Mercurial > libavcodec.hg
annotate aac.c @ 10795:342c2dbe0f94 libavcodec
More doxy
author | michael |
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date | Thu, 07 Jan 2010 15:12:31 +0000 |
parents | 750102456f00 |
children | dc84199b3306 |
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 /** | |
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24 * @file libavcodec/aac.c |
7501 | 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 | |
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44 * Y frequency domain prediction |
7501 | 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" | |
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80 #include "internal.h" |
9428 | 81 #include "get_bits.h" |
7501 | 82 #include "dsputil.h" |
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83 #include "lpc.h" |
7501 | 84 |
85 #include "aac.h" | |
86 #include "aactab.h" | |
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87 #include "aacdectab.h" |
7501 | 88 #include "mpeg4audio.h" |
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89 #include "aac_parser.h" |
7501 | 90 |
91 #include <assert.h> | |
92 #include <errno.h> | |
93 #include <math.h> | |
94 #include <string.h> | |
95 | |
10004 | 96 union float754 { |
97 float f; | |
98 uint32_t i; | |
99 }; | |
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100 |
7501 | 101 static VLC vlc_scalefactors; |
102 static VLC vlc_spectral[11]; | |
103 | |
104 | |
10004 | 105 static ChannelElement *get_che(AACContext *ac, int type, int elem_id) |
106 { | |
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107 if (ac->tag_che_map[type][elem_id]) { |
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108 return ac->tag_che_map[type][elem_id]; |
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109 } |
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110 if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) { |
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111 return NULL; |
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112 } |
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113 switch (ac->m4ac.chan_config) { |
10004 | 114 case 7: |
115 if (ac->tags_mapped == 3 && type == TYPE_CPE) { | |
116 ac->tags_mapped++; | |
117 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; | |
118 } | |
119 case 6: | |
120 /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] | |
121 instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have | |
122 encountered such a stream, transfer the LFE[0] element to SCE[1] */ | |
123 if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { | |
124 ac->tags_mapped++; | |
125 return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; | |
126 } | |
127 case 5: | |
128 if (ac->tags_mapped == 2 && type == TYPE_CPE) { | |
129 ac->tags_mapped++; | |
130 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; | |
131 } | |
132 case 4: | |
133 if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { | |
134 ac->tags_mapped++; | |
135 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; | |
136 } | |
137 case 3: | |
138 case 2: | |
139 if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { | |
140 ac->tags_mapped++; | |
141 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; | |
142 } else if (ac->m4ac.chan_config == 2) { | |
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143 return NULL; |
10004 | 144 } |
145 case 1: | |
146 if (!ac->tags_mapped && type == TYPE_SCE) { | |
147 ac->tags_mapped++; | |
148 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; | |
149 } | |
150 default: | |
151 return NULL; | |
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152 } |
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153 } |
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154 |
7539 | 155 /** |
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156 * Check for the channel element in the current channel position configuration. |
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157 * If it exists, make sure the appropriate element is allocated and map the |
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158 * channel order to match the internal FFmpeg channel layout. |
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159 * |
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160 * @param che_pos current channel position configuration |
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161 * @param type channel element type |
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162 * @param id channel element id |
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163 * @param channels count of the number of channels in the configuration |
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164 * |
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165 * @return Returns error status. 0 - OK, !0 - error |
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166 */ |
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167 static int che_configure(AACContext *ac, |
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168 enum ChannelPosition che_pos[4][MAX_ELEM_ID], |
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169 int type, int id, |
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170 int *channels) |
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171 { |
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172 if (che_pos[type][id]) { |
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173 if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement)))) |
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174 return AVERROR(ENOMEM); |
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175 if (type != TYPE_CCE) { |
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176 ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret; |
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177 if (type == TYPE_CPE) { |
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178 ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret; |
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179 } |
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180 } |
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181 } else |
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182 av_freep(&ac->che[type][id]); |
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183 return 0; |
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184 } |
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185 |
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186 /** |
7578 | 187 * Configure output channel order based on the current program configuration element. |
188 * | |
189 * @param che_pos current channel position configuration | |
190 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
191 * | |
192 * @return Returns error status. 0 - OK, !0 - error | |
193 */ | |
10004 | 194 static int output_configure(AACContext *ac, |
195 enum ChannelPosition che_pos[4][MAX_ELEM_ID], | |
196 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
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197 int channel_config, enum OCStatus oc_type) |
10004 | 198 { |
7578 | 199 AVCodecContext *avctx = ac->avccontext; |
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200 int i, type, channels = 0, ret; |
7578 | 201 |
202 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
203 | |
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204 if (channel_config) { |
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205 for (i = 0; i < tags_per_config[channel_config]; i++) { |
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206 if ((ret = che_configure(ac, che_pos, |
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207 aac_channel_layout_map[channel_config - 1][i][0], |
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208 aac_channel_layout_map[channel_config - 1][i][1], |
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209 &channels))) |
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210 return ret; |
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211 } |
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212 |
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213 memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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214 ac->tags_mapped = 0; |
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215 |
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216 avctx->channel_layout = aac_channel_layout[channel_config - 1]; |
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217 } else { |
10307 | 218 /* Allocate or free elements depending on if they are in the |
219 * current program configuration. | |
220 * | |
221 * Set up default 1:1 output mapping. | |
222 * | |
223 * For a 5.1 stream the output order will be: | |
224 * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] | |
225 */ | |
7578 | 226 |
10307 | 227 for (i = 0; i < MAX_ELEM_ID; i++) { |
228 for (type = 0; type < 4; type++) { | |
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229 if ((ret = che_configure(ac, che_pos, type, i, &channels))) |
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230 return ret; |
10307 | 231 } |
7578 | 232 } |
233 | |
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234 memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
10004 | 235 ac->tags_mapped = 4 * MAX_ELEM_ID; |
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236 |
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237 avctx->channel_layout = 0; |
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238 } |
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239 |
7578 | 240 avctx->channels = channels; |
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241 |
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242 ac->output_configured = oc_type; |
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243 |
7578 | 244 return 0; |
245 } | |
246 | |
247 /** | |
7539 | 248 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
249 * | |
250 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
251 * @param sce_map mono (Single Channel Element) map | |
252 * @param type speaker type/position for these channels | |
253 */ | |
254 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
10004 | 255 enum ChannelPosition *sce_map, |
256 enum ChannelPosition type, | |
257 GetBitContext *gb, int n) | |
258 { | |
259 while (n--) { | |
7539 | 260 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map |
261 map[get_bits(gb, 4)] = type; | |
262 } | |
263 } | |
264 | |
265 /** | |
266 * Decode program configuration element; reference: table 4.2. | |
267 * | |
268 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
269 * | |
270 * @return Returns error status. 0 - OK, !0 - error | |
271 */ | |
10004 | 272 static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], |
273 GetBitContext *gb) | |
274 { | |
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275 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index; |
7539 | 276 |
277 skip_bits(gb, 2); // object_type | |
278 | |
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279 sampling_index = get_bits(gb, 4); |
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280 if (ac->m4ac.sampling_index != sampling_index) |
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281 av_log(ac->avccontext, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n"); |
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282 |
7501 | 283 num_front = get_bits(gb, 4); |
284 num_side = get_bits(gb, 4); | |
285 num_back = get_bits(gb, 4); | |
286 num_lfe = get_bits(gb, 2); | |
287 num_assoc_data = get_bits(gb, 3); | |
288 num_cc = get_bits(gb, 4); | |
289 | |
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290 if (get_bits1(gb)) |
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291 skip_bits(gb, 4); // mono_mixdown_tag |
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292 if (get_bits1(gb)) |
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293 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 294 |
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295 if (get_bits1(gb)) |
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296 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 297 |
