Mercurial > libavcodec.hg
annotate aacenc.c @ 12431:9f8d9abd7984 libavcodec
cosmetic
author | aurel |
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date | Sat, 28 Aug 2010 14:07:16 +0000 |
parents | 335efeae3bb7 |
children | a3aca8bcbaf4 |
rev | line source |
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7571 | 1 /* |
2 * AAC encoder | |
3 * Copyright (C) 2008 Konstantin Shishkov | |
4 * | |
5 * This file is part of FFmpeg. | |
6 * | |
7 * FFmpeg is free software; you can redistribute it and/or | |
8 * modify it under the terms of the GNU Lesser General Public | |
9 * License as published by the Free Software Foundation; either | |
10 * version 2.1 of the License, or (at your option) any later version. | |
11 * | |
12 * FFmpeg is distributed in the hope that it will be useful, | |
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 * Lesser General Public License for more details. | |
16 * | |
17 * You should have received a copy of the GNU Lesser General Public | |
18 * License along with FFmpeg; if not, write to the Free Software | |
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
20 */ | |
21 | |
22 /** | |
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23 * @file |
7571 | 24 * AAC encoder |
25 */ | |
26 | |
27 /*********************************** | |
28 * TODOs: | |
7584 | 29 * add sane pulse detection |
7681 | 30 * add temporal noise shaping |
7571 | 31 ***********************************/ |
32 | |
33 #include "avcodec.h" | |
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34 #include "put_bits.h" |
7571 | 35 #include "dsputil.h" |
36 #include "mpeg4audio.h" | |
37 | |
38 #include "aac.h" | |
39 #include "aactab.h" | |
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40 #include "aacenc.h" |
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41 |
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42 #include "psymodel.h" |
7571 | 43 |
12095 | 44 #define AAC_MAX_CHANNELS 6 |
45 | |
7571 | 46 static const uint8_t swb_size_1024_96[] = { |
47 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, | |
48 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, | |
49 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64 | |
50 }; | |
51 | |
52 static const uint8_t swb_size_1024_64[] = { | |
53 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, | |
54 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36, | |
55 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40 | |
56 }; | |
57 | |
58 static const uint8_t swb_size_1024_48[] = { | |
59 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, | |
60 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, | |
61 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, | |
62 96 | |
63 }; | |
64 | |
65 static const uint8_t swb_size_1024_32[] = { | |
66 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, | |
67 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, | |
68 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32 | |
69 }; | |
70 | |
71 static const uint8_t swb_size_1024_24[] = { | |
72 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, | |
73 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, | |
74 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64 | |
75 }; | |
76 | |
77 static const uint8_t swb_size_1024_16[] = { | |
78 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, | |
79 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, | |
80 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64 | |
81 }; | |
82 | |
83 static const uint8_t swb_size_1024_8[] = { | |
84 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, | |
85 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28, | |
86 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80 | |
87 }; | |
88 | |
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89 static const uint8_t *swb_size_1024[] = { |
7571 | 90 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64, |
91 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32, | |
92 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16, | |
93 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8 | |
94 }; | |
95 | |
96 static const uint8_t swb_size_128_96[] = { | |
97 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36 | |
98 }; | |
99 | |
100 static const uint8_t swb_size_128_48[] = { | |
101 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16 | |
102 }; | |
103 | |
104 static const uint8_t swb_size_128_24[] = { | |
105 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20 | |
106 }; | |
107 | |
108 static const uint8_t swb_size_128_16[] = { | |
109 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20 | |
110 }; | |
111 | |
112 static const uint8_t swb_size_128_8[] = { | |
113 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20 | |
114 }; | |
115 | |
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116 static const uint8_t *swb_size_128[] = { |
7571 | 117 /* the last entry on the following row is swb_size_128_64 but is a |
118 duplicate of swb_size_128_96 */ | |
119 swb_size_128_96, swb_size_128_96, swb_size_128_96, | |
120 swb_size_128_48, swb_size_128_48, swb_size_128_48, | |
121 swb_size_128_24, swb_size_128_24, swb_size_128_16, | |
122 swb_size_128_16, swb_size_128_16, swb_size_128_8 | |
123 }; | |
124 | |
125 /** default channel configurations */ | |
126 static const uint8_t aac_chan_configs[6][5] = { | |
7585 | 127 {1, TYPE_SCE}, // 1 channel - single channel element |
128 {1, TYPE_CPE}, // 2 channels - channel pair | |
129 {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo | |
130 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center | |
131 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo | |
132 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE | |
7571 | 133 }; |
134 | |
135 /** | |
136 * Make AAC audio config object. | |
137 * @see 1.6.2.1 "Syntax - AudioSpecificConfig" | |
138 */ | |
139 static void put_audio_specific_config(AVCodecContext *avctx) | |
140 { | |
141 PutBitContext pb; | |
142 AACEncContext *s = avctx->priv_data; | |
143 | |
144 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8); | |
145 put_bits(&pb, 5, 2); //object type - AAC-LC | |
146 put_bits(&pb, 4, s->samplerate_index); //sample rate index | |
147 put_bits(&pb, 4, avctx->channels); | |
148 //GASpecificConfig | |
149 put_bits(&pb, 1, 0); //frame length - 1024 samples | |
150 put_bits(&pb, 1, 0); //does not depend on core coder | |
151 put_bits(&pb, 1, 0); //is not extension | |
152 flush_put_bits(&pb); | |
153 } | |
154 | |
