/*
  * 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

  * AAC encoder
  */
 
 /***********************************
  *              TODOs:

  * add sane pulse detection

  ***********************************/
 

 #include "libavutil/libm.h"

 #include "libavutil/thread.h"

 #include "libavutil/float_dsp.h"

 #include "libavutil/opt.h"

 #include "avcodec.h"

 #include "put_bits.h"

 #include "internal.h"

 #include "mpeg4audio.h"

 #include "kbdwin.h"

 #include "sinewin.h"

 
 #include "aac.h"
 #include "aactab.h"

 #include "aacenc.h"

 #include "aacenctab.h"

 #include "aacenc_utils.h"

 
 #include "psymodel.h"

 static AVOnce aac_table_init = AV_ONCE_INIT;
 

 /**

  * 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;

     int channels = s->channels - (s->channels == 8 ? 1 : 0);

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

     put_bits(&pb, 5, s->profile+1); //profile

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

     put_bits(&pb, 4, 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

 
     //Explicitly Mark SBR absent

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

     put_bits(&pb, 5,  AOT_SBR);
     put_bits(&pb, 1,  0);

     flush_put_bits(&pb);
 }
 

 void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
 {

     ++s->quantize_band_cost_cache_generation;
     if (s->quantize_band_cost_cache_generation == 0) {
         memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
         s->quantize_band_cost_cache_generation = 1;

     }
 }
 

 #define WINDOW_FUNC(type) \

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

                                     SingleChannelElement *sce, \
                                     const float *audio)

 WINDOW_FUNC(only_long)
 {
     const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;

     float *out = sce->ret_buf;

     fdsp->vector_fmul        (out,        audio,        lwindow, 1024);
     fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);

 }

 WINDOW_FUNC(long_start)
 {
     const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;

     float *out = sce->ret_buf;

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

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

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

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

 WINDOW_FUNC(long_stop)
 {
     const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
     const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;

     float *out = sce->ret_buf;

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

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

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

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

 }

 WINDOW_FUNC(eight_short)
 {
     const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
     const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
     const float *in = audio + 448;

     float *out = sce->ret_buf;

     int w;

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

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

         out += 128;
         in  += 128;

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

         out += 128;
     }

 }
 

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

                                      SingleChannelElement *sce,
                                      const float *audio) = {

     [ONLY_LONG_SEQUENCE]   = apply_only_long_window,
     [LONG_START_SEQUENCE]  = apply_long_start_window,
     [EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
     [LONG_STOP_SEQUENCE]   = apply_long_stop_window
 };
 

 static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
                                   float *audio)

 {

     int i;

     const float *output = sce->ret_buf;

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

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

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

     else
         for (i = 0; i < 1024; i += 128)

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

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

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

 }
 

 /**
  * Encode ics_info element.
  * @see Table 4.6 (syntax of ics_info)
  */

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

 {

     int w;

 
     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, !!info->predictor_present);

     } else {

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

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

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

     }
 }
 
 /**

  * Encode MS data.
  * @see 4.6.8.1 "Joint Coding - M/S Stereo"

  */

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

 {
     int i, w;

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

     if (cpe->ms_mode == 1)
         for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])

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

                 put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
 }
 
 /**
  * Produce integer coefficients from scalefactors provided by the model.
  */

 static void adjust_frame_information(ChannelElement *cpe, int chans)

 {
     int i, w, w2, g, ch;

     int maxsfb, cmaxsfb;
 
     for (ch = 0; ch < chans; ch++) {
         IndividualChannelStream *ics = &cpe->ch[ch].ics;
         maxsfb = 0;
         cpe->ch[ch].pulse.num_pulse = 0;
         for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
             for (w2 =  0; w2 < ics->group_len[w]; w2++) {

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

             }
         }
         ics->max_sfb = maxsfb;
 
         //adjust zero bands for window groups

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

                 i = 1;

                 for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
                     if (!cpe->ch[ch].zeroes[w2*16 + g]) {

                         i = 0;
                         break;
                     }
                 }
                 cpe->ch[ch].zeroes[w*16 + g] = i;
             }
         }
     }
 

