FFmpeg  4.3
wavdec.c
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1 /*
2  * WAV demuxer
3  * Copyright (c) 2001, 2002 Fabrice Bellard
4  *
5  * Sony Wave64 demuxer
6  * RF64 demuxer
7  * Copyright (c) 2009 Daniel Verkamp
8  *
9  * This file is part of FFmpeg.
10  *
11  * FFmpeg is free software; you can redistribute it and/or
12  * modify it under the terms of the GNU Lesser General Public
13  * License as published by the Free Software Foundation; either
14  * version 2.1 of the License, or (at your option) any later version.
15  *
16  * FFmpeg is distributed in the hope that it will be useful,
17  * but WITHOUT ANY WARRANTY; without even the implied warranty of
18  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
19  * Lesser General Public License for more details.
20  *
21  * You should have received a copy of the GNU Lesser General Public
22  * License along with FFmpeg; if not, write to the Free Software
23  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24  */
25 
26 #include <stdint.h>
27 
28 #include "libavutil/avassert.h"
29 #include "libavutil/dict.h"
30 #include "libavutil/intreadwrite.h"
31 #include "libavutil/log.h"
32 #include "libavutil/mathematics.h"
33 #include "libavutil/opt.h"
34 #include "avformat.h"
35 #include "avio.h"
36 #include "avio_internal.h"
37 #include "id3v2.h"
38 #include "internal.h"
39 #include "metadata.h"
40 #include "pcm.h"
41 #include "riff.h"
42 #include "w64.h"
43 #include "spdif.h"
44 
45 typedef struct WAVDemuxContext {
46  const AVClass *class;
47  int64_t data_end;
48  int w64;
49  int64_t smv_data_ofs;
52  int smv_block;
54  int smv_eof;
55  int audio_eof;
57  int spdif;
60  int unaligned; // e.g. if an odd number of bytes ID3 tag was prepended
61  int rifx; // RIFX: integer byte order for parameters is big endian
63 
65 {
66  if (CONFIG_SPDIF_DEMUXER && s->streams[0]->codecpar->codec_tag == 1) {
67  enum AVCodecID codec;
68  int len = 1<<16;
69  int ret = ffio_ensure_seekback(s->pb, len);
70 
71  if (ret >= 0) {
72  uint8_t *buf = av_malloc(len);
73  if (!buf) {
74  ret = AVERROR(ENOMEM);
75  } else {
76  int64_t pos = avio_tell(s->pb);
77  len = ret = avio_read(s->pb, buf, len);
78  if (len >= 0) {
79  ret = ff_spdif_probe(buf, len, &codec);
81  s->streams[0]->codecpar->codec_id = codec;
82  wav->spdif = 1;
83  }
84  }
85  avio_seek(s->pb, pos, SEEK_SET);
86  av_free(buf);
87  }
88  }
89 
90  if (ret < 0)
91  av_log(s, AV_LOG_WARNING, "Cannot check for SPDIF\n");
92  }
93 }
94 
95 #if CONFIG_WAV_DEMUXER
96 
97 static int64_t next_tag(AVIOContext *pb, uint32_t *tag, int big_endian)
98 {
99  *tag = avio_rl32(pb);
100  if (!big_endian) {
101  return avio_rl32(pb);
102  } else {
103  return avio_rb32(pb);
104  }
105 }
106 
107 /* RIFF chunks are always at even offsets relative to where they start. */
108 static int64_t wav_seek_tag(WAVDemuxContext * wav, AVIOContext *s, int64_t offset, int whence)
109 {
110  offset += offset < INT64_MAX && offset + wav->unaligned & 1;
111 
112  return avio_seek(s, offset, whence);
113 }
114 
115 /* return the size of the found tag */
116 static int64_t find_tag(WAVDemuxContext * wav, AVIOContext *pb, uint32_t tag1)
117 {
118  unsigned int tag;
119  int64_t size;
120 
121  for (;;) {
122  if (avio_feof(pb))
123  return AVERROR_EOF;
124  size = next_tag(pb, &tag, wav->rifx);
125  if (tag == tag1)
126  break;
127  wav_seek_tag(wav, pb, size, SEEK_CUR);
128  }
129  return size;
130 }
131 
132 static int wav_probe(const AVProbeData *p)
133 {
134  /* check file header */
135  if (p->buf_size <= 32)
136  return 0;
137  if (!memcmp(p->buf + 8, "WAVE", 4)) {
138  if (!memcmp(p->buf, "RIFF", 4) || !memcmp(p->buf, "RIFX", 4))
139  /* Since the ACT demuxer has a standard WAV header at the top of
140  * its own, the returned score is decreased to avoid a probe
141  * conflict between ACT and WAV. */
142  return AVPROBE_SCORE_MAX - 1;
143  else if (!memcmp(p->buf, "RF64", 4) &&
144  !memcmp(p->buf + 12, "ds64", 4))
145  return AVPROBE_SCORE_MAX;
146  }
147  return 0;
148 }
149 
150 static void handle_stream_probing(AVStream *st)
151 {
154  st->probe_packets = FFMIN(st->probe_packets, 32);
155  }
156 }
157 
158 static int wav_parse_fmt_tag(AVFormatContext *s, int64_t size, AVStream **st)
159 {
160  AVIOContext *pb = s->pb;
161  WAVDemuxContext *wav = s->priv_data;
162  int ret;
163 
164  /* parse fmt header */
165  *st = avformat_new_stream(s, NULL);
166  if (!*st)
167  return AVERROR(ENOMEM);
168 
169  ret = ff_get_wav_header(s, pb, (*st)->codecpar, size, wav->rifx);
170  if (ret < 0)
171  return ret;
172  handle_stream_probing(*st);
173 
174  (*st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
175 
176  avpriv_set_pts_info(*st, 64, 1, (*st)->codecpar->sample_rate);
177 
178  return 0;
179 }
180 
181 static int wav_parse_xma2_tag(AVFormatContext *s, int64_t size, AVStream **st)
182 {
183  AVIOContext *pb = s->pb;
184  int version, num_streams, i, channels = 0, ret;
185 
186  if (size < 36)
187  return AVERROR_INVALIDDATA;
188 
189  *st = avformat_new_stream(s, NULL);
190  if (!*st)
191  return AVERROR(ENOMEM);
192 
193  (*st)->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
194  (*st)->codecpar->codec_id = AV_CODEC_ID_XMA2;
195  (*st)->need_parsing = AVSTREAM_PARSE_FULL_RAW;
196 
197  version = avio_r8(pb);
198  if (version != 3 && version != 4)
199  return AVERROR_INVALIDDATA;
200  num_streams = avio_r8(pb);
201  if (size != (32 + ((version==3)?0:8) + 4*num_streams))
202  return AVERROR_INVALIDDATA;
203  avio_skip(pb, 10);
204  (*st)->codecpar->sample_rate = avio_rb32(pb);
205  if (version == 4)
206  avio_skip(pb, 8);
207  avio_skip(pb, 4);
208  (*st)->duration = avio_rb32(pb);
209  avio_skip(pb, 8);
210 
211  for (i = 0; i < num_streams; i++) {
212  channels += avio_r8(pb);
213  avio_skip(pb, 3);
214  }
215  (*st)->codecpar->channels = channels;
216 
217  if ((*st)->codecpar->channels <= 0 || (*st)->codecpar->sample_rate <= 0)
218  return AVERROR_INVALIDDATA;
219 
220  avpriv_set_pts_info(*st, 64, 1, (*st)->codecpar->sample_rate);
221 
222  avio_seek(pb, -size, SEEK_CUR);
223  if ((ret = ff_get_extradata(s, (*st)->codecpar, pb, size)) < 0)
224  return ret;
225 
226  return 0;
227 }
228 
229 static inline int wav_parse_bext_string(AVFormatContext *s, const char *key,
230  int length)
231 {
232  char temp[257];
233  int ret;
234 
235  av_assert0(length < sizeof(temp));
236  if ((ret = avio_read(s->pb, temp, length)) != length)
237  return ret < 0 ? ret : AVERROR_INVALIDDATA;
238 
239  temp[length] = 0;
240 
241  if (strlen(temp))