155 static av_cold int aac_encode_init(AVCodecContext *avctx) | |
156 { | |
157 AACEncContext *s = avctx->priv_data; | |
158 int i; | |
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159 const uint8_t *sizes[2]; |
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160 int lengths[2]; |
7571 | 161 |
162 avctx->frame_size = 1024; | |
163 | |
9936 | 164 for (i = 0; i < 16; i++) |
165 if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i]) | |
7571 | 166 break; |
9936 | 167 if (i == 16) { |
7571 | 168 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate); |
169 return -1; | |
170 } | |
12095 | 171 if (avctx->channels > AAC_MAX_CHANNELS) { |
7571 | 172 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels); |
173 return -1; | |
174 } | |
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175 if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) { |
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176 av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile); |
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177 return -1; |
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178 } |
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179 if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) { |
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180 av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n"); |
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181 return -1; |
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182 } |
7571 | 183 s->samplerate_index = i; |
184 | |
185 dsputil_init(&s->dsp, avctx); | |
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186 ff_mdct_init(&s->mdct1024, 11, 0, 1.0); |
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187 ff_mdct_init(&s->mdct128, 8, 0, 1.0); |
7584 | 188 // window init |
189 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
190 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
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191 ff_init_ff_sine_windows(10); |
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192 ff_init_ff_sine_windows(7); |
7571 | 193 |
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194 s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0])); |
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195 s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]); |
11994 | 196 avctx->extradata = av_mallocz(2 + FF_INPUT_BUFFER_PADDING_SIZE); |
7571 | 197 avctx->extradata_size = 2; |
198 put_audio_specific_config(avctx); | |
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199 |
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200 sizes[0] = swb_size_1024[i]; |
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201 sizes[1] = swb_size_128[i]; |
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202 lengths[0] = ff_aac_num_swb_1024[i]; |
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203 lengths[1] = ff_aac_num_swb_128[i]; |
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204 ff_psy_init(&s->psy, avctx, 2, sizes, lengths); |
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205 s->psypp = ff_psy_preprocess_init(avctx); |
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206 s->coder = &ff_aac_coders[2]; |
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207 |
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208 s->lambda = avctx->global_quality ? avctx->global_quality : 120; |
11932 | 209 |
210 ff_aac_tableinit(); | |
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211 |
7571 | 212 return 0; |
213 } | |
214 | |
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215 static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s, |
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216 SingleChannelElement *sce, short *audio) |
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217 { |
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218 int i, k; |
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219 const int chans = avctx->channels; |
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220 const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; |
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221 const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
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222 const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; |
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223 |
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224 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
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225 memcpy(s->output, sce->saved, sizeof(float)*1024); |
9936 | 226 if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) { |
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227 memset(s->output, 0, sizeof(s->output[0]) * 448); |
9936 | 228 for (i = 448; i < 576; i++) |
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229 s->output[i] = sce->saved[i] * pwindow[i - 448]; |
9936 | 230 for (i = 576; i < 704; i++) |
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231 s->output[i] = sce->saved[i]; |
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232 } |
9936 | 233 if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) { |
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234 for (i = 0; i < 1024; i++) { |
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235 s->output[i+1024] = audio[i * chans] * lwindow[1024 - i - 1]; |
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236 sce->saved[i] = audio[i * chans] * lwindow[i]; |
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237 } |
9936 | 238 } else { |
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239 for (i = 0; i < 448; i++) |
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240 s->output[i+1024] = audio[i * chans]; |
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241 for (; i < 576; i++) |
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242 s->output[i+1024] = audio[i * chans] * swindow[576 - i - 1]; |
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243 memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448); |
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244 for (i = 0; i < 1024; i++) |
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245 sce->saved[i] = audio[i * chans]; |
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246 } |
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247 ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output); |
9936 | 248 } else { |
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249 for (k = 0; k < 1024; k += 128) { |
