1 = mult * exponents[(-(1 << bsize)) >> esize];
701  for (i = 0; i < n; i++) {
702  *coefs++ = s->noise_table[s->noise_index] * mult1;
703  s->noise_index = (s->noise_index + 1) & (NOISE_TAB_SIZE - 1);
704  }
705  } else {
706  /* XXX: optimize more */
707  for (i = 0; i < s->coefs_start; i++)
708  *coefs++ = 0.0;
709  n = nb_coefs[ch];
710  for (i = 0; i < n; i++)
711  *coefs++ = coefs1[i] * exponents[i << bsize >> esize] * mult;
712  n = s->block_len - s->coefs_end[bsize];
713  for (i = 0; i < n; i++)
714  *coefs++ = 0.0;
715  }
716  }
717  }
718 
719 #ifdef TRACE
720  for (ch = 0; ch < s->avctx->channels; ch++) {
721  if (s->channel_coded[ch]) {
722  dump_floats(s, "exponents", 3, s->exponents[ch], s->block_len);
723  dump_floats(s, "coefs", 1, s->coefs[ch], s->block_len);
724  }
725  }
726 #endif /* TRACE */
727 
728  if (s->ms_stereo && s->channel_coded[1]) {
729  /* nominal case for ms stereo: we do it before mdct */
730  /* no need to optimize this case because it should almost
731  * never happen */
732  if (!s->channel_coded[0]) {
733  ff_tlog(s->avctx, "rare ms-stereo case happened\n");
734  memset(s->coefs[0], 0, sizeof(float) * s->block_len);
735  s->channel_coded[0] = 1;
736  }
737 
738  s->fdsp->butterflies_float(s->coefs[0], s->coefs[1], s->block_len);
739  }
740 
741 next:
742  mdct = &s->mdct_ctx[bsize];
743 
744  for (ch = 0; ch < s->avctx->channels; ch++) {
745  int n4, index;
746 
747  n4 = s->block_len / 2;
748  if (s->channel_coded[ch])
749  mdct->imdct_calc(mdct, s->output, s->coefs[ch]);
750  else if (!(s->ms_stereo && ch == 1))
751  memset(s->output, 0, sizeof(s->output));
752 
753  /* multiply by the window and add in the frame */
754  index = (s->frame_len / 2) + s->block_pos - n4;
755  wma_window(s, &s->frame_out[ch][index]);
756  }
757 
758  /* update block number */
759  s->block_num++;
760  s->block_pos += s->block_len;
761  if (s->block_pos >= s->frame_len)
762  return 1;
763  else
764  return 0;
765 }
766 
767 /* decode a frame of frame_len samples */
769  int samples_offset)
770 {
771  int ret, ch;
772 
773 #ifdef TRACE
774  ff_tlog(s->avctx, "***decode_frame: %d size=%d\n",
775  s->frame_count++, s->frame_len);
776 #endif /* TRACE */
777 
778  /* read each block */
779  s->block_num = 0;
780  s->block_pos = 0;
781  for (;;) {
783  if (ret < 0)
784  return -1;
785  if (ret)
786  break;
787  }
788 
789  for (ch = 0; ch < s->avctx->channels; ch++) {
790  /* copy current block to output */
791  memcpy(samples[ch] + samples_offset, s->frame_out[ch],
792  s->frame_len * sizeof(*s->frame_out[ch]));
793  /* prepare for next block */
794  memmove(&s->frame_out[ch][0], &s->frame_out[ch][s->frame_len],
795  s->frame_len * sizeof(*s->frame_out[ch]));
796 
797 #ifdef TRACE
798  dump_floats(s, "samples", 6, samples[ch] + samples_offset,
799  s->frame_len);
800 #endif /* TRACE */
801  }
802 
803  return 0;
804 }
805 
806 static int wma_decode_superframe(AVCodecContext *avctx, void *data,
807  int *got_frame_ptr, AVPacket *avpkt)
808 {
809  AVFrame *frame = data;
810  const uint8_t *buf = avpkt->data;
811  int buf_size = avpkt->size;
812  WMACodecContext *s = avctx->priv_data;
813  int nb_frames, bit_offset, i, pos, len, ret;
814  uint8_t *q;
815  float **samples;
816  int samples_offset;
817 
818  ff_tlog(avctx, "***decode_superframe:\n");
819 
820  if (buf_size == 0) {
821  s->last_superframe_len = 0;
822  return 0;
823  }
