FFmpeg
audiotoolboxdec.c
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1 /*
2  * Audio Toolbox system codecs
3  *
4  * copyright (c) 2016 Rodger Combs
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #include <AudioToolbox/AudioToolbox.h>
24 
25 #include "config.h"
26 #include "avcodec.h"
27 #include "ac3_parser_internal.h"
28 #include "bytestream.h"
29 #include "internal.h"
30 #include "mpegaudiodecheader.h"
31 #include "libavutil/avassert.h"
32 #include "libavutil/opt.h"
33 #include "libavutil/log.h"
34 
35 #if __MAC_OS_X_VERSION_MIN_REQUIRED < 101100
36 #define kAudioFormatEnhancedAC3 'ec-3'
37 #endif
38 
39 typedef struct ATDecodeContext {
41 
42  AudioConverterRef converter;
43  AudioStreamPacketDescription pkt_desc;
46  char *decoded_data;
47  int channel_map[64];
48 
51 
52  int64_t last_pts;
53  int eof;
55 
56 static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile)
57 {
58  switch (codec) {
59  case AV_CODEC_ID_AAC:
60  return kAudioFormatMPEG4AAC;
61  case AV_CODEC_ID_AC3:
62  return kAudioFormatAC3;
64  return kAudioFormatAppleIMA4;
65  case AV_CODEC_ID_ALAC:
66  return kAudioFormatAppleLossless;
67  case AV_CODEC_ID_AMR_NB:
68  return kAudioFormatAMR;
69  case AV_CODEC_ID_EAC3:
71  case AV_CODEC_ID_GSM_MS:
72  return kAudioFormatMicrosoftGSM;
73  case AV_CODEC_ID_ILBC:
74  return kAudioFormatiLBC;
75  case AV_CODEC_ID_MP1:
76  return kAudioFormatMPEGLayer1;
77  case AV_CODEC_ID_MP2:
78  return kAudioFormatMPEGLayer2;
79  case AV_CODEC_ID_MP3:
80  return kAudioFormatMPEGLayer3;
82  return kAudioFormatALaw;
84  return kAudioFormatULaw;
85  case AV_CODEC_ID_QDMC:
86  return kAudioFormatQDesign;
87  case AV_CODEC_ID_QDM2:
88  return kAudioFormatQDesign2;
89  default:
90  av_assert0(!"Invalid codec ID!");
91  return 0;
92  }
93 }
94 
95 static int ffat_get_channel_id(AudioChannelLabel label)
96 {
97  if (label == 0)
98  return -1;
99  else if (label <= kAudioChannelLabel_LFEScreen)
100  return label - 1;
101  else if (label <= kAudioChannelLabel_RightSurround)
102  return label + 4;
103  else if (label <= kAudioChannelLabel_CenterSurround)
104  return label + 1;
105  else if (label <= kAudioChannelLabel_RightSurroundDirect)
106  return label + 23;
107  else if (label <= kAudioChannelLabel_TopBackRight)
108  return label - 1;
109  else if (label < kAudioChannelLabel_RearSurroundLeft)
110  return -1;
111  else if (label <= kAudioChannelLabel_RearSurroundRight)
112  return label - 29;
113  else if (label <= kAudioChannelLabel_RightWide)
114  return label - 4;
115  else if (label == kAudioChannelLabel_LFE2)
117  else if (label == kAudioChannelLabel_Mono)
119  else
120  return -1;
121 }
122 
123 static int ffat_compare_channel_descriptions(const void* a, const void* b)
124 {
125  const AudioChannelDescription* da = a;
126  const AudioChannelDescription* db = b;
127  return ffat_get_channel_id(da->mChannelLabel) - ffat_get_channel_id(db->mChannelLabel);
128 }
129 
130 static AudioChannelLayout *ffat_convert_layout(AudioChannelLayout *layout, UInt32* size)
131 {
132  AudioChannelLayoutTag tag = layout->mChannelLayoutTag;
133  AudioChannelLayout *new_layout;
134  if (tag == kAudioChannelLayoutTag_UseChannelDescriptions)
135  return layout;
136  else if (tag == kAudioChannelLayoutTag_UseChannelBitmap)
137  AudioFormatGetPropertyInfo(kAudioFormatProperty_ChannelLayoutForBitmap,