9936 | 250 for (i = 448 + k; i < 448 + k + 256; i++) |
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251 s->output[i - 448 - k] = (i < 1024) |
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252 ? sce->saved[i] |
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253 : audio[(i-1024)*chans]; |
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254 s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128); |
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255 s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128); |
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256 ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output); |
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257 } |
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258 for (i = 0; i < 1024; i++) |
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259 sce->saved[i] = audio[i * chans]; |
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260 } |
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261 } |
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262 |
7571 | 263 /** |
264 * Encode ics_info element. | |
265 * @see Table 4.6 (syntax of ics_info) | |
266 */ | |
7681 | 267 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info) |
7571 | 268 { |
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269 int w; |
7571 | 270 |
271 put_bits(&s->pb, 1, 0); // ics_reserved bit | |
272 put_bits(&s->pb, 2, info->window_sequence[0]); | |
273 put_bits(&s->pb, 1, info->use_kb_window[0]); | |
9936 | 274 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
7571 | 275 put_bits(&s->pb, 6, info->max_sfb); |
276 put_bits(&s->pb, 1, 0); // no prediction | |
9936 | 277 } else { |
7571 | 278 put_bits(&s->pb, 4, info->max_sfb); |
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279 for (w = 1; w < 8; w++) |
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280 put_bits(&s->pb, 1, !info->group_len[w]); |
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281 } |
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282 } |
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283 |
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284 /** |
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285 * Encode MS data. |
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286 * @see 4.6.8.1 "Joint Coding - M/S Stereo" |
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287 */ |
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288 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe) |
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289 { |
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290 int i, w; |
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291 |
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292 put_bits(pb, 2, cpe->ms_mode); |
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293 if (cpe->ms_mode == 1) |
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294 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w]) |
9936 | 295 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++) |
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296 put_bits(pb, 1, cpe->ms_mask[w*16 + i]); |
7571 | 297 } |
298 | |
299 /** | |
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300 * Produce integer coefficients from scalefactors provided by the model. |
7681 | 301 */ |
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302 static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans) |
7681 | 303 { |
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304 int i, w, w2, g, ch; |
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305 int start, sum, maxsfb, cmaxsfb; |
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306 |
9936 | 307 for (ch = 0; ch < chans; ch++) { |
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308 IndividualChannelStream *ics = &cpe->ch[ch].ics; |
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309 start = 0; |
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310 maxsfb = 0; |
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311 cpe->ch[ch].pulse.num_pulse = 0; |
9936 | 312 for (w = 0; w < ics->num_windows*16; w += 16) { |
313 for (g = 0; g < ics->num_swb; g++) { | |
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314 sum = 0; |
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315 //apply M/S |
9936 | 316 if (!ch && cpe->ms_mask[w + g]) { |
317 for (i = 0; i < ics->swb_sizes[g]; i++) { | |
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318 cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0; |
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319 cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i]; |
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320 } |
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321 } |
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322 start += ics->swb_sizes[g]; |
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323 } |
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324 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--) |
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325 ; |
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326 maxsfb = FFMAX(maxsfb, cmaxsfb); |
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327 } |
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328 ics->max_sfb = maxsfb; |
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329 |
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330 //adjust zero bands for window groups |
9936 | 331 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) { |
332 for (g = 0; g < ics->max_sfb; g++) { | |
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333 i = 1; |
9936 | 334 for (w2 = w; w2 < w + ics->group_len[w]; w2++) { |
335 if (!cpe->ch[ch].zeroes[w2*16 + g]) { | |
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336 i = 0; |
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337 break; |
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338 } |
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339 } |
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340 cpe->ch[ch].zeroes[w*16 + g] = i; |
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341 } |
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342 } |
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343 } |
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344 |
9936 | 345 if (chans > 1 && cpe->common_window) { |
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346 IndividualChannelStream *ics0 = &cpe->ch[0].ics; |
