Mercurial > libavcodec.hg
annotate aacenc.c @ 12043:f9a0bd0888a4 libavcodec
mpegaudio: call ff_mpegaudiodec_init_mmx() only from float decoder
The mmx code is floating-point only, and this function does not know
from which decoder it is called. Without this change, the integer
decoder only "works" because the size of the context struct is smaller
in this case, and the mmx init function writes the function pointer
outside the allocated context.
author | mru |
---|---|
date | Thu, 01 Jul 2010 23:21:17 +0000 |
parents | 949d048b7611 |
children | 108e7da64995 |
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 |
44 static const uint8_t swb_size_1024_96[] = { | |
45 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, | |
46 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, | |
47 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64 | |
48 }; | |
49 | |
50 static const uint8_t swb_size_1024_64[] = { | |
51 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, | |
52 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36, | |
53 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40 | |
54 }; | |
55 | |
56 static const uint8_t swb_size_1024_48[] = { | |
57 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, | |
58 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, | |
59 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, | |
60 96 | |
61 }; | |
62 | |
63 static const uint8_t swb_size_1024_32[] = { | |
64 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, | |
65 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, | |
66 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32 | |
67 }; | |
68 | |
69 static const uint8_t swb_size_1024_24[] = { | |
70 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, | |
71 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28, | |
72 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64 | |
73 }; | |
74 | |
75 static const uint8_t swb_size_1024_16[] = { | |
76 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, | |
77 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28, | |
78 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64 | |
79 }; | |
80 | |
81 static const uint8_t swb_size_1024_8[] = { | |
82 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, | |
83 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28, | |
84 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80 | |
85 }; | |
86 | |
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87 static const uint8_t *swb_size_1024[] = { |
7571 | 88 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64, |
89 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32, | |
90 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16, | |
91 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8 | |
92 }; | |
93 | |
94 static const uint8_t swb_size_128_96[] = { | |
95 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36 | |
96 }; | |
97 | |
98 static const uint8_t swb_size_128_48[] = { | |
99 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16 | |
100 }; | |
101 | |
102 static const uint8_t swb_size_128_24[] = { | |
103 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20 | |
104 }; | |
105 | |
106 static const uint8_t swb_size_128_16[] = { | |
107 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20 | |
108 }; | |
109 | |
110 static const uint8_t swb_size_128_8[] = { | |
111 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20 | |
112 }; | |
113 | |
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114 static const uint8_t *swb_size_128[] = { |
7571 | 115 /* the last entry on the following row is swb_size_128_64 but is a |
116 duplicate of swb_size_128_96 */ | |
117 swb_size_128_96, swb_size_128_96, swb_size_128_96, | |
118 swb_size_128_48, swb_size_128_48, swb_size_128_48, | |
119 swb_size_128_24, swb_size_128_24, swb_size_128_16, | |
120 swb_size_128_16, swb_size_128_16, swb_size_128_8 | |
121 }; | |
122 | |
123 /** default channel configurations */ | |
124 static const uint8_t aac_chan_configs[6][5] = { | |
7585 | 125 {1, TYPE_SCE}, // 1 channel - single channel element |
126 {1, TYPE_CPE}, // 2 channels - channel pair | |
127 {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo | |
128 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center | |
129 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo | |
130 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE | |
7571 | 131 }; |
132 | |
133 /** | |
134 * Make AAC audio config object. | |
135 * @see 1.6.2.1 "Syntax - AudioSpecificConfig" | |
136 */ | |
137 static void put_audio_specific_config(AVCodecContext *avctx) | |
138 { | |
139 PutBitContext pb; | |
140 AACEncContext *s = avctx->priv_data; | |
141 | |
142 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8); | |
143 put_bits(&pb, 5, 2); //object type - AAC-LC | |
144 put_bits(&pb, 4, s->samplerate_index); //sample rate index | |
145 put_bits(&pb, 4, avctx->channels); | |
146 //GASpecificConfig | |
147 put_bits(&pb, 1, 0); //frame length - 1024 samples | |
148 put_bits(&pb, 1, 0); //does not depend on core coder | |
149 put_bits(&pb, 1, 0); //is not extension | |
150 flush_put_bits(&pb); | |
151 } | |
152 | |
153 static av_cold int aac_encode_init(AVCodecContext *avctx) | |
154 { | |
155 AACEncContext *s = avctx->priv_data; | |
156 int i; | |
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157 const uint8_t *sizes[2]; |
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158 int lengths[2]; |
7571 | 159 |
160 avctx->frame_size = 1024; | |
161 | |
9936 | 162 for (i = 0; i < 16; i++) |
163 if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i]) | |
7571 | 164 break; |
9936 | 165 if (i == 16) { |
7571 | 166 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate); |
167 return -1; | |
168 } | |
9936 | 169 if (avctx->channels > 6) { |
7571 | 170 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels); |
171 return -1; | |
172 } | |
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173 if (avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW) { |
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174 av_log(avctx, AV_LOG_ERROR, "Unsupported profile %d\n", avctx->profile); |
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175 return -1; |
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176 } |
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177 if (1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * avctx->channels) { |
