FFmpeg  4.4.6
aacdec.c
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1 /*
2  * AAC decoder
3  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
5  * Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
6  *
7  * AAC LATM decoder
8  * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
9  * Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
10  *
11  * This file is part of FFmpeg.
12  *
13  * FFmpeg is free software; you can redistribute it and/or
14  * modify it under the terms of the GNU Lesser General Public
15  * License as published by the Free Software Foundation; either
16  * version 2.1 of the License, or (at your option) any later version.
17  *
18  * FFmpeg is distributed in the hope that it will be useful,
19  * but WITHOUT ANY WARRANTY; without even the implied warranty of
20  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
21  * Lesser General Public License for more details.
22  *
23  * You should have received a copy of the GNU Lesser General Public
24  * License along with FFmpeg; if not, write to the Free Software
25  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
26  */
27 
28 /**
29  * @file
30  * AAC decoder
31  * @author Oded Shimon ( ods15 ods15 dyndns org )
32  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
33  */
34 
35 #define FFT_FLOAT 1
36 #define FFT_FIXED_32 0
37 #define USE_FIXED 0
38 
39 #include "libavutil/float_dsp.h"
40 #include "libavutil/opt.h"
41 #include "avcodec.h"
42 #include "internal.h"
43 #include "get_bits.h"
44 #include "fft.h"
45 #include "mdct15.h"
46 #include "lpc.h"
47 #include "kbdwin.h"
48 #include "sinewin.h"
49 
50 #include "aac.h"
51 #include "aactab.h"
52 #include "aacdectab.h"
53 #include "adts_header.h"
54 #include "cbrt_data.h"
55 #include "sbr.h"
56 #include "aacsbr.h"
57 #include "mpeg4audio.h"
58 #include "profiles.h"
59 #include "libavutil/intfloat.h"
60 
61 #include <errno.h>
62 #include <math.h>
63 #include <stdint.h>
64 #include <string.h>
65 
66 #if ARCH_ARM
67 # include "arm/aac.h"
68 #elif ARCH_MIPS
69 # include "mips/aacdec_mips.h"
70 #endif
71 
76 
78 {
79  ps->r0 = 0.0f;
80  ps->r1 = 0.0f;
81  ps->cor0 = 0.0f;
82  ps->cor1 = 0.0f;
83  ps->var0 = 1.0f;
84  ps->var1 = 1.0f;
85 }
86 
87 #ifndef VMUL2
88 static inline float *VMUL2(float *dst, const float *v, unsigned idx,
89  const float *scale)
90 {
91  float s = *scale;
92  *dst++ = v[idx & 15] * s;
93  *dst++ = v[idx>>4 & 15] * s;
94  return dst;
95 }
96 #endif
97 
98 #ifndef VMUL4
99 static inline float *VMUL4(float *dst, const float *v, unsigned idx,
100  const float *scale)
101 {
102  float s = *scale;
103  *dst++ = v[idx & 3] * s;
104  *dst++ = v[idx>>2 & 3] * s;
105  *dst++ = v[idx>>4 & 3] * s;
106  *dst++ = v[idx>>6 & 3] * s;
107  return dst;
108 }
109 #endif
110 
111 #ifndef VMUL2S
112 static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
113  unsigned sign, const float *scale)
114 {
115  union av_intfloat32 s0, s1;
116 
117  s0.f = s1.f = *scale;
118  s0.i ^= sign >> 1 << 31;
119  s1.i ^= sign << 31;
120 
121  *dst++ = v[idx & 15] * s0.f;
122  *dst++ = v[idx>>4 & 15] * s1.f;
123 
124  return dst;
125 }
126 #endif
127 
128 #ifndef VMUL4S
129 static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
130  unsigned sign, const float *scale)
