Mercurial > libavcodec.hg
annotate aacenc.c @ 8651:7b19ec801069 libavcodec
flacdec: remove another unneeded debugging statement
author | jbr |
---|---|
date | Sat, 24 Jan 2009 16:02:30 +0000 |
parents | dc1a7a6ec58d |
children | e9d9d946f213 |
rev | line source |
---|---|
7571 | 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 aacenc.c | |
24 * AAC encoder | |
25 */ | |
26 | |
27 /*********************************** | |
28 * TODOs: | |
29 * psy model selection with some option | |
7584 | 30 * add sane pulse detection |
7681 | 31 * add temporal noise shaping |
7571 | 32 ***********************************/ |
33 | |
34 #include "avcodec.h" | |
35 #include "bitstream.h" | |
36 #include "dsputil.h" | |
37 #include "mpeg4audio.h" | |
38 | |
39 #include "aacpsy.h" | |
40 #include "aac.h" | |
41 #include "aactab.h" | |
42 | |
43 static const uint8_t swb_size_1024_96[] = { | |
44 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, | |
45 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, | |
46 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64 | |
47 }; | |
48 | |
49 static const uint8_t swb_size_1024_64[] = { | |
50 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, | |
51 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36, | |
52 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40 | |
53 }; | |
54 | |
55 static const uint8_t swb_size_1024_48[] = { | |
56 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, | |
57 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, | |
58 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, | |
59 96 | |
60 }; | |
61 | |
62 static const uint8_t swb_size_1024_32[] = { | |
63 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, | |
64 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, | |
65 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32 | |
66 }; | |
67 | |
68 static const uint8_t swb_size_1024_24[] = { | |
69 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, | |
70 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, | |
71 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64 | |
72 }; | |
73 | |
74 static const uint8_t swb_size_1024_16[] = { | |
75 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, | |
76 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, | |
77 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64 | |
78 }; | |
79 | |
80 static const uint8_t swb_size_1024_8[] = { | |
81 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, | |
82 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28, | |
83 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80 | |
84 }; | |
85 | |
7978 | 86 static const uint8_t * const swb_size_1024[] = { |
7571 | 87 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64, |
88 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32, | |
89 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16, | |
90 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8 | |
91 }; | |
92 | |
93 static const uint8_t swb_size_128_96[] = { | |
94 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36 | |
95 }; | |
96 | |
97 static const uint8_t swb_size_128_48[] = { | |
98 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16 | |
99 }; | |
100 | |
101 static const uint8_t swb_size_128_24[] = { | |
102 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20 | |
103 }; | |
104 | |
105 static const uint8_t swb_size_128_16[] = { | |
106 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20 | |
107 }; | |
108 | |
109 static const uint8_t swb_size_128_8[] = { | |
110 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20 | |
111 }; | |
112 | |
7978 | 113 static const uint8_t * const swb_size_128[] = { |
7571 | 114 /* the last entry on the following row is swb_size_128_64 but is a |
115 duplicate of swb_size_128_96 */ | |
116 swb_size_128_96, swb_size_128_96, swb_size_128_96, | |
117 swb_size_128_48, swb_size_128_48, swb_size_128_48, | |
118 swb_size_128_24, swb_size_128_24, swb_size_128_16, | |
119 swb_size_128_16, swb_size_128_16, swb_size_128_8 | |
120 }; | |
121 | |
7596 | 122 /** bits needed to code codebook run value for long windows */ |
123 static const uint8_t run_value_bits_long[64] = { | |
124 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, | |
125 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10, | |
126 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, | |
