FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/aacenc.c
Date: 2021-09-16 08:47:15
Exec Total Coverage
Lines: 553 651 84.9%
Branches: 354 484 73.1%

Line Branch Exec Source
1 /*
2 * AAC encoder
3 * Copyright (C) 2008 Konstantin Shishkov
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 /**
23 * @file
24 * AAC encoder
25 */
26
27 /***********************************
28 * TODOs:
29 * add sane pulse detection
30 ***********************************/
31 #include <float.h>
32
33 #include "libavutil/channel_layout.h"
34 #include "libavutil/libm.h"
35 #include "libavutil/float_dsp.h"
36 #include "libavutil/opt.h"
37 #include "avcodec.h"
38 #include "encode.h"
39 #include "put_bits.h"
40 #include "internal.h"
41 #include "mpeg4audio.h"
42 #include "sinewin.h"
43 #include "profiles.h"
44
45 #include "aac.h"
46 #include "aactab.h"
47 #include "aacenc.h"
48 #include "aacenctab.h"
49 #include "aacenc_utils.h"
50
51 #include "psymodel.h"
52
53 static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
54 {
55 int i, j;
56 AACEncContext *s = avctx->priv_data;
57 AACPCEInfo *pce = &s->pce;
58 const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT;
59 const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT;
60
61 put_bits(pb, 4, 0);
62
63 put_bits(pb, 2, avctx->profile);
64 put_bits(pb, 4, s->samplerate_index);
65
66 put_bits(pb, 4, pce->num_ele[0]); /* Front */
67 put_bits(pb, 4, pce->num_ele[1]); /* Side */
68 put_bits(pb, 4, pce->num_ele[2]); /* Back */
69 put_bits(pb, 2, pce->num_ele[3]); /* LFE */
70 put_bits(pb, 3, 0); /* Assoc data */
71 put_bits(pb, 4, 0); /* CCs */
72
73 put_bits(pb, 1, 0); /* Stereo mixdown */
74 put_bits(pb, 1, 0); /* Mono mixdown */
75 put_bits(pb, 1, 0); /* Something else */
76
77 for (i = 0; i < 4; i++) {
78 for (j = 0; j < pce->num_ele[i]; j++) {
79 if (i < 3)
80 put_bits(pb, 1, pce->pairing[i][j]);
81 put_bits(pb, 4, pce->index[i][j]);
82 }
83 }
84
85 align_put_bits(pb);
86 put_bits(pb, 8, strlen(aux_data));
87 ff_put_string(pb, aux_data, 0);
88 }
89
90 /**
91 * Make AAC audio config object.
92 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
93 */
94 11 static int put_audio_specific_config(AVCodecContext *avctx)
95 {
96 PutBitContext pb;
97 11 AACEncContext *s = avctx->priv_data;
98 11 int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0));
99 11 const int max_size = 32;
100
101 11 avctx->extradata = av_mallocz(max_size);
102
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11 if (!avctx->extradata)
103 return AVERROR(ENOMEM);
104
105 11 init_put_bits(&pb, avctx->extradata, max_size);
106 11 put_bits(&pb, 5, s->profile+1); //profile
107 11 put_bits(&pb, 4, s->samplerate_index); //sample rate index
108 11 put_bits(&pb, 4, channels);
109 //GASpecificConfig
110 11 put_bits(&pb, 1, 0); //frame length - 1024 samples
111 11 put_bits(&pb, 1, 0); //does not depend on core coder
112 11 put_bits(&pb, 1, 0); //is not extension
113
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11 if (s->needs_pce)
114 put_pce(&pb, avctx);
115
116 //Explicitly Mark SBR absent
117 11 put_bits(&pb, 11, 0x2b7); //sync extension
118 11 put_bits(&pb, 5, AOT_SBR);
119 11 put_bits(&pb, 1, 0);
120 11 flush_put_bits(&pb);
121 11 avctx->extradata_size = put_bytes_output(&pb);
122
123 11 return 0;
124 }
125
126 10135 void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
127 {
128 10135 ++s->quantize_band_cost_cache_generation;
129
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10135 if (s->quantize_band_cost_cache_generation == 0) {
130 memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
131 s->quantize_band_cost_cache_generation = 1;
132 }
133 10135 }
134
135 #define WINDOW_FUNC(type) \
136 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
137 SingleChannelElement *sce, \
138 const float *audio)
139
140 6899 WINDOW_FUNC(only_long)
141 {
142
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6899 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
143
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6899 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
144 6899 float *out = sce->ret_buf;
145
146 6899 fdsp->vector_fmul (out, audio, lwindow, 1024);
147 6899 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
148 6899 }
149
150 119 WINDOW_FUNC(long_start)
151 {
152
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119 const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
153
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119 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
154 119 float *out = sce->ret_buf;
155
156 119 fdsp->vector_fmul(out, audio, lwindow, 1024);
157 119 memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
158 119 fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
159 119 memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
160 119 }
161
162 102 WINDOW_FUNC(long_stop)
163 {
164
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102 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
165
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102 const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
166 102 float *out = sce->ret_buf;
167
168 102 memset(out, 0, sizeof(out[0]) * 448);
169 102 fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
170 102 memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
171 102 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
172 102 }
173
174 164 WINDOW_FUNC(eight_short)
175 {
176
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164 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
177
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164 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
178 164 const float *in = audio + 448;
179 164 float *out = sce->ret_buf;
180 int w;
181
182
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1476 for (w = 0; w < 8; w++) {
183
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1312 fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
184 1312 out += 128;
185 1312 in += 128;
186 1312 fdsp->vector_fmul_reverse(out, in, swindow, 128);
187 1312 out += 128;
188 }
189 164 }
190
191 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
192 SingleChannelElement *sce,
193 const float *audio) = {
194 [ONLY_LONG_SEQUENCE] = apply_only_long_window,
195 [LONG_START_SEQUENCE] = apply_long_start_window,
196 [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
197 [LONG_STOP_SEQUENCE] = apply_long_stop_window
198 };
199
200 7284 static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
201 float *audio)
202 {
203 int i;
204 7284 const float *output = sce->ret_buf;
205
206 7284 apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
207
208
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7284 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
209 7120 s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
210 else
211
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1476 for (i = 0; i < 1024; i += 128)
212 1312 s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
213 7284 memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
214 7284 memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
215 7284 }
216
217 /**
218 * Encode ics_info element.
