Mercurial > libavcodec.hg
annotate aac.c @ 9236:7ec90057127a libavcodec
10l: don't check against current layout until after validating ch_mode.
should have been in last commit.
author | jbr |
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date | Sun, 22 Mar 2009 21:13:10 +0000 |
parents | 4172c9ef331a |
children | 3fc2f6c6b684 |
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" |
7501 | 81 #include "bitstream.h" |
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 if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]))) | |
166 return 0; /* no change */ | |
167 | |
168 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
169 | |
170 /* Allocate or free elements depending on if they are in the | |
171 * current program configuration. | |
172 * | |
173 * Set up default 1:1 output mapping. | |
174 * | |
175 * For a 5.1 stream the output order will be: | |
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176 * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] |
7578 | 177 */ |
178 | |
179 for(i = 0; i < MAX_ELEM_ID; i++) { | |
180 for(type = 0; type < 4; type++) { | |
181 if(che_pos[type][i]) { | |
182 if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) | |
183 return AVERROR(ENOMEM); | |
184 if(type != TYPE_CCE) { | |
185 ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; | |
186 if(type == TYPE_CPE) { | |
187 ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; | |
188 } | |
189 } | |
190 } else | |
191 av_freep(&ac->che[type][i]); | |
192 } | |
193 } | |
194 | |
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195 if (channel_config) { |
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196 memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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197 ac->tags_mapped = 0; |
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198 } else { |
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199 memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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200 ac->tags_mapped = 4*MAX_ELEM_ID; |
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201 } |
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202 |
7578 | 203 avctx->channels = channels; |
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204 |
7578 | 205 return 0; |
206 } | |
207 | |
208 /** | |
7539 | 209 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
210 * | |
211 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
212 * @param sce_map mono (Single Channel Element) map | |
213 * @param type speaker type/position for these channels | |
214 */ | |
215 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
216 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { | |
217 while(n--) { | |
218 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map | |
219 map[get_bits(gb, 4)] = type; | |
220 } | |
221 } | |
222 | |
223 /** | |
224 * Decode program configuration element; reference: table 4.2. | |
225 * | |
226 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
227 * | |
228 * @return Returns error status. 0 - OK, !0 - error | |
229 */ | |
230 static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
231 GetBitContext * gb) { | |
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232 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index; |
7539 | 233 |
234 skip_bits(gb, 2); // object_type | |
235 | |
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236 sampling_index = get_bits(gb, 4); |
8812 | 237 if(sampling_index > 12) { |
7539 | 238 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
239 return -1; | |
240 } | |
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241 ac->m4ac.sampling_index = sampling_index; |
7539 | 242 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; |
7501 | 243 num_front = get_bits(gb, 4); |
244 num_side = get_bits(gb, 4); | |
245 num_back = get_bits(gb, 4); | |
246 num_lfe = get_bits(gb, 2); | |
247 num_assoc_data = get_bits(gb, 3); | |
248 num_cc = get_bits(gb, 4); | |
249 | |
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250 if (get_bits1(gb)) |
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251 skip_bits(gb, 4); // mono_mixdown_tag |
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252 if (get_bits1(gb)) |
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253 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 254 |
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255 if (get_bits1(gb)) |
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256 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 257 |
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258 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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259 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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260 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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261 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 262 |
263 skip_bits_long(gb, 4 * num_assoc_data); | |
264 | |
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265 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 266 |
267 align_get_bits(gb); | |
268 | |
269 /* comment field, first byte is length */ | |
270 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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271 return 0; |
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272 } |
7501 | 273 |
7539 | 274 /** |
275 * Set up channel positions based on a default channel configuration | |
276 * as specified in table 1.17. | |
277 * | |
278 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
279 * | |
280 * @return Returns error status. 0 - OK, !0 - error | |
281 */ | |
282 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
283 int channel_config) | |
284 { | |
285 if(channel_config < 1 || channel_config > 7) { | |
286 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", | |
287 channel_config); | |
288 return -1; | |
289 } | |
290 | |
291 /* default channel configurations: | |
292 * | |
293 * 1ch : front center (mono) | |
294 * 2ch : L + R (stereo) | |
295 * 3ch : front center + L + R | |
296 * 4ch : front center + L + R + back center | |
297 * 5ch : front center + L + R + back stereo | |
298 * 6ch : front center + L + R + back stereo + LFE | |
299 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
300 */ | |
301 | |
302 if(channel_config != 2) | |
303 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) | |
304 if(channel_config > 1) | |
305 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) | |
306 if(channel_config == 4) | |
307 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center | |
308 if(channel_config > 4) | |
309 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] | |
310 = AAC_CHANNEL_BACK; // back stereo | |
311 if(channel_config > 5) | |
312 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE | |
313 if(channel_config == 7) | |
314 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right | |
315 | |
316 return 0; | |
317 } | |
318 | |
7578 | 319 /** |
320 * Decode GA "General Audio" specific configuration; reference: table 4.1. | |
321 * | |
322 * @return Returns error status. 0 - OK, !0 - error | |
323 */ | |
324 static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { | |
325 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
326 int extension_flag, ret; | |
327 | |
328 if(get_bits1(gb)) { // frameLengthFlag | |
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329 ff_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); |
7539 | 330 return -1; |
331 } | |
332 | |
333 if (get_bits1(gb)) // dependsOnCoreCoder | |
334 skip_bits(gb, 14); // coreCoderDelay | |
335 extension_flag = get_bits1(gb); | |
336 | |
337 if(ac->m4ac.object_type == AOT_AAC_SCALABLE || | |
338 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
339 skip_bits(gb, 3); // layerNr | |
340 | |
341 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
342 if (channel_config == 0) { | |
343 skip_bits(gb, 4); // element_instance_tag | |
344 if((ret = decode_pce(ac, new_che_pos, gb))) | |
345 return ret; | |
346 } else { | |
347 if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) | |
348 return ret; | |
349 } | |
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350 if((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config))) |
7539 | 351 return ret; |
352 | |
353 if (extension_flag) { | |
354 switch (ac->m4ac.object_type) { | |
355 case AOT_ER_BSAC: | |
356 skip_bits(gb, 5); // numOfSubFrame | |
357 skip_bits(gb, 11); // layer_length | |
358 break; | |
359 case AOT_ER_AAC_LC: | |
360 case AOT_ER_AAC_LTP: | |
361 case AOT_ER_AAC_SCALABLE: | |
362 case AOT_ER_AAC_LD: | |
363 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
364 * aacScalefactorDataResilienceFlag | |
365 * aacSpectralDataResilienceFlag | |
366 */ | |
367 break; | |
368 } | |
369 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
370 } | |
371 return 0; | |
372 } | |
373 | |
374 /** | |
375 * Decode audio specific configuration; reference: table 1.13. | |
376 * | |
377 * @param data pointer to AVCodecContext extradata | |
378 * @param data_size size of AVCCodecContext extradata | |
379 * | |
380 * @return Returns error status. 0 - OK, !0 - error | |
381 */ | |
382 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { | |
383 GetBitContext gb; | |
384 int i; | |
385 | |
386 init_get_bits(&gb, data, data_size * 8); | |
387 | |
388 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) | |
389 return -1; | |
8812 | 390 if(ac->m4ac.sampling_index > 12) { |
7539 | 391 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
392 return -1; | |
393 } | |
394 | |
395 skip_bits_long(&gb, i); | |
396 | |
397 switch (ac->m4ac.object_type) { | |
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398 case AOT_AAC_MAIN: |
7539 | 399 case AOT_AAC_LC: |
400 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
401 return -1; | |
402 break; | |
403 default: | |
404 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
405 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
406 return -1; | |
407 } | |
408 return 0; | |
409 } | |
410 | |
7578 | 411 /** |
412 * linear congruential pseudorandom number generator | |
413 * | |
414 * @param previous_val pointer to the current state of the generator | |
415 * | |
