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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 | * add temporal noise shaping | |
31 | ***********************************/ | |
32 | ||
33 | #include "libavutil/float_dsp.h" | |
34 | #include "libavutil/opt.h" | |
35 | #include "avcodec.h" | |
36 | #include "put_bits.h" | |
37 | #include "internal.h" | |
38 | #include "mpeg4audio.h" | |
39 | #include "kbdwin.h" | |
40 | #include "sinewin.h" | |
41 | ||
42 | #include "aac.h" | |
43 | #include "aactab.h" | |
44 | #include "aacenc.h" | |
45 | ||
46 | #include "psymodel.h" | |
47 | ||
48 | #define AAC_MAX_CHANNELS 6 | |
49 | ||
50 | #define ERROR_IF(cond, ...) \ | |
51 | if (cond) { \ | |
52 | av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \ | |
53 | return AVERROR(EINVAL); \ | |
54 | } | |
55 | ||
56 | float ff_aac_pow34sf_tab[428]; | |
57 | ||
58 | static const uint8_t swb_size_1024_96[] = { | |
59 | 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, | |
60 | 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, | |
61 | 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64 | |
62 | }; | |
63 | ||
64 | static const uint8_t swb_size_1024_64[] = { | |
65 | 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, | |
66 | 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36, | |
67 | 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40 | |
68 | }; | |
69 | ||
70 | static const uint8_t swb_size_1024_48[] = { | |
71 | 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, | |
72 | 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, | |
73 | 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, | |
74 | 96 | |
75 | }; | |
76 | ||
77 | static const uint8_t swb_size_1024_32[] = { | |
78 | 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, | |
79 | 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, | |
80 | 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32 | |
81 | }; | |
82 | ||
83 | static const uint8_t swb_size_1024_24[] = { | |
84 | 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, | |
85 | 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, | |
86 | 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64 | |
87 | }; | |
88 | ||
89 | static const uint8_t swb_size_1024_16[] = { | |
90 | 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, | |
91 | 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, | |
92 | 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64 | |
93 | }; | |
94 | ||
95 | static const uint8_t swb_size_1024_8[] = { | |
96 | 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, | |
97 | 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28, | |
98 | 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80 | |
99 | }; | |
100 | ||
101 | static const uint8_t *swb_size_1024[] = { | |
102 | swb_size_1024_96, swb_size_1024_96, swb_size_1024_64, | |
103 | swb_size_1024_48, swb_size_1024_48, swb_size_1024_32, | |
104 | swb_size_1024_24, swb_size_1024_24, swb_size_1024_16, | |
105 | swb_size_1024_16, swb_size_1024_16, swb_size_1024_8 | |
106 | }; | |
107 | ||
108 | static const uint8_t swb_size_128_96[] = { | |
109 | 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36 | |
110 | }; | |
111 | ||
112 | static const uint8_t swb_size_128_48[] = { | |
113 | 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16 | |
114 | }; | |
115 | ||
116 | static const uint8_t swb_size_128_24[] = { | |
117 | 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20 | |
118 | }; | |
119 | ||
120 | static const uint8_t swb_size_128_16[] = { | |
121 | 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20 | |
122 | }; | |
123 | ||
124 | static const uint8_t swb_size_128_8[] = { | |
125 | 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20 | |
126 | }; | |
127 | ||
128 | static const uint8_t *swb_size_128[] = { | |
129 | /* the last entry on the following row is swb_size_128_64 but is a | |