138  sizeof(UInt32), &layout->mChannelBitmap, size);
139  else
140  AudioFormatGetPropertyInfo(kAudioFormatProperty_ChannelLayoutForTag,
141  sizeof(AudioChannelLayoutTag), &tag, size);
142  new_layout = av_malloc(*size);
143  if (!new_layout) {
144  av_free(layout);
145  return NULL;
146  }
147  if (tag == kAudioChannelLayoutTag_UseChannelBitmap)
148  AudioFormatGetProperty(kAudioFormatProperty_ChannelLayoutForBitmap,
149  sizeof(UInt32), &layout->mChannelBitmap, size, new_layout);
150  else
151  AudioFormatGetProperty(kAudioFormatProperty_ChannelLayoutForTag,
152  sizeof(AudioChannelLayoutTag), &tag, size, new_layout);
153  new_layout->mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions;
154  av_free(layout);
155  return new_layout;
156 }
157 
158 static int ffat_update_ctx(AVCodecContext *avctx)
159 {
160  ATDecodeContext *at = avctx->priv_data;
161  AudioStreamBasicDescription format;
162  UInt32 size = sizeof(format);
163  if (!AudioConverterGetProperty(at->converter,
164  kAudioConverterCurrentInputStreamDescription,
165  &size, &format)) {
166  if (format.mSampleRate)
167  avctx->sample_rate = format.mSampleRate;
168  avctx->channels = format.mChannelsPerFrame;
170  avctx->frame_size = format.mFramesPerPacket;
171  }
172 
173  if (!AudioConverterGetProperty(at->converter,
174  kAudioConverterCurrentOutputStreamDescription,
175  &size, &format)) {
176  format.mSampleRate = avctx->sample_rate;
177  format.mChannelsPerFrame = avctx->channels;
178  AudioConverterSetProperty(at->converter,
179  kAudioConverterCurrentOutputStreamDescription,
180  size, &format);
181  }
182 
183  if (!AudioConverterGetPropertyInfo(at->converter, kAudioConverterOutputChannelLayout,
184  &size, NULL) && size) {
185  AudioChannelLayout *layout = av_malloc(size);
186  uint64_t layout_mask = 0;
187  int i;
188  if (!layout)
189  return AVERROR(ENOMEM);
190  AudioConverterGetProperty(at->converter, kAudioConverterOutputChannelLayout,
191  &size, layout);
192  if (!(layout = ffat_convert_layout(layout, &size)))
193  return AVERROR(ENOMEM);
194  for (i = 0; i < layout->mNumberChannelDescriptions; i++) {
195  int id = ffat_get_channel_id(layout->mChannelDescriptions[i].mChannelLabel);
196  if (id < 0)
197  goto done;
198  if (layout_mask & (1 << id))
199  goto done;
200  layout_mask |= 1 << id;
201  layout->mChannelDescriptions[i].mChannelFlags = i; // Abusing flags as index
202  }
203  avctx->channel_layout = layout_mask;
204  qsort(layout->mChannelDescriptions, layout->mNumberChannelDescriptions,
205  sizeof(AudioChannelDescription), &ffat_compare_channel_descriptions);
206  for (i = 0; i < layout->mNumberChannelDescriptions; i++)
207  at->channel_map[i] = layout->mChannelDescriptions[i].mChannelFlags;
208 done:
209  av_free(layout);
210  }
211 
212  if (!avctx->frame_size)
213  avctx->frame_size = 2048;
214 
215  return 0;
216 }
217 
218 static void put_descr(PutByteContext *pb, int tag, unsigned int size)
219 {
220  int i = 3;
221  bytestream2_put_byte(pb, tag);
222  for (; i > 0; i--)
223  bytestream2_put_byte(pb, (size >> (7 * i)) | 0x80);
224  bytestream2_put_byte(pb, size & 0x7F);
225 }
226 
227 static uint8_t* ffat_get_magic_cookie(AVCodecContext *avctx, UInt32 *cookie_size)
228 {
229  ATDecodeContext *at = avctx->priv_data;
230  if (avctx->codec_id == AV_CODEC_ID_AAC) {
231  char *extradata;