416 * @return Returns a 32-bit pseudorandom integer | |
417 */ | |
418 static av_always_inline int lcg_random(int previous_val) { | |
419 return previous_val * 1664525 + 1013904223; | |
420 } | |
421 | |
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422 static void reset_predict_state(PredictorState * ps) { |
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423 ps->r0 = 0.0f; |
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424 ps->r1 = 0.0f; |
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425 ps->cor0 = 0.0f; |
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426 ps->cor1 = 0.0f; |
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427 ps->var0 = 1.0f; |
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428 ps->var1 = 1.0f; |
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429 } |
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430 |
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431 static void reset_all_predictors(PredictorState * ps) { |
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432 int i; |
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433 for (i = 0; i < MAX_PREDICTORS; i++) |
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434 reset_predict_state(&ps[i]); |
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435 } |
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436 |
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437 static void reset_predictor_group(PredictorState * ps, int group_num) { |
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438 int i; |
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439 for (i = group_num-1; i < MAX_PREDICTORS; i+=30) |
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440 reset_predict_state(&ps[i]); |
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441 } |
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442 |
7501 | 443 static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
444 AACContext * ac = avccontext->priv_data; | |
445 int i; | |
446 | |
447 ac->avccontext = avccontext; | |
448 | |
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449 if (avccontext->extradata_size > 0) { |
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450 if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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451 return -1; |
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452 avccontext->sample_rate = ac->m4ac.sample_rate; |
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453 } else if (avccontext->channels > 0) { |
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454 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
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455 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
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456 if(set_default_channel_config(ac, new_che_pos, avccontext->channels - (avccontext->channels == 8))) |
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457 return -1; |
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458 if(output_configure(ac, ac->che_pos, new_che_pos, 1)) |
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459 return -1; |
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460 ac->m4ac.sample_rate = avccontext->sample_rate; |
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461 } else { |
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462 ff_log_missing_feature(ac->avccontext, "Implicit channel configuration is", 0); |
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463 return -1; |
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464 } |
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465 |
7539 | 466 avccontext->sample_fmt = SAMPLE_FMT_S16; |
7501 | 467 avccontext->frame_size = 1024; |
468 | |
469 AAC_INIT_VLC_STATIC( 0, 144); | |
470 AAC_INIT_VLC_STATIC( 1, 114); | |
471 AAC_INIT_VLC_STATIC( 2, 188); | |
472 AAC_INIT_VLC_STATIC( 3, 180); | |
473 AAC_INIT_VLC_STATIC( 4, 172); | |
474 AAC_INIT_VLC_STATIC( 5, 140); | |
475 AAC_INIT_VLC_STATIC( 6, 168); | |
476 AAC_INIT_VLC_STATIC( 7, 114); | |
477 AAC_INIT_VLC_STATIC( 8, 262); | |
478 AAC_INIT_VLC_STATIC( 9, 248); | |
479 AAC_INIT_VLC_STATIC(10, 384); | |
480 | |
481 dsputil_init(&ac->dsp, avccontext); | |
482 | |
7539 | 483 ac->random_state = 0x1f2e3d4c; |
484 | |
7501 | 485 // -1024 - Compensate wrong IMDCT method. |
486 // 32768 - Required to scale values to the correct range for the bias method | |
487 // for float to int16 conversion. | |
488 | |
489 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { | |
490 ac->add_bias = 385.0f; | |
491 ac->sf_scale = 1. / (-1024. * 32768.); | |
492 ac->sf_offset = 0; | |
493 } else { | |
494 ac->add_bias = 0.0f; | |
495 ac->sf_scale = 1. / -1024.; | |
496 ac->sf_offset = 60; | |
497 } | |
498 | |
8590 | 499 #if !CONFIG_HARDCODED_TABLES |
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500 for (i = 0; i < 428; i++) |
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501 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
7501 | 502 #endif /* CONFIG_HARDCODED_TABLES */ |
503 | |
8042 | 504 INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), |
7501 | 505 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), |
506 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
507 352); | |
508 | |
509 ff_mdct_init(&ac->mdct, 11, 1); | |
510 ff_mdct_init(&ac->mdct_small, 8, 1); | |
7581 | 511 // window initialization |
512 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
513 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
514 ff_sine_window_init(ff_sine_1024, 1024); | |
515 ff_sine_window_init(ff_sine_128, 128); | |
516 | |
7501 | 517 return 0; |
518 } | |
519 | |
7539 | 520 /** |
521 * Skip data_stream_element; reference: table 4.10. | |
522 */ | |
523 static void skip_data_stream_element(GetBitContext * gb) { | |
7501 | 524 int byte_align = get_bits1(gb); |
525 int count = get_bits(gb, 8); | |
526 if (count == 255) | |
527 count += get_bits(gb, 8); | |
528 if (byte_align) | |
529 align_get_bits(gb); | |
530 skip_bits_long(gb, 8 * count); | |
531 } | |
532 | |
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533 static int decode_prediction(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb) { |
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534 int sfb; |
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535 if (get_bits1(gb)) { |
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536 ics->predictor_reset_group = get_bits(gb, 5); |
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537 if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { |
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538 av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); |
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539 return -1; |
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540 } |
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541 } |
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542 for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { |
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543 ics->prediction_used[sfb] = get_bits1(gb); |
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544 } |
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545 return 0; |
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546 } |
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547 |
7501 | 548 /** |
7539 | 549 * Decode Individual Channel Stream info; reference: table 4.6. |
550 * | |
551 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
552 */ | |
553 static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { | |
554 if (get_bits1(gb)) { | |
555 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); | |
556 memset(ics, 0, sizeof(IndividualChannelStream)); | |
557 return -1; | |
558 } | |
559 ics->window_sequence[1] = ics->window_sequence[0]; | |
560 ics->window_sequence[0] = get_bits(gb, 2); | |
561 ics->use_kb_window[1] = ics->use_kb_window[0]; | |
562 ics->use_kb_window[0] = get_bits1(gb); | |
563 ics->num_window_groups = 1; | |
564 ics->group_len[0] = 1; | |
7581 | 565 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
566 int i; | |
567 ics->max_sfb = get_bits(gb, 4); | |
568 for (i = 0; i < 7; i++) { | |
569 if (get_bits1(gb)) { | |
570 ics->group_len[ics->num_window_groups-1]++; | |
571 } else { | |
572 ics->num_window_groups++; | |
573 ics->group_len[ics->num_window_groups-1] = 1; | |
574 } | |
575 } | |
576 ics->num_windows = 8; | |
577 ics->swb_offset = swb_offset_128[ac->m4ac.sampling_index]; | |
578 ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; | |
579 ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index]; | |
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580 ics->predictor_present = 0; |
7581 | 581 } else { |
582 ics->max_sfb = get_bits(gb, 6); | |
583 ics->num_windows = 1; | |
584 ics->swb_offset = swb_offset_1024[ac->m4ac.sampling_index]; | |
585 ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; | |
586 ics->tns_max_bands = tns_max_bands_1024[ac->m4ac.sampling_index]; | |
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587 ics->predictor_present = get_bits1(gb); |
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588 ics->predictor_reset_group = 0; |
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589 if (ics->predictor_present) { |
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590 if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
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591 if (decode_prediction(ac, ics, gb)) { |
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592 memset(ics, 0, sizeof(IndividualChannelStream)); |
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593 return -1; |
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594 } |
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595 } else if (ac->m4ac.object_type == AOT_AAC_LC) { |
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596 av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); |
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597 memset(ics, 0, sizeof(IndividualChannelStream)); |
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598 return -1; |
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599 } else { |
8281
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600 ff_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); |