127 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15 | |
128 }; | |
129 | |
130 /** bits needed to code codebook run value for short windows */ | |
131 static const uint8_t run_value_bits_short[16] = { | |
132 3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9 | |
133 }; | |
134 | |
7978 | 135 static const uint8_t* const run_value_bits[2] = { |
7596 | 136 run_value_bits_long, run_value_bits_short |
137 }; | |
138 | |
7571 | 139 /** default channel configurations */ |
140 static const uint8_t aac_chan_configs[6][5] = { | |
7585 | 141 {1, TYPE_SCE}, // 1 channel - single channel element |
142 {1, TYPE_CPE}, // 2 channels - channel pair | |
143 {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo | |
144 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center | |
145 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo | |
146 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE | |
7571 | 147 }; |
148 | |
149 /** | |
7596 | 150 * structure used in optimal codebook search |
151 */ | |
152 typedef struct BandCodingPath { | |
153 int prev_idx; ///< pointer to the previous path point | |
154 int codebook; ///< codebook for coding band run | |
155 int bits; ///< number of bit needed to code given number of bands | |
156 } BandCodingPath; | |
157 | |
158 /** | |
7584 | 159 * AAC encoder context |
160 */ | |
161 typedef struct { | |
162 PutBitContext pb; | |
163 MDCTContext mdct1024; ///< long (1024 samples) frame transform context | |
164 MDCTContext mdct128; ///< short (128 samples) frame transform context | |
165 DSPContext dsp; | |
7605 | 166 DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients |
167 int16_t* samples; ///< saved preprocessed input | |
168 | |
169 int samplerate_index; ///< MPEG-4 samplerate index | |
170 | |
171 ChannelElement *cpe; ///< channel elements | |
7596 | 172 AACPsyContext psy; ///< psychoacoustic model context |
173 int last_frame; | |
7584 | 174 } AACEncContext; |
175 | |
176 /** | |
7571 | 177 * Make AAC audio config object. |
178 * @see 1.6.2.1 "Syntax - AudioSpecificConfig" | |
179 */ | |
180 static void put_audio_specific_config(AVCodecContext *avctx) | |
181 { | |
182 PutBitContext pb; | |
183 AACEncContext *s = avctx->priv_data; | |
184 | |
185 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8); | |
186 put_bits(&pb, 5, 2); //object type - AAC-LC | |
187 put_bits(&pb, 4, s->samplerate_index); //sample rate index | |
188 put_bits(&pb, 4, avctx->channels); | |
189 //GASpecificConfig | |
190 put_bits(&pb, 1, 0); //frame length - 1024 samples | |
191 put_bits(&pb, 1, 0); //does not depend on core coder | |
192 put_bits(&pb, 1, 0); //is not extension | |
193 flush_put_bits(&pb); | |
194 } | |
195 | |
196 static av_cold int aac_encode_init(AVCodecContext *avctx) | |
197 { | |
198 AACEncContext *s = avctx->priv_data; | |
199 int i; | |
200 | |
201 avctx->frame_size = 1024; | |
202 | |
203 for(i = 0; i < 16; i++) | |
204 if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i]) | |
205 break; | |
206 if(i == 16){ | |
207 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate); | |
208 return -1; | |
209 } | |
210 if(avctx->channels > 6){ | |
211 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels); | |
212 return -1; | |
213 } | |
214 s->samplerate_index = i; | |
215 | |
216 dsputil_init(&s->dsp, avctx); | |
217 ff_mdct_init(&s->mdct1024, 11, 0); | |
218 ff_mdct_init(&s->mdct128, 8, 0); | |
7584 | 219 // window init |
220 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
221 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
222 ff_sine_window_init(ff_sine_1024, 1024); | |
223 ff_sine_window_init(ff_sine_128, 128); | |
7571 | 224 |
225 s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0])); | |
226 s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]); | |
7605 | 227 if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP, |
228 aac_chan_configs[avctx->channels-1][0], 0, | |
7681 | 229 swb_size_1024[i], ff_aac_num_swb_1024[i], swb_size_128[i], ff_aac_num_swb_128[i]) < 0){ |
7571 | 230 av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n"); |
231 return -1; | |
232 } | |
233 avctx->extradata = av_malloc(2); | |
234 avctx->extradata_size = 2; | |
235 put_audio_specific_config(avctx); | |
236 return 0; | |
237 } | |
238 | |
239 /** | |
240 * Encode ics_info element. | |
241 * @see Table 4.6 (syntax of ics_info) | |
242 */ | |
