Mercurial > libavcodec.hg
annotate aac.c @ 7560:0dc289443426 libavcodec
mlp: Cosmetics: Close comments in an extra line and
remove empty lines between doxy comments and function declarations.
author | ramiro |
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date | Wed, 13 Aug 2008 19:43:18 +0000 |
parents | 5298e05f931d |
children | a05954c505ab |
rev | line source |
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7501 | 1 /* |
2 * AAC decoder | |
3 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) | |
4 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) | |
5 * | |
6 * This file is part of FFmpeg. | |
7 * | |
8 * FFmpeg is free software; you can redistribute it and/or | |
9 * modify it under the terms of the GNU Lesser General Public | |
10 * License as published by the Free Software Foundation; either | |
11 * version 2.1 of the License, or (at your option) any later version. | |
12 * | |
13 * FFmpeg is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
16 * Lesser General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU Lesser General Public | |
19 * License along with FFmpeg; if not, write to the Free Software | |
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
21 */ | |
22 | |
23 /** | |
24 * @file aac.c | |
25 * AAC decoder | |
26 * @author Oded Shimon ( ods15 ods15 dyndns org ) | |
27 * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) | |
28 */ | |
29 | |
30 /* | |
31 * supported tools | |
32 * | |
33 * Support? Name | |
34 * N (code in SoC repo) gain control | |
35 * Y block switching | |
36 * Y window shapes - standard | |
37 * N window shapes - Low Delay | |
38 * Y filterbank - standard | |
39 * N (code in SoC repo) filterbank - Scalable Sample Rate | |
40 * Y Temporal Noise Shaping | |
41 * N (code in SoC repo) Long Term Prediction | |
42 * Y intensity stereo | |
43 * Y channel coupling | |
44 * N frequency domain prediction | |
45 * Y Perceptual Noise Substitution | |
46 * Y Mid/Side stereo | |
47 * N Scalable Inverse AAC Quantization | |
48 * N Frequency Selective Switch | |
49 * N upsampling filter | |
50 * Y quantization & coding - AAC | |
51 * N quantization & coding - TwinVQ | |
52 * N quantization & coding - BSAC | |
53 * N AAC Error Resilience tools | |
54 * N Error Resilience payload syntax | |
55 * N Error Protection tool | |
56 * N CELP | |
57 * N Silence Compression | |
58 * N HVXC | |
59 * N HVXC 4kbits/s VR | |
60 * N Structured Audio tools | |
61 * N Structured Audio Sample Bank Format | |
62 * N MIDI | |
63 * N Harmonic and Individual Lines plus Noise | |
64 * N Text-To-Speech Interface | |
65 * N (in progress) Spectral Band Replication | |
66 * Y (not in this code) Layer-1 | |
67 * Y (not in this code) Layer-2 | |
68 * Y (not in this code) Layer-3 | |
69 * N SinuSoidal Coding (Transient, Sinusoid, Noise) | |
70 * N (planned) Parametric Stereo | |
71 * N Direct Stream Transfer | |
72 * | |
73 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. | |
74 * - HE AAC v2 comprises LC AAC with Spectral Band Replication and | |
75 Parametric Stereo. | |
76 */ | |
77 | |
78 | |
79 #include "avcodec.h" | |
80 #include "bitstream.h" | |
81 #include "dsputil.h" | |
82 | |
83 #include "aac.h" | |
84 #include "aactab.h" | |
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85 #include "aacdectab.h" |
7501 | 86 #include "mpeg4audio.h" |
87 | |
88 #include <assert.h> | |
89 #include <errno.h> | |
90 #include <math.h> | |
91 #include <string.h> | |
92 | |
93 #ifndef CONFIG_HARDCODED_TABLES | |
94 static float ff_aac_ivquant_tab[IVQUANT_SIZE]; | |
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95 static float ff_aac_pow2sf_tab[316]; |
7501 | 96 #endif /* CONFIG_HARDCODED_TABLES */ |
97 | |
98 static VLC vlc_scalefactors; | |
99 static VLC vlc_spectral[11]; | |
100 | |
101 | |
7539 | 102 /** |
103 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. | |
104 * | |
105 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
106 * @param sce_map mono (Single Channel Element) map | |
107 * @param type speaker type/position for these channels | |
108 */ | |
109 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
110 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { | |
111 while(n--) { | |
112 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map | |
113 map[get_bits(gb, 4)] = type; | |
114 } | |
115 } | |
116 | |
117 /** | |
118 * Decode program configuration element; reference: table 4.2. | |
119 * | |
120 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
121 * | |
122 * @return Returns error status. 0 - OK, !0 - error | |
123 */ | |
124 static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
125 GetBitContext * gb) { | |
126 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; | |
127 | |
128 skip_bits(gb, 2); // object_type | |
129 | |
130 ac->m4ac.sampling_index = get_bits(gb, 4); | |
131 if(ac->m4ac.sampling_index > 11) { | |
132 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
133 return -1; | |
134 } | |
135 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; | |
7501 | 136 num_front = get_bits(gb, 4); |
137 num_side = get_bits(gb, 4); | |
138 num_back = get_bits(gb, 4); | |
139 num_lfe = get_bits(gb, 2); | |
140 num_assoc_data = get_bits(gb, 3); | |
141 num_cc = get_bits(gb, 4); | |
142 | |
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143 if (get_bits1(gb)) |
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144 skip_bits(gb, 4); // mono_mixdown_tag |
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145 if (get_bits1(gb)) |