219 * @see Table 4.6 (syntax of ics_info)
220 */
221 5693 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
222 {
223 int w;
224
225 5693 put_bits(&s->pb, 1, 0); // ics_reserved bit
226 5693 put_bits(&s->pb, 2, info->window_sequence[0]);
227 5693 put_bits(&s->pb, 1, info->use_kb_window[0]);
228
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5693 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
229 5513 put_bits(&s->pb, 6, info->max_sfb);
230 5513 put_bits(&s->pb, 1, !!info->predictor_present);
231 } else {
232 180 put_bits(&s->pb, 4, info->max_sfb);
233
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1440 for (w = 1; w < 8; w++)
234 1260 put_bits(&s->pb, 1, !info->group_len[w]);
235 }
236 5693 }
237
238 /**
239 * Encode MS data.
240 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
241 */
242 4538 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
243 {
244 int i, w;
245
246 4538 put_bits(pb, 2, cpe->ms_mode);
247
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4538 if (cpe->ms_mode == 1)
248
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823 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
249
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19581 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
250 19160 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
251 4538 }
252
253 /**
254 * Produce integer coefficients from scalefactors provided by the model.
255 */
256 5448 static void adjust_frame_information(ChannelElement *cpe, int chans)
257 {
258 int i, w, w2, g, ch;
259 int maxsfb, cmaxsfb;
260
261
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15679 for (ch = 0; ch < chans; ch++) {
262 10231 IndividualChannelStream *ics = &cpe->ch[ch].ics;
263 10231 maxsfb = 0;
264 10231 cpe->ch[ch].pulse.num_pulse = 0;
265
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21114 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
266
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22885 for (w2 = 0; w2 < ics->group_len[w]; w2++) {
267
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29945 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
268 ;
269 12002 maxsfb = FFMAX(maxsfb, cmaxsfb);
270 }
271 }
272 10231 ics->max_sfb = maxsfb;
273
274 //adjust zero bands for window groups
275
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21114 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
276
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498183 for (g = 0; g < ics->max_sfb; g++) {
277 487300 i = 1;
278
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490475 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
279
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487520 if (!cpe->ch[ch].zeroes[w2*16 + g]) {
280 484345 i = 0;
281 484345 break;
282 }
283 }
284 487300 cpe->ch[ch].zeroes[w*16 + g] = i;
285 }
286 }
287 }
288
289
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5448 if (chans > 1 && cpe->common_window) {
290 4538 IndividualChannelStream *ics0 = &cpe->ch[0].ics;
291 4538 IndividualChannelStream *ics1 = &cpe->ch[1].ics;
292 4538 int msc = 0;
293 4538 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
294 4538 ics1->max_sfb = ics0->max_sfb;
295
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9587 for (w = 0; w < ics0->num_windows*16; w += 16)
296
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225553 for (i = 0; i < ics0->max_sfb; i++)
297
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220504 if (cpe->ms_mask[w+i])
298 24406 msc++;
299
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4538 if (msc == 0 || ics0->max_sfb == 0)
300 3749 cpe->ms_mode = 0;
301 else
302
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789 cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
303 }
304 5448 }
305
306 1334 static void apply_intensity_stereo(ChannelElement *cpe)
307 {
308 int w, w2, g, i;
309 1334 IndividualChannelStream *ics = &cpe->ch[0].ics;
310
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1334 if (!cpe->common_window)
311 707 return;
312
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1288 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
313
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1386 for (w2 = 0; w2 < ics->group_len[w]; w2++) {
314 725 int start = (w+w2) * 128;
315
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32330 for (g = 0; g < ics->num_swb; g++) {
316 31605 int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
317 31605 float scale = cpe->ch[0].is_ener[w*16+g];
318
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31605 if (!cpe->is_mask[w*16 + g]) {
319 22930 start += ics->swb_sizes[g];
320 22930 continue;
321 }
322
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8675 if (cpe->ms_mask[w*16 + g])
323 2383 p *= -1;
324
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300811 for (i = 0; i < ics->swb_sizes[g]; i++) {
325 292136 float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
326 292136 cpe->ch[0].coeffs[start+i] = sum;
327 292136 cpe->ch[1].coeffs[start+i] = 0.0f;
328 }
329 8675 start += ics->swb_sizes[g];
330 }
331 }
332 }
333 }
334
335 1050 static void apply_mid_side_stereo(ChannelElement *cpe)
336 {
337 int w, w2, g, i;
338 1050 IndividualChannelStream *ics = &cpe->ch[0].ics;
339
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1050 if (!cpe->common_window)
340 548 return;
341
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1038 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
342
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1136 for (w2 = 0; w2 < ics->group_len[w]; w2++) {
343 600 int start = (w+w2) * 128;
344
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26080 for (g = 0; g < ics->num_swb; g++) {
345 /* ms_mask can be used for other purposes in PNS and I/S,
346 * so must not apply M/S if any band uses either, even if
347 * ms_mask is set.