242  return av_dict_set(&s->metadata, key, temp, 0);
243 
244  return 0;
245 }
246 
247 static int wav_parse_bext_tag(AVFormatContext *s, int64_t size)
248 {
249  char temp[131], *coding_history;
250  int ret, x;
251  uint64_t time_reference;
252  int64_t umid_parts[8], umid_mask = 0;
253 
254  if ((ret = wav_parse_bext_string(s, "description", 256)) < 0 ||
255  (ret = wav_parse_bext_string(s, "originator", 32)) < 0 ||
256  (ret = wav_parse_bext_string(s, "originator_reference", 32)) < 0 ||
257  (ret = wav_parse_bext_string(s, "origination_date", 10)) < 0 ||
258  (ret = wav_parse_bext_string(s, "origination_time", 8)) < 0)
259  return ret;
260 
261  time_reference = avio_rl64(s->pb);
262  snprintf(temp, sizeof(temp), "%"PRIu64, time_reference);
263  if ((ret = av_dict_set(&s->metadata, "time_reference", temp, 0)) < 0)
264  return ret;
265 
266  /* check if version is >= 1, in which case an UMID may be present */
267  if (avio_rl16(s->pb) >= 1) {
268  for (x = 0; x < 8; x++)
269  umid_mask |= umid_parts[x] = avio_rb64(s->pb);
270 
271  if (umid_mask) {
272  /* the string formatting below is per SMPTE 330M-2004 Annex C */
273  if (umid_parts[4] == 0 && umid_parts[5] == 0 &&
274  umid_parts[6] == 0 && umid_parts[7] == 0) {
275  /* basic UMID */
276  snprintf(temp, sizeof(temp),
277  "0x%016"PRIX64"%016"PRIX64"%016"PRIX64"%016"PRIX64,
278  umid_parts[0], umid_parts[1],
279  umid_parts[2], umid_parts[3]);
280  } else {
281  /* extended UMID */
282  snprintf(temp, sizeof(temp),
283  "0x%016"PRIX64"%016"PRIX64"%016"PRIX64"%016"PRIX64
284  "%016"PRIX64"%016"PRIX64"%016"PRIX64"%016"PRIX64,
285  umid_parts[0], umid_parts[1],
286  umid_parts[2], umid_parts[3],
287  umid_parts[4], umid_parts[5],
288  umid_parts[6], umid_parts[7]);
289  }
290 
291  if ((ret = av_dict_set(&s->metadata, "umid", temp, 0)) < 0)
292  return ret;
293  }
294 
295  avio_skip(s->pb, 190);
296  } else
297  avio_skip(s->pb, 254);
298 
299  if (size > 602) {
300  /* CodingHistory present */
301  size -= 602;
302 
303  if (!(coding_history = av_malloc(size + 1)))
304  return AVERROR(ENOMEM);
305 
306  if ((ret = avio_read(s->pb, coding_history, size)) != size) {
307  av_free(coding_history);
308  return ret < 0 ? ret : AVERROR_INVALIDDATA;
309  }
310 
311  coding_history[size] = 0;
312  if ((ret = av_dict_set(&s->metadata, "coding_history", coding_history,
314  return ret;
315  }
316 
317  return 0;
318 }
319 
320 static const AVMetadataConv wav_metadata_conv[] = {
321  { "description", "comment" },
322  { "originator", "encoded_by" },
323  { "origination_date", "date" },
324  { "origination_time", "creation_time" },
325  { 0 },
326 };
327 
328 /* wav input */
329 static int wav_read_header(AVFormatContext *s)
330 {
331  int64_t size, av_uninit(data_size);
332  int64_t sample_count = 0;
333  int rf64 = 0;
334  uint32_t tag;
335  AVIOContext *pb = s->pb;
336  AVStream *st = NULL;
337  WAVDemuxContext *wav = s->priv_data;
338  int ret, got_fmt = 0, got_xma2 = 0;
339  int64_t next_tag_ofs, data_ofs = -1;
340 
341  wav->unaligned = avio_tell(s->pb) & 1;
342 
343  wav->smv_data_ofs = -1;
344 
345  /* read chunk ID */
346  tag = avio_rl32(pb);
347  switch (tag) {
348  case MKTAG('R', 'I', 'F', 'F'):
349  break;
350  case MKTAG('R', 'I', 'F', 'X'):
351  wav->rifx = 1;
352  break;
353  case MKTAG('R', 'F', '6', '4'):
354  rf64 = 1;
355  break;
356  default:
357  av_log(s, AV_LOG_ERROR, "invalid start code %s in RIFF header\n",
358  av_fourcc2str(tag));
359  return AVERROR_INVALIDDATA;
360  }
361 
362  /* read chunk size */
363  avio_rl32(pb);
364 
365  /* read format */
366  if (avio_rl32(pb) != MKTAG('W', 'A', 'V', 'E')) {
367  av_log(s, AV_LOG_ERROR, "invalid format in RIFF header\n");
368  return AVERROR_INVALIDDATA;
369  }
370 
371  if (rf64) {
372  if (avio_rl32(pb) != MKTAG('d', 's', '6', '4'))
373  return AVERROR_INVALIDDATA;
374  size = avio_rl32(pb);
375  if (size < 24)
376  return AVERROR_INVALIDDATA;
377  avio_rl64(pb); /* RIFF size */
378 
379  data_size = avio_rl64(pb);
380  sample_count = avio_rl64(pb);
381 
382  if (data_size < 0 || sample_count < 0) {
383  av_log(s, AV_LOG_ERROR, "negative data_size and/or sample_count in "
384  "ds64: data_size = %"PRId64", sample_count = %"PRId64"\n",
385  data_size, sample_count);
386  return AVERROR_INVALIDDATA;
387  }
388  avio_skip(pb, size - 24); /* skip rest of ds64 chunk */
389 
390  }
391 
392  for (;;) {
393  AVStream *vst;
394  size = next_tag(pb, &tag, wav->rifx);
395  next_tag_ofs = avio_tell(pb) + size;
396 
397  if (avio_feof(pb))
398  break;
399 
400  switch (tag) {
401  case MKTAG('f', 'm', 't', ' '):
402  /* only parse the first 'fmt ' tag found */
403  if (!got_xma2 && !got_fmt && (ret = wav_parse_fmt_tag(s, size, &st)) < 0) {
404  return ret;
405  } else if (got_fmt)
406  av_log(s, AV_LOG_WARNING, "found more than one 'fmt ' tag\n");
407 
408  got_fmt = 1;
409  break;
410  case MKTAG('X', 'M', 'A', '2'):
411  /* only parse the first 'XMA2' tag found */
412  if (!got_fmt && !got_xma2 && (ret = wav_parse_xma2_tag(s, size, &st)) < 0) {
413  return ret;
414  } else if (got_xma2)
415  av_log(s, AV_LOG_WARNING, "found more than one 'XMA2' tag\n");
416 
417  got_xma2 = 1;
418  break;
419  case MKTAG('d', 'a', 't', 'a'):
420  if (!(pb->seekable & AVIO_SEEKABLE_NORMAL) && !got_fmt && !got_xma2) {
422  "found no 'fmt ' tag before the 'data' tag\n");
423  return AVERROR_INVALIDDATA;
424  }
425 
426  if (rf64) {
427  next_tag_ofs = wav->data_end = avio_tell(pb) + data_size;
428  } else if (size != 0xFFFFFFFF) {
429  data_size = size;
430  next_tag_ofs = wav->data_end = size ? next_tag_ofs : INT64_MAX;
431  } else {
432  av_log(s, AV_LOG_WARNING, "Ignoring maximum wav data size, "
433  "file may be invalid\n");
434  data_size = 0;
435  next_tag_ofs = wav->data_end = INT64_MAX;
436  }
437 
438  data_ofs = avio_tell(pb);
439 
440  /* don't look for footer metadata if we can't seek or if we don't
441  * know where the data tag ends
442  */
443  if (!(pb->seekable & AVIO_SEEKABLE_NORMAL) || (!rf64 && !size))
444  goto break_loop;
445  break;
446  case MKTAG('f', 'a', 'c', 't'):
447  if (!sample_count)
448  sample_count = (!wav->rifx ? avio_rl32(pb) : avio_rb32(pb));
449  break;
450  case MKTAG('b', 'e', 'x', 't'):
451  if ((ret = wav_parse_bext_tag(s, size)) < 0)
452  return ret;
453  break;
454  case MKTAG('S','M','V','0'):
455  if (!got_fmt) {
456  av_log(s, AV_LOG_ERROR, "found no 'fmt ' tag before the 'SMV0' tag\n");
457  return AVERROR_INVALIDDATA;
458  }
459  // SMV file, a wav file with video appended.