130 | duplicate of swb_size_128_96 */ | |
131 | swb_size_128_96, swb_size_128_96, swb_size_128_96, | |
132 | swb_size_128_48, swb_size_128_48, swb_size_128_48, | |
133 | swb_size_128_24, swb_size_128_24, swb_size_128_16, | |
134 | swb_size_128_16, swb_size_128_16, swb_size_128_8 | |
135 | }; | |
136 | ||
137 | /** default channel configurations */ | |
138 | static const uint8_t aac_chan_configs[6][5] = { | |
139 | {1, TYPE_SCE}, // 1 channel - single channel element | |
140 | {1, TYPE_CPE}, // 2 channels - channel pair | |
141 | {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo | |
142 | {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center | |
143 | {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo | |
144 | {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE | |
145 | }; | |
146 | ||
147 | /** | |
148 | * Table to remap channels from libavcodec's default order to AAC order. | |
149 | */ | |
150 | static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = { | |
151 | { 0 }, | |
152 | { 0, 1 }, | |
153 | { 2, 0, 1 }, | |
154 | { 2, 0, 1, 3 }, | |
155 | { 2, 0, 1, 3, 4 }, | |
156 | { 2, 0, 1, 4, 5, 3 }, | |
157 | }; | |
158 | ||
159 | /** | |
160 | * Make AAC audio config object. | |
161 | * @see 1.6.2.1 "Syntax - AudioSpecificConfig" | |
162 | */ | |
163 | static void put_audio_specific_config(AVCodecContext *avctx) | |
164 | { | |
165 | PutBitContext pb; | |
166 | AACEncContext *s = avctx->priv_data; | |
167 | ||
168 | init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8); | |
169 | put_bits(&pb, 5, 2); //object type - AAC-LC | |
170 | put_bits(&pb, 4, s->samplerate_index); //sample rate index | |
171 | put_bits(&pb, 4, s->channels); | |
172 | //GASpecificConfig | |
173 | put_bits(&pb, 1, 0); //frame length - 1024 samples | |
174 | put_bits(&pb, 1, 0); //does not depend on core coder | |
175 | put_bits(&pb, 1, 0); //is not extension | |
176 | ||
177 | //Explicitly Mark SBR absent | |
178 | put_bits(&pb, 11, 0x2b7); //sync extension | |
179 | put_bits(&pb, 5, AOT_SBR); | |
180 | put_bits(&pb, 1, 0); | |
181 | flush_put_bits(&pb); | |
182 | } | |
183 | ||
184 | #define WINDOW_FUNC(type) \ | |
185 | static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \ | |
186 | SingleChannelElement *sce, \ | |
187 | const float *audio) | |
188 | ||
189 | WINDOW_FUNC(only_long) | |
190 | { | |
191 | const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
192 | const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
193 | float *out = sce->ret_buf; | |
194 | ||
195 | fdsp->vector_fmul (out, audio, lwindow, 1024); | |
196 | fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024); | |
197 | } | |
198 | ||
199 | WINDOW_FUNC(long_start) | |
200 | { | |
201 | const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
202 | const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; | |
203 | float *out = sce->ret_buf; | |
204 | ||
205 | fdsp->vector_fmul(out, audio, lwindow, 1024); | |
206 | memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448); | |
207 | fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128); | |
208 | memset(out + 1024 + 576, 0, sizeof(out[0]) * 448); | |
209 | } | |
210 | ||
211 | WINDOW_FUNC(long_stop) | |
212 | { | |
213 | const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
214 | const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
215 | float *out = sce->ret_buf; | |
216 | ||
217 | memset(out, 0, sizeof(out[0]) * 448); | |
218 | fdsp->vector_fmul(out + 448, audio + 448, swindow, 128); | |
219 | memcpy(out + 576, audio + 576, sizeof(out[0]) * 448); | |
220 | fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024); | |
221 | } | |
222 | ||
223 | WINDOW_FUNC(eight_short) | |
224 | { | |
225 | const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; | |
226 | const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
227 | const float *in = audio + 448; | |
228 | float *out = sce->ret_buf; | |
229 | int w; | |
