FFmpeg
aacdec_fixed.c
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1 /*
2  * Copyright (c) 2013
3  * MIPS Technologies, Inc., California.
4  *
5  * Redistribution and use in source and binary forms, with or without
6  * modification, are permitted provided that the following conditions
7  * are met:
8  * 1. Redistributions of source code must retain the above copyright
9  * notice, this list of conditions and the following disclaimer.
10  * 2. Redistributions in binary form must reproduce the above copyright
11  * notice, this list of conditions and the following disclaimer in the
12  * documentation and/or other materials provided with the distribution.
13  * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
14  * contributors may be used to endorse or promote products derived from
15  * this software without specific prior written permission.
16  *
17  * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
18  * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
19  * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
20  * ARE DISCLAIMED. IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
21  * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
22  * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
23  * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
24  * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
25  * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
26  * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
27  * SUCH DAMAGE.
28  *
29  * AAC decoder fixed-point implementation
30  *
31  * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
32  * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
33  *
34  * This file is part of FFmpeg.
35  *
36  * FFmpeg is free software; you can redistribute it and/or
37  * modify it under the terms of the GNU Lesser General Public
38  * License as published by the Free Software Foundation; either
39  * version 2.1 of the License, or (at your option) any later version.
40  *
41  * FFmpeg is distributed in the hope that it will be useful,
42  * but WITHOUT ANY WARRANTY; without even the implied warranty of
43  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
44  * Lesser General Public License for more details.
45  *
46  * You should have received a copy of the GNU Lesser General Public
47  * License along with FFmpeg; if not, write to the Free Software
48  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
49  */
50 
51 /**
52  * @file
53  * AAC decoder
54  * @author Oded Shimon ( ods15 ods15 dyndns org )
55  * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
56  *
57  * Fixed point implementation
58  * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
59  */
60 
61 #define USE_FIXED 1
62 #define TX_TYPE AV_TX_INT32_MDCT
63 
64 #include "libavutil/fixed_dsp.h"
65 #include "libavutil/opt.h"
66 #include "avcodec.h"
67 #include "codec_internal.h"
68 #include "get_bits.h"
69 #include "lpc.h"
70 #include "kbdwin.h"
71 #include "sinewin_fixed_tablegen.h"
72 
73 #include "aac.h"
74 #include "aactab.h"
75 #include "aacdectab.h"
76 #include "adts_header.h"
77 #include "cbrt_data.h"
78 #include "sbr.h"
79 #include "aacsbr.h"
80 #include "mpeg4audio.h"
81 #include "profiles.h"
82 #include "libavutil/intfloat.h"
83 
84 #include <math.h>
85 #include <string.h>
86 
87 DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_long_1024))[1024];
88 DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_short_128))[128];
91 
93 {
94  ps->r0.mant = 0;
95  ps->r0.exp = 0;
96  ps->r1.mant = 0;
97  ps->r1.exp = 0;
98  ps->cor0.mant = 0;
99  ps->cor0.exp = 0;
100  ps->cor1.mant = 0;
101  ps->cor1.exp = 0;
102  ps->var0.mant = 0x20000000;
103  ps->var0.exp = 1;
104  ps->var1.mant = 0x20000000;
105  ps->var1.exp = 1;
106 }
107 
108 static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) }; // 2^0, 2^0.25, 2^0.5, 2^0.75
109 
110 static inline int *DEC_SPAIR(int *dst, unsigned idx)