460  if (size != MKTAG('0','2','0','0')) {
461  av_log(s, AV_LOG_ERROR, "Unknown SMV version found\n");
462  goto break_loop;
463  }
464  av_log(s, AV_LOG_DEBUG, "Found SMV data\n");
465  wav->smv_given_first = 0;
466  vst = avformat_new_stream(s, NULL);
467  if (!vst)
468  return AVERROR(ENOMEM);
469  avio_r8(pb);
470  vst->id = 1;
473  vst->codecpar->width = avio_rl24(pb);
474  vst->codecpar->height = avio_rl24(pb);
475  if ((ret = ff_alloc_extradata(vst->codecpar, 4)) < 0) {
476  av_log(s, AV_LOG_ERROR, "Could not allocate extradata.\n");
477  return ret;
478  }
479  size = avio_rl24(pb);
480  wav->smv_data_ofs = avio_tell(pb) + (size - 5) * 3;
481  avio_rl24(pb);
482  wav->smv_block_size = avio_rl24(pb);
483  avpriv_set_pts_info(vst, 32, 1, avio_rl24(pb));
484  vst->duration = avio_rl24(pb);
485  avio_rl24(pb);
486  avio_rl24(pb);
487  wav->smv_frames_per_jpeg = avio_rl24(pb);
488  if (wav->smv_frames_per_jpeg > 65536) {
489  av_log(s, AV_LOG_ERROR, "too many frames per jpeg\n");
490  return AVERROR_INVALIDDATA;
491  }
493  wav->smv_cur_pt = 0;
494  goto break_loop;
495  case MKTAG('L', 'I', 'S', 'T'):
496  if (size < 4) {
497  av_log(s, AV_LOG_ERROR, "too short LIST tag\n");
498  return AVERROR_INVALIDDATA;
499  }
500  switch (avio_rl32(pb)) {
501  case MKTAG('I', 'N', 'F', 'O'):
502  ff_read_riff_info(s, size - 4);
503  }
504  break;
505  case MKTAG('I', 'D', '3', ' '):
506  case MKTAG('i', 'd', '3', ' '): {
507  ID3v2ExtraMeta *id3v2_extra_meta = NULL;
508  ff_id3v2_read_dict(pb, &s->internal->id3v2_meta, ID3v2_DEFAULT_MAGIC, &id3v2_extra_meta);
509  if (id3v2_extra_meta) {
510  ff_id3v2_parse_apic(s, id3v2_extra_meta);
511  ff_id3v2_parse_chapters(s, id3v2_extra_meta);
512  ff_id3v2_parse_priv(s, id3v2_extra_meta);
513  }
514  ff_id3v2_free_extra_meta(&id3v2_extra_meta);
515  }
516  break;
517  }
518 
519  /* seek to next tag unless we know that we'll run into EOF */
520  if ((avio_size(pb) > 0 && next_tag_ofs >= avio_size(pb)) ||
521  wav_seek_tag(wav, pb, next_tag_ofs, SEEK_SET) < 0) {
522  break;
523  }
524  }
525 
526 break_loop:
527  if (!got_fmt && !got_xma2) {
528  av_log(s, AV_LOG_ERROR, "no 'fmt ' or 'XMA2' tag found\n");
529  return AVERROR_INVALIDDATA;
530  }
531 
532  if (data_ofs < 0) {
533  av_log(s, AV_LOG_ERROR, "no 'data' tag found\n");
534  return AVERROR_INVALIDDATA;
535  }
536 
537  avio_seek(pb, data_ofs, SEEK_SET);
538 
539  if (data_size > (INT64_MAX>>3)) {
540  av_log(s, AV_LOG_WARNING, "Data size %"PRId64" is too large\n", data_size);
541  data_size = 0;
542  }
543 
544  if ( st->codecpar->bit_rate > 0 && data_size > 0
545  && st->codecpar->sample_rate > 0
546  && sample_count > 0 && st->codecpar->channels > 1
547  && sample_count % st->codecpar->channels == 0) {
548  if (fabs(8.0 * data_size * st->codecpar->channels * st->codecpar->sample_rate /
549  sample_count /st->codecpar->bit_rate - 1.0) < 0.3)
550  sample_count /= st->codecpar->channels;
551  }
552 
553  if ( data_size > 0 && sample_count && st->codecpar->channels
554  && (data_size << 3) / sample_count / st->codecpar->channels > st->codecpar->bits_per_coded_sample + 1) {
555  av_log(s, AV_LOG_WARNING, "ignoring wrong sample_count %"PRId64"\n", sample_count);
556  sample_count = 0;
557  }
558 
559  /* G.729 hack (for Ticket4577)
560  * FIXME: Come up with cleaner, more general solution */
561  if (st->codecpar->codec_id == AV_CODEC_ID_G729 && sample_count && (data_size << 3) > sample_count) {
562  av_log(s, AV_LOG_WARNING, "ignoring wrong sample_count %"PRId64"\n", sample_count);
563  sample_count = 0;
564  }
565 
566  if (!sample_count || av_get_exact_bits_per_sample(st->codecpar->codec_id) > 0)
567  if ( st->codecpar->channels
568  && data_size
570  && wav->data_end <= avio_size(pb))
571  sample_count = (data_size << 3)
572  /
573  (st->codecpar->channels * (uint64_t)av_get_bits_per_sample(st->codecpar->codec_id));
574 
575  if (sample_count)
576  st->duration = sample_count;
577 
579  st->codecpar->block_align == st->codecpar->channels * 4 &&
580  st->codecpar->bits_per_coded_sample == 32 &&
581  st->codecpar->extradata_size == 2 &&
582  AV_RL16(st->codecpar->extradata) == 1) {
585  } else if (st->codecpar->codec_id == AV_CODEC_ID_PCM_S24LE &&
586  st->codecpar->block_align == st->codecpar->channels * 4 &&
587  st->codecpar->bits_per_coded_sample == 24) {
589  } else if (st->codecpar->codec_id == AV_CODEC_ID_XMA1 ||
591  st->codecpar->block_align = 2048;
592  } else if (st->codecpar->codec_id == AV_CODEC_ID_ADPCM_MS && st->codecpar->channels > 2) {
593  st->codecpar->block_align *= st->codecpar->channels;
594  }
595 
596  ff_metadata_conv_ctx(s, NULL, wav_metadata_conv);
598 
599  set_spdif(s, wav);
600 
601  return 0;
602 }
603 
604 /**
605  * Find chunk with w64 GUID by skipping over other chunks.
