falkTX
/
FFmpeg
mirror of https://github.com/falkTX/FFmpeg.git
1
0
Fork 0
Code Issues 0 Releases 338 Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
74077 Commits
32 Branches
283MB
Tree: e36db49b7b
Branches Tags
${ item.name }
Create branch ${ searchTerm }
from 'e36db49b7b'
${ noResults }
FFmpeg/libavcodec/aacenc.c

952 lines
34KB
Raw Blame History 

  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. /**

  23.  * @file

  24.  * AAC encoder

  25.  */

  26. 

  27. /***********************************

  28.  *              TODOs:

  29.  * add sane pulse detection

  30.  * add temporal noise shaping

  31.  ***********************************/

  32. 

  33. #include "libavutil/float_dsp.h"

  34. #include "libavutil/opt.h"

  35. #include "avcodec.h"

  36. #include "put_bits.h"

  37. #include "internal.h"

  38. #include "mpeg4audio.h"

  39. #include "kbdwin.h"

  40. #include "sinewin.h"

  41. 

  42. #include "aac.h"

  43. #include "aactab.h"

  44. #include "aacenc.h"

  45. 

  46. #include "psymodel.h"

  47. 

  48. #define AAC_MAX_CHANNELS 6

  49. 

  50. #define ERROR_IF(cond, ...) \

  51.     if (cond) { \

  52.         av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \

  53.         return AVERROR(EINVAL); \

  54.     }

  55. 

  56. #define WARN_IF(cond, ...) \

  57.     if (cond) { \

  58.         av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \

  59.     }

  60. 

  61. 

  62. static const uint8_t swb_size_1024_96[] = {

  63.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,

  64.     12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,

  65.     64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64

  66. };

  67. 

  68. static const uint8_t swb_size_1024_64[] = {

  69.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,

  70.     12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,

  71.     40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40

  72. };

  73. 

  74. static const uint8_t swb_size_1024_48[] = {

  75.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,

  76.     12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,

  77.     32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,

  78.     96

  79. };

  80. 

  81. static const uint8_t swb_size_1024_32[] = {

  82.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,

  83.     12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,

  84.     32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32

  85. };

  86. 

  87. static const uint8_t swb_size_1024_24[] = {

  88.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,

  89.     12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,

  90.     32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64

  91. };

  92. 

  93. static const uint8_t swb_size_1024_16[] = {

  94.     8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,

  95.     12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,

  96.     32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64

  97. };

  98. 

  99. static const uint8_t swb_size_1024_8[] = {

  100.     12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,

  101.     16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,

  102.     32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80

  103. };

  104. 

  105. static const uint8_t *swb_size_1024[] = {

  106.     swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,

  107.     swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,

  108.     swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,

  109.     swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,

  110.     swb_size_1024_8

  111. };

  112. 

  113. static const uint8_t swb_size_128_96[] = {

  114.     4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36

  115. };

  116. 

  117. static const uint8_t swb_size_128_48[] = {

  118.     4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16

  119. };

  120. 

  121. static const uint8_t swb_size_128_24[] = {

  122.     4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20

  123. };

  124. 

  125. static const uint8_t swb_size_128_16[] = {

  126.     4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20

  127. };

  128. 

  129. static const uint8_t swb_size_128_8[] = {

  130.     4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20

  131. };

  132. 

  133. static const uint8_t *swb_size_128[] = {

  134.     /* the last entry on the following row is swb_size_128_64 but is a

  135.        duplicate of swb_size_128_96 */

  136.     swb_size_128_96, swb_size_128_96, swb_size_128_96,

  137.     swb_size_128_48, swb_size_128_48, swb_size_128_48,

  138.     swb_size_128_24, swb_size_128_24, swb_size_128_16,

  139.     swb_size_128_16, swb_size_128_16, swb_size_128_8,

  140.     swb_size_128_8

  141. };

  142. 

  143. /** default channel configurations */

  144. static const uint8_t aac_chan_configs[6][5] = {

  145.  {1, TYPE_SCE},                               // 1 channel  - single channel element

  146.  {1, TYPE_CPE},                               // 2 channels - channel pair

  147.  {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo

  148.  {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center

  149.  {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo

  150.  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE

  151. };

  152. 

  153. /**

  154.  * Table to remap channels from libavcodec's default order to AAC order.