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178 av_log(avctx, AV_LOG_ERROR, "Too many bits per frame requested\n"); |
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179 return -1; |
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180 } |
7571 | 181 s->samplerate_index = i; |
182 | |
183 dsputil_init(&s->dsp, avctx); | |
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184 ff_mdct_init(&s->mdct1024, 11, 0, 1.0); |
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185 ff_mdct_init(&s->mdct128, 8, 0, 1.0); |
7584 | 186 // window init |
187 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
188 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
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189 ff_init_ff_sine_windows(10); |
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190 ff_init_ff_sine_windows(7); |
7571 | 191 |
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192 s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0])); |
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193 s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]); |
11994 | 194 avctx->extradata = av_mallocz(2 + FF_INPUT_BUFFER_PADDING_SIZE); |
7571 | 195 avctx->extradata_size = 2; |
196 put_audio_specific_config(avctx); | |
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197 |
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198 sizes[0] = swb_size_1024[i]; |
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199 sizes[1] = swb_size_128[i]; |
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200 lengths[0] = ff_aac_num_swb_1024[i]; |
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201 lengths[1] = ff_aac_num_swb_128[i]; |
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202 ff_psy_init(&s->psy, avctx, 2, sizes, lengths); |
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203 s->psypp = ff_psy_preprocess_init(avctx); |
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204 s->coder = &ff_aac_coders[2]; |
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205 |
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206 s->lambda = avctx->global_quality ? avctx->global_quality : 120; |
11932 | 207 |
208 ff_aac_tableinit(); | |
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209 |
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210 if (avctx->channels > 5) |
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211 av_log(avctx, AV_LOG_ERROR, "This encoder does not yet enforce the restrictions on LFEs. " |
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212 "The output will most likely be an illegal bitstream.\n"); |
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213 |
7571 | 214 return 0; |
215 } | |
216 | |
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217 static void apply_window_and_mdct(AVCodecContext *avctx, AACEncContext *s, |
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218 SingleChannelElement *sce, short *audio, int channel) |
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219 { |
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220 int i, j, k; |
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221 const float * lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; |
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222 const float * swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
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223 const float * pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; |
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224 |
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225 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
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226 memcpy(s->output, sce->saved, sizeof(float)*1024); |
9936 | 227 if (sce->ics.window_sequence[0] == LONG_STOP_SEQUENCE) { |
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228 memset(s->output, 0, sizeof(s->output[0]) * 448); |
9936 | 229 for (i = 448; i < 576; i++) |
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230 s->output[i] = sce->saved[i] * pwindow[i - 448]; |
9936 | 231 for (i = 576; i < 704; i++) |
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232 s->output[i] = sce->saved[i]; |
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233 } |
9936 | 234 if (sce->ics.window_sequence[0] != LONG_START_SEQUENCE) { |
11837 | 235 for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) { |
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236 s->output[i+1024] = audio[j] * lwindow[1024 - i - 1]; |
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237 sce->saved[i] = audio[j] * lwindow[i]; |
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238 } |
9936 | 239 } else { |
11837 | 240 for (i = 0, j = channel; i < 448; i++, j += avctx->channels) |
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241 s->output[i+1024] = audio[j]; |
11837 | 242 for (; i < 576; i++, j += avctx->channels) |
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243 s->output[i+1024] = audio[j] * swindow[576 - i - 1]; |
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244 memset(s->output+1024+576, 0, sizeof(s->output[0]) * 448); |
11837 | 245 for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) |
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246 sce->saved[i] = audio[j]; |
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247 } |
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248 ff_mdct_calc(&s->mdct1024, sce->coeffs, s->output); |
9936 | 249 } else { |
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250 for (k = 0; k < 1024; k += 128) { |
9936 | 251 for (i = 448 + k; i < 448 + k + 256; i++) |
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252 s->output[i - 448 - k] = (i < 1024) |
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253 ? sce->saved[i] |
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254 : audio[channel + (i-1024)*avctx->channels]; |
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255 s->dsp.vector_fmul (s->output, k ? swindow : pwindow, 128); |
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256 s->dsp.vector_fmul_reverse(s->output+128, s->output+128, swindow, 128); |
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257 ff_mdct_calc(&s->mdct128, sce->coeffs + k, s->output); |
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258 } |
11837 | 259 for (i = 0, j = channel; i < 1024; i++, j += avctx->channels) |
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260 sce->saved[i] = audio[j]; |