606  * @return the size of the found chunk
607  */
608 static int64_t find_guid(AVIOContext *pb, const uint8_t guid1[16])
609 {
610  uint8_t guid[16];
611  int64_t size;
612 
613  while (!avio_feof(pb)) {
614  avio_read(pb, guid, 16);
615  size = avio_rl64(pb);
616  if (size <= 24)
617  return AVERROR_INVALIDDATA;
618  if (!memcmp(guid, guid1, 16))
619  return size;
620  avio_skip(pb, FFALIGN(size, INT64_C(8)) - 24);
621  }
622  return AVERROR_EOF;
623 }
624 
625 #define MAX_SIZE 4096
626 
627 static int wav_read_packet(AVFormatContext *s, AVPacket *pkt)
628 {
629  int ret, size;
630  int64_t left;
631  AVStream *st;
632  WAVDemuxContext *wav = s->priv_data;
633 
634  if (CONFIG_SPDIF_DEMUXER && wav->spdif == 1)
635  return ff_spdif_read_packet(s, pkt);
636 
637  if (wav->smv_data_ofs > 0) {
638  int64_t audio_dts, video_dts;
639 smv_retry:
640  audio_dts = (int32_t)s->streams[0]->cur_dts;
641  video_dts = (int32_t)s->streams[1]->cur_dts;
642 
644  /*We always return a video frame first to get the pixel format first*/
645  wav->smv_last_stream = wav->smv_given_first ?
646  av_compare_ts(video_dts, s->streams[1]->time_base,
647  audio_dts, s->streams[0]->time_base) > 0 : 0;
648  wav->smv_given_first = 1;
649  }
650  wav->smv_last_stream = !wav->smv_last_stream;
651  wav->smv_last_stream |= wav->audio_eof;
652  wav->smv_last_stream &= !wav->smv_eof;
653  if (wav->smv_last_stream) {
654  uint64_t old_pos = avio_tell(s->pb);
655  uint64_t new_pos = wav->smv_data_ofs +
656  wav->smv_block * wav->smv_block_size;
657  if (avio_seek(s->pb, new_pos, SEEK_SET) < 0) {
658  ret = AVERROR_EOF;
659  goto smv_out;
660  }
661  size = avio_rl24(s->pb);
662  ret = av_get_packet(s->pb, pkt, size);
663  if (ret < 0)
664  goto smv_out;
665  pkt->pos -= 3;
666  pkt->pts = wav->smv_block * wav->smv_frames_per_jpeg + wav->smv_cur_pt;
667  wav->smv_cur_pt++;
668  if (wav->smv_frames_per_jpeg > 0)
669  wav->smv_cur_pt %= wav->smv_frames_per_jpeg;
670  if (!wav->smv_cur_pt)
671  wav->smv_block++;
672 
673  pkt->stream_index = 1;
674 smv_out:
675  avio_seek(s->pb, old_pos, SEEK_SET);
676  if (ret == AVERROR_EOF) {
677  wav->smv_eof = 1;
678  goto smv_retry;
679  }
680  return ret;
681  }
682  }
683 
684  st = s->streams[0];
685 
686  left = wav->data_end - avio_tell(s->pb);
687  if (wav->ignore_length)
688  left = INT_MAX;
689  if (left <= 0) {
690  if (CONFIG_W64_DEMUXER && wav->w64)
691  left = find_guid(s->pb, ff_w64_guid_data) - 24;
692  else
693  left = find_tag(wav, s->pb, MKTAG('d', 'a', 't', 'a'));
694  if (left < 0) {
695  wav->audio_eof = 1;
696  if (wav->smv_data_ofs > 0 && !wav->smv_eof)
697  goto smv_retry;
698  return AVERROR_EOF;
699  }
700  wav->data_end = avio_tell(s->pb) + left;
701  }
702 
703  size = MAX_SIZE;
704  if (st->codecpar->block_align > 1) {
705  if (size < st->codecpar->block_align)
706  size = st->codecpar->block_align;
707  size = (size / st->codecpar->block_align) * st->codecpar->block_align;
708  }
709  size = FFMIN(size, left);
710  ret = av_get_packet(s->pb, pkt, size);
711  if (ret < 0)
712  return ret;
713  pkt->stream_index = 0;
714 
715  return ret;
716 }
717 
718 static int wav_read_seek(AVFormatContext *s,
719  int stream_index, int64_t timestamp, int flags)
720 {
721  WAVDemuxContext *wav = s->priv_data;
722  AVStream *st;
723  wav->smv_eof = 0;
724  wav->audio_eof = 0;
725  if (wav->smv_data_ofs > 0) {
726  int64_t smv_timestamp = timestamp;
727  if (stream_index == 0)
728  smv_timestamp = av_rescale_q(timestamp, s->streams[0]->time_base, s->streams[1]->time_base);
729  else
730  timestamp = av_rescale_q(smv_timestamp, s->streams[1]->time_base, s->streams[0]->time_base);
731  if (wav->smv_frames_per_jpeg > 0) {
732  wav->smv_block = smv_timestamp / wav->smv_frames_per_jpeg;
733  wav->smv_cur_pt = smv_timestamp % wav->smv_frames_per_jpeg;
734  }
735  }
736 
737  st = s->streams[0];
738  switch (st->codecpar->codec_id) {
739  case AV_CODEC_ID_MP2:
740  case AV_CODEC_ID_MP3:
741  case AV_CODEC_ID_AC3:
742  case AV_CODEC_ID_DTS:
743  case AV_CODEC_ID_XMA2:
744  /* use generic seeking with dynamically generated indexes */
745  return -1;
746  default:
747  break;
748  }
749  return ff_pcm_read_seek(s, stream_index, timestamp, flags);
750 }
751 
752 #define OFFSET(x) offsetof(WAVDemuxContext, x)
753 #define DEC AV_OPT_FLAG_DECODING_PARAM
754 static const AVOption demux_options[] = {
755  { "ignore_length", "Ignore length", OFFSET(ignore_length), AV_OPT_TYPE_BOOL, { .i64 = 0 }, 0, 1, DEC },
756  { NULL },
757 };
758 
759 static const AVClass wav_demuxer_class = {
760  .class_name = "WAV demuxer",
761  .item_name = av_default_item_name,
762  .option = demux_options,
763  .version = LIBAVUTIL_VERSION_INT,
764 };
766  .name = "wav",
767  .long_name = NULL_IF_CONFIG_SMALL("WAV / WAVE (Waveform Audio)"),
768  .priv_data_size = sizeof(WAVDemuxContext),
769  .read_probe = wav_probe,
770  .read_header = wav_read_header,
771  .read_packet = wav_read_packet,
772  .read_seek = wav_read_seek,
774  .codec_tag = (const AVCodecTag * const []) { ff_codec_wav_tags, 0 },
775  .priv_class = &wav_demuxer_class,
776 };
777 #endif /* CONFIG_WAV_DEMUXER */
778 
779 #if CONFIG_W64_DEMUXER
780 static int w64_probe(const AVProbeData *p)
781 {
782  if (p->buf_size <= 40)
783  return 0;
784  if (!memcmp(p->buf, ff_w64_guid_riff, 16) &&
785  !memcmp(p->buf + 24, ff_w64_guid_wave, 16))
786  return AVPROBE_SCORE_MAX;
787  else
788  return 0;
789 }
790 
791 static int w64_read_header(AVFormatContext *s)
792 {
793  int64_t size, data_ofs = 0;
794  AVIOContext *pb = s->pb;
795  WAVDemuxContext *wav = s->priv_data;
796  AVStream *st;
797  uint8_t guid[16];
