FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/aacenc.c
Date: 2022-07-06 18:02:43
Exec Total Coverage
Lines: 553 653 84.7%
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 "codec_internal.h"
39 #include "encode.h"
40 #include "put_bits.h"
41 #include "mpeg4audio.h"
42 #include "sinewin.h"
43 #include "profiles.h"
44 #include "version.h"
45
46 #include "aac.h"
47 #include "aactab.h"
48 #include "aacenc.h"
49 #include "aacenctab.h"
50 #include "aacenc_utils.h"
51
52 #include "psymodel.h"
53
54 static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
55 {
56 int i, j;
57 AACEncContext *s = avctx->priv_data;
58 AACPCEInfo *pce = &s->pce;
59 const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT;
60 const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT;
61
62 put_bits(pb, 4, 0);
63
64 put_bits(pb, 2, avctx->profile);
65 put_bits(pb, 4, s->samplerate_index);
66
67 put_bits(pb, 4, pce->num_ele[0]); /* Front */
68 put_bits(pb, 4, pce->num_ele[1]); /* Side */
69 put_bits(pb, 4, pce->num_ele[2]); /* Back */
70 put_bits(pb, 2, pce->num_ele[3]); /* LFE */
71 put_bits(pb, 3, 0); /* Assoc data */
72 put_bits(pb, 4, 0); /* CCs */
73
74 put_bits(pb, 1, 0); /* Stereo mixdown */
75 put_bits(pb, 1, 0); /* Mono mixdown */
76 put_bits(pb, 1, 0); /* Something else */
77
78 for (i = 0; i < 4; i++) {
79 for (j = 0; j < pce->num_ele[i]; j++) {
80 if (i < 3)
81 put_bits(pb, 1, pce->pairing[i][j]);
82 put_bits(pb, 4, pce->index[i][j]);
83 }
84 }
85
86 align_put_bits(pb);
87 put_bits(pb, 8, strlen(aux_data));
88 ff_put_string(pb, aux_data, 0);
89 }
90
91 /**
92 * Make AAC audio config object.
93 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
94 */
95 11 static int put_audio_specific_config(AVCodecContext *avctx)
96 {
97 PutBitContext pb;
98 11 AACEncContext *s = avctx->priv_data;
99 11 int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0));
100 11 const int max_size = 32;
101
102 11 avctx->extradata = av_mallocz(max_size);
103
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11 if (!avctx->extradata)
104 return AVERROR(ENOMEM);
105
106 11 init_put_bits(&pb, avctx->extradata, max_size);
107 11 put_bits(&pb, 5, s->profile+1); //profile
108 11 put_bits(&pb, 4, s->samplerate_index); //sample rate index
109 11 put_bits(&pb, 4, channels);
110 //GASpecificConfig
111 11 put_bits(&pb, 1, 0); //frame length - 1024 samples
112 11 put_bits(&pb, 1, 0); //does not depend on core coder
113 11 put_bits(&pb, 1, 0); //is not extension
114
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11 if (s->needs_pce)
115 put_pce(&pb, avctx);
116
117 //Explicitly Mark SBR absent
118 11 put_bits(&pb, 11, 0x2b7); //sync extension
119 11 put_bits(&pb, 5, AOT_SBR);
120 11 put_bits(&pb, 1, 0);
121 11 flush_put_bits(&pb);
122 11 avctx->extradata_size = put_bytes_output(&pb);
123
124 11 return 0;
125 }
126
127 10135 void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
128 {
129 10135 ++s->quantize_band_cost_cache_generation;
130
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10135 if (s->quantize_band_cost_cache_generation == 0) {
131 memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
132 s->quantize_band_cost_cache_generation = 1;
133 }
134 10135 }
135
136 #define WINDOW_FUNC(type) \
137 static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
138 SingleChannelElement *sce, \
139 const float *audio)
140
141 6899 WINDOW_FUNC(only_long)
142 {
143
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6899 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
144
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6899 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
145 6899 float *out = sce->ret_buf;
146
147 6899 fdsp->vector_fmul (out, audio, lwindow, 1024);
148 6899 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
149 6899 }
150
151 119 WINDOW_FUNC(long_start)
152 {
153
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119 const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
154
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119 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
155 119 float *out = sce->ret_buf;
156
157 119 fdsp->vector_fmul(out, audio, lwindow, 1024);
158 119 memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
159 119 fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
160 119 memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
161 119 }
162
163 102 WINDOW_FUNC(long_stop)
164 {
165
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102 const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
166
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102 const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
167 102 float *out = sce->ret_buf;
168
169 102 memset(out, 0, sizeof(out[0]) * 448);
170 102 fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
171 102 memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
172 102 fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
173 102 }
174
175 164 WINDOW_FUNC(eight_short)
176 {
177
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164 const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
178
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164 const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
179 164 const float *in = audio + 448;
180 164 float *out = sce->ret_buf;
181 int w;
182
183
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1476 for (w = 0; w < 8; w++) {
184
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1312 fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
185 1312 out += 128;
186 1312 in += 128;
187 1312 fdsp->vector_fmul_reverse(out, in, swindow, 128);
188 1312 out += 128;
189 }
190 164 }
191
192 static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
193 SingleChannelElement *sce,
194 const float *audio) = {
195 [ONLY_LONG_SEQUENCE] = apply_only_long_window,
196 [LONG_START_SEQUENCE] = apply_long_start_window,
197 [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
198 [LONG_STOP_SEQUENCE] = apply_long_stop_window
199 };
200
201 7284 static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
202 float *audio)
203 {
204 int i;
205 7284 const float *output = sce->ret_buf;
206
207 7284 apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
208
209
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7284 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
210 7120 s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
211 else
212
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1476 for (i = 0; i < 1024; i += 128)
213 1312 s->mdct128.mdct_calc(&s->mdct128, &sce->coeffs[i], output + i*2);
214 7284 memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
215 7284 memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
216 7284 }
217
218 /**
219 * Encode ics_info element.