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347 IndividualChannelStream *ics1 = &cpe->ch[1].ics; |
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348 int msc = 0; |
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349 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb); |
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350 ics1->max_sfb = ics0->max_sfb; |
9936 | 351 for (w = 0; w < ics0->num_windows*16; w += 16) |
352 for (i = 0; i < ics0->max_sfb; i++) | |
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353 if (cpe->ms_mask[w+i]) |
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354 msc++; |
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355 if (msc == 0 || ics0->max_sfb == 0) |
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356 cpe->ms_mode = 0; |
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357 else |
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358 cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2; |
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359 } |
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360 } |
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361 |
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362 /** |
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363 * Encode scalefactor band coding type. |
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364 */ |
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365 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce) |
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366 { |
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367 int w; |
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368 |
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369 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) |
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370 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda); |
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371 } |
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372 |
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373 /** |
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374 * Encode scalefactors. |
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375 */ |
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376 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, |
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377 SingleChannelElement *sce) |
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378 { |
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379 int off = sce->sf_idx[0], diff; |
7681 | 380 int i, w; |
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381 |
9936 | 382 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { |
383 for (i = 0; i < sce->ics.max_sfb; i++) { | |
384 if (!sce->zeroes[w*16 + i]) { | |
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385 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO; |
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386 if (diff < 0 || diff > 120) |
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387 av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n"); |
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388 off = sce->sf_idx[w*16 + i]; |
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389 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]); |
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390 } |
7681 | 391 } |
392 } | |
393 } | |
394 | |
395 /** | |
7584 | 396 * Encode pulse data. |
397 */ | |
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398 static void encode_pulses(AACEncContext *s, Pulse *pulse) |
7584 | 399 { |
400 int i; | |
401 | |
402 put_bits(&s->pb, 1, !!pulse->num_pulse); | |
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403 if (!pulse->num_pulse) |
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404 return; |
7584 | 405 |
406 put_bits(&s->pb, 2, pulse->num_pulse - 1); | |
407 put_bits(&s->pb, 6, pulse->start); | |
9936 | 408 for (i = 0; i < pulse->num_pulse; i++) { |
7585 | 409 put_bits(&s->pb, 5, pulse->pos[i]); |
7584 | 410 put_bits(&s->pb, 4, pulse->amp[i]); |
411 } | |
412 } | |
413 | |
414 /** | |
415 * Encode spectral coefficients processed by psychoacoustic model. | |
416 */ | |
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417 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce) |
7584 | 418 { |
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419 int start, i, w, w2; |
7584 | 420 |
9936 | 421 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { |
7584 | 422 start = 0; |
9936 | 423 for (i = 0; i < sce->ics.max_sfb; i++) { |
424 if (sce->zeroes[w*16 + i]) { | |
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425 start += sce->ics.swb_sizes[i]; |
7584 | 426 continue; |
427 } | |
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428 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) |
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|
429 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128, |
9937
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430 sce->ics.swb_sizes[i], |
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|
431 sce->sf_idx[w*16 + i], |
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parents:
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|
432 sce->band_type[w*16 + i], |
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parents:
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diff
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|
433 s->lambda); |
7672
1c01b74dc78c
Do not pass unnecessary arguments to AAC encoder functions
kostya
parents:
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diff
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|
434 start += sce->ics.swb_sizes[i]; |
7584 | 435 } |
436 } | |
437 } | |
438 | |
439 /** | |
9935
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|
440 * Encode one channel of audio data. |
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|
441 */ |
9937
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parents:
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|
442 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, |
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parents:
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|
443 SingleChannelElement *sce, |
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diego
parents:
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diff
changeset
|