8200 | 601 memset(ics, 0, sizeof(IndividualChannelStream)); |
602 return -1; | |
8199
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603 } |
7578 | 604 } |
605 } | |
606 | |
607 if(ics->max_sfb > ics->num_swb) { | |
608 av_log(ac->avccontext, AV_LOG_ERROR, | |
609 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", | |
610 ics->max_sfb, ics->num_swb); | |
611 memset(ics, 0, sizeof(IndividualChannelStream)); | |
612 return -1; | |
613 } | |
614 | |
7539 | 615 return 0; |
616 } | |
617 | |
618 /** | |
619 * Decode band types (section_data payload); reference: table 4.46. | |
620 * | |
621 * @param band_type array of the used band type | |
622 * @param band_type_run_end array of the last scalefactor band of a band type run | |
623 * | |
624 * @return Returns error status. 0 - OK, !0 - error | |
625 */ | |
626 static int decode_band_types(AACContext * ac, enum BandType band_type[120], | |
7523
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627 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
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628 int g, idx = 0; |
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629 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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630 for (g = 0; g < ics->num_window_groups; g++) { |
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631 int k = 0; |
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632 while (k < ics->max_sfb) { |
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633 uint8_t sect_len = k; |
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634 int sect_len_incr; |
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635 int sect_band_type = get_bits(gb, 4); |
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636 if (sect_band_type == 12) { |
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637 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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638 return -1; |
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639 } |
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640 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
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641 sect_len += sect_len_incr; |
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642 sect_len += sect_len_incr; |
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643 if (sect_len > ics->max_sfb) { |
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644 av_log(ac->avccontext, AV_LOG_ERROR, |
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645 "Number of bands (%d) exceeds limit (%d).\n", |
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646 sect_len, ics->max_sfb); |
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647 return -1; |
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648 } |
7581 | 649 for (; k < sect_len; k++) { |
650 band_type [idx] = sect_band_type; | |
651 band_type_run_end[idx++] = sect_len; | |
652 } | |
7539 | 653 } |
654 } | |
655 return 0; | |
656 } | |
7523
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657 |
7539 | 658 /** |
659 * Decode scalefactors; reference: table 4.47. | |
7523
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660 * |
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661 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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662 * @param band_type array of the used band type |
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663 * @param band_type_run_end array of the last scalefactor band of a band type run |
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664 * @param sf array of scalefactors or intensity stereo positions |
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665 * |
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666 * @return Returns error status. 0 - OK, !0 - error |
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667 */ |
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668 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
7540 | 669 unsigned int global_gain, IndividualChannelStream * ics, |
7523
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670 enum BandType band_type[120], int band_type_run_end[120]) { |
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671 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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672 int g, i, idx = 0; |
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673 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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674 int noise_flag = 1; |
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675 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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676 for (g = 0; g < ics->num_window_groups; g++) { |
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677 for (i = 0; i < ics->max_sfb;) { |
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678 int run_end = band_type_run_end[idx]; |
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679 if (band_type[idx] == ZERO_BT) { |
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680 for(; i < run_end; i++, idx++) |
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681 sf[idx] = 0.; |
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682 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
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683 for(; i < run_end; i++, idx++) { |
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684 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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685 if(offset[2] > 255U) { |
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686 av_log(ac->avccontext, AV_LOG_ERROR, |
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687 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
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688 return -1; |
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689 } |
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690 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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691 } |
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692 }else if(band_type[idx] == NOISE_BT) { |
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693 for(; i < run_end; i++, idx++) { |
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694 if(noise_flag-- > 0) |
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695 offset[1] += get_bits(gb, 9) - 256; |
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696 else |
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697 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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698 if(offset[1] > 255U) { |
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699 av_log(ac->avccontext, AV_LOG_ERROR, |
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700 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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701 return -1; |
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702 } |
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7baa25290231
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703 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset + 100]; |
7523
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704 } |
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705 }else { |
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706 for(; i < run_end; i++, idx++) { |
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707 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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708 if(offset[0] > 255U) { |
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709 av_log(ac->avccontext, AV_LOG_ERROR, |
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710 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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711 return -1; |
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712 } |
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713 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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714 } |
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715 } |
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716 } |
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717 } |
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718 return 0; |
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719 } |
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720 |
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721 /** |
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722 * Decode pulse data; reference: table 4.7. |
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723 */ |
7871
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724 static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) { |
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725 int i, pulse_swb; |
7523
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726 pulse->num_pulse = get_bits(gb, 2) + 1; |
7871
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727 pulse_swb = get_bits(gb, 6); |
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728 if (pulse_swb >= num_swb) |
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729 return -1; |
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730 pulse->pos[0] = swb_offset[pulse_swb]; |
7856
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731 pulse->pos[0] += get_bits(gb, 5); |
7871
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732 if (pulse->pos[0] > 1023) |
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733 return -1; |
7580 | 734 pulse->amp[0] = get_bits(gb, 4); |
735 for (i = 1; i < pulse->num_pulse; i++) { | |
736 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; | |
7871
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737 if (pulse->pos[i] > 1023) |
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|
738 return -1; |
7580 | 739 pulse->amp[i] = get_bits(gb, 4); |
7523
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
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diff
changeset
|
740 } |
7871
8277c41b7160
Validate pulse position and error out if an invalid position is encountered.