220 * @see Table 4.6 (syntax of ics_info)
221 */
222 5693 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
223 {
224 int w;
225
226 5693 put_bits(&s->pb, 1, 0); // ics_reserved bit
227 5693 put_bits(&s->pb, 2, info->window_sequence[0]);
228 5693 put_bits(&s->pb, 1, info->use_kb_window[0]);
229
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5693 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
230 5513 put_bits(&s->pb, 6, info->max_sfb);
231 5513 put_bits(&s->pb, 1, !!info->predictor_present);
232 } else {
233 180 put_bits(&s->pb, 4, info->max_sfb);
234
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1440 for (w = 1; w < 8; w++)
235 1260 put_bits(&s->pb, 1, !info->group_len[w]);
236 }
237 5693 }
238
239 /**
240 * Encode MS data.
241 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
242 */
243 4538 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
244 {
245 int i, w;
246
247 4538 put_bits(pb, 2, cpe->ms_mode);
248
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4538 if (cpe->ms_mode == 1)
249
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823 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
250
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19581 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
251 19160 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
252 4538 }
253
254 /**
255 * Produce integer coefficients from scalefactors provided by the model.
256 */
257 5448 static void adjust_frame_information(ChannelElement *cpe, int chans)
258 {
259 int i, w, w2, g, ch;
260 int maxsfb, cmaxsfb;
261
262
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15679 for (ch = 0; ch < chans; ch++) {
263 10231 IndividualChannelStream *ics = &cpe->ch[ch].ics;
264 10231 maxsfb = 0;
265 10231 cpe->ch[ch].pulse.num_pulse = 0;
266
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21114 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
267
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22885 for (w2 = 0; w2 < ics->group_len[w]; w2++) {
268
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29945 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
269 ;
270 12002 maxsfb = FFMAX(maxsfb, cmaxsfb);
271 }
272 }
273 10231 ics->max_sfb = maxsfb;
274
275 //adjust zero bands for window groups
276
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21114 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
277
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498183 for (g = 0; g < ics->max_sfb; g++) {
278 487300 i = 1;
279
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490475 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
280
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487520 if (!cpe->ch[ch].zeroes[w2*16 + g]) {
281 484345 i = 0;
282 484345 break;
283 }
284 }
285 487300 cpe->ch[ch].zeroes[w*16 + g] = i;
286 }
287 }
288 }
289
290
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5448 if (chans > 1 && cpe->common_window) {
291 4538 IndividualChannelStream *ics0 = &cpe->ch[0].ics;
292 4538 IndividualChannelStream *ics1 = &cpe->ch[1].ics;
293 4538 int msc = 0;
294 4538 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
295 4538 ics1->max_sfb = ics0->max_sfb;
296
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9587 for (w = 0; w < ics0->num_windows*16; w += 16)
297
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225553 for (i = 0; i < ics0->max_sfb; i++)
298
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220504 if (cpe->ms_mask[w+i])
299 24406 msc++;
300
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4538 if (msc == 0 || ics0->max_sfb == 0)
301 3749 cpe->ms_mode = 0;
302 else
303
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789 cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
304 }
305 5448 }
306
307 1334 static void apply_intensity_stereo(ChannelElement *cpe)
308 {
309 int w, w2, g, i;
310 1334 IndividualChannelStream *ics = &cpe->ch[0].ics;
311
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1334 if (!cpe->common_window)
312 707 return;
313
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1288 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
314
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1386 for (w2 = 0; w2 < ics->group_len[w]; w2++) {
315 725 int start = (w+w2) * 128;
316
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32330 for (g = 0; g < ics->num_swb; g++) {
317 31605 int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
318 31605 float scale = cpe->ch[0].is_ener[w*16+g];
319
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31605 if (!cpe->is_mask[w*16 + g]) {
320 22930 start += ics->swb_sizes[g];
321 22930 continue;
322 }
323
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8675 if (cpe->ms_mask[w*16 + g])
324 2383 p *= -1;
325
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300811 for (i = 0; i < ics->swb_sizes[g]; i++) {
326 292136 float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
327 292136 cpe->ch[0].coeffs[start+i] = sum;
328 292136 cpe->ch[1].coeffs[start+i] = 0.0f;
329 }
330 8675 start += ics->swb_sizes[g];
331 }
332 }
333 }
334 }
335
336 1050 static void apply_mid_side_stereo(ChannelElement *cpe)
337 {
338 int w, w2, g, i;
339 1050 IndividualChannelStream *ics = &cpe->ch[0].ics;
340
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1050 if (!cpe->common_window)
341 548 return;
342
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1038 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
343
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1136 for (w2 = 0; w2 < ics->group_len[w]; w2++) {
344 600 int start = (w+w2) * 128;
345
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26080 for (g = 0; g < ics->num_swb; g++) {
346 /* ms_mask can be used for other purposes in PNS and I/S,
347 * so must not apply M/S if any band uses either, even if
348 * ms_mask is set.