798  int ret;
799 
800  avio_read(pb, guid, 16);
801  if (memcmp(guid, ff_w64_guid_riff, 16))
802  return AVERROR_INVALIDDATA;
803 
804  /* riff + wave + fmt + sizes */
805  if (avio_rl64(pb) < 16 + 8 + 16 + 8 + 16 + 8)
806  return AVERROR_INVALIDDATA;
807 
808  avio_read(pb, guid, 16);
809  if (memcmp(guid, ff_w64_guid_wave, 16)) {
810  av_log(s, AV_LOG_ERROR, "could not find wave guid\n");
811  return AVERROR_INVALIDDATA;
812  }
813 
814  wav->w64 = 1;
815 
816  st = avformat_new_stream(s, NULL);
817  if (!st)
818  return AVERROR(ENOMEM);
819 
820  while (!avio_feof(pb)) {
821  if (avio_read(pb, guid, 16) != 16)
822  break;
823  size = avio_rl64(pb);
824  if (size <= 24 || INT64_MAX - size < avio_tell(pb))
825  return AVERROR_INVALIDDATA;
826 
827  if (!memcmp(guid, ff_w64_guid_fmt, 16)) {
828  /* subtract chunk header size - normal wav file doesn't count it */
829  ret = ff_get_wav_header(s, pb, st->codecpar, size - 24, 0);
830  if (ret < 0)
831  return ret;
832  avio_skip(pb, FFALIGN(size, INT64_C(8)) - size);
833 
834  avpriv_set_pts_info(st, 64, 1, st->codecpar->sample_rate);
835  } else if (!memcmp(guid, ff_w64_guid_fact, 16)) {
836  int64_t samples;
837 
838  samples = avio_rl64(pb);
839  if (samples > 0)
840  st->duration = samples;
841  avio_skip(pb, FFALIGN(size, INT64_C(8)) - 32);
842  } else if (!memcmp(guid, ff_w64_guid_data, 16)) {
843  wav->data_end = avio_tell(pb) + size - 24;
844 
845  data_ofs = avio_tell(pb);
846  if (!(pb->seekable & AVIO_SEEKABLE_NORMAL))
847  break;
848 
849  avio_skip(pb, size - 24);
850  } else if (!memcmp(guid, ff_w64_guid_summarylist, 16)) {
851  int64_t start, end, cur;
852  uint32_t count, chunk_size, i;
853 
854  start = avio_tell(pb);
855  end = start + FFALIGN(size, INT64_C(8)) - 24;
856  count = avio_rl32(pb);
857 
858  for (i = 0; i < count; i++) {
859  char chunk_key[5], *value;
860 
861  if (avio_feof(pb) || (cur = avio_tell(pb)) < 0 || cur > end - 8 /* = tag + size */)
862  break;
863 
864  chunk_key[4] = 0;
865  avio_read(pb, chunk_key, 4);
866  chunk_size = avio_rl32(pb);
867  if (chunk_size == UINT32_MAX)
868  return AVERROR_INVALIDDATA;
869 
870  value = av_mallocz(chunk_size + 1);
871  if (!value)
872  return AVERROR(ENOMEM);
873 
874  ret = avio_get_str16le(pb, chunk_size, value, chunk_size);
875  avio_skip(pb, chunk_size - ret);
876 
877  av_dict_set(&s->metadata, chunk_key, value, AV_DICT_DONT_STRDUP_VAL);
878  }
879 
880  avio_skip(pb, end - avio_tell(pb));
881  } else {
882  av_log(s, AV_LOG_DEBUG, "unknown guid: "FF_PRI_GUID"\n", FF_ARG_GUID(guid));
883  avio_skip(pb, FFALIGN(size, INT64_C(8)) - 24);
884  }
885  }
886 
887  if (!data_ofs)
888  return AVERROR_EOF;
889 
890  ff_metadata_conv_ctx(s, NULL, wav_metadata_conv);
892 
893  handle_stream_probing(st);
895 
896  avio_seek(pb, data_ofs, SEEK_SET);
897 
898  set_spdif(s, wav);
899 
900  return 0;
901 }
902 
904  .name = "w64",
905  .long_name = NULL_IF_CONFIG_SMALL("Sony Wave64"),
906  .priv_data_size = sizeof(WAVDemuxContext),
907  .read_probe = w64_probe,
908  .read_header = w64_read_header,
909  .read_packet = wav_read_packet,
910  .read_seek = wav_read_seek,
912  .codec_tag = (const AVCodecTag * const []) { ff_codec_wav_tags, 0 },
913 };
914 #endif /* CONFIG_W64_DEMUXER */
WAVDemuxContext
Definition: wavdec.c:45
AV_CODEC_ID_PCM_S16LE
@ AV_CODEC_ID_PCM_S16LE
Definition: codec_id.h:301
WAVDemuxContext::unaligned
int unaligned
Definition: wavdec.c:60
WAVDemuxContext::smv_block
int smv_block
Definition: wavdec.c:52
AV_LOG_WARNING
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
AV_CODEC_ID_ADPCM_MS
@ AV_CODEC_ID_ADPCM_MS
Definition: codec_id.h:346
AVCodecParameters::extradata
uint8_t * extradata
Extra binary data needed for initializing the decoder, codec-dependent.
Definition: codec_par.h:74
AV_CODEC_ID_AC3
@ AV_CODEC_ID_AC3
Definition: codec_id.h:413
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
opt.h
avformat_new_stream
AVStream * avformat_new_stream(AVFormatContext *s, const AVCodec *c)
Add a new stream to a media file.
Definition: utils.c:4519
pcm.h
AVCodecParameters::codec_type
enum AVMediaType codec_type
General type of the encoded data.
Definition: codec_par.h:56
AV_WL32
#define AV_WL32(p, v)
Definition: intreadwrite.h:426
av_compare_ts
int av_compare_ts(int64_t ts_a, AVRational tb_a, int64_t ts_b, AVRational tb_b)
Compare two timestamps each in its own time base.
Definition: mathematics.c:147
ff_get_extradata
int ff_get_extradata(AVFormatContext *s, AVCodecParameters *par, AVIOContext *pb, int size)
Allocate extradata with additional AV_INPUT_BUFFER_PADDING_SIZE at end which is always set to 0 and f...
Definition: utils.c:3339
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:55
MKTAG
#define MKTAG(a, b, c, d)
Definition: common.h:406
WAVDemuxContext::smv_data_ofs
int64_t smv_data_ofs
Definition: wavdec.c:49
audio_dts
int64_t audio_dts
Definition: movenc.c:60
WAVDemuxContext::smv_frames_per_jpeg
int smv_frames_per_jpeg
Definition: wavdec.c:51
WAVDemuxContext::smv_cur_pt
int smv_cur_pt
Definition: wavdec.c:58
id3v2.h
end
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
AVOption
AVOption.
Definition: opt.h:246
AVMetadataConv
Definition: metadata.h:34
ff_codec_wav_tags
const AVCodecTag ff_codec_wav_tags[]
Definition: riff.c:506
samples
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new samples
Definition: fate.txt:139
mathematics.h
AVProbeData::buf_size
int buf_size
Size of buf except extra allocated bytes.
Definition: avformat.h:444
ff_id3v2_parse_chapters
int ff_id3v2_parse_chapters(AVFormatContext *s, ID3v2ExtraMeta *extra_meta)
Create chapters for all CHAP tags found in the ID3v2 header.