131 {
132  unsigned nz = idx >> 12;
133  union av_intfloat32 s = { .f = *scale };
134  union av_intfloat32 t;
135 
136  t.i = s.i ^ (sign & 1U<<31);
137  *dst++ = v[idx & 3] * t.f;
138 
139  sign <<= nz & 1; nz >>= 1;
140  t.i = s.i ^ (sign & 1U<<31);
141  *dst++ = v[idx>>2 & 3] * t.f;
142 
143  sign <<= nz & 1; nz >>= 1;
144  t.i = s.i ^ (sign & 1U<<31);
145  *dst++ = v[idx>>4 & 3] * t.f;
146 
147  sign <<= nz & 1;
148  t.i = s.i ^ (sign & 1U<<31);
149  *dst++ = v[idx>>6 & 3] * t.f;
150 
151  return dst;
152 }
153 #endif
154 
155 static av_always_inline float flt16_round(float pf)
156 {
157  union av_intfloat32 tmp;
158  tmp.f = pf;
159  tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
160  return tmp.f;
161 }
162 
163 static av_always_inline float flt16_even(float pf)
164 {
165  union av_intfloat32 tmp;
166  tmp.f = pf;
167  tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
168  return tmp.f;
169 }
170 
171 static av_always_inline float flt16_trunc(float pf)
172 {
173  union av_intfloat32 pun;
174  pun.f = pf;
175  pun.i &= 0xFFFF0000U;
176  return pun.f;
177 }
178 
179 static av_always_inline void predict(PredictorState *ps, float *coef,
180  int output_enable)
181 {
182  const float a = 0.953125; // 61.0 / 64
183  const float alpha = 0.90625; // 29.0 / 32
184  float e0, e1;
185  float pv;
186  float k1, k2;
187  float r0 = ps->r0, r1 = ps->r1;
188  float cor0 = ps->cor0, cor1 = ps->cor1;
189  float var0 = ps->var0, var1 = ps->var1;
190 
191  k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
192  k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
193 
194  pv = flt16_round(k1 * r0 + k2 * r1);
195  if (output_enable)
196  *coef += pv;
197 
198  e0 = *coef;
199  e1 = e0 - k1 * r0;
200 
201  ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
202  ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
203  ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
204  ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
205 
206  ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
207  ps->r0 = flt16_trunc(a * e0);
208 }
209 
210 /**
211  * Apply dependent channel coupling (applied before IMDCT).
212  *
213  * @param index index into coupling gain array
214  */
216  SingleChannelElement *target,
217  ChannelElement *cce, int index)
218 {
219  IndividualChannelStream *ics = &cce->ch[0].ics;
220  const uint16_t *offsets = ics->swb_offset;
221  float *dest = target->coeffs;
222  const float *src = cce->ch[0].coeffs;
223  int g, i, group, k, idx = 0;
224  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
226  "Dependent coupling is not supported together with LTP\n");
227  return;
228  }
229  for (g = 0; g < ics->num_window_groups; g++) {
230  for (i = 0; i < ics->max_sfb; i++, idx++) {
231  if (cce->ch[0].band_type[idx] != ZERO_BT) {
232  const float gain = cce->coup.gain[index][idx];
233  for (group = 0; group < ics->group_len[g]; group++) {
234  for (k = offsets[i]; k < offsets[i + 1]; k++) {
235  // FIXME: SIMDify
236  dest[group * 128 + k] += gain * src[group * 128 + k];
237  }
238  }
239  }
240  }
241  dest += ics->group_len[g] * 128;
242  src += ics->group_len[g] * 128;
243  }
244 }
245 
246 /**
247  * Apply independent channel coupling (applied after IMDCT).