superdump
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7858
diff
changeset
|
741 return 0; |
7523
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
parents:
7501
diff
changeset
|
742 } |
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
parents:
7501
diff
changeset
|
743 |
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
parents:
7501
diff
changeset
|
744 /** |
7609 | 745 * Decode Temporal Noise Shaping data; reference: table 4.48. |
746 * | |
747 * @return Returns error status. 0 - OK, !0 - error | |
748 */ | |
749 static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns, | |
750 GetBitContext * gb, const IndividualChannelStream * ics) { | |
751 int w, filt, i, coef_len, coef_res, coef_compress; | |
752 const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; | |
753 const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; | |
754 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
|
755 if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
7609 | 756 coef_res = get_bits1(gb); |
757 | |
7638 | 758 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
759 int tmp2_idx; | |
760 tns->length[w][filt] = get_bits(gb, 6 - 2*is8); | |
7609 | 761 |
7638 | 762 if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) { |
763 av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", | |
764 tns->order[w][filt], tns_max_order); | |
765 tns->order[w][filt] = 0; | |
766 return -1; | |
767 } | |
7850
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diff
changeset
|
768 if (tns->order[w][filt]) { |
7851 | 769 tns->direction[w][filt] = get_bits1(gb); |
770 coef_compress = get_bits1(gb); | |
771 coef_len = coef_res + 3 - coef_compress; | |
772 tmp2_idx = 2*coef_compress + coef_res; | |
7638 | 773 |
7851 | 774 for (i = 0; i < tns->order[w][filt]; i++) |
775 tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; | |
7850
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Only read some of TNS bitstream data in the case that the TNS filter order is
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parents:
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diff
changeset
|
776 } |
7609 | 777 } |
7637
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Work around GCC variable may be used uninitialised warning
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parents:
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diff
changeset
|
778 } |
7609 | 779 } |
780 return 0; | |
781 } | |
782 | |
783 /** | |
7539 | 784 * Decode Mid/Side data; reference: table 4.54. |
785 * | |
786 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
787 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
788 * [3] reserved for scalable AAC | |
789 */ | |
790 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, | |
791 int ms_present) { | |
7578 | 792 int idx; |
793 if (ms_present == 1) { | |
794 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) | |
795 cpe->ms_mask[idx] = get_bits1(gb); | |
796 } else if (ms_present == 2) { | |
797 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); | |
798 } | |
799 } | |
7539 | 800 |
801 /** | |
7581 | 802 * Decode spectral data; reference: table 4.50. |
803 * Dequantize and scale spectral data; reference: 4.6.3.3. | |
804 * | |
805 * @param coef array of dequantized, scaled spectral data | |
806 * @param sf array of scalefactors or intensity stereo positions | |
807 * @param pulse_present set if pulses are present | |
808 * @param pulse pointer to pulse data struct | |
809 * @param band_type array of the used band type | |
810 * | |
811 * @return Returns error status. 0 - OK, !0 - error | |
812 */ | |
813 static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120], | |
814 int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) { | |
815 int i, k, g, idx = 0; | |
816 const int c = 1024/ics->num_windows; | |
817 const uint16_t * offsets = ics->swb_offset; | |
818 float *coef_base = coef; | |
8284 | 819 static const float sign_lookup[] = { 1.0f, -1.0f }; |
7581 | 820 |
821 for (g = 0; g < ics->num_windows; g++) | |
822 memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb])); | |
823 | |
824 for (g = 0; g < ics->num_window_groups; g++) { | |
825 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
826 const int cur_band_type = band_type[idx]; | |
827 const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; | |
828 const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); | |
829 int group; | |
8413
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parents:
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diff
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|
830 if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) { |
7581 | 831 for (group = 0; group < ics->group_len[g]; group++) { |
832 memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); | |
833 } | |
834 }else if (cur_band_type == NOISE_BT) { | |
835 for (group = 0; group < ics->group_len[g]; group++) { | |
8109
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Fix perceptual noise shaping scaling including an oversight of the offset for
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parents:
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diff
changeset
|
836 float scale; |
7baa25290231
Fix perceptual noise shaping scaling including an oversight of the offset for
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parents:
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diff
changeset
|
837 float band_energy = 0; |
7581 | 838 for (k = offsets[i]; k < offsets[i+1]; k++) { |
839 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
superdump
parents:
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diff
changeset
|
840 coef[group*128+k] = ac->random_state; |
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Fix perceptual noise shaping scaling including an oversight of the offset for
superdump
parents:
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diff
changeset
|
841 band_energy += coef[group*128+k]*coef[group*128+k]; |
7baa25290231
Fix perceptual noise shaping scaling including an oversight of the offset for
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diff
changeset
|
842 } |
7baa25290231
Fix perceptual noise shaping scaling including an oversight of the offset for
superdump
parents:
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diff
changeset
|
843 scale = sf[idx] / sqrtf(band_energy); |
7baa25290231
Fix perceptual noise shaping scaling including an oversight of the offset for
superdump
parents:
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diff
changeset
|
844 for (k = offsets[i]; k < offsets[i+1]; k++) { |
7baa25290231
Fix perceptual noise shaping scaling including an oversight of the offset for
superdump
parents:
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diff
changeset
|
845 coef[group*128+k] *= scale; |
7581 | 846 } |
847 } | |
8413
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diff
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|
848 }else { |
7581 | 849 for (group = 0; group < ics->group_len[g]; group++) { |
850 for (k = offsets[i]; k < offsets[i+1]; k += dim) { | |
851 const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); | |
852 const int coef_tmp_idx = (group << 7) + k; | |
853 const float *vq_ptr; | |
854 int j; | |
855 if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { | |
856 av_log(ac->avccontext, AV_LOG_ERROR, | |
857 "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", | |
858 cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); | |
859 return -1; | |
860 } | |
861 vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; | |
862 if (is_cb_unsigned) { | |
8284 | 863 if (vq_ptr[0]) coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)]; |
864 if (vq_ptr[1]) coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)]; | |
8283
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
865 if (dim == 4) { |
8284 | 866 if (vq_ptr[2]) coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)]; |
867 if (vq_ptr[3]) coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)]; | |
8283
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
868 } |
9053 | 869 if (cur_band_type == ESC_BT) { |
870 for (j = 0; j < 2; j++) { | |
871 if (vq_ptr[j] == 64.0f) { | |
872 int n = 4; | |
873 /* The total length of escape_sequence must be < 22 bits according | |
874 to the specification (i.e. max is 11111111110xxxxxxxxxx). */ | |
875 while (get_bits1(gb) && n < 15) n++; | |
876 if(n == 15) { | |
877 av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); | |
878 return -1; | |
879 } | |
880 n = (1<<n) + get_bits(gb, n); | |
881 coef[coef_tmp_idx + j] *= cbrtf(n) * n; | |
882 }else | |
883 coef[coef_tmp_idx + j] *= vq_ptr[j]; | |
884 } | |
885 }else | |
886 { | |
887 coef[coef_tmp_idx ] *= vq_ptr[0]; | |
888 coef[coef_tmp_idx + 1] *= vq_ptr[1]; | |
889 if (dim == 4) { | |
890 coef[coef_tmp_idx + 2] *= vq_ptr[2]; | |
891 coef[coef_tmp_idx + 3] *= vq_ptr[3]; | |
892 } | |
7581 | 893 } |
9052
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
894 }else { |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
895 coef[coef_tmp_idx ] = vq_ptr[0]; |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
896 coef[coef_tmp_idx + 1] = vq_ptr[1]; |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
897 if (dim == 4) { |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
898 coef[coef_tmp_idx + 2] = vq_ptr[2]; |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
899 coef[coef_tmp_idx + 3] = vq_ptr[3]; |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
900 } |
54973ad4b7cf
Refactor AAC unsigned band type decoding for a modest speedup.
alexc
parents:
9051
diff
changeset
|
901 } |
8283
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
902 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
|
903 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
|
904 if (dim == 4) { |
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
parents:
8282
diff
changeset
|
905 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
|
906 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
|
907 } |
7581 | 908 } |
909 } | |
910 } | |
911 } | |
912 coef += ics->group_len[g]<<7; | |
913 } | |
914 | |
915 if (pulse_present) { | |
7857
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
changeset
|
916 idx = 0; |
7581 | 917 for(i = 0; i < pulse->num_pulse; i++){ |
918 float co = coef_base[ pulse->pos[i] ]; | |
7857
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
changeset
|
919 while(offsets[idx + 1] <= pulse->pos[i]) |
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
changeset
|
920 idx++; |
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
changeset
|
921 if (band_type[idx] != NOISE_BT && sf[idx]) { |
7858 | 922 float ico = -pulse->amp[i]; |
923 if (co) { | |
924 co /= sf[idx]; | |
925 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); | |
926 } | |
927 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
changeset
|
928 } |
7581 | 929 } |
930 } | |
931 return 0; | |
932 } | |
933 | |
8199
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
934 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
|
935 union float754 tmp; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
936 tmp.f = pf; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
937 tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
938 return tmp.f; |
8199
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
939 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
940 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
941 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
|
942 union float754 tmp; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
943 tmp.f = pf; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
944 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
|
945 return tmp.f; |
8199
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
946 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
947 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
948 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
|
949 union float754 pun; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
950 pun.f = pf; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
951 pun.i &= 0xFFFF0000U; |
4172c9ef331a
AAC: IEEE-754 type punning for 16-bit floating point rounding.