444 int common_window) |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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|
445 { |
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|
446 put_bits(&s->pb, 8, sce->sf_idx[0]); |
9937
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parents:
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|
447 if (!common_window) |
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diego
parents:
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diff
changeset
|
448 put_ics_info(s, &sce->ics); |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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|
449 encode_band_info(s, sce); |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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|
450 encode_scale_factors(avctx, s, sce); |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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|
451 encode_pulses(s, &sce->pulse); |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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|
452 put_bits(&s->pb, 1, 0); //tns |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
453 put_bits(&s->pb, 1, 0); //ssr |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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|
454 encode_spectral_coeffs(s, sce); |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
455 return 0; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
456 } |
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|
457 |
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|
458 /** |
7571 | 459 * Write some auxiliary information about the created AAC file. |
460 */ | |
9937
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461 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, |
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|
462 const char *name) |
7571 | 463 { |
464 int i, namelen, padbits; | |
465 | |
466 namelen = strlen(name) + 2; | |
7585 | 467 put_bits(&s->pb, 3, TYPE_FIL); |
7571 | 468 put_bits(&s->pb, 4, FFMIN(namelen, 15)); |
9936 | 469 if (namelen >= 15) |
7571 | 470 put_bits(&s->pb, 8, namelen - 16); |
471 put_bits(&s->pb, 4, 0); //extension type - filler | |
472 padbits = 8 - (put_bits_count(&s->pb) & 7); | |
473 align_put_bits(&s->pb); | |
9936 | 474 for (i = 0; i < namelen - 2; i++) |
7571 | 475 put_bits(&s->pb, 8, name[i]); |
476 put_bits(&s->pb, 12 - padbits, 0); | |
477 } | |
478 | |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
479 static int aac_encode_frame(AVCodecContext *avctx, |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
480 uint8_t *frame, int buf_size, void *data) |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
481 { |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
482 AACEncContext *s = avctx->priv_data; |
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|
483 int16_t *samples = s->samples, *samples2, *la; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
484 ChannelElement *cpe; |
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|
485 int i, j, chans, tag, start_ch; |
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|
486 const uint8_t *chan_map = aac_chan_configs[avctx->channels-1]; |
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|
487 int chan_el_counter[4]; |
12095 | 488 FFPsyWindowInfo windows[AAC_MAX_CHANNELS]; |
9935
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parents:
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|
489 |
9936 | 490 if (s->last_frame) |
9935
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|
491 return 0; |
9936 | 492 if (data) { |
493 if (!s->psypp) { | |
9937
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494 memcpy(s->samples + 1024 * avctx->channels, data, |
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|
495 1024 * avctx->channels * sizeof(s->samples[0])); |
9936 | 496 } else { |
9935
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|
497 start_ch = 0; |
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|
498 samples2 = s->samples + 1024 * avctx->channels; |
9936 | 499 for (i = 0; i < chan_map[0]; i++) { |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
500 tag = chan_map[i+1]; |
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|
501 chans = tag == TYPE_CPE ? 2 : 1; |
9937
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502 ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, |
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|
503 samples2 + start_ch, start_ch, chans); |
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|
504 start_ch += chans; |
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|
505 } |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
506 } |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
507 } |
9936 | 508 if (!avctx->frame_number) { |
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|
509 memcpy(s->samples, s->samples + 1024 * avctx->channels, |
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|
510 1024 * avctx->channels * sizeof(s->samples[0])); |
9935
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|
511 return 0; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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changeset
|
512 } |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
513 |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
514 start_ch = 0; |
9936 | 515 for (i = 0; i < chan_map[0]; i++) { |
9940
580fad942ae6
Refactor windowing and early psy from bitstream writing parts of the AAC encoder.
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parents:
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diff
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|
516 FFPsyWindowInfo* wi = windows + start_ch; |
9937
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517 tag = chan_map[i+1]; |
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518 chans = tag == TYPE_CPE ? 2 : 1; |
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|
519 cpe = &s->cpe[i]; |
9936 | 520 for (j = 0; j < chans; j++) { |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
521 IndividualChannelStream *ics = &cpe->ch[j].ics; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
522 int k; |
12191
335efeae3bb7
aacenc: Adjust array offsets for the current channel before calling ff_psy_suggest_window().
alexc
parents:
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diff
changeset
|
523 int cur_channel = start_ch + j; |
335efeae3bb7
aacenc: Adjust array offsets for the current channel before calling ff_psy_suggest_window().
