libavcodec/psymodel.c

   1 /*
   2  * audio encoder psychoacoustic model
   3  * Copyright (C) 2008 Konstantin Shishkov
   4  *
   5  * This file is part of Libav.
   6  *
   7  * Libav 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  * Libav 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 Libav; if not, write to the Free Software
  19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  20  */
  21
  22 #include <string.h>
  23
  24 #include "avcodec.h"
  25 #include "psymodel.h"
  26 #include "iirfilter.h"
  27 #include "libavutil/mem.h"
  28
  29 extern const FFPsyModel ff_aac_psy_model;
  30
  31 av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
  32                         const uint8_t **bands, const int* num_bands,
  33                         int num_groups, const uint8_t *group_map)
  34 {
  35     int i, j, k = 0;
  36
  37     ctx->avctx = avctx;
  38     ctx->ch        = av_mallocz(sizeof(ctx->ch[0]) * avctx->channels * 2);
  39     ctx->group     = av_mallocz(sizeof(ctx->group[0]) * num_groups);
  40     ctx->bands     = av_malloc (sizeof(ctx->bands[0])     * num_lens);
  41     ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
  42     memcpy(ctx->bands,     bands,     sizeof(ctx->bands[0])     *  num_lens);
  43     memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) *  num_lens);
  44
  45     /* assign channels to groups (with virtual channels for coupling) */
  46     for (i = 0; i < num_groups; i++) {
  47         /* NOTE: Add 1 to handle the AAC chan_config without modification.
  48          *       This has the side effect of allowing an array of 0s to map
  49          *       to one channel per group.
  50          */
  51         ctx->group[i].num_ch = group_map[i] + 1;
  52         for (j = 0; j < ctx->group[i].num_ch * 2; j++)
  53             ctx->group[i].ch[j]  = &ctx->ch[k++];
  54     }
  55
  56     switch (ctx->avctx->codec_id) {
  57     case AV_CODEC_ID_AAC:
  58         ctx->model = &ff_aac_psy_model;
  59         break;
  60     }
  61     if (ctx->model->init)
  62         return ctx->model->init(ctx);
  63     return 0;
  64 }
  65
  66 FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
  67 {
  68     int i = 0, ch = 0;
  69
  70     while (ch <= channel)
  71         ch += ctx->group[i++].num_ch;
  72
  73     return &ctx->group[i-1];
  74 }
  75
  76 av_cold void ff_psy_end(FFPsyContext *ctx)
  77 {
  78     if (ctx->model->end)
  79         ctx->model->end(ctx);
  80     av_freep(&ctx->bands);
  81     av_freep(&ctx->num_bands);
  82     av_freep(&ctx->group);
  83     av_freep(&ctx->ch);
  84 }
  85
  86 typedef struct FFPsyPreprocessContext{
  87     AVCodecContext *avctx;
  88     float stereo_att;
  89     struct FFIIRFilterCoeffs *fcoeffs;
  90     struct FFIIRFilterState **fstate;
  91 }FFPsyPreprocessContext;
  92
  93 #define FILT_ORDER 4
  94
  95 av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
  96 {
  97     FFPsyPreprocessContext *ctx;
  98     int i;
  99     float cutoff_coeff = 0;
 100     ctx        = av_mallocz(sizeof(FFPsyPreprocessContext));
 101     ctx->avctx = avctx;
 102
 103     if (avctx->cutoff > 0)
 104         cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
 105
 106     if (cutoff_coeff)
 107     ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
 108                                              FF_FILTER_MODE_LOWPASS, FILT_ORDER,
 109                                              cutoff_coeff, 0.0, 0.0);
 110     if (ctx->fcoeffs) {
 111         ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
 112         for (i = 0; i < avctx->channels; i++)
 113             ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
 114     }
 115     return ctx;
 116 }
 117
 118 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
 119 {
 120     int ch;
 121     int frame_size = ctx->avctx->frame_size;
 122
 123     if (ctx->fstate) {
 124         for (ch = 0; ch < channels; ch++)
 125             ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
 126                               &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
 127     }
 128 }
 129
 130 av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
 131 {
 132     int i;
 133     ff_iir_filter_free_coeffs(ctx->fcoeffs);
 134     if (ctx->fstate)
 135         for (i = 0; i < ctx->avctx->channels; i++)
 136             ff_iir_filter_free_state(ctx->fstate[i]);
 137     av_freep(&ctx->fstate);
 138     av_free(ctx);
 139 }