Mercurial > libavcodec.hg
annotate aac.c @ 9757:8e4d442554b3 libavcodec
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author | reimar |
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date | Sun, 31 May 2009 09:57:42 +0000 |
parents | d45e44454d13 |
children | 4cac4001dd23 |
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 | |
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96 union float754 { float f; uint32_t i; }; |
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97 |
7501 | 98 static VLC vlc_scalefactors; |
99 static VLC vlc_spectral[11]; | |
100 | |
101 | |
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102 static ChannelElement* get_che(AACContext *ac, int type, int elem_id) { |
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103 static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 }; |
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104 if (ac->tag_che_map[type][elem_id]) { |
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105 return ac->tag_che_map[type][elem_id]; |
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106 } |
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107 if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) { |
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108 return NULL; |
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109 } |
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110 switch (ac->m4ac.chan_config) { |
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111 case 7: |
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112 if (ac->tags_mapped == 3 && type == TYPE_CPE) { |
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113 ac->tags_mapped++; |
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114 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; |
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115 } |
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116 case 6: |
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117 /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] |
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118 instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have |
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119 encountered such a stream, transfer the LFE[0] element to SCE[1] */ |
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120 if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { |
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121 ac->tags_mapped++; |
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122 return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; |
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123 } |
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124 case 5: |
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125 if (ac->tags_mapped == 2 && type == TYPE_CPE) { |
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126 ac->tags_mapped++; |
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127 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; |
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128 } |
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129 case 4: |
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130 if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { |
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131 ac->tags_mapped++; |
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132 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; |
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133 } |
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134 case 3: |
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135 case 2: |
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136 if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { |
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137 ac->tags_mapped++; |
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138 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; |
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139 } else if (ac->m4ac.chan_config == 2) { |
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140 return NULL; |
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141 } |
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142 case 1: |
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143 if (!ac->tags_mapped && type == TYPE_SCE) { |
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144 ac->tags_mapped++; |
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145 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; |
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146 } |
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147 default: |
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148 return NULL; |
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149 } |
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150 } |
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151 |
7539 | 152 /** |
7578 | 153 * Configure output channel order based on the current program configuration element. |
154 * | |
155 * @param che_pos current channel position configuration | |
156 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
157 * | |
158 * @return Returns error status. 0 - OK, !0 - error | |
159 */ | |
160 static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], | |
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161 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], int channel_config) { |
7578 | 162 AVCodecContext *avctx = ac->avccontext; |
163 int i, type, channels = 0; | |
164 | |
165 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
166 | |
167 /* Allocate or free elements depending on if they are in the | |
168 * current program configuration. | |
169 * | |
170 * Set up default 1:1 output mapping. | |
171 * | |
172 * For a 5.1 stream the output order will be: | |
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173 * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] |
7578 | 174 */ |
175 | |
176 for(i = 0; i < MAX_ELEM_ID; i++) { | |
177 for(type = 0; type < 4; type++) { | |
178 if(che_pos[type][i]) { | |
179 if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) | |
180 return AVERROR(ENOMEM); | |
181 if(type != TYPE_CCE) { | |
182 ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; | |
183 if(type == TYPE_CPE) { | |
184 ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; | |
185 } | |
186 } | |
187 } else | |
188 av_freep(&ac->che[type][i]); | |
189 } | |
190 } | |
191 | |
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192 if (channel_config) { |
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193 memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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194 ac->tags_mapped = 0; |
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195 } else { |
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196 memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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197 ac->tags_mapped = 4*MAX_ELEM_ID; |
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198 } |
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199 |
7578 | 200 avctx->channels = channels; |
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201 |
7578 | 202 return 0; |
203 } | |
204 | |
205 /** | |
7539 | 206 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
207 * | |
208 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
209 * @param sce_map mono (Single Channel Element) map | |
210 * @param type speaker type/position for these channels | |
211 */ | |
212 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
213 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { | |
214 while(n--) { | |
215 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map | |
216 map[get_bits(gb, 4)] = type; | |
217 } | |
218 } | |
219 | |
220 /** | |
221 * Decode program configuration element; reference: table 4.2. | |
222 * | |
223 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
224 * | |
225 * @return Returns error status. 0 - OK, !0 - error | |
226 */ | |
227 static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
228 GetBitContext * gb) { | |
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229 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index; |
7539 | 230 |
231 skip_bits(gb, 2); // object_type | |
232 | |
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233 sampling_index = get_bits(gb, 4); |
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234 if (ac->m4ac.sampling_index != sampling_index) |
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235 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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236 |
7501 | 237 num_front = get_bits(gb, 4); |
238 num_side = get_bits(gb, 4); | |
239 num_back = get_bits(gb, 4); | |
240 num_lfe = get_bits(gb, 2); | |
241 num_assoc_data = get_bits(gb, 3); | |
242 num_cc = get_bits(gb, 4); | |
243 | |
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244 if (get_bits1(gb)) |
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245 skip_bits(gb, 4); // mono_mixdown_tag |
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246 if (get_bits1(gb)) |
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247 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 248 |
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249 if (get_bits1(gb)) |
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250 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 251 |
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252 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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253 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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254 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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255 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 256 |
257 skip_bits_long(gb, 4 * num_assoc_data); | |
258 | |
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259 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 260 |
261 align_get_bits(gb); | |
262 | |
263 /* comment field, first byte is length */ | |
264 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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265 return 0; |
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266 } |
7501 | 267 |
7539 | 268 /** |
269 * Set up channel positions based on a default channel configuration | |
270 * as specified in table 1.17. | |
271 * | |
272 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
273 * | |
274 * @return Returns error status. 0 - OK, !0 - error | |
275 */ | |
276 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
277 int channel_config) | |
278 { | |
279 if(channel_config < 1 || channel_config > 7) { | |
280 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", | |
281 channel_config); | |
282 return -1; | |
283 } | |
284 | |
285 /* default channel configurations: | |
286 * | |
287 * 1ch : front center (mono) | |
288 * 2ch : L + R (stereo) | |
289 * 3ch : front center + L + R | |
290 * 4ch : front center + L + R + back center | |
291 * 5ch : front center + L + R + back stereo | |
292 * 6ch : front center + L + R + back stereo + LFE | |
293 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
294 */ | |
295 | |
296 if(channel_config != 2) | |
297 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) | |
298 if(channel_config > 1) | |
