61e0f52cc42d454f61953b5ce583a4c5ab470b3b
[libav.git] / libavcodec / aacenc.c
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 libavcodec/aacenc.c
24 * AAC encoder
25 */
26
27 /***********************************
28 * TODOs:
29 * add sane pulse detection
30 * add temporal noise shaping
31 ***********************************/
32
33 #include "avcodec.h"
34 #include "put_bits.h"
35 #include "dsputil.h"
36 #include "mpeg4audio.h"
37
38 #include "aac.h"
39 #include "aactab.h"
40 #include "aacenc.h"
41
42 #include "psymodel.h"
43
44 static const uint8_t swb_size_1024_96[] = {
45 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
46 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
47 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
48 };
49
50 static const uint8_t swb_size_1024_64[] = {
51 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
52 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
53 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
54 };
55
56 static const uint8_t swb_size_1024_48[] = {
57 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
58 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
59 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
60 96
61 };
62
63 static const uint8_t swb_size_1024_32[] = {
64 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
65 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
66 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
67 };
68
69 static const uint8_t swb_size_1024_24[] = {
70 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
71 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
72 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
73 };
74
75 static const uint8_t swb_size_1024_16[] = {
76 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
77 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
78 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
79 };
80
81 static const uint8_t swb_size_1024_8[] = {
82 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
83 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
84 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
85 };
86
87 static const uint8_t *swb_size_1024[] = {
88 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
89 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
90 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
91 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
92 };
93
94 static const uint8_t swb_size_128_96[] = {
95 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
96 };
97
98 static const uint8_t swb_size_128_48[] = {
99 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
100 };
101
102 static const uint8_t swb_size_128_24[] = {
103 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
104 };
105
106 static const uint8_t swb_size_128_16[] = {
107 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
108 };
109
110 static const uint8_t swb_size_128_8[] = {
111 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
112 };
113
114 static const uint8_t *swb_size_128[] = {
115 /* the last entry on the following row is swb_size_128_64 but is a
116 duplicate of swb_size_128_96 */
117 swb_size_128_96, swb_size_128_96, swb_size_128_96,
118 swb_size_128_48, swb_size_128_48, swb_size_128_48,
119 swb_size_128_24, swb_size_128_24, swb_size_128_16,
120 swb_size_128_16, swb_size_128_16, swb_size_128_8
121 };
122
123 /** default channel configurations */
124 static const uint8_t aac_chan_configs[6][5] = {
125 {1, TYPE_SCE}, // 1 channel - single channel element
126 {1, TYPE_CPE}, // 2 channels - channel pair
127 {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
128 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
129 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
130 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
131 };
132
133 /**
134 * Make AAC audio config object.
135 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
136 */
137 static void put_audio_specific_config(AVCodecContext *avctx)
138 {
139 PutBitContext pb;
140 AACEncContext *s = avctx->priv_data;
141
142 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
143 put_bits(&pb, 5, 2); //object type - AAC-LC
144 put_bits(&pb, 4, s->samplerate_index); //sample rate index
145 put_bits(&pb, 4, avctx->channels);
146 //GASpecificConfig
147 put_bits(&pb, 1, 0); //frame length - 1024 samples
148 put_bits(&pb, 1, 0); //does not depend on core coder
149 put_bits(&pb, 1, 0); //is not extension
150 flush_put_bits(&pb);
151 }
152
153 static av_cold int aac_encode_init(AVCodecContext *avctx)
154 {
155 AACEncContext *s = avctx->priv_data;
156 int i;
157 const uint8_t *sizes[2];
158 int lengths[2];
159
160 avctx->frame_size = 1024;
161
162 for (i = 0; i < 16; i++)
163 if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
164 break;
165 if (i == 16) {
166 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
167 return -1;
168 }
169 if (avctx->channels > 6) {
170 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
171 return -1;
172 }
173 s->samplerate_index = i;
174
175 dsputil_init(&s->dsp, avctx);
176 ff_mdct_init(&s->mdct1024, 11, 0, 1.0);
177 ff_mdct_init(&s->mdct128, 8, 0, 1.0);
178 // window init
179 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
180 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
181 ff_sine_window_init(ff_sine_1024, 1024);
182 ff_sine_window_init(ff_sine_128, 128);
183
184 s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
185 s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
186 avctx->extradata = av_malloc(2);
187 avctx->extradata_size = 2;