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261 } |
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262 } |
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263 |
7571 | 264 /** |
265 * Encode ics_info element. | |
266 * @see Table 4.6 (syntax of ics_info) | |
267 */ | |
7681 | 268 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info) |
7571 | 269 { |
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270 int w; |
7571 | 271 |
272 put_bits(&s->pb, 1, 0); // ics_reserved bit | |
273 put_bits(&s->pb, 2, info->window_sequence[0]); | |
274 put_bits(&s->pb, 1, info->use_kb_window[0]); | |
9936 | 275 if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
7571 | 276 put_bits(&s->pb, 6, info->max_sfb); |
277 put_bits(&s->pb, 1, 0); // no prediction | |
9936 | 278 } else { |
7571 | 279 put_bits(&s->pb, 4, info->max_sfb); |
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280 for (w = 1; w < 8; w++) |
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281 put_bits(&s->pb, 1, !info->group_len[w]); |
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282 } |
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283 } |
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284 |
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285 /** |
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286 * Encode MS data. |
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287 * @see 4.6.8.1 "Joint Coding - M/S Stereo" |
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288 */ |
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289 static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe) |
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290 { |
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291 int i, w; |
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292 |
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293 put_bits(pb, 2, cpe->ms_mode); |
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294 if (cpe->ms_mode == 1) |
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295 for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w]) |
9936 | 296 for (i = 0; i < cpe->ch[0].ics.max_sfb; i++) |
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297 put_bits(pb, 1, cpe->ms_mask[w*16 + i]); |
7571 | 298 } |
299 | |
300 /** | |
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301 * Produce integer coefficients from scalefactors provided by the model. |
7681 | 302 */ |
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303 static void adjust_frame_information(AACEncContext *apc, ChannelElement *cpe, int chans) |
7681 | 304 { |
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305 int i, w, w2, g, ch; |
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306 int start, sum, maxsfb, cmaxsfb; |
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307 |
9936 | 308 for (ch = 0; ch < chans; ch++) { |
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309 IndividualChannelStream *ics = &cpe->ch[ch].ics; |
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310 start = 0; |
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311 maxsfb = 0; |
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312 cpe->ch[ch].pulse.num_pulse = 0; |
9936 | 313 for (w = 0; w < ics->num_windows*16; w += 16) { |
314 for (g = 0; g < ics->num_swb; g++) { | |
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315 sum = 0; |
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316 //apply M/S |
9936 | 317 if (!ch && cpe->ms_mask[w + g]) { |
318 for (i = 0; i < ics->swb_sizes[g]; i++) { | |
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319 cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0; |
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320 cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i]; |
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321 } |
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322 } |
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323 start += ics->swb_sizes[g]; |
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324 } |
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325 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--) |
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326 ; |
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327 maxsfb = FFMAX(maxsfb, cmaxsfb); |
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328 } |
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329 ics->max_sfb = maxsfb; |
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330 |
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331 //adjust zero bands for window groups |
9936 | 332 for (w = 0; w < ics->num_windows; w += ics->group_len[w]) { |
333 for (g = 0; g < ics->max_sfb; g++) { | |
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334 i = 1; |
9936 | 335 for (w2 = w; w2 < w + ics->group_len[w]; w2++) { |
336 if (!cpe->ch[ch].zeroes[w2*16 + g]) { | |
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337 i = 0; |
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338 break; |
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339 } |
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340 } |
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341 cpe->ch[ch].zeroes[w*16 + g] = i; |
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342 } |
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343 } |
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344 } |
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345 |
9936 | 346 if (chans > 1 && cpe->common_window) { |
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347 IndividualChannelStream *ics0 = &cpe->ch[0].ics; |
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348 IndividualChannelStream *ics1 = &cpe->ch[1].ics; |
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349 int msc = 0; |
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350 ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb); |
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351 ics1->max_sfb = ics0->max_sfb; |
9936 | 352 for (w = 0; w < ics0->num_windows*16; w += 16) |
353 for (i = 0; i < ics0->max_sfb; i++) | |
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354 if (cpe->ms_mask[w+i]) |
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355 msc++; |
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356 if (msc == 0 || ics0->max_sfb == 0) |
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357 cpe->ms_mode = 0; |