299 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) | |
300 if(channel_config == 4) | |
301 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center | |
302 if(channel_config > 4) | |
303 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] | |
304 = AAC_CHANNEL_BACK; // back stereo | |
305 if(channel_config > 5) | |
306 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE | |
307 if(channel_config == 7) | |
308 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right | |
309 | |
310 return 0; | |
311 } | |
312 | |
7578 | 313 /** |
314 * Decode GA "General Audio" specific configuration; reference: table 4.1. | |
315 * | |
316 * @return Returns error status. 0 - OK, !0 - error | |
317 */ | |
318 static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { | |
319 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
320 int extension_flag, ret; | |
321 | |
322 if(get_bits1(gb)) { // frameLengthFlag | |
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323 ff_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); |
7539 | 324 return -1; |
325 } | |
326 | |
327 if (get_bits1(gb)) // dependsOnCoreCoder | |
328 skip_bits(gb, 14); // coreCoderDelay | |
329 extension_flag = get_bits1(gb); | |
330 | |
331 if(ac->m4ac.object_type == AOT_AAC_SCALABLE || | |
332 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
333 skip_bits(gb, 3); // layerNr | |
334 | |
335 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
336 if (channel_config == 0) { | |
337 skip_bits(gb, 4); // element_instance_tag | |
338 if((ret = decode_pce(ac, new_che_pos, gb))) | |
339 return ret; | |
340 } else { | |
341 if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) | |
342 return ret; | |
343 } | |
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344 if((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config))) |
7539 | 345 return ret; |
346 | |
347 if (extension_flag) { | |
348 switch (ac->m4ac.object_type) { | |
349 case AOT_ER_BSAC: | |
350 skip_bits(gb, 5); // numOfSubFrame | |
351 skip_bits(gb, 11); // layer_length | |
352 break; | |
353 case AOT_ER_AAC_LC: | |
354 case AOT_ER_AAC_LTP: | |
355 case AOT_ER_AAC_SCALABLE: | |
356 case AOT_ER_AAC_LD: | |
357 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
358 * aacScalefactorDataResilienceFlag | |
359 * aacSpectralDataResilienceFlag | |
360 */ | |
361 break; | |
362 } | |
363 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
364 } | |
365 return 0; | |
366 } | |
367 | |
368 /** | |
369 * Decode audio specific configuration; reference: table 1.13. | |
370 * | |
371 * @param data pointer to AVCodecContext extradata | |
372 * @param data_size size of AVCCodecContext extradata | |
373 * | |
374 * @return Returns error status. 0 - OK, !0 - error | |
375 */ | |
376 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { | |
377 GetBitContext gb; | |
378 int i; | |
379 | |
380 init_get_bits(&gb, data, data_size * 8); | |
381 | |
382 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) | |
383 return -1; | |
8812 | 384 if(ac->m4ac.sampling_index > 12) { |
7539 | 385 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
386 return -1; | |
387 } | |
388 | |
389 skip_bits_long(&gb, i); | |
390 | |
391 switch (ac->m4ac.object_type) { | |
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392 case AOT_AAC_MAIN: |
7539 | 393 case AOT_AAC_LC: |
394 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
395 return -1; | |
396 break; | |
397 default: | |
398 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
399 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
400 return -1; | |
401 } | |
402 return 0; | |
403 } | |
404 | |
7578 | 405 /** |
406 * linear congruential pseudorandom number generator | |
407 * | |
408 * @param previous_val pointer to the current state of the generator | |
409 * | |
410 * @return Returns a 32-bit pseudorandom integer | |
411 */ | |
412 static av_always_inline int lcg_random(int previous_val) { | |
413 return previous_val * 1664525 + 1013904223; | |
414 } | |
415 | |
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416 static void reset_predict_state(PredictorState * ps) { |
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417 ps->r0 = 0.0f; |
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418 ps->r1 = 0.0f; |
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419 ps->cor0 = 0.0f; |
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420 ps->cor1 = 0.0f; |
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421 ps->var0 = 1.0f; |
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422 ps->var1 = 1.0f; |
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423 } |
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424 |
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425 static void reset_all_predictors(PredictorState * ps) { |
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426 int i; |
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427 for (i = 0; i < MAX_PREDICTORS; i++) |
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428 reset_predict_state(&ps[i]); |
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429 } |
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430 |
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431 static void reset_predictor_group(PredictorState * ps, int group_num) { |
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432 int i; |
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433 for (i = group_num-1; i < MAX_PREDICTORS; i+=30) |
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434 reset_predict_state(&ps[i]); |
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435 } |
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436 |
7501 | 437 static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
438 AACContext * ac = avccontext->priv_data; | |
439 int i; | |
440 | |
441 ac->avccontext = avccontext; | |
442 | |
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443 if (avccontext->extradata_size > 0) { |
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444 if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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445 return -1; |
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446 avccontext->sample_rate = ac->m4ac.sample_rate; |
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447 } else if (avccontext->channels > 0) { |
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448 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
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449 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
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450 if(set_default_channel_config(ac, new_che_pos, avccontext->channels - (avccontext->channels == 8))) |
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451 return -1; |
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452 if(output_configure(ac, ac->che_pos, new_che_pos, 1)) |
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453 return -1; |
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454 ac->m4ac.sample_rate = avccontext->sample_rate; |
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455 } |
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456 |
7539 | 457 avccontext->sample_fmt = SAMPLE_FMT_S16; |
7501 | 458 avccontext->frame_size = 1024; |
459 | |
460 AAC_INIT_VLC_STATIC( 0, 144); | |
461 AAC_INIT_VLC_STATIC( 1, 114); | |
462 AAC_INIT_VLC_STATIC( 2, 188); | |
463 AAC_INIT_VLC_STATIC( 3, 180); | |
464 AAC_INIT_VLC_STATIC( 4, 172); | |
465 AAC_INIT_VLC_STATIC( 5, 140); | |
466 AAC_INIT_VLC_STATIC( 6, 168); | |
467 AAC_INIT_VLC_STATIC( 7, 114); | |
468 AAC_INIT_VLC_STATIC( 8, 262); | |
469 AAC_INIT_VLC_STATIC( 9, 248); | |
470 AAC_INIT_VLC_STATIC(10, 384); | |
471 | |
472 dsputil_init(&ac->dsp, avccontext); | |
473 | |
7539 | 474 ac->random_state = 0x1f2e3d4c; |
475 | |
7501 | 476 // -1024 - Compensate wrong IMDCT method. |
477 // 32768 - Required to scale values to the correct range for the bias method | |
478 // for float to int16 conversion. | |
479 | |
480 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { | |
481 ac->add_bias = 385.0f; | |
482 ac->sf_scale = 1. / (-1024. * 32768.); | |
483 ac->sf_offset = 0; | |
484 } else { | |
485 ac->add_bias = 0.0f; | |
486 ac->sf_scale = 1. / -1024.; | |
487 ac->sf_offset = 60; | |
488 } | |
489 | |
8590 | 490 #if !CONFIG_HARDCODED_TABLES |
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491 for (i = 0; i < 428; i++) |
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492 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
7501 | 493 #endif /* CONFIG_HARDCODED_TABLES */ |
494 | |
8042 | 495 INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), |
7501 | 496 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), |
497 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
498 352); | |
499 | |
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500 ff_mdct_init(&ac->mdct, 11, 1, 1.0); |
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501 ff_mdct_init(&ac->mdct_small, 8, 1, 1.0); |
7581 | 502 // window initialization |
503 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
504 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
505 ff_sine_window_init(ff_sine_1024, 1024); | |
506 ff_sine_window_init(ff_sine_128, 128); | |
507 | |
7501 | 508 return 0; |
509 } | |
510 | |
7539 | 511 /** |
512 * Skip data_stream_element; reference: table 4.10. | |
513 */ | |
514 static void skip_data_stream_element(GetBitContext * gb) { | |
7501 | 515 int byte_align = get_bits1(gb); |
516 int count = get_bits(gb, 8); | |
517 if (count == 255) | |
518 count += get_bits(gb, 8); | |
519 if (byte_align) | |
520 align_get_bits(gb); | |
521 skip_bits_long(gb, 8 * count); | |
522 } | |
523 | |
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524 static int decode_prediction(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb) { |
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525 int sfb; |
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526 if (get_bits1(gb)) { |
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527 ics->predictor_reset_group = get_bits(gb, 5); |
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528 if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { |
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529 av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); |
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530 return -1; |
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531 } |
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532 } |
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533 for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { |
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534 ics->prediction_used[sfb] = get_bits1(gb); |
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535 } |
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536 return 0; |
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537 } |
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538 |
7501 | 539 /** |
7539 | 540 * Decode Individual Channel Stream info; reference: table 4.6. |
541 * | |
542 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
543 */ | |
544 static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { | |
545 if (get_bits1(gb)) { | |
546 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); | |
547 memset(ics, 0, sizeof(IndividualChannelStream)); | |
548 return -1; | |
549 } | |
550 ics->window_sequence[1] = ics->window_sequence[0]; | |
551 ics->window_sequence[0] = get_bits(gb, 2); | |
