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  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. float ff_aac_pow34sf_tab[428];
  57.  
  58. static const uint8_t swb_size_1024_96[] = {
  59.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
  60.     12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
  61.     64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
  62. };
  63.  
  64. static const uint8_t swb_size_1024_64[] = {
  65.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
  66.     12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
  67.     40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
  68. };
  69.  
  70. static const uint8_t swb_size_1024_48[] = {
  71.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
  72.     12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
  73.     32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
  74.     96
  75. };
  76.  
  77. static const uint8_t swb_size_1024_32[] = {
  78.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
  79.     12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
  80.     32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
  81. };
  82.  
  83. static const uint8_t swb_size_1024_24[] = {
  84.     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
  85.     12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
  86.     32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
  87. };
  88.  
  89. static const uint8_t swb_size_1024_16[] = {
  90.     8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
  91.     12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
  92.     32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
  93. };
  94.  
  95. static const uint8_t swb_size_1024_8[] = {
  96.     12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
  97.     16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
  98.     32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
  99. };
  100.  
  101. static const uint8_t *swb_size_1024[] = {
  102.     swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
  103.     swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
  104.     swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
  105.     swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
  106. };
  107.  
  108. static const uint8_t swb_size_128_96[] = {
  109.     4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
  110. };
  111.  
  112. static const uint8_t swb_size_128_48[] = {
  113.     4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
  114. };
  115.  
  116. static const uint8_t swb_size_128_24[] = {
  117.     4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
  118. };
  119.  
  120. static const uint8_t swb_size_128_16[] = {
  121.     4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
  122. };
  123.  
  124. static const uint8_t swb_size_128_8[] = {
  125.     4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
  126. };
  127.  
  128. static const uint8_t *swb_size_128[] = {
  129.     /* the last entry on the following row is swb_size_128_64 but is a
  130.        duplicate of swb_size_128_96 */
  131.     swb_size_128_96, swb_size_128_96, swb_size_128_96,
  132.     swb_size_128_48, swb_size_128_48, swb_size_128_48,
  133.     swb_size_128_24, swb_size_128_24, swb_size_128_16,
  134.     swb_size_128_16, swb_size_128_16, swb_size_128_8
  135. };
  136.  
  137. /** default channel configurations */
  138. static const uint8_t aac_chan_configs[6][5] = {
  139.  {1, TYPE_SCE},                               // 1 channel  - single channel element
  140.  {1, TYPE_CPE},                               // 2 channels - channel pair
  141.  {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
  142.  {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
  143.  {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
  144.  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
  145. };
  146.  
  147. /**
  148.  * Table to remap channels from libavcodec's default order to AAC order.
  149.  */
  150. static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
  151.     { 0 },
  152.     { 0, 1 },
  153.     { 2, 0, 1 },
  154.     { 2, 0, 1, 3 },
  155.     { 2, 0, 1, 3, 4 },
  156.     { 2, 0, 1, 4, 5, 3 },
  157. };
  158.  
  159. /**
  160.  * Make AAC audio config object.
  161.  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
  162.  */
  163. static void put_audio_specific_config(AVCodecContext *avctx)
  164. {
  165.     PutBitContext pb;
  166.     AACEncContext *s = avctx->priv_data;
  167.  
  168.     init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
  169.     put_bits(&pb, 5, 2); //object type - AAC-LC
  170.     put_bits(&pb, 4, s->samplerate_index); //sample rate index
  171.     put_bits(&pb, 4, s->channels);
  172.     //GASpecificConfig
  173.     put_bits(&pb, 1, 0); //frame length - 1024 samples
  174.     put_bits(&pb, 1, 0); //does not depend on core coder
  175.     put_bits(&pb, 1, 0); //is not extension
  176.  
  177.     //Explicitly Mark SBR absent
  178.     put_bits(&pb, 11, 0x2b7); //sync extension
  179.     put_bits(&pb, 5,  AOT_SBR);
  180.     put_bits(&pb, 1,  0);
  181.     flush_put_bits(&pb);
  182. }
  183.  