Definition: id3v2.c:1170
avio_size
int64_t avio_size(AVIOContext *s)
Get the filesize.
Definition: aviobuf.c:334
WAVDemuxContext::data_end
int64_t data_end
Definition: wavdec.c:47
video_dts
int64_t video_dts
Definition: movenc.c:60
av_malloc
#define av_malloc(s)
Definition: tableprint_vlc.h:31
WAVDemuxContext::w64
int w64
Definition: wavdec.c:48
ff_wav_demuxer
AVInputFormat ff_wav_demuxer
CONFIG_W64_DEMUXER
#define CONFIG_W64_DEMUXER
Definition: config.h:2349
AVPROBE_SCORE_MAX
#define AVPROBE_SCORE_MAX
maximum score
Definition: avformat.h:453
return
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 it should return
Definition: filter_design.txt:264
AVCodecParameters::channels
int channels
Audio only.
Definition: codec_par.h:166
ff_w64_guid_summarylist
const uint8_t ff_w64_guid_summarylist[16]
Definition: w64.c:47
read_seek
static int read_seek(AVFormatContext *ctx, int stream_index, int64_t timestamp, int flags)
Definition: libcdio.c:153
x
FFmpeg Automated Testing Environment ************************************Introduction Using FATE from your FFmpeg source directory Submitting the results to the FFmpeg result aggregation server Uploading new samples to the fate suite FATE makefile targets and variables Makefile targets Makefile variables Examples Introduction **************FATE is an extended regression suite on the client side and a means for results aggregation and presentation on the server side The first part of this document explains how you can use FATE from your FFmpeg source directory to test your ffmpeg binary The second part describes how you can run FATE to submit the results to FFmpeg’s FATE server In any way you can have a look at the publicly viewable FATE results by visiting this as it can be seen if some test on some platform broke with their recent contribution This usually happens on the platforms the developers could not test on The second part of this document describes how you can run FATE to submit your results to FFmpeg’s FATE server If you want to submit your results be sure to check that your combination of OS and compiler is not already listed on the above mentioned website In the third part you can find a comprehensive listing of FATE makefile targets and variables Using FATE from your FFmpeg source directory **********************************************If you want to run FATE on your machine you need to have the samples in place You can get the samples via the build target fate rsync Use this command from the top level source this will cause FATE to fail NOTE To use a custom wrapper to run the pass ‘ target exec’ to ‘configure’ or set the TARGET_EXEC Make variable Submitting the results to the FFmpeg result aggregation server ****************************************************************To submit your results to the server you should run fate through the shell script ‘tests fate sh’ from the FFmpeg sources This script needs to be invoked with a configuration file as its first argument tests fate sh path to fate_config A configuration file template with comments describing the individual configuration variables can be found at ‘doc fate_config sh template’ Create a configuration that suits your based on the configuration template The ‘slot’ configuration variable can be any string that is not yet but it is suggested that you name it adhering to the following pattern ‘ARCH OS COMPILER COMPILER VERSION’ The configuration file itself will be sourced in a shell therefore all shell features may be used This enables you to setup the environment as you need it for your build For your first test runs the ‘fate_recv’ variable should be empty or commented out This will run everything as normal except that it will omit the submission of the results to the server The following files should be present in $workdir as specified in the configuration it may help to try out the ‘ssh’ command with one or more ‘ v’ options You should get detailed output concerning your SSH configuration and the authentication process The only thing left is to automate the execution of the fate sh script and the synchronisation of the samples directory Uploading new samples to the fate suite *****************************************If you need a sample uploaded send a mail to samples request This is for developers who have an account on the fate suite server If you upload new please make sure they are as small as space on each network bandwidth and so on benefit from smaller test cases Also keep in mind older checkouts use existing sample that means in practice generally do not remove or overwrite files as it likely would break older checkouts or releases Also all needed samples for a commit should be ideally before the push If you need an account for frequently uploading samples or you wish to help others by doing that send a mail to ffmpeg devel rsync vauL Duo x
Definition: fate.txt:150
avio_tell
static av_always_inline int64_t avio_tell(AVIOContext *s)
ftell() equivalent for AVIOContext.
Definition: avio.h:557
AVFMT_GENERIC_INDEX
#define AVFMT_GENERIC_INDEX
Use generic index building code.
Definition: avformat.h:463
AV_CODEC_ID_MP3
@ AV_CODEC_ID_MP3
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: codec_id.h:411
AVStream::duration
int64_t duration
Decoding: duration of the stream, in stream time base.
Definition: avformat.h:914
avio_rl16
unsigned int avio_rl16(AVIOContext *s)
Definition: aviobuf.c:731
CONFIG_SPDIF_DEMUXER
#define CONFIG_SPDIF_DEMUXER
Definition: config.h:2316
AV_DICT_DONT_STRDUP_VAL
#define AV_DICT_DONT_STRDUP_VAL
Take ownership of a value that's been allocated with av_malloc() or another memory allocation functio...
Definition: dict.h:74
ff_w64_demuxer
AVInputFormat ff_w64_demuxer
avassert.h
avio_rb32
unsigned int avio_rb32(AVIOContext *s)
Definition: aviobuf.c:778
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
AVInputFormat
Definition: avformat.h:636
AVCodecTag
Definition: internal.h:42
ID3v2ExtraMeta
Definition: id3v2.h:84
avio_get_str16le
int avio_get_str16le(AVIOContext *pb, int maxlen, char *buf, int buflen)
Read a UTF-16 string from pb and convert it to UTF-8.
FF_ARG_GUID
#define FF_ARG_GUID(g)
Definition: riff.h:105
intreadwrite.h
s
#define s(width, name)
Definition: cbs_vp9.c:257
AV_CODEC_ID_XMA1
@ AV_CODEC_ID_XMA1
Definition: codec_id.h:490
AVInputFormat::name
const char * name
A comma separated list of short names for the format.
Definition: avformat.h:641
AVProbeData::buf
unsigned char * buf
Buffer must have AVPROBE_PADDING_SIZE of extra allocated bytes filled with zero.
Definition: avformat.h:443
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVCodecParameters::width
int width
Video only.
Definition: codec_par.h:126
AV_CODEC_ID_MP2
@ AV_CODEC_ID_MP2
Definition: codec_id.h:410
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
AVStream::need_parsing
enum AVStreamParseType need_parsing
Definition: avformat.h:1083
AV_CODEC_ID_PCM_F24LE
@ AV_CODEC_ID_PCM_F24LE
Definition: codec_id.h:336
channels
channels
Definition: aptx.h:33
av_mallocz
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:237
av_rescale_q
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
AV_RL16
#define AV_RL16
Definition: intreadwrite.h:42
ff_read_riff_info
int ff_read_riff_info(AVFormatContext *s, int64_t size)
Definition: riffdec.c:228
MAX_SIZE
#define MAX_SIZE
Definition: vf_unsharp.c:305
key
const char * key
Definition: hwcontext_opencl.c:168
int32_t
int32_t
Definition: audio_convert.c:194
if
if(ret)
Definition: filter_design.txt:179
AVFormatContext
Format I/O context.
Definition: avformat.h:1335
internal.h
ff_metadata_conv_ctx
void ff_metadata_conv_ctx(AVFormatContext *ctx, const AVMetadataConv *d_conv, const AVMetadataConv *s_conv)
Definition: metadata.c:59
AVStream::codecpar
AVCodecParameters * codecpar
Codec parameters associated with this stream.
Definition: avformat.h:1012
LIBAVUTIL_VERSION_INT
#define LIBAVUTIL_VERSION_INT
Definition: version.h:85
read_header
static int read_header(FFV1Context *f)
Definition: ffv1dec.c:527
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
NULL
#define NULL
Definition: coverity.c:32
read_probe
static int read_probe(const AVProbeData *pd)
Definition: jvdec.c:55
av_default_item_name
const char * av_default_item_name(void *ptr)
Return the context name.