348 */
349
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25480 if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
350
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22915 || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
351
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22915 || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
352 2565 start += ics->swb_sizes[g];
353 2565 continue;
354 }
355
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484399 for (i = 0; i < ics->swb_sizes[g]; i++) {
356 461484 float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
357 461484 float R = L - cpe->ch[1].coeffs[start+i];
358 461484 cpe->ch[0].coeffs[start+i] = L;
359 461484 cpe->ch[1].coeffs[start+i] = R;
360 }
361 22915 start += ics->swb_sizes[g];
362 }
363 }
364 }
365 }
366
367 /**
368 * Encode scalefactor band coding type.
369 */
370 10231 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
371 {
372 int w;
373
374
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10231 if (s->coder->set_special_band_scalefactors)
375 10231 s->coder->set_special_band_scalefactors(s, sce);
376
377
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21114 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
378 10883 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
379 10231 }
380
381 /**
382 * Encode scalefactors.
383 */
384 10231 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
385 SingleChannelElement *sce)
386 {
387 10231 int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
388 10231 int off_is = 0, noise_flag = 1;
389 int i, w;
390
391
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21114 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
392
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498188 for (i = 0; i < sce->ics.max_sfb; i++) {
393
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487305 if (!sce->zeroes[w*16 + i]) {
394
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476374 if (sce->band_type[w*16 + i] == NOISE_BT) {
395 6375 diff = sce->sf_idx[w*16 + i] - off_pns;
396 6375 off_pns = sce->sf_idx[w*16 + i];
397
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6375 if (noise_flag-- > 0) {
398 1122 put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
399 1122 continue;
400 }
401
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469999 } else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
402
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463673 sce->band_type[w*16 + i] == INTENSITY_BT2) {
403 8564 diff = sce->sf_idx[w*16 + i] - off_is;
404 8564 off_is = sce->sf_idx[w*16 + i];
405 } else {
406 461435 diff = sce->sf_idx[w*16 + i] - off_sf;
407 461435 off_sf = sce->sf_idx[w*16 + i];
408 }
409 475252 diff += SCALE_DIFF_ZERO;
410
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475252 av_assert0(diff >= 0 && diff <= 120);
411 475252 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
412 }
413 }
414 }
415 10231 }
416
417 /**
418 * Encode pulse data.
419 */
420 10231 static void encode_pulses(AACEncContext *s, Pulse *pulse)
421 {
422 int i;
423
424 10231 put_bits(&s->pb, 1, !!pulse->num_pulse);
425
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10231 if (!pulse->num_pulse)
426 10231 return;
427
428 put_bits(&s->pb, 2, pulse->num_pulse - 1);
429 put_bits(&s->pb, 6, pulse->start);
430 for (i = 0; i < pulse->num_pulse; i++) {
431 put_bits(&s->pb, 5, pulse->pos[i]);
432 put_bits(&s->pb, 4, pulse->amp[i]);
433 }
434 }
435
436 /**
437 * Encode spectral coefficients processed by psychoacoustic model.
438 */
439 10231 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
440 {
441 int start, i, w, w2;
442
443
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21114 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
444 10883 start = 0;
445
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498188 for (i = 0; i < sce->ics.max_sfb; i++) {
446
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487305 if (sce->zeroes[w*16 + i]) {
447 10931 start += sce->ics.swb_sizes[i];
448 10931 continue;
449 }
450
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960754 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
451 484380 s->coder->quantize_and_encode_band(s, &s->pb,
452 484380 &sce->coeffs[start + w2*128],
453 484380 NULL, sce->ics.swb_sizes[i],
454 484380 sce->sf_idx[w*16 + i],
455 484380 sce->band_type[w*16 + i],
456 s->lambda,
457 484380 sce->ics.window_clipping[w]);
458 }
459 476374 start += sce->ics.swb_sizes[i];
460 }
461 }
462 10231 }
463
464 /**
465 * Downscale spectral coefficients for near-clipping windows to avoid artifacts
466 */
467 7284 static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
468 {
469 int start, i, j, w;
470
471
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7284 if (sce->ics.clip_avoidance_factor < 1.0f) {
472
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392 for (w = 0; w < sce->ics.num_windows; w++) {
473 224 start = 0;
474
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8690 for (i = 0; i < sce->ics.max_sfb; i++) {
475 8466 float *swb_coeffs = &sce->coeffs[start + w*128];
476
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168658 for (j = 0; j < sce->ics.swb_sizes[i]; j++)
477 160192 swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
478 8466 start += sce->ics.swb_sizes[i];
479 }
480 }
481 }
482 7284 }
483
484 /**
485 * Encode one channel of audio data.