  184. #define WINDOW_FUNC(type) \
  185. static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
  186.                                     SingleChannelElement *sce, \
  187.                                     const float *audio)
  188.  
  189. WINDOW_FUNC(only_long)
  190. {
  191.     const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
  192.     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
  193.     float *out = sce->ret_buf;
  194.  
  195.     fdsp->vector_fmul        (out,        audio,        lwindow, 1024);
  196.     fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
  197. }
  198.  
  199. WINDOW_FUNC(long_start)
  200. {
  201.     const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
  202.     const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
  203.     float *out = sce->ret_buf;
  204.  
  205.     fdsp->vector_fmul(out, audio, lwindow, 1024);
  206.     memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
  207.     fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
  208.     memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
  209. }
  210.  
  211. WINDOW_FUNC(long_stop)
  212. {
  213.     const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
  214.     const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
  215.     float *out = sce->ret_buf;
  216.  
  217.     memset(out, 0, sizeof(out[0]) * 448);
  218.     fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
  219.     memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
  220.     fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
  221. }
  222.  
  223. WINDOW_FUNC(eight_short)
  224. {
  225.     const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
  226.     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
  227.     const float *in = audio + 448;
  228.     float *out = sce->ret_buf;
  229.     int w;
  230.  
  231.     for (w = 0; w < 8; w++) {
  232.         fdsp->vector_fmul        (out, in, w ? pwindow : swindow, 128);
  233.         out += 128;
  234.         in  += 128;
  235.         fdsp->vector_fmul_reverse(out, in, swindow, 128);
  236.         out += 128;
  237.     }
  238. }
  239.  
  240. static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
  241.                                      SingleChannelElement *sce,
  242.                                      const float *audio) = {
  243.     [ONLY_LONG_SEQUENCE]   = apply_only_long_window,
  244.     [LONG_START_SEQUENCE]  = apply_long_start_window,
  245.     [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
  246.     [LONG_STOP_SEQUENCE]   = apply_long_stop_window
  247. };
  248.  
  249. static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
  250.                                   float *audio)
  251. {
  252.     int i;
  253.     float *output = sce->ret_buf;
  254.  
  255.     apply_window[sce->ics.window_sequence[0]](&s->fdsp, sce, audio);
  256.  
  257.     if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
  258.         s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
  259.     else
  260.         for (i = 0; i < 1024; i += 128)
  261.             s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
  262.     memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
  263. }
  264.  
  265. /**
  266.  * Encode ics_info element.
  267.  * @see Table 4.6 (syntax of ics_info)
  268.  */
  269. static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
  270. {
  271.     int w;
  272.  
  273.     put_bits(&s->pb, 1, 0);                // ics_reserved bit
  274.     put_bits(&s->pb, 2, info->window_sequence[0]);
  275.     put_bits(&s->pb, 1, info->use_kb_window[0]);
  276.     if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
  277.         put_bits(&s->pb, 6, info->max_sfb);
  278.         put_bits(&s->pb, 1, 0);            // no prediction
  279.     } else {
  280.         put_bits(&s->pb, 4, info->max_sfb);
  281.         for (w = 1; w < 8; w++)
  282.             put_bits(&s->pb, 1, !info->group_len[w]);
  283.     }
  284. }
  285.  
  286. /**
  287.  * Encode MS data.
  288.  * @see 4.6.8.1 "Joint Coding - M/S Stereo"
  289.  */
  290. static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
  291. {
  292.     int i, w;
  293.  
  294.     put_bits(pb, 2, cpe->ms_mode);
  295.     if (cpe->ms_mode == 1)
  296.         for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
  297.             for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
  298.                 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
  299. }
  300.  
  301. /**
  302.  * Produce integer coefficients from scalefactors provided by the model.
  303.  */
  304. static void adjust_frame_information(ChannelElement *cpe, int chans)
  305. {
  306.     int i, w, w2, g, ch;
  307.     int start, maxsfb, cmaxsfb;
  308.  