552 ics->use_kb_window[1] = ics->use_kb_window[0]; | |
553 ics->use_kb_window[0] = get_bits1(gb); | |
554 ics->num_window_groups = 1; | |
555 ics->group_len[0] = 1; | |
7581 | 556 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
557 int i; | |
558 ics->max_sfb = get_bits(gb, 4); | |
559 for (i = 0; i < 7; i++) { | |
560 if (get_bits1(gb)) { | |
561 ics->group_len[ics->num_window_groups-1]++; | |
562 } else { | |
563 ics->num_window_groups++; | |
564 ics->group_len[ics->num_window_groups-1] = 1; | |
565 } | |
566 } | |
567 ics->num_windows = 8; | |
568 ics->swb_offset = swb_offset_128[ac->m4ac.sampling_index]; | |
569 ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; | |
570 ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index]; | |
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571 ics->predictor_present = 0; |
7581 | 572 } else { |
573 ics->max_sfb = get_bits(gb, 6); | |
574 ics->num_windows = 1; | |
575 ics->swb_offset = swb_offset_1024[ac->m4ac.sampling_index]; | |
576 ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; | |
577 ics->tns_max_bands = tns_max_bands_1024[ac->m4ac.sampling_index]; | |
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578 ics->predictor_present = get_bits1(gb); |
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579 ics->predictor_reset_group = 0; |
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580 if (ics->predictor_present) { |
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581 if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
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AAC: Frequency domain prediction and hence Main profile support
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582 if (decode_prediction(ac, ics, gb)) { |
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583 memset(ics, 0, sizeof(IndividualChannelStream)); |
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584 return -1; |
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585 } |
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586 } else if (ac->m4ac.object_type == AOT_AAC_LC) { |
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587 av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); |
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588 memset(ics, 0, sizeof(IndividualChannelStream)); |
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589 return -1; |
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590 } else { |
8281
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591 ff_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); |
8200 | 592 memset(ics, 0, sizeof(IndividualChannelStream)); |
593 return -1; | |
8199
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594 } |
7578 | 595 } |
596 } | |
597 | |
598 if(ics->max_sfb > ics->num_swb) { | |
599 av_log(ac->avccontext, AV_LOG_ERROR, | |
600 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", | |
601 ics->max_sfb, ics->num_swb); | |
602 memset(ics, 0, sizeof(IndividualChannelStream)); | |
603 return -1; | |
604 } | |
605 | |
7539 | 606 return 0; |
607 } | |
608 | |
609 /** | |
610 * Decode band types (section_data payload); reference: table 4.46. | |
611 * | |
612 * @param band_type array of the used band type | |
613 * @param band_type_run_end array of the last scalefactor band of a band type run | |
614 * | |
615 * @return Returns error status. 0 - OK, !0 - error | |
616 */ | |
617 static int decode_band_types(AACContext * ac, enum BandType band_type[120], | |
7523
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618 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
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619 int g, idx = 0; |
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620 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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621 for (g = 0; g < ics->num_window_groups; g++) { |
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622 int k = 0; |
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623 while (k < ics->max_sfb) { |
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624 uint8_t sect_len = k; |
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625 int sect_len_incr; |
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626 int sect_band_type = get_bits(gb, 4); |
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627 if (sect_band_type == 12) { |
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628 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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629 return -1; |
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630 } |
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631 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
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632 sect_len += sect_len_incr; |
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633 sect_len += sect_len_incr; |
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634 if (sect_len > ics->max_sfb) { |
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635 av_log(ac->avccontext, AV_LOG_ERROR, |
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636 "Number of bands (%d) exceeds limit (%d).\n", |
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637 sect_len, ics->max_sfb); |
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638 return -1; |
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639 } |
7581 | 640 for (; k < sect_len; k++) { |
641 band_type [idx] = sect_band_type; | |
642 band_type_run_end[idx++] = sect_len; | |
643 } | |
7539 | 644 } |
645 } | |
646 return 0; | |
647 } | |
7523
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648 |
7539 | 649 /** |
650 * Decode scalefactors; reference: table 4.47. | |
7523
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651 * |
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652 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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653 * @param band_type array of the used band type |
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654 * @param band_type_run_end array of the last scalefactor band of a band type run |
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655 * @param sf array of scalefactors or intensity stereo positions |
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656 * |
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657 * @return Returns error status. 0 - OK, !0 - error |
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658 */ |
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659 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
7540 | 660 unsigned int global_gain, IndividualChannelStream * ics, |
7523
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661 enum BandType band_type[120], int band_type_run_end[120]) { |
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662 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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663 int g, i, idx = 0; |
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664 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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665 int noise_flag = 1; |
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666 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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667 for (g = 0; g < ics->num_window_groups; g++) { |
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668 for (i = 0; i < ics->max_sfb;) { |
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669 int run_end = band_type_run_end[idx]; |
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670 if (band_type[idx] == ZERO_BT) { |
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671 for(; i < run_end; i++, idx++) |
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672 sf[idx] = 0.; |
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673 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
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674 for(; i < run_end; i++, idx++) { |
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675 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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676 if(offset[2] > 255U) { |
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677 av_log(ac->avccontext, AV_LOG_ERROR, |
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678 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
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679 return -1; |
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680 } |
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681 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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682 } |
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683 }else if(band_type[idx] == NOISE_BT) { |
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684 for(; i < run_end; i++, idx++) { |
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685 if(noise_flag-- > 0) |
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686 offset[1] += get_bits(gb, 9) - 256; |
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687 else |
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688 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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689 if(offset[1] > 255U) { |
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690 av_log(ac->avccontext, AV_LOG_ERROR, |
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691 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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692 return -1; |
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693 } |
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694 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset + 100]; |
7523
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695 } |
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696 }else { |
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697 for(; i < run_end; i++, idx++) { |
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698 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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699 if(offset[0] > 255U) { |
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700 av_log(ac->avccontext, AV_LOG_ERROR, |
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701 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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702 return -1; |
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703 } |
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704 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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705 } |
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706 } |
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707 } |
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708 } |
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709 return 0; |
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710 } |
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711 |
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712 /** |
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713 * Decode pulse data; reference: table 4.7. |
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714 */ |
7871
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715 static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) { |
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716 int i, pulse_swb; |