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

         IndividualChannelStream *ics0 = &cpe->ch[0].ics;
         IndividualChannelStream *ics1 = &cpe->ch[1].ics;
         int msc = 0;
         ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
         ics1->max_sfb = ics0->max_sfb;

         for (w = 0; w < ics0->num_windows*16; w += 16)
             for (i = 0; i < ics0->max_sfb; i++)

                 if (cpe->ms_mask[w+i])
                     msc++;

         if (msc == 0 || ics0->max_sfb == 0)
             cpe->ms_mode = 0;
         else

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

     }
 }
 

 static void apply_intensity_stereo(ChannelElement *cpe)
 {
     int w, w2, g, i;
     IndividualChannelStream *ics = &cpe->ch[0].ics;
     if (!cpe->common_window)
         return;
     for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
         for (w2 =  0; w2 < ics->group_len[w]; w2++) {
             int start = (w+w2) * 128;
             for (g = 0; g < ics->num_swb; g++) {
                 int p  = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
                 float scale = cpe->ch[0].is_ener[w*16+g];
                 if (!cpe->is_mask[w*16 + g]) {
                     start += ics->swb_sizes[g];
                     continue;
                 }

                 if (cpe->ms_mask[w*16 + g])
                     p *= -1;

                 for (i = 0; i < ics->swb_sizes[g]; i++) {
                     float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
                     cpe->ch[0].coeffs[start+i] = sum;
                     cpe->ch[1].coeffs[start+i] = 0.0f;
                 }
                 start += ics->swb_sizes[g];
             }
         }
     }
 }
 
 static void apply_mid_side_stereo(ChannelElement *cpe)
 {
     int w, w2, g, i;
     IndividualChannelStream *ics = &cpe->ch[0].ics;
     if (!cpe->common_window)
         return;
     for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
         for (w2 =  0; w2 < ics->group_len[w]; w2++) {
             int start = (w+w2) * 128;
             for (g = 0; g < ics->num_swb; g++) {

                 /* ms_mask can be used for other purposes in PNS and I/S,
                  * so must not apply M/S if any band uses either, even if
                  * ms_mask is set.
                  */
                 if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]

                     || cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
                     || cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {

                     start += ics->swb_sizes[g];
                     continue;
                 }
                 for (i = 0; i < ics->swb_sizes[g]; i++) {
                     float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
                     float R = L - cpe->ch[1].coeffs[start+i];
                     cpe->ch[0].coeffs[start+i] = L;
                     cpe->ch[1].coeffs[start+i] = R;
                 }
                 start += ics->swb_sizes[g];
             }
         }
     }
 }
 

 /**
  * Encode scalefactor band coding type.
  */
 static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
 {
     int w;
 

     if (s->coder->set_special_band_scalefactors)
         s->coder->set_special_band_scalefactors(s, sce);
 

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

         s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
 }
 
 /**
  * Encode scalefactors.
  */

 static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
                                  SingleChannelElement *sce)

 {

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

     int off_is = 0, noise_flag = 1;

     int i, w;
 

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

                 if (sce->band_type[w*16 + i] == NOISE_BT) {
                     diff = sce->sf_idx[w*16 + i] - off_pns;
                     off_pns = sce->sf_idx[w*16 + i];
                     if (noise_flag-- > 0) {
                         put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
                         continue;
                     }

                 } else if (sce->band_type[w*16 + i] == INTENSITY_BT  ||
                            sce->band_type[w*16 + i] == INTENSITY_BT2) {
                     diff = sce->sf_idx[w*16 + i] - off_is;
                     off_is = sce->sf_idx[w*16 + i];

                 } else {
                     diff = sce->sf_idx[w*16 + i] - off_sf;
                     off_sf = sce->sf_idx[w*16 + i];
                 }
                 diff += SCALE_DIFF_ZERO;

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

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

         }
     }
 }
 
 /**

  * 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;

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

         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[w]; w2++) {

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

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

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

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

             }

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

         }
     }
 }
 
 /**

  * Downscale spectral coefficients for near-clipping windows to avoid artifacts
  */
 static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
 {
     int start, i, j, w;
 
     if (sce->ics.clip_avoidance_factor < 1.0f) {
         for (w = 0; w < sce->ics.num_windows; w++) {
             start = 0;
             for (i = 0; i < sce->ics.max_sfb; i++) {

                 float *swb_coeffs = &sce->coeffs[start + w*128];

                 for (j = 0; j < sce->ics.swb_sizes[i]; j++)
                     swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
                 start += sce->ics.swb_sizes[i];
             }
         }
     }
 }
 