248  *
249  * @param index index into coupling gain array
250  */
252  SingleChannelElement *target,
253  ChannelElement *cce, int index)
254 {
255  const float gain = cce->coup.gain[index][0];
256  const float *src = cce->ch[0].ret;
257  float *dest = target->ret;
258  const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
259 
260  ac->fdsp->vector_fmac_scalar(dest, src, gain, len);
261 }
262 
263 #include "aacdec_template.c"
264 
265 #define LOAS_SYNC_WORD 0x2b7 ///< 11 bits LOAS sync word
266 
267 struct LATMContext {
268  AACContext aac_ctx; ///< containing AACContext
269  int initialized; ///< initialized after a valid extradata was seen
270 
271  // parser data
272  int audio_mux_version_A; ///< LATM syntax version
273  int frame_length_type; ///< 0/1 variable/fixed frame length
274  int frame_length; ///< frame length for fixed frame length
275 };
276 
277 static inline uint32_t latm_get_value(GetBitContext *b)
278 {
279  int length = get_bits(b, 2);
280 
281  return get_bits_long(b, (length+1)*8);
282 }
283 
285  GetBitContext *gb, int asclen)
286 {
287  AACContext *ac = &latmctx->aac_ctx;
288  AVCodecContext *avctx = ac->avctx;
289  MPEG4AudioConfig m4ac = { 0 };
290  GetBitContext gbc;
291  int config_start_bit = get_bits_count(gb);
292  int sync_extension = 0;
293  int bits_consumed, esize, i;
294 
295  if (asclen > 0) {
296  sync_extension = 1;
297  asclen = FFMIN(asclen, get_bits_left(gb));
298  init_get_bits(&gbc, gb->buffer, config_start_bit + asclen);
299  skip_bits_long(&gbc, config_start_bit);
300  } else if (asclen == 0) {
301  gbc = *gb;
302  } else {
303  return AVERROR_INVALIDDATA;
304  }
305 
306  if (get_bits_left(gb) <= 0)
307  return AVERROR_INVALIDDATA;
308 
309  bits_consumed = decode_audio_specific_config_gb(NULL, avctx, &m4ac,
310  &gbc, config_start_bit,
311  sync_extension);
312 
313  if (bits_consumed < config_start_bit)
314  return AVERROR_INVALIDDATA;
315  bits_consumed -= config_start_bit;
316 
317  if (asclen == 0)
318  asclen = bits_consumed;
319 
320  if (!latmctx->initialized ||
321  ac->oc[1].m4ac.sample_rate != m4ac.sample_rate ||
322  ac->oc[1].m4ac.chan_config != m4ac.chan_config) {
323 
324  if (latmctx->initialized) {
325  av_log(avctx, AV_LOG_INFO, "audio config changed (sample_rate=%d, chan_config=%d)\n", m4ac.sample_rate, m4ac.chan_config);
326  } else {
327  av_log(avctx, AV_LOG_DEBUG, "initializing latmctx\n");
328  }
329  latmctx->initialized = 0;
330 
331  esize = (asclen + 7) / 8;
332 
333  if (avctx->extradata_size < esize) {
334  av_free(avctx->extradata);
336  if (!avctx->extradata)
337  return AVERROR(ENOMEM);
338  }
339 
340  avctx->extradata_size = esize;
341  gbc = *gb;
342  for (i = 0; i < esize; i++) {
343  avctx->extradata[i] = get_bits(&gbc, 8);
344  }
345  memset(avctx->extradata+esize, 0, AV_INPUT_BUFFER_PADDING_SIZE);
346  }
347  skip_bits_long(gb, asclen);
348 
349  return 0;
350 }
351 
352 static int read_stream_mux_config(struct LATMContext *latmctx,
353  GetBitContext *gb)
354 {
355  int ret, audio_mux_version = get_bits(gb, 1);
356 
357  latmctx->audio_mux_version_A = 0;
358  if (audio_mux_version)
359  latmctx->audio_mux_version_A = get_bits(gb, 1);
360 
361  if (!latmctx->audio_mux_version_A) {
362 
363  if (audio_mux_version)
