/*
  * AAC encoder
  * Copyright (C) 2008 Konstantin Shishkov
  *
  * This file is part of FFmpeg.
  *
  * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
  * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
  * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 /**

  * @file libavcodec/aacenc.c

  * AAC encoder
  */
 
 /***********************************
  *              TODOs:
  * psy model selection with some option

  * add sane pulse detection

  * add temporal noise shaping

  ***********************************/
 
 #include "avcodec.h"
 #include "bitstream.h"
 #include "dsputil.h"
 #include "mpeg4audio.h"
 
 #include "aacpsy.h"
 #include "aac.h"
 #include "aactab.h"
 
 static const uint8_t swb_size_1024_96[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
     64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
 };
 
 static const uint8_t swb_size_1024_64[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
     12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
     40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
 };
 
 static const uint8_t swb_size_1024_48[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
     32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
     96
 };
 
 static const uint8_t swb_size_1024_32[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
     32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
 };
 
 static const uint8_t swb_size_1024_24[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
     32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
 };
 
 static const uint8_t swb_size_1024_16[] = {
     8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
     12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
     32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
 };
 
 static const uint8_t swb_size_1024_8[] = {
     12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
     16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
     32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
 };
 

 static const uint8_t * const swb_size_1024[] = {

     swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
     swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
     swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
     swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
 };
 
 static const uint8_t swb_size_128_96[] = {
     4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
 };
 
 static const uint8_t swb_size_128_48[] = {
     4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
 };
 
 static const uint8_t swb_size_128_24[] = {
     4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
 };
 
 static const uint8_t swb_size_128_16[] = {
     4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
 };
 
 static const uint8_t swb_size_128_8[] = {
     4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
 };
 

 static const uint8_t * const swb_size_128[] = {

     /* the last entry on the following row is swb_size_128_64 but is a
        duplicate of swb_size_128_96 */
     swb_size_128_96, swb_size_128_96, swb_size_128_96,
     swb_size_128_48, swb_size_128_48, swb_size_128_48,
     swb_size_128_24, swb_size_128_24, swb_size_128_16,
     swb_size_128_16, swb_size_128_16, swb_size_128_8
 };
 

 /** bits needed to code codebook run value for long windows */
 static const uint8_t run_value_bits_long[64] = {
      5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,
      5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5,  5, 10,
     10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
     10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
 };
 
 /** bits needed to code codebook run value for short windows */
 static const uint8_t run_value_bits_short[16] = {
     3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
 };
 

 static const uint8_t* const run_value_bits[2] = {

     run_value_bits_long, run_value_bits_short
 };
 

 /** default channel configurations */
 static const uint8_t aac_chan_configs[6][5] = {

  {1, TYPE_SCE},                               // 1 channel  - single channel element
  {1, TYPE_CPE},                               // 2 channels - channel pair
  {2, TYPE_SCE, TYPE_CPE},                     // 3 channels - center + stereo
  {3, TYPE_SCE, TYPE_CPE, TYPE_SCE},           // 4 channels - front center + stereo + back center
  {3, TYPE_SCE, TYPE_CPE, TYPE_CPE},           // 5 channels - front center + stereo + back stereo
  {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE

 };
 
 /**

  * structure used in optimal codebook search
  */
 typedef struct BandCodingPath {
     int prev_idx; ///< pointer to the previous path point
     int codebook; ///< codebook for coding band run
     int bits;     ///< number of bit needed to code given number of bands
 } BandCodingPath;
 
 /**

  * AAC encoder context
  */
 typedef struct {
     PutBitContext pb;
     MDCTContext mdct1024;                        ///< long (1024 samples) frame transform context
     MDCTContext mdct128;                         ///< short (128 samples) frame transform context
     DSPContext  dsp;

     DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
     int16_t* samples;                            ///< saved preprocessed input
 
     int samplerate_index;                        ///< MPEG-4 samplerate index
 
     ChannelElement *cpe;                         ///< channel elements

     AACPsyContext psy;                           ///< psychoacoustic model context
     int last_frame;

 } AACEncContext;
 
 /**

  * Make AAC audio config object.
  * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
  */
 static void put_audio_specific_config(AVCodecContext *avctx)
 {
     PutBitContext pb;
     AACEncContext *s = avctx->priv_data;
 
     init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
     put_bits(&pb, 5, 2); //object type - AAC-LC
     put_bits(&pb, 4, s->samplerate_index); //sample rate index
     put_bits(&pb, 4, avctx->channels);
     //GASpecificConfig
     put_bits(&pb, 1, 0); //frame length - 1024 samples
     put_bits(&pb, 1, 0); //does not depend on core coder
     put_bits(&pb, 1, 0); //is not extension
     flush_put_bits(&pb);
 }
 
 static av_cold int aac_encode_init(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
     int i;
 
     avctx->frame_size = 1024;
 
     for(i = 0; i < 16; i++)
         if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
             break;
     if(i == 16){
         av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
         return -1;
     }
     if(avctx->channels > 6){
         av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
         return -1;
     }
     s->samplerate_index = i;
 
     dsputil_init(&s->dsp, avctx);
     ff_mdct_init(&s->mdct1024, 11, 0);
     ff_mdct_init(&s->mdct128,   8, 0);