  309.     for (ch = 0; ch < chans; ch++) {
  310.         IndividualChannelStream *ics = &cpe->ch[ch].ics;
  311.         start = 0;
  312.         maxsfb = 0;
  313.         cpe->ch[ch].pulse.num_pulse = 0;
  314.         for (w = 0; w < ics->num_windows*16; w += 16) {
  315.             for (g = 0; g < ics->num_swb; g++) {
  316.                 //apply M/S
  317.                 if (cpe->common_window && !ch && cpe->ms_mask[w + g]) {
  318.                     for (i = 0; i < ics->swb_sizes[g]; i++) {
  319.                         cpe->ch[0].coeffs[start+i] = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) / 2.0;
  320.                         cpe->ch[1].coeffs[start+i] =  cpe->ch[0].coeffs[start+i] - cpe->ch[1].coeffs[start+i];
  321.                     }
  322.                 }
  323.                 start += ics->swb_sizes[g];
  324.             }
  325.             for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w+cmaxsfb-1]; cmaxsfb--)
  326.                 ;
  327.             maxsfb = FFMAX(maxsfb, cmaxsfb);
  328.         }
  329.         ics->max_sfb = maxsfb;
  330.  
  331.         //adjust zero bands for window groups
  332.         for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
  333.             for (g = 0; g < ics->max_sfb; g++) {
  334.                 i = 1;
  335.                 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
  336.                     if (!cpe->ch[ch].zeroes[w2*16 + g]) {
  337.                         i = 0;
  338.                         break;
  339.                     }
  340.                 }
  341.                 cpe->ch[ch].zeroes[w*16 + g] = i;
  342.             }
  343.         }
  344.     }
  345.  
  346.     if (chans > 1 && cpe->common_window) {
  347.         IndividualChannelStream *ics0 = &cpe->ch[0].ics;
  348.         IndividualChannelStream *ics1 = &cpe->ch[1].ics;
  349.         int msc = 0;
  350.         ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
  351.         ics1->max_sfb = ics0->max_sfb;
  352.         for (w = 0; w < ics0->num_windows*16; w += 16)
  353.             for (i = 0; i < ics0->max_sfb; i++)
  354.                 if (cpe->ms_mask[w+i])
  355.                     msc++;
  356.         if (msc == 0 || ics0->max_sfb == 0)
  357.             cpe->ms_mode = 0;
  358.         else
  359.             cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
  360.     }
  361. }
  362.  
  363. /**
  364.  * Encode scalefactor band coding type.
  365.  */
  366. static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
  367. {
  368.     int w;
  369.  
  370.     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
  371.         s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
  372. }
  373.  
  374. /**
  375.  * Encode scalefactors.
  376.  */
  377. static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
  378.                                  SingleChannelElement *sce)
  379. {
  380.     int off = sce->sf_idx[0], diff;
  381.     int i, w;
  382.  
  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]) {
  386.                 diff = sce->sf_idx[w*16 + i] - off + SCALE_DIFF_ZERO;
  387.                 av_assert0(diff >= 0 && diff <= 120);
  388.                 off = sce->sf_idx[w*16 + i];
  389.                 put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
  390.             }
  391.         }
  392.     }
  393. }
  394.  
  395. /**
  396.  * Encode pulse data.
  397.  */
  398. static void encode_pulses(AACEncContext *s, Pulse *pulse)
  399. {
  400.     int i;
  401.  
  402.     put_bits(&s->pb, 1, !!pulse->num_pulse);
  403.     if (!pulse->num_pulse)
  404.         return;
  405.  
  406.     put_bits(&s->pb, 2, pulse->num_pulse - 1);
  407.     put_bits(&s->pb, 6, pulse->start);
  408.     for (i = 0; i < pulse->num_pulse; i++) {
  409.         put_bits(&s->pb, 5, pulse->pos[i]);
  410.         put_bits(&s->pb, 4, pulse->amp[i]);
  411.     }
  412. }
  413.  
  414. /**
  415.  * Encode spectral coefficients processed by psychoacoustic model.
  416.  */
  417. static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
  418. {
  419.     int start, i, w, w2;
  420.  