824  if (buf_size < avctx->block_align) {
825  av_log(avctx, AV_LOG_ERROR,
826  "Input packet size too small (%d < %d)\n",
827  buf_size, avctx->block_align);
828  return AVERROR_INVALIDDATA;
829  }
830  if (avctx->block_align)
831  buf_size = avctx->block_align;
832 
833  init_get_bits(&s->gb, buf, buf_size * 8);
834 
835  if (s->use_bit_reservoir) {
836  /* read super frame header */
837  skip_bits(&s->gb, 4); /* super frame index */
838  nb_frames = get_bits(&s->gb, 4) - (s->last_superframe_len <= 0);
839  if (nb_frames <= 0) {
840  int is_error = nb_frames < 0 || get_bits_left(&s->gb) <= 8;
841  av_log(avctx, is_error ? AV_LOG_ERROR : AV_LOG_WARNING,
842  "nb_frames is %d bits left %d\n",
843  nb_frames, get_bits_left(&s->gb));
844  if (is_error)
845  return AVERROR_INVALIDDATA;
846 
847  if ((s->last_superframe_len + buf_size - 1) >
849  goto fail;
850 
851  q = s->last_superframe + s->last_superframe_len;
852  len = buf_size - 1;
853  while (len > 0) {
854  *q++ = get_bits (&s->gb, 8);
855  len --;
856  }
857  memset(q, 0, AV_INPUT_BUFFER_PADDING_SIZE);
858 
859  s->last_superframe_len += 8*buf_size - 8;
860 // s->reset_block_lengths = 1; //XXX is this needed ?
861  *got_frame_ptr = 0;
862  return buf_size;
863  }
864  } else
865  nb_frames = 1;
866 
867  /* get output buffer */
868  frame->nb_samples = nb_frames * s->frame_len;
869  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
870  return ret;
871  samples = (float **) frame->extended_data;
872  samples_offset = 0;
873 
874  if (s->use_bit_reservoir) {
875  bit_offset = get_bits(&s->gb, s->byte_offset_bits + 3);
876  if (bit_offset > get_bits_left(&s->gb)) {
877  av_log(avctx, AV_LOG_ERROR,
878  "Invalid last frame bit offset %d > buf size %d (%d)\n",
879  bit_offset, get_bits_left(&s->gb), buf_size);
880  goto fail;
881  }
882 
883  if (s->last_superframe_len > 0) {
884  /* add bit_offset bits to last frame */
885  if ((s->last_superframe_len + ((bit_offset + 7) >> 3)) >
887  goto fail;
888  q = s->last_superframe + s->last_superframe_len;
889  len = bit_offset;
890  while (len > 7) {
891  *q++ = get_bits(&s->gb, 8);
892  len -= 8;
893  }
894  if (len > 0)
895  *q++ = get_bits(&s->gb, len) << (8 - len);
896  memset(q, 0, AV_INPUT_BUFFER_PADDING_SIZE);
897 
898  /* XXX: bit_offset bits into last frame */
899  init_get_bits(&s->gb, s->last_superframe,
900  s->last_superframe_len * 8 + bit_offset);
901  /* skip unused bits */
902  if (s->last_bitoffset > 0)
903  skip_bits(&s->gb, s->last_bitoffset);
904  /* this frame is stored in the last superframe and in the
905  * current one */
906  if (wma_decode_frame(s, samples, samples_offset) < 0)
907  goto fail;
908  samples_offset += s->frame_len;
909  nb_frames--;
910  }
911 
912  /* read each frame starting from bit_offset */
913  pos = bit_offset + 4 + 4 + s->byte_offset_bits + 3;
914  if (pos >= MAX_CODED_SUPERFRAME_SIZE * 8 || pos > buf_size * 8)
915  return AVERROR_INVALIDDATA;
916  init_get_bits(&s->gb, buf + (pos >> 3), (buf_size - (pos >> 3)) * 8);
917  len = pos & 7;
918  if (len > 0)
919  skip_bits(&s->gb, len);
920 
921  s->reset_block_lengths = 1;
922  for (i = 0; i < nb_frames; i++) {
923  if (wma_decode_frame(s, samples, samples_offset) < 0)
924  goto fail;
925  samples_offset += s->frame_len;
926  }
927 
928  /* we copy the end of the frame in the last frame buffer */
929  pos = get_bits_count(&s->gb) +
930  ((bit_offset + 4 + 4 + s->byte_offset_bits + 3) & ~7);