349 */
350
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25480 if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
351
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22915 || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
352
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22915 || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
353 2565 start += ics->swb_sizes[g];
354 2565 continue;
355 }
356
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484399 for (i = 0; i < ics->swb_sizes[g]; i++) {
357 461484 float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
358 461484 float R = L - cpe->ch[1].coeffs[start+i];
359 461484 cpe->ch[0].coeffs[start+i] = L;
360 461484 cpe->ch[1].coeffs[start+i] = R;
361 }
362 22915 start += ics->swb_sizes[g];
363 }
364 }
365 }
366 }
367
368 /**
369 * Encode scalefactor band coding type.
370 */
371 10231 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
372 {
373 int w;
374
375
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10231 if (s->coder->set_special_band_scalefactors)
376 10231 s->coder->set_special_band_scalefactors(s, sce);
377
378
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21114 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
379 10883 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
380 10231 }
381
382 /**
383 * Encode scalefactors.
384 */
385 10231 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
386 SingleChannelElement *sce)
387 {
388 10231 int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
389 10231 int off_is = 0, noise_flag = 1;
390 int i, w;
391
392
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21114 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
393
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498188 for (i = 0; i < sce->ics.max_sfb; i++) {
394
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487305 if (!sce->zeroes[w*16 + i]) {
395
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476374 if (sce->band_type[w*16 + i] == NOISE_BT) {
396 6375 diff = sce->sf_idx[w*16 + i] - off_pns;
397 6375 off_pns = sce->sf_idx[w*16 + i];
398
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6375 if (noise_flag-- > 0) {
399 1122 put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
400 1122 continue;
401 }
402
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469999 } else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
403
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463673 sce->band_type[w*16 + i] == INTENSITY_BT2) {
404 8564 diff = sce->sf_idx[w*16 + i] - off_is;
405 8564 off_is = sce->sf_idx[w*16 + i];
406 } else {
407 461435 diff = sce->sf_idx[w*16 + i] - off_sf;
408 461435 off_sf = sce->sf_idx[w*16 + i];
409 }
410 475252 diff += SCALE_DIFF_ZERO;
411
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475252 av_assert0(diff >= 0 && diff <= 120);
412 475252 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
413 }
414 }
415 }
416 10231 }
417
418 /**
419 * Encode pulse data.
420 */
421 10231 static void encode_pulses(AACEncContext *s, Pulse *pulse)
422 {
423 int i;
424
425 10231 put_bits(&s->pb, 1, !!pulse->num_pulse);
426
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10231 if (!pulse->num_pulse)
427 10231 return;
428
429 put_bits(&s->pb, 2, pulse->num_pulse - 1);
430 put_bits(&s->pb, 6, pulse->start);
431 for (i = 0; i < pulse->num_pulse; i++) {
432 put_bits(&s->pb, 5, pulse->pos[i]);
433 put_bits(&s->pb, 4, pulse->amp[i]);
434 }
435 }
436
437 /**
438 * Encode spectral coefficients processed by psychoacoustic model.
439 */
440 10231 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
441 {
442 int start, i, w, w2;
443
444
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21114 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
445 10883 start = 0;
446
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498188 for (i = 0; i < sce->ics.max_sfb; i++) {
447
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487305 if (sce->zeroes[w*16 + i]) {
448 10931 start += sce->ics.swb_sizes[i];
449 10931 continue;
450 }
451
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960754 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
452 484380 s->coder->quantize_and_encode_band(s, &s->pb,
453 484380 &sce->coeffs[start + w2*128],
454 484380 NULL, sce->ics.swb_sizes[i],
455 484380 sce->sf_idx[w*16 + i],
456 484380 sce->band_type[w*16 + i],
457 s->lambda,
458 484380 sce->ics.window_clipping[w]);
459 }
460 476374 start += sce->ics.swb_sizes[i];
461 }
462 }
463 10231 }
464
465 /**
466 * Downscale spectral coefficients for near-clipping windows to avoid artifacts
467 */
468 7284 static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
469 {
470 int start, i, j, w;
471
472
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7284 if (sce->ics.clip_avoidance_factor < 1.0f) {
473
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392 for (w = 0; w < sce->ics.num_windows; w++) {
474 224 start = 0;
475
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8690 for (i = 0; i < sce->ics.max_sfb; i++) {
476 8466 float *swb_coeffs = &sce->coeffs[start + w*128];
477
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168658 for (j = 0; j < sce->ics.swb_sizes[i]; j++)
478 160192 swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
479 8466 start += sce->ics.swb_sizes[i];
480 }
481 }
482 }
483 7284 }
484
485 /**
486 * Encode one channel of audio data.