232  PutByteContext pb;
233  *cookie_size = 5 + 3 + 5+13 + 5+at->extradata_size;
234  if (!(extradata = av_malloc(*cookie_size)))
235  return NULL;
236 
237  bytestream2_init_writer(&pb, extradata, *cookie_size);
238 
239  // ES descriptor
240  put_descr(&pb, 0x03, 3 + 5+13 + 5+at->extradata_size);
241  bytestream2_put_be16(&pb, 0);
242  bytestream2_put_byte(&pb, 0x00); // flags (= no flags)
243 
244  // DecoderConfig descriptor
245  put_descr(&pb, 0x04, 13 + 5+at->extradata_size);
246 
247  // Object type indication
248  bytestream2_put_byte(&pb, 0x40);
249 
250  bytestream2_put_byte(&pb, 0x15); // flags (= Audiostream)
251 
252  bytestream2_put_be24(&pb, 0); // Buffersize DB
253 
254  bytestream2_put_be32(&pb, 0); // maxbitrate
255  bytestream2_put_be32(&pb, 0); // avgbitrate
256 
257  // DecoderSpecific info descriptor
258  put_descr(&pb, 0x05, at->extradata_size);
260  return extradata;
261  } else {
262  *cookie_size = at->extradata_size;
263  return at->extradata;
264  }
265 }
266 
268 {
269  ATDecodeContext *at = avctx->priv_data;
270  return at->extradata_size &&
271  (avctx->codec_id == AV_CODEC_ID_ALAC ||
272  avctx->codec_id == AV_CODEC_ID_QDM2 ||
273  avctx->codec_id == AV_CODEC_ID_QDMC ||
274  avctx->codec_id == AV_CODEC_ID_AAC);
275 }
276 
278 {
279  ATDecodeContext *at = avctx->priv_data;
280  if (ffat_usable_extradata(avctx)) {
281  OSStatus status;
282  UInt32 cookie_size;
283  uint8_t *cookie = ffat_get_magic_cookie(avctx, &cookie_size);
284  if (!cookie)
285  return AVERROR(ENOMEM);
286 
287  status = AudioConverterSetProperty(at->converter,
288  kAudioConverterDecompressionMagicCookie,
289  cookie_size, cookie);
290  if (status != 0)
291  av_log(avctx, AV_LOG_WARNING, "AudioToolbox cookie error: %i\n", (int)status);
292 
293  if (cookie != at->extradata)
294  av_free(cookie);
295  }
296  return 0;
297 }
298 
300 {
301  ATDecodeContext *at = avctx->priv_data;
302  OSStatus status;
303  int i;
304 
305  enum AVSampleFormat sample_fmt = (avctx->bits_per_raw_sample == 32) ?
307 
308  AudioStreamBasicDescription in_format = {
309  .mFormatID = ffat_get_format_id(avctx->codec_id, avctx->profile),
310  .mBytesPerPacket = (avctx->codec_id == AV_CODEC_ID_ILBC) ? avctx->block_align : 0,
311  };
312  AudioStreamBasicDescription out_format = {
313  .mFormatID = kAudioFormatLinearPCM,
314  .mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked,
315  .mFramesPerPacket = 1,
316  .mBitsPerChannel = av_get_bytes_per_sample(sample_fmt) * 8,
317  };
318 
319  avctx->sample_fmt = sample_fmt;
320 
321  if (ffat_usable_extradata(avctx)) {
322  UInt32 format_size = sizeof(in_format);
323  UInt32 cookie_size;
324  uint8_t *cookie = ffat_get_magic_cookie(avctx, &cookie_size);
325  if (!cookie)
326  return AVERROR(ENOMEM);
327  status = AudioFormatGetProperty(kAudioFormatProperty_FormatInfo,
328  cookie_size, cookie, &format_size, &in_format);
329  if (cookie != at->extradata)
330  av_free(cookie);
331  if (status != 0) {
332  av_log(avctx, AV_LOG_ERROR, "AudioToolbox header-parse error: %i\n", (int)status);
333  return AVERROR_UNKNOWN;
334  }
335 #if CONFIG_MP1_AT_DECODER || CONFIG_MP2_AT_DECODER || CONFIG_MP3_AT_DECODER
336  } else if (pkt && pkt->size >= 4 &&
337  (avctx->codec_id == AV_CODEC_ID_MP1 ||
338  avctx->codec_id == AV_CODEC_ID_MP2 ||
339  avctx->codec_id == AV_CODEC_ID_MP3)) {