 /**

  * Encode one channel of audio data.
  */

 static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
                                      SingleChannelElement *sce,
                                      int common_window)

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

     if (!common_window) {

         put_ics_info(s, &sce->ics);

         if (s->coder->encode_main_pred)
             s->coder->encode_main_pred(s, sce);

         if (s->coder->encode_ltp_info)
             s->coder->encode_ltp_info(s, sce, 0);

     }

     encode_band_info(s, sce);
     encode_scale_factors(avctx, s, sce);
     encode_pulses(s, &sce->pulse);

     put_bits(&s->pb, 1, !!sce->tns.present);

     if (s->coder->encode_tns_info)
         s->coder->encode_tns_info(s, sce);

     put_bits(&s->pb, 1, 0); //ssr
     encode_spectral_coeffs(s, sce);
     return 0;
 }
 
 /**

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

 static void put_bitstream_info(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 - 14);

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

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

     avpriv_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);
 }
 

 /*

  * Copy input samples.

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

  */

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

 {

     int ch;
     int end = 2048 + (frame ? frame->nb_samples : 0);
     const uint8_t *channel_map = aac_chan_maps[s->channels - 1];

     /* copy and remap input samples */
     for (ch = 0; ch < s->channels; ch++) {

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

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

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

         if (frame) {

             memcpy(&s->planar_samples[ch][2048],
                    frame->extended_data[channel_map[ch]],
                    frame->nb_samples * sizeof(s->planar_samples[0][0]));

         }

         memset(&s->planar_samples[ch][end], 0,
                (3072 - end) * sizeof(s->planar_samples[0][0]));

     }
 }
 

 static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
                             const AVFrame *frame, int *got_packet_ptr)

 {
     AACEncContext *s = avctx->priv_data;

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

     ChannelElement *cpe;

     SingleChannelElement *sce;

     IndividualChannelStream *ics;

     int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;

     int target_bits, rate_bits, too_many_bits, too_few_bits;

     int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;

     int chan_el_counter[4];

     FFPsyWindowInfo windows[AAC_MAX_CHANNELS];

     /* add current frame to queue */
     if (frame) {

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

             return ret;

     } else {
         if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
             return 0;

     }
 

     copy_input_samples(s, frame);

     if (s->psypp)
         ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);

 
     if (!avctx->frame_number)

         return 0;
 
     start_ch = 0;

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

         FFPsyWindowInfo* wi = windows + start_ch;

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

         chans    = tag == TYPE_CPE ? 2 : 1;
         cpe      = &s->cpe[i];

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

             int k;

             float clip_avoidance_factor;

             sce = &cpe->ch[ch];
             ics = &sce->ics;
             s->cur_channel = start_ch + ch;
             overlap  = &samples[s->cur_channel][0];

             samples2 = overlap + 1024;

             la       = samples2 + (448+64);

             if (!frame)

                 la = NULL;

             if (tag == TYPE_LFE) {

                 wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;

                 wi[ch].window_shape   = 0;
                 wi[ch].num_windows    = 1;
                 wi[ch].grouping[0]    = 1;

                 wi[ch].clipping[0]    = 0;

 
                 /* Only the lowest 12 coefficients are used in a LFE channel.
                  * The expression below results in only the bottom 8 coefficients
                  * being used for 11.025kHz to 16kHz sample rates.
                  */
                 ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;

             } else {

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

                                               ics->window_sequence[0]);

             }

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

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

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

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

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

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

             ics->max_sfb            = FFMIN(ics->max_sfb, ics->num_swb);

             ics->swb_offset         = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
                                         ff_swb_offset_128 [s->samplerate_index]:
                                         ff_swb_offset_1024[s->samplerate_index];

             ics->tns_max_bands      = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
                                         ff_tns_max_bands_128 [s->samplerate_index]:
                                         ff_tns_max_bands_1024[s->samplerate_index];