  155.  */

  156. static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {

  157.     { 0 },

  158.     { 0, 1 },

  159.     { 2, 0, 1 },

  160.     { 2, 0, 1, 3 },

  161.     { 2, 0, 1, 3, 4 },

  162.     { 2, 0, 1, 4, 5, 3 },

  163. };

  164. 

  165. /**

  166.  * Make AAC audio config object.

  167.  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"

  168.  */

  169. static void put_audio_specific_config(AVCodecContext *avctx)

  170. {

  171.     PutBitContext pb;

  172.     AACEncContext *s = avctx->priv_data;

  173. 

  174.     init_put_bits(&pb, avctx->extradata, avctx->extradata_size);

  175.     put_bits(&pb, 5, 2); //object type - AAC-LC

  176.     put_bits(&pb, 4, s->samplerate_index); //sample rate index

  177.     put_bits(&pb, 4, s->channels);

  178.     //GASpecificConfig

  179.     put_bits(&pb, 1, 0); //frame length - 1024 samples

  180.     put_bits(&pb, 1, 0); //does not depend on core coder

  181.     put_bits(&pb, 1, 0); //is not extension

  182. 

  183.     //Explicitly Mark SBR absent

  184.     put_bits(&pb, 11, 0x2b7); //sync extension

  185.     put_bits(&pb, 5,  AOT_SBR);

  186.     put_bits(&pb, 1,  0);

  187.     flush_put_bits(&pb);

  188. }

  189. 

  190. #define WINDOW_FUNC(type) \

  191. static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \

  192.                                     SingleChannelElement *sce, \

  193.                                     const float *audio)

  194. 

  195. WINDOW_FUNC(only_long)

  196. {

  197.     const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;

  198.     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;

  199.     float *out = sce->ret_buf;

  200. 

  201.     fdsp->vector_fmul        (out,        audio,        lwindow, 1024);

  202.     fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);

  203. }

  204. 

  205. WINDOW_FUNC(long_start)

  206. {

  207.     const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;

  208.     const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;

  209.     float *out = sce->ret_buf;

  210. 

  211.     fdsp->vector_fmul(out, audio, lwindow, 1024);

  212.     memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);

  213.     fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);

  214.     memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);

  215. }

  216. 

  217. WINDOW_FUNC(long_stop)

  218. {

  219.     const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;

  220.     const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;

  221.     float *out = sce->ret_buf;

  222. 

  223.     memset(out, 0, sizeof(out[0]) * 448);

  224.     fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);

  225.     memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);

  226.     fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);

  227. }

  228. 

  229. WINDOW_FUNC(eight_short)

  230. {

  231.     const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;

  232.     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;

  233.     const float *in = audio + 448;

  234.     float *out = sce->ret_buf;

  235.     int w;

  236. 

  237.     for (w = 0; w < 8; w++) {

  238.         fdsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);

  239.         out += 128;

  240.         in  += 128;

  241.         fdsp->vector_fmul_reverse(out, in, swindow, 128);

  242.         out += 128;

  243.     }

  244. }

  245. 

  246. static void (*const apply_window[4])(AVFloatDSPContext *fdsp,

  247.                                      SingleChannelElement *sce,

  248.                                      const float *audio) = {

  249.     [ONLY_LONG_SEQUENCE]   = apply_only_long_window,

  250.     [LONG_START_SEQUENCE]  = apply_long_start_window,

  251.     [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,

  252.     [LONG_STOP_SEQUENCE]   = apply_long_stop_window

  253. };

  254. 

  255. static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,

  256.                                   float *audio)

  257. {

  258.     int i;

  259.     float *output = sce->ret_buf;

  260. 

  261.     apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);

  262. 

  263.     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)

  264.         s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);

  265.     else

  266.         for (i = 0; i < 1024; i += 128)

  267.             s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);

  268.     memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);

  269.     memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));

  270. }

  271. 

  272. /**

  273.  * Encode ics_info element.

  274.  * @see Table 4.6 (syntax of ics_info)

  275.  */

  276. static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)

  277. {

  278.     int w;

  279. 

  280.     put_bits(&s->pb, 1, 0);                // ics_reserved bit

  281.     put_bits(&s->pb, 2, info->window_sequence[0]);

  282.     put_bits(&s->pb, 1, info->use_kb_window[0]);

  283.     if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {

  284.         put_bits(&s->pb, 6, info->max_sfb);

  285.         put_bits(&s->pb, 1, 0);            // no prediction

  286.     } else {

  287.         put_bits(&s->pb, 4, info->max_sfb);

  288.         for (w = 1; w < 8; w++)

  289.             put_bits(&s->pb, 1, !info->group_len[w]);

  290.     }

  291. }

  292. 