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146 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 147 |
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148 if (get_bits1(gb)) |
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149 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 150 |
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151 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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152 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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153 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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154 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 155 |
156 skip_bits_long(gb, 4 * num_assoc_data); | |
157 | |
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158 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 159 |
160 align_get_bits(gb); | |
161 | |
162 /* comment field, first byte is length */ | |
163 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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164 return 0; |
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165 } |
7501 | 166 |
7539 | 167 /** |
168 * Set up channel positions based on a default channel configuration | |
169 * as specified in table 1.17. | |
170 * | |
171 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
172 * | |
173 * @return Returns error status. 0 - OK, !0 - error | |
174 */ | |
175 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
176 int channel_config) | |
177 { | |
178 if(channel_config < 1 || channel_config > 7) { | |
179 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", | |
180 channel_config); | |
181 return -1; | |
182 } | |
183 | |
184 /* default channel configurations: | |
185 * | |
186 * 1ch : front center (mono) | |
187 * 2ch : L + R (stereo) | |
188 * 3ch : front center + L + R | |
189 * 4ch : front center + L + R + back center | |
190 * 5ch : front center + L + R + back stereo | |
191 * 6ch : front center + L + R + back stereo + LFE | |
192 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
193 */ | |
194 | |
195 if(channel_config != 2) | |
196 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) | |
197 if(channel_config > 1) | |
198 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) | |
199 if(channel_config == 4) | |
200 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center | |
201 if(channel_config > 4) | |
202 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] | |
203 = AAC_CHANNEL_BACK; // back stereo | |
204 if(channel_config > 5) | |
205 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE | |
206 if(channel_config == 7) | |
207 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right | |
208 | |
209 return 0; | |
210 } | |
211 | |
212 return -1; | |
213 } | |
214 | |
215 if (get_bits1(gb)) // dependsOnCoreCoder | |
216 skip_bits(gb, 14); // coreCoderDelay | |
217 extension_flag = get_bits1(gb); | |
218 | |
219 if(ac->m4ac.object_type == AOT_AAC_SCALABLE || | |
220 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
221 skip_bits(gb, 3); // layerNr | |
222 | |
223 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
224 if (channel_config == 0) { | |
225 skip_bits(gb, 4); // element_instance_tag | |
226 if((ret = decode_pce(ac, new_che_pos, gb))) | |
227 return ret; | |
228 } else { | |
229 if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) | |
230 return ret; | |
231 } | |
232 if((ret = output_configure(ac, ac->che_pos, new_che_pos))) | |
233 return ret; | |
234 | |
235 if (extension_flag) { | |
236 switch (ac->m4ac.object_type) { | |
237 case AOT_ER_BSAC: | |
238 skip_bits(gb, 5); // numOfSubFrame | |
239 skip_bits(gb, 11); // layer_length | |
240 break; | |
241 case AOT_ER_AAC_LC: | |
242 case AOT_ER_AAC_LTP: | |
243 case AOT_ER_AAC_SCALABLE: | |
244 case AOT_ER_AAC_LD: | |
245 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
246 * aacScalefactorDataResilienceFlag | |
247 * aacSpectralDataResilienceFlag | |
248 */ | |
249 break; | |
250 } | |
251 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
252 } | |
253 return 0; | |
254 } | |
255 | |
256 /** | |
257 * Decode audio specific configuration; reference: table 1.13. | |
258 * | |
259 * @param data pointer to AVCodecContext extradata | |
260 * @param data_size size of AVCCodecContext extradata | |
261 * | |
262 * @return Returns error status. 0 - OK, !0 - error | |
263 */ | |
264 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { | |
265 GetBitContext gb; | |
266 int i; | |
267 | |
268 init_get_bits(&gb, data, data_size * 8); | |
269 | |
270 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) | |
271 return -1; | |
272 if(ac->m4ac.sampling_index > 11) { | |
273 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
274 return -1; | |
275 } | |
276 | |
277 skip_bits_long(&gb, i); | |
278 | |
279 switch (ac->m4ac.object_type) { | |
280 case AOT_AAC_LC: | |
281 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
282 return -1; | |
283 break; | |
284 default: | |
285 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
286 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
287 return -1; | |
288 } | |
289 return 0; | |
290 } | |
291 | |
7501 | 292 static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
293 AACContext * ac = avccontext->priv_data; | |
294 int i; | |
295 | |
296 ac->avccontext = avccontext; | |
297 | |