230 | ||
231 | for (w = 0; w < 8; w++) { | |
232 | fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128); | |
233 | out += 128; | |
234 | in += 128; | |
235 | fdsp->vector_fmul_reverse(out, in, swindow, 128); | |
236 | out += 128; | |
237 | } | |
238 | } | |
239 | ||
240 | static void (*const apply_window[4])(AVFloatDSPContext *fdsp, | |
241 | SingleChannelElement *sce, | |
242 | const float *audio) = { | |
243 | [ONLY_LONG_SEQUENCE] = apply_only_long_window, | |
244 | [LONG_START_SEQUENCE] = apply_long_start_window, | |
245 | [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window, | |
246 | [LONG_STOP_SEQUENCE] = apply_long_stop_window | |
247 | }; | |
248 | ||
249 | static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce, | |
250 | float *audio) | |
251 | { | |
252 | int i; | |
253 | float *output = sce->ret_buf; | |
254 | ||
f6fa7814 | 255 | apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio); |
2ba45a60 DM |
256 | |
257 | if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) | |
258 | s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output); | |
259 | else | |
260 | for (i = 0; i < 1024; i += 128) | |
261 | s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2); | |
262 | memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024); | |
263 | } | |
264 | ||
265 | /** | |
266 | * Encode ics_info element. | |
267 | * @see Table 4.6 (syntax of ics_info) | |
268 | */ | |
269 | static void put_ics_info(AACEncContext *s, IndividualChannelStream *info) | |
270 | { | |
271 | int w; | |
272 | ||
273 | put_bits(&s->pb, 1, 0); // ics_reserved bit | |
274 | put_bits(&s->pb, 2, info->window_sequence[0]); | |
275 | put_bits(&s->pb, 1, info->use_kb_window[0]); | |
276 | if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) { | |
277 | put_bits(&s->pb, 6, info->max_sfb); | |
278 | put_bits(&s->pb, 1, 0); // no prediction | |
279 | } else { | |
280 | put_bits(&s->pb, 4, info->max_sfb); | |
281 | for (w = 1; w < 8; w++) | |
282 | put_bits(&s->pb, 1, !info->group_len[w]); | |
283 | } | |
284 | } | |
285 | ||
286 | /** | |
287 | * Encode MS data. | |
288 | * @see 4.6.8.1 "Joint Coding - M/S Stereo" | |
289 | */ | |
290 | static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe) | |
291 | { | |
292 | int i, w; | |
293 | ||
294 | put_bits(pb, 2, cpe->ms_mode); | |
295 | if (cpe->ms_mode == 1) | |
296 | for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w]) | |
297 | for (i = 0; i < cpe->ch[0].ics.max_sfb; i++) | |
298 | put_bits(pb, 1, cpe->ms_mask[w*16 + i]); | |
299 | } | |
300 | ||
301 | /** | |
302 | * Produce integer coefficients from scalefactors provided by the model. | |
303 | */ | |
304 | static void adjust_frame_information(ChannelElement *cpe, int chans) | |
305 | { | |
306 | int i, w, w2, g, ch; | |
307 | int start, maxsfb, cmaxsfb; | |
308 | ||
309 | for (ch = 0; ch < chans; ch++) { | |
310 | IndividualChannelStream *ics = &cpe->ch[ch].ics; | |
311 | start = 0; | |
312 | maxsfb = 0; | |
313 | cpe->ch[ch].pulse.num_pulse = 0; | |
314 | for (w = 0; w < ics->num_windows*16; w += 16) { | |
315 | for (g = 0; g < ics->num_swb; g++) { | |
316 | //apply M/S | |
317 | if (cpe->common_window && !ch && cpe->ms_mask[w + g]) { | |
318 | for (i = 0; i < ics->swb_sizes[g]; i++) { | |
319 | cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0; | |
320 | cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i]; | |
321 | } | |
322 | } | |
323 | start += ics->swb_sizes[g]; | |
324 | } | |
325 | for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--) | |
326 | ; | |
327 | maxsfb = FFMAX(maxsfb, cmaxsfb); | |
328 | } | |
329 | ics->max_sfb = maxsfb; | |
330 | ||
331 | //adjust zero bands for window groups | |
332 | for (w = 0; w < ics->num_windows; w += ics->group_len[w]) { | |
333 | for (g = 0; g < ics->max_sfb; g++) { | |
334 | i = 1; | |
335 | for (w2 = w; w2 < w + ics->group_len[w]; w2++) { | |