  293. /**

  294.  * Encode MS data.

  295.  * @see 4.6.8.1 "Joint Coding - M/S Stereo"

  296.  */

  297. static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)

  298. {

  299.     int i, w;

  300. 

  301.     put_bits(pb, 2, cpe->ms_mode);

  302.     if (cpe->ms_mode == 1)

  303.         for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])

  304.             for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)

  305.                 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);

  306. }

  307. 

  308. /**

  309.  * Produce integer coefficients from scalefactors provided by the model.

  310.  */

  311. static void adjust_frame_information(ChannelElement *cpe, int chans)

  312. {

  313.     int i, w, w2, g, ch;

  314.     int maxsfb, cmaxsfb;

  315.     IndividualChannelStream *ics;

  316. 

  317.     if (cpe->common_window) {

  318.         ics = &cpe->ch[0].ics;

  319.         for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {

  320.             for (w2 =  0; w2 < ics->group_len[w]; w2++) {

  321.                 int start = (w+w2) * 128;

  322.                 for (g = 0; g < ics->num_swb; g++) {

  323.                     //apply Intensity stereo coeffs transformation

  324.                     if (cpe->is_mask[w*16 + g]) {

  325.                         int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);

  326.                         float scale = cpe->ch[0].is_ener[w*16+g];

  327.                         for (i = 0; i < ics->swb_sizes[g]; i++) {

  328.                             cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + p*cpe->ch[1].pcoeffs[start+i]) * scale;

  329.                             cpe->ch[1].coeffs[start+i] = 0.0f;

  330.                         }

  331.                     } else if (cpe->ms_mask[w*16 + g] &&

  332.                                cpe->ch[0].band_type[w*16 + g] < NOISE_BT &&

  333.                                cpe->ch[1].band_type[w*16 + g] < NOISE_BT) {

  334.                         for (i = 0; i < ics->swb_sizes[g]; i++) {

  335.                             cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + cpe->ch[1].pcoeffs[start+i]) * 0.5f;

  336.                             cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].pcoeffs[start+i];

  337.                         }

  338.                     }

  339.                     start += ics->swb_sizes[g];

  340.                 }

  341.             }

  342.         }

  343.     }

  344. 

  345.     for (ch = 0; ch < chans; ch++) {

  346.         IndividualChannelStream *ics = &cpe->ch[ch].ics;

  347.         maxsfb = 0;

  348.         cpe->ch[ch].pulse.num_pulse = 0;

  349.         for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {

  350.             for (w2 =  0; w2 < ics->group_len[w]; w2++) {

  351.                 for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)

  352.                     ;

  353.                 maxsfb = FFMAX(maxsfb, cmaxsfb);

  354.             }

  355.         }

  356.         ics->max_sfb = maxsfb;

  357. 

  358.         //adjust zero bands for window groups

  359.         for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {

  360.             for (g = 0; g < ics->max_sfb; g++) {

  361.                 i = 1;

  362.                 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {

  363.                     if (!cpe->ch[ch].zeroes[w2*16 + g]) {

  364.                         i = 0;

  365.                         break;

  366.                     }

  367.                 }

  368.                 cpe->ch[ch].zeroes[w*16 + g] = i;

  369.             }

  370.         }

  371.     }

  372. 

  373.     if (chans > 1 && cpe->common_window) {

  374.         IndividualChannelStream *ics0 = &cpe->ch[0].ics;

  375.         IndividualChannelStream *ics1 = &cpe->ch[1].ics;

  376.         int msc = 0;

  377.         ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);

  378.         ics1->max_sfb = ics0->max_sfb;

  379.         for (w = 0; w < ics0->num_windows*16; w += 16)

  380.             for (i = 0; i < ics0->max_sfb; i++)

  381.                 if (cpe->ms_mask[w+i])

  382.                     msc++;

  383.         if (msc == 0 || ics0->max_sfb == 0)

  384.             cpe->ms_mode = 0;

  385.         else

  386.             cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;

  387.     }

  388. }

  389. 

  390. /**

  391.  * Encode scalefactor band coding type.

  392.  */

  393. static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)

  394. {

  395.     int w;

  396. 

  397.     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])

  398.         s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);

  399. }

  400. 

  401. /**

  402.  * Encode scalefactors.