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298 if (avccontext->extradata_size <= 0 || |
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299 decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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300 return -1; |
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301 |
7539 | 302 avccontext->sample_fmt = SAMPLE_FMT_S16; |
7501 | 303 avccontext->sample_rate = ac->m4ac.sample_rate; |
304 avccontext->frame_size = 1024; | |
305 | |
306 AAC_INIT_VLC_STATIC( 0, 144); | |
307 AAC_INIT_VLC_STATIC( 1, 114); | |
308 AAC_INIT_VLC_STATIC( 2, 188); | |
309 AAC_INIT_VLC_STATIC( 3, 180); | |
310 AAC_INIT_VLC_STATIC( 4, 172); | |
311 AAC_INIT_VLC_STATIC( 5, 140); | |
312 AAC_INIT_VLC_STATIC( 6, 168); | |
313 AAC_INIT_VLC_STATIC( 7, 114); | |
314 AAC_INIT_VLC_STATIC( 8, 262); | |
315 AAC_INIT_VLC_STATIC( 9, 248); | |
316 AAC_INIT_VLC_STATIC(10, 384); | |
317 | |
318 dsputil_init(&ac->dsp, avccontext); | |
319 | |
7539 | 320 ac->random_state = 0x1f2e3d4c; |
321 | |
7501 | 322 // -1024 - Compensate wrong IMDCT method. |
323 // 32768 - Required to scale values to the correct range for the bias method | |
324 // for float to int16 conversion. | |
325 | |
326 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { | |
327 ac->add_bias = 385.0f; | |
328 ac->sf_scale = 1. / (-1024. * 32768.); | |
329 ac->sf_offset = 0; | |
330 } else { | |
331 ac->add_bias = 0.0f; | |
332 ac->sf_scale = 1. / -1024.; | |
333 ac->sf_offset = 60; | |
334 } | |
335 | |
336 #ifndef CONFIG_HARDCODED_TABLES | |
337 for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++) | |
338 ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] = cbrt(fabs(i)) * i; | |
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339 for (i = 0; i < 316; i++) |
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340 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
7501 | 341 #endif /* CONFIG_HARDCODED_TABLES */ |
342 | |
343 INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]), | |
344 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), | |
345 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
346 352); | |
347 | |
348 ff_mdct_init(&ac->mdct, 11, 1); | |
349 ff_mdct_init(&ac->mdct_small, 8, 1); | |
350 return 0; | |
351 } | |
352 | |
7539 | 353 /** |
354 * Skip data_stream_element; reference: table 4.10. | |
355 */ | |
356 static void skip_data_stream_element(GetBitContext * gb) { | |
7501 | 357 int byte_align = get_bits1(gb); |
358 int count = get_bits(gb, 8); | |
359 if (count == 255) | |
360 count += get_bits(gb, 8); | |
361 if (byte_align) | |
362 align_get_bits(gb); | |
363 skip_bits_long(gb, 8 * count); | |
364 } | |
365 | |
366 /** | |
7539 | 367 * Decode Individual Channel Stream info; reference: table 4.6. |
368 * | |
369 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
370 */ | |
371 static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { | |
372 if (get_bits1(gb)) { | |
373 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); | |
374 memset(ics, 0, sizeof(IndividualChannelStream)); | |
375 return -1; | |
376 } | |
377 ics->window_sequence[1] = ics->window_sequence[0]; | |
378 ics->window_sequence[0] = get_bits(gb, 2); | |
379 ics->use_kb_window[1] = ics->use_kb_window[0]; | |
380 ics->use_kb_window[0] = get_bits1(gb); | |
381 ics->num_window_groups = 1; | |
382 ics->group_len[0] = 1; | |
383 | |
384 return 0; | |
385 } | |
386 | |
387 /** | |
7501 | 388 * inverse quantization |
389 * | |
390 * @param a quantized value to be dequantized | |
391 * @return Returns dequantized value. | |
392 */ | |
393 static inline float ivquant(int a) { | |
394 if (a + (unsigned int)IVQUANT_SIZE/2 - 1 < (unsigned int)IVQUANT_SIZE - 1) | |
395 return ff_aac_ivquant_tab[a + IVQUANT_SIZE/2 - 1]; | |
396 else | |
397 return cbrtf(fabsf(a)) * a; | |
398 } | |
399 | |
7539 | 400 /** |
401 * Decode band types (section_data payload); reference: table 4.46. | |
402 * | |
403 * @param band_type array of the used band type | |
404 * @param band_type_run_end array of the last scalefactor band of a band type run | |
405 * | |
406 * @return Returns error status. 0 - OK, !0 - error | |
407 */ | |
408 static int decode_band_types(AACContext * ac, enum BandType band_type[120], | |
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409 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
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410 int g, idx = 0; |
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411 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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412 for (g = 0; g < ics->num_window_groups; g++) { |
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413 int k = 0; |
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414 while (k < ics->max_sfb) { |
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415 uint8_t sect_len = k; |
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416 int sect_len_incr; |
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417 int sect_band_type = get_bits(gb, 4); |
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418 if (sect_band_type == 12) { |
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419 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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420 return -1; |
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421 } |
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422 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