487 */
488 10231 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
489 SingleChannelElement *sce,
490 int common_window)
491 {
492 10231 put_bits(&s->pb, 8, sce->sf_idx[0]);
493
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10231 if (!common_window) {
494 1155 put_ics_info(s, &sce->ics);
495
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1155 if (s->coder->encode_main_pred)
496 1155 s->coder->encode_main_pred(s, sce);
497
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1155 if (s->coder->encode_ltp_info)
498 1155 s->coder->encode_ltp_info(s, sce, 0);
499 }
500 10231 encode_band_info(s, sce);
501 10231 encode_scale_factors(avctx, s, sce);
502 10231 encode_pulses(s, &sce->pulse);
503 10231 put_bits(&s->pb, 1, !!sce->tns.present);
504
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10231 if (s->coder->encode_tns_info)
505 10231 s->coder->encode_tns_info(s, sce);
506 10231 put_bits(&s->pb, 1, 0); //ssr
507 10231 encode_spectral_coeffs(s, sce);
508 10231 return 0;
509 }
510
511 /**
512 * Write some auxiliary information about the created AAC file.
513 */
514 static void put_bitstream_info(AACEncContext *s, const char *name)
515 {
516 int i, namelen, padbits;
517
518 namelen = strlen(name) + 2;
519 put_bits(&s->pb, 3, TYPE_FIL);
520 put_bits(&s->pb, 4, FFMIN(namelen, 15));
521 if (namelen >= 15)
522 put_bits(&s->pb, 8, namelen - 14);
523 put_bits(&s->pb, 4, 0); //extension type - filler
524 padbits = -put_bits_count(&s->pb) & 7;
525 align_put_bits(&s->pb);
526 for (i = 0; i < namelen - 2; i++)
527 put_bits(&s->pb, 8, name[i]);
528 put_bits(&s->pb, 12 - padbits, 0);
529 }
530
531 /*
532 * Copy input samples.
533 * Channels are reordered from libavcodec's default order to AAC order.
534 */
535 3816 static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
536 {
537 int ch;
538
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3816 int end = 2048 + (frame ? frame->nb_samples : 0);
539 3816 const uint8_t *channel_map = s->reorder_map;
540
541 /* copy and remap input samples */
542
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11125 for (ch = 0; ch < s->channels; ch++) {
543 /* copy last 1024 samples of previous frame to the start of the current frame */
544 7309 memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
545
546 /* copy new samples and zero any remaining samples */
547
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7309 if (frame) {
548 7259 memcpy(&s->planar_samples[ch][2048],
549 7259 frame->extended_data[channel_map[ch]],
550 7259 frame->nb_samples * sizeof(s->planar_samples[0][0]));
551 }
552 7309 memset(&s->planar_samples[ch][end], 0,
553 7309 (3072 - end) * sizeof(s->planar_samples[0][0]));
554 }
555 3816 }
556
557 3827 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
558 const AVFrame *frame, int *got_packet_ptr)
559 {
560 3827 AACEncContext *s = avctx->priv_data;
561 3827 float **samples = s->planar_samples, *samples2, *la, *overlap;
562 ChannelElement *cpe;
563 SingleChannelElement *sce;
564 IndividualChannelStream *ics;
565 int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
566 int target_bits, rate_bits, too_many_bits, too_few_bits;
567 3827 int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
568 int chan_el_counter[4];
569 FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
570
571 /* add current frame to queue */
572
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3827 if (frame) {
573
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3794 if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
574 return ret;
575 } else {
576
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33 if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
577 11 return 0;
578 }
579
580 3816 copy_input_samples(s, frame);
581
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3816 if (s->psypp)
582 3816 ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
583
584
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3816 if (!avctx->frame_number)
585 11 return 0;
586
587 3805 start_ch = 0;
588
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7754 for (i = 0; i < s->chan_map[0]; i++) {
589 3949 FFPsyWindowInfo* wi = windows + start_ch;
590 3949 tag = s->chan_map[i+1];
591
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3949 chans = tag == TYPE_CPE ? 2 : 1;
592 3949 cpe = &s->cpe[i];
593
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11233 for (ch = 0; ch < chans; ch++) {
594 int k;
595 float clip_avoidance_factor;
596 7284 sce = &cpe->ch[ch];
597 7284 ics = &sce->ics;
598 7284 s->cur_channel = start_ch + ch;
599 7284 overlap = &samples[s->cur_channel][0];
600 7284 samples2 = overlap + 1024;
601 7284 la = samples2 + (448+64);
602
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7284 if (!frame)
603 50 la = NULL;
604
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7284 if (tag == TYPE_LFE) {
605 48 wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
606 48 wi[ch].window_shape = 0;
607 48 wi[ch].num_windows = 1;
608 48 wi[ch].grouping[0] = 1;
609 48 wi[ch].clipping[0] = 0;
610
611 /* Only the lowest 12 coefficients are used in a LFE channel.
612 * The expression below results in only the bottom 8 coefficients
613 * being used for 11.025kHz to 16kHz sample rates.