  403.  */

  404. static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,

  405.                                  SingleChannelElement *sce)

  406. {

  407.     int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;

  408.     int off_is = 0, noise_flag = 1;

  409.     int i, w;

  410. 

  411.     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {

  412.         for (i = 0; i < sce->ics.max_sfb; i++) {

  413.             if (!sce->zeroes[w*16 + i]) {

  414.                 if (sce->band_type[w*16 + i] == NOISE_BT) {

  415.                     diff = sce->sf_idx[w*16 + i] - off_pns;

  416.                     off_pns = sce->sf_idx[w*16 + i];

  417.                     if (noise_flag-- > 0) {

  418.                         put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);

  419.                         continue;

  420.                     }

  421.                 } else if (sce->band_type[w*16 + i] == INTENSITY_BT  ||

  422.                            sce->band_type[w*16 + i] == INTENSITY_BT2) {

  423.                     diff = sce->sf_idx[w*16 + i] - off_is;

  424.                     off_is = sce->sf_idx[w*16 + i];

  425.                 } else {

  426.                     diff = sce->sf_idx[w*16 + i] - off_sf;

  427.                     off_sf = sce->sf_idx[w*16 + i];

  428.                 }

  429.                 diff += SCALE_DIFF_ZERO;

  430.                 av_assert0(diff >= 0 && diff <= 120);

  431.                 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);

  432.             }

  433.         }

  434.     }

  435. }

  436. 

  437. /**

  438.  * Encode pulse data.

  439.  */

  440. static void encode_pulses(AACEncContext *s, Pulse *pulse)

  441. {

  442.     int i;

  443. 

  444.     put_bits(&s->pb, 1, !!pulse->num_pulse);

  445.     if (!pulse->num_pulse)

  446.         return;

  447. 

  448.     put_bits(&s->pb, 2, pulse->num_pulse - 1);

  449.     put_bits(&s->pb, 6, pulse->start);

  450.     for (i = 0; i < pulse->num_pulse; i++) {

  451.         put_bits(&s->pb, 5, pulse->pos[i]);

  452.         put_bits(&s->pb, 4, pulse->amp[i]);

  453.     }

  454. }

  455. 

  456. /**

  457.  * Encode spectral coefficients processed by psychoacoustic model.

  458.  */

  459. static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)

  460. {

  461.     int start, i, w, w2;

  462. 

  463.     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {

  464.         start = 0;

  465.         for (i = 0; i < sce->ics.max_sfb; i++) {

  466.             if (sce->zeroes[w*16 + i]) {

  467.                 start += sce->ics.swb_sizes[i];

  468.                 continue;

  469.             }

  470.             for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)

  471.                 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,

  472.                                                    sce->ics.swb_sizes[i],

  473.                                                    sce->sf_idx[w*16 + i],

  474.                                                    sce->band_type[w*16 + i],

  475.                                                    s->lambda, sce->ics.window_clipping[w]);

  476.             start += sce->ics.swb_sizes[i];

  477.         }

  478.     }

  479. }

  480. 

  481. /**

  482.  * Downscale spectral coefficients for near-clipping windows to avoid artifacts

  483.  */

  484. static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)

  485. {

  486.     int start, i, j, w;

  487. 

  488.     if (sce->ics.clip_avoidance_factor < 1.0f) {

  489.         for (w = 0; w < sce->ics.num_windows; w++) {

  490.             start = 0;

  491.             for (i = 0; i < sce->ics.max_sfb; i++) {

  492.                 float *swb_coeffs = sce->coeffs + start + w*128;

  493.                 for (j = 0; j < sce->ics.swb_sizes[i]; j++)

  494.                     swb_coeffs[j] *= sce->ics.clip_avoidance_factor;

  495.                 start += sce->ics.swb_sizes[i];

  496.             }

  497.         }

  498.     }

  499. }

  500. 

  501. /**

  502.  * Encode one channel of audio data.

  503.  */

  504. static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,

  505.                                      SingleChannelElement *sce,

  506.                                      int common_window)

  507. {

  508.     put_bits(&s->pb, 8, sce->sf_idx[0]);

  509.     if (!common_window)

  510.         put_ics_info(s, &sce->ics);

  511.     encode_band_info(s, sce);

  512.     encode_scale_factors(avctx, s, sce);

  513.     encode_pulses(s, &sce->pulse);

  514.     put_bits(&s->pb, 1, 0); //tns

  515.     put_bits(&s->pb, 1, 0); //ssr

  516.     encode_spectral_coeffs(s, sce);

  517.     return 0;

  518. }

  519. 