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

 
             /* Calculate input sample maximums and evaluate clipping risk */
             clip_avoidance_factor = 0.0f;
             for (w = 0; w < ics->num_windows; w++) {
                 const float *wbuf = overlap + w * 128;
                 const int wlen = 2048 / ics->num_windows;
                 float max = 0;
                 int j;
                 /* mdct input is 2 * output */
                 for (j = 0; j < wlen; j++)
                     max = FFMAX(max, fabsf(wbuf[j]));
                 wi[ch].clipping[w] = max;
             }

             for (w = 0; w < ics->num_windows; w++) {
                 if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
                     ics->window_clipping[w] = 1;
                     clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
                 } else {
                     ics->window_clipping[w] = 0;
                 }
             }
             if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
                 ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
             } else {
                 ics->clip_avoidance_factor = 1.0f;
             }

             apply_window_and_mdct(s, sce, overlap);

 
             if (s->options.ltp && s->coder->update_ltp) {
                 s->coder->update_ltp(s, sce);
                 apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]);
                 s->mdct1024.mdct_calc(&s->mdct1024, sce->lcoeffs, sce->ret_buf);
             }
 

             for (k = 0; k < 1024; k++) {

                 if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
                     av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");

                     return AVERROR(EINVAL);
                 }

             }

             avoid_clipping(s, sce);

         }
         start_ch += chans;
     }

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

         return ret;

     frame_bits = its = 0;

     do {

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

         if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))

             put_bitstream_info(s, LIBAVCODEC_IDENT);

         start_ch = 0;

         target_bits = 0;

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

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

             FFPsyWindowInfo* wi = windows + start_ch;

             const float *coeffs[2];

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

             chans    = tag == TYPE_CPE ? 2 : 1;
             cpe      = &s->cpe[i];

             cpe->common_window = 0;

             memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
             memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));

             put_bits(&s->pb, 3, tag);
             put_bits(&s->pb, 4, chan_el_counter[tag]++);

             for (ch = 0; ch < chans; ch++) {
                 sce = &cpe->ch[ch];
                 coeffs[ch] = sce->coeffs;

                 sce->ics.predictor_present = 0;

                 sce->ics.ltp.present = 0;
                 memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used));

                 memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used));

                 memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));

                 for (w = 0; w < 128; w++)
                     if (sce->band_type[w] > RESERVED_BT)
                         sce->band_type[w] = 0;
             }

             s->psy.bitres.alloc = -1;

             s->psy.bitres.bits = s->last_frame_pb_count / s->channels;

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

             if (s->psy.bitres.alloc > 0) {
                 /* Lambda unused here on purpose, we need to take psy's unscaled allocation */

                 target_bits += s->psy.bitres.alloc
                     * (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));

                 s->psy.bitres.alloc /= chans;
             }
             s->cur_type = tag;

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

                 s->cur_channel = start_ch + ch;

                 if (s->options.pns && s->coder->mark_pns)
                     s->coder->mark_pns(s, avctx, &cpe->ch[ch]);

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

             }
             if (chans > 1
                 && wi[0].window_type[0] == wi[1].window_type[0]
                 && wi[0].window_shape   == wi[1].window_shape) {
 
                 cpe->common_window = 1;

                 for (w = 0; w < wi[0].num_windows; w++) {
                     if (wi[0].grouping[w] != wi[1].grouping[w]) {

                         cpe->common_window = 0;
                         break;
                     }

                 }
             }

             for (ch = 0; ch < chans; ch++) { /* TNS and PNS */

                 sce = &cpe->ch[ch];
                 s->cur_channel = start_ch + ch;
                 if (s->options.tns && s->coder->search_for_tns)
                     s->coder->search_for_tns(s, sce);

                 if (s->options.tns && s->coder->apply_tns_filt)

                     s->coder->apply_tns_filt(s, sce);

                 if (sce->tns.present)
                     tns_mode = 1;

                 if (s->options.pns && s->coder->search_for_pns)
                     s->coder->search_for_pns(s, avctx, sce);

             }

             s->cur_channel = start_ch;

             if (s->options.intensity_stereo) { /* Intensity Stereo */
                 if (s->coder->search_for_is)
                     s->coder->search_for_is(s, avctx, cpe);

                 if (cpe->is_mode) is_mode = 1;

                 apply_intensity_stereo(cpe);