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717 pulse->num_pulse = get_bits(gb, 2) + 1; |
7871
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718 pulse_swb = get_bits(gb, 6); |
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719 if (pulse_swb >= num_swb) |
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720 return -1; |
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721 pulse->pos[0] = swb_offset[pulse_swb]; |
7856
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722 pulse->pos[0] += get_bits(gb, 5); |
7871
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723 if (pulse->pos[0] > 1023) |
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724 return -1; |
7580 | 725 pulse->amp[0] = get_bits(gb, 4); |
726 for (i = 1; i < pulse->num_pulse; i++) { | |
727 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; | |
7871
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728 if (pulse->pos[i] > 1023) |
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729 return -1; |
7580 | 730 pulse->amp[i] = get_bits(gb, 4); |
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731 } |
7871
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732 return 0; |
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diff
changeset
|
733 } |
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
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diff
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|
734 |
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
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diff
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|
735 /** |
7609 | 736 * Decode Temporal Noise Shaping data; reference: table 4.48. |
737 * | |
738 * @return Returns error status. 0 - OK, !0 - error | |
739 */ | |
740 static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns, | |
741 GetBitContext * gb, const IndividualChannelStream * ics) { | |
742 int w, filt, i, coef_len, coef_res, coef_compress; | |
743 const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; | |
744 const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; | |
745 for (w = 0; w < ics->num_windows; w++) { | |
7637
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Work around GCC variable may be used uninitialised warning
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diff
changeset
|
746 if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
7609 | 747 coef_res = get_bits1(gb); |
748 | |
7638 | 749 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
750 int tmp2_idx; | |
751 tns->length[w][filt] = get_bits(gb, 6 - 2*is8); | |
7609 | 752 |
7638 | 753 if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) { |
754 av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", | |
755 tns->order[w][filt], tns_max_order); | |
756 tns->order[w][filt] = 0; | |
757 return -1; | |
758 } | |
7850
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diff
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|
759 if (tns->order[w][filt]) { |
7851 | 760 tns->direction[w][filt] = get_bits1(gb); |
761 coef_compress = get_bits1(gb); | |
762 coef_len = coef_res + 3 - coef_compress; | |
763 tmp2_idx = 2*coef_compress + coef_res; | |
7638 | 764 |
7851 | 765 for (i = 0; i < tns->order[w][filt]; i++) |
766 tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; | |
7850
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diff
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|
767 } |
7609 | 768 } |
7637
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Work around GCC variable may be used uninitialised warning
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diff
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|
769 } |
7609 | 770 } |
771 return 0; | |
772 } | |
773 | |
774 /** | |
7539 | 775 * Decode Mid/Side data; reference: table 4.54. |
776 * | |
777 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
778 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
779 * [3] reserved for scalable AAC | |
780 */ | |
781 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, | |
782 int ms_present) { | |
7578 | 783 int idx; |
784 if (ms_present == 1) { | |
785 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) | |
786 cpe->ms_mask[idx] = get_bits1(gb); | |
787 } else if (ms_present == 2) { | |
788 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); | |
789 } | |
790 } | |
7539 | 791 |
792 /** | |
7581 | 793 * Decode spectral data; reference: table 4.50. |
794 * Dequantize and scale spectral data; reference: 4.6.3.3. | |
795 * | |
796 * @param coef array of dequantized, scaled spectral data | |
797 * @param sf array of scalefactors or intensity stereo positions | |
798 * @param pulse_present set if pulses are present | |
799 * @param pulse pointer to pulse data struct | |
800 * @param band_type array of the used band type | |
801 * | |
802 * @return Returns error status. 0 - OK, !0 - error | |
803 */ | |
804 static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120], | |
805 int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) { | |
806 int i, k, g, idx = 0; | |
807 const int c = 1024/ics->num_windows; | |
808 const uint16_t * offsets = ics->swb_offset; | |
809 float *coef_base = coef; | |
8284 | 810 static const float sign_lookup[] = { 1.0f, -1.0f }; |
7581 | 811 |
812 for (g = 0; g < ics->num_windows; g++) | |
813 memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb])); | |
814 | |
815 for (g = 0; g < ics->num_window_groups; g++) { | |
816 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
817 const int cur_band_type = band_type[idx]; | |
818 const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; | |
819 const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); | |
820 int group; | |
8413
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|
821 if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) { |
7581 | 822 for (group = 0; group < ics->group_len[g]; group++) { |
823 memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); | |
824 } | |
825 }else if (cur_band_type == NOISE_BT) { | |
826 for (group = 0; group < ics->group_len[g]; group++) { | |
8109
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|
827 float scale; |
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Fix perceptual noise shaping scaling including an oversight of the offset for
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diff
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|
828 float band_energy = 0; |
7581 | 829 for (k = offsets[i]; k < offsets[i+1]; k++) { |
830 ac->random_state = lcg_random(ac->random_state); | |
8109
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Fix perceptual noise shaping scaling including an oversight of the offset for
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|
831 coef[group*128+k] = ac->random_state; |
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Fix perceptual noise shaping scaling including an oversight of the offset for
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diff
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|
832 band_energy += coef[group*128+k]*coef[group*128+k]; |
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diff
changeset
|
833 } |
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Fix perceptual noise shaping scaling including an oversight of the offset for
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diff
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|
834 scale = sf[idx] / sqrtf(band_energy); |
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Fix perceptual noise shaping scaling including an oversight of the offset for
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diff
changeset
|
835 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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Fix perceptual noise shaping scaling including an oversight of the offset for
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diff
changeset
|
836 coef[group*128+k] *= scale; |
7581 | 837 } |
838 } | |
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|
839 }else { |
7581 | 840 for (group = 0; group < ics->group_len[g]; group++) { |
841 for (k = offsets[i]; k < offsets[i+1]; k += dim) { | |
842 const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); | |
843 const int coef_tmp_idx = (group << 7) + k; | |
844 const float *vq_ptr; | |
845 int j; | |
846 if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { | |
847 av_log(ac->avccontext, AV_LOG_ERROR, | |
848 "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", | |
849 cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); | |
850 return -1; | |
851 } | |
852 vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; | |
853 if (is_cb_unsigned) { | |
8284 | 854 if (vq_ptr[0]) coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)]; |
855 if (vq_ptr[1]) 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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parents:
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diff
changeset
|
856 if (dim == 4) { |
8284 | 857 if (vq_ptr[2]) coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)]; |
858 if (vq_ptr[3]) 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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8282
diff
changeset
|
859 } |
9053 | 860 if (cur_band_type == ESC_BT) { |
861 for (j = 0; j < 2; j++) { | |
862 if (vq_ptr[j] == 64.0f) { | |
863 int n = 4; | |
864 /* The total length of escape_sequence must be < 22 bits according | |
865 to the specification (i.e. max is 11111111110xxxxxxxxxx). */ | |
866 while (get_bits1(gb) && n < 15) n++; | |
867 if(n == 15) { | |
868 av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); | |
869 return -1; | |
870 } | |
871 n = (1<<n) + get_bits(gb, n); | |
872 coef[coef_tmp_idx + j] *= cbrtf(n) * n; | |
873 }else | |
874 coef[coef_tmp_idx + j] *= vq_ptr[j]; | |
875 } | |
876 }else | |
877 { | |
878 coef[coef_tmp_idx ] *= vq_ptr[0]; | |
879 coef[coef_tmp_idx + 1] *= vq_ptr[1]; | |
880 if (dim == 4) { | |
881 coef[coef_tmp_idx + 2] *= vq_ptr[2]; | |
882 coef[coef_tmp_idx + 3] *= vq_ptr[3]; | |
883 } | |
7581 | 884 } |
9052
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
885 }else { |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
886 coef[coef_tmp_idx ] = vq_ptr[0]; |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
887 coef[coef_tmp_idx + 1] = vq_ptr[1]; |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
888 if (dim == 4) { |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
889 coef[coef_tmp_idx + 2] = vq_ptr[2]; |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
890 coef[coef_tmp_idx + 3] = vq_ptr[3]; |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
891 } |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
892 } |
8283
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
893 coef[coef_tmp_idx ] *= sf[idx]; |
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
894 coef[coef_tmp_idx + 1] *= sf[idx]; |
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
895 if (dim == 4) { |
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
896 coef[coef_tmp_idx + 2] *= sf[idx]; |
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
897 coef[coef_tmp_idx + 3] *= sf[idx]; |
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
898 } |
7581 | 899 } |
900 } | |
901 } | |
902 } | |
903 coef += ics->group_len[g]<<7; | |
904 } | |
905 | |
906 if (pulse_present) { | |
7857
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
changeset
|
907 idx = 0; |