188 put_audio_specific_config(avctx);
189
190 sizes[0] = swb_size_1024[i];
191 sizes[1] = swb_size_128[i];
192 lengths[0] = ff_aac_num_swb_1024[i];
193 lengths[1] = ff_aac_num_swb_128[i];
194 ff_psy_init(&s->psy, avctx, 2, sizes, lengths);
195 s->psypp = ff_psy_preprocess_init(avctx);
196 s->coder = &ff_aac_coders[0];
197
198 s->lambda = avctx->global_quality ? avctx->global_quality : 120;
199 #if !CONFIG_HARDCODED_TABLES
200 for (i = 0; i < 428; i++)
201 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
202 #endif /* CONFIG_HARDCODED_TABLES */
203
204 if (avctx->channels > 5)
205 av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. "
206 "The output will most likely be an illegal bitstream.\n");
207
208 return 0;
209 }
210
211 static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s,
212 SingleChannelElement *sce, short *audio, int channel)
213 {
214 int i, j, k;
215 const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
216 const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
217 const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
218
219 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
220 memcpy(s->output, sce->saved, sizeof(float)*1024);
221 if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) {
222 memset(s->output, 0, sizeof(s->output[0]) * 448);
223 for (i = 448; i < 576; i++)
224 s->output[i] = sce->saved[i] * pwindow[i - 448];
225 for (i = 576; i < 704; i++)
226 s->output[i] = sce->saved[i];
227 }
228 if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) {
229 j = channel;
230 for (i = 0; i < 1024; i++, j += avctx->channels) {
231 s->output[i+1024] = audio[j] * lwindow[1024 - i - 1];
232 sce->saved[i] = audio[j] * lwindow[i];
233 }
234 } else {
235 j = channel;
236 for (i = 0; i < 448; i++, j += avctx->channels)
237 s->output[i+1024] = audio[j];
238 for (i = 448; i < 576; i++, j += avctx->channels)
239 s->output[i+1024] = audio[j] * swindow[576 - i - 1];
240 memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448);
241 j = channel;
242 for (i = 0; i < 1024; i++, j += avctx->channels)
243 sce->saved[i] = audio[j];
244 }
245 ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output);
246 } else {
247 j = channel;
248 for (k = 0; k < 1024; k += 128) {
249 for (i = 448 + k; i < 448 + k + 256; i++)
250 s->output[i - 448 - k] = (i < 1024)
251 ? sce->saved[i]
252 : audio[channel + (i-1024)*avctx->channels];
253 s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128);
254 s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128);
255 ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output);
256 }
257 j = channel;
258 for (i = 0; i < 1024; i++, j += avctx->channels)
259 sce->saved[i] = audio[j];
260 }
261 }
262
263 /**
264 * Encode ics_info element.
265 * @see Table 4.6 (syntax of ics_info)
266 */
267 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
268 {
269 int w;
270
271 put_bits(&s->pb, 1, 0); // ics_reserved bit
272 put_bits(&s->pb, 2, info->window_sequence[0]);
273 put_bits(&s->pb, 1, info->use_kb_window[0]);
274 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
275 put_bits(&s->pb, 6, info->max_sfb);
276 put_bits(&s->pb, 1, 0); // no prediction
277 } else {
278 put_bits(&s->pb, 4, info->max_sfb);
279 for (w = 1; w < 8; w++)
280 put_bits(&s->pb, 1, !info->group_len[w]);
281 }
282 }
283
284 /**
285 * Encode MS data.
286 * @see 4.6.8.1 "Joint Coding - M/S Stereo"
287 */
288 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
289 {
290 int i, w;
291
292 put_bits(pb, 2, cpe->ms_mode);
293 if (cpe->ms_mode == 1)
294 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
295 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
296 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
297 }
298
299 /**
300 * Produce integer coefficients from scalefactors provided by the model.
301 */
302 static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans)
303 {
304 int i, w, w2, g, ch;
305 int start, sum, maxsfb, cmaxsfb;
306
307 for (ch = 0; ch < chans; ch++) {
308 IndividualChannelStream *ics = &cpe->ch[ch].ics;
309 start = 0;
310 maxsfb = 0;
311 cpe->ch[ch].pulse.num_pulse = 0;
312 for (w = 0; w < ics->num_windows*16; w += 16) {
313 for (g = 0; g < ics->num_swb; g++) {
314 sum = 0;
315 //apply M/S
316 if (!ch && cpe->ms_mask[w + g]) {
317 for (i = 0; i < ics->swb_sizes[g]; i++) {
318 cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
319 cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
320 }
321 }
322 start += ics->swb_sizes[g];
323 }
324 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
325 ;
326 maxsfb = FFMAX(maxsfb, cmaxsfb);
327 }
328 ics->max_sfb = maxsfb;
329
330 //adjust zero bands for window groups
331 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
332 for (g = 0; g < ics->max_sfb; g++) {
333 i = 1;
334 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
335 if (!cpe->ch[ch].zeroes[w2*16 + g]) {
336 i = 0;
337 break;
338 }
339 }
340 cpe->ch[ch].zeroes[w*16 + g] = i;
341 }
342 }
343 }
344
345 if (chans > 1 && cpe->common_window) {
346 IndividualChannelStream *ics0 = &cpe->ch[0].ics;
347 IndividualChannelStream *ics1 = &cpe->ch[1].ics;
348 int msc = 0;
349 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
350 ics1->max_sfb = ics0->max_sfb;
351 for (w = 0; w < ics0->num_windows*16; w += 16)
352 for (i = 0; i < ics0->max_sfb; i++)
353 if (cpe->ms_mask[w+i])
354 msc++;
355 if (msc == 0 || ics0->max_sfb == 0)
356 cpe->ms_mode = 0;
357 else
358 cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2;
359 }
360 }
361
362 /**
363 * Encode scalefactor band coding type.