931  s->last_bitoffset = pos & 7;
932  pos >>= 3;
933  len = buf_size - pos;
934  if (len > MAX_CODED_SUPERFRAME_SIZE || len < 0) {
935  av_log(s->avctx, AV_LOG_ERROR, "len %d invalid\n", len);
936  goto fail;
937  }
938  s->last_superframe_len = len;
939  memcpy(s->last_superframe, buf + pos, len);
940  } else {
941  /* single frame decode */
942  if (wma_decode_frame(s, samples, samples_offset) < 0)
943  goto fail;
944  samples_offset += s->frame_len;
945  }
946 
947  ff_dlog(s->avctx, "%d %d %d %d outbytes:%"PTRDIFF_SPECIFIER" eaten:%d\n",
948  s->frame_len_bits, s->block_len_bits, s->frame_len, s->block_len,
949  (int8_t *) samples - (int8_t *) data, avctx->block_align);
950 
951  *got_frame_ptr = 1;
952 
953  return buf_size;
954 
955 fail:
956  /* when error, we reset the bit reservoir */
957  s->last_superframe_len = 0;
958  return -1;
959 }
960 
961 static av_cold void flush(AVCodecContext *avctx)
962 {
963  WMACodecContext *s = avctx->priv_data;
964 
965  s->last_bitoffset =
966  s->last_superframe_len = 0;
967 }
968 
969 #if CONFIG_WMAV1_DECODER
971  .name = "wmav1",
972  .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 1"),
973  .type = AVMEDIA_TYPE_AUDIO,
974  .id = AV_CODEC_ID_WMAV1,
975  .priv_data_size = sizeof(WMACodecContext),
977  .close = ff_wma_end,
979  .flush = flush,
980  .capabilities = AV_CODEC_CAP_DR1,
981  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
983  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
984 };
985 #endif
986 #if CONFIG_WMAV2_DECODER
988  .name = "wmav2",
989  .long_name = NULL_IF_CONFIG_SMALL("Windows Media Audio 2"),
990  .type = AVMEDIA_TYPE_AUDIO,
991  .id = AV_CODEC_ID_WMAV2,
992  .priv_data_size = sizeof(WMACodecContext),
994  .close = ff_wma_end,
996  .flush = flush,
997  .capabilities = AV_CODEC_CAP_DR1,
998  .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1000  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
1001 };
1002 #endif
AVCodec
AVCodec.
Definition: codec.h:197
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:200
name
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf default minimum maximum flags name is the option name
Definition: writing_filters.txt:88
ff_exp10
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
Definition: ffmath.h:42
init
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
get_bits_left
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
out
FILE * out
Definition: movenc.c:54
u
#define u(width, name, range_min, range_max)
Definition: cbs_h2645.c:264
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
flush
static av_cold void flush(AVCodecContext *avctx)
Definition: wmadec.c:961
get_bits_count
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
w
uint8_t w
Definition: llviddspenc.c:39
internal.h
AVPacket::data
uint8_t * data
Definition: packet.h:369
wma_lsp_to_curve_init
static void wma_lsp_to_curve_init(WMACodecContext *s, int frame_len)
Definition: wmadec.c:155
init_vlc
#define init_vlc(vlc, nb_bits, nb_codes, bits, bits_wrap, bits_size, codes, codes_wrap, codes_size, flags)
Definition: vlc.h:38
b
#define b
Definition: input.c:41
data
const char data[16]
Definition: mxf.c:142
ff_mdct_init
#define ff_mdct_init
Definition: fft.h:161
get_vlc2
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:797
AV_CODEC_ID_WMAV2
@ AV_CODEC_ID_WMAV2
Definition: codec_id.h:432
ff_tlog
#define ff_tlog(ctx,...)