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298 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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299 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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300 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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301 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 302 |
303 skip_bits_long(gb, 4 * num_assoc_data); | |
304 | |
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305 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 306 |
307 align_get_bits(gb); | |
308 | |
309 /* comment field, first byte is length */ | |
310 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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311 return 0; |
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312 } |
7501 | 313 |
7539 | 314 /** |
315 * Set up channel positions based on a default channel configuration | |
316 * as specified in table 1.17. | |
317 * | |
318 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
319 * | |
320 * @return Returns error status. 0 - OK, !0 - error | |
321 */ | |
10004 | 322 static int set_default_channel_config(AACContext *ac, |
323 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
324 int channel_config) | |
7539 | 325 { |
10004 | 326 if (channel_config < 1 || channel_config > 7) { |
7539 | 327 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", |
328 channel_config); | |
329 return -1; | |
330 } | |
331 | |
332 /* default channel configurations: | |
333 * | |
334 * 1ch : front center (mono) | |
335 * 2ch : L + R (stereo) | |
336 * 3ch : front center + L + R | |
337 * 4ch : front center + L + R + back center | |
338 * 5ch : front center + L + R + back stereo | |
339 * 6ch : front center + L + R + back stereo + LFE | |
340 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
341 */ | |
342 | |
10004 | 343 if (channel_config != 2) |
7539 | 344 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) |
10004 | 345 if (channel_config > 1) |
7539 | 346 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) |
10004 | 347 if (channel_config == 4) |
7539 | 348 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center |
10004 | 349 if (channel_config > 4) |
7539 | 350 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] |
10004 | 351 = AAC_CHANNEL_BACK; // back stereo |
352 if (channel_config > 5) | |
7539 | 353 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE |
10004 | 354 if (channel_config == 7) |
7539 | 355 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right |
356 | |
357 return 0; | |
358 } | |
359 | |
7578 | 360 /** |
361 * Decode GA "General Audio" specific configuration; reference: table 4.1. | |
362 * | |
363 * @return Returns error status. 0 - OK, !0 - error | |
364 */ | |
10004 | 365 static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb, |
366 int channel_config) | |
367 { | |
7578 | 368 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
369 int extension_flag, ret; | |
370 | |
10004 | 371 if (get_bits1(gb)) { // frameLengthFlag |
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372 av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); |
7539 | 373 return -1; |
374 } | |
375 | |
376 if (get_bits1(gb)) // dependsOnCoreCoder | |
377 skip_bits(gb, 14); // coreCoderDelay | |
378 extension_flag = get_bits1(gb); | |
379 | |
10004 | 380 if (ac->m4ac.object_type == AOT_AAC_SCALABLE || |
381 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
7539 | 382 skip_bits(gb, 3); // layerNr |
383 | |
384 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
385 if (channel_config == 0) { | |
386 skip_bits(gb, 4); // element_instance_tag | |
10004 | 387 if ((ret = decode_pce(ac, new_che_pos, gb))) |
7539 | 388 return ret; |
389 } else { | |
10004 | 390 if ((ret = set_default_channel_config(ac, new_che_pos, channel_config))) |
7539 | 391 return ret; |
392 } | |
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393 if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR))) |
7539 | 394 return ret; |
395 | |
396 if (extension_flag) { | |
397 switch (ac->m4ac.object_type) { | |
10004 | 398 case AOT_ER_BSAC: |
399 skip_bits(gb, 5); // numOfSubFrame | |
400 skip_bits(gb, 11); // layer_length | |
401 break; | |
402 case AOT_ER_AAC_LC: | |
403 case AOT_ER_AAC_LTP: | |
404 case AOT_ER_AAC_SCALABLE: | |
405 case AOT_ER_AAC_LD: | |
406 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
7539 | 407 * aacScalefactorDataResilienceFlag |
408 * aacSpectralDataResilienceFlag | |
409 */ | |
10004 | 410 break; |
7539 | 411 } |
412 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
413 } | |
414 return 0; | |
415 } | |
416 | |
417 /** | |
418 * Decode audio specific configuration; reference: table 1.13. | |
419 * | |
420 * @param data pointer to AVCodecContext extradata | |
421 * @param data_size size of AVCCodecContext extradata | |
422 * | |
423 * @return Returns error status. 0 - OK, !0 - error | |
424 */ | |
10004 | 425 static int decode_audio_specific_config(AACContext *ac, void *data, |
426 int data_size) | |
427 { | |
7539 | 428 GetBitContext gb; |
429 int i; | |
430 | |
431 init_get_bits(&gb, data, data_size * 8); | |
432 | |
10004 | 433 if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) |
7539 | 434 return -1; |
10004 | 435 if (ac->m4ac.sampling_index > 12) { |
7539 | 436 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
437 return -1; | |
438 } | |
439 | |
440 skip_bits_long(&gb, i); | |
441 | |
442 switch (ac->m4ac.object_type) { | |
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443 case AOT_AAC_MAIN: |
7539 | 444 case AOT_AAC_LC: |
445 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
446 return -1; | |
447 break; | |
448 default: | |
449 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
450 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
451 return -1; | |
452 } | |
453 return 0; | |
454 } | |
455 | |
7578 | 456 /** |
457 * linear congruential pseudorandom number generator | |
458 * | |
459 * @param previous_val pointer to the current state of the generator | |
460 * | |
461 * @return Returns a 32-bit pseudorandom integer | |
462 */ | |
10004 | 463 static av_always_inline int lcg_random(int previous_val) |
464 { | |
7578 | 465 return previous_val * 1664525 + 1013904223; |
466 } | |
467 | |
10004 | 468 static void reset_predict_state(PredictorState *ps) |
469 { | |
470 ps->r0 = 0.0f; | |
471 ps->r1 = 0.0f; | |
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472 ps->cor0 = 0.0f; |
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473 ps->cor1 = 0.0f; |
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474 ps->var0 = 1.0f; |
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475 ps->var1 = 1.0f; |
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476 } |
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477 |
10004 | 478 static void reset_all_predictors(PredictorState *ps) |
479 { | |
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480 int i; |
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481 for (i = 0; i < MAX_PREDICTORS; i++) |
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482 reset_predict_state(&ps[i]); |
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483 } |
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484 |
10004 | 485 static void reset_predictor_group(PredictorState *ps, int group_num) |
486 { | |
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487 int i; |
10004 | 488 for (i = group_num - 1; i < MAX_PREDICTORS; i += 30) |
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489 reset_predict_state(&ps[i]); |
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490 } |
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491 |
10004 | 492 static av_cold int aac_decode_init(AVCodecContext *avccontext) |
493 { | |
494 AACContext *ac = avccontext->priv_data; | |
7501 | 495 int i; |
496 | |
497 ac->avccontext = avccontext; | |
498 | |
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499 if (avccontext->extradata_size > 0) { |
10004 | 500 if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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501 return -1; |
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502 avccontext->sample_rate = ac->m4ac.sample_rate; |
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503 } else if (avccontext->channels > 0) { |
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504 ac->m4ac.sample_rate = avccontext->sample_rate; |
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505 } |
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506 |
10004 | 507 avccontext->sample_fmt = SAMPLE_FMT_S16; |
508 avccontext->frame_size = 1024; | |
7501 | 509 |
510 AAC_INIT_VLC_STATIC( 0, 144); | |
511 AAC_INIT_VLC_STATIC( 1, 114); | |
512 AAC_INIT_VLC_STATIC( 2, 188); | |
513 AAC_INIT_VLC_STATIC( 3, 180); | |
514 AAC_INIT_VLC_STATIC( 4, 172); | |
515 AAC_INIT_VLC_STATIC( 5, 140); | |
516 AAC_INIT_VLC_STATIC( 6, 168); | |
517 AAC_INIT_VLC_STATIC( 7, 114); | |
518 AAC_INIT_VLC_STATIC( 8, 262); | |
519 AAC_INIT_VLC_STATIC( 9, 248); | |
520 AAC_INIT_VLC_STATIC(10, 384); | |
521 | |
522 dsputil_init(&ac->dsp, avccontext); | |
523 | |
7539 | 524 ac->random_state = 0x1f2e3d4c; |
525 | |
7501 | 526 // -1024 - Compensate wrong IMDCT method. |
527 // 32768 - Required to scale values to the correct range for the bias method | |
528 // for float to int16 conversion. | |
529 | |
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530 if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { |
10004 | 531 ac->add_bias = 385.0f; |
532 ac->sf_scale = 1. / (-1024. * 32768.); | |
7501 | 533 ac->sf_offset = 0; |
534 } else { | |
10004 | 535 ac->add_bias = 0.0f; |
536 ac->sf_scale = 1. / -1024.; | |
7501 | 537 ac->sf_offset = 60; |
538 } | |
539 | |
8590 | 540 #if !CONFIG_HARDCODED_TABLES |
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541 for (i = 0; i < 428; i++) |
10004 | 542 ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.); |
7501 | 543 #endif /* CONFIG_HARDCODED_TABLES */ |
544 | |
8042 | 545 INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), |
10004 | 546 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), |
547 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
548 352); | |
7501 | 549 |
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550 ff_mdct_init(&ac->mdct, 11, 1, 1.0); |
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551 ff_mdct_init(&ac->mdct_small, 8, 1, 1.0); |
7581 | 552 // window initialization |
553 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
554 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
555 ff_sine_window_init(ff_sine_1024, 1024); | |
556 ff_sine_window_init(ff_sine_128, 128); | |
557 | |
7501 | 558 return 0; |
559 } | |
560 | |
7539 | 561 /** |
562 * Skip data_stream_element; reference: table 4.10. | |
563 */ | |
10004 | 564 static void skip_data_stream_element(GetBitContext *gb) |
565 { | |
7501 | 566 int byte_align = get_bits1(gb); |
567 int count = get_bits(gb, 8); | |
568 if (count == 255) | |
569 count += get_bits(gb, 8); | |
570 if (byte_align) | |
571 align_get_bits(gb); | |
572 skip_bits_long(gb, 8 * count); | |
573 } | |
574 | |
10004 | 575 static int decode_prediction(AACContext *ac, IndividualChannelStream *ics, |
576 GetBitContext *gb) | |
577 { | |
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578 int sfb; |
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579 if (get_bits1(gb)) { |
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580 ics->predictor_reset_group = get_bits(gb, 5); |
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581 if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { |
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582 av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); |
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583 return -1; |
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584 } |
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585 } |
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586 for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { |
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587 ics->prediction_used[sfb] = get_bits1(gb); |
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588 } |
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589 return 0; |
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590 } |
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591 |
7501 | 592 /** |
7539 | 593 * Decode Individual Channel Stream info; reference: table 4.6. |
594 * | |
595 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
596 */ | |
10004 | 597 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics, |
598 GetBitContext *gb, int common_window) | |
599 { | |
7539 | 600 if (get_bits1(gb)) { |
601 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); | |
602 memset(ics, 0, sizeof(IndividualChannelStream)); | |
603 return -1; | |
604 } | |
605 ics->window_sequence[1] = ics->window_sequence[0]; | |
606 ics->window_sequence[0] = get_bits(gb, 2); | |
10004 | 607 ics->use_kb_window[1] = ics->use_kb_window[0]; |
608 ics->use_kb_window[0] = get_bits1(gb); | |
609 ics->num_window_groups = 1; | |
610 ics->group_len[0] = 1; | |
7581 | 611 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
612 int i; | |
613 ics->max_sfb = get_bits(gb, 4); | |
614 for (i = 0; i < 7; i++) { | |
615 if (get_bits1(gb)) { | |
10004 | 616 ics->group_len[ics->num_window_groups - 1]++; |
7581 | 617 } else { |
618 ics->num_window_groups++; | |
10004 | 619 ics->group_len[ics->num_window_groups - 1] = 1; |
7581 | 620 } |
621 } | |
10004 | 622 ics->num_windows = 8; |
623 ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index]; | |
624 ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; | |
625 ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index]; | |
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626 ics->predictor_present = 0; |
7581 | 627 } else { |
10004 | 628 ics->max_sfb = get_bits(gb, 6); |
629 ics->num_windows = 1; | |
630 ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index]; | |
631 ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; | |
632 ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index]; | |
633 ics->predictor_present = get_bits1(gb); | |
8199
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634 ics->predictor_reset_group = 0; |
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635 if (ics->predictor_present) { |
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636 if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
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637 if (decode_prediction(ac, ics, gb)) { |
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638 memset(ics, 0, sizeof(IndividualChannelStream)); |
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639 return -1; |
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640 } |
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641 } else if (ac->m4ac.object_type == AOT_AAC_LC) { |
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642 av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); |
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643 memset(ics, 0, sizeof(IndividualChannelStream)); |
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644 return -1; |
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645 } else { |
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646 av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); |
8200 | 647 memset(ics, 0, sizeof(IndividualChannelStream)); |
648 return -1; | |
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649 } |
7578 | 650 } |
651 } | |
652 | |
10004 | 653 if (ics->max_sfb > ics->num_swb) { |
7578 | 654 av_log(ac->avccontext, AV_LOG_ERROR, |
10004 | 655 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", |
656 ics->max_sfb, ics->num_swb); | |
7578 | 657 memset(ics, 0, sizeof(IndividualChannelStream)); |
658 return -1; | |
659 } | |
660 | |
7539 | 661 return 0; |
662 } | |
663 | |
664 /** | |
665 * Decode band types (section_data payload); reference: table 4.46. | |
666 * | |
667 * @param band_type array of the used band type | |
668 * @param band_type_run_end array of the last scalefactor band of a band type run | |
669 * | |
670 * @return Returns error status. 0 - OK, !0 - error | |
671 */ | |
10004 | 672 static int decode_band_types(AACContext *ac, enum BandType band_type[120], |
673 int band_type_run_end[120], GetBitContext *gb, | |
674 IndividualChannelStream *ics) | |
675 { | |
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676 int g, idx = 0; |
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677 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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678 for (g = 0; g < ics->num_window_groups; g++) { |
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679 int k = 0; |
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680 while (k < ics->max_sfb) { |
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681 uint8_t sect_end = k; |
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682 int sect_len_incr; |
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683 int sect_band_type = get_bits(gb, 4); |
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684 if (sect_band_type == 12) { |
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685 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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686 return -1; |
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687 } |
10004 | 688 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1) |
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689 sect_end += sect_len_incr; |
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690 sect_end += sect_len_incr; |
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691 if (sect_end > ics->max_sfb) { |
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692 av_log(ac->avccontext, AV_LOG_ERROR, |
10004 | 693 "Number of bands (%d) exceeds limit (%d).\n", |
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694 sect_end, ics->max_sfb); |
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695 return -1; |
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696 } |
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697 for (; k < sect_end; k++) { |
7581 | 698 band_type [idx] = sect_band_type; |
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699 band_type_run_end[idx++] = sect_end; |
7581 | 700 } |
7539 | 701 } |
702 } | |
703 return 0; | |
704 } | |
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705 |
7539 | 706 /** |
707 * Decode scalefactors; reference: table 4.47. | |
7523
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708 * |
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709 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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710 * @param band_type array of the used band type |
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711 * @param band_type_run_end array of the last scalefactor band of a band type run |
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712 * @param sf array of scalefactors or intensity stereo positions |
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713 * |
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714 * @return Returns error status. 0 - OK, !0 - error |
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715 */ |
10004 | 716 static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb, |
717 unsigned int global_gain, | |
718 IndividualChannelStream *ics, | |
719 enum BandType band_type[120], | |
720 int band_type_run_end[120]) | |
721 { | |
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722 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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723 int g, i, idx = 0; |
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724 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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725 int noise_flag = 1; |
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726 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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727 for (g = 0; g < ics->num_window_groups; g++) { |
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728 for (i = 0; i < ics->max_sfb;) { |
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729 int run_end = band_type_run_end[idx]; |
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730 if (band_type[idx] == ZERO_BT) { |
10004 | 731 for (; i < run_end; i++, idx++) |
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732 sf[idx] = 0.; |
10004 | 733 } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
734 for (; i < run_end; i++, idx++) { | |
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735 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
10004 | 736 if (offset[2] > 255U) { |
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737 av_log(ac->avccontext, AV_LOG_ERROR, |
10004 | 738 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
7523
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739 return -1; |
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740 } |
10004 | 741 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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742 } |
10004 | 743 } else if (band_type[idx] == NOISE_BT) { |
744 for (; i < run_end; i++, idx++) { | |
745 if (noise_flag-- > 0) | |
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746 offset[1] += get_bits(gb, 9) - 256; |
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747 else |
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748 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
10004 | 749 if (offset[1] > 255U) { |
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750 av_log(ac->avccontext, AV_LOG_ERROR, |
10004 | 751 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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752 return -1; |
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753 } |
10004 | 754 sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100]; |
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755 } |
10004 | 756 } else { |
757 for (; i < run_end; i++, idx++) { | |
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758 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
10004 | 759 if (offset[0] > 255U) { |
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760 av_log(ac->avccontext, AV_LOG_ERROR, |
10004 | 761 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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762 return -1; |
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763 } |
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764 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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765 } |
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766 } |
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767 } |
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768 } |
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769 return 0; |
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770 } |
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771 |
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772 /** |
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773 * Decode pulse data; reference: table 4.7. |
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774 */ |
10004 | 775 static int decode_pulses(Pulse *pulse, GetBitContext *gb, |
776 const uint16_t *swb_offset, int num_swb) | |
777 { | |
7871
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778 int i, pulse_swb; |
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779 pulse->num_pulse = get_bits(gb, 2) + 1; |
7871
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780 pulse_swb = get_bits(gb, 6); |
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781 if (pulse_swb >= num_swb) |
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782 return -1; |
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783 pulse->pos[0] = swb_offset[pulse_swb]; |
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784 pulse->pos[0] += get_bits(gb, 5); |
7871
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785 if (pulse->pos[0] > 1023) |
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786 return -1; |