614 */
615
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48 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
616 } else {
617 7236 wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
618 7236 ics->window_sequence[0]);
619 }
620 7284 ics->window_sequence[1] = ics->window_sequence[0];
621 7284 ics->window_sequence[0] = wi[ch].window_type[0];
622 7284 ics->use_kb_window[1] = ics->use_kb_window[0];
623 7284 ics->use_kb_window[0] = wi[ch].window_shape;
624 7284 ics->num_windows = wi[ch].num_windows;
625
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7284 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
626
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7284 ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
627 7284 ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb);
628 14568 ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
629
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7284 ff_swb_offset_128 [s->samplerate_index]:
630 7120 ff_swb_offset_1024[s->samplerate_index];
631 14568 ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
632
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7284 ff_tns_max_bands_128 [s->samplerate_index]:
633 7120 ff_tns_max_bands_1024[s->samplerate_index];
634
635
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15716 for (w = 0; w < ics->num_windows; w++)
636 8432 ics->group_len[w] = wi[ch].grouping[w];
637
638 /* Calculate input sample maximums and evaluate clipping risk */
639 7284 clip_avoidance_factor = 0.0f;
640
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15716 for (w = 0; w < ics->num_windows; w++) {
641 8432 const float *wbuf = overlap + w * 128;
642 8432 const int wlen = 2048 / ics->num_windows;
643 8432 float max = 0;
644 int j;
645 /* mdct input is 2 * output */
646
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14926064 for (j = 0; j < wlen; j++)
647
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14917632 max = FFMAX(max, fabsf(wbuf[j]));
648 8432 wi[ch].clipping[w] = max;
649 }
650
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15716 for (w = 0; w < ics->num_windows; w++) {
651
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8432 if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
652 176 ics->window_clipping[w] = 1;
653
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176 clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
654 } else {
655 8256 ics->window_clipping[w] = 0;
656 }
657 }
658
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7284 if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
659 168 ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
660 } else {
661 7116 ics->clip_avoidance_factor = 1.0f;
662 }
663
664 7284 apply_window_and_mdct(s, sce, overlap);
665
666
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7284 if (s->options.ltp && s->coder->update_ltp) {
667 s->coder->update_ltp(s, sce);
668 apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
669 s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
670 }
671
672
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7466100 for (k = 0; k < 1024; k++) {
673
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7458816 if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
674 av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
675 return AVERROR(EINVAL);
676 }
677 }
678 7284 avoid_clipping(s, sce);
679 }
680 3949 start_ch += chans;
681 }
682
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3805 if ((ret = ff_alloc_packet(avctx, avpkt, 8192 * s->channels)) < 0)
683 return ret;
684 3805 frame_bits = its = 0;
685 do {
686 5232 init_put_bits(&s->pb, avpkt->data, avpkt->size);
687
688
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5232 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
689 put_bitstream_info(s, LIBAVCODEC_IDENT);
690 5232 start_ch = 0;
691 5232 target_bits = 0;
692 5232 memset(chan_el_counter, 0, sizeof(chan_el_counter));
693
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10680 for (i = 0; i < s->chan_map[0]; i++) {
694 5448 FFPsyWindowInfo* wi = windows + start_ch;
695 const float *coeffs[2];
696 5448 tag = s->chan_map[i+1];
697
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5448 chans = tag == TYPE_CPE ? 2 : 1;
698 5448 cpe = &s->cpe[i];
699 5448 cpe->common_window = 0;
700 5448 memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
701 5448 memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
702 5448 put_bits(&s->pb, 3, tag);
703 5448 put_bits(&s->pb, 4, chan_el_counter[tag]++);
704
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15679 for (ch = 0; ch < chans; ch++) {
705 10231 sce = &cpe->ch[ch];
706 10231 coeffs[ch] = sce->coeffs;
707 10231 sce->ics.predictor_present = 0;
708 10231 sce->ics.ltp.present = 0;
709 10231 memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));
710 10231 memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));
711 10231 memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
712
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1319799 for (w = 0; w < 128; w++)
713
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1309568 if (sce->band_type[w] > RESERVED_BT)
714 14871 sce->band_type[w] = 0;
715 }
716 5448 s->psy.bitres.alloc = -1;
717 5448 s->psy.bitres.bits = s->last_frame_pb_count / s->channels;
718 5448 s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
719
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5448 if (s->psy.bitres.alloc > 0) {
720 /* Lambda unused here on purpose, we need to take psy's unscaled allocation */
721 10896 target_bits += s->psy.bitres.alloc
722
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5448 * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
723 5448 s->psy.bitres.alloc /= chans;
724 }
725 5448 s->cur_type = tag;
726
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15679 for (ch = 0; ch < chans; ch++) {
727 10231 s->cur_channel = start_ch + ch;
728
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10231 if (s->options.pns && s->coder->mark_pns)
729 2003 s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
730 10231 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
731 }
732
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✓ Branch 1 taken 665 times.
5448 if (chans > 1
733
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✓ Branch 1 taken 236 times.