7581 | 908 for(i = 0; i < pulse->num_pulse; i++){ |
909 float co = coef_base[ pulse->pos[i] ]; | |
7857
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
changeset
|
910 while(offsets[idx + 1] <= pulse->pos[i]) |
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
changeset
|
911 idx++; |
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
changeset
|
912 if (band_type[idx] != NOISE_BT && sf[idx]) { |
7858 | 913 float ico = -pulse->amp[i]; |
914 if (co) { | |
915 co /= sf[idx]; | |
916 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); | |
917 } | |
918 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; | |
7857
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
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|
919 } |
7581 | 920 } |
921 } | |
922 return 0; | |
923 } | |
924 | |
8199
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
925 static av_always_inline float flt16_round(float pf) { |
9179
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
926 union float754 tmp; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
927 tmp.f = pf; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
928 tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
929 return tmp.f; |
8199
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AAC: Frequency domain prediction and hence Main profile support
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parents:
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diff
changeset
|
930 } |
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AAC: Frequency domain prediction and hence Main profile support
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parents:
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diff
changeset
|
931 |
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AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
932 static av_always_inline float flt16_even(float pf) { |
9179
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
933 union float754 tmp; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
934 tmp.f = pf; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
935 tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U>>16)) & 0xFFFF0000U; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
936 return tmp.f; |
8199
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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parents:
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diff
changeset
|
937 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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parents:
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diff
changeset
|
938 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
939 static av_always_inline float flt16_trunc(float pf) { |
9179
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
940 union float754 pun; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
941 pun.f = pf; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
942 pun.i &= 0xFFFF0000U; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
943 return pun.f; |
8199
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
944 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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parents:
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diff
changeset
|
945 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
changeset
|
946 static void predict(AACContext * ac, PredictorState * ps, float* coef, int output_enable) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
947 const float a = 0.953125; // 61.0/64 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
948 const float alpha = 0.90625; // 29.0/32 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
949 float e0, e1; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
950 float pv; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
951 float k1, k2; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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parents:
8109
diff
changeset
|
952 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
953 k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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parents:
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diff
changeset
|
954 k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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parents:
8109
diff
changeset
|
955 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
956 pv = flt16_round(k1 * ps->r0 + k2 * ps->r1); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
957 if (output_enable) |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
958 *coef += pv * ac->sf_scale; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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parents:
8109
diff
changeset
|
959 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
960 e0 = *coef / ac->sf_scale; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
961 e1 = e0 - k1 * ps->r0; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
962 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
963 ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
964 ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1)); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
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diff
changeset
|
965 ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
966 ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0)); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
967 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
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|
968 ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0)); |
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969 ps->r0 = flt16_trunc(a * e0); |
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970 } |
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971 |
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|
972 /** |
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973 * Apply AAC-Main style frequency domain prediction. |
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|
974 */ |
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975 static void apply_prediction(AACContext * ac, SingleChannelElement * sce) { |
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976 int sfb, k; |
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977 |
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978 if (!sce->ics.predictor_initialized) { |
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979 reset_all_predictors(sce->predictor_state); |
8199
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980 sce->ics.predictor_initialized = 1; |
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|
981 } |
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982 |
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983 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
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984 for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { |
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985 for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { |
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986 predict(ac, &sce->predictor_state[k], &sce->coeffs[k], |
8199
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987 sce->ics.predictor_present && sce->ics.prediction_used[sfb]); |
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988 } |
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989 } |
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990 if (sce->ics.predictor_reset_group) |
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991 reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group); |
8199
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992 } else |
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993 reset_all_predictors(sce->predictor_state); |
8199
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994 } |
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995 |
7581 | 996 /** |
7539 | 997 * Decode an individual_channel_stream payload; reference: table 4.44. |
998 * | |
999 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
1000 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
1001 * | |
1002 * @return Returns error status. 0 - OK, !0 - error | |
1003 */ | |
1004 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { | |
1005 Pulse pulse; | |
1006 TemporalNoiseShaping * tns = &sce->tns; | |
1007 IndividualChannelStream * ics = &sce->ics; | |
1008 float * out = sce->coeffs; | |
1009 int global_gain, pulse_present = 0; | |
1010 | |
7580 | 1011 /* This assignment is to silence a GCC warning about the variable being used |
1012 * uninitialized when in fact it always is. | |
7539 | 1013 */ |
1014 pulse.num_pulse = 0; | |
1015 | |
1016 global_gain = get_bits(gb, 8); | |
1017 | |
1018 if (!common_window && !scale_flag) { | |
1019 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
1020 return -1; | |
1021 } | |
1022 | |
1023 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
1024 return -1; | |
1025 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
1026 return -1; | |
1027 | |
1028 pulse_present = 0; | |
1029 if (!scale_flag) { | |
1030 if ((pulse_present = get_bits1(gb))) { | |
1031 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
1032 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
1033 return -1; | |
1034 } | |
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1035 if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { |
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1036 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); |
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1037 return -1; |
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1038 } |
7539 | 1039 } |
1040 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
1041 return -1; | |
1042 if (get_bits1(gb)) { | |
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1043 ff_log_missing_feature(ac->avccontext, "SSR", 1); |
7539 | 1044 return -1; |
1045 } | |
1046 } | |
1047 | |
7580 | 1048 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
7539 | 1049 return -1; |
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1050 |
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1051 if(ac->m4ac.object_type == AOT_AAC_MAIN && !common_window) |
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1052 apply_prediction(ac, sce); |
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1053 |
7539 | 1054 return 0; |
1055 } | |
1056 | |
1057 /** | |
7581 | 1058 * Mid/Side stereo decoding; reference: 4.6.8.1.3. |
1059 */ | |
1060 static void apply_mid_side_stereo(ChannelElement * cpe) { | |
1061 const IndividualChannelStream * ics = &cpe->ch[0].ics; | |
1062 float *ch0 = cpe->ch[0].coeffs; | |
1063 float *ch1 = cpe->ch[1].coeffs; | |
1064 int g, i, k, group, idx = 0; | |
1065 const uint16_t * offsets = ics->swb_offset; | |
1066 for (g = 0; g < ics->num_window_groups; g++) { | |
1067 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
1068 if (cpe->ms_mask[idx] && | |
1069 cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { | |
1070 for (group = 0; group < ics->group_len[g]; group++) { | |
1071 for (k = offsets[i]; k < offsets[i+1]; k++) { | |
1072 float tmp = ch0[group*128 + k] - ch1[group*128 + k]; | |
1073 ch0[group*128 + k] += ch1[group*128 + k]; | |
1074 ch1[group*128 + k] = tmp; | |
1075 } | |
1076 } | |
1077 } | |
1078 } | |
1079 ch0 += ics->group_len[g]*128; | |
1080 ch1 += ics->group_len[g]*128; | |
1081 } | |
1082 } | |
1083 | |