364  latm_get_value(gb); // taraFullness
365 
366  skip_bits(gb, 1); // allStreamSameTimeFraming
367  skip_bits(gb, 6); // numSubFrames
368  // numPrograms
369  if (get_bits(gb, 4)) { // numPrograms
370  avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple programs");
371  return AVERROR_PATCHWELCOME;
372  }
373 
374  // for each program (which there is only one in DVB)
375 
376  // for each layer (which there is only one in DVB)
377  if (get_bits(gb, 3)) { // numLayer
378  avpriv_request_sample(latmctx->aac_ctx.avctx, "Multiple layers");
379  return AVERROR_PATCHWELCOME;
380  }
381 
382  // for all but first stream: use_same_config = get_bits(gb, 1);
383  if (!audio_mux_version) {
384  if ((ret = latm_decode_audio_specific_config(latmctx, gb, 0)) < 0)
385  return ret;
386  } else {
387  int ascLen = latm_get_value(gb);
388  if ((ret = latm_decode_audio_specific_config(latmctx, gb, ascLen)) < 0)
389  return ret;
390  }
391 
392  latmctx->frame_length_type = get_bits(gb, 3);
393  switch (latmctx->frame_length_type) {
394  case 0:
395  skip_bits(gb, 8); // latmBufferFullness
396  break;
397  case 1:
398  latmctx->frame_length = get_bits(gb, 9);
399  break;
400  case 3:
401  case 4:
402  case 5:
403  skip_bits(gb, 6); // CELP frame length table index
404  break;
405  case 6:
406  case 7:
407  skip_bits(gb, 1); // HVXC frame length table index
408  break;
409  }
410 
411  if (get_bits(gb, 1)) { // other data
412  if (audio_mux_version) {
413  latm_get_value(gb); // other_data_bits
414  } else {
415  int esc;
416  do {
417  if (get_bits_left(gb) < 9)
418  return AVERROR_INVALIDDATA;
419  esc = get_bits(gb, 1);
420  skip_bits(gb, 8);
421  } while (esc);
422  }
423  }
424 
425  if (get_bits(gb, 1)) // crc present
426  skip_bits(gb, 8); // config_crc
427  }
428 
429  return 0;
430 }
431 
433 {
434  uint8_t tmp;
435 
436  if (ctx->frame_length_type == 0) {
437  int mux_slot_length = 0;
438  do {
439  if (get_bits_left(gb) < 8)
440  return AVERROR_INVALIDDATA;
441  tmp = get_bits(gb, 8);
442  mux_slot_length += tmp;
443  } while (tmp == 255);
444  return mux_slot_length;
445  } else if (ctx->frame_length_type == 1) {
446  return ctx->frame_length;
447  } else if (ctx->frame_length_type == 3 ||
448  ctx->frame_length_type == 5 ||
449  ctx->frame_length_type == 7) {
450  skip_bits(gb, 2); // mux_slot_length_coded
451  }
452  return 0;
453 }
454 
455 static int read_audio_mux_element(struct LATMContext *latmctx,
456  GetBitContext *gb)
457 {
458  int err;
459  uint8_t use_same_mux = get_bits(gb, 1);
460  if (!use_same_mux) {
461  if ((err = read_stream_mux_config(latmctx, gb)) < 0)
462  return err;
463  } else if (!latmctx->aac_ctx.avctx->extradata) {
464  av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
465  "no decoder config found\n");
466  return 1;
467  }
468  if (latmctx->audio_mux_version_A == 0) {
469  int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
470  if (mux_slot_length_bytes < 0 || mux_slot_length_bytes * 8LL > get_bits_left(gb)) {
471  av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
472  return AVERROR_INVALIDDATA;
473  } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
474  av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
475  "frame length mismatch %d << %d\n",
476  mux_slot_length_bytes * 8, get_bits_left(gb));