4783 && wi[0].window_type[0] == wi[1].window_type[0]
734
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4547 && wi[0].window_shape == wi[1].window_shape) {
735
736 4547 cpe->common_window = 1;
737
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9596 for (w = 0; w < wi[0].num_windows; w++) {
738
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5058 if (wi[0].grouping[w] != wi[1].grouping[w]) {
739 9 cpe->common_window = 0;
740 9 break;
741 }
742 }
743 }
744
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15679 for (ch = 0; ch < chans; ch++) { /* TNS and PNS */
745 10231 sce = &cpe->ch[ch];
746 10231 s->cur_channel = start_ch + ch;
747
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10231 if (s->options.tns && s->coder->search_for_tns)
748 2003 s->coder->search_for_tns(s, sce);
749
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10231 if (s->options.tns && s->coder->apply_tns_filt)
750 2003 s->coder->apply_tns_filt(s, sce);
751
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10231 if (sce->tns.present)
752 40 tns_mode = 1;
753
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10231 if (s->options.pns && s->coder->search_for_pns)
754 2003 s->coder->search_for_pns(s, avctx, sce);
755 }
756 5448 s->cur_channel = start_ch;
757
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5448 if (s->options.intensity_stereo) { /* Intensity Stereo */
758
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1334 if (s->coder->search_for_is)
759 1334 s->coder->search_for_is(s, avctx, cpe);
760
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1334 if (cpe->is_mode) is_mode = 1;
761 1334 apply_intensity_stereo(cpe);
762 }
763
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5448 if (s->options.pred) { /* Prediction */
764
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1248 for (ch = 0; ch < chans; ch++) {
765 832 sce = &cpe->ch[ch];
766 832 s->cur_channel = start_ch + ch;
767
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832 if (s->options.pred && s->coder->search_for_pred)
768 832 s->coder->search_for_pred(s, sce);
769
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832 if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
770 }
771
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416 if (s->coder->adjust_common_pred)
772 416 s->coder->adjust_common_pred(s, cpe);
773
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1248 for (ch = 0; ch < chans; ch++) {
774 832 sce = &cpe->ch[ch];
775 832 s->cur_channel = start_ch + ch;
776
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832 if (s->options.pred && s->coder->apply_main_pred)
777 832 s->coder->apply_main_pred(s, sce);
778 }
779 416 s->cur_channel = start_ch;
780 }
781
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5448 if (s->options.mid_side) { /* Mid/Side stereo */
782
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1050 if (s->options.mid_side == -1 && s->coder->search_for_ms)
783 630 s->coder->search_for_ms(s, cpe);
784
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420 else if (cpe->common_window)
785 393 memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
786 1050 apply_mid_side_stereo(cpe);
787 }
788 5448 adjust_frame_information(cpe, chans);
789
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5448 if (s->options.ltp) { /* LTP */
790 for (ch = 0; ch < chans; ch++) {
791 sce = &cpe->ch[ch];
792 s->cur_channel = start_ch + ch;
793 if (s->coder->search_for_ltp)
794 s->coder->search_for_ltp(s, sce, cpe->common_window);
795 if (sce->ics.ltp.present) pred_mode = 1;
796 }
797 s->cur_channel = start_ch;
798 if (s->coder->adjust_common_ltp)
799 s->coder->adjust_common_ltp(s, cpe);
800 }
801
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5448 if (chans == 2) {
802 4783 put_bits(&s->pb, 1, cpe->common_window);
803
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4783 if (cpe->common_window) {
804 4538 put_ics_info(s, &cpe->ch[0].ics);
805
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4538 if (s->coder->encode_main_pred)
806 4538 s->coder->encode_main_pred(s, &cpe->ch[0]);
807
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4538 if (s->coder->encode_ltp_info)
808 4538 s->coder->encode_ltp_info(s, &cpe->ch[0], 1);
809 4538 encode_ms_info(&s->pb, cpe);
810
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4538 if (cpe->ms_mode) ms_mode = 1;
811 }
812 }
813
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15679 for (ch = 0; ch < chans; ch++) {
814 10231 s->cur_channel = start_ch + ch;
815 10231 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
816 }
817 5448 start_ch += chans;
818 }
819
820
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5232 if (avctx->flags & AV_CODEC_FLAG_QSCALE) {
821 /* When using a constant Q-scale, don't mess with lambda */
822 break;
823 }
824
825 /* rate control stuff
826 * allow between the nominal bitrate, and what psy's bit reservoir says to target
827 * but drift towards the nominal bitrate always
828 */
829 5232 frame_bits = put_bits_count(&s->pb);
830 5232 rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
831
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5232 rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
832 5232 too_many_bits = FFMAX(target_bits, rate_bits);
833
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5232 too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
834 5232 too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
835
836 /* When using ABR, be strict (but only for increasing) */
837 5232 too_few_bits = too_few_bits - too_few_bits/8;
838 5232 too_many_bits = too_many_bits + too_many_bits/2;
839
840
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5232 if ( its == 0 /* for steady-state Q-scale tracking */
841
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1427 || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
842
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289 || frame_bits >= 6144 * s->channels - 3 )
843 {
844 4943 float ratio = ((float)rate_bits) / frame_bits;
845
846
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4943 if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
847 /*
848 * This path is for steady-state Q-scale tracking
849 * When frame bits fall within the stable range, we still need to adjust
850 * lambda to maintain it like so in a stable fashion (large jumps in lambda