1084 /** | |
1085 * intensity stereo decoding; reference: 4.6.8.2.3 | |
1086 * | |
1087 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
1088 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
1089 * [3] reserved for scalable AAC | |
1090 */ | |
1091 static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) { | |
1092 const IndividualChannelStream * ics = &cpe->ch[1].ics; | |
1093 SingleChannelElement * sce1 = &cpe->ch[1]; | |
1094 float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; | |
1095 const uint16_t * offsets = ics->swb_offset; | |
1096 int g, group, i, k, idx = 0; | |
1097 int c; | |
1098 float scale; | |
1099 for (g = 0; g < ics->num_window_groups; g++) { | |
1100 for (i = 0; i < ics->max_sfb;) { | |
1101 if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { | |
1102 const int bt_run_end = sce1->band_type_run_end[idx]; | |
1103 for (; i < bt_run_end; i++, idx++) { | |
1104 c = -1 + 2 * (sce1->band_type[idx] - 14); | |
1105 if (ms_present) | |
1106 c *= 1 - 2 * cpe->ms_mask[idx]; | |
1107 scale = c * sce1->sf[idx]; | |
1108 for (group = 0; group < ics->group_len[g]; group++) | |
1109 for (k = offsets[i]; k < offsets[i+1]; k++) | |
1110 coef1[group*128 + k] = scale * coef0[group*128 + k]; | |
1111 } | |
1112 } else { | |
1113 int bt_run_end = sce1->band_type_run_end[idx]; | |
1114 idx += bt_run_end - i; | |
1115 i = bt_run_end; | |
1116 } | |
1117 } | |
1118 coef0 += ics->group_len[g]*128; | |
1119 coef1 += ics->group_len[g]*128; | |
1120 } | |
1121 } | |
1122 | |
1123 /** | |
7539 | 1124 * Decode a channel_pair_element; reference: table 4.4. |
1125 * | |
1126 * @param elem_id Identifies the instance of a syntax element. | |
1127 * | |
1128 * @return Returns error status. 0 - OK, !0 - error | |
1129 */ | |
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1130 static int decode_cpe(AACContext * ac, GetBitContext * gb, ChannelElement * cpe) { |
7539 | 1131 int i, ret, common_window, ms_present = 0; |
1132 | |
1133 common_window = get_bits1(gb); | |
1134 if (common_window) { | |
1135 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
1136 return -1; | |
1137 i = cpe->ch[1].ics.use_kb_window[0]; | |
1138 cpe->ch[1].ics = cpe->ch[0].ics; | |
1139 cpe->ch[1].ics.use_kb_window[1] = i; | |
1140 ms_present = get_bits(gb, 2); | |
1141 if(ms_present == 3) { | |
1142 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); | |
1143 return -1; | |
1144 } else if(ms_present) | |
1145 decode_mid_side_stereo(cpe, gb, ms_present); | |
1146 } | |
1147 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
1148 return ret; | |
1149 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
1150 return ret; | |
1151 | |
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1152 if (common_window) { |
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1153 if (ms_present) |
8414 | 1154 apply_mid_side_stereo(cpe); |
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1155 if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
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1156 apply_prediction(ac, &cpe->ch[0]); |
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1157 apply_prediction(ac, &cpe->ch[1]); |
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1158 } |
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1159 } |
7539 | 1160 |
7580 | 1161 apply_intensity_stereo(cpe, ms_present); |
7539 | 1162 return 0; |
1163 } | |
1164 | |
7581 | 1165 /** |
1166 * Decode coupling_channel_element; reference: table 4.8. | |
1167 * | |
1168 * @param elem_id Identifies the instance of a syntax element. | |
1169 * | |
1170 * @return Returns error status. 0 - OK, !0 - error | |
1171 */ | |
1172 static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) { | |
1173 int num_gain = 0; | |
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1174 int c, g, sfb, ret; |
7581 | 1175 int sign; |
1176 float scale; | |
1177 SingleChannelElement * sce = &che->ch[0]; | |
1178 ChannelCoupling * coup = &che->coup; | |
1179 | |
7578 | 1180 coup->coupling_point = 2*get_bits1(gb); |
1181 coup->num_coupled = get_bits(gb, 3); | |
1182 for (c = 0; c <= coup->num_coupled; c++) { | |
1183 num_gain++; | |
1184 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; | |
1185 coup->id_select[c] = get_bits(gb, 4); | |
1186 if (coup->type[c] == TYPE_CPE) { | |
1187 coup->ch_select[c] = get_bits(gb, 2); | |
1188 if (coup->ch_select[c] == 3) | |
1189 num_gain++; | |
1190 } else | |
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1191 coup->ch_select[c] = 2; |
7578 | 1192 } |
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1193 coup->coupling_point += get_bits1(gb) || (coup->coupling_point>>1); |
7578 | 1194 |
1195 sign = get_bits(gb, 1); | |
7889
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1196 scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); |
7578 | 1197 |
1198 if ((ret = decode_ics(ac, sce, gb, 0, 0))) | |
1199 return ret; | |
1200 | |
1201 for (c = 0; c < num_gain; c++) { | |
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1202 int idx = 0; |
7578 | 1203 int cge = 1; |
1204 int gain = 0; | |
1205 float gain_cache = 1.; | |
1206 if (c) { | |
1207 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); | |
1208 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; | |
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1209 gain_cache = pow(scale, -gain); |
7578 | 1210 } |
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1211 if (coup->coupling_point == AFTER_IMDCT) { |
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1212 coup->gain[c][0] = gain_cache; |
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1213 } else { |
8823 | 1214 for (g = 0; g < sce->ics.num_window_groups; g++) { |
1215 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { | |
1216 if (sce->band_type[idx] != ZERO_BT) { | |
1217 if (!cge) { | |
1218 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; | |
1219 if (t) { | |
1220 int s = 1; | |
1221 t = gain += t; | |
1222 if (sign) { | |
1223 s -= 2 * (t & 0x1); | |
1224 t >>= 1; | |
1225 } | |
1226 gain_cache = pow(scale, -t) * s; | |
7578 | 1227 } |
1228 } | |
8823 | 1229 coup->gain[c][idx] = gain_cache; |
7578 | 1230 } |
1231 } | |
7890 | 1232 } |
1233 } | |
7578 | 1234 } |
1235 return 0; | |
1236 } | |
1237 | |
7539 | 1238 /** |
1239 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
7523
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1240 * |
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1241 * @param crc flag indicating the presence of CRC checksum |
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1242 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 1243 * |
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1244 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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1245 */ |
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1246 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
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1247 // TODO : sbr_extension implementation |
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1248 ff_log_missing_feature(ac->avccontext, "SBR", 0); |
7523
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1249 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
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1250 return cnt; |
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1251 } |
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1252 |
7539 | 1253 /** |
7578 | 1254 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. |
1255 * | |
1256 * @return Returns number of bytes consumed. | |
1257 */ | |
1258 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { | |
1259 int i; | |
1260 int num_excl_chan = 0; | |
1261 | |
1262 do { | |
1263 for (i = 0; i < 7; i++) | |
1264 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); | |
1265 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); | |
1266 | |
1267 return num_excl_chan / 7; | |
1268 } | |
1269 | |
1270 /** | |
7539 | 1271 * Decode dynamic range information; reference: table 4.52. |
1272 * | |
1273 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1274 * | |
1275 * @return Returns number of bytes consumed. | |
1276 */ | |
1277 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { | |
1278 int n = 1; | |
1279 int drc_num_bands = 1; | |
1280 int i; | |
1281 | |
1282 /* pce_tag_present? */ | |
1283 if(get_bits1(gb)) { | |
1284 che_drc->pce_instance_tag = get_bits(gb, 4); | |
1285 skip_bits(gb, 4); // tag_reserved_bits | |
1286 n++; | |
1287 } | |
1288 | |
1289 /* excluded_chns_present? */ | |
1290 if(get_bits1(gb)) { | |
1291 n += decode_drc_channel_exclusions(che_drc, gb); | |
1292 } | |
1293 | |
1294 /* drc_bands_present? */ | |
1295 if (get_bits1(gb)) { | |
1296 che_drc->band_incr = get_bits(gb, 4); | |
1297 che_drc->interpolation_scheme = get_bits(gb, 4); | |
1298 n++; | |
1299 drc_num_bands += che_drc->band_incr; | |
1300 for (i = 0; i < drc_num_bands; i++) { | |
1301 che_drc->band_top[i] = get_bits(gb, 8); | |
1302 n++; | |
1303 } | |
1304 } | |
1305 | |
1306 /* prog_ref_level_present? */ | |
1307 if (get_bits1(gb)) { | |
1308 che_drc->prog_ref_level = get_bits(gb, 7); | |
1309 skip_bits1(gb); // prog_ref_level_reserved_bits | |
1310 n++; | |
1311 } | |
1312 | |
1313 for (i = 0; i < drc_num_bands; i++) { | |
1314 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
1315 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
1316 n++; | |
1317 } | |
1318 | |
1319 return n; | |
1320 } | |
1321 | |
1322 /** | |
1323 * Decode extension data (incomplete); reference: table 4.51. | |
1324 * | |
1325 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1326 * | |
1327 * @return Returns number of bytes consumed | |
1328 */ | |
1329 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { | |
7523
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1330 int crc_flag = 0; |
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1331 int res = cnt; |
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1332 switch (get_bits(gb, 4)) { // extension type |
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1333 case EXT_SBR_DATA_CRC: |
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1334 crc_flag++; |
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1335 case EXT_SBR_DATA: |
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1336 res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
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1337 break; |
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1338 case EXT_DYNAMIC_RANGE: |
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1339 res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
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1340 break; |
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1341 case EXT_FILL: |
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1342 case EXT_FILL_DATA: |
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1343 case EXT_DATA_ELEMENT: |
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1344 default: |
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1345 skip_bits_long(gb, 8*cnt - 4); |
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1346 break; |
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1347 }; |
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1348 return res; |
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1349 } |
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1350 |
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1351 /** |