340  enum AVCodecID codec_id;
341  int bit_rate;
342  if (ff_mpa_decode_header(AV_RB32(pkt->data), &avctx->sample_rate,
343  &in_format.mChannelsPerFrame, &avctx->frame_size,
344  &bit_rate, &codec_id) < 0)
345  return AVERROR_INVALIDDATA;
346  avctx->bit_rate = bit_rate;
347  in_format.mSampleRate = avctx->sample_rate;
348 #endif
349 #if CONFIG_AC3_AT_DECODER || CONFIG_EAC3_AT_DECODER
350  } else if (pkt && pkt->size >= 7 &&
351  (avctx->codec_id == AV_CODEC_ID_AC3 ||
352  avctx->codec_id == AV_CODEC_ID_EAC3)) {
353  AC3HeaderInfo hdr;
354  GetBitContext gbc;
355  init_get_bits(&gbc, pkt->data, pkt->size);
356  if (ff_ac3_parse_header(&gbc, &hdr) < 0)
357  return AVERROR_INVALIDDATA;
358  in_format.mSampleRate = hdr.sample_rate;
359  in_format.mChannelsPerFrame = hdr.channels;
360  avctx->frame_size = hdr.num_blocks * 256;
361  avctx->bit_rate = hdr.bit_rate;
362 #endif
363  } else {
364  in_format.mSampleRate = avctx->sample_rate ? avctx->sample_rate : 44100;
365  in_format.mChannelsPerFrame = avctx->channels ? avctx->channels : 1;
366  }
367 
368  avctx->sample_rate = out_format.mSampleRate = in_format.mSampleRate;
369  avctx->channels = out_format.mChannelsPerFrame = in_format.mChannelsPerFrame;
370 
371  if (avctx->codec_id == AV_CODEC_ID_ADPCM_IMA_QT)
372  in_format.mFramesPerPacket = 64;
373 
374  status = AudioConverterNew(&in_format, &out_format, &at->converter);
375 
376  if (status != 0) {
377  av_log(avctx, AV_LOG_ERROR, "AudioToolbox init error: %i\n", (int)status);
378  return AVERROR_UNKNOWN;
379  }
380 
381  if ((status = ffat_set_extradata(avctx)) < 0)
382  return status;
383 
384  for (i = 0; i < (sizeof(at->channel_map) / sizeof(at->channel_map[0])); i++)
385  at->channel_map[i] = i;
386 
387  ffat_update_ctx(avctx);
388 
390  * avctx->frame_size * avctx->channels)))
391  return AVERROR(ENOMEM);
392 
393  at->last_pts = AV_NOPTS_VALUE;
394 
395  return 0;
396 }
397 
399 {
400  ATDecodeContext *at = avctx->priv_data;
401  if (avctx->extradata_size) {
403  if (!at->extradata)
404  return AVERROR(ENOMEM);
405  at->extradata_size = avctx->extradata_size;
406  memcpy(at->extradata, avctx->extradata, avctx->extradata_size);
407  }
408 
409  if ((avctx->channels && avctx->sample_rate) || ffat_usable_extradata(avctx))
410  return ffat_create_decoder(avctx, NULL);
411  else
412  return 0;
413 }
414 
415 static OSStatus ffat_decode_callback(AudioConverterRef converter, UInt32 *nb_packets,
416  AudioBufferList *data,
417  AudioStreamPacketDescription **packets,
418  void *inctx)
419 {
420  AVCodecContext *avctx = inctx;
421  ATDecodeContext *at = avctx->priv_data;
422 
423  if (at->eof) {
424  *nb_packets = 0;
425  if (packets) {
426  *packets = &at->pkt_desc;
427  at->pkt_desc.mDataByteSize = 0;
428  }
429  return 0;
430  }
431 
432  av_packet_unref(&at->in_pkt);
434 
435  if (!at->in_pkt.data) {
436  *nb_packets = 0;
437  return 1;
438  }
439 
440  data->mNumberBuffers = 1;
441  data->mBuffers[0].mNumberChannels = 0;
442  data->mBuffers[0].mDataByteSize = at->in_pkt.size;
443  data->mBuffers[0].mData = at->in_pkt.data;
444  *nb_packets = 1;
445 
446  if (packets) {
447  *packets = &at->pkt_desc;
448  at->pkt_desc.mDataByteSize = at->in_pkt.size;
449  }
450 
451  return 0;
452 }
453 
454 #define COPY_SAMPLES(type) \
455  type *in_ptr = (type*)at->decoded_data; \