486 */
487 10231 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
488 SingleChannelElement *sce,
489 int common_window)
490 {
491 10231 put_bits(&s->pb, 8, sce->sf_idx[0]);
492
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10231 if (!common_window) {
493 1155 put_ics_info(s, &sce->ics);
494
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1155 if (s->coder->encode_main_pred)
495 1155 s->coder->encode_main_pred(s, sce);
496
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1155 if (s->coder->encode_ltp_info)
497 1155 s->coder->encode_ltp_info(s, sce, 0);
498 }
499 10231 encode_band_info(s, sce);
500 10231 encode_scale_factors(avctx, s, sce);
501 10231 encode_pulses(s, &sce->pulse);
502 10231 put_bits(&s->pb, 1, !!sce->tns.present);
503
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10231 if (s->coder->encode_tns_info)
504 10231 s->coder->encode_tns_info(s, sce);
505 10231 put_bits(&s->pb, 1, 0); //ssr
506 10231 encode_spectral_coeffs(s, sce);
507 10231 return 0;
508 }
509
510 /**
511 * Write some auxiliary information about the created AAC file.
512 */
513 static void put_bitstream_info(AACEncContext *s, const char *name)
514 {
515 int i, namelen, padbits;
516
517 namelen = strlen(name) + 2;
518 put_bits(&s->pb, 3, TYPE_FIL);
519 put_bits(&s->pb, 4, FFMIN(namelen, 15));
520 if (namelen >= 15)
521 put_bits(&s->pb, 8, namelen - 14);
522 put_bits(&s->pb, 4, 0); //extension type - filler
523 padbits = -put_bits_count(&s->pb) & 7;
524 align_put_bits(&s->pb);
525 for (i = 0; i < namelen - 2; i++)
526 put_bits(&s->pb, 8, name[i]);
527 put_bits(&s->pb, 12 - padbits, 0);
528 }
529
530 /*
531 * Copy input samples.
532 * Channels are reordered from libavcodec's default order to AAC order.
533 */
534 3816 static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
535 {
536 int ch;
537
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3816 int end = 2048 + (frame ? frame->nb_samples : 0);
538 3816 const uint8_t *channel_map = s->reorder_map;
539
540 /* copy and remap input samples */
541
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11125 for (ch = 0; ch < s->channels; ch++) {
542 /* copy last 1024 samples of previous frame to the start of the current frame */
543 7309 memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
544
545 /* copy new samples and zero any remaining samples */
546
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7309 if (frame) {
547 7259 memcpy(&s->planar_samples[ch][2048],
548 7259 frame->extended_data[channel_map[ch]],
549 7259 frame->nb_samples * sizeof(s->planar_samples[0][0]));
550 }
551 7309 memset(&s->planar_samples[ch][end], 0,
552 7309 (3072 - end) * sizeof(s->planar_samples[0][0]));
553 }
554 3816 }
555
556 3827 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
557 const AVFrame *frame, int *got_packet_ptr)
558 {
559 3827 AACEncContext *s = avctx->priv_data;
560 3827 float **samples = s->planar_samples, *samples2, *la, *overlap;
561 ChannelElement *cpe;
562 SingleChannelElement *sce;
563 IndividualChannelStream *ics;
564 int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
565 int target_bits, rate_bits, too_many_bits, too_few_bits;
566 3827 int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
567 int chan_el_counter[4];
568 FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
569
570 /* add current frame to queue */
571
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3827 if (frame) {
572
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3794 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
573 return ret;
574 } else {
575
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33 if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
576 11 return 0;
577 }
578
579 3816 copy_input_samples(s, frame);
580
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3816 if (s->psypp)
581 3816 ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
582
583
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3816 if (!avctx->frame_number)
584 11 return 0;
585
586 3805 start_ch = 0;
587
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7754 for (i = 0; i < s->chan_map[0]; i++) {
588 3949 FFPsyWindowInfo* wi = windows + start_ch;
589 3949 tag = s->chan_map[i+1];
590
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3949 chans = tag == TYPE_CPE ? 2 : 1;
591 3949 cpe = &s->cpe[i];
592
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11233 for (ch = 0; ch < chans; ch++) {
593 int k;
594 float clip_avoidance_factor;
595 7284 sce = &cpe->ch[ch];
596 7284 ics = &sce->ics;
597 7284 s->cur_channel = start_ch + ch;
598 7284 overlap = &samples[s->cur_channel][0];
599 7284 samples2 = overlap + 1024;
600 7284 la = samples2 + (448+64);
601
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7284 if (!frame)
602 50 la = NULL;
603
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7284 if (tag == TYPE_LFE) {
604 48 wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
605 48 wi[ch].window_shape = 0;
606 48 wi[ch].num_windows = 1;
607 48 wi[ch].grouping[0] = 1;
608 48 wi[ch].clipping[0] = 0;
609
610 /* Only the lowest 12 coefficients are used in a LFE channel.
611 * The expression below results in only the bottom 8 coefficients
612 * being used for 11.025kHz to 16kHz sample rates.