477  return AVERROR_INVALIDDATA;
478  }
479  }
480  return 0;
481 }
482 
483 
484 static int latm_decode_frame(AVCodecContext *avctx, void *out,
485  int *got_frame_ptr, AVPacket *avpkt)
486 {
487  struct LATMContext *latmctx = avctx->priv_data;
488  int muxlength, err;
489  GetBitContext gb;
490 
491  if ((err = init_get_bits8(&gb, avpkt->data, avpkt->size)) < 0)
492  return err;
493 
494  // check for LOAS sync word
495  if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
496  return AVERROR_INVALIDDATA;
497 
498  muxlength = get_bits(&gb, 13) + 3;
499  // not enough data, the parser should have sorted this out
500  if (muxlength > avpkt->size)
501  return AVERROR_INVALIDDATA;
502 
503  if ((err = read_audio_mux_element(latmctx, &gb)))
504  return (err < 0) ? err : avpkt->size;
505 
506  if (!latmctx->initialized) {
507  if (!avctx->extradata) {
508  *got_frame_ptr = 0;
509  return avpkt->size;
510  } else {
512  if ((err = decode_audio_specific_config(
513  &latmctx->aac_ctx, avctx, &latmctx->aac_ctx.oc[1].m4ac,
514  avctx->extradata, avctx->extradata_size*8LL, 1)) < 0) {
516  return err;
517  }
518  latmctx->initialized = 1;
519  }
520  }
521 
522  if (show_bits(&gb, 12) == 0xfff) {
523  av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
524  "ADTS header detected, probably as result of configuration "
525  "misparsing\n");
526  return AVERROR_INVALIDDATA;
527  }
528 
529  switch (latmctx->aac_ctx.oc[1].m4ac.object_type) {
530  case AOT_ER_AAC_LC:
531  case AOT_ER_AAC_LTP:
532  case AOT_ER_AAC_LD:
533  case AOT_ER_AAC_ELD:
534  err = aac_decode_er_frame(avctx, out, got_frame_ptr, &gb);
535  break;
536  default:
537  err = aac_decode_frame_int(avctx, out, got_frame_ptr, &gb, avpkt);
538  }
539  if (err < 0)
540  return err;
541 
542  return muxlength;
543 }
544 
546 {
547  struct LATMContext *latmctx = avctx->priv_data;
548  int ret = aac_decode_init(avctx);
549 
550  if (avctx->extradata_size > 0)
551  latmctx->initialized = !ret;
552 
553  return ret;
554 }
555 
557  .name = "aac",
558  .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
559  .type = AVMEDIA_TYPE_AUDIO,
560  .id = AV_CODEC_ID_AAC,
561  .priv_data_size = sizeof(AACContext),
563  .close = aac_decode_close,
565  .sample_fmts = (const enum AVSampleFormat[]) {
567  },
570  .channel_layouts = aac_channel_layout,
571  .flush = flush,
572  .priv_class = &aac_decoder_class,
574 };
575 
576 /*
577  Note: This decoder filter is intended to decode LATM streams transferred
578  in MPEG transport streams which only contain one program.
579  To do a more complex LATM demuxing a separate LATM demuxer should be used.
580 */
582  .name = "aac_latm",
583  .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Coding LATM syntax)"),
584  .type = AVMEDIA_TYPE_AUDIO,
585  .id = AV_CODEC_ID_AAC_LATM,
586  .priv_data_size = sizeof(struct LATMContext),
588  .close = aac_decode_close,
589  .decode = latm_decode_frame,
590  .sample_fmts = (const enum AVSampleFormat[]) {
592  },
595  .channel_layouts = aac_channel_layout,
596  .flush = flush,
598 };
@ ZERO_BT
Scalefactors and spectral data are all zero.
Definition: aac.h:84
#define AAC_RENAME(x)
Definition: aac_defines.h:85
float INTFLOAT
Definition: aac_defines.h:88
Reference: libavcodec/aacdec.c.