336 | if (!cpe->ch[ch].zeroes[w2*16 + g]) { | |
337 | i = 0; | |
338 | break; | |
339 | } | |
340 | } | |
341 | cpe->ch[ch].zeroes[w*16 + g] = i; | |
342 | } | |
343 | } | |
344 | } | |
345 | ||
346 | if (chans > 1 && cpe->common_window) { | |
347 | IndividualChannelStream *ics0 = &cpe->ch[0].ics; | |
348 | IndividualChannelStream *ics1 = &cpe->ch[1].ics; | |
349 | int msc = 0; | |
350 | ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb); | |
351 | ics1->max_sfb = ics0->max_sfb; | |
352 | for (w = 0; w < ics0->num_windows*16; w += 16) | |
353 | for (i = 0; i < ics0->max_sfb; i++) | |
354 | if (cpe->ms_mask[w+i]) | |
355 | msc++; | |
356 | if (msc == 0 || ics0->max_sfb == 0) | |
357 | cpe->ms_mode = 0; | |
358 | else | |
359 | cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2; | |
360 | } | |
361 | } | |
362 | ||
363 | /** | |
364 | * Encode scalefactor band coding type. | |
365 | */ | |
366 | static void encode_band_info(AACEncContext *s, SingleChannelElement *sce) | |
367 | { | |
368 | int w; | |
369 | ||
370 | for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) | |
371 | s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda); | |
372 | } | |
373 | ||
374 | /** | |
375 | * Encode scalefactors. | |
376 | */ | |
377 | static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, | |
378 | SingleChannelElement *sce) | |
379 | { | |
380 | int off = sce->sf_idx[0], diff; | |
381 | int i, w; | |
382 | ||
383 | for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { | |
384 | for (i = 0; i < sce->ics.max_sfb; i++) { | |
385 | if (!sce->zeroes[w*16 + i]) { | |
386 | diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO; | |
387 | av_assert0(diff >= 0 && diff <= 120); | |
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(AACEncContext *s, const char *name) | |
462 | { | |
463 | int i, namelen, padbits; | |
464 | ||
465 | namelen = strlen(name) + 2; | |
466 | put_bits(&s->pb, 3, TYPE_FIL); | |
467 | put_bits(&s->pb, 4, FFMIN(namelen, 15)); | |
468 | if (namelen >= 15) | |
469 | put_bits(&s->pb, 8, namelen - 14); | |
470 | put_bits(&s->pb, 4, 0); //extension type - filler | |
471 | padbits = -put_bits_count(&s->pb) & 7; | |
472 | avpriv_align_put_bits(&s->pb); | |
473 | for (i = 0; i < namelen - 2; i++) | |
474 | put_bits(&s->pb, 8, name[i]); | |
475 | put_bits(&s->pb, 12 - padbits, 0); | |
476 | } | |
477 | ||
478 | /* | |
479 | * Copy input samples. | |
480 | * Channels are reordered from libavcodec's default order to AAC order. | |
481 | */ | |
482 | static void copy_input_samples(AACEncContext *s, const AVFrame *frame) | |
483 | { | |
484 | int ch; | |
485 | int end = 2048 + (frame ? frame->nb_samples : 0); | |
486 | const uint8_t *channel_map = aac_chan_maps[s->channels - 1]; | |
487 | ||
488 | /* copy and remap input samples */ | |
489 | for (ch = 0; ch < s->channels; ch++) { | |
490 | /* copy last 1024 samples of previous frame to the start of the current frame */ | |
491 | memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0])); | |
492 | ||
493 | /* copy new samples and zero any remaining samples */ | |
494 | if (frame) { | |
495 | memcpy(&s->planar_samples[ch][2048], | |
496 | frame->extended_data[channel_map[ch]], | |
497 | frame->nb_samples * sizeof(s->planar_samples[0][0])); | |
498 | } | |
499 | memset(&s->planar_samples[ch][end], 0, | |
500 | (3072 - end) * sizeof(s->planar_samples[0][0])); | |
501 | } | |
502 | } | |
503 | ||
504 | static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt, | |
505 | const AVFrame *frame, int *got_packet_ptr) | |
506 | { | |
507 | AACEncContext *s = avctx->priv_data; | |
508 | float **samples = s->planar_samples, *samples2, *la, *overlap; | |
509 | ChannelElement *cpe; | |
510 | int i, ch, w, g, chans, tag, start_ch, ret; | |
511 | int chan_el_counter[4]; | |
512 | FFPsyWindowInfo windows[AAC_MAX_CHANNELS]; | |
513 | ||
514 | if (s->last_frame == 2) | |
515 | return 0; | |
516 | ||
