3 * Copyright (C) 2008 Konstantin Shishkov
5 * This file is part of FFmpeg.
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.
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.
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
27 /***********************************
29 * psy model selection with some option
30 * change greedy codebook search into something more optimal, like Viterbi algorithm
31 * determine run lengths along with codebook
32 ***********************************/
35 #include "bitstream.h"
37 #include "mpeg4audio.h"
43 static const uint8_t swb_size_1024_96[] = {
44 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
45 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
46 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
49 static const uint8_t swb_size_1024_64[] = {
50 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
51 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
52 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
55 static const uint8_t swb_size_1024_48[] = {
56 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
57 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
58 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
62 static const uint8_t swb_size_1024_32[] = {
63 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
64 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
65 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
68 static const uint8_t swb_size_1024_24[] = {
69 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
70 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
71 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
74 static const uint8_t swb_size_1024_16[] = {
75 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
76 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
77 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
80 static const uint8_t swb_size_1024_8[] = {
81 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
82 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
83 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
86 static const uint8_t *swb_size_1024[] = {
87 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
88 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
89 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
90 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
93 static const uint8_t swb_size_128_96[] = {
94 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
97 static const uint8_t swb_size_128_48[] = {
98 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
101 static const uint8_t swb_size_128_24[] = {
102 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
105 static const uint8_t swb_size_128_16[] = {
106 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
109 static const uint8_t swb_size_128_8[] = {
110 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
113 static const uint8_t *swb_size_128[] = {
114 /* the last entry on the following row is swb_size_128_64 but is a
115 duplicate of swb_size_128_96 */
116 swb_size_128_96, swb_size_128_96, swb_size_128_96,
117 swb_size_128_48, swb_size_128_48, swb_size_128_48,
118 swb_size_128_24, swb_size_128_24, swb_size_128_16,
119 swb_size_128_16, swb_size_128_16, swb_size_128_8
122 #define CB_UNSIGNED 0x01 ///< coefficients are coded as absolute values
123 #define CB_PAIRS 0x02 ///< coefficients are grouped into pairs before coding (quads by default)
124 #define CB_ESCAPE 0x04 ///< codebook allows escapes
126 /** spectral coefficients codebook information */
127 static const struct {
128 int16_t maxval; ///< maximum possible value
129 int8_t cb_num; ///< codebook number
130 uint8_t flags; ///< codebook features
132 { 0, -1, CB_UNSIGNED }, // zero codebook
135 { 2, 2, CB_UNSIGNED },
136 { 2, 3, CB_UNSIGNED },
139 { 7, 6, CB_PAIRS | CB_UNSIGNED },
140 { 7, 7, CB_PAIRS | CB_UNSIGNED },
141 { 12, 8, CB_PAIRS | CB_UNSIGNED },
142 { 12, 9, CB_PAIRS | CB_UNSIGNED },
143 { 8191, 10, CB_PAIRS | CB_UNSIGNED | CB_ESCAPE },
144 { -1, -1, 0 }, // reserved
145 { -1, -1, 0 }, // perceptual noise substitution
146 { -1, -1, 0 }, // intensity out-of-phase
147 { -1, -1, 0 }, // intensity in-phase
150 /** default channel configurations */
151 static const uint8_t aac_chan_configs[6][5] = {
152 {1, ID_SCE}, // 1 channel - single channel element
153 {1, ID_CPE}, // 2 channels - channel pair
154 {2, ID_SCE, ID_CPE}, // 3 channels - center + stereo
155 {3, ID_SCE, ID_CPE, ID_SCE}, // 4 channels - front center + stereo + back center
156 {3, ID_SCE, ID_CPE, ID_CPE}, // 5 channels - front center + stereo + back stereo
157 {4, ID_SCE, ID_CPE, ID_CPE, ID_LFE}, // 6 channels - front center + stereo + back stereo + LFE
161 * AAC encoder context
165 MDCTContext mdct1024; ///< long (1024 samples) frame transform context
166 MDCTContext mdct128; ///< short (128 samples) frame transform context
168 DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
169 DECLARE_ALIGNED_16(FFTSample, tmp[1024]); ///< temporary buffer used by MDCT
170 int16_t* samples; ///< saved preprocessed input
172 int samplerate_index; ///< MPEG-4 samplerate index
173 const uint8_t *swb_sizes1024; ///< scalefactor band sizes for long frame
174 int swb_num1024; ///< number of scalefactor bands for long frame
175 const uint8_t *swb_sizes128; ///< scalefactor band sizes for short frame
176 int swb_num128; ///< number of scalefactor bands for short frame
178 ChannelElement *cpe; ///< channel elements
179 AACPsyContext psy; ///< psychoacoustic model context
184 * Make AAC audio config object.