111 {
112  dst[0] = (idx & 15) - 4;
113  dst[1] = (idx >> 4 & 15) - 4;
114 
115  return dst + 2;
116 }
117 
118 static inline int *DEC_SQUAD(int *dst, unsigned idx)
119 {
120  dst[0] = (idx & 3) - 1;
121  dst[1] = (idx >> 2 & 3) - 1;
122  dst[2] = (idx >> 4 & 3) - 1;
123  dst[3] = (idx >> 6 & 3) - 1;
124 
125  return dst + 4;
126 }
127 
128 static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
129 {
130  dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
131  dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
132 
133  return dst + 2;
134 }
135 
136 static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
137 {
138  unsigned nz = idx >> 12;
139 
140  dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
141  sign <<= nz & 1;
142  nz >>= 1;
143  dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
144  sign <<= nz & 1;
145  nz >>= 1;
146  dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
147  sign <<= nz & 1;
148  nz >>= 1;
149  dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
150 
151  return dst + 4;
152 }
153 
154 static void vector_pow43(int *coefs, int len)
155 {
156  int i, coef;
157 
158  for (i=0; i<len; i++) {
159  coef = coefs[i];
160  if (coef < 0)
161  coef = -(int)ff_cbrt_tab_fixed[(-coef) & 8191];
162  else
163  coef = (int)ff_cbrt_tab_fixed[ coef & 8191];
164  coefs[i] = coef;
165  }
166 }
167 
168 static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
169 {
170  int ssign = scale < 0 ? -1 : 1;
171  int s = FFABS(scale);
172  unsigned int round;
173  int i, out, c = exp2tab[s & 3];
174 
175  s = offset - (s >> 2);
176 
177  if (s > 31) {
178  for (i=0; i<len; i++) {
179  dst[i] = 0;
180  }
181  } else if (s > 0) {
182  round = 1 << (s-1);
183  for (i=0; i<len; i++) {
184  out = (int)(((int64_t)src[i] * c) >> 32);
185  dst[i] = ((int)(out+round) >> s) * ssign;
186  }
187  } else if (s > -32) {
188  s = s + 32;
189  round = 1U << (s-1);
190  for (i=0; i<len; i++) {
191  out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
192  dst[i] = out * (unsigned)ssign;
193  }
194  } else {
195  av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
196  }
197 }
198 
199 static void noise_scale(int *coefs, int scale, int band_energy, int len)
200 {
201  int s = -scale;
202  unsigned int round;
203  int i, out, c = exp2tab[s & 3];
204  int nlz = 0;
205 
206  av_assert0(s >= 0);
207  while (band_energy > 0x7fff) {
208  band_energy >>= 1;
209  nlz++;
210  }
211  c /= band_energy;
212  s = 21 + nlz - (s >> 2);
213 
214  if (s > 31) {
215  for (i=0; i<len; i++) {
216  coefs[i] = 0;
217  }
218  } else if (s >= 0) {
219  round = s ? 1 << (s-1) : 0;
220  for (i=0; i<len; i++) {
221  out = (int)(((int64_t)coefs[i] * c) >> 32);
222  coefs[i] = -((int)(out+round) >> s);
223  }
224  }
225  else {
226  s = s + 32;
227  if (s > 0) {
228  round = 1 << (s-1);
229  for (i=0; i<len; i++) {
230  out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
231  coefs[i] = -out;
232  }
233  } else {
234  for (i=0; i<len; i++)
235  coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
236  }
237  }
238 }
239 
241 {
242  SoftFloat tmp;
243  int s;
244 
245  tmp.exp = pf.exp;
246  s = pf.mant >> 31;
247  tmp.mant = (pf.mant ^ s) - s;
248  tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
249  tmp.mant = (tmp.mant ^ s) - s;
250 
251  return tmp;
252 }
253 
255 {
256  SoftFloat tmp;
257  int s;
258 
259  tmp.exp = pf.exp;
260  s = pf.mant >> 31;
261  tmp.mant = (pf.mant ^ s) - s;
262  tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
263  tmp.mant = (tmp.mant ^ s) - s;
264 
265  return tmp;
266 }
267 
269 {
270  SoftFloat pun;
271  int s;
272 
273  pun.exp = pf.exp;
274  s = pf.mant >> 31;
275  pun.mant = (pf.mant ^ s) - s;
276  pun.mant = pun.mant & 0xFFC00000U;
277  pun.mant = (pun.mant ^ s) - s;
278 
279  return pun;
280 }
281 
282 static av_always_inline void predict(PredictorState *ps, int *coef,