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423 sect_len += sect_len_incr; |
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424 sect_len += sect_len_incr; |
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425 if (sect_len > ics->max_sfb) { |
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426 av_log(ac->avccontext, AV_LOG_ERROR, |
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427 "Number of bands (%d) exceeds limit (%d).\n", |
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428 sect_len, ics->max_sfb); |
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429 return -1; |
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430 } |
7539 | 431 } |
432 } | |
433 return 0; | |
434 } | |
7523
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435 |
7539 | 436 /** |
437 * Decode scalefactors; reference: table 4.47. | |
7523
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438 * |
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439 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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440 * @param band_type array of the used band type |
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441 * @param band_type_run_end array of the last scalefactor band of a band type run |
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442 * @param sf array of scalefactors or intensity stereo positions |
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443 * |
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444 * @return Returns error status. 0 - OK, !0 - error |
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445 */ |
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446 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
7540 | 447 unsigned int global_gain, IndividualChannelStream * ics, |
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448 enum BandType band_type[120], int band_type_run_end[120]) { |
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449 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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450 int g, i, idx = 0; |
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451 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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452 int noise_flag = 1; |
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453 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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454 ics->intensity_present = 0; |
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455 for (g = 0; g < ics->num_window_groups; g++) { |
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456 for (i = 0; i < ics->max_sfb;) { |
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457 int run_end = band_type_run_end[idx]; |
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458 if (band_type[idx] == ZERO_BT) { |
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459 for(; i < run_end; i++, idx++) |
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460 sf[idx] = 0.; |
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461 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
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462 ics->intensity_present = 1; |
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463 for(; i < run_end; i++, idx++) { |
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464 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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465 if(offset[2] > 255U) { |
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466 av_log(ac->avccontext, AV_LOG_ERROR, |
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467 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
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468 return -1; |
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469 } |
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470 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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471 } |
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472 }else if(band_type[idx] == NOISE_BT) { |
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473 for(; i < run_end; i++, idx++) { |
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474 if(noise_flag-- > 0) |
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475 offset[1] += get_bits(gb, 9) - 256; |
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476 else |
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477 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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478 if(offset[1] > 255U) { |
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479 av_log(ac->avccontext, AV_LOG_ERROR, |
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480 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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481 return -1; |
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482 } |
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483 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset]; |
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484 } |
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485 }else { |
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486 for(; i < run_end; i++, idx++) { |
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487 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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488 if(offset[0] > 255U) { |