  421.     for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
  422.         start = 0;
  423.         for (i = 0; i < sce->ics.max_sfb; i++) {
  424.             if (sce->zeroes[w*16 + i]) {
  425.                 start += sce->ics.swb_sizes[i];
  426.                 continue;
  427.             }
  428.             for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++)
  429.                 s->coder->quantize_and_encode_band(s, &s->pb, sce->coeffs + start + w2*128,
  430.                                                    sce->ics.swb_sizes[i],
  431.                                                    sce->sf_idx[w*16 + i],
  432.                                                    sce->band_type[w*16 + i],
  433.                                                    s->lambda);
  434.             start += sce->ics.swb_sizes[i];
  435.         }
  436.     }
  437. }
  438.  
  439. /**
  440.  * Encode one channel of audio data.
  441.  */
  442. static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
  443.                                      SingleChannelElement *sce,
  444.                                      int common_window)
  445. {
  446.     put_bits(&s->pb, 8, sce->sf_idx[0]);
  447.     if (!common_window)
  448.         put_ics_info(s, &sce->ics);
  449.     encode_band_info(s, sce);
  450.     encode_scale_factors(avctx, s, sce);
  451.     encode_pulses(s, &sce->pulse);
  452.     put_bits(&s->pb, 1, 0); //tns
  453.     put_bits(&s->pb, 1, 0); //ssr
  454.     encode_spectral_coeffs(s, sce);
  455.     return 0;
  456. }
  457.  
  458. /**
  459.  * Write some auxiliary information about the created AAC file.
  460.  */
  461. static void put_bitstream_info(AACEncContext *s, const char *name)
  462. {
  463.     int i, namelen, padbits;
  464.  
  465.     namelen = strlen(name) + 2;
  466.     put_bits(&s->pb, 3, TYPE_FIL);
  467.     put_bits(&s->pb, 4, FFMIN(namelen, 15));
  468.     if (namelen >= 15)
  469.         put_bits(&s->pb, 8, namelen - 14);
  470.     put_bits(&s->pb, 4, 0); //extension type - filler
  471.     padbits = -put_bits_count(&s->pb) & 7;
  472.     avpriv_align_put_bits(&s->pb);
  473.     for (i = 0; i < namelen - 2; i++)
  474.         put_bits(&s->pb, 8, name[i]);
  475.     put_bits(&s->pb, 12 - padbits, 0);
  476. }
  477.  
  478. /*
  479.  * Copy input samples.
  480.  * Channels are reordered from libavcodec's default order to AAC order.
  481.  */
  482. static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
  483. {
  484.     int ch;
  485.     int end = 2048 + (frame ? frame->nb_samples : 0);
  486.     const uint8_t *channel_map = aac_chan_maps[s->channels - 1];
  487.  
  488.     /* copy and remap input samples */
  489.     for (ch = 0; ch < s->channels; ch++) {
  490.         /* copy last 1024 samples of previous frame to the start of the current frame */
  491.         memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
  492.  
  493.         /* copy new samples and zero any remaining samples */
  494.         if (frame) {
  495.             memcpy(&s->planar_samples[ch][2048],
  496.                    frame->extended_data[channel_map[ch]],
  497.                    frame->nb_samples * sizeof(s->planar_samples[0][0]));
  498.         }
  499.         memset(&s->planar_samples[ch][end], 0,
  500.                (3072 - end) * sizeof(s->planar_samples[0][0]));
  501.     }
  502. }
  503.  
  504. static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
  505.                             const AVFrame *frame, int *got_packet_ptr)
  506. {
  507.     AACEncContext *s = avctx->priv_data;
  508.     float **samples = s->planar_samples, *samples2, *la, *overlap;
  509.     ChannelElement *cpe;
  510.     int i, ch, w, g, chans, tag, start_ch, ret;
  511.     int chan_el_counter[4];
  512.     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
  513.  
  514.     if (s->last_frame == 2)
  515.         return 0;
  516.  
  517.     /* add current frame to queue */
  518.     if (frame) {
  519.         if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
  520.             return ret;
  521.     }
  522.  
  523.     copy_input_samples(s, frame);
  524.     if (s->psypp)
  525.         ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
  526.  