283  int output_enable)
284 {
285  const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
286  const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
287  SoftFloat e0, e1;
288  SoftFloat pv;
289  SoftFloat k1, k2;
290  SoftFloat r0 = ps->r0, r1 = ps->r1;
291  SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
292  SoftFloat var0 = ps->var0, var1 = ps->var1;
293  SoftFloat tmp;
294 
295  if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
296  k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
297  }
298  else {
299  k1.mant = 0;
300  k1.exp = 0;
301  }
302 
303  if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
304  k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
305  }
306  else {
307  k2.mant = 0;
308  k2.exp = 0;
309  }
310 
311  tmp = av_mul_sf(k1, r0);
312  pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
313  if (output_enable) {
314  int shift = 28 - pv.exp;
315 
316  if (shift < 31) {
317  if (shift > 0) {
318  *coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
319  } else
320  *coef += (unsigned)pv.mant << -shift;
321  }
322  }
323 
324  e0 = av_int2sf(*coef, 2);
325  e1 = av_sub_sf(e0, tmp);
326 
327  ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
328  tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
329  tmp.exp--;
330  ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
331  ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
332  tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
333  tmp.exp--;
334  ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
335 
336  ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
337  ps->r0 = flt16_trunc(av_mul_sf(a, e0));
338 }
339 
340 
341 static const int cce_scale_fixed[8] = {
342  Q30(1.0), //2^(0/8)
343  Q30(1.0905077327), //2^(1/8)
344  Q30(1.1892071150), //2^(2/8)
345  Q30(1.2968395547), //2^(3/8)
346  Q30(1.4142135624), //2^(4/8)
347  Q30(1.5422108254), //2^(5/8)
348  Q30(1.6817928305), //2^(6/8)
349  Q30(1.8340080864), //2^(7/8)
350 };
351 
352 /**
353  * Apply dependent channel coupling (applied before IMDCT).
354  *
355  * @param index index into coupling gain array
356  */
358  SingleChannelElement *target,
359  ChannelElement *cce, int index)
360 {
361  IndividualChannelStream *ics = &cce->ch[0].ics;
362  const uint16_t *offsets = ics->swb_offset;
363  int *dest = target->coeffs;
364  const int *src = cce->ch[0].coeffs;
365  int g, i, group, k, idx = 0;
366  if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
368  "Dependent coupling is not supported together with LTP\n");
369  return;
370  }
371  for (g = 0; g < ics->num_window_groups; g++) {
372  for (i = 0; i < ics->max_sfb; i++, idx++) {
373  if (cce->ch[0].band_type[idx] != ZERO_BT) {
374  const int gain = cce->coup.gain[index][idx];
375  int shift, round, c, tmp;
376 
377  if (gain < 0) {
378  c = -cce_scale_fixed[-gain & 7];
379  shift = (-gain-1024) >> 3;
380  }
381  else {
382  c = cce_scale_fixed[gain & 7];
383  shift = (gain-1024) >> 3;
384  }
385 
386  if (shift < -31) {
387  // Nothing to do
388  } else if (shift < 0) {
389  shift = -shift;
390  round = 1 << (shift - 1);
391 
392  for (group = 0; group < ics->group_len[g]; group++) {
393  for (k = offsets[i]; k < offsets[i + 1]; k++) {
394  tmp = (int)(((int64_t)src[group * 128 + k] * c + \
395  (int64_t)0x1000000000) >> 37);
396  dest[group * 128 + k] += (tmp + (int64_t)round) >> shift;
397  }
398  }
399  }
400  else {
401  for (group = 0; group < ics->group_len[g]; group++) {
402  for (k = offsets[i]; k < offsets[i + 1]; k++) {
403  tmp = (int)(((int64_t)src[group * 128 + k] * c + \
404  (int64_t)0x1000000000) >> 37);
405  dest[group * 128 + k] += tmp * (1U << shift);
406  }
407  }
408  }
409  }
410  }
411  dest += ics->group_len[g] * 128;
412  src += ics->group_len[g] * 128;
413  }
414 }
415 
416 /**
417  * Apply independent channel coupling (applied after IMDCT).