7580 | 787 pulse->amp[0] = get_bits(gb, 4); |
788 for (i = 1; i < pulse->num_pulse; i++) { | |
10004 | 789 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1]; |
7871
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790 if (pulse->pos[i] > 1023) |
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791 return -1; |
7580 | 792 pulse->amp[i] = get_bits(gb, 4); |
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793 } |
7871
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794 return 0; |
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795 } |
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796 |
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797 /** |
7609 | 798 * Decode Temporal Noise Shaping data; reference: table 4.48. |
799 * | |
800 * @return Returns error status. 0 - OK, !0 - error | |
801 */ | |
10004 | 802 static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns, |
803 GetBitContext *gb, const IndividualChannelStream *ics) | |
804 { | |
7609 | 805 int w, filt, i, coef_len, coef_res, coef_compress; |
806 const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; | |
807 const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; | |
808 for (w = 0; w < ics->num_windows; w++) { | |
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809 if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
7609 | 810 coef_res = get_bits1(gb); |
811 | |
7638 | 812 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
813 int tmp2_idx; | |
10004 | 814 tns->length[w][filt] = get_bits(gb, 6 - 2 * is8); |
7609 | 815 |
10004 | 816 if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) { |
7638 | 817 av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", |
818 tns->order[w][filt], tns_max_order); | |
819 tns->order[w][filt] = 0; | |
820 return -1; | |
821 } | |
7850
77030651ddd0
Only read some of TNS bitstream data in the case that the TNS filter order is
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|
822 if (tns->order[w][filt]) { |
7851 | 823 tns->direction[w][filt] = get_bits1(gb); |
824 coef_compress = get_bits1(gb); | |
825 coef_len = coef_res + 3 - coef_compress; | |
10004 | 826 tmp2_idx = 2 * coef_compress + coef_res; |
7638 | 827 |
7851 | 828 for (i = 0; i < tns->order[w][filt]; i++) |
829 tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; | |
7850
77030651ddd0
Only read some of TNS bitstream data in the case that the TNS filter order is
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diff
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|
830 } |
7609 | 831 } |
7637
1fd8e319e8b6
Work around GCC variable may be used uninitialised warning
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|
832 } |
7609 | 833 } |
834 return 0; | |
835 } | |
836 | |
837 /** | |
7539 | 838 * Decode Mid/Side data; reference: table 4.54. |
839 * | |
840 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
841 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
842 * [3] reserved for scalable AAC | |
843 */ | |
10004 | 844 static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb, |
845 int ms_present) | |
846 { | |
7578 | 847 int idx; |
848 if (ms_present == 1) { | |
849 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) | |
850 cpe->ms_mask[idx] = get_bits1(gb); | |
851 } else if (ms_present == 2) { | |
852 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); | |
853 } | |
854 } | |
7539 | 855 |
856 /** | |
7581 | 857 * Decode spectral data; reference: table 4.50. |
858 * Dequantize and scale spectral data; reference: 4.6.3.3. | |
859 * | |
860 * @param coef array of dequantized, scaled spectral data | |
861 * @param sf array of scalefactors or intensity stereo positions | |
862 * @param pulse_present set if pulses are present | |
863 * @param pulse pointer to pulse data struct | |
864 * @param band_type array of the used band type | |
865 * | |
866 * @return Returns error status. 0 - OK, !0 - error | |
867 */ | |
10004 | 868 static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024], |
869 GetBitContext *gb, float sf[120], | |
870 int pulse_present, const Pulse *pulse, | |
871 const IndividualChannelStream *ics, | |
872 enum BandType band_type[120]) | |
873 { | |
7581 | 874 int i, k, g, idx = 0; |
10004 | 875 const int c = 1024 / ics->num_windows; |
876 const uint16_t *offsets = ics->swb_offset; | |
7581 | 877 float *coef_base = coef; |
8284 | 878 static const float sign_lookup[] = { 1.0f, -1.0f }; |
7581 | 879 |
880 for (g = 0; g < ics->num_windows; g++) | |
10004 | 881 memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb])); |
7581 | 882 |
883 for (g = 0; g < ics->num_window_groups; g++) { | |
884 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
885 const int cur_band_type = band_type[idx]; | |
886 const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; | |
887 const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); | |
888 int group; | |
8413
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889 if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) { |
7581 | 890 for (group = 0; group < ics->group_len[g]; group++) { |
10004 | 891 memset(coef + group * 128 + offsets[i], 0, (offsets[i + 1] - offsets[i]) * sizeof(float)); |
7581 | 892 } |
10004 | 893 } else if (cur_band_type == NOISE_BT) { |
7581 | 894 for (group = 0; group < ics->group_len[g]; group++) { |
8109
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Fix perceptual noise shaping scaling including an oversight of the offset for
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|
895 float scale; |
10252 | 896 float band_energy; |
10227 | 897 float *cf = coef + group * 128 + offsets[i]; |
10220 | 898 int len = offsets[i+1] - offsets[i]; |
899 | |
10252 | 900 for (k = 0; k < len; k++) { |
7581 | 901 ac->random_state = lcg_random(ac->random_state); |
10252 | 902 cf[k] = ac->random_state; |
8109
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Fix perceptual noise shaping scaling including an oversight of the offset for
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|
903 } |
10220 | 904 |
10252 | 905 band_energy = ac->dsp.scalarproduct_float(cf, cf, len); |
8109
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Fix perceptual noise shaping scaling including an oversight of the offset for
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906 scale = sf[idx] / sqrtf(band_energy); |
10220 | 907 ac->dsp.vector_fmul_scalar(cf, cf, scale, len); |
7581 | 908 } |
10004 | 909 } else { |
7581 | 910 for (group = 0; group < ics->group_len[g]; group++) { |
10220 | 911 const float *vq[96]; |
912 const float **vqp = vq; | |
913 float *cf = coef + (group << 7) + offsets[i]; | |
914 int len = offsets[i + 1] - offsets[i]; | |
915 | |
10004 | 916 for (k = offsets[i]; k < offsets[i + 1]; k += dim) { |
7581 | 917 const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); |
918 const int coef_tmp_idx = (group << 7) + k; | |
919 const float *vq_ptr; | |
920 int j; | |
10004 | 921 if (index >= ff_aac_spectral_sizes[cur_band_type - 1]) { |
7581 | 922 av_log(ac->avccontext, AV_LOG_ERROR, |
10004 | 923 "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", |
924 cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); | |
7581 | 925 return -1; |
926 } | |
927 vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; | |
10220 | 928 *vqp++ = vq_ptr; |
7581 | 929 if (is_cb_unsigned) { |
10004 | 930 if (vq_ptr[0]) |
931 coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)]; | |
932 if (vq_ptr[1]) | |
933 coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)]; | |
8283
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AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
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|
934 if (dim == 4) { |
10004 | 935 if (vq_ptr[2]) |
936 coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)]; | |
937 if (vq_ptr[3]) | |
938 coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)]; | |
8283
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AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
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|
939 } |
9053 | 940 if (cur_band_type == ESC_BT) { |
941 for (j = 0; j < 2; j++) { | |
942 if (vq_ptr[j] == 64.0f) { | |
943 int n = 4; | |
944 /* The total length of escape_sequence must be < 22 bits according | |
945 to the specification (i.e. max is 11111111110xxxxxxxxxx). */ | |
946 while (get_bits1(gb) && n < 15) n++; | |
10004 | 947 if (n == 15) { |
9053 | 948 av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); |
949 return -1; | |
950 } | |
10004 | 951 n = (1 << n) + get_bits(gb, n); |
9053 | 952 coef[coef_tmp_idx + j] *= cbrtf(n) * n; |
10004 | 953 } else |
9053 | 954 coef[coef_tmp_idx + j] *= vq_ptr[j]; |
955 } | |
9052
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
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|
956 } |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
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|
957 } |
10220 | 958 } |
959 | |
960 if (is_cb_unsigned && cur_band_type != ESC_BT) { | |
961 ac->dsp.vector_fmul_sv_scalar[dim>>2]( | |
962 cf, cf, vq, sf[idx], len); | |
10222
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AAC: Simplify some logic from r19956. (The escape codebook is unsigned.)
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963 } else if (cur_band_type == ESC_BT) { |
10220 | 964 ac->dsp.vector_fmul_scalar(cf, cf, sf[idx], len); |
965 } else { /* !is_cb_unsigned */ | |
966 ac->dsp.sv_fmul_scalar[dim>>2](cf, vq, sf[idx], len); | |
7581 | 967 } |
968 } | |
969 } | |
970 } | |
10004 | 971 coef += ics->group_len[g] << 7; |
7581 | 972 } |
973 | |
974 if (pulse_present) { | |
7857
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Apply pulses to unscaled coefficients rather than scaled.
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975 idx = 0; |
10004 | 976 for (i = 0; i < pulse->num_pulse; i++) { |
977 float co = coef_base[ pulse->pos[i] ]; | |
978 while (offsets[idx + 1] <= pulse->pos[i]) | |
7857
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
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979 idx++; |
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
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980 if (band_type[idx] != NOISE_BT && sf[idx]) { |
7858 | 981 float ico = -pulse->amp[i]; |
982 if (co) { | |
983 co /= sf[idx]; | |
984 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); | |
985 } | |
986 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; | |
7857
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Apply pulses to unscaled coefficients rather than scaled.
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|
987 } |
7581 | 988 } |
989 } | |
990 return 0; | |
991 } | |
992 | |
10004 | 993 static av_always_inline float flt16_round(float pf) |
994 { | |
9179
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AAC: IEEE-754 type punning for 16-bit floating point rounding.
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995 union float754 tmp; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
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996 tmp.f = pf; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
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997 tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; |
4172c9ef331a
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998 return tmp.f; |
8199
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|
999 } |
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1000 |
10004 | 1001 static av_always_inline float flt16_even(float pf) |
1002 { | |
9179
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1003 union float754 tmp; |
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1004 tmp.f = pf; |
10004 | 1005 tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U; |
9179
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1006 return tmp.f; |
8199
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1007 } |
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1008 |
10004 | 1009 static av_always_inline float flt16_trunc(float pf) |
1010 { | |
9179
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AAC: IEEE-754 type punning for 16-bit floating point rounding.