613 */
614
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48 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
615 } else {
616 7236 wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
617 7236 ics->window_sequence[0]);
618 }
619 7284 ics->window_sequence[1] = ics->window_sequence[0];
620 7284 ics->window_sequence[0] = wi[ch].window_type[0];
621 7284 ics->use_kb_window[1] = ics->use_kb_window[0];
622 7284 ics->use_kb_window[0] = wi[ch].window_shape;
623 7284 ics->num_windows = wi[ch].num_windows;
624
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7284 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
625
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7284 ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
626 7284 ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb);
627 14568 ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
628
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7284 ff_swb_offset_128 [s->samplerate_index]:
629 7120 ff_swb_offset_1024[s->samplerate_index];
630 14568 ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
631
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7284 ff_tns_max_bands_128 [s->samplerate_index]:
632 7120 ff_tns_max_bands_1024[s->samplerate_index];
633
634
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15716 for (w = 0; w < ics->num_windows; w++)
635 8432 ics->group_len[w] = wi[ch].grouping[w];
636
637 /* Calculate input sample maximums and evaluate clipping risk */
638 7284 clip_avoidance_factor = 0.0f;
639
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15716 for (w = 0; w < ics->num_windows; w++) {
640 8432 const float *wbuf = overlap + w * 128;
641 8432 const int wlen = 2048 / ics->num_windows;
642 8432 float max = 0;
643 int j;
644 /* mdct input is 2 * output */
645
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14926064 for (j = 0; j < wlen; j++)
646
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14917632 max = FFMAX(max, fabsf(wbuf[j]));
647 8432 wi[ch].clipping[w] = max;
648 }
649
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15716 for (w = 0; w < ics->num_windows; w++) {
650
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8432 if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
651 176 ics->window_clipping[w] = 1;
652
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176 clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
653 } else {
654 8256 ics->window_clipping[w] = 0;
655 }
656 }
657
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7284 if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
658 168 ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
659 } else {
660 7116 ics->clip_avoidance_factor = 1.0f;
661 }
662
663 7284 apply_window_and_mdct(s, sce, overlap);
664
665
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7284 if (s->options.ltp && s->coder->update_ltp) {
666 s->coder->update_ltp(s, sce);
667 apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
668 s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
669 }
670
671
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7466100 for (k = 0; k < 1024; k++) {
672
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7458816 if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
673 av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
674 return AVERROR(EINVAL);
675 }
676 }
677 7284 avoid_clipping(s, sce);
678 }
679 3949 start_ch += chans;
680 }
681
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3805 if ((ret = ff_alloc_packet(avctx, avpkt, 8192 * s->channels)) < 0)
682 return ret;
683 3805 frame_bits = its = 0;
684 do {
685 5232 init_put_bits(&s->pb, avpkt->data, avpkt->size);
686
687
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5232 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
688 put_bitstream_info(s, LIBAVCODEC_IDENT);
689 5232 start_ch = 0;
690 5232 target_bits = 0;
691 5232 memset(chan_el_counter, 0, sizeof(chan_el_counter));
692
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10680 for (i = 0; i < s->chan_map[0]; i++) {
693 5448 FFPsyWindowInfo* wi = windows + start_ch;
694 const float *coeffs[2];
695 5448 tag = s->chan_map[i+1];
696
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5448 chans = tag == TYPE_CPE ? 2 : 1;
697 5448 cpe = &s->cpe[i];
698 5448 cpe->common_window = 0;
699 5448 memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
700 5448 memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
701 5448 put_bits(&s->pb, 3, tag);
702 5448 put_bits(&s->pb, 4, chan_el_counter[tag]++);
703
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15679 for (ch = 0; ch < chans; ch++) {
704 10231 sce = &cpe->ch[ch];
705 10231 coeffs[ch] = sce->coeffs;
706 10231 sce->ics.predictor_present = 0;
707 10231 sce->ics.ltp.present = 0;
708 10231 memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
709 10231 memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
710 10231 memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
711
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1319799 for (w = 0; w < 128; w++)
712
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1309568 if (sce->band_type[w] > RESERVED_BT)
713 14871 sce->band_type[w] = 0;
714 }
715 5448 s->psy.bitres.alloc = -1;
716 5448 s->psy.bitres.bits = s->last_frame_pb_count / s->channels;
717 5448 s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
718
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5448 if (s->psy.bitres.alloc > 0) {
719 /* Lambda unused here on purpose, we need to take psy's unscaled allocation */
720 10896 target_bits += s->psy.bitres.alloc
721
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5448 * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
722 5448 s->psy.bitres.alloc /= chans;
723 }
724 5448 s->cur_type = tag;
725
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15679 for (ch = 0; ch < chans; ch++) {
726 10231 s->cur_channel = start_ch + ch;
727
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10231 if (s->options.pns && s->coder->mark_pns)
728 2003 s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
729 10231 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
730 }
731
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✓ Branch 1 taken 665 times.