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489 av_log(ac->avccontext, AV_LOG_ERROR, |
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490 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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491 return -1; |
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492 } |
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493 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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494 } |
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495 } |
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496 } |
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497 } |
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498 return 0; |
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499 } |
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500 |
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501 /** |
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502 * Decode pulse data; reference: table 4.7. |
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503 */ |
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504 static void decode_pulses(Pulse * pulse, GetBitContext * gb) { |
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505 int i; |
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506 pulse->num_pulse = get_bits(gb, 2) + 1; |
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507 pulse->start = get_bits(gb, 6); |
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508 for (i = 0; i < pulse->num_pulse; i++) { |
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509 pulse->offset[i] = get_bits(gb, 5); |
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510 pulse->amp [i] = get_bits(gb, 4); |
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511 } |
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512 } |
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513 |
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514 /** |
7539 | 515 * Decode Mid/Side data; reference: table 4.54. |
516 * | |
517 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
518 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
519 * [3] reserved for scalable AAC | |
520 */ | |
521 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, | |
522 int ms_present) { | |
523 | |
524 /** | |
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525 * Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3. |
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526 * |
7501 | 527 * @param pulse pointer to pulse data struct |
528 * @param icoef array of quantized spectral data | |
529 */ | |
530 static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) { | |
531 int i, off = ics->swb_offset[pulse->start]; | |
532 for (i = 0; i < pulse->num_pulse; i++) { | |
533 int ic; | |
534 off += pulse->offset[i]; | |
535 ic = (icoef[off] - 1)>>31; | |
536 icoef[off] += (pulse->amp[i]^ic) - ic; | |
537 } | |
538 } | |
539 | |
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540 /** |
7539 | 541 * Decode an individual_channel_stream payload; reference: table 4.44. |
542 * | |
543 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
544 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
545 * | |
546 * @return Returns error status. 0 - OK, !0 - error | |
547 */ | |
548 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { | |
549 int icoeffs[1024]; | |
550 Pulse pulse; | |
551 TemporalNoiseShaping * tns = &sce->tns; | |
552 IndividualChannelStream * ics = &sce->ics; | |
553 float * out = sce->coeffs; | |
554 int global_gain, pulse_present = 0; | |
555 | |
556 /* These two assignments are to silence some GCC warnings about the | |
557 * variables being used uninitialised when in fact they always are. | |
558 */ | |
559 pulse.num_pulse = 0; | |
560 pulse.start = 0; | |
561 | |
562 global_gain = get_bits(gb, 8); | |
563 | |
564 if (!common_window && !scale_flag) { | |
565 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
566 return -1; | |
567 } | |
568 | |
569 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
570 return -1; | |
571 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
572 return -1; | |
573 | |
574 pulse_present = 0; | |
575 if (!scale_flag) { | |
576 if ((pulse_present = get_bits1(gb))) { | |
577 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
578 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
579 return -1; | |
580 } | |
581 decode_pulses(&pulse, gb); | |
582 } | |
583 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
584 return -1; | |
585 if (get_bits1(gb)) { | |
586 av_log_missing_feature(ac->avccontext, "SSR", 1); | |
587 return -1; | |
588 } | |
589 } | |
590 | |
591 if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0) | |
592 return -1; | |
593 if (pulse_present) | |
594 add_pulses(icoeffs, &pulse, ics); | |
595 dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type); | |
596 return 0; | |
597 } | |
598 | |
599 /** | |
600 * Decode a channel_pair_element; reference: table 4.4. | |
601 * | |
602 * @param elem_id Identifies the instance of a syntax element. | |
603 * | |
604 * @return Returns error status. 0 - OK, !0 - error | |
605 */ | |
606 static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) { | |
607 int i, ret, common_window, ms_present = 0; | |
608 ChannelElement * cpe; | |
609 | |
610 cpe = ac->che[TYPE_CPE][elem_id]; | |
611 common_window = get_bits1(gb); | |
612 if (common_window) { | |