  527.     if (!avctx->frame_number)
  528.         return 0;
  529.  
  530.     start_ch = 0;
  531.     for (i = 0; i < s->chan_map[0]; i++) {
  532.         FFPsyWindowInfo* wi = windows + start_ch;
  533.         tag      = s->chan_map[i+1];
  534.         chans    = tag == TYPE_CPE ? 2 : 1;
  535.         cpe      = &s->cpe[i];
  536.         for (ch = 0; ch < chans; ch++) {
  537.             IndividualChannelStream *ics = &cpe->ch[ch].ics;
  538.             int cur_channel = start_ch + ch;
  539.             overlap  = &samples[cur_channel][0];
  540.             samples2 = overlap + 1024;
  541.             la       = samples2 + (448+64);
  542.             if (!frame)
  543.                 la = NULL;
  544.             if (tag == TYPE_LFE) {
  545.                 wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
  546.                 wi[ch].window_shape   = 0;
  547.                 wi[ch].num_windows    = 1;
  548.                 wi[ch].grouping[0]    = 1;
  549.  
  550.                 /* Only the lowest 12 coefficients are used in a LFE channel.
  551.                  * The expression below results in only the bottom 8 coefficients
  552.                  * being used for 11.025kHz to 16kHz sample rates.
  553.                  */
  554.                 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
  555.             } else {
  556.                 wi[ch] = s->psy.model->window(&s->psy, samples2, la, cur_channel,
  557.                                               ics->window_sequence[0]);
  558.             }
  559.             ics->window_sequence[1] = ics->window_sequence[0];
  560.             ics->window_sequence[0] = wi[ch].window_type[0];
  561.             ics->use_kb_window[1]   = ics->use_kb_window[0];
  562.             ics->use_kb_window[0]   = wi[ch].window_shape;
  563.             ics->num_windows        = wi[ch].num_windows;
  564.             ics->swb_sizes          = s->psy.bands    [ics->num_windows == 8];
  565.             ics->num_swb            = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
  566.             for (w = 0; w < ics->num_windows; w++)
  567.                 ics->group_len[w] = wi[ch].grouping[w];
  568.  
  569.             apply_window_and_mdct(s, &cpe->ch[ch], overlap);
  570.         }
  571.         start_ch += chans;
  572.     }
  573.     if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)
  574.         return ret;
  575.     do {
  576.         int frame_bits;
  577.  
  578.         init_put_bits(&s->pb, avpkt->data, avpkt->size);
  579.  
  580.         if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
  581.             put_bitstream_info(s, LIBAVCODEC_IDENT);
  582.         start_ch = 0;
  583.         memset(chan_el_counter, 0, sizeof(chan_el_counter));
  584.         for (i = 0; i < s->chan_map[0]; i++) {
  585.             FFPsyWindowInfo* wi = windows + start_ch;
  586.             const float *coeffs[2];
  587.             tag      = s->chan_map[i+1];
  588.             chans    = tag == TYPE_CPE ? 2 : 1;
  589.             cpe      = &s->cpe[i];
  590.             put_bits(&s->pb, 3, tag);
  591.             put_bits(&s->pb, 4, chan_el_counter[tag]++);
  592.             for (ch = 0; ch < chans; ch++)
  593.                 coeffs[ch] = cpe->ch[ch].coeffs;
  594.             s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
  595.             for (ch = 0; ch < chans; ch++) {
  596.                 s->cur_channel = start_ch + ch;
  597.                 s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
  598.             }
  599.             cpe->common_window = 0;
  600.             if (chans > 1
  601.                 && wi[0].window_type[0] == wi[1].window_type[0]
  602.                 && wi[0].window_shape   == wi[1].window_shape) {
  603.  