517 | /* add current frame to queue */ | |
518 | if (frame) { | |
519 | if ((ret = ff_af_queue_add(&s->afq, frame)) < 0) | |
520 | return ret; | |
521 | } | |
522 | ||
523 | copy_input_samples(s, frame); | |
524 | if (s->psypp) | |
525 | ff_psy_preprocess(s->psypp, s->planar_samples, s->channels); | |
526 | ||
527 | if (!avctx->frame_number) | |
528 | return 0; | |
529 | ||
530 | start_ch = 0; | |
531 | for (i = 0; i < s->chan_map[0]; i++) { | |
532 | FFPsyWindowInfo* wi = windows + start_ch; | |
533 | tag = s->chan_map[i+1]; | |
534 | chans = tag == TYPE_CPE ? 2 : 1; | |
535 | cpe = &s->cpe[i]; | |
536 | for (ch = 0; ch < chans; ch++) { | |
537 | IndividualChannelStream *ics = &cpe->ch[ch].ics; | |
538 | int cur_channel = start_ch + ch; | |
539 | overlap = &samples[cur_channel][0]; | |
540 | samples2 = overlap + 1024; | |
541 | la = samples2 + (448+64); | |
542 | if (!frame) | |
543 | la = NULL; | |
544 | if (tag == TYPE_LFE) { | |
545 | wi[ch].window_type[0] = ONLY_LONG_SEQUENCE; | |
546 | wi[ch].window_shape = 0; | |
547 | wi[ch].num_windows = 1; | |
548 | wi[ch].grouping[0] = 1; | |
549 | ||
550 | /* Only the lowest 12 coefficients are used in a LFE channel. | |
551 | * The expression below results in only the bottom 8 coefficients | |
552 | * being used for 11.025kHz to 16kHz sample rates. | |
553 | */ | |
554 | ics->num_swb = s->samplerate_index >= 8 ? 1 : 3; | |
555 | } else { | |
556 | wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel, | |
557 | ics->window_sequence[0]); | |
558 | } | |
559 | ics->window_sequence[1] = ics->window_sequence[0]; | |
560 | ics->window_sequence[0] = wi[ch].window_type[0]; | |
561 | ics->use_kb_window[1] = ics->use_kb_window[0]; | |
562 | ics->use_kb_window[0] = wi[ch].window_shape; | |
563 | ics->num_windows = wi[ch].num_windows; | |
564 | ics->swb_sizes = s->psy.bands [ics->num_windows == 8]; | |
565 | ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8]; | |
566 | for (w = 0; w < ics->num_windows; w++) | |
567 | ics->group_len[w] = wi[ch].grouping[w]; | |
568 | ||
569 | apply_window_and_mdct(s, &cpe->ch[ch], overlap); | |
f6fa7814 DM |
570 | if (isnan(cpe->ch->coeffs[0])) { |
571 | av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n"); | |
572 | return AVERROR(EINVAL); | |
573 | } | |
2ba45a60 DM |
574 | } |
575 | start_ch += chans; | |
576 | } | |
577 | if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0) | |
578 | return ret; | |
579 | do { | |
580 | int frame_bits; | |
581 | ||
582 | init_put_bits(&s->pb, avpkt->data, avpkt->size); | |
583 | ||
584 | if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)) | |
585 | put_bitstream_info(s, LIBAVCODEC_IDENT); | |
586 | start_ch = 0; | |
587 | memset(chan_el_counter, 0, sizeof(chan_el_counter)); | |
588 | for (i = 0; i < s->chan_map[0]; i++) { | |
589 | FFPsyWindowInfo* wi = windows + start_ch; | |
590 | const float *coeffs[2]; | |
591 | tag = s->chan_map[i+1]; | |
592 | chans = tag == TYPE_CPE ? 2 : 1; | |
593 | cpe = &s->cpe[i]; | |
594 | put_bits(&s->pb, 3, tag); | |
595 | put_bits(&s->pb, 4, chan_el_counter[tag]++); | |
596 | for (ch = 0; ch < chans; ch++) | |
597 | coeffs[ch] = cpe->ch[ch].coeffs; | |
598 | s->psy.model->analyze(&s->psy, start_ch, coeffs, wi); | |
599 | for (ch = 0; ch < chans; ch++) { | |
600 | s->cur_channel = start_ch + ch; | |
601 | s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda); | |
602 | } | |
603 | cpe->common_window = 0; | |
604 | if (chans > 1 | |
605 | && wi[0].window_type[0] == wi[1].window_type[0] | |
606 | && wi[0].window_shape == wi[1].window_shape) { | |
607 | ||
608 | cpe->common_window = 1; | |
609 | for (w = 0; w < wi[0].num_windows; w++) { | |
610 | if (wi[0].grouping[w] != wi[1].grouping[w]) { | |
611 | cpe->common_window = 0; | |
612 | break; | |
613 | } | |
614 | } | |
615 | } | |
616 | s->cur_channel = start_ch; | |
617 | if (s->options.stereo_mode && cpe->common_window) { | |
618 | if (s->options.stereo_mode > 0) { | |