alexc
parents:
12190
diff
changeset
|
524 samples2 = samples + cur_channel; |
335efeae3bb7
aacenc: Adjust array offsets for the current channel before calling ff_psy_suggest_window().
alexc
parents:
12190
diff
changeset
|
525 la = samples2 + (448+64) * avctx->channels; |
335efeae3bb7
aacenc: Adjust array offsets for the current channel before calling ff_psy_suggest_window().
alexc
parents:
12190
diff
changeset
|
526 if (!data) |
335efeae3bb7
aacenc: Adjust array offsets for the current channel before calling ff_psy_suggest_window().
alexc
parents:
12190
diff
changeset
|
527 la = NULL; |
12109 | 528 if (tag == TYPE_LFE) { |
529 wi[j].window_type[0] = ONLY_LONG_SEQUENCE; | |
530 wi[j].window_shape = 0; | |
531 wi[j].num_windows = 1; | |
532 wi[j].grouping[0] = 1; | |
533 } else { | |
12191
335efeae3bb7
aacenc: Adjust array offsets for the current channel before calling ff_psy_suggest_window().
alexc
parents:
12190
diff
changeset
|
534 wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, cur_channel, |
12110 | 535 ics->window_sequence[0]); |
12109 | 536 } |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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changeset
|
537 ics->window_sequence[1] = ics->window_sequence[0]; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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diff
changeset
|
538 ics->window_sequence[0] = wi[j].window_type[0]; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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diff
changeset
|
539 ics->use_kb_window[1] = ics->use_kb_window[0]; |
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
changeset
|
540 ics->use_kb_window[0] = wi[j].window_shape; |
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
changeset
|
541 ics->num_windows = wi[j].num_windows; |
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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diff
changeset
|
542 ics->swb_sizes = s->psy.bands [ics->num_windows == 8]; |
12109 | 543 ics->num_swb = tag == TYPE_LFE ? 12 : s->psy.num_bands[ics->num_windows == 8]; |
9936 | 544 for (k = 0; k < ics->num_windows; k++) |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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diff
changeset
|
545 ics->group_len[k] = wi[j].grouping[k]; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
changeset
|
546 |
12191
335efeae3bb7
aacenc: Adjust array offsets for the current channel before calling ff_psy_suggest_window().
alexc
parents:
12190
diff
changeset
|
547 s->cur_channel = cur_channel; |
335efeae3bb7
aacenc: Adjust array offsets for the current channel before calling ff_psy_suggest_window().
alexc
parents:
12190
diff
changeset
|
548 apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2); |
9940
580fad942ae6
Refactor windowing and early psy from bitstream writing parts of the AAC encoder.
alexc
parents:
9939
diff
changeset
|
549 } |
580fad942ae6
Refactor windowing and early psy from bitstream writing parts of the AAC encoder.
alexc
parents:
9939
diff
changeset
|
550 start_ch += chans; |
580fad942ae6
Refactor windowing and early psy from bitstream writing parts of the AAC encoder.
alexc
parents:
9939
diff
changeset
|
551 } |
9941 | 552 do { |
553 int frame_bits; | |
9942 | 554 init_put_bits(&s->pb, frame, buf_size*8); |
555 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)) | |
556 put_bitstream_info(avctx, s, LIBAVCODEC_IDENT); | |
557 start_ch = 0; | |
558 memset(chan_el_counter, 0, sizeof(chan_el_counter)); | |
559 for (i = 0; i < chan_map[0]; i++) { | |
560 FFPsyWindowInfo* wi = windows + start_ch; | |
561 tag = chan_map[i+1]; | |
562 chans = tag == TYPE_CPE ? 2 : 1; | |
563 cpe = &s->cpe[i]; | |
564 for (j = 0; j < chans; j++) { | |
11706
15e7486e5c7a
Set cur_channel in the AAC encoder context where needed.