5448 if (chans > 1
732
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4783 && wi[0].window_type[0] == wi[1].window_type[0]
733
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4547 && wi[0].window_shape == wi[1].window_shape) {
734
735 4547 cpe->common_window = 1;
736
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9596 for (w = 0; w < wi[0].num_windows; w++) {
737
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5058 if (wi[0].grouping[w] != wi[1].grouping[w]) {
738 9 cpe->common_window = 0;
739 9 break;
740 }
741 }
742 }
743
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15679 for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
744 10231 sce = &cpe->ch[ch];
745 10231 s->cur_channel = start_ch + ch;
746
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10231 if (s->options.tns && s->coder->search_for_tns)
747 2003 s->coder->search_for_tns(s, sce);
748
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10231 if (s->options.tns && s->coder->apply_tns_filt)
749 2003 s->coder->apply_tns_filt(s, sce);
750
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10231 if (sce->tns.present)
751 40 tns_mode = 1;
752
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10231 if (s->options.pns && s->coder->search_for_pns)
753 2003 s->coder->search_for_pns(s, avctx, sce);
754 }
755 5448 s->cur_channel = start_ch;
756
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5448 if (s->options.intensity_stereo) { /* Intensity Stereo */
757
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1334 if (s->coder->search_for_is)
758 1334 s->coder->search_for_is(s, avctx, cpe);
759
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1334 if (cpe->is_mode) is_mode = 1;
760 1334 apply_intensity_stereo(cpe);
761 }
762
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5448 if (s->options.pred) { /* Prediction */
763
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1248 for (ch = 0; ch < chans; ch++) {
764 832 sce = &cpe->ch[ch];
765 832 s->cur_channel = start_ch + ch;
766
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832 if (s->options.pred && s->coder->search_for_pred)
767 832 s->coder->search_for_pred(s, sce);
768
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832 if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
769 }
770
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416 if (s->coder->adjust_common_pred)
771 416 s->coder->adjust_common_pred(s, cpe);
772
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1248 for (ch = 0; ch < chans; ch++) {
773 832 sce = &cpe->ch[ch];
774 832 s->cur_channel = start_ch + ch;
775
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832 if (s->options.pred && s->coder->apply_main_pred)
776 832 s->coder->apply_main_pred(s, sce);
777 }
778 416 s->cur_channel = start_ch;
779 }
780
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5448 if (s->options.mid_side) { /* Mid/Side stereo */
781
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1050 if (s->options.mid_side == -1 && s->coder->search_for_ms)
782 630 s->coder->search_for_ms(s, cpe);
783
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420 else if (cpe->common_window)
784 393 memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
785 1050 apply_mid_side_stereo(cpe);
786 }
787 5448 adjust_frame_information(cpe, chans);
788
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5448 if (s->options.ltp) { /* LTP */
789 for (ch = 0; ch < chans; ch++) {
790 sce = &cpe->ch[ch];
791 s->cur_channel = start_ch + ch;
792 if (s->coder->search_for_ltp)
793 s->coder->search_for_ltp(s, sce, cpe->common_window);
794 if (sce->ics.ltp.present) pred_mode = 1;
795 }
796 s->cur_channel = start_ch;
797 if (s->coder->adjust_common_ltp)
798 s->coder->adjust_common_ltp(s, cpe);
799 }
800
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5448 if (chans == 2) {
801 4783 put_bits(&s->pb, 1, cpe->common_window);
802
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4783 if (cpe->common_window) {
803 4538 put_ics_info(s, &cpe->ch[0].ics);
804
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4538 if (s->coder->encode_main_pred)
805 4538 s->coder->encode_main_pred(s, &cpe->ch[0]);
806
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4538 if (s->coder->encode_ltp_info)
807 4538 s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
808 4538 encode_ms_info(&s->pb, cpe);
809
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4538 if (cpe->ms_mode) ms_mode = 1;
810 }
811 }
812
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15679 for (ch = 0; ch < chans; ch++) {
813 10231 s->cur_channel = start_ch + ch;
814 10231 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
815 }
816 5448 start_ch += chans;
817 }
818
819
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5232 if (avctx->flags & AV_CODEC_FLAG_QSCALE) {
820 /* When using a constant Q-scale, don't mess with lambda */
821 break;
822 }
823
824 /* rate control stuff
825 * allow between the nominal bitrate, and what psy's bit reservoir says to target
826 * but drift towards the nominal bitrate always
827 */
828 5232 frame_bits = put_bits_count(&s->pb);
829 5232 rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
830
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5232 rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
831 5232 too_many_bits = FFMAX(target_bits, rate_bits);
832
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5232 too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
833 5232 too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
834
835 /* When using ABR, be strict (but only for increasing) */
836 5232 too_few_bits = too_few_bits - too_few_bits/8;
837 5232 too_many_bits = too_many_bits + too_many_bits/2;
838
839
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5232 if ( its == 0 /* for steady-state Q-scale tracking */
840
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1427 || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
841
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289 || frame_bits >= 6144 * s->channels - 3 )
842 {
843 4943 float ratio = ((float)rate_bits) / frame_bits;
844
845
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4943 if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
846 /*
847 * This path is for steady-state Q-scale tracking
848 * When frame bits fall within the stable range, we still need to adjust