Definition: internal.h:96
wma_decode_frame
static int wma_decode_frame(WMACodecContext *s, float **samples, int samples_offset)
Definition: wmadec.c:768
init_get_bits
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
ff_wma_hgain_hufftab
const uint8_t ff_wma_hgain_hufftab[37][2]
Definition: wmadata.h:54
skip_bits
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
get_bits
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
HIGH_BAND_MAX_SIZE
#define HIGH_BAND_MAX_SIZE
Definition: wma.h:41
decode_exp_lsp
static void decode_exp_lsp(WMACodecContext *s, int ch)
decode exponents coded with LSP coefficients (same idea as Vorbis)
Definition: wmadec.c:216
AVCodecContext::codec
const struct AVCodec * codec
Definition: avcodec.h:545
WMACoef
float WMACoef
type for decoded coefficients, int16_t would be enough for wma 1/2
Definition: wma.h:58
fail
#define fail()
Definition: checkasm.h:133
tab
static const struct twinvq_data tab
Definition: twinvq_data.h:10345
val
static double val(void *priv, double ch)
Definition: aeval.c:76
WMACodecContext
Definition: wma.h:68
ff_init_vlc_from_lengths
int ff_init_vlc_from_lengths(VLC *vlc_arg, int nb_bits, int nb_codes, const int8_t *lens, int lens_wrap, const void *symbols, int symbols_wrap, int symbols_size, int offset, int flags, void *logctx)
Build VLC decoding tables suitable for use with get_vlc2()
Definition: bitstream.c:381
mult
static int16_t mult(Float11 *f1, Float11 *f2)
Definition: g726.c:55
AV_CODEC_ID_WMAV1
@ AV_CODEC_ID_WMAV1
Definition: codec_id.h:431
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
FF_ARRAY_ELEMS
#define FF_ARRAY_ELEMS(a)
Definition: sinewin_tablegen.c:29
av_cold
#define av_cold
Definition: attributes.h:90
decode
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
AVCodecContext::extradata_size
int extradata_size
Definition: avcodec.h:638
EXPVLCBITS
#define EXPVLCBITS
Definition: wmadec.c:43
s
#define s(width, name)
Definition: cbs_vp9.c:257
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
exp2f
#define exp2f(x)
Definition: libm.h:293
AV_RL16
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_RL16
Definition: bytestream.h:94
wma.h
ff_wma_total_gain_to_bits
int ff_wma_total_gain_to_bits(int total_gain)
Definition: wma.c:350
f
#define f(width, name)
Definition: cbs_vp9.c:255
if
if(ret)
Definition: filter_design.txt:179
PTRDIFF_SPECIFIER
#define PTRDIFF_SPECIFIER
Definition: internal.h:192
HGAINMAX
#define HGAINMAX
Definition: wmadec.c:47
get_bits1
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
index
int index
Definition: gxfenc.c:89
MAX_CODED_SUPERFRAME_SIZE
#define MAX_CODED_SUPERFRAME_SIZE
Definition: wma.h:46
ff_wma_end
int ff_wma_end(AVCodecContext *avctx)
Definition: wma.c:364
ff_dlog
#define ff_dlog(a,...)
Definition: tableprint_vlc.h:29
FFTContext::imdct_calc
void(* imdct_calc)(struct FFTContext *s, FFTSample *output, const FFTSample *input)
Definition: fft.h:102
ff_aac_scalefactor_bits
const uint8_t ff_aac_scalefactor_bits[121]
Definition: aactab.c:111
ff_get_buffer
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1900
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
AVPacket::size
int size
Definition: packet.h:370
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
wma_lsp_to_curve
static void wma_lsp_to_curve(WMACodecContext *s, float *out, float *val_max_ptr, int n, float *lsp)
NOTE: We use the same code as Vorbis here.