619 | IndividualChannelStream *ics = &cpe->ch[0].ics; | |
620 | for (w = 0; w < ics->num_windows; w += ics->group_len[w]) | |
621 | for (g = 0; g < ics->num_swb; g++) | |
622 | cpe->ms_mask[w*16+g] = 1; | |
623 | } else if (s->coder->search_for_ms) { | |
624 | s->coder->search_for_ms(s, cpe, s->lambda); | |
625 | } | |
626 | } | |
627 | adjust_frame_information(cpe, chans); | |
628 | if (chans == 2) { | |
629 | put_bits(&s->pb, 1, cpe->common_window); | |
630 | if (cpe->common_window) { | |
631 | put_ics_info(s, &cpe->ch[0].ics); | |
632 | encode_ms_info(&s->pb, cpe); | |
633 | } | |
634 | } | |
635 | for (ch = 0; ch < chans; ch++) { | |
636 | s->cur_channel = start_ch + ch; | |
637 | encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window); | |
638 | } | |
639 | start_ch += chans; | |
640 | } | |
641 | ||
642 | frame_bits = put_bits_count(&s->pb); | |
643 | if (frame_bits <= 6144 * s->channels - 3) { | |
644 | s->psy.bitres.bits = frame_bits / s->channels; | |
645 | break; | |
646 | } | |
647 | ||
648 | s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits; | |
649 | ||
650 | } while (1); | |
651 | ||
652 | put_bits(&s->pb, 3, TYPE_END); | |
653 | flush_put_bits(&s->pb); | |
654 | avctx->frame_bits = put_bits_count(&s->pb); | |
655 | ||
656 | // rate control stuff | |
657 | if (!(avctx->flags & CODEC_FLAG_QSCALE)) { | |
658 | float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits; | |
659 | s->lambda *= ratio; | |
660 | s->lambda = FFMIN(s->lambda, 65536.f); | |
661 | } | |
662 | ||
663 | if (!frame) | |
664 | s->last_frame++; | |
665 | ||
666 | ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, | |
667 | &avpkt->duration); | |
668 | ||
669 | avpkt->size = put_bits_count(&s->pb) >> 3; | |
670 | *got_packet_ptr = 1; | |
671 | return 0; | |
672 | } | |
673 | ||
674 | static av_cold int aac_encode_end(AVCodecContext *avctx) | |
675 | { | |
676 | AACEncContext *s = avctx->priv_data; | |
677 | ||
678 | ff_mdct_end(&s->mdct1024); | |
679 | ff_mdct_end(&s->mdct128); | |
680 | ff_psy_end(&s->psy); | |
681 | if (s->psypp) | |
682 | ff_psy_preprocess_end(s->psypp); | |
683 | av_freep(&s->buffer.samples); | |
684 | av_freep(&s->cpe); | |
f6fa7814 | 685 | av_freep(&s->fdsp); |
2ba45a60 DM |
686 | ff_af_queue_close(&s->afq); |
687 | return 0; | |
688 | } | |
689 | ||
690 | static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s) | |
691 | { | |
692 | int ret = 0; | |
693 | ||
f6fa7814 DM |
694 | s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT); |
695 | if (!s->fdsp) | |
696 | return AVERROR(ENOMEM); | |
2ba45a60 DM |
697 | |
698 | // window init | |
699 | ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
700 | ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
701 | ff_init_ff_sine_windows(10); | |
702 | ff_init_ff_sine_windows(7); | |
703 | ||
704 | if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) | |
705 | return ret; | |
706 | if (ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) | |
707 | return ret; | |
708 | ||
709 | return 0; | |
710 | } | |
711 | ||
712 | static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s) | |
713 | { | |
714 | int ch; | |
715 | FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail); | |
716 | FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail); | |
717 | FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail); | |
718 | ||
719 | for(ch = 0; ch < s->channels; ch++) | |
720 | s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch; | |
721 | ||
722 | return 0; | |
723 | alloc_fail: | |
724 | return AVERROR(ENOMEM); | |
725 | } | |
726 | ||
727 | static av_cold int aac_encode_init(AVCodecContext *avctx) | |
728 | { | |
729 | AACEncContext *s = avctx->priv_data; | |
730 | int i, ret = 0; | |
731 | const uint8_t *sizes[2]; | |
732 | uint8_t grouping[AAC_MAX_CHANNELS]; | |
733 | int lengths[2]; | |
734 | ||
735 | avctx->frame_size = 1024; | |
736 | ||