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358 else |
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359 cpe->ms_mode = msc < ics0->max_sfb ? 1 : 2; |
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360 } |
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361 } |
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362 |
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363 /** |
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364 * Encode scalefactor band coding type. |
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365 */ |
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366 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce) |
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367 { |
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368 int w; |
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369 |
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370 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) |
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371 s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda); |
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372 } |
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373 |
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374 /** |
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375 * Encode scalefactors. |
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376 */ |
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377 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s, |
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378 SingleChannelElement *sce) |
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379 { |
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380 int off = sce->sf_idx[0], diff; |
7681 | 381 int i, w; |
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382 |
9936 | 383 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { |
384 for (i = 0; i < sce->ics.max_sfb; i++) { | |
385 if (!sce->zeroes[w*16 + i]) { | |
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386 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO; |
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387 if (diff < 0 || diff > 120) |
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388 av_log(avctx, AV_LOG_ERROR, "Scalefactor difference is too big to be coded\n"); |
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389 off = sce->sf_idx[w*16 + i]; |
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390 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]); |
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391 } |
7681 | 392 } |
393 } | |
394 } | |
395 | |
396 /** | |
7584 | 397 * Encode pulse data. |
398 */ | |
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399 static void encode_pulses(AACEncContext *s, Pulse *pulse) |
7584 | 400 { |
401 int i; | |
402 | |
403 put_bits(&s->pb, 1, !!pulse->num_pulse); | |
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404 if (!pulse->num_pulse) |
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405 return; |
7584 | 406 |
407 put_bits(&s->pb, 2, pulse->num_pulse - 1); | |
408 put_bits(&s->pb, 6, pulse->start); | |
9936 | 409 for (i = 0; i < pulse->num_pulse; i++) { |
7585 | 410 put_bits(&s->pb, 5, pulse->pos[i]); |
7584 | 411 put_bits(&s->pb, 4, pulse->amp[i]); |
412 } | |
413 } | |
414 | |
415 /** | |
416 * Encode spectral coefficients processed by psychoacoustic model. | |
417 */ | |
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418 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce) |
7584 | 419 { |
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420 int start, i, w, w2; |
7584 | 421 |
9936 | 422 for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) { |
7584 | 423 start = 0; |
9936 | 424 for (i = 0; i < sce->ics.max_sfb; i++) { |
425 if (sce->zeroes[w*16 + i]) { | |
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426 start += sce->ics.swb_sizes[i]; |
7584 | 427 continue; |
428 } | |
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429 for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) |
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430 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128, |
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|
431 sce->ics.swb_sizes[i], |
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parents:
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|
432 sce->sf_idx[w*16 + i], |
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parents:
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|
433 sce->band_type[w*16 + i], |
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diego
parents:
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|
434 s->lambda); |
7672
1c01b74dc78c
Do not pass unnecessary arguments to AAC encoder functions
kostya
parents:
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diff
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|
435 start += sce->ics.swb_sizes[i]; |
7584 | 436 } |
437 } | |
438 } | |
439 | |
440 /** | |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
441 * Encode one channel of audio data. |
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|
442 */ |
9937
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diego
parents:
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diff
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|
443 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s, |
3e39dbd2d9eb
cosmetics: prettyprinting, K&R style, break overly long lines
diego
parents:
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|
444 SingleChannelElement *sce, |
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diego
parents:
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diff
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|
445 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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diff
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|
446 { |
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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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|
447 put_bits(&s->pb, 8, sce->sf_idx[0]); |
9937
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diego
parents:
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|
448 if (!common_window) |
3e39dbd2d9eb
cosmetics: prettyprinting, K&R style, break overly long lines
diego
parents:
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|
449 put_ics_info(s, &sce->ics); |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
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|
450 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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|
451 encode_scale_factors(avctx, s, sce); |
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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|
452 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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|
453 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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parents:
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|
454 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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|
455 encode_spectral_coeffs(s, sce); |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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|
456 return 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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|
457 } |
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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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|
458 |
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|
459 /** |
7571 | 460 * Write some auxiliary information about the created AAC file. |
461 */ | |
9937
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|
462 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, |
3e39dbd2d9eb
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|
463 const char *name) |
7571 | 464 { |
465 int i, namelen, padbits; | |
466 | |
467 namelen = strlen(name) + 2; | |
7585 | 468 put_bits(&s->pb, 3, TYPE_FIL); |
7571 | 469 put_bits(&s->pb, 4, FFMIN(namelen, 15)); |
9936 | 470 if (namelen >= 15) |
7571 | 471 put_bits(&s->pb, 8, namelen - 16); |
472 put_bits(&s->pb, 4, 0); //extension type - filler | |
473 padbits = 8 - (put_bits_count(&s->pb) & 7); | |
474 align_put_bits(&s->pb); | |
9936 | 475 for (i = 0; i < namelen - 2; i++) |
7571 | 476 put_bits(&s->pb, 8, name[i]); |
477 put_bits(&s->pb, 12 - padbits, 0); | |
478 } | |
479 | |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
480 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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|
481 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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|
482 { |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
483 AACEncContext *s = avctx->priv_data; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
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|
484 int16_t *samples = s->samples, *samples2, *la; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
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|
485 ChannelElement *cpe; |
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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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|
486 int i, j, chans, tag, start_ch; |
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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
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|
487 const uint8_t *chan_map = aac_chan_configs[avctx->channels-1]; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
488 int chan_el_counter[4]; |
9940
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Refactor windowing and early psy from bitstream writing parts of the AAC encoder.
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489 FFPsyWindowInfo windows[avctx->channels]; |
9935
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parents:
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|
490 |
9936 | 491 if (s->last_frame) |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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492 return 0; |
9936 | 493 if (data) { |
494 if (!s->psypp) { | |
9937
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|
495 memcpy(s->samples + 1024 * avctx->channels, data, |
3e39dbd2d9eb
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|
496 1024 * avctx->channels * sizeof(s->samples[0])); |
9936 | 497 } else { |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
498 start_ch = 0; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
499 samples2 = s->samples + 1024 * avctx->channels; |
9936 | 500 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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|
501 tag = chan_map[i+1]; |
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parents:
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|
502 chans = tag == TYPE_CPE ? 2 : 1; |
9937
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503 ff_psy_preprocess(s->psypp, (uint16_t*)data + start_ch, |
3e39dbd2d9eb
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504 samples2 + start_ch, start_ch, chans); |
9935
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505 start_ch += chans; |
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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 } |
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|
508 } |
9936 | 509 if (!avctx->frame_number) { |
9937
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510 memcpy(s->samples, s->samples + 1024 * avctx->channels, |
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|
511 1024 * avctx->channels * sizeof(s->samples[0])); |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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|
512 return 0; |
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|
513 } |
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|
514 |
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|
515 start_ch = 0; |
9936 | 516 for (i = 0; i < chan_map[0]; i++) { |
9940
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|
517 FFPsyWindowInfo* wi = windows + start_ch; |
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518 tag = chan_map[i+1]; |
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519 chans = tag == TYPE_CPE ? 2 : 1; |
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parents:
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|
520 cpe = &s->cpe[i]; |
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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|
521 samples2 = samples + start_ch; |
11999
949d048b7611
aacenc: Start the lookahead pointer 1/4 of the way into the first short window.