alexc
parents:
9151
diff
changeset
|
952 return pun.f; |
8199
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
953 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
954 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
955 static void predict(AACContext * ac, PredictorState * ps, float* coef, int output_enable) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
956 const float a = 0.953125; // 61.0/64 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
957 const float alpha = 0.90625; // 29.0/32 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
958 float e0, e1; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
959 float pv; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
960 float k1, k2; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
961 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
962 k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
963 k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
964 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
965 pv = flt16_round(k1 * ps->r0 + k2 * ps->r1); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
966 if (output_enable) |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
967 *coef += pv * ac->sf_scale; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
968 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
969 e0 = *coef / ac->sf_scale; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
970 e1 = e0 - k1 * ps->r0; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
971 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
superdump
parents:
8109
diff
changeset
|
972 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
|
973 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:
8109
diff
changeset
|
974 ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0); |
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975 ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0)); |
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976 |
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977 ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0)); |
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978 ps->r0 = flt16_trunc(a * e0); |
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|
979 } |
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980 |
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981 /** |
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982 * Apply AAC-Main style frequency domain prediction. |
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|
983 */ |
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984 static void apply_prediction(AACContext * ac, SingleChannelElement * sce) { |
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985 int sfb, k; |
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986 |
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987 if (!sce->ics.predictor_initialized) { |
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988 reset_all_predictors(sce->predictor_state); |
8199
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989 sce->ics.predictor_initialized = 1; |
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|
990 } |
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991 |
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992 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
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993 for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { |
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994 for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { |
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995 predict(ac, &sce->predictor_state[k], &sce->coeffs[k], |
8199
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996 sce->ics.predictor_present && sce->ics.prediction_used[sfb]); |
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997 } |
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998 } |
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999 if (sce->ics.predictor_reset_group) |
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1000 reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group); |
8199
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1001 } else |
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1002 reset_all_predictors(sce->predictor_state); |
8199
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1003 } |
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1004 |
7581 | 1005 /** |
7539 | 1006 * Decode an individual_channel_stream payload; reference: table 4.44. |
1007 * | |
1008 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
1009 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
1010 * | |
1011 * @return Returns error status. 0 - OK, !0 - error | |
1012 */ | |
1013 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { | |
1014 Pulse pulse; | |
1015 TemporalNoiseShaping * tns = &sce->tns; | |
1016 IndividualChannelStream * ics = &sce->ics; | |
1017 float * out = sce->coeffs; | |
1018 int global_gain, pulse_present = 0; | |
1019 | |
7580 | 1020 /* This assignment is to silence a GCC warning about the variable being used |
1021 * uninitialized when in fact it always is. | |
7539 | 1022 */ |
1023 pulse.num_pulse = 0; | |
1024 | |
1025 global_gain = get_bits(gb, 8); | |
1026 | |
1027 if (!common_window && !scale_flag) { | |
1028 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
1029 return -1; | |
1030 } | |
1031 | |
1032 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
1033 return -1; | |
1034 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
1035 return -1; | |
1036 | |
1037 pulse_present = 0; | |
1038 if (!scale_flag) { | |
1039 if ((pulse_present = get_bits1(gb))) { | |
1040 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
1041 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
1042 return -1; | |
1043 } | |
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1044 if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { |
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1045 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); |
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1046 return -1; |
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1047 } |
7539 | 1048 } |
1049 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
1050 return -1; | |
1051 if (get_bits1(gb)) { | |
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1052 ff_log_missing_feature(ac->avccontext, "SSR", 1); |
7539 | 1053 return -1; |
1054 } | |
1055 } | |
1056 | |
7580 | 1057 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
7539 | 1058 return -1; |
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1059 |
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1060 if(ac->m4ac.object_type == AOT_AAC_MAIN && !common_window) |
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1061 apply_prediction(ac, sce); |
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1062 |
7539 | 1063 return 0; |
1064 } | |
1065 | |
1066 /** | |
7581 | 1067 * Mid/Side stereo decoding; reference: 4.6.8.1.3. |
1068 */ | |
1069 static void apply_mid_side_stereo(ChannelElement * cpe) { | |
1070 const IndividualChannelStream * ics = &cpe->ch[0].ics; | |
1071 float *ch0 = cpe->ch[0].coeffs; | |
1072 float *ch1 = cpe->ch[1].coeffs; | |
1073 int g, i, k, group, idx = 0; | |
1074 const uint16_t * offsets = ics->swb_offset; | |
1075 for (g = 0; g < ics->num_window_groups; g++) { | |
1076 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
1077 if (cpe->ms_mask[idx] && | |
1078 cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { | |
1079 for (group = 0; group < ics->group_len[g]; group++) { | |
1080 for (k = offsets[i]; k < offsets[i+1]; k++) { | |
1081 float tmp = ch0[group*128 + k] - ch1[group*128 + k]; | |
1082 ch0[group*128 + k] += ch1[group*128 + k]; | |
1083 ch1[group*128 + k] = tmp; | |
1084 } | |
1085 } | |
1086 } | |
1087 } | |
1088 ch0 += ics->group_len[g]*128; | |
1089 ch1 += ics->group_len[g]*128; | |
1090 } | |
1091 } | |
1092 | |
1093 /** | |
1094 * intensity stereo decoding; reference: 4.6.8.2.3 | |
1095 * | |
1096 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
1097 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
1098 * [3] reserved for scalable AAC | |
1099 */ | |
1100 static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) { | |
1101 const IndividualChannelStream * ics = &cpe->ch[1].ics; | |
1102 SingleChannelElement * sce1 = &cpe->ch[1]; | |
1103 float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; | |
1104 const uint16_t * offsets = ics->swb_offset; | |
1105 int g, group, i, k, idx = 0; | |
1106 int c; | |
1107 float scale; | |
1108 for (g = 0; g < ics->num_window_groups; g++) { | |
1109 for (i = 0; i < ics->max_sfb;) { | |
1110 if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { | |
1111 const int bt_run_end = sce1->band_type_run_end[idx]; | |
1112 for (; i < bt_run_end; i++, idx++) { | |
1113 c = -1 + 2 * (sce1->band_type[idx] - 14); | |
1114 if (ms_present) | |
1115 c *= 1 - 2 * cpe->ms_mask[idx]; | |
1116 scale = c * sce1->sf[idx]; | |
1117 for (group = 0; group < ics->group_len[g]; group++) | |
1118 for (k = offsets[i]; k < offsets[i+1]; k++) | |
1119 coef1[group*128 + k] = scale * coef0[group*128 + k]; | |
1120 } | |
1121 } else { | |
1122 int bt_run_end = sce1->band_type_run_end[idx]; | |
1123 idx += bt_run_end - i; | |
1124 i = bt_run_end; | |
1125 } | |
1126 } | |
1127 coef0 += ics->group_len[g]*128; | |
1128 coef1 += ics->group_len[g]*128; | |
1129 } | |
1130 } | |
1131 | |
1132 /** | |
7539 | 1133 * Decode a channel_pair_element; reference: table 4.4. |
1134 * | |
1135 * @param elem_id Identifies the instance of a syntax element. | |
1136 * | |
1137 * @return Returns error status. 0 - OK, !0 - error | |
1138 */ | |
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1139 static int decode_cpe(AACContext * ac, GetBitContext * gb, ChannelElement * cpe) { |
7539 | 1140 int i, ret, common_window, ms_present = 0; |
1141 | |
1142 common_window = get_bits1(gb); | |
1143 if (common_window) { | |
1144 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
1145 return -1; | |
1146 i = cpe->ch[1].ics.use_kb_window[0]; | |
1147 cpe->ch[1].ics = cpe->ch[0].ics; | |
1148 cpe->ch[1].ics.use_kb_window[1] = i; | |
1149 ms_present = get_bits(gb, 2); | |
1150 if(ms_present == 3) { | |
1151 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); | |
1152 return -1; | |
1153 } else if(ms_present) | |
1154 decode_mid_side_stereo(cpe, gb, ms_present); | |
1155 } | |
1156 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
1157 return ret; | |
1158 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
1159 return ret; | |
1160 | |
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1161 if (common_window) { |
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|
1162 if (ms_present) |