alexc
parents:
11687
diff
changeset
|
565 s->cur_channel = start_ch + j; |
11728 | 566 ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]); |
9942 | 567 s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda); |
568 } | |
569 cpe->common_window = 0; | |
570 if (chans > 1 | |
571 && wi[0].window_type[0] == wi[1].window_type[0] | |
572 && wi[0].window_shape == wi[1].window_shape) { | |
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|
573 |
9942 | 574 cpe->common_window = 1; |
575 for (j = 0; j < wi[0].num_windows; j++) { | |
576 if (wi[0].grouping[j] != wi[1].grouping[j]) { | |
577 cpe->common_window = 0; | |
578 break; | |
579 } | |
9935
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parents:
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diff
changeset
|
580 } |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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diff
changeset
|
581 } |
11706
15e7486e5c7a
Set cur_channel in the AAC encoder context where needed.
alexc
parents:
11687
diff
changeset
|
582 s->cur_channel = start_ch; |
9942 | 583 if (cpe->common_window && s->coder->search_for_ms) |
584 s->coder->search_for_ms(s, cpe, s->lambda); | |
585 adjust_frame_information(s, cpe, chans); | |
586 put_bits(&s->pb, 3, tag); | |
587 put_bits(&s->pb, 4, chan_el_counter[tag]++); | |
588 if (chans == 2) { | |
589 put_bits(&s->pb, 1, cpe->common_window); | |
590 if (cpe->common_window) { | |
591 put_ics_info(s, &cpe->ch[0].ics); | |
592 encode_ms_info(&s->pb, cpe); | |
593 } | |
594 } | |
595 for (j = 0; j < chans; j++) { | |
596 s->cur_channel = start_ch + j; | |
597 encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window); | |
598 } | |
599 start_ch += chans; | |
9935
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|
600 } |
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diff
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|
601 |
9941 | 602 frame_bits = put_bits_count(&s->pb); |
603 if (frame_bits <= 6144 * avctx->channels - 3) | |
604 break; | |
605 | |
606 s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits; | |
607 | |
608 } while (1); | |
609 | |
9935
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changeset
|
610 put_bits(&s->pb, 3, TYPE_END); |
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diff
changeset
|
611 flush_put_bits(&s->pb); |
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parents:
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diff
changeset
|
612 avctx->frame_bits = put_bits_count(&s->pb); |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
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diff
changeset
|
613 |
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alexc
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diff
changeset
|
614 // rate control stuff |
9936 | 615 if (!(avctx->flags & CODEC_FLAG_QSCALE)) { |
9935
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changeset
|
616 float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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changeset
|
617 s->lambda *= ratio; |
9944
c5ca5e520fe1
Change fminf/fmaxf to FFMIN/FFMAX to fix the build on broken operating systems.
alexc
parents:
9942
diff
changeset
|
618 s->lambda = FFMIN(s->lambda, 65536.f); |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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diff
changeset
|
619 } |
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alexc
parents:
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changeset
|
620 |
9936 | 621 if (!data) |
9935
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changeset
|
622 s->last_frame = 1; |
9937
3e39dbd2d9eb
cosmetics: prettyprinting, K&R style, break overly long lines
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623 memcpy(s->samples, s->samples + 1024 * avctx->channels, |
3e39dbd2d9eb
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624 1024 * avctx->channels * sizeof(s->samples[0])); |
9935
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
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625 return put_bits_count(&s->pb)>>3; |
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
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626 } |
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
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627 |
7571 | 628 static av_cold int aac_encode_end(AVCodecContext *avctx) |
629 { | |
630 AACEncContext *s = avctx->priv_data; | |
631 | |
632 ff_mdct_end(&s->mdct1024); | |
633 ff_mdct_end(&s->mdct128); | |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
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634 ff_psy_end(&s->psy); |
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635 ff_psy_preprocess_end(s->psypp); |
7571 | 636 av_freep(&s->samples); |
637 av_freep(&s->cpe); | |
638 return 0; | |
639 } | |
640 | |
641 AVCodec aac_encoder = { | |
642 "aac", | |
11560
8a4984c5cacc
Define AVMediaType enum, and use it instead of enum CodecType, which
stefano
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643 AVMEDIA_TYPE_AUDIO, |
7571 | 644 CODEC_ID_AAC, |
645 sizeof(AACEncContext), | |
646 aac_encode_init, | |
647 aac_encode_frame, | |
648 aac_encode_end, | |
11786 | 649 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL, |
10145
7955db355703
Make sample_fmts and channel_layouts compound literals const to reduce size of
reimar
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650 .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7571 | 651 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), |
652 }; |