Definition: log.c:235
ff_id3v2_parse_apic
int ff_id3v2_parse_apic(AVFormatContext *s, ID3v2ExtraMeta *extra_meta)
Create a stream for each APIC (attached picture) extracted from the ID3v2 header.
Definition: id3v2.c:1130
avio_rb64
uint64_t avio_rb64(AVIOContext *s)
Definition: aviobuf.c:899
AVProbeData
This structure contains the data a format has to probe a file.
Definition: avformat.h:441
AVPROBE_SCORE_EXTENSION
#define AVPROBE_SCORE_EXTENSION
score for file extension
Definition: avformat.h:451
AVCodecParameters::sample_rate
int sample_rate
Audio only.
Definition: codec_par.h:170
AVCodecID
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: codec_id.h:46
ff_w64_guid_fmt
const uint8_t ff_w64_guid_fmt[16]
Definition: w64.c:33
AVCodecParameters::extradata_size
int extradata_size
Size of the extradata content in bytes.
Definition: codec_par.h:78
av_get_bits_per_sample
int av_get_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: utils.c:1552
WAVDemuxContext::smv_last_stream
int smv_last_stream
Definition: wavdec.c:53
avio_rl32
unsigned int avio_rl32(AVIOContext *s)
Definition: aviobuf.c:747
AVIOContext
Bytestream IO Context.
Definition: avio.h:161
AV_CODEC_ID_PCM_S24LE
@ AV_CODEC_ID_PCM_S24LE
Definition: codec_id.h:313
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:186
AVIOContext::seekable
int seekable
A combination of AVIO_SEEKABLE_ flags or 0 when the stream is not seekable.
Definition: avio.h:260
ff_spdif_probe
int ff_spdif_probe(const uint8_t *p_buf, int buf_size, enum AVCodecID *codec)
Definition: spdifdec.c:116
WAVDemuxContext::smv_eof
int smv_eof
Definition: wavdec.c:54
avpriv_set_pts_info
void avpriv_set_pts_info(AVStream *s, int pts_wrap_bits, unsigned int pts_num, unsigned int pts_den)
Set the time base and wrapping info for a given stream.
Definition: utils.c:4938
AV_CODEC_ID_DTS
@ AV_CODEC_ID_DTS
Definition: codec_id.h:414
AVStream::probe_packets
int probe_packets
Number of packets to buffer for codec probing.
Definition: avformat.h:1075
size
int size
Definition: twinvq_data.h:11134
AV_CODEC_ID_SMVJPEG
@ AV_CODEC_ID_SMVJPEG
Definition: codec_id.h:258
avio.h
ID3v2_DEFAULT_MAGIC
#define ID3v2_DEFAULT_MAGIC
Default magic bytes for ID3v2 header: "ID3".
Definition: id3v2.h:35
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
ff_riff_info_conv
const AVMetadataConv ff_riff_info_conv[]
Definition: riff.c:591
avio_r8
int avio_r8(AVIOContext *s)
Definition: aviobuf.c:616
FFMIN
#define FFMIN(a, b)
Definition: common.h:96
ffio_ensure_seekback
int ffio_ensure_seekback(AVIOContext *s, int64_t buf_size)
Ensures that the requested seekback buffer size will be available.
Definition: aviobuf.c:982
offset
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 offset
Definition: writing_filters.txt:86
version
version
Definition: libkvazaar.c:292
ff_spdif_read_packet
int ff_spdif_read_packet(AVFormatContext *s, AVPacket *pkt)
Definition: spdifdec.c:173
WAVDemuxContext::rifx
int rifx
Definition: wavdec.c:61
log.h
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
AVPacket::pts
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: packet.h:348
avio_rl24
unsigned int avio_rl24(AVIOContext *s)
Definition: aviobuf.c:739
avio_internal.h
WAVDemuxContext::audio_eof
int audio_eof
Definition: wavdec.c:55
ff_w64_guid_wave
const uint8_t ff_w64_guid_wave[16]
Definition: w64.c:28
AVCodecParameters::height
int height
Definition: codec_par.h:127
AVCodecParameters::block_align
int block_align
Audio only.
Definition: codec_par.h:177
WAVDemuxContext::smv_block_size
int smv_block_size
Definition: wavdec.c:50
ff_id3v2_read_dict
void ff_id3v2_read_dict(AVIOContext *pb, AVDictionary **metadata, const char *magic, ID3v2ExtraMeta **extra_meta)
Read an ID3v2 tag into specified dictionary and retrieve supported extra metadata.
Definition: id3v2.c:1102
value
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 value
Definition: writing_filters.txt:86
uint8_t
uint8_t
Definition: audio_convert.c:194
AV_CODEC_ID_PCM_F16LE
@ AV_CODEC_ID_PCM_F16LE
Definition: codec_id.h:335
ff_w64_guid_fact
const uint8_t ff_w64_guid_fact[16]
Definition: w64.c:38
len
int len
Definition: vorbis_enc_data.h:452
ff_get_wav_header
int ff_get_wav_header(AVFormatContext *s, AVIOContext *pb, AVCodecParameters *par, int size, int big_endian)
Definition: riffdec.c:91
av_get_exact_bits_per_sample
int av_get_exact_bits_per_sample(enum AVCodecID codec_id)
Return codec bits per sample.
Definition: utils.c:1468
av_get_packet
int av_get_packet(AVIOContext *s, AVPacket *pkt, int size)
Allocate and read the payload of a packet and initialize its fields with default values.
Definition: utils.c:304
av_uninit
#define av_uninit(x)
Definition: attributes.h:154
tag
uint32_t tag
Definition: movenc.c:1532
AVStream::id
int id
Format-specific stream ID.
Definition: avformat.h:872
ret
ret
Definition: filter_design.txt:187
read_packet
static int read_packet(void *opaque, uint8_t *buf, int buf_size)
Definition: avio_reading.c:42
AVStream
Stream structure.
Definition: avformat.h:865
WAVDemuxContext::spdif
int spdif
Definition: wavdec.c:57
avio_seek
int64_t avio_seek(AVIOContext *s, int64_t offset, int whence)
fseek() equivalent for AVIOContext.
Definition: aviobuf.c:241
AVClass::class_name
const char * class_name
The name of the class; usually it is the same name as the context structure type to which the AVClass...