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|
1011 union float754 pun; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
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|
1012 pun.f = pf; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
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|
1013 pun.i &= 0xFFFF0000U; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
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1014 return pun.f; |
8199
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|
1015 } |
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1016 |
10004 | 1017 static void predict(AACContext *ac, PredictorState *ps, float *coef, |
1018 int output_enable) | |
1019 { | |
1020 const float a = 0.953125; // 61.0 / 64 | |
1021 const float alpha = 0.90625; // 29.0 / 32 | |
8199
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1022 float e0, e1; |
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AAC: Frequency domain prediction and hence Main profile support
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1023 float pv; |
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1024 float k1, k2; |
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1025 |
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1026 k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0; |
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1027 k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0; |
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1028 |
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1029 pv = flt16_round(k1 * ps->r0 + k2 * ps->r1); |
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1030 if (output_enable) |
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1031 *coef += pv * ac->sf_scale; |
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1032 |
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1033 e0 = *coef / ac->sf_scale; |
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1034 e1 = e0 - k1 * ps->r0; |
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1035 |
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1036 ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1); |
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1037 ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1)); |
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1038 ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0); |
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1039 ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0)); |
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1040 |
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1041 ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0)); |
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1042 ps->r0 = flt16_trunc(a * e0); |
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1043 } |
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1044 |
8a8d40fe77d9
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1045 /** |
8a8d40fe77d9
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1046 * Apply AAC-Main style frequency domain prediction. |
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1047 */ |
10004 | 1048 static void apply_prediction(AACContext *ac, SingleChannelElement *sce) |
1049 { | |
8199
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1050 int sfb, k; |
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1051 |
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1052 if (!sce->ics.predictor_initialized) { |
8413
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1053 reset_all_predictors(sce->predictor_state); |
8199
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1054 sce->ics.predictor_initialized = 1; |
8a8d40fe77d9
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|
1055 } |
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|
1056 |
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1057 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
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AAC: Frequency domain prediction and hence Main profile support
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1058 for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { |
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1059 for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { |
8413
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|
1060 predict(ac, &sce->predictor_state[k], &sce->coeffs[k], |
10004 | 1061 sce->ics.predictor_present && sce->ics.prediction_used[sfb]); |
8199
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|
1062 } |
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AAC: Frequency domain prediction and hence Main profile support
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|
1063 } |
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|
1064 if (sce->ics.predictor_reset_group) |
8413
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|
1065 reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group); |
8199
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1066 } else |
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1067 reset_all_predictors(sce->predictor_state); |
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1068 } |
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1069 |
7581 | 1070 /** |
7539 | 1071 * Decode an individual_channel_stream payload; reference: table 4.44. |
1072 * | |
1073 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
1074 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
1075 * | |
1076 * @return Returns error status. 0 - OK, !0 - error | |
1077 */ | |
10004 | 1078 static int decode_ics(AACContext *ac, SingleChannelElement *sce, |
1079 GetBitContext *gb, int common_window, int scale_flag) | |
1080 { | |
7539 | 1081 Pulse pulse; |
10004 | 1082 TemporalNoiseShaping *tns = &sce->tns; |
1083 IndividualChannelStream *ics = &sce->ics; | |
1084 float *out = sce->coeffs; | |
7539 | 1085 int global_gain, pulse_present = 0; |
1086 | |
7580 | 1087 /* This assignment is to silence a GCC warning about the variable being used |
1088 * uninitialized when in fact it always is. | |
7539 | 1089 */ |
1090 pulse.num_pulse = 0; | |
1091 | |
1092 global_gain = get_bits(gb, 8); | |
1093 | |
1094 if (!common_window && !scale_flag) { | |
1095 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
1096 return -1; | |
1097 } | |
1098 | |
1099 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
1100 return -1; | |
1101 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
1102 return -1; | |
1103 | |
1104 pulse_present = 0; | |
1105 if (!scale_flag) { | |
1106 if ((pulse_present = get_bits1(gb))) { | |
1107 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
1108 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
1109 return -1; | |
1110 } | |
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1111 if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { |
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1112 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); |
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1113 return -1; |
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1114 } |
7539 | 1115 } |
1116 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
1117 return -1; | |
1118 if (get_bits1(gb)) { | |
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1119 av_log_missing_feature(ac->avccontext, "SSR", 1); |
7539 | 1120 return -1; |
1121 } | |
1122 } | |
1123 | |
7580 | 1124 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
7539 | 1125 return -1; |
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1126 |
10004 | 1127 if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window) |
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1128 apply_prediction(ac, sce); |
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1129 |
7539 | 1130 return 0; |
1131 } | |
1132 | |
1133 /** | |
7581 | 1134 * Mid/Side stereo decoding; reference: 4.6.8.1.3. |
1135 */ | |
10220 | 1136 static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe) |
10004 | 1137 { |
1138 const IndividualChannelStream *ics = &cpe->ch[0].ics; | |
7581 | 1139 float *ch0 = cpe->ch[0].coeffs; |
1140 float *ch1 = cpe->ch[1].coeffs; | |
10220 | 1141 int g, i, group, idx = 0; |
10004 | 1142 const uint16_t *offsets = ics->swb_offset; |
7581 | 1143 for (g = 0; g < ics->num_window_groups; g++) { |
1144 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
1145 if (cpe->ms_mask[idx] && | |
10004 | 1146 cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { |
7581 | 1147 for (group = 0; group < ics->group_len[g]; group++) { |
10220 | 1148 ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i], |
1149 ch1 + group * 128 + offsets[i], | |
1150 offsets[i+1] - offsets[i]); | |
7581 | 1151 } |
1152 } | |
1153 } | |
10004 | 1154 ch0 += ics->group_len[g] * 128; |
1155 ch1 += ics->group_len[g] * 128; | |
7581 | 1156 } |
1157 } | |
1158 | |
1159 /** | |
1160 * intensity stereo decoding; reference: 4.6.8.2.3 | |
1161 * | |
1162 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
1163 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
1164 * [3] reserved for scalable AAC | |
1165 */ | |
10004 | 1166 static void apply_intensity_stereo(ChannelElement *cpe, int ms_present) |
1167 { | |
1168 const IndividualChannelStream *ics = &cpe->ch[1].ics; | |
1169 SingleChannelElement *sce1 = &cpe->ch[1]; | |
7581 | 1170 float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; |
10004 | 1171 const uint16_t *offsets = ics->swb_offset; |
7581 | 1172 int g, group, i, k, idx = 0; |
1173 int c; | |
1174 float scale; | |
1175 for (g = 0; g < ics->num_window_groups; g++) { | |
1176 for (i = 0; i < ics->max_sfb;) { | |
1177 if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { | |
1178 const int bt_run_end = sce1->band_type_run_end[idx]; | |
1179 for (; i < bt_run_end; i++, idx++) { | |
1180 c = -1 + 2 * (sce1->band_type[idx] - 14); | |
1181 if (ms_present) | |
1182 c *= 1 - 2 * cpe->ms_mask[idx]; | |
1183 scale = c * sce1->sf[idx]; | |
1184 for (group = 0; group < ics->group_len[g]; group++) | |
10004 | 1185 for (k = offsets[i]; k < offsets[i + 1]; k++) |
1186 coef1[group * 128 + k] = scale * coef0[group * 128 + k]; | |
7581 | 1187 } |
1188 } else { | |
1189 int bt_run_end = sce1->band_type_run_end[idx]; | |
1190 idx += bt_run_end - i; | |
1191 i = bt_run_end; | |
1192 } | |
1193 } | |
10004 | 1194 coef0 += ics->group_len[g] * 128; |
1195 coef1 += ics->group_len[g] * 128; | |
7581 | 1196 } |
1197 } | |
1198 | |
1199 /** | |
7539 | 1200 * Decode a channel_pair_element; reference: table 4.4. |
1201 * | |
1202 * @param elem_id Identifies the instance of a syntax element. | |
1203 * | |
1204 * @return Returns error status. 0 - OK, !0 - error | |
1205 */ | |
10004 | 1206 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe) |
1207 { | |
7539 | 1208 int i, ret, common_window, ms_present = 0; |
1209 | |
1210 common_window = get_bits1(gb); | |
1211 if (common_window) { | |
1212 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
1213 return -1; | |
1214 i = cpe->ch[1].ics.use_kb_window[0]; | |
1215 cpe->ch[1].ics = cpe->ch[0].ics; | |