8414 | 1163 apply_mid_side_stereo(cpe); |
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1164 if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
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1165 apply_prediction(ac, &cpe->ch[0]); |
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1166 apply_prediction(ac, &cpe->ch[1]); |
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1167 } |
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|
1168 } |
7539 | 1169 |
7580 | 1170 apply_intensity_stereo(cpe, ms_present); |
7539 | 1171 return 0; |
1172 } | |
1173 | |
7581 | 1174 /** |
1175 * Decode coupling_channel_element; reference: table 4.8. | |
1176 * | |
1177 * @param elem_id Identifies the instance of a syntax element. | |
1178 * | |
1179 * @return Returns error status. 0 - OK, !0 - error | |
1180 */ | |
1181 static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) { | |
1182 int num_gain = 0; | |
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1183 int c, g, sfb, ret; |
7581 | 1184 int sign; |
1185 float scale; | |
1186 SingleChannelElement * sce = &che->ch[0]; | |
1187 ChannelCoupling * coup = &che->coup; | |
1188 | |
7578 | 1189 coup->coupling_point = 2*get_bits1(gb); |
1190 coup->num_coupled = get_bits(gb, 3); | |
1191 for (c = 0; c <= coup->num_coupled; c++) { | |
1192 num_gain++; | |
1193 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; | |
1194 coup->id_select[c] = get_bits(gb, 4); | |
1195 if (coup->type[c] == TYPE_CPE) { | |
1196 coup->ch_select[c] = get_bits(gb, 2); | |
1197 if (coup->ch_select[c] == 3) | |
1198 num_gain++; | |
1199 } else | |
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1200 coup->ch_select[c] = 2; |
7578 | 1201 } |
1202 coup->coupling_point += get_bits1(gb); | |
1203 | |
1204 if (coup->coupling_point == 2) { | |
1205 av_log(ac->avccontext, AV_LOG_ERROR, | |
1206 "Independently switched CCE with 'invalid' domain signalled.\n"); | |
1207 memset(coup, 0, sizeof(ChannelCoupling)); | |
1208 return -1; | |
1209 } | |
1210 | |
1211 sign = get_bits(gb, 1); | |
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1212 scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); |
7578 | 1213 |
1214 if ((ret = decode_ics(ac, sce, gb, 0, 0))) | |
1215 return ret; | |
1216 | |
1217 for (c = 0; c < num_gain; c++) { | |
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1218 int idx = 0; |
7578 | 1219 int cge = 1; |
1220 int gain = 0; | |
1221 float gain_cache = 1.; | |
1222 if (c) { | |
1223 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); | |
1224 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; | |
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1225 gain_cache = pow(scale, -gain); |
7578 | 1226 } |
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1227 if (coup->coupling_point == AFTER_IMDCT) { |
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1228 coup->gain[c][0] = gain_cache; |
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1229 } else { |
8823 | 1230 for (g = 0; g < sce->ics.num_window_groups; g++) { |
1231 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { | |
1232 if (sce->band_type[idx] != ZERO_BT) { | |
1233 if (!cge) { | |
1234 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; | |
1235 if (t) { | |
1236 int s = 1; | |
1237 t = gain += t; | |
1238 if (sign) { | |
1239 s -= 2 * (t & 0x1); | |
1240 t >>= 1; | |
1241 } | |
1242 gain_cache = pow(scale, -t) * s; | |
7578 | 1243 } |
1244 } | |
8823 | 1245 coup->gain[c][idx] = gain_cache; |
7578 | 1246 } |
1247 } | |
7890 | 1248 } |
1249 } | |
7578 | 1250 } |
1251 return 0; | |
1252 } | |
1253 | |
7539 | 1254 /** |
1255 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
7523
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1256 * |
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1257 * @param crc flag indicating the presence of CRC checksum |
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1258 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 1259 * |
7523
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1260 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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1261 */ |
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1262 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
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1263 // TODO : sbr_extension implementation |
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1264 ff_log_missing_feature(ac->avccontext, "SBR", 0); |
7523
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1265 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
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1266 return cnt; |
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1267 } |
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1268 |
7539 | 1269 /** |
7578 | 1270 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. |
1271 * | |
1272 * @return Returns number of bytes consumed. | |
1273 */ | |
1274 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { | |
1275 int i; | |
1276 int num_excl_chan = 0; | |
1277 | |
1278 do { | |
1279 for (i = 0; i < 7; i++) | |
1280 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); | |
1281 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); | |
1282 | |
1283 return num_excl_chan / 7; | |
1284 } | |
1285 | |
1286 /** | |
7539 | 1287 * Decode dynamic range information; reference: table 4.52. |
1288 * | |
1289 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1290 * | |
1291 * @return Returns number of bytes consumed. | |
1292 */ | |
1293 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { | |
1294 int n = 1; | |
1295 int drc_num_bands = 1; | |
1296 int i; | |
1297 | |
1298 /* pce_tag_present? */ | |
1299 if(get_bits1(gb)) { | |
1300 che_drc->pce_instance_tag = get_bits(gb, 4); | |
1301 skip_bits(gb, 4); // tag_reserved_bits | |
1302 n++; | |
1303 } | |
1304 | |
1305 /* excluded_chns_present? */ | |
1306 if(get_bits1(gb)) { | |
1307 n += decode_drc_channel_exclusions(che_drc, gb); | |
1308 } | |
1309 | |
1310 /* drc_bands_present? */ | |
1311 if (get_bits1(gb)) { | |
1312 che_drc->band_incr = get_bits(gb, 4); | |
1313 che_drc->interpolation_scheme = get_bits(gb, 4); | |
1314 n++; | |
1315 drc_num_bands += che_drc->band_incr; | |
1316 for (i = 0; i < drc_num_bands; i++) { | |
1317 che_drc->band_top[i] = get_bits(gb, 8); | |
1318 n++; | |
1319 } | |
1320 } | |
1321 | |
1322 /* prog_ref_level_present? */ | |
1323 if (get_bits1(gb)) { | |
1324 che_drc->prog_ref_level = get_bits(gb, 7); | |
1325 skip_bits1(gb); // prog_ref_level_reserved_bits | |
1326 n++; | |
1327 } | |
1328 | |
1329 for (i = 0; i < drc_num_bands; i++) { | |
1330 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
1331 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
1332 n++; | |
1333 } | |
1334 | |
1335 return n; | |
1336 } | |
1337 | |
1338 /** | |
1339 * Decode extension data (incomplete); reference: table 4.51. | |
1340 * | |
1341 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1342 * | |
1343 * @return Returns number of bytes consumed | |
1344 */ | |
1345 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { | |
7523
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1346 int crc_flag = 0; |
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1347 int res = cnt; |
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1348 switch (get_bits(gb, 4)) { // extension type |
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1349 case EXT_SBR_DATA_CRC: |
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1350 crc_flag++; |
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1351 case EXT_SBR_DATA: |
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1352 res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
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1353 break; |
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1354 case EXT_DYNAMIC_RANGE: |
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1355 res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
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1356 break; |
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1357 case EXT_FILL: |
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1358 case EXT_FILL_DATA: |
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1359 case EXT_DATA_ELEMENT: |
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1360 default: |
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1361 skip_bits_long(gb, 8*cnt - 4); |
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1362 break; |
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1363 }; |
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1364 return res; |
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1365 } |
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1366 |
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1367 /** |
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1368 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. |
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1369 * |
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1370 * @param decode 1 if tool is used normally, 0 if tool is used in LTP. |
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1371 * @param coef spectral coefficients |
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1372 */ |
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1373 static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) { |
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1374 const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); |
7636 | 1375 int w, filt, m, i; |
7632
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1376 int bottom, top, order, start, end, size, inc; |
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1377 float lpc[TNS_MAX_ORDER]; |
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1378 |
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1379 for (w = 0; w < ics->num_windows; w++) { |
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1380 bottom = ics->num_swb; |
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1381 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
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1382 top = bottom; |
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1383 bottom = FFMAX(0, top - tns->length[w][filt]); |
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1384 order = tns->order[w][filt]; |
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1385 if (order == 0) |
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1386 continue; |
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1387 |
7788
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1388 // tns_decode_coef |
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1389 compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); |
7632