418  *
419  * @param index index into coupling gain array
420  */
422  SingleChannelElement *target,
423  ChannelElement *cce, int index)
424 {
425  int i, c, shift, round, tmp;
426  const int gain = cce->coup.gain[index][0];
427  const int *src = cce->ch[0].ret;
428  unsigned int *dest = target->ret;
429  const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
430 
431  c = cce_scale_fixed[gain & 7];
432  shift = (gain-1024) >> 3;
433  if (shift < -31) {
434  return;
435  } else if (shift < 0) {
436  shift = -shift;
437  round = 1 << (shift - 1);
438 
439  for (i = 0; i < len; i++) {
440  tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
441  dest[i] += (tmp + round) >> shift;
442  }
443  }
444  else {
445  for (i = 0; i < len; i++) {
446  tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
447  dest[i] += tmp * (1U << shift);
448  }
449  }
450 }
451 
452 #include "aacdec_template.c"
453 
455  .p.name = "aac_fixed",
456  CODEC_LONG_NAME("AAC (Advanced Audio Coding)"),
457  .p.type = AVMEDIA_TYPE_AUDIO,
458  .p.id = AV_CODEC_ID_AAC,
459  .priv_data_size = sizeof(AACContext),
461  .close = aac_decode_close,
463  .p.sample_fmts = (const enum AVSampleFormat[]) {
465  },
466  .p.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
467  .caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
468  CODEC_OLD_CHANNEL_LAYOUTS_ARRAY(aac_channel_layout)
469  .p.ch_layouts = aac_ch_layout,
470  .p.priv_class = &aac_decoder_class,
471  .p.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
472  .flush = flush,
473 };
vector_pow43
static void vector_pow43(int *coefs, int len)
Definition: aacdec_fixed.c:154
flt16_even
static av_always_inline SoftFloat flt16_even(SoftFloat pf)
Definition: aacdec_fixed.c:254
FF_CODEC_CAP_INIT_CLEANUP
#define FF_CODEC_CAP_INIT_CLEANUP
The codec allows calling the close function for deallocation even if the init function returned a fai...
Definition: codec_internal.h:42
opt.h
out
FILE * out
Definition: movenc.c:54
aac_decode_init
static av_cold int aac_decode_init(AVCodecContext *avctx)
Definition: aacdec_template.c:1181
aac_kbd_short_120
static INTFLOAT aac_kbd_short_120[120]
Definition: aacdec.c:72
PredictorState::var1
AAC_FLOAT var1
Definition: aac.h:137
aacsbr.h
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:28
PredictorState::var0
AAC_FLOAT var0
Definition: aac.h:136
av_sub_sf
static av_const SoftFloat av_sub_sf(SoftFloat a, SoftFloat b)
Definition: softfloat.h:173
aacdectab.h
AV_SAMPLE_FMT_S32P
@ AV_SAMPLE_FMT_S32P
signed 32 bits, planar
Definition: samplefmt.h:65
FFCodec
Definition: codec_internal.h:119
ff_aac_profiles
const AVProfile ff_aac_profiles[]
Definition: profiles.c:26
SoftFloat::mant
int32_t mant
Definition: softfloat.h:35
DEC_SQUAD
static int * DEC_SQUAD(int *dst, unsigned idx)
Definition: aacdec_fixed.c:118
lpc.h
SingleChannelElement::ret
INTFLOAT * ret
PCM output.
Definition: aac.h:267
intfloat.h
apply_independent_coupling_fixed
static void apply_independent_coupling_fixed(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply independent channel coupling (applied after IMDCT).
Definition: aacdec_fixed.c:421
sbr.h
apply_dependent_coupling_fixed
static void apply_dependent_coupling_fixed(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)
Apply dependent channel coupling (applied before IMDCT).