AAC decoder.
static int aac_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
static int aac_decode_er_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb)
static int decode_audio_specific_config_gb(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, GetBitContext *gb, int get_bit_alignment, int sync_extension)
Decode audio specific configuration; reference: table 1.13.
static int aac_decode_frame_int(AVCodecContext *avctx, void *data, int *got_frame_ptr, GetBitContext *gb, const AVPacket *avpkt)
static av_cold int aac_decode_close(AVCodecContext *avctx)
static av_cold int aac_decode_init(AVCodecContext *avctx)
static int decode_audio_specific_config(AACContext *ac, AVCodecContext *avctx, MPEG4AudioConfig *m4ac, const uint8_t *data, int64_t bit_size, int sync_extension)
static int push_output_configuration(AACContext *ac)
Save current output configuration if and only if it has been locked.
static void pop_output_configuration(AACContext *ac)
Restore the previous output configuration if and only if the current configuration is unlocked.
static const AVClass aac_decoder_class
static void flush(AVCodecContext *avctx)
AAC decoder data.
static const uint64_t aac_channel_layout[16]
Definition: aacdectab.h:75
AAC Spectral Band Replication function declarations.
AAC data declarations.
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:925
#define av_always_inline
Definition: attributes.h:45
#define av_cold
Definition: attributes.h:88
uint8_t
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
#define s(width, name)
Definition: cbs_vp9.c:257
#define FFMIN(a, b)
Definition: common.h:105
#define NULL
Definition: coverity.c:32
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
bitstream reader API header.
static unsigned int get_bits_long(GetBitContext *s, int n)
Read 0-32 bits.
Definition: get_bits.h:546
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
static void skip_bits_long(GetBitContext *s, int n)
Skips the specified number of bits.
Definition: get_bits.h:291
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
static int init_get_bits8(GetBitContext *s, const uint8_t *buffer, int byte_size)
Initialize GetBitContext.
Definition: get_bits.h:677
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
static unsigned int show_bits(GetBitContext *s, int n)
Show 1-25 bits.
Definition: get_bits.h:446
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition: codec.h:104
@ AV_CODEC_ID_AAC
Definition: codec_id.h:426
@ AV_CODEC_ID_AAC_LATM
Definition: codec_id.h:473
#define AV_INPUT_BUFFER_PADDING_SIZE
Required number of additionally allocated bytes at the end of the input bitstream for decoding.
Definition: avcodec.h:215
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AVERROR(e)
Definition: error.h:43
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
#define AV_LOG_INFO
Standard information.
Definition: log.h:205
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:117
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
int index
Definition: gxfenc.c:89
static const int offsets[]
Definition: hevc_pel.c:34
static const int16_t alpha[]
Definition: ilbcdata.h:55
int i
Definition: input.c:407
static int read_audio_mux_element(struct LATMContext *latmctx, GetBitContext *gb)
Definition: aacdec.c:455
static av_always_inline void reset_predict_state(PredictorState *ps)
Definition: aacdec.c:77
static av_cold int latm_decode_init(AVCodecContext *avctx)
Definition: aacdec.c:545
static av_always_inline float flt16_even(float pf)
Definition: aacdec.c:163
static float * VMUL4S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:129
static uint32_t latm_get_value(GetBitContext *b)
Definition: aacdec.c:277
#define LOAS_SYNC_WORD
11 bits LOAS sync word
Definition: aacdec.c:265
static void apply_dependent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply dependent channel coupling (applied before IMDCT).
Definition: aacdec.c:215
static float * VMUL4(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:99
static INTFLOAT aac_kbd_short_120[120]
Definition: aacdec.c:75
AVCodec ff_aac_decoder
Definition: aacdec.c:556
static av_always_inline void predict(PredictorState *ps, float *coef, int output_enable)
Definition: aacdec.c:179
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
Definition: aacdec.c:432
static void apply_independent_coupling(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply independent channel coupling (applied after IMDCT).