Definition: wmadec.c:187
AVCodecContext::sample_fmt
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
ff_wma_init
av_cold int ff_wma_init(AVCodecContext *avctx, int flags2)
Definition: wma.c:77
NOISE_TAB_SIZE
#define NOISE_TAB_SIZE
Definition: wma.h:50
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
attributes.h
MAX_CHANNELS
#define MAX_CHANNELS
Definition: aac.h:48
wma_decode_block
static int wma_decode_block(WMACodecContext *s)
Definition: wmadec.c:427
M_PI
#define M_PI
Definition: mathematics.h:52
pow_tab
static const float pow_tab[]
pow(10, i / 16.0) for i in -60..95
Definition: wmadec.c:234
AVCodec::id
enum AVCodecID id
Definition: codec.h:211
in
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Definition: audio_convert.c:326
LSP_POW_BITS
#define LSP_POW_BITS
Definition: wma.h:52
FFTContext
Definition: fft.h:83
i
int i
Definition: input.c:407
code
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some it can consider them to be part of the FIFO and delay acknowledging a status change accordingly Example code
Definition: filter_design.txt:178
AVCodecContext::extradata
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:637
FF_CODEC_CAP_INIT_CLEANUP
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: internal.h:49
ff_wmav1_decoder
AVCodec ff_wmav1_decoder
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
uint8_t
uint8_t
Definition: audio_convert.c:194
ff_wmav2_decoder
AVCodec ff_wmav2_decoder
wma_window
static void wma_window(WMACodecContext *s, float *out)
Apply MDCT window and add into output.
Definition: wmadec.c:377
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:204
len
int len
Definition: vorbis_enc_data.h:452
avcodec.h
ret
ret
Definition: filter_design.txt:187
AVCodecContext::block_align
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs.
Definition: avcodec.h:1233
frame
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame For filters that do not use the this method is called when a frame is wanted on an output For a it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return or at least make progress towards producing a frame
Definition: filter_design.txt:264
HGAINVLCBITS
#define HGAINVLCBITS
Definition: wmadec.c:46
align_get_bits
static const uint8_t * align_get_bits(GetBitContext *s)
Definition: get_bits.h:693
pos
unsigned int pos
Definition: spdifenc.c:412
EXPMAX
#define EXPMAX
Definition: wmadec.c:44
AV_INPUT_BUFFER_PADDING_SIZE
#define AV_INPUT_BUFFER_PADDING_SIZE
Definition: avcodec.h:215
AVCodecContext
main external API structure.
Definition: avcodec.h:536
noise
static int noise(AVBSFContext *ctx, AVPacket *pkt)
Definition: noise_bsf.c:36
decode_exp_vlc
static int decode_exp_vlc(WMACodecContext *s, int ch)
decode exponents coded with VLC codes
Definition: wmadec.c:318
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
ffmath.h
ff_wma_lsp_codebook
const float ff_wma_lsp_codebook[NB_LSP_COEFS][16]
Definition: wmadata.h:64
ff_wma_run_level_decode
int ff_wma_run_level_decode(AVCodecContext *avctx, GetBitContext *gb, VLC *vlc, const float *level_table, const uint16_t *run_table, int version, WMACoef *ptr, int offset, int num_coefs, int block_len, int frame_len_bits, int coef_nb_bits)
Decode run level compressed coefficients.
Definition: wma.c:424
AVPacket
This structure stores compressed data.
Definition: packet.h:346
AVCodecContext::priv_data
void * priv_data
Definition: avcodec.h:563
NB_LSP_COEFS
#define NB_LSP_COEFS
Definition: wma.h:43
wma_decode_init
static av_cold int wma_decode_init(AVCodecContext *avctx)
Definition: wmadec.c:70
pow_m1_4
static float pow_m1_4(WMACodecContext *s, float x)
compute x^-0.25 with an exponent and mantissa table.
Definition: wmadec.c:136
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
AVERROR_INVALIDDATA
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
int
int
Definition: ffmpeg_filter.c:170
av_log2
int av_log2(unsigned v)
Definition: intmath.c:26
wma_decode_superframe
static int wma_decode_superframe(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: wmadec.c:806
ff_aac_scalefactor_code
const uint32_t ff_aac_scalefactor_code[121]
Definition: aactab.c:92