364 */
365 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
366 {
367 int w;
368
369 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
370 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
371 }
372
373 /**
374 * Encode scalefactors.
375 */
376 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
377 SingleChannelElement *sce)
378 {
379 int off = sce->sf_idx[0], diff;
380 int i, w;
381
382 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
383 for (i = 0; i < sce->ics.max_sfb; i++) {
384 if (!sce->zeroes[w*16 + i]) {
385 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
386 if (diff < 0 || diff > 120)
387 av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n");
388 off = sce->sf_idx[w*16 + i];
389 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
390 }
391 }
392 }
393 }
394
395 /**
396 * Encode pulse data.
397 */
398 static void encode_pulses(AACEncContext *s, Pulse *pulse)
399 {
400 int i;
401
402 put_bits(&s->pb, 1, !!pulse->num_pulse);
403 if (!pulse->num_pulse)
404 return;
405
406 put_bits(&s->pb, 2, pulse->num_pulse - 1);
407 put_bits(&s->pb, 6, pulse->start);
408 for (i = 0; i < pulse->num_pulse; i++) {
409 put_bits(&s->pb, 5, pulse->pos[i]);
410 put_bits(&s->pb, 4, pulse->amp[i]);
411 }
412 }
413
414 /**
415 * Encode spectral coefficients processed by psychoacoustic model.
416 */
417 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
418 {
419 int start, i, w, w2;
420
421 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
422 start = 0;
423 for (i = 0; i < sce->ics.max_sfb; i++) {
424 if (sce->zeroes[w*16 + i]) {
425 start += sce->ics.swb_sizes[i];
426 continue;
427 }
428 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
429 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
430 sce->ics.swb_sizes[i],
431 sce->sf_idx[w*16 + i],
432 sce->band_type[w*16 + i],
433 s->lambda);
434 start += sce->ics.swb_sizes[i];
435 }
436 }
437 }
438
439 /**
440 * Encode one channel of audio data.
441 */
442 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
443 SingleChannelElement *sce,
444 int common_window)
445 {
446 put_bits(&s->pb, 8, sce->sf_idx[0]);
447 if (!common_window)
448 put_ics_info(s, &sce->ics);
449 encode_band_info(s, sce);
450 encode_scale_factors(avctx, s, sce);
451 encode_pulses(s, &sce->pulse);
452 put_bits(&s->pb, 1, 0); //tns
453 put_bits(&s->pb, 1, 0); //ssr
454 encode_spectral_coeffs(s, sce);
455 return 0;
456 }
457
458 /**
459 * Write some auxiliary information about the created AAC file.
460 */
461 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s,
462 const char *name)
463 {
464 int i, namelen, padbits;
465
466 namelen = strlen(name) + 2;
467 put_bits(&s->pb, 3, TYPE_FIL);
468 put_bits(&s->pb, 4, FFMIN(namelen, 15));
469 if (namelen >= 15)
470 put_bits(&s->pb, 8, namelen - 16);
471 put_bits(&s->pb, 4, 0); //extension type - filler
472 padbits = 8 - (put_bits_count(&s->pb) & 7);
473 align_put_bits(&s->pb);
474 for (i = 0; i < namelen - 2; i++)
475 put_bits(&s->pb, 8, name[i]);
476 put_bits(&s->pb, 12 - padbits, 0);
477 }
478
479 static int aac_encode_frame(AVCodecContext *avctx,
480 uint8_t *frame, int buf_size, void *data)
481 {
482 AACEncContext *s = avctx->priv_data;
483 int16_t *samples = s->samples, *samples2, *la;
484 ChannelElement *cpe;
485 int i, j, chans, tag, start_ch;
486 const uint8_t *chan_map = aac_chan_configs[avctx->channels-1];
487 int chan_el_counter[4];
488
489 if (s->last_frame)
490 return 0;
491 if (data) {
492 if (!s->psypp) {
493 memcpy(s->samples + 1024 * avctx->channels, data,
494 1024 * avctx->channels * sizeof(s->samples[0]));
495 } else {
496 start_ch = 0;
497 samples2 = s->samples + 1024 * avctx->channels;
498 for (i = 0; i < chan_map[0]; i++) {
499 tag = chan_map[i+1];
500 chans = tag == TYPE_CPE ? 2 : 1;
501 ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch,
502 samples2 + start_ch, start_ch, chans);