849 * lambda to maintain it like so in a stable fashion (large jumps in lambda
850 * create artifacts and should be avoided), but slowly
851 */
852 2925 ratio = sqrtf(sqrtf(ratio));
853 2925 ratio = av_clipf(ratio, 0.9f, 1.1f);
854 } else {
855 /* Not so fast though */
856 2018 ratio = sqrtf(ratio);
857 }
858 4943 s->lambda = av_clipf(s->lambda * ratio, FLT_EPSILON, 65536.f);
859
860 /* Keep iterating if we must reduce and lambda is in the sky */
861
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4943 if (ratio > 0.9f && ratio < 1.1f) {
862 break;
863 } else {
864
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1427 if (is_mode || ms_mode || tns_mode || pred_mode) {
865
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45 for (i = 0; i < s->chan_map[0]; i++) {
866 // Must restore coeffs
867
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30 chans = tag == TYPE_CPE ? 2 : 1;
868 30 cpe = &s->cpe[i];
869
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70 for (ch = 0; ch < chans; ch++)
870 40 memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
871 }
872 }
873 1427 its++;
874 }
875 } else {
876 break;
877 }
878 } while (1);
879
880
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3805 if (s->options.ltp && s->coder->ltp_insert_new_frame)
881 s->coder->ltp_insert_new_frame(s);
882
883 3805 put_bits(&s->pb, 3, TYPE_END);
884 3805 flush_put_bits(&s->pb);
885
886 3805 s->last_frame_pb_count = put_bits_count(&s->pb);
887 3805 avpkt->size = put_bytes_output(&s->pb);
888
889 3805 s->lambda_sum += s->lambda;
890 3805 s->lambda_count++;
891
892 3805 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
893 &avpkt->duration);
894
895 3805 *got_packet_ptr = 1;
896 3805 return 0;
897 }
898
899 11 static av_cold int aac_encode_end(AVCodecContext *avctx)
900 {
901 11 AACEncContext *s = avctx->priv_data;
902
903
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11 av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_count ? s->lambda_sum / s->lambda_count : NAN);
904
905 11 ff_mdct_end(&s->mdct1024);
906 11 ff_mdct_end(&s->mdct128);
907 11 ff_psy_end(&s->psy);
908 11 ff_lpc_end(&s->lpc);
909
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11 if (s->psypp)
910 11 ff_psy_preprocess_end(s->psypp);
911 11 av_freep(&s->buffer.samples);
912 11 av_freep(&s->cpe);
913 11 av_freep(&s->fdsp);
914 11 ff_af_queue_close(&s->afq);
915 11 return 0;
916 }
917
918 11 static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
919 {
920 11 int ret = 0;
921
922 11 s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
923
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11 if (!s->fdsp)
924 return AVERROR(ENOMEM);
925
926 // window init
927 11 ff_aac_float_common_init();
928
929
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11 if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
930 return ret;
931
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11 if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
932 return ret;
933
934 11 return 0;
935 }
936
937 11 static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
938 {
939 int ch;
940
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11 if (!FF_ALLOCZ_TYPED_ARRAY(s->buffer.samples, s->channels * 3 * 1024) ||
941
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11 !FF_ALLOCZ_TYPED_ARRAY(s->cpe, s->chan_map[0]))
942 return AVERROR(ENOMEM);
943
944
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36 for(ch = 0; ch < s->channels; ch++)
945 25 s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
946
947 11 return 0;
948 }
949
950 11 static av_cold int aac_encode_init(AVCodecContext *avctx)
951 {
952 11 AACEncContext *s = avctx->priv_data;
953 11 int i, ret = 0;
954 const uint8_t *sizes[2];
955 uint8_t grouping[AAC_MAX_CHANNELS];
956 int lengths[2];
957
958 /* Constants */
959 11 s->last_frame_pb_count = 0;
960 11 avctx->frame_size = 1024;
961 11 avctx->initial_padding = 1024;
962
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11 s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
963
964 /* Channel map and unspecified bitrate guessing */
965 11 s->channels = avctx->channels;
966
967 11 s->needs_pce = 1;
968
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25 for (i = 0; i < FF_ARRAY_ELEMS(aac_normal_chan_layouts); i++) {
969
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25 if (avctx->channel_layout == aac_normal_chan_layouts[i]) {
970 11 s->needs_pce = s->options.pce;
971 11 break;
972 }
973 }
974
975
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11 if (s->needs_pce) {
976 char buf[64];
977 for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++)
978 if (avctx->channel_layout == aac_pce_configs[i].layout)
979 break;
980 av_get_channel_layout_string(buf, sizeof(buf), -1, avctx->channel_layout);
981 ERROR_IF(i == FF_ARRAY_ELEMS(aac_pce_configs), "Unsupported channel layout \"%s\"\n", buf);
982 av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout \"%s\"\n", buf);
983 s->pce = aac_pce_configs[i];
984 s->reorder_map = s->pce.reorder_map;
985 s->chan_map = s->pce.config_map;
986 } else {
987 11 s->reorder_map = aac_chan_maps[s->channels - 1];
988 11 s->chan_map = aac_chan_configs[s->channels - 1];
989 }
990
991
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11 if (!avctx->bit_rate) {
992
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9 for (i = 1; i <= s->chan_map[0]; i++) {
993
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9 avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
994
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3 s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */
995 69000 ; /* SCE */
996 }
997 }
998
999 /* Samplerate */
1000
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54 for (i = 0; i < 16; i++)
1001
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54 if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
1002 11 break;
1003 11 s->samplerate_index = i;
1004
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11 ERROR_IF(s->samplerate_index == 16 ||
1005 s->samplerate_index >= ff_aac_swb_size_1024_len ||
1006 s->samplerate_index >= ff_aac_swb_size_128_len,
1007 "Unsupported sample rate %d\n", avctx->sample_rate);
1008
1009 /* Bitrate limiting */
1010
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11 WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