Definition: aacdec_fixed.c:357
FFCodec::p
AVCodec p
The public AVCodec.
Definition: codec_internal.h:123
mpeg4audio.h
ChannelElement::coup
ChannelCoupling coup
Definition: aac.h:284
av_div_sf
static av_const SoftFloat av_div_sf(SoftFloat a, SoftFloat b)
b has to be normalized and not zero.
Definition: softfloat.h:116
scale
static av_always_inline float scale(float x, float s)
Definition: vf_v360.c:1389
PredictorState::r0
AAC_FLOAT r0
Definition: aac.h:138
SingleChannelElement::ics
IndividualChannelStream ics
Definition: aac.h:247
aac_decode_frame
static int aac_decode_frame(AVCodecContext *avctx, AVFrame *frame, int *got_frame_ptr, AVPacket *avpkt)
Definition: aacdec_template.c:3346
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:180
ZERO_BT
@ ZERO_BT
Scalefactors and spectral data are all zero.
Definition: aac.h:82
ff_cbrt_tab_fixed
uint32_t ff_cbrt_tab_fixed[1<< 13]
FF_CODEC_DECODE_CB
#define FF_CODEC_DECODE_CB(func)
Definition: codec_internal.h:298
aac_decoder_class
static const AVClass aac_decoder_class
Definition: aacdec_template.c:3477
s
#define s(width, name)
Definition: cbs_vp9.c:256
SingleChannelElement::coeffs
INTFLOAT coeffs[1024]
coefficients for IMDCT, maybe processed
Definition: aac.h:260
offsets
static const int offsets[]
Definition: hevc_pel.c:34
g
const char * g
Definition: vf_curves.c:127
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
aac_ch_layout
static const AVChannelLayout aac_ch_layout[]
Definition: aacdectab.h:113
init
int(* init)(AVBSFContext *ctx)
Definition: dts2pts_bsf.c:363
IndividualChannelStream::group_len
uint8_t group_len[8]
Definition: aac.h:177
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:37
PredictorState
Predictor State.
Definition: aac.h:133
get_bits.h
predict
static av_always_inline void predict(PredictorState *ps, int *coef, int output_enable)
Definition: aacdec_fixed.c:282
kbdwin.h
fixed_dsp.h
IndividualChannelStream
Individual Channel Stream.
Definition: aac.h:172
IndividualChannelStream::swb_offset
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:179
CODEC_LONG_NAME
#define CODEC_LONG_NAME(str)
Definition: codec_internal.h:264
FFABS
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:64
IndividualChannelStream::num_window_groups
int num_window_groups
Definition: aac.h:176
sinewin_fixed_tablegen.h
aac_decode_close
static av_cold int aac_decode_close(AVCodecContext *avctx)
Definition: aacdec_template.c:3409
profiles.h
aac.h
aactab.h
SoftFloat::exp
int32_t exp
Definition: softfloat.h:36
PredictorState::r1
AAC_FLOAT r1
Definition: aac.h:139
index
int index
Definition: gxfenc.c:89
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
AV_CODEC_CAP_CHANNEL_CONF
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition: codec.h:109
AV_CODEC_ID_AAC
@ AV_CODEC_ID_AAC
Definition: codec_id.h:437
AACContext::avctx
AVCodecContext * avctx
Definition: aac.h:298
AV_CODEC_CAP_DR1
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
ChannelElement::ch
SingleChannelElement ch[2]
Definition: aac.h:282
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:115
reset_predict_state
static av_always_inline void reset_predict_state(PredictorState *ps)
Definition: aacdec_fixed.c:92
codec_internal.h
shift
static int shift(int a, int b)
Definition: bonk.c:253
noise_scale
static void noise_scale(int *coefs, int scale, int band_energy, int len)
Definition: aacdec_fixed.c:199
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:56
AAC_RENAME2
#define AAC_RENAME2(x)
Definition: aac_defines.h:81
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
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
cbrt_data.h
aac_kbd_long_960
static INTFLOAT aac_kbd_long_960[960]
Definition: aacdec.c:71
DECLARE_ALIGNED
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:116
cce_scale_fixed
static const int cce_scale_fixed[8]
Definition: aacdec_fixed.c:341
CODEC_OLD_CHANNEL_LAYOUTS_ARRAY
#define CODEC_OLD_CHANNEL_LAYOUTS_ARRAY(array)
Definition: codec_internal.h:295
SoftFloat
Definition: softfloat.h:34
SingleChannelElement
Single Channel Element - used for both SCE and LFE elements.