alexc
parents:
11994
diff
changeset
|
522 la = samples2 + (448+64) * avctx->channels + start_ch; |
9938
6c1ac45b3097
cosmetics: Remove unnecessary {} around if/for blocks;
diego
parents:
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diff
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|
523 if (!data) |
6c1ac45b3097
cosmetics: Remove unnecessary {} around if/for blocks;
diego
parents:
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diff
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|
524 la = NULL; |
9936 | 525 for (j = 0; j < chans; j++) { |
9935
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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|
526 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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parents:
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|
527 int k; |
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|
528 wi[j] = ff_psy_suggest_window(&s->psy, samples2, la, start_ch + j, ics->window_sequence[0]); |
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|
529 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
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|
530 ics->window_sequence[0] = wi[j].window_type[0]; |
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
changeset
|
531 ics->use_kb_window[1] = ics->use_kb_window[0]; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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diff
changeset
|
532 ics->use_kb_window[0] = wi[j].window_shape; |
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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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|
533 ics->num_windows = wi[j].num_windows; |
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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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|
534 ics->swb_sizes = s->psy.bands [ics->num_windows == 8]; |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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diff
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|
535 ics->num_swb = s->psy.num_bands[ics->num_windows == 8]; |
9936 | 536 for (k = 0; k < ics->num_windows; k++) |
9935
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
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parents:
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diff
changeset
|
537 ics->group_len[k] = wi[j].grouping[k]; |
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
changeset
|
538 |
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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 s->cur_channel = start_ch + j; |
d09283aeeef8
Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
changeset
|
540 apply_window_and_mdct(avctx, s, &cpe->ch[j], samples2, j); |
9940
580fad942ae6
Refactor windowing and early psy from bitstream writing parts of the AAC encoder.
alexc
parents:
9939
diff
changeset
|
541 } |
580fad942ae6
Refactor windowing and early psy from bitstream writing parts of the AAC encoder.
alexc
parents:
9939
diff
changeset
|
542 start_ch += chans; |
580fad942ae6
Refactor windowing and early psy from bitstream writing parts of the AAC encoder.
alexc
parents:
9939
diff
changeset
|
543 } |
9941 | 544 do { |
545 int frame_bits; | |
9942 | 546 init_put_bits(&s->pb, frame, buf_size*8); |
547 if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT)) | |
548 put_bitstream_info(avctx, s, LIBAVCODEC_IDENT); | |
549 start_ch = 0; | |
550 memset(chan_el_counter, 0, sizeof(chan_el_counter)); | |
551 for (i = 0; i < chan_map[0]; i++) { | |
552 FFPsyWindowInfo* wi = windows + start_ch; | |
553 tag = chan_map[i+1]; | |
554 chans = tag == TYPE_CPE ? 2 : 1; | |
555 cpe = &s->cpe[i]; | |
556 for (j = 0; j < chans; j++) { | |
11706
15e7486e5c7a
Set cur_channel in the AAC encoder context where needed.
alexc
parents:
11687
diff
changeset
|
557 s->cur_channel = start_ch + j; |
11728 | 558 ff_psy_set_band_info(&s->psy, s->cur_channel, cpe->ch[j].coeffs, &wi[j]); |
9942 | 559 s->coder->search_for_quantizers(avctx, s, &cpe->ch[j], s->lambda); |
560 } | |
561 cpe->common_window = 0; | |
562 if (chans > 1 | |
563 && wi[0].window_type[0] == wi[1].window_type[0] | |
564 && wi[0].window_shape == wi[1].window_shape) { | |
9935
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parents:
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diff
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|
565 |
9942 | 566 cpe->common_window = 1; |
567 for (j = 0; j < wi[0].num_windows; j++) { | |
568 if (wi[0].grouping[j] != wi[1].grouping[j]) { | |
569 cpe->common_window = 0; | |
570 break; | |
571 } | |
9935
d09283aeeef8
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parents:
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diff
changeset
|
572 } |
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parents:
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|
573 } |
11706
15e7486e5c7a
Set cur_channel in the AAC encoder context where needed.