Definition: aacdec.c:251
static int read_stream_mux_config(struct LATMContext *latmctx, GetBitContext *gb)
Definition: aacdec.c:352
static int latm_decode_audio_specific_config(struct LATMContext *latmctx, GetBitContext *gb, int asclen)
Definition: aacdec.c:284
static INTFLOAT aac_kbd_long_960[960]
Definition: aacdec.c:74
static INTFLOAT sine_960[960]
Definition: aacdec.c:73
AVCodec ff_aac_latm_decoder
Definition: aacdec.c:581
static av_always_inline float flt16_round(float pf)
Definition: aacdec.c:155
static INTFLOAT sine_120[120]
Definition: aacdec.c:72
static float * VMUL2(float *dst, const float *v, unsigned idx, const float *scale)
Definition: aacdec.c:88
static float * VMUL2S(float *dst, const float *v, unsigned idx, unsigned sign, const float *scale)
Definition: aacdec.c:112
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *got_frame_ptr, AVPacket *avpkt)
Definition: aacdec.c:484
static av_always_inline float flt16_trunc(float pf)
Definition: aacdec.c:171
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:41
#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
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
@ AOT_AAC_LTP
Y Long Term Prediction.
Definition: mpeg4audio.h:93
@ AOT_ER_AAC_LD
N Error Resilient Low Delay.
Definition: mpeg4audio.h:109
@ AOT_ER_AAC_ELD
N Error Resilient Enhanced Low Delay.
Definition: mpeg4audio.h:125
@ AOT_ER_AAC_LC
N Error Resilient Low Complexity.
Definition: mpeg4audio.h:104
@ AOT_ER_AAC_LTP
N Error Resilient Long Term Prediction.
Definition: mpeg4audio.h:105
AVOptions.
const AVProfile ff_aac_profiles[]
Definition: profiles.c:26
#define s1
Definition: regdef.h:38
#define s0
Definition: regdef.h:37
#define pv
Definition: regdef.h:60
Spectral Band Replication definitions and structures.
main AAC context
Definition: aac.h:294
AVCodecContext * avctx
Definition: aac.h:296
OutputConfiguration oc[2]
Definition: aac.h:357
AVFloatDSPContext * fdsp
Definition: aac.h:334
main external API structure.
Definition: avcodec.h:536
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:637
int extradata_size
Definition: avcodec.h:638
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
void(* vector_fmac_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float and add to destination vector.
Definition: float_dsp.h:54
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
INTFLOAT gain[16][120]
Definition: aac.h:243
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:276
ChannelCoupling coup
Definition: aac.h:287
SingleChannelElement ch[2]
Definition: aac.h:285
const uint8_t * buffer
Definition: get_bits.h:62
Individual Channel Stream.
Definition: aac.h:175
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:176
uint8_t group_len[8]
Definition: aac.h:180
const uint16_t * swb_offset
table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular wind...
Definition: aac.h:182
int audio_mux_version_A
LATM syntax version.
Definition: aacdec.c:272
AACContext aac_ctx
containing AACContext
Definition: aacdec.c:268
int frame_length_type
0/1 variable/fixed frame length
Definition: aacdec.c:273
int initialized
initialized after a valid extradata was seen
Definition: aacdec.c:269
int frame_length
frame length for fixed frame length
Definition: aacdec.c:274
int sbr
-1 implicit, 1 presence
Definition: mpeg4audio.h:38
MPEG4AudioConfig m4ac
Definition: aac.h:125
Predictor State.
Definition: aac.h:136
AAC_FLOAT cor1
Definition: aac.h:138
AAC_FLOAT r1
Definition: aac.h:142
AAC_FLOAT var1
Definition: aac.h:140
AAC_FLOAT var0
Definition: aac.h:139
AAC_FLOAT r0
Definition: aac.h:141
AAC_FLOAT cor0
Definition: aac.h:137
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:249
INTFLOAT * ret
PCM output.
Definition: aac.h:270
enum BandType band_type[128]
band types
Definition: aac.h:253
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:263
IndividualChannelStream ics
Definition: aac.h:250
#define av_free(p)
#define avpriv_request_sample(...)
#define av_malloc(s)
#define av_log(a,...)
static uint8_t tmp[11]
Definition: aes_ctr.c:27
#define src
Definition: vp8dsp.c:255
FILE * out
Definition: movenc.c:54
AVFormatContext * ctx
Definition: movenc.c:48
uint32_t i
Definition: intfloat.h:28
const char * b
Definition: vf_curves.c:118
const char * g
Definition: vf_curves.c:117
int len