  520. /**

  521.  * Write some auxiliary information about the created AAC file.

  522.  */

  523. static void put_bitstream_info(AACEncContext *s, const char *name)

  524. {

  525.     int i, namelen, padbits;

  526. 

  527.     namelen = strlen(name) + 2;

  528.     put_bits(&s->pb, 3, TYPE_FIL);

  529.     put_bits(&s->pb, 4, FFMIN(namelen, 15));

  530.     if (namelen >= 15)

  531.         put_bits(&s->pb, 8, namelen - 14);

  532.     put_bits(&s->pb, 4, 0); //extension type - filler

  533.     padbits = -put_bits_count(&s->pb) & 7;

  534.     avpriv_align_put_bits(&s->pb);

  535.     for (i = 0; i < namelen - 2; i++)

  536.         put_bits(&s->pb, 8, name[i]);

  537.     put_bits(&s->pb, 12 - padbits, 0);

  538. }

  539. 

  540. /*

  541.  * Copy input samples.

  542.  * Channels are reordered from libavcodec's default order to AAC order.

  543.  */

  544. static void copy_input_samples(AACEncContext *s, const AVFrame *frame)

  545. {

  546.     int ch;

  547.     int end = 2048 + (frame ? frame->nb_samples : 0);

  548.     const uint8_t *channel_map = aac_chan_maps[s->channels - 1];

  549. 

  550.     /* copy and remap input samples */

  551.     for (ch = 0; ch < s->channels; ch++) {

  552.         /* copy last 1024 samples of previous frame to the start of the current frame */

  553.         memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));

  554. 

  555.         /* copy new samples and zero any remaining samples */

  556.         if (frame) {

  557.             memcpy(&s->planar_samples[ch][2048],

  558.                    frame->extended_data[channel_map[ch]],

  559.                    frame->nb_samples * sizeof(s->planar_samples[0][0]));

  560.         }

  561.         memset(&s->planar_samples[ch][end], 0,

  562.                (3072 - end) * sizeof(s->planar_samples[0][0]));

  563.     }

  564. }

  565. 

  566. static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,

  567.                             const AVFrame *frame, int *got_packet_ptr)

  568. {

  569.     AACEncContext *s = avctx->priv_data;

  570.     float **samples = s->planar_samples, *samples2, *la, *overlap;

  571.     ChannelElement *cpe;

  572.     int i, ch, w, g, chans, tag, start_ch, ret, ms_mode = 0, is_mode = 0;

  573.     int chan_el_counter[4];

  574.     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];

  575. 

  576.     if (s->last_frame == 2)

  577.         return 0;

  578. 

  579.     /* add current frame to queue */

  580.     if (frame) {

  581.         if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)

  582.             return ret;

  583.     }

  584. 

  585.     copy_input_samples(s, frame);

  586.     if (s->psypp)

  587.         ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);

  588. 

  589.     if (!avctx->frame_number)

  590.         return 0;

  591. 

  592.     start_ch = 0;

  593.     for (i = 0; i < s->chan_map[0]; i++) {

  594.         FFPsyWindowInfo* wi = windows + start_ch;

  595.         tag      = s->chan_map[i+1];

  596.         chans    = tag == TYPE_CPE ? 2 : 1;

  597.         cpe      = &s->cpe[i];

  598.         for (ch = 0; ch < chans; ch++) {

  599.             IndividualChannelStream *ics = &cpe->ch[ch].ics;

  600.             int cur_channel = start_ch + ch;

  601.             float clip_avoidance_factor;

  602.             overlap  = &samples[cur_channel][0];

  603.             samples2 = overlap + 1024;

  604.             la       = samples2 + (448+64);

  605.             if (!frame)

  606.                 la = NULL;

  607.             if (tag == TYPE_LFE) {

  608.                 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;

  609.                 wi[ch].window_shape   = 0;

  610.                 wi[ch].num_windows    = 1;

  611.                 wi[ch].grouping[0]    = 1;

  612. 

  613.                 /* Only the lowest 12 coefficients are used in a LFE channel.

  614.                  * The expression below results in only the bottom 8 coefficients

  615.                  * being used for 11.025kHz to 16kHz sample rates.