1216 cpe->ch[1].ics.use_kb_window[1] = i; | |
1217 ms_present = get_bits(gb, 2); | |
10004 | 1218 if (ms_present == 3) { |
7539 | 1219 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); |
1220 return -1; | |
10004 | 1221 } else if (ms_present) |
7539 | 1222 decode_mid_side_stereo(cpe, gb, ms_present); |
1223 } | |
1224 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
1225 return ret; | |
1226 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
1227 return ret; | |
1228 | |
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1229 if (common_window) { |
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1230 if (ms_present) |
10220 | 1231 apply_mid_side_stereo(ac, cpe); |
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1232 if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
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1233 apply_prediction(ac, &cpe->ch[0]); |
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1234 apply_prediction(ac, &cpe->ch[1]); |
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1235 } |
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1236 } |
7539 | 1237 |
7580 | 1238 apply_intensity_stereo(cpe, ms_present); |
7539 | 1239 return 0; |
1240 } | |
1241 | |
7581 | 1242 /** |
1243 * Decode coupling_channel_element; reference: table 4.8. | |
1244 * | |
1245 * @param elem_id Identifies the instance of a syntax element. | |
1246 * | |
1247 * @return Returns error status. 0 - OK, !0 - error | |
1248 */ | |
10004 | 1249 static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che) |
1250 { | |
7581 | 1251 int num_gain = 0; |
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1252 int c, g, sfb, ret; |
7581 | 1253 int sign; |
1254 float scale; | |
10004 | 1255 SingleChannelElement *sce = &che->ch[0]; |
1256 ChannelCoupling *coup = &che->coup; | |
7581 | 1257 |
10004 | 1258 coup->coupling_point = 2 * get_bits1(gb); |
7578 | 1259 coup->num_coupled = get_bits(gb, 3); |
1260 for (c = 0; c <= coup->num_coupled; c++) { | |
1261 num_gain++; | |
1262 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; | |
1263 coup->id_select[c] = get_bits(gb, 4); | |
1264 if (coup->type[c] == TYPE_CPE) { | |
1265 coup->ch_select[c] = get_bits(gb, 2); | |
1266 if (coup->ch_select[c] == 3) | |
1267 num_gain++; | |
1268 } else | |
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1269 coup->ch_select[c] = 2; |
7578 | 1270 } |
10004 | 1271 coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1); |
7578 | 1272 |
10004 | 1273 sign = get_bits(gb, 1); |
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1274 scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); |
7578 | 1275 |
1276 if ((ret = decode_ics(ac, sce, gb, 0, 0))) | |
1277 return ret; | |
1278 | |
1279 for (c = 0; c < num_gain; c++) { | |
10004 | 1280 int idx = 0; |
1281 int cge = 1; | |
7578 | 1282 int gain = 0; |
1283 float gain_cache = 1.; | |
1284 if (c) { | |
1285 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); | |
1286 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; | |
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1287 gain_cache = pow(scale, -gain); |
7578 | 1288 } |
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1289 if (coup->coupling_point == AFTER_IMDCT) { |
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1290 coup->gain[c][0] = gain_cache; |
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1291 } else { |
8823 | 1292 for (g = 0; g < sce->ics.num_window_groups; g++) { |
1293 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { | |
1294 if (sce->band_type[idx] != ZERO_BT) { | |
1295 if (!cge) { | |
1296 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; | |
10004 | 1297 if (t) { |
8823 | 1298 int s = 1; |
1299 t = gain += t; | |
1300 if (sign) { | |
1301 s -= 2 * (t & 0x1); | |
1302 t >>= 1; | |
1303 } | |
1304 gain_cache = pow(scale, -t) * s; | |
7578 | 1305 } |
1306 } | |
8823 | 1307 coup->gain[c][idx] = gain_cache; |
7578 | 1308 } |
1309 } | |
7890 | 1310 } |
1311 } | |
7578 | 1312 } |
1313 return 0; | |
1314 } | |
1315 | |
7539 | 1316 /** |
1317 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
7523
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1318 * |
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1319 * @param crc flag indicating the presence of CRC checksum |
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1320 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 1321 * |
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1322 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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1323 */ |
10004 | 1324 static int decode_sbr_extension(AACContext *ac, GetBitContext *gb, |
1325 int crc, int cnt) | |
1326 { | |
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1327 // TODO : sbr_extension implementation |
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1328 av_log_missing_feature(ac->avccontext, "SBR", 0); |
10004 | 1329 skip_bits_long(gb, 8 * cnt - 4); // -4 due to reading extension type |
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1330 return cnt; |
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1331 } |
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1332 |
7539 | 1333 /** |
7578 | 1334 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. |
1335 * | |
1336 * @return Returns number of bytes consumed. | |
1337 */ | |
10004 | 1338 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, |
1339 GetBitContext *gb) | |
1340 { | |
7578 | 1341 int i; |
1342 int num_excl_chan = 0; | |
1343 | |
1344 do { | |
1345 for (i = 0; i < 7; i++) | |
1346 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); | |
1347 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); | |
1348 | |
1349 return num_excl_chan / 7; | |
1350 } | |
1351 | |
1352 /** | |
7539 | 1353 * Decode dynamic range information; reference: table 4.52. |
1354 * | |
1355 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1356 * | |
1357 * @return Returns number of bytes consumed. | |
1358 */ | |
10004 | 1359 static int decode_dynamic_range(DynamicRangeControl *che_drc, |
1360 GetBitContext *gb, int cnt) | |
1361 { | |
1362 int n = 1; | |
7539 | 1363 int drc_num_bands = 1; |
1364 int i; | |
1365 | |
1366 /* pce_tag_present? */ | |
10004 | 1367 if (get_bits1(gb)) { |
7539 | 1368 che_drc->pce_instance_tag = get_bits(gb, 4); |
1369 skip_bits(gb, 4); // tag_reserved_bits | |
1370 n++; | |
1371 } | |
1372 | |
1373 /* excluded_chns_present? */ | |
10004 | 1374 if (get_bits1(gb)) { |
7539 | 1375 n += decode_drc_channel_exclusions(che_drc, gb); |
1376 } | |
1377 | |
1378 /* drc_bands_present? */ | |
1379 if (get_bits1(gb)) { | |
1380 che_drc->band_incr = get_bits(gb, 4); | |
1381 che_drc->interpolation_scheme = get_bits(gb, 4); | |
1382 n++; | |
1383 drc_num_bands += che_drc->band_incr; | |
1384 for (i = 0; i < drc_num_bands; i++) { | |
1385 che_drc->band_top[i] = get_bits(gb, 8); | |
1386 n++; | |
1387 } | |
1388 } | |
1389 | |
1390 /* prog_ref_level_present? */ | |
1391 if (get_bits1(gb)) { | |
1392 che_drc->prog_ref_level = get_bits(gb, 7); | |
1393 skip_bits1(gb); // prog_ref_level_reserved_bits | |
1394 n++; | |
1395 } | |
1396 | |
1397 for (i = 0; i < drc_num_bands; i++) { | |
1398 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
1399 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
1400 n++; | |
1401 } | |
1402 | |
1403 return n; | |
1404 } | |
1405 | |
1406 /** | |
1407 * Decode extension data (incomplete); reference: table 4.51. | |
1408 * | |
1409 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1410 * | |
1411 * @return Returns number of bytes consumed | |
1412 */ | |
10004 | 1413 static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt) |
1414 { | |
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1415 int crc_flag = 0; |
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1416 int res = cnt; |
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1417 switch (get_bits(gb, 4)) { // extension type |
10004 | 1418 case EXT_SBR_DATA_CRC: |
1419 crc_flag++; | |
1420 case EXT_SBR_DATA: | |
1421 res = decode_sbr_extension(ac, gb, crc_flag, cnt); | |
1422 break; | |
1423 case EXT_DYNAMIC_RANGE: | |
1424 res = decode_dynamic_range(&ac->che_drc, gb, cnt); | |
1425 break; | |
1426 case EXT_FILL: | |
1427 case EXT_FILL_DATA: | |
1428 case EXT_DATA_ELEMENT: | |
1429 default: | |
1430 skip_bits_long(gb, 8 * cnt - 4); | |
1431 break; | |
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1432 }; |
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1433 return res; |
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1434 } |
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1435 |
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1436 /** |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1437 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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diff
changeset
|
1438 * |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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diff
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|
1439 * @param decode 1 if tool is used normally, 0 if tool is used in LTP. |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1440 * @param coef spectral coefficients |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1441 */ |
10004 | 1442 static void apply_tns(float coef[1024], TemporalNoiseShaping *tns, |
1443 IndividualChannelStream *ics, int decode) | |
1444 { | |
1445 const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); | |
7636 | 1446 int w, filt, m, i; |
7632
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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diff
changeset
|
1447 int bottom, top, order, start, end, size, inc; |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1448 float lpc[TNS_MAX_ORDER]; |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1449 |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1450 for (w = 0; w < ics->num_windows; w++) { |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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diff
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|
1451 bottom = ics->num_swb; |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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changeset
|
1452 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1453 top = bottom; |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1454 bottom = FFMAX(0, top - tns->length[w][filt]); |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1455 order = tns->order[w][filt]; |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1456 if (order == 0) |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1457 continue; |
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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|
1458 |
7788
ffd4b1364b62
Avoid duplicating compute_lpc_coefs() function in both the RA288 and AAC decoders.
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|
1459 // tns_decode_coef |
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Avoid duplicating compute_lpc_coefs() function in both the RA288 and AAC decoders.