851 * create artifacts and should be avoided), but slowly
852 */
853 2925 ratio = sqrtf(sqrtf(ratio));
854 2925 ratio = av_clipf(ratio, 0.9f, 1.1f);
855 } else {
856 /* Not so fast though */
857 2018 ratio = sqrtf(ratio);
858 }
859 4943 s->lambda = av_clipf(s->lambda * ratio, FLT_EPSILON, 65536.f);
860
861 /* Keep iterating if we must reduce and lambda is in the sky */
862
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4943 if (ratio > 0.9f && ratio < 1.1f) {
863 break;
864 } else {
865
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1427 if (is_mode || ms_mode || tns_mode || pred_mode) {
866
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45 for (i = 0; i < s->chan_map[0]; i++) {
867 // Must restore coeffs
868
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30 chans = tag == TYPE_CPE ? 2 : 1;
869 30 cpe = &s->cpe[i];
870
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70 for (ch = 0; ch < chans; ch++)
871 40 memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
872 }
873 }
874 1427 its++;
875 }
876 } else {
877 break;
878 }
879 } while (1);
880
881
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3805 if (s->options.ltp && s->coder->ltp_insert_new_frame)
882 s->coder->ltp_insert_new_frame(s);
883
884 3805 put_bits(&s->pb, 3, TYPE_END);
885 3805 flush_put_bits(&s->pb);
886
887 3805 s->last_frame_pb_count = put_bits_count(&s->pb);
888 3805 avpkt->size = put_bytes_output(&s->pb);
889
890 3805 s->lambda_sum += s->lambda;
891 3805 s->lambda_count++;
892
893 3805 ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
894 &avpkt->duration);
895
896 3805 *got_packet_ptr = 1;
897 3805 return 0;
898 }
899
900 11 static av_cold int aac_encode_end(AVCodecContext *avctx)
901 {
902 11 AACEncContext *s = avctx->priv_data;
903
904
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11 av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_count ? s->lambda_sum / s->lambda_count : NAN);
905
906 11 ff_mdct_end(&s->mdct1024);
907 11 ff_mdct_end(&s->mdct128);
908 11 ff_psy_end(&s->psy);
909 11 ff_lpc_end(&s->lpc);
910
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11 if (s->psypp)
911 11 ff_psy_preprocess_end(s->psypp);
912 11 av_freep(&s->buffer.samples);
913 11 av_freep(&s->cpe);
914 11 av_freep(&s->fdsp);
915 11 ff_af_queue_close(&s->afq);
916 11 return 0;
917 }
918
919 11 static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
920 {
921 11 int ret = 0;
922
923 11 s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
924
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11 if (!s->fdsp)
925 return AVERROR(ENOMEM);
926
927 // window init
928 11 ff_aac_float_common_init();
929
930
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11 if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
931 return ret;
932
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11 if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
933 return ret;
934
935 11 return 0;
936 }
937
938 11 static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
939 {
940 int ch;
941
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11 if (!FF_ALLOCZ_TYPED_ARRAY(s->buffer.samples, s->channels * 3 * 1024) ||
942
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11 !FF_ALLOCZ_TYPED_ARRAY(s->cpe, s->chan_map[0]))
943 return AVERROR(ENOMEM);
944
945
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36 for(ch = 0; ch < s->channels; ch++)
946 25 s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
947
948 11 return 0;
949 }
950
951 11 static av_cold int aac_encode_init(AVCodecContext *avctx)
952 {
953 11 AACEncContext *s = avctx->priv_data;
954 11 int i, ret = 0;
955 const uint8_t *sizes[2];
956 uint8_t grouping[AAC_MAX_CHANNELS];
957 int lengths[2];
958
959 /* Constants */
960 11 s->last_frame_pb_count = 0;
961 11 avctx->frame_size = 1024;
962 11 avctx->initial_padding = 1024;
963
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11 s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
964
965 /* Channel map and unspecified bitrate guessing */
966 11 s->channels = avctx->ch_layout.nb_channels;
967
968 11 s->needs_pce = 1;
969
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25 for (i = 0; i < FF_ARRAY_ELEMS(aac_normal_chan_layouts); i++) {
970
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25 if (!av_channel_layout_compare(&avctx->ch_layout, &aac_normal_chan_layouts[i])) {
971 11 s->needs_pce = s->options.pce;
972 11 break;
973 }
974 }
975
976
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11 if (s->needs_pce) {
977 char buf[64];
978 for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++)
979 if (!av_channel_layout_compare(&avctx->ch_layout, &aac_pce_configs[i].layout))
980 break;
981 av_channel_layout_describe(&avctx->ch_layout, buf, sizeof(buf));
982 if (i == FF_ARRAY_ELEMS(aac_pce_configs)) {
983 av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout \"%s\"\n", buf);
984 return AVERROR(EINVAL);
985 }
986 av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout \"%s\"\n", buf);
987 s->pce = aac_pce_configs[i];
988 s->reorder_map = s->pce.reorder_map;
989 s->chan_map = s->pce.config_map;
990 } else {
991 11 s->reorder_map = aac_chan_maps[s->channels - 1];
992 11 s->chan_map = aac_chan_configs[s->channels - 1];
993 }
994
995
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11 if (!avctx->bit_rate) {
996
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9 for (i = 1; i <= s->chan_map[0]; i++) {
997
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9 avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
998
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3 s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */
999 69000 ; /* SCE */
1000 }
1001 }
1002
1003 /* Samplerate */
1004
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54 for (i = 0; i < 16; i++)
1005
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54 if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
1006 11 break;
1007 11 s->samplerate_index = i;
1008
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11 ERROR_IF(s->samplerate_index == 16 ||
1009 s->samplerate_index >= ff_aac_swb_size_1024_len ||
1010 s->samplerate_index >= ff_aac_swb_size_128_len,
1011 "Unsupported sample rate %d\n", avctx->sample_rate);
1012
1013 /* Bitrate limiting */
1014
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11 WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