  616.                  */

  617.                 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;

  618.             } else {

  619.                 wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,

  620.                                               ics->window_sequence[0]);

  621.             }

  622.             ics->window_sequence[1] = ics->window_sequence[0];

  623.             ics->window_sequence[0] = wi[ch].window_type[0];

  624.             ics->use_kb_window[1]   = ics->use_kb_window[0];

  625.             ics->use_kb_window[0]   = wi[ch].window_shape;

  626.             ics->num_windows        = wi[ch].num_windows;

  627.             ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];

  628.             ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];

  629.             clip_avoidance_factor = 0.0f;

  630.             for (w = 0; w < ics->num_windows; w++)

  631.                 ics->group_len[w] = wi[ch].grouping[w];

  632.             for (w = 0; w < ics->num_windows; w++) {

  633.                 if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {

  634.                     ics->window_clipping[w] = 1;

  635.                     clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);

  636.                 } else {

  637.                     ics->window_clipping[w] = 0;

  638.                 }

  639.             }

  640.             if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {

  641.                 ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;

  642.             } else {

  643.                 ics->clip_avoidance_factor = 1.0f;

  644.             }

  645. 

  646.             apply_window_and_mdct(s, &cpe->ch[ch], overlap);

  647.             if (isnan(cpe->ch->coeffs[0])) {

  648.                 av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");

  649.                 return AVERROR(EINVAL);

  650.             }

  651.             avoid_clipping(s, &cpe->ch[ch]);

  652.         }

  653.         start_ch += chans;

  654.     }

  655.     if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels, 0)) < 0)

  656.         return ret;

  657.     do {

  658.         int frame_bits;

  659. 

  660.         init_put_bits(&s->pb, avpkt->data, avpkt->size);

  661. 

  662.         if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))

  663.             put_bitstream_info(s, LIBAVCODEC_IDENT);

  664.         start_ch = 0;

  665.         memset(chan_el_counter, 0, sizeof(chan_el_counter));

  666.         for (i = 0; i < s->chan_map[0]; i++) {

  667.             FFPsyWindowInfo* wi = windows + start_ch;

  668.             const float *coeffs[2];

  669.             tag      = s->chan_map[i+1];

  670.             chans    = tag == TYPE_CPE ? 2 : 1;

  671.             cpe      = &s->cpe[i];

  672.             memset(cpe->is_mask, 0, sizeof(cpe->is_mask));

  673.             memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));

  674.             put_bits(&s->pb, 3, tag);

  675.             put_bits(&s->pb, 4, chan_el_counter[tag]++);

  676.             for (ch = 0; ch < chans; ch++)

  677.                 coeffs[ch] = cpe->ch[ch].coeffs;

  678.             s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);

  679.             for (ch = 0; ch < chans; ch++) {

  680.                 s->cur_channel = start_ch + ch;

  681.                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);

  682.             }

  683.             cpe->common_window = 0;

  684.             if (chans > 1

  685.                 && wi[0].window_type[0] == wi[1].window_type[0]

  686.                 && wi[0].window_shape   == wi[1].window_shape) {

  687. 

  688.                 cpe->common_window = 1;

  689.                 for (w = 0; w < wi[0].num_windows; w++) {

  690.                     if (wi[0].grouping[w] != wi[1].grouping[w]) {

  691.                         cpe->common_window = 0;

  692.                         break;

  693.                     }

  694.                 }

  695.             }

  696.             if (s->options.pns && s->coder->search_for_pns) {

  697.                 for (ch = 0; ch < chans; ch++) {

  698.                     s->cur_channel = start_ch + ch;

  699.                     s->coder->search_for_pns(s, avctx, &cpe->ch[ch], s->lambda);

  700.                 }

  701.             }

  702.             s->cur_channel = start_ch;

  703.             if (s->options.stereo_mode && cpe->common_window) {

  704.                 if (s->options.stereo_mode > 0) {

  705.                     IndividualChannelStream *ics = &cpe->ch[0].ics;

  706.                     for (w = 0; w < ics->num_windows; w += ics->group_len[w])

  707.                         for (g = 0;  g < ics->num_swb; g++)

  708.                             cpe->ms_mask[w*16+g] = 1;

  709.                 } else if (s->coder->search_for_ms) {

  710.                     s->coder->search_for_ms(s, cpe, s->lambda);

  711.                 }

  712.             }

  713.             if (chans > 1 && s->options.intensity_stereo && s->coder->search_for_is) {

  714.                 s->coder->search_for_is(s, avctx, cpe, s->lambda);

