/*
  * audio encoder psychoacoustic model
  * 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
  */
 

 #include <string.h>
 

 #include "avcodec.h"
 #include "psymodel.h"
 #include "iirfilter.h"

 #include "libavutil/mem.h"

 
 extern const FFPsyModel ff_aac_psy_model;
 

 av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
                         const uint8_t **bands, const int* num_bands,
                         int num_groups, const uint8_t *group_map)

 {

     int i, j, k = 0;
 

     ctx->avctx = avctx;

     ctx->ch        = av_mallocz(sizeof(ctx->ch[0]) * avctx->channels * 2);
     ctx->group     = av_mallocz(sizeof(ctx->group[0]) * num_groups);

     ctx->bands     = av_malloc (sizeof(ctx->bands[0])     * num_lens);
     ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
     memcpy(ctx->bands,     bands,     sizeof(ctx->bands[0])     *  num_lens);
     memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) *  num_lens);

 
     /* assign channels to groups (with virtual channels for coupling) */
     for (i = 0; i < num_groups; i++) {
         /* NOTE: Add 1 to handle the AAC chan_config without modification.
          *       This has the side effect of allowing an array of 0s to map
          *       to one channel per group.
          */
         ctx->group[i].num_ch = group_map[i] + 1;
         for (j = 0; j < ctx->group[i].num_ch * 2; j++)
             ctx->group[i].ch[j]  = &ctx->ch[k++];
     }
 

     switch (ctx->avctx->codec_id) {

     case AV_CODEC_ID_AAC:

         ctx->model = &ff_aac_psy_model;
         break;
     }

     if (ctx->model->init)

         return ctx->model->init(ctx);
     return 0;
 }
 

 FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
 {
     int i = 0, ch = 0;
 
     while (ch <= channel)
         ch += ctx->group[i++].num_ch;
 
     return &ctx->group[i-1];
 }
 

 av_cold void ff_psy_end(FFPsyContext *ctx)
 {

     if (ctx->model->end)

         ctx->model->end(ctx);
     av_freep(&ctx->bands);
     av_freep(&ctx->num_bands);

     av_freep(&ctx->group);
     av_freep(&ctx->ch);

 }
 
 typedef struct FFPsyPreprocessContext{
     AVCodecContext *avctx;
     float stereo_att;
     struct FFIIRFilterCoeffs *fcoeffs;
     struct FFIIRFilterState **fstate;
 }FFPsyPreprocessContext;
 
 #define FILT_ORDER 4
 
 av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
 {
     FFPsyPreprocessContext *ctx;
     int i;

     float cutoff_coeff = 0;

     ctx        = av_mallocz(sizeof(FFPsyPreprocessContext));

     ctx->avctx = avctx;
 

     if (avctx->cutoff > 0)
         cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;

     if (!cutoff_coeff && avctx->codec_id == AV_CODEC_ID_AAC)

         cutoff_coeff = 2.0 * AAC_CUTOFF(avctx) / avctx->sample_rate;
 

     if (cutoff_coeff && cutoff_coeff < 0.98)

     ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
                                              FF_FILTER_MODE_LOWPASS, FILT_ORDER,
                                              cutoff_coeff, 0.0, 0.0);

     if (ctx->fcoeffs) {

         ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);

         for (i = 0; i < avctx->channels; i++)

             ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
     }
     return ctx;
 }
 

 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)

 {

     int ch;
     int frame_size = ctx->avctx->frame_size;
 

     if (ctx->fstate) {

         for (ch = 0; ch < channels; ch++)

             ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
                               &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);

     }
 }
 
 av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
 {
     int i;
     ff_iir_filter_free_coeffs(ctx->fcoeffs);
     if (ctx->fstate)
         for (i = 0; i < ctx->avctx->channels; i++)
             ff_iir_filter_free_state(ctx->fstate[i]);
     av_freep(&ctx->fstate);

     av_free(ctx);

 }