613 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
614 return -1; | |
615 i = cpe->ch[1].ics.use_kb_window[0]; | |
616 cpe->ch[1].ics = cpe->ch[0].ics; | |
617 cpe->ch[1].ics.use_kb_window[1] = i; | |
618 ms_present = get_bits(gb, 2); | |
619 if(ms_present == 3) { | |
620 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); | |
621 return -1; | |
622 } else if(ms_present) | |
623 decode_mid_side_stereo(cpe, gb, ms_present); | |
624 } | |
625 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
626 return ret; | |
627 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
628 return ret; | |
629 | |
630 if (common_window && ms_present) | |
631 apply_mid_side_stereo(cpe); | |
632 | |
633 if (cpe->ch[1].ics.intensity_present) | |
634 apply_intensity_stereo(cpe, ms_present); | |
635 return 0; | |
636 } | |
637 | |
638 /** | |
639 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
7523
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640 * |
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641 * @param crc flag indicating the presence of CRC checksum |
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642 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 643 * |
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644 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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645 */ |
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646 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
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647 // TODO : sbr_extension implementation |
7540 | 648 av_log_missing_feature(ac->avccontext, "SBR", 0); |
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649 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
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650 return cnt; |
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651 } |
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652 |
7539 | 653 /** |
654 * Decode dynamic range information; reference: table 4.52. | |
655 * | |
656 * @param cnt length of TYPE_FIL syntactic element in bytes | |
657 * | |
658 * @return Returns number of bytes consumed. | |
659 */ | |
660 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { | |
661 int n = 1; | |
662 int drc_num_bands = 1; | |
663 int i; | |
664 | |
665 /* pce_tag_present? */ | |
666 if(get_bits1(gb)) { | |
667 che_drc->pce_instance_tag = get_bits(gb, 4); | |
668 skip_bits(gb, 4); // tag_reserved_bits | |
669 n++; | |
670 } | |
671 | |
672 /* excluded_chns_present? */ | |
673 if(get_bits1(gb)) { | |
674 n += decode_drc_channel_exclusions(che_drc, gb); | |
675 } | |
676 | |
677 /* drc_bands_present? */ | |
678 if (get_bits1(gb)) { | |
679 che_drc->band_incr = get_bits(gb, 4); | |
680 che_drc->interpolation_scheme = get_bits(gb, 4); | |
681 n++; | |
682 drc_num_bands += che_drc->band_incr; | |
683 for (i = 0; i < drc_num_bands; i++) { | |
684 che_drc->band_top[i] = get_bits(gb, 8); | |
685 n++; | |
686 } | |
687 } | |
688 | |
689 /* prog_ref_level_present? */ | |
690 if (get_bits1(gb)) { | |
691 che_drc->prog_ref_level = get_bits(gb, 7); | |
692 skip_bits1(gb); // prog_ref_level_reserved_bits | |
693 n++; | |
694 } | |
695 | |
696 for (i = 0; i < drc_num_bands; i++) { | |
697 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
698 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
699 n++; | |
700 } | |
701 | |
702 return n; | |
703 } | |
704 | |
705 /** | |
706 * Decode extension data (incomplete); reference: table 4.51. | |
707 * | |
708 * @param cnt length of TYPE_FIL syntactic element in bytes | |
709 * | |
710 * @return Returns number of bytes consumed | |
711 */ | |
712 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { | |
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713 int crc_flag = 0; |
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714 int res = cnt; |
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715 switch (get_bits(gb, 4)) { // extension type |
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716 case EXT_SBR_DATA_CRC: |
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717 crc_flag++; |
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718 case EXT_SBR_DATA: |
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719 res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
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720 break; |
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721 case EXT_DYNAMIC_RANGE: |
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722 res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
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723 break; |
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724 case EXT_FILL: |
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725 case EXT_FILL_DATA: |
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726 case EXT_DATA_ELEMENT: |
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727 default: |
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728 skip_bits_long(gb, 8*cnt - 4); |
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729 break; |
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730 }; |
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731 return res; |
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732 } |
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733 |
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734 /** |
7539 | 735 * Conduct IMDCT and windowing. |
736 */ | |