alexc
parents:
11687
diff
changeset
|
574 s->cur_channel = start_ch; |
9942 | 575 if (cpe->common_window && s->coder->search_for_ms) |
576 s->coder->search_for_ms(s, cpe, s->lambda); | |
577 adjust_frame_information(s, cpe, chans); | |
578 put_bits(&s->pb, 3, tag); | |
579 put_bits(&s->pb, 4, chan_el_counter[tag]++); | |
580 if (chans == 2) { | |
581 put_bits(&s->pb, 1, cpe->common_window); | |
582 if (cpe->common_window) { | |
583 put_ics_info(s, &cpe->ch[0].ics); | |
584 encode_ms_info(&s->pb, cpe); | |
585 } | |
586 } | |
587 for (j = 0; j < chans; j++) { | |
588 s->cur_channel = start_ch + j; | |
589 encode_individual_channel(avctx, s, &cpe->ch[j], cpe->common_window); | |
590 } | |
591 start_ch += chans; | |
9935
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|
592 } |
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|
593 |
9941 | 594 frame_bits = put_bits_count(&s->pb); |
595 if (frame_bits <= 6144 * avctx->channels - 3) | |
596 break; | |
597 | |
598 s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits; | |
599 | |
600 } while (1); | |
601 | |
9935
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|
602 put_bits(&s->pb, 3, TYPE_END); |
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diff
changeset
|
603 flush_put_bits(&s->pb); |
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diff
changeset
|
604 avctx->frame_bits = put_bits_count(&s->pb); |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
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changeset
|
605 |
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diff
changeset
|
606 // rate control stuff |
9936 | 607 if (!(avctx->flags & CODEC_FLAG_QSCALE)) { |
9935
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parents:
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changeset
|
608 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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diff
changeset
|
609 s->lambda *= ratio; |
9944
c5ca5e520fe1
Change fminf/fmaxf to FFMIN/FFMAX to fix the build on broken operating systems.
alexc
parents:
9942
diff
changeset
|
610 s->lambda = FFMIN(s->lambda, 65536.f); |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
changeset
|
611 } |
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
changeset
|
612 |
9936 | 613 if (!data) |
9935
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Merge the AAC encoder from SoC svn. It is still considered experimental.
alexc
parents:
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diff
changeset
|
614 s->last_frame = 1; |
9937
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diego
parents:
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diff
changeset
|
615 memcpy(s->samples, s->samples + 1024 * avctx->channels, |
3e39dbd2d9eb
cosmetics: prettyprinting, K&R style, break overly long lines
diego
parents:
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diff
changeset
|
616 1024 * avctx->channels * sizeof(s->samples[0])); |
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617 return put_bits_count(&s->pb)>>3; |
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618 } |
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619 |
7571 | 620 static av_cold int aac_encode_end(AVCodecContext *avctx) |
621 { | |
622 AACEncContext *s = avctx->priv_data; | |
623 | |
624 ff_mdct_end(&s->mdct1024); | |
625 ff_mdct_end(&s->mdct128); | |
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626 ff_psy_end(&s->psy); |
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627 ff_psy_preprocess_end(s->psypp); |
7571 | 628 av_freep(&s->samples); |
629 av_freep(&s->cpe); | |
630 return 0; | |
631 } | |
632 | |
633 AVCodec aac_encoder = { | |
634 "aac", | |
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635 AVMEDIA_TYPE_AUDIO, |
7571 | 636 CODEC_ID_AAC, |
637 sizeof(AACEncContext), | |
638 aac_encode_init, | |
639 aac_encode_frame, | |
640 aac_encode_end, | |
11786 | 641 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY | CODEC_CAP_EXPERIMENTAL, |
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642 .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7571 | 643 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), |
644 }; |