503 start_ch += chans;
504 }
505 }
506 }
507 if (!avctx->frame_number) {
508 memcpy(s->samples, s->samples + 1024 * avctx->channels,
509 1024 * avctx->channels * sizeof(s->samples[0]));
510 return 0;
511 }
512
513 init_put_bits(&s->pb, frame, buf_size*8);
514 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
515 put_bitstream_info(avctx, s, LIBAVCODEC_IDENT);
516 start_ch = 0;
517 memset(chan_el_counter, 0, sizeof(chan_el_counter));
518 for (i = 0; i < chan_map[0]; i++) {
519 FFPsyWindowInfo wi[2];
520 tag = chan_map[i+1];
521 chans = tag == TYPE_CPE ? 2 : 1;
522 cpe = &s->cpe[i];
523 samples2 = samples + start_ch;
524 la = samples2 + 1024 * avctx->channels + start_ch;
525 if (!data)
526 la = NULL;
527 for (j = 0; j < chans; j++) {
528 IndividualChannelStream *ics = &cpe->ch[j].ics;
529 int k;
530 wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]);
531 ics->window_sequence[1] = ics->window_sequence[0];
532 ics->window_sequence[0] = wi[j].window_type[0];
533 ics->use_kb_window[1] = ics->use_kb_window[0];
534 ics->use_kb_window[0] = wi[j].window_shape;
535 ics->num_windows = wi[j].num_windows;
536 ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
537 ics->num_swb = s->psy.num_bands[ics->num_windows == 8];
538 for (k = 0; k < ics->num_windows; k++)
539 ics->group_len[k] = wi[j].grouping[k];
540
541 s->cur_channel = start_ch + j;
542 apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j);
543 s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda);
544 }
545 cpe->common_window = 0;
546 if (chans > 1
547 && wi[0].window_type[0] == wi[1].window_type[0]
548 && wi[0].window_shape == wi[1].window_shape) {
549
550 cpe->common_window = 1;
551 for (j = 0; j < wi[0].num_windows; j++) {
552 if (wi[0].grouping[j] != wi[1].grouping[j]) {
553 cpe->common_window = 0;
554 break;
555 }
556 }
557 }
558 if (cpe->common_window && s->coder->search_for_ms)
559 s->coder->search_for_ms(s, cpe, s->lambda);
560 adjust_frame_information(s, cpe, chans);
561 put_bits(&s->pb, 3, tag);
562 put_bits(&s->pb, 4, chan_el_counter[tag]++);
563 if (chans == 2) {
564 put_bits(&s->pb, 1, cpe->common_window);
565 if (cpe->common_window) {
566 put_ics_info(s, &cpe->ch[0].ics);
567 encode_ms_info(&s->pb, cpe);
568 }
569 }
570 for (j = 0; j < chans; j++) {
571 s->cur_channel = start_ch + j;
572 ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]);
573 encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window);
574 }
575 start_ch += chans;
576 }
577
578 put_bits(&s->pb, 3, TYPE_END);
579 flush_put_bits(&s->pb);
580 avctx->frame_bits = put_bits_count(&s->pb);
581
582 // rate control stuff
583 if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
584 float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
585 s->lambda *= ratio;
586 s->lambda = fminf(s->lambda, 65536.f);
587 }
588
589 if (avctx->frame_bits > 6144*avctx->channels)
590 av_log(avctx, AV_LOG_ERROR, "input buffer violation %d > %d.\n",
591 avctx->frame_bits, 6144*avctx->channels);
592
593 if (!data)
594 s->last_frame = 1;
595 memcpy(s->samples, s->samples + 1024 * avctx->channels,
596 1024 * avctx->channels * sizeof(s->samples[0]));
597 return put_bits_count(&s->pb)>>3;
598 }
599
600 static av_cold int aac_encode_end(AVCodecContext *avctx)
601 {
602 AACEncContext *s = avctx->priv_data;
603
604 ff_mdct_end(&s->mdct1024);
605 ff_mdct_end(&s->mdct128);
606 ff_psy_end(&s->psy);
607 ff_psy_preprocess_end(s->psypp);
608 av_freep(&s->samples);
609 av_freep(&s->cpe);
610 return 0;
611 }
612
613 AVCodec aac_encoder = {
614 "aac",
615 CODEC_TYPE_AUDIO,
616 CODEC_ID_AAC,
617 sizeof(AACEncContext),
618 aac_encode_init,
619 aac_encode_frame,
620 aac_encode_end,
621 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
622 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
623 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
624 };