7681 | 243 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info) |
7571 | 244 { |
245 int i; | |
246 | |
247 put_bits(&s->pb, 1, 0); // ics_reserved bit | |
248 put_bits(&s->pb, 2, info->window_sequence[0]); | |
249 put_bits(&s->pb, 1, info->use_kb_window[0]); | |
250 if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){ | |
251 put_bits(&s->pb, 6, info->max_sfb); | |
252 put_bits(&s->pb, 1, 0); // no prediction | |
253 }else{ | |
254 put_bits(&s->pb, 4, info->max_sfb); | |
255 for(i = 1; i < info->num_windows; i++) | |
256 put_bits(&s->pb, 1, info->group_len[i]); | |
257 } | |
258 } | |
259 | |
260 /** | |
7681 | 261 * Calculate the number of bits needed to code all coefficient signs in current band. |
262 */ | |
263 static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce, | |
264 int group_len, int start, int size) | |
265 { | |
266 int bits = 0; | |
267 int i, w; | |
268 for(w = 0; w < group_len; w++){ | |
269 for(i = 0; i < size; i++){ | |
270 if(sce->icoefs[start + i]) | |
271 bits++; | |
272 } | |
273 start += 128; | |
274 } | |
275 return bits; | |
276 } | |
277 | |
278 /** | |
7584 | 279 * Encode pulse data. |
280 */ | |
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281 static void encode_pulses(AACEncContext *s, Pulse *pulse) |
7584 | 282 { |
283 int i; | |
284 | |
285 put_bits(&s->pb, 1, !!pulse->num_pulse); | |
286 if(!pulse->num_pulse) return; | |
287 | |
288 put_bits(&s->pb, 2, pulse->num_pulse - 1); | |
289 put_bits(&s->pb, 6, pulse->start); | |
290 for(i = 0; i < pulse->num_pulse; i++){ | |
7585 | 291 put_bits(&s->pb, 5, pulse->pos[i]); |
7584 | 292 put_bits(&s->pb, 4, pulse->amp[i]); |
293 } | |
294 } | |
295 | |
296 /** | |
297 * Encode spectral coefficients processed by psychoacoustic model. | |
298 */ | |
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299 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce) |
7584 | 300 { |
301 int start, i, w, w2, wg; | |
302 | |
303 w = 0; | |
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304 for(wg = 0; wg < sce->ics.num_window_groups; wg++){ |
7584 | 305 start = 0; |
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306 for(i = 0; i < sce->ics.max_sfb; i++){ |
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307 if(sce->zeroes[w*16 + i]){ |
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308 start += sce->ics.swb_sizes[i]; |
7584 | 309 continue; |
310 } | |
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311 for(w2 = w; w2 < w + sce->ics.group_len[wg]; w2++){ |
7681 | 312 encode_band_coeffs(s, sce, start + w2*128, |
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313 sce->ics.swb_sizes[i], |
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314 sce->band_type[w*16 + i]); |
7584 | 315 } |
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316 start += sce->ics.swb_sizes[i]; |
7584 | 317 } |
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318 w += sce->ics.group_len[wg]; |
7584 | 319 } |
320 } | |
321 | |
322 /** | |
7571 | 323 * Write some auxiliary information about the created AAC file. |
324 */ | |
325 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name) | |
326 { | |
327 int i, namelen, padbits; | |
328 | |
329 namelen = strlen(name) + 2; | |
7585 | 330 put_bits(&s->pb, 3, TYPE_FIL); |
7571 | 331 put_bits(&s->pb, 4, FFMIN(namelen, 15)); |
332 if(namelen >= 15) | |
333 put_bits(&s->pb, 8, namelen - 16); | |
334 put_bits(&s->pb, 4, 0); //extension type - filler | |
335 padbits = 8 - (put_bits_count(&s->pb) & 7); | |
336 align_put_bits(&s->pb); | |
337 for(i = 0; i < namelen - 2; i++) | |
338 put_bits(&s->pb, 8, name[i]); | |
339 put_bits(&s->pb, 12 - padbits, 0); | |
340 } | |
341 | |
342 static av_cold int aac_encode_end(AVCodecContext *avctx) | |
343 { | |
344 AACEncContext *s = avctx->priv_data; | |
345 | |
346 ff_mdct_end(&s->mdct1024); | |
347 ff_mdct_end(&s->mdct128); | |
348 ff_aac_psy_end(&s->psy); | |
349 av_freep(&s->samples); | |
350 av_freep(&s->cpe); | |
351 return 0; | |
352 } | |
353 | |
354 AVCodec aac_encoder = { | |
355 "aac", | |
356 CODEC_TYPE_AUDIO, | |
357 CODEC_ID_AAC, | |
358 sizeof(AACEncContext), | |
359 aac_encode_init, | |
360 aac_encode_frame, | |
361 aac_encode_end, | |
362 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY, | |
363 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, | |
364 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
365 }; |