185 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
187 static void put_audio_specific_config(AVCodecContext *avctx)
190 AACEncContext *s = avctx->priv_data;
192 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
193 put_bits(&pb, 5, 2); //object type - AAC-LC
194 put_bits(&pb, 4, s->samplerate_index); //sample rate index
195 put_bits(&pb, 4, avctx->channels);
197 put_bits(&pb, 1, 0); //frame length - 1024 samples
198 put_bits(&pb, 1, 0); //does not depend on core coder
199 put_bits(&pb, 1, 0); //is not extension
203 static av_cold int aac_encode_init(AVCodecContext *avctx)
205 AACEncContext *s = avctx->priv_data;
208 avctx->frame_size = 1024;
210 for(i = 0; i < 16; i++)
211 if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
214 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
217 if(avctx->channels > 6){
218 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
221 s->samplerate_index = i;
222 s->swb_sizes1024 = swb_size_1024[i];
223 s->swb_num1024 = ff_aac_num_swb_1024[i];
224 s->swb_sizes128 = swb_size_128[i];
225 s->swb_num128 = ff_aac_num_swb_128[i];
227 dsputil_init(&s->dsp, avctx);
228 ff_mdct_init(&s->mdct1024, 11, 0);
229 ff_mdct_init(&s->mdct128, 8, 0);
231 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
232 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
233 ff_sine_window_init(ff_aac_sine_long_1024, 1024);
234 ff_sine_window_init(ff_aac_sine_short_128, 128);
236 s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
237 s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
238 if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP, aac_chan_configs[avctx->channels-1][0], 0, s->swb_sizes1024, s->swb_num1024, s->swb_sizes128, s->swb_num128) < 0){
239 av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n");
242 avctx->extradata = av_malloc(2);
243 avctx->extradata_size = 2;
244 put_audio_specific_config(avctx);
249 * Encode ics_info element.
250 * @see Table 4.6 (syntax of ics_info)
252 static void put_ics_info(AVCodecContext *avctx, IndividualChannelStream *info)
254 AACEncContext *s = avctx->priv_data;
257 put_bits(&s->pb, 1, 0); // ics_reserved bit
258 put_bits(&s->pb, 2, info->window_sequence[0]);
259 put_bits(&s->pb, 1, info->use_kb_window[0]);
260 if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){
261 put_bits(&s->pb, 6, info->max_sfb);
262 put_bits(&s->pb, 1, 0); // no prediction
264 put_bits(&s->pb, 4, info->max_sfb);
265 for(i = 1; i < info->num_windows; i++)
266 put_bits(&s->pb, 1, info->group_len[i]);
271 * Write some auxiliary information about the created AAC file.
273 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
275 int i, namelen, padbits;
277 namelen = strlen(name) + 2;
278 put_bits(&s->pb, 3, ID_FIL);
279 put_bits(&s->pb, 4, FFMIN(namelen, 15));
281 put_bits(&s->pb, 8, namelen - 16);
282 put_bits(&s->pb, 4, 0); //extension type - filler
283 padbits = 8 - (put_bits_count(&s->pb) & 7);
284 align_put_bits(&s->pb);
285 for(i = 0; i < namelen - 2; i++)
286 put_bits(&s->pb, 8, name[i]);
287 put_bits(&s->pb, 12 - padbits, 0);
290 static av_cold int aac_encode_end(AVCodecContext *avctx)
292 AACEncContext *s = avctx->priv_data;
294 ff_mdct_end(&s->mdct1024);
295 ff_mdct_end(&s->mdct128);
296 ff_aac_psy_end(&s->psy);
297 av_freep(&s->samples);
302 AVCodec aac_encoder = {
306 sizeof(AACEncContext),
310 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
311 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
312 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),