             }

             if (s->options.pred) { /* Prediction */
                 for (ch = 0; ch < chans; ch++) {
                     sce = &cpe->ch[ch];
                     s->cur_channel = start_ch + ch;
                     if (s->options.pred && s->coder->search_for_pred)
                         s->coder->search_for_pred(s, sce);
                     if (cpe->ch[ch].ics.predictor_present) pred_mode = 1;
                 }

                 if (s->coder->adjust_common_pred)
                     s->coder->adjust_common_pred(s, cpe);

                 for (ch = 0; ch < chans; ch++) {
                     sce = &cpe->ch[ch];
                     s->cur_channel = start_ch + ch;
                     if (s->options.pred && s->coder->apply_main_pred)
                         s->coder->apply_main_pred(s, sce);
                 }
                 s->cur_channel = start_ch;

             }

             if (s->options.mid_side) { /* Mid/Side stereo */
                 if (s->options.mid_side == -1 && s->coder->search_for_ms)

                     s->coder->search_for_ms(s, cpe);
                 else if (cpe->common_window)
                     memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
                 apply_mid_side_stereo(cpe);

             }

             adjust_frame_information(cpe, chans);

             if (s->options.ltp) { /* LTP */
                 for (ch = 0; ch < chans; ch++) {
                     sce = &cpe->ch[ch];
                     s->cur_channel = start_ch + ch;
                     if (s->coder->search_for_ltp)
                         s->coder->search_for_ltp(s, sce, cpe->common_window);
                     if (sce->ics.ltp.present) pred_mode = 1;
                 }
                 s->cur_channel = start_ch;
                 if (s->coder->adjust_common_ltp)
                     s->coder->adjust_common_ltp(s, cpe);
             }

             if (chans == 2) {
                 put_bits(&s->pb, 1, cpe->common_window);
                 if (cpe->common_window) {
                     put_ics_info(s, &cpe->ch[0].ics);

                     if (s->coder->encode_main_pred)
                         s->coder->encode_main_pred(s, &cpe->ch[0]);

                     if (s->coder->encode_ltp_info)
                         s->coder->encode_ltp_info(s, &cpe->ch[0], 1);

                     encode_ms_info(&s->pb, cpe);

                     if (cpe->ms_mode) ms_mode = 1;

                 }

             }

             for (ch = 0; ch < chans; ch++) {
                 s->cur_channel = start_ch + ch;
                 encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);

             }
             start_ch += chans;

         }
 

         if (avctx->flags & AV_CODEC_FLAG_QSCALE) {

             /* When using a constant Q-scale, don't mess with lambda */

             break;

         }

         /* rate control stuff

          * allow between the nominal bitrate, and what psy's bit reservoir says to target
          * but drift towards the nominal bitrate always

          */
         frame_bits = put_bits_count(&s->pb);

         rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
         rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
         too_many_bits = FFMAX(target_bits, rate_bits);
         too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
         too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);

 
         /* When using ABR, be strict (but only for increasing) */

         too_few_bits = too_few_bits - too_few_bits/8;
         too_many_bits = too_many_bits + too_many_bits/2;

 
         if (   its == 0 /* for steady-state Q-scale tracking */
             || (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
             || frame_bits >= 6144 * s->channels - 3  )
         {

             float ratio = ((float)rate_bits) / frame_bits;

 
             if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
                 /*
                  * This path is for steady-state Q-scale tracking
                  * When frame bits fall within the stable range, we still need to adjust
                  * lambda to maintain it like so in a stable fashion (large jumps in lambda
                  * create artifacts and should be avoided), but slowly
                  */
                 ratio = sqrtf(sqrtf(ratio));
                 ratio = av_clipf(ratio, 0.9f, 1.1f);
             } else {
                 /* Not so fast though */
                 ratio = sqrtf(ratio);
             }
             s->lambda = FFMIN(s->lambda * ratio, 65536.f);

             /* Keep iterating if we must reduce and lambda is in the sky */

             if (ratio > 0.9f && ratio < 1.1f) {

                 break;
             } else {
                 if (is_mode || ms_mode || tns_mode || pred_mode) {
                     for (i = 0; i < s->chan_map[0]; i++) {
                         // Must restore coeffs
                         chans = tag == TYPE_CPE ? 2 : 1;
                         cpe = &s->cpe[i];
                         for (ch = 0; ch < chans; ch++)
                             memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
                     }
                 }
                 its++;
             }
         } else {
             break;
         }