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1352 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. |
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1353 * |
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1354 * @param decode 1 if tool is used normally, 0 if tool is used in LTP. |
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1355 * @param coef spectral coefficients |
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1356 */ |
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1357 static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) { |
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1358 const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); |
7636 | 1359 int w, filt, m, i; |
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1360 int bottom, top, order, start, end, size, inc; |
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1361 float lpc[TNS_MAX_ORDER]; |
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1362 |
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1363 for (w = 0; w < ics->num_windows; w++) { |
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1364 bottom = ics->num_swb; |
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1365 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
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1366 top = bottom; |
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1367 bottom = FFMAX(0, top - tns->length[w][filt]); |
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1368 order = tns->order[w][filt]; |
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1369 if (order == 0) |
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1370 continue; |
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1371 |
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1372 // tns_decode_coef |
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1373 compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); |
7632
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1374 |
7609 | 1375 start = ics->swb_offset[FFMIN(bottom, mmm)]; |
1376 end = ics->swb_offset[FFMIN( top, mmm)]; | |
1377 if ((size = end - start) <= 0) | |
1378 continue; | |
1379 if (tns->direction[w][filt]) { | |
1380 inc = -1; start = end - 1; | |
1381 } else { | |
1382 inc = 1; | |
1383 } | |
1384 start += w * 128; | |
1385 | |
1386 // ar filter | |
1387 for (m = 0; m < size; m++, start += inc) | |
1388 for (i = 1; i <= FFMIN(m, order); i++) | |
7632
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1389 coef[start] -= coef[start - i*inc] * lpc[i-1]; |
7609 | 1390 } |
1391 } | |
1392 } | |
1393 | |
7523
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1394 /** |
7539 | 1395 * Conduct IMDCT and windowing. |
1396 */ | |
1397 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { | |
1398 IndividualChannelStream * ics = &sce->ics; | |
1399 float * in = sce->coeffs; | |
1400 float * out = sce->ret; | |
1401 float * saved = sce->saved; | |
7580 | 1402 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
1403 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
1404 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
7539 | 1405 float * buf = ac->buf_mdct; |
8286
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1406 float * temp = ac->temp; |
7539 | 1407 int i; |
1408 | |
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1409 // imdct |
7578 | 1410 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1411 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) | |
1412 av_log(ac->avccontext, AV_LOG_WARNING, | |
1413 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " | |
1414 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); | |
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1415 for (i = 0; i < 1024; i += 128) |
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1416 ff_imdct_half(&ac->mdct_small, buf + i, in + i); |
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1417 } else |
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1418 ff_imdct_half(&ac->mdct, buf, in); |
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1419 |
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1420 /* window overlapping |
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1421 * NOTE: To simplify the overlapping code, all 'meaningless' short to long |
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1422 * and long to short transitions are considered to be short to short |
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1423 * transitions. This leaves just two cases (long to long and short to short) |
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1424 * with a little special sauce for EIGHT_SHORT_SEQUENCE. |
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1425 */ |
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1426 if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
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1427 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
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1428 ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512); |
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1429 } else { |
7662 | 1430 for (i = 0; i < 448; i++) |
1431 out[i] = saved[i] + ac->add_bias; | |
7578 | 1432 |
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1433 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
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1434 ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64); |
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1435 ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, ac->add_bias, 64); |
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1436 ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, ac->add_bias, 64); |
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1437 ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, ac->add_bias, 64); |
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1438 ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64); |
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1439 memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
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1440 } else { |
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1441 ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64); |
7662 | 1442 for (i = 576; i < 1024; i++) |
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1443 out[i] = buf[i-512] + ac->add_bias; |
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1444 } |
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1445 } |
7578 | 1446 |
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1447 // buffer update |
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1448 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
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1449 for (i = 0; i < 64; i++) |
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1450 saved[i] = temp[64 + i] - ac->add_bias; |
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1451 ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); |
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1452 ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); |
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1453 ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); |
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1454 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
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1455 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
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|
1456 memcpy( saved, buf + 512, 448 * sizeof(float)); |
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1457 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
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1458 } else { // LONG_STOP or ONLY_LONG |
7663
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1459 memcpy( saved, buf + 512, 512 * sizeof(float)); |
7578 | 1460 } |
1461 } | |
1462 | |
7539 | 1463 /** |
7523
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1464 * Apply dependent channel coupling (applied before IMDCT). |
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1465 * |
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1466 * @param index index into coupling gain array |
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1467 */ |
8006
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1468 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
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1469 IndividualChannelStream * ics = &cce->ch[0].ics; |
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1470 const uint16_t * offsets = ics->swb_offset; |
8006
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1471 float * dest = target->coeffs; |
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1472 const float * src = cce->ch[0].coeffs; |
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1473 int g, i, group, k, idx = 0; |
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1474 if(ac->m4ac.object_type == AOT_AAC_LTP) { |
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1475 av_log(ac->avccontext, AV_LOG_ERROR, |
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1476 "Dependent coupling is not supported together with LTP\n"); |
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1477 return; |
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1478 } |
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1479 for (g = 0; g < ics->num_window_groups; g++) { |
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1480 for (i = 0; i < ics->max_sfb; i++, idx++) { |
8006
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|
1481 if (cce->ch[0].band_type[idx] != ZERO_BT) { |
9143 | 1482 const float gain = cce->coup.gain[index][idx]; |
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1483 for (group = 0; group < ics->group_len[g]; group++) { |
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1484 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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1485 // XXX dsputil-ize |
9143 | 1486 dest[group*128+k] += gain * src[group*128+k]; |
7523
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1487 } |
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1488 } |
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1489 } |
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1490 } |
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1491 dest += ics->group_len[g]*128; |
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1492 src += ics->group_len[g]*128; |
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1493 } |
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1494 } |
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1495 |
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1496 /** |
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1497 * Apply independent channel coupling (applied after IMDCT). |
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1498 * |
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1499 * @param index index into coupling gain array |
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1500 */ |
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1501 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
7523
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1502 int i; |
9051
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1503 const float gain = cce->coup.gain[index][0]; |