456  type *end_ptr = in_ptr + frame->nb_samples * avctx->channels; \
457  type *out_ptr = (type*)frame->data[0]; \
458  for (; in_ptr < end_ptr; in_ptr += avctx->channels, out_ptr += avctx->channels) { \
459  int c; \
460  for (c = 0; c < avctx->channels; c++) \
461  out_ptr[c] = in_ptr[at->channel_map[c]]; \
462  }
463 
465 {
466  ATDecodeContext *at = avctx->priv_data;
467  if (avctx->sample_fmt == AV_SAMPLE_FMT_S32) {
469  } else {
470  COPY_SAMPLES(int16_t);
471  }
472 }
473 
474 static int ffat_decode(AVCodecContext *avctx, void *data,
475  int *got_frame_ptr, AVPacket *avpkt)
476 {
477  ATDecodeContext *at = avctx->priv_data;
478  AVFrame *frame = data;
479  int pkt_size = avpkt->size;
480  OSStatus ret;
481  AudioBufferList out_buffers;
482 
483  if (avctx->codec_id == AV_CODEC_ID_AAC) {
484  if (!at->extradata_size) {
485  uint8_t *side_data;
486  int side_data_size = 0;
487 
489  &side_data_size);
490  if (side_data_size) {
491  at->extradata = av_mallocz(side_data_size + AV_INPUT_BUFFER_PADDING_SIZE);
492  if (!at->extradata)
493  return AVERROR(ENOMEM);
494  at->extradata_size = side_data_size;
495  memcpy(at->extradata, side_data, side_data_size);
496  }
497  }
498  }
499 
500  if (!at->converter) {
501  if ((ret = ffat_create_decoder(avctx, avpkt)) < 0) {
502  return ret;
503  }
504  }
505 
506  out_buffers = (AudioBufferList){
507  .mNumberBuffers = 1,
508  .mBuffers = {
509  {
510  .mNumberChannels = avctx->channels,
511  .mDataByteSize = av_get_bytes_per_sample(avctx->sample_fmt) * avctx->frame_size
512  * avctx->channels,
513  }
514  }
515  };
516 
518 
519  if (avpkt->size) {
520  if ((ret = av_packet_ref(&at->new_in_pkt, avpkt)) < 0) {
521  return ret;
522  }
523  } else {
524  at->eof = 1;
525  }
526 
527  frame->sample_rate = avctx->sample_rate;
528 
529  frame->nb_samples = avctx->frame_size;
530 
531  out_buffers.mBuffers[0].mData = at->decoded_data;
532 
533  ret = AudioConverterFillComplexBuffer(at->converter, ffat_decode_callback, avctx,
534  &frame->nb_samples, &out_buffers, NULL);
535  if ((!ret || ret == 1) && frame->nb_samples) {
536  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
537  return ret;
538  ffat_copy_samples(avctx, frame);
539  *got_frame_ptr = 1;
540  if (at->last_pts != AV_NOPTS_VALUE) {
541  frame->pts = at->last_pts;
542 #if FF_API_PKT_PTS
544  frame->pkt_pts = at->last_pts;
546 #endif
547  at->last_pts = avpkt->pts;
548  }
549  } else if (ret && ret != 1) {
550  av_log(avctx, AV_LOG_WARNING, "Decode error: %i\n", ret);
551  } else {
552  at->last_pts = avpkt->pts;
553  }
554 
555  return pkt_size;
556 }
557 
559 {
560  ATDecodeContext *at = avctx->priv_data;
561  AudioConverterReset(at->converter);
563  av_packet_unref(&at->in_pkt);
564 }
565 
567 {
568  ATDecodeContext *at = avctx->priv_data;
569  if (at->converter)
570  AudioConverterDispose(at->converter);
572  av_packet_unref(&at->in_pkt);
573  av_freep(&at->decoded_data);
574  av_freep(&at->extradata);
575  return 0;
576 }
577 
578 #define FFAT_DEC_CLASS(NAME) \
579  static const AVClass ffat_##NAME##_dec_class = { \
580  .class_name = "at_" #NAME "_dec", \
581  .version = LIBAVUTIL_VERSION_INT, \
582  };
583 
584 #define FFAT_DEC(NAME, ID, bsf_name) \
585  FFAT_DEC_CLASS(NAME) \
586  AVCodec ff_##NAME##_at_decoder = { \
587  .name = #NAME "_at", \