Definition: aac.h:246
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
round
static av_always_inline av_const double round(double x)
Definition: libm.h:444
ChannelElement
channel element - generic struct for SCE/CPE/CCE/LFE
Definition: aac.h:273
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:55
av_always_inline
#define av_always_inline
Definition: attributes.h:49
av_int2sf
static av_const SoftFloat av_int2sf(int v, int frac_bits)
Converts a mantisse and exponent to a SoftFloat.
Definition: softfloat.h:185
AVCodec::name
const char * name
Name of the codec implementation.
Definition: codec.h:211
AACContext::oc
OutputConfiguration oc[2]
Definition: aac.h:369
len
int len
Definition: vorbis_enc_data.h:426
ff_aac_fixed_decoder
const FFCodec ff_aac_fixed_decoder
Definition: aacdec_fixed.c:454
subband_scale
static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
Definition: aacdec_fixed.c:168
avcodec.h
pv
#define pv
Definition: regdef.h:60
exp2tab
static const int exp2tab[4]
Definition: aacdec_fixed.c:108
MPEG4AudioConfig::object_type
int object_type
Definition: mpeg4audio.h:30
U
#define U(x)
Definition: vpx_arith.h:37
av_add_sf
static av_const SoftFloat av_add_sf(SoftFloat a, SoftFloat b)
Definition: softfloat.h:162
ChannelCoupling::gain
INTFLOAT gain[16][120]
Definition: aac.h:240
MPEG4AudioConfig::sbr
int sbr
-1 implicit, 1 presence
Definition: mpeg4audio.h:34
Q31
#define Q31(x)
Definition: aac_defines.h:92
OutputConfiguration::m4ac
MPEG4AudioConfig m4ac
Definition: aac.h:123
DEC_UQUAD
static int * DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:136
aacdec_template.c
flush
void(* flush)(AVBSFContext *ctx)
Definition: dts2pts_bsf.c:365
PredictorState::cor1
AAC_FLOAT cor1
Definition: aac.h:135
adts_header.h
alpha
static const int16_t alpha[]
Definition: ilbcdata.h:55
src
INIT_CLIP pixel * src
Definition: h264pred_template.c:418
AACContext
main AAC context
Definition: aac.h:296
IndividualChannelStream::max_sfb
uint8_t max_sfb
number of scalefactor bands per group
Definition: aac.h:173
flt16_round
static av_always_inline SoftFloat flt16_round(SoftFloat pf)
Definition: aacdec_fixed.c:240
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:27
DEC_UPAIR
static int * DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
Definition: aacdec_fixed.c:128
DEC_SPAIR
static int * DEC_SPAIR(int *dst, unsigned idx)
Definition: aacdec_fixed.c:110
int
int
Definition: ffmpeg_filter.c:156
SingleChannelElement::band_type
enum BandType band_type[128]
band types
Definition: aac.h:250
av_mul_sf
static av_const SoftFloat av_mul_sf(SoftFloat a, SoftFloat b)
Definition: softfloat.h:102
flt16_trunc
static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
Definition: aacdec_fixed.c:268
AOT_AAC_LTP
@ AOT_AAC_LTP
Y Long Term Prediction.
Definition: mpeg4audio.h:76
Q30
#define Q30(x)
Definition: aac_defines.h:91
PredictorState::cor0
AAC_FLOAT cor0
Definition: aac.h:134