  715.                 if (cpe->is_mode) is_mode = 1;

  716.             }

  717.             if (s->coder->set_special_band_scalefactors)

  718.                 for (ch = 0; ch < chans; ch++)

  719.                     s->coder->set_special_band_scalefactors(s, &cpe->ch[ch]);

  720.             adjust_frame_information(cpe, chans);

  721.             if (chans == 2) {

  722.                 put_bits(&s->pb, 1, cpe->common_window);

  723.                 if (cpe->common_window) {

  724.                     put_ics_info(s, &cpe->ch[0].ics);

  725.                     encode_ms_info(&s->pb, cpe);

  726.                     if (cpe->ms_mode) ms_mode = 1;

  727.                 }

  728.             }

  729.             for (ch = 0; ch < chans; ch++) {

  730.                 s->cur_channel = start_ch + ch;

  731.                 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);

  732.             }

  733.             start_ch += chans;

  734.         }

  735. 

  736.         frame_bits = put_bits_count(&s->pb);

  737.         if (frame_bits <= 6144 * s->channels - 3) {

  738.             s->psy.bitres.bits = frame_bits / s->channels;

  739.             break;

  740.         }

  741.         if (is_mode || ms_mode) {

  742.             for (i = 0; i < s->chan_map[0]; i++) {

  743.                 // Must restore coeffs

  744.                 chans = tag == TYPE_CPE ? 2 : 1;

  745.                 cpe = &s->cpe[i];

  746.                 for (ch = 0; ch < chans; ch++)

  747.                     memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));

  748.             }

  749.         }

  750. 

  751.         s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;

  752. 

  753.     } while (1);

  754. 

  755.     put_bits(&s->pb, 3, TYPE_END);

  756.     flush_put_bits(&s->pb);

  757.     avctx->frame_bits = put_bits_count(&s->pb);

  758. 

  759.     // rate control stuff

  760.     if (!(avctx->flags & CODEC_FLAG_QSCALE)) {

  761.         float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;

  762.         s->lambda *= ratio;

  763.         s->lambda = FFMIN(s->lambda, 65536.f);

  764.     }

  765. 

  766.     if (!frame)

  767.         s->last_frame++;

  768. 

  769.     ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,

  770.                        &avpkt->duration);

  771. 

  772.     avpkt->size = put_bits_count(&s->pb) >> 3;

  773.     *got_packet_ptr = 1;

  774.     return 0;

  775. }

  776. 

  777. static av_cold int aac_encode_end(AVCodecContext *avctx)

  778. {

  779.     AACEncContext *s = avctx->priv_data;

  780. 

  781.     ff_mdct_end(&s->mdct1024);

  782.     ff_mdct_end(&s->mdct128);

  783.     ff_psy_end(&s->psy);

  784.     if (s->psypp)

  785.         ff_psy_preprocess_end(s->psypp);

  786.     av_freep(&s->buffer.samples);

  787.     av_freep(&s->cpe);

  788.     av_freep(&s->fdsp);

  789.     ff_af_queue_close(&s->afq);

  790.     return 0;

  791. }

  792. 

  793. static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)

  794. {

  795.     int ret = 0;

  796. 

  797.     s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);

  798.     if (!s->fdsp)

  799.         return AVERROR(ENOMEM);

  800. 

  801.     // window init

  802.     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);

  803.     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);

  804.     ff_init_ff_sine_windows(10);

  805.     ff_init_ff_sine_windows(7);

  806. 

  807.     if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)

  808.         return ret;

  809.     if ((ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0)) < 0)

  810.         return ret;

  811. 

  812.     return 0;

  813. }

  814. 

  815. static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)

  816. {

  817.     int ch;

  818.     FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);

  819.     FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);

  820.     FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);

  821. 

  822.     for(ch = 0; ch < s->channels; ch++)

  823.         s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;

  824. 

  825.     return 0;

  826. alloc_fail:

  827.     return AVERROR(ENOMEM);

  828. }

  829. 

  830. static av_cold int aac_encode_init(AVCodecContext *avctx)

  831. {

  832.     AACEncContext *s = avctx->priv_data;

  833.     int i, ret = 0;

  834.     const uint8_t *sizes[2];

  835.     uint8_t grouping[AAC_MAX_CHANNELS];

  836.     int lengths[2];

  837. 

  838.     avctx->frame_size = 1024;

  839. 

  840.     for (i = 0; i < 16; i++)

  841.         if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])

  842.             break;

  843. 