737 | for (i = 0; i < 16; i++) | |
738 | if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i]) | |
739 | break; | |
740 | ||
741 | s->channels = avctx->channels; | |
742 | ||
743 | ERROR_IF(i == 16, | |
744 | "Unsupported sample rate %d\n", avctx->sample_rate); | |
745 | ERROR_IF(s->channels > AAC_MAX_CHANNELS, | |
746 | "Unsupported number of channels: %d\n", s->channels); | |
747 | ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW, | |
748 | "Unsupported profile %d\n", avctx->profile); | |
749 | ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels, | |
750 | "Too many bits per frame requested\n"); | |
751 | ||
752 | s->samplerate_index = i; | |
753 | ||
754 | s->chan_map = aac_chan_configs[s->channels-1]; | |
755 | ||
756 | if (ret = dsp_init(avctx, s)) | |
757 | goto fail; | |
758 | ||
759 | if (ret = alloc_buffers(avctx, s)) | |
760 | goto fail; | |
761 | ||
762 | avctx->extradata_size = 5; | |
763 | put_audio_specific_config(avctx); | |
764 | ||
765 | sizes[0] = swb_size_1024[i]; | |
766 | sizes[1] = swb_size_128[i]; | |
767 | lengths[0] = ff_aac_num_swb_1024[i]; | |
768 | lengths[1] = ff_aac_num_swb_128[i]; | |
769 | for (i = 0; i < s->chan_map[0]; i++) | |
770 | grouping[i] = s->chan_map[i + 1] == TYPE_CPE; | |
771 | if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping)) | |
772 | goto fail; | |
773 | s->psypp = ff_psy_preprocess_init(avctx); | |
774 | s->coder = &ff_aac_coders[s->options.aac_coder]; | |
775 | ||
776 | if (HAVE_MIPSDSPR1) | |
777 | ff_aac_coder_init_mips(s); | |
778 | ||
779 | s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120; | |
780 | ||
781 | ff_aac_tableinit(); | |
782 | ||
783 | for (i = 0; i < 428; i++) | |
784 | ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i])); | |
785 | ||
f6fa7814 | 786 | avctx->initial_padding = 1024; |
2ba45a60 DM |
787 | ff_af_queue_init(avctx, &s->afq); |
788 | ||
789 | return 0; | |
790 | fail: | |
791 | aac_encode_end(avctx); | |
792 | return ret; | |
793 | } | |
794 | ||
795 | #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM | |
796 | static const AVOption aacenc_options[] = { | |
797 | {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"}, | |
798 | {"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, | |
799 | {"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, | |
800 | {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"}, | |
801 | {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"}, | |
802 | {"faac", "FAAC-inspired method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, | |
803 | {"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, | |
804 | {"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, | |
805 | {"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"}, | |
806 | {NULL} | |
807 | }; | |
808 | ||
809 | static const AVClass aacenc_class = { | |
810 | "AAC encoder", | |
811 | av_default_item_name, | |
812 | aacenc_options, | |
813 | LIBAVUTIL_VERSION_INT, | |
814 | }; | |
815 | ||
816 | /* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build | |
817 | * failures */ | |
818 | static const int mpeg4audio_sample_rates[16] = { | |
819 | 96000, 88200, 64000, 48000, 44100, 32000, | |
820 | 24000, 22050, 16000, 12000, 11025, 8000, 7350 | |
821 | }; | |
822 | ||
823 | AVCodec ff_aac_encoder = { | |
824 | .name = "aac", | |
825 | .long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), | |
826 | .type = AVMEDIA_TYPE_AUDIO, | |
827 | .id = AV_CODEC_ID_AAC, | |
828 | .priv_data_size = sizeof(AACEncContext), | |
829 | .init = aac_encode_init, | |
830 | .encode2 = aac_encode_frame, | |
831 | .close = aac_encode_end, | |
832 | .supported_samplerates = mpeg4audio_sample_rates, | |
833 | .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | | |
834 | CODEC_CAP_EXPERIMENTAL, | |
835 | .sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, | |
836 | AV_SAMPLE_FMT_NONE }, | |
837 | .priv_class = &aacenc_class, | |
838 | }; |