  604.                 cpe->common_window = 1;
  605.                 for (w = 0; w < wi[0].num_windows; w++) {
  606.                     if (wi[0].grouping[w] != wi[1].grouping[w]) {
  607.                         cpe->common_window = 0;
  608.                         break;
  609.                     }
  610.                 }
  611.             }
  612.             s->cur_channel = start_ch;
  613.             if (s->options.stereo_mode && cpe->common_window) {
  614.                 if (s->options.stereo_mode > 0) {
  615.                     IndividualChannelStream *ics = &cpe->ch[0].ics;
  616.                     for (w = 0; w < ics->num_windows; w += ics->group_len[w])
  617.                         for (g = 0;  g < ics->num_swb; g++)
  618.                             cpe->ms_mask[w*16+g] = 1;
  619.                 } else if (s->coder->search_for_ms) {
  620.                     s->coder->search_for_ms(s, cpe, s->lambda);
  621.                 }
  622.             }
  623.             adjust_frame_information(cpe, chans);
  624.             if (chans == 2) {
  625.                 put_bits(&s->pb, 1, cpe->common_window);
  626.                 if (cpe->common_window) {
  627.                     put_ics_info(s, &cpe->ch[0].ics);
  628.                     encode_ms_info(&s->pb, cpe);
  629.                 }
  630.             }
  631.             for (ch = 0; ch < chans; ch++) {
  632.                 s->cur_channel = start_ch + ch;
  633.                 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
  634.             }
  635.             start_ch += chans;
  636.         }
  637.  
  638.         frame_bits = put_bits_count(&s->pb);
  639.         if (frame_bits <= 6144 * s->channels - 3) {
  640.             s->psy.bitres.bits = frame_bits / s->channels;
  641.             break;
  642.         }
  643.  
  644.         s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
  645.  
  646.     } while (1);
  647.  
  648.     put_bits(&s->pb, 3, TYPE_END);
  649.     flush_put_bits(&s->pb);
  650.     avctx->frame_bits = put_bits_count(&s->pb);
  651.  
  652.     // rate control stuff
  653.     if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
  654.         float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
  655.         s->lambda *= ratio;
  656.         s->lambda = FFMIN(s->lambda, 65536.f);
  657.     }
  658.  
  659.     if (!frame)
  660.         s->last_frame++;
  661.  
  662.     ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
  663.                        &avpkt->duration);
  664.  
  665.     avpkt->size = put_bits_count(&s->pb) >> 3;
  666.     *got_packet_ptr = 1;
  667.     return 0;
  668. }
  669.  
  670. static av_cold int aac_encode_end(AVCodecContext *avctx)
  671. {
  672.     AACEncContext *s = avctx->priv_data;
  673.  
  674.     ff_mdct_end(&s->mdct1024);
  675.     ff_mdct_end(&s->mdct128);
  676.     ff_psy_end(&s->psy);
  677.     if (s->psypp)
  678.         ff_psy_preprocess_end(s->psypp);
  679.     av_freep(&s->buffer.samples);
  680.     av_freep(&s->cpe);
  681.     ff_af_queue_close(&s->afq);
  682.     return 0;
  683. }
  684.  
  685. static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
  686. {
  687.     int ret = 0;
  688.  
  689.     avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
  690.  
  691.     // window init
  692.     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
  693.     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
  694.     ff_init_ff_sine_windows(10);
  695.     ff_init_ff_sine_windows(7);
  696.  
  697.     if (ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0))
  698.         return ret;
  699.     if (ret = ff_mdct_init(&s->mdct128,   8, 0, 32768.0))
  700.         return ret;
  701.  
  702.     return 0;
  703. }
  704.  
  705. static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
  706. {
  707.     int ch;
  708.     FF_ALLOCZ_OR_GOTO(avctx, s->buffer.samples, 3 * 1024 * s->channels * sizeof(s->buffer.samples[0]), alloc_fail);
  709.     FF_ALLOCZ_OR_GOTO(avctx, s->cpe, sizeof(ChannelElement) * s->chan_map[0], alloc_fail);
  710.     FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
  711.  
  712.     for(ch = 0; ch < s->channels; ch++)
  713.         s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
  714.  
  715.     return 0;
  716. alloc_fail:
  717.     return AVERROR(ENOMEM);
  718. }
  719.  
  720. static av_cold int aac_encode_init(AVCodecContext *avctx)
  721. {
  722.     AACEncContext *s = avctx->priv_data;
  723.     int i, ret = 0;
  724.     const uint8_t *sizes[2];
  725.     uint8_t grouping[AAC_MAX_CHANNELS];
  726.     int lengths[2];
  727.  