588  .long_name = NULL_IF_CONFIG_SMALL(#NAME " (AudioToolbox)"), \
589  .type = AVMEDIA_TYPE_AUDIO, \
590  .id = ID, \
591  .priv_data_size = sizeof(ATDecodeContext), \
592  .init = ffat_init_decoder, \
593  .close = ffat_close_decoder, \
594  .decode = ffat_decode, \
595  .flush = ffat_decode_flush, \
596  .priv_class = &ffat_##NAME##_dec_class, \
597  .bsfs = bsf_name, \
598  .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY, \
599  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP, \
600  .wrapper_name = "at", \
601  };
602 
603 FFAT_DEC(aac, AV_CODEC_ID_AAC, "aac_adtstoasc")
605 FFAT_DEC(adpcm_ima_qt, AV_CODEC_ID_ADPCM_IMA_QT, NULL)
static uint8_t * ffat_get_magic_cookie(AVCodecContext *avctx, UInt32 *cookie_size)
#define NULL
Definition: coverity.c:32
uint8_t * extradata
int ff_ac3_parse_header(GetBitContext *gbc, AC3HeaderInfo *hdr)
Parse AC-3 frame header.
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:100
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1615
enum AVCodecID codec_id
Definition: qsv.c:77
int size
Definition: avcodec.h:1478
static av_always_inline void bytestream2_init_writer(PutByteContext *p, uint8_t *buf, int buf_size)
Definition: bytestream.h:143
static av_cold int ffat_create_decoder(AVCodecContext *avctx, AVPacket *pkt)
#define AV_CH_LOW_FREQUENCY_2
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:36
void * av_mallocz(size_t size)
Allocate a memory block with alignment suitable for all memory accesses (including vectors if availab...
Definition: mem.c:236
static AVPacket pkt
int bits_per_raw_sample
Bits per sample/pixel of internal libavcodec pixel/sample format.
Definition: avcodec.h:2796
int profile
profile
Definition: avcodec.h:2898
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 format(the sample packing is implied by the sample format) and sample rate.The lists are not just lists
int block_align
number of bytes per packet if constant and known or 0 Used by some WAV based audio codecs...
Definition: avcodec.h:2262
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2233
uint8_t
#define av_cold
Definition: attributes.h:82
#define av_malloc(s)
AVOptions.
#define COPY_SAMPLES(type)
static AudioChannelLayout * ffat_convert_layout(AudioChannelLayout *layout, UInt32 *size)
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
AVClass * av_class
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1666
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
uint8_t * data
Definition: avcodec.h:1477
void av_packet_move_ref(AVPacket *dst, AVPacket *src)
Move every field in src to dst and reset src.
Definition: avpacket.c:655
uint32_t tag
Definition: movenc.c:1496
static av_cold void ffat_decode_flush(AVCodecContext *avctx)
ptrdiff_t size
Definition: opengl_enc.c:100
static void put_descr(PutByteContext *pb, int tag, unsigned int size)
signed 32 bits
Definition: samplefmt.h:62
#define av_log(a,...)
int av_packet_ref(AVPacket *dst, const AVPacket *src)
Setup a new reference to the data described by a given packet.
Definition: avpacket.c:608
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
AVCodecID
Identify the syntax and semantics of the bitstream.
Definition: avcodec.h:215
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
uint8_t * av_packet_get_side_data(const AVPacket *pkt, enum AVPacketSideDataType type, int *size)
Get side information from packet.