Definition: log.h:72
ff_pcm_read_seek
int ff_pcm_read_seek(AVFormatContext *s, int stream_index, int64_t timestamp, int flags)
Definition: pcm.c:52
metadata.h
pos
unsigned int pos
Definition: spdifenc.c:410
avformat.h
dict.h
left
Tag MUST be and< 10hcoeff half pel interpolation filter coefficients, hcoeff[0] are the 2 middle coefficients[1] are the next outer ones and so on, resulting in a filter like:...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... the sign of the coefficients is not explicitly stored but alternates after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... hcoeff[0] is not explicitly stored but found by subtracting the sum of all stored coefficients with signs from 32 hcoeff[0]=32 - hcoeff[1] - hcoeff[2] - ... a good choice for hcoeff and htaps is htaps=6 hcoeff={40,-10, 2} an alternative which requires more computations at both encoder and decoder side and may or may not be better is htaps=8 hcoeff={42,-14, 6,-2}ref_frames minimum of the number of available reference frames and max_ref_frames for example the first frame after a key frame always has ref_frames=1spatial_decomposition_type wavelet type 0 is a 9/7 symmetric compact integer wavelet 1 is a 5/3 symmetric compact integer wavelet others are reserved stored as delta from last, last is reset to 0 if always_reset||keyframeqlog quality(logarithmic quantizer scale) stored as delta from last, last is reset to 0 if always_reset||keyframemv_scale stored as delta from last, last is reset to 0 if always_reset||keyframe FIXME check that everything works fine if this changes between framesqbias dequantization bias stored as delta from last, last is reset to 0 if always_reset||keyframeblock_max_depth maximum depth of the block tree stored as delta from last, last is reset to 0 if always_reset||keyframequant_table quantization tableHighlevel bitstream structure:==============================--------------------------------------------|Header|--------------------------------------------|------------------------------------|||Block0||||split?||||yes no||||......... intra?||||:Block01 :yes no||||:Block02 :....... ..........||||:Block03 ::y DC ::ref index:||||:Block04 ::cb DC ::motion x :||||......... :cr DC ::motion y :||||....... ..........|||------------------------------------||------------------------------------|||Block1|||...|--------------------------------------------|------------ ------------ ------------|||Y subbands||Cb subbands||Cr subbands||||--- ---||--- ---||--- ---|||||LL0||HL0||||LL0||HL0||||LL0||HL0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||LH0||HH0||||LH0||HH0||||LH0||HH0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HL1||LH1||||HL1||LH1||||HL1||LH1|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HH1||HL2||||HH1||HL2||||HH1||HL2|||||...||...||...|||------------ ------------ ------------|--------------------------------------------Decoding process:=================------------|||Subbands|------------||||------------|Intra DC||||LL0 subband prediction ------------|\ Dequantization ------------------- \||Reference frames|\ IDWT|------- -------|Motion \|||Frame 0||Frame 1||Compensation . OBMC v -------|------- -------|--------------. \------> Frame n output Frame Frame<----------------------------------/|...|------------------- Range Coder:============Binary Range Coder:------------------- The implemented range coder is an adapted version based upon "Range encoding: an algorithm for removing redundancy from a digitised message." by G. N. N. Martin. The symbols encoded by the Snow range coder are bits(0|1). The associated probabilities are not fix but change depending on the symbol mix seen so far. bit seen|new state ---------+----------------------------------------------- 0|256 - state_transition_table[256 - old_state];1|state_transition_table[old_state];state_transition_table={ 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, 195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, 210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, 226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};FIXME Range Coding of integers:------------------------- FIXME Neighboring Blocks:===================left and top are set to the respective blocks unless they are outside of the image in which case they are set to the Null block top-left is set to the top left block unless it is outside of the image in which case it is set to the left block if this block has no larger parent block or it is at the left side of its parent block and the top right block is not outside of the image then the top right block is used for top-right else the top-left block is used Null block y, cb, cr are 128 level, ref, mx and my are 0 Motion Vector Prediction:=========================1. the motion vectors of all the neighboring blocks are scaled to compensate for the difference of reference frames scaled_mv=(mv *(256 *(current_reference+1)/(mv.reference+1))+128)> the median of the scaled left
Definition: snow.txt:386
set_spdif
static void set_spdif(AVFormatContext *s, WAVDemuxContext *wav)
Definition: wavdec.c:64
OFFSET
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 keep it simple and lowercase description are in without and describe what they for example set the foo of the bar offset is the offset of the field in your see the OFFSET() macro
WAVDemuxContext::smv_given_first
int smv_given_first
Definition: wavdec.c:59
AV_CODEC_ID_G729
@ AV_CODEC_ID_G729
Definition: codec_id.h:463
w64.h
WAVDemuxContext::ignore_length
int ignore_length
Definition: wavdec.c:56
pkt
static AVPacket pkt
Definition: demuxing_decoding.c:54
AVIO_SEEKABLE_NORMAL
#define AVIO_SEEKABLE_NORMAL
Seeking works like for a local file.
Definition: avio.h:40
avio_read
int avio_read(AVIOContext *s, unsigned char *buf, int size)
Read size bytes from AVIOContext into buf.
Definition: aviobuf.c:625
temp
else temp
Definition: vf_mcdeint.c:256
AVSTREAM_PARSE_FULL_RAW
@ AVSTREAM_PARSE_FULL_RAW
full parsing and repack with timestamp and position generation by parser for raw this assumes that ea...
Definition: avformat.h:790
AVStream::request_probe
int request_probe
stream probing state -1 -> probing finished 0 -> no probing requested rest -> perform probing with re...
Definition: avformat.h:1122
AVPacket::stream_index
int stream_index
Definition: packet.h:357
avio_skip
int64_t avio_skip(AVIOContext *s, int64_t offset)
Skip given number of bytes forward.
Definition: aviobuf.c:329
AVMEDIA_TYPE_VIDEO
@ AVMEDIA_TYPE_VIDEO
Definition: avutil.h:201
ff_w64_guid_data
const uint8_t ff_w64_guid_data[16]
Definition: w64.c:42
AV_CODEC_ID_PCM_S32LE
@ AV_CODEC_ID_PCM_S32LE
Definition: codec_id.h:309
AVCodecParameters::bits_per_coded_sample
int bits_per_coded_sample
The number of bits per sample in the codedwords.
Definition: codec_par.h:102
find_guid
static const GUIDParseTable * find_guid(ff_asf_guid guid)
Definition: asfdec_o.c:1658
AV_CODEC_ID_XMA2
@ AV_CODEC_ID_XMA2
Definition: codec_id.h:491
ff_w64_guid_riff
const uint8_t ff_w64_guid_riff[16]
Definition: w64.c:23
av_free
#define av_free(p)
Definition: tableprint_vlc.h:34
FFALIGN
#define FFALIGN(x, a)
Definition: macros.h:48
AVCodecParameters::codec_id
enum AVCodecID codec_id
Specific type of the encoded data (the codec used).
Definition: codec_par.h:60
AVPacket
This structure stores compressed data.
Definition: packet.h:332
AV_OPT_TYPE_BOOL
@ AV_OPT_TYPE_BOOL
Definition: opt.h:240
av_dict_set
int av_dict_set(AVDictionary **pm, const char *key, const char *value, int flags)
Set the given entry in *pm, overwriting an existing entry.
Definition: dict.c:70
riff.h
ff_id3v2_free_extra_meta
void ff_id3v2_free_extra_meta(ID3v2ExtraMeta **extra_meta)
Free memory allocated parsing special (non-text) metadata.
Definition: id3v2.c:1114
AVPacket::pos
int64_t pos
byte position in stream, -1 if unknown
Definition: packet.h:375
avio_rl64
uint64_t avio_rl64(AVIOContext *s)
Definition: aviobuf.c:755
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:564
AVCodecParameters::bit_rate
int64_t bit_rate
The average bitrate of the encoded data (in bits per second).
Definition: codec_par.h:89
ff_id3v2_parse_priv
int ff_id3v2_parse_priv(AVFormatContext *s, ID3v2ExtraMeta *extra_meta)
Add metadata for all PRIV tags in the ID3v2 header.
Definition: id3v2.c:1261
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
snprintf
#define snprintf
Definition: snprintf.h:34
spdif.h
FF_PRI_GUID
#define FF_PRI_GUID
Definition: riff.h:101
av_fourcc2str
#define av_fourcc2str(fourcc)
Definition: avutil.h:348
ff_alloc_extradata
int ff_alloc_extradata(AVCodecParameters *par, int size)
Allocate extradata with additional AV_INPUT_BUFFER_PADDING_SIZE at end which is always set to 0.
Definition: utils.c:3321
avio_feof
int avio_feof(AVIOContext *s)
Similar to feof() but also returns nonzero on read errors.
Definition: aviobuf.c:356
DEC
#define DEC
Definition: librsvgdec.c:105