     } while (1);
 

     if (s->options.ltp && s->coder->ltp_insert_new_frame)
         s->coder->ltp_insert_new_frame(s);
 

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

     s->last_frame_pb_count = put_bits_count(&s->pb);

     s->lambda_sum += s->lambda;
     s->lambda_count++;

     ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
                        &avpkt->duration);
 
     avpkt->size = put_bits_count(&s->pb) >> 3;
     *got_packet_ptr = 1;
     return 0;

 }
 

 static av_cold int aac_encode_end(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
 

     av_log(avctx, AV_LOG_INFO, "Qavg: %.3f\n", s->lambda_sum / s->lambda_count);
 

     ff_mdct_end(&s->mdct1024);
     ff_mdct_end(&s->mdct128);

     ff_psy_end(&s->psy);

     ff_lpc_end(&s->lpc);

     if (s->psypp)
         ff_psy_preprocess_end(s->psypp);

     av_freep(&s->buffer.samples);

     av_freep(&s->cpe);

     av_freep(&s->fdsp);

     ff_af_queue_close(&s->afq);

     return 0;
 }
 

 static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
 {
     int ret = 0;
 

     s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);

     if (!s->fdsp)
         return AVERROR(ENOMEM);

 
     // 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_init_ff_sine_windows(10);
     ff_init_ff_sine_windows(7);
 

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

         return ret;

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

         return ret;
 
     return 0;
 }
 
 static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
 {

     int ch;

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

     FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + AV_INPUT_BUFFER_PADDING_SIZE, alloc_fail);

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

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

     return 0;
 alloc_fail:
     return AVERROR(ENOMEM);
 }
 

 static av_cold void aac_encode_init_tables(void)
 {
     ff_aac_tableinit();
 }
 

 static av_cold int aac_encode_init(AVCodecContext *avctx)
 {
     AACEncContext *s = avctx->priv_data;
     int i, ret = 0;
     const uint8_t *sizes[2];
     uint8_t grouping[AAC_MAX_CHANNELS];
     int lengths[2];
 

     /* Constants */

     s->last_frame_pb_count = 0;

     avctx->extradata_size = 5;

     avctx->frame_size = 1024;

     avctx->initial_padding = 1024;

     s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
 
     /* Channel map and unspecified bitrate guessing */
     s->channels = avctx->channels;
     ERROR_IF(s->channels > AAC_MAX_CHANNELS || s->channels == 7,
              "Unsupported number of channels: %d\n", s->channels);
     s->chan_map = aac_chan_configs[s->channels-1];
     if (!avctx->bit_rate) {
         for (i = 1; i <= s->chan_map[0]; i++) {
             avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
                                s->chan_map[i] == TYPE_LFE ? 16000  : /* LFE  */
                                                             69000  ; /* SCE  */
         }
     }

     /* Samplerate */

     for (i = 0; i < 16; i++)
         if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
             break;

     s->samplerate_index = i;
     ERROR_IF(s->samplerate_index == 16 ||
              s->samplerate_index >= ff_aac_swb_size_1024_len ||
              s->samplerate_index >= ff_aac_swb_size_128_len,

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

 
     /* Bitrate limiting */

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

              "Too many bits %f > %d per frame requested, clamping to max\n",

              1024.0 * avctx->bit_rate / avctx->sample_rate,
              6144 * s->channels);

     avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
                                      avctx->bit_rate);

     /* Profile and option setting */
     avctx->profile = avctx->profile == FF_PROFILE_UNKNOWN ? FF_PROFILE_AAC_LOW :
                      avctx->profile;

     for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
         if (avctx->profile == aacenc_profiles[i])

             break;