     // window init
     ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
     ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
     ff_sine_window_init(ff_sine_1024, 1024);
     ff_sine_window_init(ff_sine_128, 128);

 
     s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
     s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);

     if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP,
                        aac_chan_configs[avctx->channels-1][0], 0,

                        swb_size_1024[i], ff_aac_num_swb_1024[i], swb_size_128[i], ff_aac_num_swb_128[i]) < 0){

         av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n");
         return -1;
     }
     avctx->extradata = av_malloc(2);
     avctx->extradata_size = 2;
     put_audio_specific_config(avctx);
     return 0;
 }
 
 /**
  * Encode ics_info element.
  * @see Table 4.6 (syntax of ics_info)
  */

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

 {
     int i;
 
     put_bits(&s->pb, 1, 0);                // ics_reserved bit
     put_bits(&s->pb, 2, info->window_sequence[0]);
     put_bits(&s->pb, 1, info->use_kb_window[0]);
     if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){
         put_bits(&s->pb, 6, info->max_sfb);
         put_bits(&s->pb, 1, 0);            // no prediction
     }else{
         put_bits(&s->pb, 4, info->max_sfb);
         for(i = 1; i < info->num_windows; i++)
             put_bits(&s->pb, 1, info->group_len[i]);
     }
 }
 
 /**

  * Calculate the number of bits needed to code all coefficient signs in current band.
  */
 static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
                                     int group_len, int start, int size)
 {
     int bits = 0;
     int i, w;
     for(w = 0; w < group_len; w++){
         for(i = 0; i < size; i++){
             if(sce->icoefs[start + i])
                 bits++;
         }
         start += 128;
     }
     return bits;
 }
 
 /**

  * Encode pulse data.
  */

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

 {
     int i;
 
     put_bits(&s->pb, 1, !!pulse->num_pulse);
     if(!pulse->num_pulse) return;
 
     put_bits(&s->pb, 2, pulse->num_pulse - 1);
     put_bits(&s->pb, 6, pulse->start);
     for(i = 0; i < pulse->num_pulse; i++){

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

         put_bits(&s->pb, 4, pulse->amp[i]);
     }
 }
 
 /**
  * Encode spectral coefficients processed by psychoacoustic model.
  */

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

 {
     int start, i, w, w2, wg;
 
     w = 0;

     for(wg = 0; wg < sce->ics.num_window_groups; wg++){

         start = 0;

         for(i = 0; i < sce->ics.max_sfb; i++){
             if(sce->zeroes[w*16 + i]){
                 start += sce->ics.swb_sizes[i];

                 continue;
             }

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

                 encode_band_coeffs(s, sce, start + w2*128,

                                    sce->ics.swb_sizes[i],
                                    sce->band_type[w*16 + i]);

             }

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

         }

         w += sce->ics.group_len[wg];

     }
 }
 
 /**

  * Write some auxiliary information about the created AAC file.
  */
 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
 {
     int i, namelen, padbits;
 
     namelen = strlen(name) + 2;

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

     put_bits(&s->pb, 4, FFMIN(namelen, 15));
     if(namelen >= 15)
         put_bits(&s->pb, 8, namelen - 16);
     put_bits(&s->pb, 4, 0); //extension type - filler
     padbits = 8 - (put_bits_count(&s->pb) & 7);
     align_put_bits(&s->pb);
     for(i = 0; i < namelen - 2; i++)
         put_bits(&s->pb, 8, name[i]);
     put_bits(&s->pb, 12 - padbits, 0);
 }
 
 static av_cold int aac_encode_end(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
 
     ff_mdct_end(&s->mdct1024);
     ff_mdct_end(&s->mdct128);
     ff_aac_psy_end(&s->psy);
     av_freep(&s->samples);
     av_freep(&s->cpe);
     return 0;
 }
 
 AVCodec aac_encoder = {
     "aac",
     CODEC_TYPE_AUDIO,
     CODEC_ID_AAC,
     sizeof(AACEncContext),
     aac_encode_init,
     aac_encode_frame,
     aac_encode_end,
     .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
     .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
     .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
 };