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1390 |
7609 | 1391 start = ics->swb_offset[FFMIN(bottom, mmm)]; |
1392 end = ics->swb_offset[FFMIN( top, mmm)]; | |
1393 if ((size = end - start) <= 0) | |
1394 continue; | |
1395 if (tns->direction[w][filt]) { | |
1396 inc = -1; start = end - 1; | |
1397 } else { | |
1398 inc = 1; | |
1399 } | |
1400 start += w * 128; | |
1401 | |
1402 // ar filter | |
1403 for (m = 0; m < size; m++, start += inc) | |
1404 for (i = 1; i <= FFMIN(m, order); i++) | |
7632
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1405 coef[start] -= coef[start - i*inc] * lpc[i-1]; |
7609 | 1406 } |
1407 } | |
1408 } | |
1409 | |
7523
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1410 /** |
7539 | 1411 * Conduct IMDCT and windowing. |
1412 */ | |
1413 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { | |
1414 IndividualChannelStream * ics = &sce->ics; | |
1415 float * in = sce->coeffs; | |
1416 float * out = sce->ret; | |
1417 float * saved = sce->saved; | |
7580 | 1418 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
1419 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
1420 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
7539 | 1421 float * buf = ac->buf_mdct; |
8286
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1422 float * temp = ac->temp; |
7539 | 1423 int i; |
1424 | |
7661
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1425 // imdct |
7578 | 1426 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1427 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) | |
1428 av_log(ac->avccontext, AV_LOG_WARNING, | |
1429 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " | |
1430 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); | |
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1431 for (i = 0; i < 1024; i += 128) |
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1432 ff_imdct_half(&ac->mdct_small, buf + i, in + i); |
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1433 } else |
7663
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1434 ff_imdct_half(&ac->mdct, buf, in); |
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1435 |
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1436 /* window overlapping |
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1437 * NOTE: To simplify the overlapping code, all 'meaningless' short to long |
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1438 * and long to short transitions are considered to be short to short |
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1439 * transitions. This leaves just two cases (long to long and short to short) |
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1440 * with a little special sauce for EIGHT_SHORT_SEQUENCE. |
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1441 */ |
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1442 if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
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1443 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
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1444 ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512); |
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1445 } else { |
7662 | 1446 for (i = 0; i < 448; i++) |
1447 out[i] = saved[i] + ac->add_bias; | |
7578 | 1448 |
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1449 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
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1450 ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64); |
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1451 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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1452 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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1453 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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1454 ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64); |
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1455 memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
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1456 } else { |
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1457 ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64); |
7662 | 1458 for (i = 576; i < 1024; i++) |
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1459 out[i] = buf[i-512] + ac->add_bias; |
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1460 } |
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1461 } |
7578 | 1462 |
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1463 // buffer update |
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1464 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
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1465 for (i = 0; i < 64; i++) |
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1466 saved[i] = temp[64 + i] - ac->add_bias; |
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1467 ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); |
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1468 ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); |
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|
1469 ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); |
66fb1f3937fa
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|
1470 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
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1471 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
7663
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parents:
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diff
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|
1472 memcpy( saved, buf + 512, 448 * sizeof(float)); |
66fb1f3937fa
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parents:
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|
1473 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
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1474 } else { // LONG_STOP or ONLY_LONG |
7663
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|
1475 memcpy( saved, buf + 512, 512 * sizeof(float)); |
7578 | 1476 } |
1477 } | |
1478 | |
7539 | 1479 /** |
7523
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1480 * Apply dependent channel coupling (applied before IMDCT). |
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1481 * |
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1482 * @param index index into coupling gain array |
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1483 */ |
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1484 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
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1485 IndividualChannelStream * ics = &cce->ch[0].ics; |
7523
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1486 const uint16_t * offsets = ics->swb_offset; |
8006
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1487 float * dest = target->coeffs; |
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1488 const float * src = cce->ch[0].coeffs; |
7523
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1489 int g, i, group, k, idx = 0; |
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1490 if(ac->m4ac.object_type == AOT_AAC_LTP) { |
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1491 av_log(ac->avccontext, AV_LOG_ERROR, |
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1492 "Dependent coupling is not supported together with LTP\n"); |
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1493 return; |
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1494 } |
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1495 for (g = 0; g < ics->num_window_groups; g++) { |
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1496 for (i = 0; i < ics->max_sfb; i++, idx++) { |
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1497 if (cce->ch[0].band_type[idx] != ZERO_BT) { |
9143 | 1498 const float gain = cce->coup.gain[index][idx]; |
7523
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1499 for (group = 0; group < ics->group_len[g]; group++) { |
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1500 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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1501 // XXX dsputil-ize |
9143 | 1502 dest[group*128+k] += gain * src[group*128+k]; |
7523
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1503 } |
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1504 } |
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1505 } |
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1506 } |
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1507 dest += ics->group_len[g]*128; |
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1508 src += ics->group_len[g]*128; |
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1509 } |
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1510 } |
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1511 |
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1512 /** |
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1513 * Apply independent channel coupling (applied after IMDCT). |
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1514 * |
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1515 * @param index index into coupling gain array |
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1516 */ |
8006
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1517 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
7523
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1518 int i; |
9051
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1519 const float gain = cce->coup.gain[index][0]; |
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1520 const float bias = ac->add_bias; |
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1521 const float* src = cce->ch[0].ret; |
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1522 float* dest = target->ret; |
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1523 |
7523
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1524 for (i = 0; i < 1024; i++) |
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1525 dest[i] += gain * (src[i] - bias); |
7523
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1526 } |
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1527 |
7581 | 1528 /** |
1529 * channel coupling transformation interface | |
1530 * | |
1531 * @param index index into coupling gain array | |
1532 * @param apply_coupling_method pointer to (in)dependent coupling function | |
1533 */ | |
1534 static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, | |
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1535 enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, |
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1536 void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index)) |