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|
1460 compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); |
7632
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Last hunk of the AAC decoder code to be OKed and build system and documentation
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1461 |
7609 | 1462 start = ics->swb_offset[FFMIN(bottom, mmm)]; |
1463 end = ics->swb_offset[FFMIN( top, mmm)]; | |
1464 if ((size = end - start) <= 0) | |
1465 continue; | |
1466 if (tns->direction[w][filt]) { | |
10004 | 1467 inc = -1; |
1468 start = end - 1; | |
7609 | 1469 } else { |
1470 inc = 1; | |
1471 } | |
1472 start += w * 128; | |
1473 | |
1474 // ar filter | |
1475 for (m = 0; m < size; m++, start += inc) | |
1476 for (i = 1; i <= FFMIN(m, order); i++) | |
10004 | 1477 coef[start] -= coef[start - i * inc] * lpc[i - 1]; |
7609 | 1478 } |
1479 } | |
1480 } | |
1481 | |
7523
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1482 /** |
7539 | 1483 * Conduct IMDCT and windowing. |
1484 */ | |
10004 | 1485 static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce) |
1486 { | |
1487 IndividualChannelStream *ics = &sce->ics; | |
1488 float *in = sce->coeffs; | |
1489 float *out = sce->ret; | |
1490 float *saved = sce->saved; | |
1491 const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; | |
1492 const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
1493 const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
1494 float *buf = ac->buf_mdct; | |
1495 float *temp = ac->temp; | |
7539 | 1496 int i; |
1497 | |
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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1498 // imdct |
7578 | 1499 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1500 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) | |
1501 av_log(ac->avccontext, AV_LOG_WARNING, | |
1502 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " | |
1503 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); | |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1504 for (i = 0; i < 1024; i += 128) |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1505 ff_imdct_half(&ac->mdct_small, buf + i, in + i); |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1506 } else |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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changeset
|
1507 ff_imdct_half(&ac->mdct, buf, in); |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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changeset
|
1508 |
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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changeset
|
1509 /* window overlapping |
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1510 * NOTE: To simplify the overlapping code, all 'meaningless' short to long |
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1511 * and long to short transitions are considered to be short to short |
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1512 * transitions. This leaves just two cases (long to long and short to short) |
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1513 * with a little special sauce for EIGHT_SHORT_SEQUENCE. |
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1514 */ |
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|
1515 if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
10004 | 1516 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1517 ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512); |
7661
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1518 } else { |
7662 | 1519 for (i = 0; i < 448; i++) |
1520 out[i] = saved[i] + ac->add_bias; | |
7578 | 1521 |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1522 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1523 ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1524 ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, ac->add_bias, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1525 ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, ac->add_bias, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1526 ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, ac->add_bias, 64); |
66fb1f3937fa
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|
1527 ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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changeset
|
1528 memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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changeset
|
1529 } else { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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changeset
|
1530 ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64); |
7662 | 1531 for (i = 576; i < 1024; i++) |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1532 out[i] = buf[i-512] + ac->add_bias; |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1533 } |
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|
1534 } |
7578 | 1535 |
7661
258eb30e6f7d
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|
1536 // buffer update |
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|
1537 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1538 for (i = 0; i < 64; i++) |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1539 saved[i] = temp[64 + i] - ac->add_bias; |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1540 ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1541 ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1542 ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1543 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1544 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1545 memcpy( saved, buf + 512, 448 * sizeof(float)); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1546 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1547 } else { // LONG_STOP or ONLY_LONG |
7663
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|
1548 memcpy( saved, buf + 512, 512 * sizeof(float)); |
7578 | 1549 } |
1550 } | |
1551 | |
7539 | 1552 /** |
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1553 * Apply dependent channel coupling (applied before IMDCT). |
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|
1554 * |
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|
1555 * @param index index into coupling gain array |
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|
1556 */ |
10004 | 1557 static void apply_dependent_coupling(AACContext *ac, |
1558 SingleChannelElement *target, | |
1559 ChannelElement *cce, int index) | |
1560 { | |
1561 IndividualChannelStream *ics = &cce->ch[0].ics; | |
1562 const uint16_t *offsets = ics->swb_offset; | |
1563 float *dest = target->coeffs; | |
1564 const float *src = cce->ch[0].coeffs; | |
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|
1565 int g, i, group, k, idx = 0; |
10004 | 1566 if (ac->m4ac.object_type == AOT_AAC_LTP) { |
7523
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|
1567 av_log(ac->avccontext, AV_LOG_ERROR, |
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|
1568 "Dependent coupling is not supported together with LTP\n"); |
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|
1569 return; |
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|
1570 } |
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|
1571 for (g = 0; g < ics->num_window_groups; g++) { |
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|
1572 for (i = 0; i < ics->max_sfb; i++, idx++) { |
8006
c7c1e85d14bc
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|
1573 if (cce->ch[0].band_type[idx] != ZERO_BT) { |
9143 | 1574 const float gain = cce->coup.gain[index][idx]; |
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|
1575 for (group = 0; group < ics->group_len[g]; group++) { |
10004 | 1576 for (k = offsets[i]; k < offsets[i + 1]; k++) { |
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|
1577 // XXX dsputil-ize |
10004 | 1578 dest[group * 128 + k] += gain * src[group * 128 + k]; |
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|
1579 } |
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1580 } |
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1581 } |
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1582 } |
10004 | 1583 dest += ics->group_len[g] * 128; |
1584 src += ics->group_len[g] * 128; | |
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1585 } |
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1586 } |
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1587 |
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1588 /** |
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1589 * Apply independent channel coupling (applied after IMDCT). |
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1590 * |
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1591 * @param index index into coupling gain array |
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1592 */ |
10004 | 1593 static void apply_independent_coupling(AACContext *ac, |
1594 SingleChannelElement *target, | |
1595 ChannelElement *cce, int index) | |
1596 { | |
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|
1597 int i; |
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1598 const float gain = cce->coup.gain[index][0]; |
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1599 const float bias = ac->add_bias; |
10004 | 1600 const float *src = cce->ch[0].ret; |
1601 float *dest = target->ret; | |
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1602 |
7523
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Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
parents:
7501
diff
changeset
|
1603 for (i = 0; i < 1024; i++) |
9051
0cffed11ee33
Use const to better allow the compiler to optimize AAC independent coupling.
alexc
parents:
8823
diff
changeset
|
1604 dest[i] += gain * (src[i] - bias); |
7523
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
parents:
7501
diff
changeset
|
1605 } |
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
parents:
7501
diff
changeset
|
1606 |
7581 | 1607 /** |
1608 * channel coupling transformation interface | |
1609 * | |
1610 * @param index index into coupling gain array | |
1611 * @param apply_coupling_method pointer to (in)dependent coupling function | |
1612 */ | |
10004 | 1613 static void apply_channel_coupling(AACContext *ac, ChannelElement *cc, |
1614 enum RawDataBlockType type, int elem_id, | |
1615 enum CouplingPoint coupling_point, | |
1616 void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index)) | |
7581 | 1617 { |
8004
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1618 int i, c; |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1619 |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1620 for (i = 0; i < MAX_ELEM_ID; i++) { |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1621 ChannelElement *cce = ac->che[TYPE_CCE][i]; |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1622 int index = 0; |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1623 |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1624 if (cce && cce->coup.coupling_point == coupling_point) { |
10004 | 1625 ChannelCoupling *coup = &cce->coup; |
8004
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1626 |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1627 for (c = 0; c <= coup->num_coupled; c++) { |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1628 if (coup->type[c] == type && coup->id_select[c] == elem_id) { |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1629 if (coup->ch_select[c] != 1) { |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1630 apply_coupling_method(ac, &cc->ch[0], cce, index); |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1631 if (coup->ch_select[c] != 0) |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1632 index++; |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1633 } |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1634 if (coup->ch_select[c] != 2) |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1635 apply_coupling_method(ac, &cc->ch[1], cce, index++); |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1636 } else |
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1637 index += 1 + (coup->ch_select[c] == 3); |
7581 | 1638 } |
1639 } | |
1640 } | |
1641 } | |
1642 | |
1643 /** | |
1644 * Convert spectral data to float samples, applying all supported tools as appropriate. | |
1645 */ | |
10004 | 1646 static void spectral_to_sample(AACContext *ac) |
1647 { | |
10688 | 1648 int i, type; |
1649 for (type = 3; type >= 0; type--) { | |
8004
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1650 for (i = 0; i < MAX_ELEM_ID; i++) { |
7581 | 1651 ChannelElement *che = ac->che[type][i]; |
10004 | 1652 if (che) { |
1653 if (type <= TYPE_CPE) | |
8004
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1654 apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); |
10004 | 1655 if (che->ch[0].tns.present) |
7581 | 1656 apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); |
10004 | 1657 if (che->ch[1].tns.present) |
7581 | 1658 apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); |
10004 | 1659 if (type <= TYPE_CPE) |
8004
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1660 apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); |
10004 | 1661 if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) |
8004
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1662 imdct_and_windowing(ac, &che->ch[0]); |
10004 | 1663 if (type == TYPE_CPE) |
7581 | 1664 imdct_and_windowing(ac, &che->ch[1]); |
10004 | 1665 if (type <= TYPE_CCE) |
8004
8fd8f23be794
Corrections to channel coupling code to attain conformance for appropriate
superdump
parents:
7891
diff
changeset
|
1666 apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); |
7578 | 1667 } |
1668 } | |
1669 } | |
1670 } | |
1671 | |
10004 | 1672 static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb) |
1673 { | |
8549
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
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parents:
8502
diff
changeset
|
1674 int size; |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1675 AACADTSHeaderInfo hdr_info; |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1676 |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1677 size = ff_aac_parse_header(gb, &hdr_info); |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1678 if (size > 0) { |
10491
74e51b86b934
Don't lock the channel output configuration based on the first value seen for
alexc
parents:
10378
diff
changeset
|
1679 if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) { |
9948
98fd723f72e7
Ignore subsequent channel configurations after the first in an AAC file. The
alexc
parents:
9891
diff
changeset
|
1680 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
98fd723f72e7
Ignore subsequent channel configurations after the first in an AAC file. The
alexc
parents:
9891
diff
changeset
|
1681 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
8549
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1682 ac->m4ac.chan_config = hdr_info.chan_config; |
9948
98fd723f72e7
Ignore subsequent channel configurations after the first in an AAC file. The
alexc
parents:
9891
diff
changeset
|
1683 if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config)) |
98fd723f72e7
Ignore subsequent channel configurations after the first in an AAC file. The
alexc
parents:
9891
diff
changeset
|
1684 return -7; |
10491
74e51b86b934
Don't lock the channel output configuration based on the first value seen for
alexc
parents:
10378
diff
changeset
|
1685 if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME)) |
9948
98fd723f72e7
Ignore subsequent channel configurations after the first in an AAC file. The
alexc
parents:
9891
diff
changeset
|
1686 return -7; |
10491
74e51b86b934
Don't lock the channel output configuration based on the first value seen for
alexc
parents:
10378
diff
changeset
|
1687 } else if (ac->output_configured != OC_LOCKED) { |
74e51b86b934
Don't lock the channel output configuration based on the first value seen for
alexc
parents:
10378
diff
changeset
|
1688 ac->output_configured = OC_NONE; |
9948
98fd723f72e7
Ignore subsequent channel configurations after the first in an AAC file. The
alexc
parents:
9891
diff
changeset
|
1689 } |
10621
62b3faabb663
Set the SBR mode to implicit on ADTS frames occurring before the output is locked.