1011 "Too many bits %f > %d per frame requested, clamping to max\n",
1012 1024.0 * avctx->bit_rate / avctx->sample_rate,
1013 6144 * s->channels);
1014
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11 avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
1015 avctx->bit_rate);
1016
1017 /* Profile and option setting */
1018
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11 avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
1019 avctx->profile;
1020
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21 for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
1021
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21 if (avctx->profile == aacenc_profiles[i])
1022 11 break;
1023
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11 if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
1024 avctx->profile = FF_PROFILE_AAC_LOW;
1025 ERROR_IF(s->options.pred,
1026 "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1027 ERROR_IF(s->options.ltp,
1028 "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1029 WARN_IF(s->options.pns,
1030 "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
1031 s->options.pns = 0;
1032
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11 } else if (avctx->profile == FF_PROFILE_AAC_LTP) {
1033 s->options.ltp = 1;
1034 ERROR_IF(s->options.pred,
1035 "Main prediction unavailable in the \"aac_ltp\" profile\n");
1036
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11 } else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
1037 1 s->options.pred = 1;
1038
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1 ERROR_IF(s->options.ltp,
1039 "LTP prediction unavailable in the \"aac_main\" profile\n");
1040
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10 } else if (s->options.ltp) {
1041 avctx->profile = FF_PROFILE_AAC_LTP;
1042 WARN_IF(1,
1043 "Chainging profile to \"aac_ltp\"\n");
1044 ERROR_IF(s->options.pred,
1045 "Main prediction unavailable in the \"aac_ltp\" profile\n");
1046
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10 } else if (s->options.pred) {
1047 avctx->profile = FF_PROFILE_AAC_MAIN;
1048 WARN_IF(1,
1049 "Chainging profile to \"aac_main\"\n");
1050 ERROR_IF(s->options.ltp,
1051 "LTP prediction unavailable in the \"aac_main\" profile\n");
1052 }
1053 11 s->profile = avctx->profile;
1054
1055 /* Coder limitations */
1056 11 s->coder = &ff_aac_coders[s->options.coder];
1057
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11 if (s->options.coder == AAC_CODER_ANMR) {
1058 ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
1059 "The ANMR coder is considered experimental, add -strict -2 to enable!\n");
1060 s->options.intensity_stereo = 0;
1061 s->options.pns = 0;
1062 }
1063
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11 ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
1064 "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
1065
1066 /* M/S introduces horrible artifacts with multichannel files, this is temporary */
1067
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11 if (s->channels > 3)
1068 1 s->options.mid_side = 0;
1069
1070
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11 if ((ret = dsp_init(avctx, s)) < 0)
1071 return ret;
1072
1073
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11 if ((ret = alloc_buffers(avctx, s)) < 0)
1074 return ret;
1075
1076
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11 if ((ret = put_audio_specific_config(avctx)))
1077 return ret;
1078
1079 11 sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
1080 11 sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
1081 11 lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
1082 11 lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
1083
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25 for (i = 0; i < s->chan_map[0]; i++)
1084 14 grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
1085
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11 if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
1086 11 s->chan_map[0], grouping)) < 0)
1087 return ret;
1088 11 s->psypp = ff_psy_preprocess_init(avctx);
1089 11 ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
1090 11 s->random_state = 0x1f2e3d4c;
1091
1092 11 s->abs_pow34 = abs_pow34_v;
1093 11 s->quant_bands = quantize_bands;
1094
1095 if (ARCH_X86)
1096 11 ff_aac_dsp_init_x86(s);
1097
1098 if (HAVE_MIPSDSP)
1099 ff_aac_coder_init_mips(s);
1100
1101 11 ff_af_queue_init(avctx, &s->afq);
1102 11 ff_aac_tableinit();
1103
1104 11 return 0;
1105 }
1106
1107 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1108 static const AVOption aacenc_options[] = {
1109 {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
1110 {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1111 {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1112 {"fast", "Default fast search", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1113 {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
1114 {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1115 {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1116 {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1117 {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1118 {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1119 {"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1120 FF_AAC_PROFILE_OPTS
1121 {NULL}
1122 };
1123
1124 static const AVClass aacenc_class = {
1125 .class_name = "AAC encoder",
1126 .item_name = av_default_item_name,
1127 .option = aacenc_options,
1128 .version = LIBAVUTIL_VERSION_INT,
1129 };
1130
1131 static const AVCodecDefault aac_encode_defaults[] = {
1132 { "b", "0" },
1133 { NULL }
1134 };
1135
1136 const AVCodec ff_aac_encoder = {
1137 .name = "aac",
1138 .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
1139 .type = AVMEDIA_TYPE_AUDIO,
1140 .id = AV_CODEC_ID_AAC,
1141 .priv_data_size = sizeof(AACEncContext),
1142 .init = aac_encode_init,
1143 .encode2 = aac_encode_frame,
1144 .close = aac_encode_end,
1145 .defaults = aac_encode_defaults,
1146 .supported_samplerates = mpeg4audio_sample_rates,
1147 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
1148 .capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
1149 .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1150 AV_SAMPLE_FMT_NONE },
1151 .priv_class = &aacenc_class,
1152 };
1153