  728.     avctx->frame_size = 1024;
  729.  
  730.     for (i = 0; i < 16; i++)
  731.         if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
  732.             break;
  733.  
  734.     s->channels = avctx->channels;
  735.  
  736.     ERROR_IF(i == 16,
  737.              "Unsupported sample rate %d\n", avctx->sample_rate);
  738.     ERROR_IF(s->channels > AAC_MAX_CHANNELS,
  739.              "Unsupported number of channels: %d\n", s->channels);
  740.     ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
  741.              "Unsupported profile %d\n", avctx->profile);
  742.     ERROR_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
  743.              "Too many bits per frame requested\n");
  744.  
  745.     s->samplerate_index = i;
  746.  
  747.     s->chan_map = aac_chan_configs[s->channels-1];
  748.  
  749.     if (ret = dsp_init(avctx, s))
  750.         goto fail;
  751.  
  752.     if (ret = alloc_buffers(avctx, s))
  753.         goto fail;
  754.  
  755.     avctx->extradata_size = 5;
  756.     put_audio_specific_config(avctx);
  757.  
  758.     sizes[0]   = swb_size_1024[i];
  759.     sizes[1]   = swb_size_128[i];
  760.     lengths[0] = ff_aac_num_swb_1024[i];
  761.     lengths[1] = ff_aac_num_swb_128[i];
  762.     for (i = 0; i < s->chan_map[0]; i++)
  763.         grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
  764.     if (ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths, s->chan_map[0], grouping))
  765.         goto fail;
  766.     s->psypp = ff_psy_preprocess_init(avctx);
  767.     s->coder = &ff_aac_coders[s->options.aac_coder];
  768.  
  769.     if (HAVE_MIPSDSPR1)
  770.         ff_aac_coder_init_mips(s);
  771.  
  772.     s->lambda = avctx->global_quality ? avctx->global_quality : 120;
  773.  
  774.     ff_aac_tableinit();
  775.  
  776.     for (i = 0; i < 428; i++)
  777.         ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
  778.  
  779.     avctx->delay = 1024;
  780.     ff_af_queue_init(avctx, &s->afq);
  781.  
  782.     return 0;
  783. fail:
  784.     aac_encode_end(avctx);
  785.     return ret;
  786. }
  787.  
  788. #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
  789. static const AVOption aacenc_options[] = {
  790.     {"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
  791.         {"auto",     "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
  792.         {"ms_off",   "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 =  0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
  793.         {"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"},
  794.     {"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
  795.         {"faac",     "FAAC-inspired method",      0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
  796.         {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
  797.         {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
  798.         {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
  799.     {NULL}
  800. };
  801.  
  802. static const AVClass aacenc_class = {
  803.     "AAC encoder",
  804.     av_default_item_name,
  805.     aacenc_options,
  806.     LIBAVUTIL_VERSION_INT,
  807. };
  808.  
  809. /* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
  810.  * failures */
  811. static const int mpeg4audio_sample_rates[16] = {
  812.     96000, 88200, 64000, 48000, 44100, 32000,
  813.     24000, 22050, 16000, 12000, 11025, 8000, 7350
  814. };
  815.  
  816. AVCodec ff_aac_encoder = {
  817.     .name           = "aac",
  818.     .long_name      = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
  819.     .type           = AVMEDIA_TYPE_AUDIO,
  820.     .id             = AV_CODEC_ID_AAC,
  821.     .priv_data_size = sizeof(AACEncContext),
  822.     .init           = aac_encode_init,
  823.     .encode2        = aac_encode_frame,
  824.     .close          = aac_encode_end,
  825.     .supported_samplerates = mpeg4audio_sample_rates,
  826.     .capabilities   = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
  827.                       CODEC_CAP_EXPERIMENTAL,
  828.     .sample_fmts    = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
  829.                                                      AV_SAMPLE_FMT_NONE },
  830.     .priv_class     = &aacenc_class,
  831. };
  832.