7581 | 1537 { |
8004
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1538 int i, c; |
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1539 |
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1540 for (i = 0; i < MAX_ELEM_ID; i++) { |
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1541 ChannelElement *cce = ac->che[TYPE_CCE][i]; |
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1542 int index = 0; |
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1543 |
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1544 if (cce && cce->coup.coupling_point == coupling_point) { |
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1545 ChannelCoupling * coup = &cce->coup; |
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1546 |
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1547 for (c = 0; c <= coup->num_coupled; c++) { |
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1548 if (coup->type[c] == type && coup->id_select[c] == elem_id) { |
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1549 if (coup->ch_select[c] != 1) { |
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1550 apply_coupling_method(ac, &cc->ch[0], cce, index); |
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1551 if (coup->ch_select[c] != 0) |
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1552 index++; |
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1553 } |
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1554 if (coup->ch_select[c] != 2) |
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1555 apply_coupling_method(ac, &cc->ch[1], cce, index++); |
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1556 } else |
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1557 index += 1 + (coup->ch_select[c] == 3); |
7581 | 1558 } |
1559 } | |
1560 } | |
1561 } | |
1562 | |
1563 /** | |
1564 * Convert spectral data to float samples, applying all supported tools as appropriate. | |
1565 */ | |
1566 static void spectral_to_sample(AACContext * ac) { | |
8502 | 1567 int i, type; |
8004
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1568 for(type = 3; type >= 0; type--) { |
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1569 for (i = 0; i < MAX_ELEM_ID; i++) { |
7581 | 1570 ChannelElement *che = ac->che[type][i]; |
1571 if(che) { | |
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1572 if(type <= TYPE_CPE) |
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1573 apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); |
7581 | 1574 if(che->ch[0].tns.present) |
1575 apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); | |
1576 if(che->ch[1].tns.present) | |
1577 apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); | |
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1578 if(type <= TYPE_CPE) |
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1579 apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); |
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1580 if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) |
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1581 imdct_and_windowing(ac, &che->ch[0]); |
7581 | 1582 if(type == TYPE_CPE) |
1583 imdct_and_windowing(ac, &che->ch[1]); | |
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1584 if(type <= TYPE_CCE) |
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1585 apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); |
7578 | 1586 } |
1587 } | |
1588 } | |
1589 } | |
1590 | |
8549
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1591 static int parse_adts_frame_header(AACContext * ac, GetBitContext * gb) { |
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1592 |
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1593 int size; |
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1594 AACADTSHeaderInfo hdr_info; |
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1595 |
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1596 size = ff_aac_parse_header(gb, &hdr_info); |
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1597 if (size > 0) { |
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1598 if (hdr_info.chan_config) |
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1599 ac->m4ac.chan_config = hdr_info.chan_config; |
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1600 ac->m4ac.sample_rate = hdr_info.sample_rate; |
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1601 ac->m4ac.sampling_index = hdr_info.sampling_index; |
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1602 ac->m4ac.object_type = hdr_info.object_type; |
9151 | 1603 if (hdr_info.num_aac_frames == 1) { |
1604 if (!hdr_info.crc_absent) | |
1605 skip_bits(gb, 16); | |
1606 } else { | |
1607 ff_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0); | |
1608 return -1; | |
1609 } | |
9150
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1610 } |
8549
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1611 return size; |
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1612 } |
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1613 |
7578 | 1614 static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) { |
1615 AACContext * ac = avccontext->priv_data; | |
9149
955835e2bc99
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diff
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1616 ChannelElement * che = NULL; |
7578 | 1617 GetBitContext gb; |
1618 enum RawDataBlockType elem_type; | |
1619 int err, elem_id, data_size_tmp; | |
1620 | |
1621 init_get_bits(&gb, buf, buf_size*8); | |
1622 | |
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1623 if (show_bits(&gb, 12) == 0xfff) { |
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1624 if ((err = parse_adts_frame_header(ac, &gb)) < 0) { |
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1625 av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); |
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1626 return -1; |
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1627 } |
8812 | 1628 if (ac->m4ac.sampling_index > 12) { |
8789 | 1629 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
1630 return -1; | |
1631 } | |
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1632 } |
2cbfa4cd4670
Support ADTS AAC files in the ffaac decoder (limited to streams containing one
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1633 |
7578 | 1634 // parse |
1635 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { | |
1636 elem_id = get_bits(&gb, 4); | |
1637 err = -1; | |
1638 | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
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9143
diff
changeset
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1639 if(elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) { |
8814 | 1640 av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); |
1641 return -1; | |
7578 | 1642 } |
1643 | |
1644 switch (elem_type) { | |
1645 | |
1646 case TYPE_SCE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
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9143
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changeset
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1647 err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
7578 | 1648 break; |
1649 | |
1650 case TYPE_CPE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
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9143
diff
changeset
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1651 err = decode_cpe(ac, &gb, che); |
7578 | 1652 break; |
1653 | |
1654 case TYPE_CCE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
parents:
9143
diff
changeset
|
1655 err = decode_cce(ac, &gb, che); |
7578 | 1656 break; |
1657 | |
1658 case TYPE_LFE: | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
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9143
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changeset
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1659 err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
7578 | 1660 break; |
1661 | |
1662 case TYPE_DSE: | |
1663 skip_data_stream_element(&gb); | |
1664 err = 0; | |
1665 break; | |
1666 | |
1667 case TYPE_PCE: | |
1668 { | |
1669 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
1670 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
1671 if((err = decode_pce(ac, new_che_pos, &gb))) | |
1672 break; | |
9149
955835e2bc99
Fix the channel allocation bug/assumption (issue 800).
alexc
parents:
9143
diff
changeset
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1673 err = output_configure(ac, ac->che_pos, new_che_pos, 0); |
7578 | 1674 break; |
1675 } | |
1676 | |
1677 case TYPE_FIL: | |
1678 if (elem_id == 15) | |
1679 elem_id += get_bits(&gb, 8) - 1; | |
1680 while (elem_id > 0) | |
1681 elem_id -= decode_extension_payload(ac, &gb, elem_id); | |
1682 err = 0; /* FIXME */ | |
1683 break; | |
1684 | |
1685 default: | |
1686 err = -1; /* should not happen, but keeps compiler happy */ | |
1687 break; | |
1688 } | |
1689 | |
1690 if(err) | |
1691 return err; | |
1692 } | |
1693 | |
1694 spectral_to_sample(ac); | |
1695 | |
7539 | 1696 if (!ac->is_saved) { |
1697 ac->is_saved = 1; | |
1698 *data_size = 0; | |
7580 | 1699 return buf_size; |
7539 | 1700 } |
1701 | |
1702 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
1703 if(*data_size < data_size_tmp) { | |
1704 av_log(avccontext, AV_LOG_ERROR, | |
1705 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
1706 *data_size, data_size_tmp); | |
1707 return -1; | |
1708 } | |
1709 *data_size = data_size_tmp; | |
1710 | |
1711 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
1712 | |
1713 return buf_size; | |
1714 } | |
1715 | |
7501 | 1716 static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
1717 AACContext * ac = avccontext->priv_data; | |
7540 | 1718 int i, type; |
7501 | 1719 |
7523
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
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7501
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1720 for (i = 0; i < MAX_ELEM_ID; i++) { |
7540 | 1721 for(type = 0; type < 4; type++) |
1722 av_freep(&ac->che[type][i]); | |
7501 | 1723 } |
1724 | |
1725 ff_mdct_end(&ac->mdct); | |
1726 ff_mdct_end(&ac->mdct_small); | |
1727 return 0 ; | |
1728 } | |
1729 | |
1730 AVCodec aac_decoder = { | |
1731 "aac", | |
1732 CODEC_TYPE_AUDIO, | |
1733 CODEC_ID_AAC, | |
1734 sizeof(AACContext), | |
1735 aac_decode_init, | |
1736 NULL, | |
1737 aac_decode_close, | |
1738 aac_decode_frame, | |
1739 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
7523
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
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7501
diff
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1740 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7501 | 1741 }; |