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1504 const float bias = ac->add_bias; |
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1505 const float* src = cce->ch[0].ret; |
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1506 float* dest = target->ret; |
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1507 |
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1508 for (i = 0; i < 1024; i++) |
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1509 dest[i] += gain * (src[i] - bias); |
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1510 } |
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1511 |
7581 | 1512 /** |
1513 * channel coupling transformation interface | |
1514 * | |
1515 * @param index index into coupling gain array | |
1516 * @param apply_coupling_method pointer to (in)dependent coupling function | |
1517 */ | |
1518 static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, | |
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1519 enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, |
8006
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1520 void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index)) |
7581 | 1521 { |
8004
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1522 int i, c; |
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1523 |
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1524 for (i = 0; i < MAX_ELEM_ID; i++) { |
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1525 ChannelElement *cce = ac->che[TYPE_CCE][i]; |
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1526 int index = 0; |
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1527 |
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1528 if (cce && cce->coup.coupling_point == coupling_point) { |
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1529 ChannelCoupling * coup = &cce->coup; |
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1530 |
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1531 for (c = 0; c <= coup->num_coupled; c++) { |
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1532 if (coup->type[c] == type && coup->id_select[c] == elem_id) { |
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1533 if (coup->ch_select[c] != 1) { |
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1534 apply_coupling_method(ac, &cc->ch[0], cce, index); |
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1535 if (coup->ch_select[c] != 0) |
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1536 index++; |
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1537 } |
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1538 if (coup->ch_select[c] != 2) |
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1539 apply_coupling_method(ac, &cc->ch[1], cce, index++); |
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1540 } else |
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1541 index += 1 + (coup->ch_select[c] == 3); |
7581 | 1542 } |
1543 } | |
1544 } | |
1545 } | |
1546 | |
1547 /** | |
1548 * Convert spectral data to float samples, applying all supported tools as appropriate. | |
1549 */ | |
1550 static void spectral_to_sample(AACContext * ac) { | |
8502 | 1551 int i, type; |
8004
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1552 for(type = 3; type >= 0; type--) { |
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1553 for (i = 0; i < MAX_ELEM_ID; i++) { |
7581 | 1554 ChannelElement *che = ac->che[type][i]; |
1555 if(che) { | |
8004
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1556 if(type <= TYPE_CPE) |
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1557 apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); |
7581 | 1558 if(che->ch[0].tns.present) |
1559 apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); | |
1560 if(che->ch[1].tns.present) | |
1561 apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); | |
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1562 if(type <= TYPE_CPE) |
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1563 apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); |
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1564 if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) |
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1565 imdct_and_windowing(ac, &che->ch[0]); |
7581 | 1566 if(type == TYPE_CPE) |
1567 imdct_and_windowing(ac, &che->ch[1]); | |
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1568 if(type <= TYPE_CCE) |
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1569 apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); |
7578 | 1570 } |
1571 } | |
1572 } | |
1573 } | |
1574 | |
8549
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1575 static int parse_adts_frame_header(AACContext * ac, GetBitContext * gb) { |
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1576 |
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1577 int size; |
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1578 AACADTSHeaderInfo hdr_info; |
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1579 |
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1580 size = ff_aac_parse_header(gb, &hdr_info); |
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1581 if (size > 0) { |
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1582 if (hdr_info.chan_config) |
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1583 ac->m4ac.chan_config = hdr_info.chan_config; |
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1584 ac->m4ac.sample_rate = hdr_info.sample_rate; |
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1585 ac->m4ac.sampling_index = hdr_info.sampling_index; |
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1586 ac->m4ac.object_type = hdr_info.object_type; |
9151 | 1587 if (hdr_info.num_aac_frames == 1) { |
1588 if (!hdr_info.crc_absent) | |
1589 skip_bits(gb, 16); | |
1590 } else { | |
1591 ff_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0); | |
1592 return -1; | |
1593 } | |
9150
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1594 } |
8549
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1595 return size; |
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1596 } |
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1597 |
9355
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|
1598 static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, AVPacket *avpkt) { |
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1599 const uint8_t *buf = avpkt->data; |
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1600 int buf_size = avpkt->size; |
7578 | 1601 AACContext * ac = avccontext->priv_data; |
9149
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1602 ChannelElement * che = NULL; |
7578 | 1603 GetBitContext gb; |
1604 enum RawDataBlockType elem_type; | |
1605 int err, elem_id, data_size_tmp; | |
1606 | |
1607 init_get_bits(&gb, buf, buf_size*8); | |
1608 | |
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1609 if (show_bits(&gb, 12) == 0xfff) { |
9510 | 1610 if (parse_adts_frame_header(ac, &gb) < 0) { |
8549
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1611 av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); |
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1612 return -1; |
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1613 } |
8812 | 1614 if (ac->m4ac.sampling_index > 12) { |
8789 | 1615 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
1616 return -1; | |
1617 } | |
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1618 } |
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1619 |
7578 | 1620 // parse |
1621 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { | |
1622 elem_id = get_bits(&gb, 4); | |
1623 | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
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changeset
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1624 if(elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) { |
8814 | 1625 av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); |
1626 return -1; | |
7578 | 1627 } |
1628 | |
1629 switch (elem_type) { | |
1630 | |
1631 case TYPE_SCE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
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1632 err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
7578 | 1633 break; |
1634 | |
1635 case TYPE_CPE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
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1636 err = decode_cpe(ac, &gb, che); |
7578 | 1637 break; |
1638 | |
1639 case TYPE_CCE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
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1640 err = decode_cce(ac, &gb, che); |
7578 | 1641 break; |
1642 | |
1643 case TYPE_LFE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
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1644 err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
7578 | 1645 break; |
1646 | |
1647 case TYPE_DSE: | |
1648 skip_data_stream_element(&gb); | |
1649 err = 0; | |
1650 break; | |
1651 | |
1652 case TYPE_PCE: | |
1653 { | |
1654 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
1655 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
1656 if((err = decode_pce(ac, new_che_pos, &gb))) | |
1657 break; | |
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955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
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1658 err = output_configure(ac, ac->che_pos, new_che_pos, 0); |
7578 | 1659 break; |
1660 } | |
1661 | |
1662 case TYPE_FIL: | |
1663 if (elem_id == 15) | |
1664 elem_id += get_bits(&gb, 8) - 1; | |
1665 while (elem_id > 0) | |
1666 elem_id -= decode_extension_payload(ac, &gb, elem_id); | |
1667 err = 0; /* FIXME */ | |
1668 break; | |
1669 | |
1670 default: | |
1671 err = -1; /* should not happen, but keeps compiler happy */ | |
1672 break; | |
1673 } | |
1674 | |
1675 if(err) | |
1676 return err; | |
1677 } | |
1678 | |
1679 spectral_to_sample(ac); | |
1680 | |
7539 | 1681 if (!ac->is_saved) { |
1682 ac->is_saved = 1; | |
1683 *data_size = 0; | |
7580 | 1684 return buf_size; |
7539 | 1685 } |
1686 | |
1687 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
1688 if(*data_size < data_size_tmp) { | |
1689 av_log(avccontext, AV_LOG_ERROR, | |
1690 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
1691 *data_size, data_size_tmp); | |
1692 return -1; | |
1693 } | |
1694 *data_size = data_size_tmp; | |
1695 | |
1696 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
1697 | |
1698 return buf_size; | |
1699 } | |
1700 | |
7501 | 1701 static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
1702 AACContext * ac = avccontext->priv_data; | |
7540 | 1703 int i, type; |
7501 | 1704 |
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a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
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1705 for (i = 0; i < MAX_ELEM_ID; i++) { |
7540 | 1706 for(type = 0; type < 4; type++) |
1707 av_freep(&ac->che[type][i]); | |
7501 | 1708 } |
1709 | |
1710 ff_mdct_end(&ac->mdct); | |
1711 ff_mdct_end(&ac->mdct_small); | |
1712 return 0 ; | |
1713 } | |
1714 | |
1715 AVCodec aac_decoder = { | |
1716 "aac", | |
1717 CODEC_TYPE_AUDIO, | |
1718 CODEC_ID_AAC, | |
1719 sizeof(AACContext), | |
1720 aac_decode_init, | |
1721 NULL, | |
1722 aac_decode_close, | |
1723 aac_decode_frame, | |
1724 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
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1725 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7501 | 1726 }; |