Definition: avpacket.c:350
preferred ID for decoding MPEG audio layer 1, 2 or 3
Definition: avcodec.h:565
simple assert() macros that are a bit more flexible than ISO C assert().
Coded AC-3 header values up to the lfeon element, plus derived values.
Definition: ac3.h:177
uint16_t sample_rate
Definition: ac3.h:201
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2276
#define FFAT_DEC(NAME, ID, bsf_name)
static int ffat_get_channel_id(AudioChannelLabel label)
#define b
Definition: input.c:41
static int ffat_update_ctx(AVCodecContext *avctx)
#define ff_ctzll
Definition: intmath.h:126
uint32_t bit_rate
Definition: ac3.h:202
static av_cold int ffat_usable_extradata(AVCodecContext *avctx)
AudioStreamPacketDescription pkt_desc
int32_t
AudioConverterRef converter
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
#define AV_CH_FRONT_CENTER
static av_always_inline unsigned int bytestream2_put_buffer(PutByteContext *p, const uint8_t *src, unsigned int size)
Definition: bytestream.h:282
int frame_size
Number of samples per channel in an audio frame.
Definition: avcodec.h:2245
The AV_PKT_DATA_NEW_EXTRADATA is used to notify the codec or the format that the extradata buffer was...
Definition: avcodec.h:1199
Libavcodec external API header.
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
enum AVCodecID codec_id
Definition: avcodec.h:1575
int sample_rate
samples per second
Definition: avcodec.h:2225
static av_cold int ffat_init_decoder(AVCodecContext *avctx)
main external API structure.
Definition: avcodec.h:1565
void av_packet_unref(AVPacket *pkt)
Wipe the packet.
Definition: avpacket.c:599
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1963
static int ffat_set_extradata(AVCodecContext *avctx)
int extradata_size
Definition: avcodec.h:1667
Describe the class of an AVClass context structure.
Definition: log.h:67
int sample_rate
Sample rate of the audio data.
Definition: frame.h:467
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
they must not be accessed directly The fifo field contains the frames that are queued in the input for processing by the filter The status_in and status_out fields contains the queued status(EOF or error) of the link
mfxU16 profile
Definition: qsvenc.c:44
#define kAudioFormatEnhancedAC3
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)
Return number of bytes per sample.
Definition: samplefmt.c:106
attribute_deprecated int64_t pkt_pts
PTS copied from the AVPacket that was decoded to produce this frame.
Definition: frame.h:396
MPEG Audio header decoder.
static int ffat_decode(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
#define FF_DISABLE_DEPRECATION_WARNINGS
Definition: internal.h:84
common internal api header.
signed 16 bits
Definition: samplefmt.h:61
#define AVERROR_UNKNOWN
Unknown error, typically from an external library.
Definition: error.h:71
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding...
Definition: avcodec.h:790
void * priv_data
Definition: avcodec.h:1592
static OSStatus ffat_decode_callback(AudioConverterRef converter, UInt32 *nb_packets, AudioBufferList *data, AudioStreamPacketDescription **packets, void *inctx)
int ff_mpa_decode_header(uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bit_rate, enum AVCodecID *codec_id)
#define av_free(p)
#define FF_ENABLE_DEPRECATION_WARNINGS
Definition: internal.h:85
int channels
number of audio channels
Definition: avcodec.h:2226
uint8_t channels
Definition: ac3.h:203
static int ffat_compare_channel_descriptions(const void *a, const void *b)
static av_cold int ffat_close_decoder(AVCodecContext *avctx)
int64_t av_get_default_channel_layout(int nb_channels)
Return default channel layout for a given number of channels.
static UInt32 ffat_get_format_id(enum AVCodecID codec, int profile)
#define av_freep(p)
int num_blocks
number of audio blocks
Definition: ac3.h:193
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
static void ffat_copy_samples(AVCodecContext *avctx, AVFrame *frame)
enum AVCodecID id
This structure stores compressed data.
Definition: avcodec.h:1454
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
int64_t pts
Presentation timestamp in AVStream->time_base units; the time at which the decompressed packet will b...
Definition: avcodec.h:1470
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 layout
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248