alexc
parents:
10551
diff
changeset
|
1690 if (ac->output_configured != OC_LOCKED) |
62b3faabb663
Set the SBR mode to implicit on ADTS frames occurring before the output is locked.
alexc
parents:
10551
diff
changeset
|
1691 ac->m4ac.sbr = -1; |
8549
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1692 ac->m4ac.sample_rate = hdr_info.sample_rate; |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1693 ac->m4ac.sampling_index = hdr_info.sampling_index; |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1694 ac->m4ac.object_type = hdr_info.object_type; |
10657
168ea860d58f
aac: Set sample rate in avctx from ADTS header (if not already set by some
andoma
parents:
10622
diff
changeset
|
1695 if (!ac->avccontext->sample_rate) |
168ea860d58f
aac: Set sample rate in avctx from ADTS header (if not already set by some
andoma
parents:
10622
diff
changeset
|
1696 ac->avccontext->sample_rate = hdr_info.sample_rate; |
9151 | 1697 if (hdr_info.num_aac_frames == 1) { |
1698 if (!hdr_info.crc_absent) | |
1699 skip_bits(gb, 16); | |
1700 } else { | |
9891
7ad7d4094d1f
Rename ff_log_missing_feature() to av_log_missing_feature().
rbultje
parents:
9847
diff
changeset
|
1701 av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0); |
9151 | 1702 return -1; |
1703 } | |
9150
4fe6dee5f1dd
If we get an error from ff_aac_parse_header() we should not trust the header info that it provides.
alexc
parents:
9149
diff
changeset
|
1704 } |
8549
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1705 return size; |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1706 } |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
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parents:
8502
diff
changeset
|
1707 |
10004 | 1708 static int aac_decode_frame(AVCodecContext *avccontext, void *data, |
1709 int *data_size, AVPacket *avpkt) | |
1710 { | |
9355
54bc8a2727b0
Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
rbultje
parents:
9251
diff
changeset
|
1711 const uint8_t *buf = avpkt->data; |
54bc8a2727b0
Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
rbultje
parents:
9251
diff
changeset
|
1712 int buf_size = avpkt->size; |
10004 | 1713 AACContext *ac = avccontext->priv_data; |
1714 ChannelElement *che = NULL; | |
7578 | 1715 GetBitContext gb; |
1716 enum RawDataBlockType elem_type; | |
1717 int err, elem_id, data_size_tmp; | |
1718 | |
10004 | 1719 init_get_bits(&gb, buf, buf_size * 8); |
7578 | 1720 |
8549
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1721 if (show_bits(&gb, 12) == 0xfff) { |
9510 | 1722 if (parse_adts_frame_header(ac, &gb) < 0) { |
8549
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1723 av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1724 return -1; |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
superdump
parents:
8502
diff
changeset
|
1725 } |
8812 | 1726 if (ac->m4ac.sampling_index > 12) { |
8789 | 1727 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
1728 return -1; | |
1729 } | |
8549
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
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parents:
8502
diff
changeset
|
1730 } |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
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parents:
8502
diff
changeset
|
1731 |
7578 | 1732 // parse |
1733 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { | |
1734 elem_id = get_bits(&gb, 4); | |
1735 | |
10004 | 1736 if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) { |
8814 | 1737 av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); |
1738 return -1; | |
7578 | 1739 } |
1740 | |
1741 switch (elem_type) { | |
1742 | |
1743 case TYPE_SCE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
parents:
9143
diff
changeset
|
1744 err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
7578 | 1745 break; |
1746 | |
1747 case TYPE_CPE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
parents:
9143
diff
changeset
|
1748 err = decode_cpe(ac, &gb, che); |
7578 | 1749 break; |
1750 | |
1751 case TYPE_CCE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
parents:
9143
diff
changeset
|
1752 err = decode_cce(ac, &gb, che); |
7578 | 1753 break; |
1754 | |
1755 case TYPE_LFE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
parents:
9143
diff
changeset
|
1756 err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
7578 | 1757 break; |
1758 | |
1759 case TYPE_DSE: | |
1760 skip_data_stream_element(&gb); | |
1761 err = 0; | |
1762 break; | |
1763 | |
10004 | 1764 case TYPE_PCE: { |
7578 | 1765 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
1766 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
10004 | 1767 if ((err = decode_pce(ac, new_che_pos, &gb))) |
7578 | 1768 break; |
10551
a0dd118e82ae
10l: Fix inverted if-condition from r20448. Fixes issue 1562.
alexc
parents:
10519
diff
changeset
|
1769 if (ac->output_configured > OC_TRIAL_PCE) |
9948
98fd723f72e7
Ignore subsequent channel configurations after the first in an AAC file. The
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parents:
9891
diff
changeset
|
1770 av_log(avccontext, AV_LOG_ERROR, |
98fd723f72e7
Ignore subsequent channel configurations after the first in an AAC file. The
alexc
parents:
9891
diff
changeset
|
1771 "Not evaluating a further program_config_element as this construct is dubious at best.\n"); |
98fd723f72e7
Ignore subsequent channel configurations after the first in an AAC file. The
alexc
parents:
9891
diff
changeset
|
1772 else |
10491
74e51b86b934
Don't lock the channel output configuration based on the first value seen for
alexc
parents:
10378
diff
changeset
|
1773 err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE); |
7578 | 1774 break; |
1775 } | |
1776 | |
1777 case TYPE_FIL: | |
1778 if (elem_id == 15) | |
1779 elem_id += get_bits(&gb, 8) - 1; | |
1780 while (elem_id > 0) | |
1781 elem_id -= decode_extension_payload(ac, &gb, elem_id); | |
1782 err = 0; /* FIXME */ | |
1783 break; | |
1784 | |
1785 default: | |
1786 err = -1; /* should not happen, but keeps compiler happy */ | |
1787 break; | |
1788 } | |
1789 | |
10004 | 1790 if (err) |
7578 | 1791 return err; |
1792 } | |
1793 | |
1794 spectral_to_sample(ac); | |
1795 | |
7539 | 1796 if (!ac->is_saved) { |
1797 ac->is_saved = 1; | |
1798 *data_size = 0; | |
7580 | 1799 return buf_size; |
7539 | 1800 } |
1801 | |
1802 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
10004 | 1803 if (*data_size < data_size_tmp) { |
7539 | 1804 av_log(avccontext, AV_LOG_ERROR, |
1805 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
1806 *data_size, data_size_tmp); | |
1807 return -1; | |
1808 } | |
1809 *data_size = data_size_tmp; | |
1810 | |
1811 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
1812 | |
10491
74e51b86b934
Don't lock the channel output configuration based on the first value seen for
alexc
parents:
10378
diff
changeset
|
1813 if (ac->output_configured) |
74e51b86b934
Don't lock the channel output configuration based on the first value seen for
alexc
parents:
10378
diff
changeset
|
1814 ac->output_configured = OC_LOCKED; |
74e51b86b934
Don't lock the channel output configuration based on the first value seen for
alexc
parents:
10378
diff
changeset
|
1815 |
7539 | 1816 return buf_size; |
1817 } | |
1818 | |
10004 | 1819 static av_cold int aac_decode_close(AVCodecContext *avccontext) |
1820 { | |
1821 AACContext *ac = avccontext->priv_data; | |
7540 | 1822 int i, type; |
7501 | 1823 |
7523
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
parents:
7501
diff
changeset
|
1824 for (i = 0; i < MAX_ELEM_ID; i++) { |
10004 | 1825 for (type = 0; type < 4; type++) |
7540 | 1826 av_freep(&ac->che[type][i]); |
7501 | 1827 } |
1828 | |
1829 ff_mdct_end(&ac->mdct); | |
1830 ff_mdct_end(&ac->mdct_small); | |
10004 | 1831 return 0; |
7501 | 1832 } |
1833 | |
1834 AVCodec aac_decoder = { | |
1835 "aac", | |
1836 CODEC_TYPE_AUDIO, | |
1837 CODEC_ID_AAC, | |
1838 sizeof(AACContext), | |
1839 aac_decode_init, | |
1840 NULL, | |
1841 aac_decode_close, | |
1842 aac_decode_frame, | |
1843 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
10145
7955db355703
Make sample_fmts and channel_layouts compound literals const to reduce size of
reimar
parents:
10004
diff
changeset
|
1844 .sample_fmts = (const enum SampleFormat[]) { |
10004 | 1845 SAMPLE_FMT_S16,SAMPLE_FMT_NONE |
1846 }, | |
10306
c97a6cabee62
AAC: Add channel layout support for channel configuration as defined in the
superdump
parents:
10252
diff
changeset
|
1847 .channel_layouts = aac_channel_layout, |
7501 | 1848 }; |