     if (avctx->profile == FF_PROFILE_MPEG2_AAC_LOW) {
         avctx->profile = FF_PROFILE_AAC_LOW;
         ERROR_IF(s->options.pred,
                  "Main prediction unavailable in the \"mpeg2_aac_low\" profile\n");
         ERROR_IF(s->options.ltp,
                  "LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n");
         WARN_IF(s->options.pns,
                 "PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
         s->options.pns = 0;
     } else if (avctx->profile == FF_PROFILE_AAC_LTP) {
         s->options.ltp = 1;
         ERROR_IF(s->options.pred,
                  "Main prediction unavailable in the \"aac_ltp\" profile\n");
     } else if (avctx->profile == FF_PROFILE_AAC_MAIN) {
         s->options.pred = 1;
         ERROR_IF(s->options.ltp,
                  "LTP prediction unavailable in the \"aac_main\" profile\n");
     } else if (s->options.ltp) {
         avctx->profile = FF_PROFILE_AAC_LTP;
         WARN_IF(1,
                 "Chainging profile to \"aac_ltp\"\n");
         ERROR_IF(s->options.pred,
                  "Main prediction unavailable in the \"aac_ltp\" profile\n");
     } else if (s->options.pred) {
         avctx->profile = FF_PROFILE_AAC_MAIN;
         WARN_IF(1,
                 "Chainging profile to \"aac_main\"\n");

         ERROR_IF(s->options.ltp,

                  "LTP prediction unavailable in the \"aac_main\" profile\n");

     }

     s->profile = avctx->profile;

     /* Coder limitations */
     s->coder = &ff_aac_coders[s->options.coder];

     if (s->options.coder == AAC_CODER_ANMR) {

         ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,

                  "The ANMR coder is considered experimental, add -strict -2 to enable!\n");

         s->options.intensity_stereo = 0;
         s->options.pns = 0;
     }

     ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL,
              "The LPT profile requires experimental compliance, add -strict -2 to enable!\n");
 

     /* M/S introduces horrible artifacts with multichannel files, this is temporary */
     if (s->channels > 3)
         s->options.mid_side = 0;
 

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

         goto fail;
 

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

         goto fail;
 
     put_audio_specific_config(avctx);
 

     sizes[0]   = ff_aac_swb_size_1024[s->samplerate_index];
     sizes[1]   = ff_aac_swb_size_128[s->samplerate_index];
     lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
     lengths[1] = ff_aac_num_swb_128[s->samplerate_index];

     for (i = 0; i < s->chan_map[0]; i++)
         grouping[i] = s->chan_map[i + 1] == TYPE_CPE;

     if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
                            s->chan_map[0], grouping)) < 0)

         goto fail;
     s->psypp = ff_psy_preprocess_init(avctx);

     ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);

     s->random_state = 0x1f2e3d4c;

     s->abs_pow34   = abs_pow34_v;
     s->quant_bands = quantize_bands;
 
     if (ARCH_X86)
         ff_aac_dsp_init_x86(s);
 

     if (HAVE_MIPSDSP)

         ff_aac_coder_init_mips(s);
 

     if ((ret = ff_thread_once(&aac_table_init, &aac_encode_init_tables)) != 0)

         return AVERROR_UNKNOWN;

     ff_af_queue_init(avctx, &s->afq);
 

     return 0;
 fail:
     aac_encode_end(avctx);
     return ret;
 }
 

 #define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
 static const AVOption aacenc_options[] = {

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

         {"anmr",     "ANMR method",               0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR},    INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
         {"twoloop",  "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},
         {"fast",     "Constant quantizer",        0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST},    INT_MIN, INT_MAX, AACENC_FLAGS, "coder"},

     {"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},

     {"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
     {"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},

     {"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},

     {"aac_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
     {"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},

     {NULL}
 };
 
 static const AVClass aacenc_class = {
     "AAC encoder",
     av_default_item_name,
     aacenc_options,
     LIBAVUTIL_VERSION_INT,
 };
 

 static const AVCodecDefault aac_encode_defaults[] = {
     { "b", "0" },
     { NULL }
 };
 

 AVCodec ff_aac_encoder = {

     .name           = "aac",

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

     .type           = AVMEDIA_TYPE_AUDIO,

     .id             = AV_CODEC_ID_AAC,

     .priv_data_size = sizeof(AACEncContext),
     .init           = aac_encode_init,

     .encode2        = aac_encode_frame,

     .close          = aac_encode_end,

     .defaults       = aac_encode_defaults,

     .supported_samplerates = mpeg4audio_sample_rates,

     .caps_internal  = FF_CODEC_CAP_INIT_THREADSAFE,

     .capabilities   = AV_CODEC_CAP_SMALL_LAST_FRAME | AV_CODEC_CAP_DELAY,

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

                                                      AV_SAMPLE_FMT_NONE },
     .priv_class     = &aacenc_class,

 };