1015 "Too many bits %f > %d per frame requested, clamping to max\n",
1016 1024.0 * avctx->bit_rate / avctx->sample_rate,
1017 6144 * s->channels);
1018
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11 avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
1019 avctx->bit_rate);
1020
1021 /* Profile and option setting */
1022
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11 avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
1023 avctx->profile;
1024
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21 for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
1025
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21 if (avctx->profile == aacenc_profiles[i])
1026 11 break;
1027
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11 if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
1028 avctx->profile = FF_PROFILE_AAC_LOW;
1029 ERROR_IF(s->options.pred,
1030 "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1031 ERROR_IF(s->options.ltp,
1032 "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
1033 WARN_IF(s->options.pns,
1034 "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
1035 s->options.pns = 0;
1036
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11 } else if (avctx->profile == FF_PROFILE_AAC_LTP) {
1037 s->options.ltp = 1;
1038 ERROR_IF(s->options.pred,
1039 "Main prediction unavailable in the \"aac_ltp\" profile\n");
1040
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11 } else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
1041 1 s->options.pred = 1;
1042
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1 ERROR_IF(s->options.ltp,
1043 "LTP prediction unavailable in the \"aac_main\" profile\n");
1044
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10 } else if (s->options.ltp) {
1045 avctx->profile = FF_PROFILE_AAC_LTP;
1046 WARN_IF(1,
1047 "Chainging profile to \"aac_ltp\"\n");
1048 ERROR_IF(s->options.pred,
1049 "Main prediction unavailable in the \"aac_ltp\" profile\n");
1050
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10 } else if (s->options.pred) {
1051 avctx->profile = FF_PROFILE_AAC_MAIN;
1052 WARN_IF(1,
1053 "Chainging profile to \"aac_main\"\n");
1054 ERROR_IF(s->options.ltp,
1055 "LTP prediction unavailable in the \"aac_main\" profile\n");
1056 }
1057 11 s->profile = avctx->profile;
1058
1059 /* Coder limitations */
1060 11 s->coder = &ff_aac_coders[s->options.coder];
1061
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11 if (s->options.coder == AAC_CODER_ANMR) {
1062 ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
1063 "The ANMR coder is considered experimental, add -strict -2 to enable!\n");
1064 s->options.intensity_stereo = 0;
1065 s->options.pns = 0;
1066 }
1067
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11 ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
1068 "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
1069
1070 /* M/S introduces horrible artifacts with multichannel files, this is temporary */
1071
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11 if (s->channels > 3)
1072 1 s->options.mid_side = 0;
1073
1074
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11 if ((ret = dsp_init(avctx, s)) < 0)
1075 return ret;
1076
1077
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11 if ((ret = alloc_buffers(avctx, s)) < 0)
1078 return ret;
1079
1080
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11 if ((ret = put_audio_specific_config(avctx)))
1081 return ret;
1082
1083 11 sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
1084 11 sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
1085 11 lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
1086 11 lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
1087
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25 for (i = 0; i < s->chan_map[0]; i++)
1088 14 grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
1089
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11 if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
1090 11 s->chan_map[0], grouping)) < 0)
1091 return ret;
1092 11 s->psypp = ff_psy_preprocess_init(avctx);
1093 11 ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
1094 11 s->random_state = 0x1f2e3d4c;
1095
1096 11 s->abs_pow34 = abs_pow34_v;
1097 11 s->quant_bands = quantize_bands;
1098
1099 #if ARCH_X86
1100 11 ff_aac_dsp_init_x86(s);
1101 #endif
1102
1103 #if HAVE_MIPSDSP
1104 ff_aac_coder_init_mips(s);
1105 #endif
1106
1107 11 ff_af_queue_init(avctx, &s->afq);
1108 11 ff_aac_tableinit();
1109
1110 11 return 0;
1111 }
1112
1113 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
1114 static const AVOption aacenc_options[] = {
1115 {"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "coder"},
1116 {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1117 {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1118 {"fast", "Default fast search", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
1119 {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
1120 {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1121 {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1122 {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
1123 {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1124 {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1125 {"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
1126 FF_AAC_PROFILE_OPTS
1127 {NULL}
1128 };
1129
1130 static const AVClass aacenc_class = {
1131 .class_name = "AAC encoder",
1132 .item_name = av_default_item_name,
1133 .option = aacenc_options,
1134 .version = LIBAVUTIL_VERSION_INT,
1135 };
1136
1137 static const FFCodecDefault aac_encode_defaults[] = {
1138 { "b", "0" },
1139 { NULL }
1140 };
1141
1142 const FFCodec ff_aac_encoder = {
1143 .p.name = "aac",
1144 .p.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
1145 .p.type = AVMEDIA_TYPE_AUDIO,
1146 .p.id = AV_CODEC_ID_AAC,
1147 .priv_data_size = sizeof(AACEncContext),
1148 .init = aac_encode_init,
1149 FF_CODEC_ENCODE_CB(aac_encode_frame),
1150 .close = aac_encode_end,
1151 .defaults = aac_encode_defaults,
1152 .p.supported_samplerates = ff_mpeg4audio_sample_rates,
1153 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
1154 .p.capabilities = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,
1155 .p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
1156 AV_SAMPLE_FMT_NONE },
1157 .p.priv_class = &aacenc_class,
1158 };
1159