  844.     s->channels = avctx->channels;

  845. 

  846.     ERROR_IF(i == 16

  847.                 || i >= (sizeof(swb_size_1024) / sizeof(*swb_size_1024))

  848.                 || i >= (sizeof(swb_size_128) / sizeof(*swb_size_128)),

  849.              "Unsupported sample rate %d\n", avctx->sample_rate);

  850.     ERROR_IF(s->channels > AAC_MAX_CHANNELS,

  851.              "Unsupported number of channels: %d\n", s->channels);

  852.     ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,

  853.              "Unsupported profile %d\n", avctx->profile);

  854.     WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,

  855.              "Too many bits per frame requested, clamping to max\n");

  856. 

  857.     avctx->bit_rate = (int)FFMIN(

  858.         6144 * s->channels / 1024.0 * avctx->sample_rate,

  859.         avctx->bit_rate);

  860. 

  861.     s->samplerate_index = i;

  862. 

  863.     s->chan_map = aac_chan_configs[s->channels-1];

  864. 

  865.     if ((ret = dsp_init(avctx, s)) < 0)

  866.         goto fail;

  867. 

  868.     if ((ret = alloc_buffers(avctx, s)) < 0)

  869.         goto fail;

  870. 

  871.     avctx->extradata_size = 5;

  872.     put_audio_specific_config(avctx);

  873. 

  874.     sizes[0]   = swb_size_1024[i];

  875.     sizes[1]   = swb_size_128[i];

  876.     lengths[0] = ff_aac_num_swb_1024[i];

  877.     lengths[1] = ff_aac_num_swb_128[i];

  878.     for (i = 0; i < s->chan_map[0]; i++)

  879.         grouping[i] = s->chan_map[i + 1] == TYPE_CPE;

  880.     if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,

  881.                            s->chan_map[0], grouping)) < 0)

  882.         goto fail;

  883.     s->psypp = ff_psy_preprocess_init(avctx);

  884.     s->coder = &ff_aac_coders[s->options.aac_coder];

  885. 

  886.     if (HAVE_MIPSDSPR1)

  887.         ff_aac_coder_init_mips(s);

  888. 

  889.     s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;

  890. 

  891.     ff_aac_tableinit();

  892. 

  893.     avctx->initial_padding = 1024;

  894.     ff_af_queue_init(avctx, &s->afq);

  895. 

  896.     return 0;

  897. fail:

  898.     aac_encode_end(avctx);

  899.     return ret;

  900. }

  901. 

  902. #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM

  903. static const AVOption aacenc_options[] = {

  904.     {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},

  905.         {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

  906.         {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

  907.         {"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},

  908.     {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},

  909.         {"faac",     "FAAC-inspired method",      0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},

  910.         {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},

  911.         {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},

  912.         {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},

  913.     {"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pns"},

  914.         {"disable",  "Disable perceptual noise substitution", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},

  915.         {"enable",   "Enable perceptual noise substitution",  0, AV_OPT_TYPE_CONST, {.i64 =  1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},

  916.     {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "intensity_stereo"},

  917.         {"disable",  "Disable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},

  918.         {"enable",   "Enable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},

  919.     {NULL}

  920. };

  921. 

  922. static const AVClass aacenc_class = {

  923.     "AAC encoder",

  924.     av_default_item_name,

  925.     aacenc_options,

  926.     LIBAVUTIL_VERSION_INT,

  927. };

  928. 

  929. /* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build

  930.  * failures */

  931. static const int mpeg4audio_sample_rates[16] = {

  932.     96000, 88200, 64000, 48000, 44100, 32000,

  933.     24000, 22050, 16000, 12000, 11025, 8000, 7350

  934. };

  935. 

  936. AVCodec ff_aac_encoder = {

  937.     .name           = "aac",

  938.     .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),

  939.     .type           = AVMEDIA_TYPE_AUDIO,

  940.     .id             = AV_CODEC_ID_AAC,

  941.     .priv_data_size = sizeof(AACEncContext),

  942.     .init           = aac_encode_init,

  943.     .encode2        = aac_encode_frame,

  944.     .close          = aac_encode_end,

  945.     .supported_samplerates = mpeg4audio_sample_rates,

  946.     .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |

  947.                       CODEC_CAP_EXPERIMENTAL,

  948.     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,

  949.                                                      AV_SAMPLE_FMT_NONE },

  950.     .priv_class     = &aacenc_class,

  951. };