737 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { | |
738 IndividualChannelStream * ics = &sce->ics; | |
739 float * in = sce->coeffs; | |
740 float * out = sce->ret; | |
741 float * saved = sce->saved; | |
742 const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024; | |
743 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128; | |
744 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024; | |
745 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128; | |
746 float * buf = ac->buf_mdct; | |
747 int i; | |
748 | |
749 /** | |
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750 * Apply dependent channel coupling (applied before IMDCT). |
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751 * |
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752 * @param index index into coupling gain array |
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753 */ |
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754 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) { |
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755 IndividualChannelStream * ics = &cc->ch[0].ics; |
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756 const uint16_t * offsets = ics->swb_offset; |
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757 float * dest = sce->coeffs; |
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758 const float * src = cc->ch[0].coeffs; |
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759 int g, i, group, k, idx = 0; |
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760 if(ac->m4ac.object_type == AOT_AAC_LTP) { |
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761 av_log(ac->avccontext, AV_LOG_ERROR, |
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762 "Dependent coupling is not supported together with LTP\n"); |
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763 return; |
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764 } |
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765 for (g = 0; g < ics->num_window_groups; g++) { |
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766 for (i = 0; i < ics->max_sfb; i++, idx++) { |
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767 if (cc->ch[0].band_type[idx] != ZERO_BT) { |
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768 for (group = 0; group < ics->group_len[g]; group++) { |
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769 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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770 // XXX dsputil-ize |
7540 | 771 dest[group*128+k] += cc->coup.gain[index][idx] * src[group*128+k]; |
7523
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772 } |
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773 } |
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774 } |
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775 } |
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776 dest += ics->group_len[g]*128; |
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777 src += ics->group_len[g]*128; |
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778 } |
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779 } |
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780 |
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781 /** |
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782 * Apply independent channel coupling (applied after IMDCT). |
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783 * |
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784 * @param index index into coupling gain array |
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785 */ |
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786 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) { |
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787 int i; |
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788 for (i = 0; i < 1024; i++) |
7540 | 789 sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias); |
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790 } |
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791 |
7539 | 792 if (!ac->is_saved) { |
793 ac->is_saved = 1; | |
794 *data_size = 0; | |
795 return 0; | |
796 } | |
797 | |
798 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
799 if(*data_size < data_size_tmp) { | |
800 av_log(avccontext, AV_LOG_ERROR, | |
801 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
802 *data_size, data_size_tmp); | |
803 return -1; | |
804 } | |
805 *data_size = data_size_tmp; | |
806 | |
807 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
808 | |
809 return buf_size; | |
810 } | |
811 | |
7501 | 812 static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
813 AACContext * ac = avccontext->priv_data; | |
7540 | 814 int i, type; |
7501 | 815 |
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816 for (i = 0; i < MAX_ELEM_ID; i++) { |
7540 | 817 for(type = 0; type < 4; type++) |
818 av_freep(&ac->che[type][i]); | |
7501 | 819 } |
820 | |
821 ff_mdct_end(&ac->mdct); | |
822 ff_mdct_end(&ac->mdct_small); | |
823 return 0 ; | |
824 } | |
825 | |
826 AVCodec aac_decoder = { | |
827 "aac", | |
828 CODEC_TYPE_AUDIO, | |
829 CODEC_ID_AAC, | |
830 sizeof(AACContext), | |
831 aac_decode_init, | |
832 NULL, | |
833 aac_decode_close, | |
834 aac_decode_frame, | |
835 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
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836 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7501 | 837 }; |