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libavcodec/psymodel.c

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00001 /*
00002  * audio encoder psychoacoustic model
00003  * Copyright (C) 2008 Konstantin Shishkov
00004  *
00005  * This file is part of Libav.
00006  *
00007  * Libav is free software; you can redistribute it and/or
00008  * modify it under the terms of the GNU Lesser General Public
00009  * License as published by the Free Software Foundation; either
00010  * version 2.1 of the License, or (at your option) any later version.
00011  *
00012  * Libav is distributed in the hope that it will be useful,
00013  * but WITHOUT ANY WARRANTY; without even the implied warranty of
00014  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
00015  * Lesser General Public License for more details.
00016  *
00017  * You should have received a copy of the GNU Lesser General Public
00018  * License along with Libav; if not, write to the Free Software
00019  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
00020  */
00021 
00022 #include "avcodec.h"
00023 #include "psymodel.h"
00024 #include "iirfilter.h"
00025 
00026 extern const FFPsyModel ff_aac_psy_model;
00027 
00028 av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
00029                         const uint8_t **bands, const int* num_bands,
00030                         int num_groups, const uint8_t *group_map)
00031 {
00032     int i, j, k = 0;
00033 
00034     ctx->avctx = avctx;
00035     ctx->ch        = av_mallocz(sizeof(ctx->ch[0]) * avctx->channels * 2);
00036     ctx->group     = av_mallocz(sizeof(ctx->group[0]) * num_groups);
00037     ctx->bands     = av_malloc (sizeof(ctx->bands[0])     * num_lens);
00038     ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
00039     memcpy(ctx->bands,     bands,     sizeof(ctx->bands[0])     *  num_lens);
00040     memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) *  num_lens);
00041 
00042     /* assign channels to groups (with virtual channels for coupling) */
00043     for (i = 0; i < num_groups; i++) {
00044         /* NOTE: Add 1 to handle the AAC chan_config without modification.
00045          *       This has the side effect of allowing an array of 0s to map
00046          *       to one channel per group.
00047          */
00048         ctx->group[i].num_ch = group_map[i] + 1;
00049         for (j = 0; j < ctx->group[i].num_ch * 2; j++)
00050             ctx->group[i].ch[j]  = &ctx->ch[k++];
00051     }
00052 
00053     switch (ctx->avctx->codec_id) {
00054     case CODEC_ID_AAC:
00055         ctx->model = &ff_aac_psy_model;
00056         break;
00057     }
00058     if (ctx->model->init)
00059         return ctx->model->init(ctx);
00060     return 0;
00061 }
00062 
00063 FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
00064 {
00065     int i = 0, ch = 0;
00066 
00067     while (ch <= channel)
00068         ch += ctx->group[i++].num_ch;
00069 
00070     return &ctx->group[i-1];
00071 }
00072 
00073 av_cold void ff_psy_end(FFPsyContext *ctx)
00074 {
00075     if (ctx->model->end)
00076         ctx->model->end(ctx);
00077     av_freep(&ctx->bands);
00078     av_freep(&ctx->num_bands);
00079     av_freep(&ctx->group);
00080     av_freep(&ctx->ch);
00081 }
00082 
00083 typedef struct FFPsyPreprocessContext{
00084     AVCodecContext *avctx;
00085     float stereo_att;
00086     struct FFIIRFilterCoeffs *fcoeffs;
00087     struct FFIIRFilterState **fstate;
00088 }FFPsyPreprocessContext;
00089 
00090 #define FILT_ORDER 4
00091 
00092 av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
00093 {
00094     FFPsyPreprocessContext *ctx;
00095     int i;
00096     float cutoff_coeff = 0;
00097     ctx        = av_mallocz(sizeof(FFPsyPreprocessContext));
00098     ctx->avctx = avctx;
00099 
00100     if (avctx->cutoff > 0)
00101         cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
00102 
00103     if (cutoff_coeff)
00104     ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
00105                                              FF_FILTER_MODE_LOWPASS, FILT_ORDER,
00106                                              cutoff_coeff, 0.0, 0.0);
00107     if (ctx->fcoeffs) {
00108         ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
00109         for (i = 0; i < avctx->channels; i++)
00110             ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
00111     }
00112     return ctx;
00113 }
00114 
00115 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
00116                        const int16_t *audio, int16_t *dest,
00117                        int tag, int channels)
00118 {
00119     int ch, i;
00120     if (ctx->fstate) {
00121         for (ch = 0; ch < channels; ch++)
00122             ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
00123                           audio + ch, ctx->avctx->channels,
00124                           dest  + ch, ctx->avctx->channels);
00125     } else {
00126         for (ch = 0; ch < channels; ch++)
00127             for (i = 0; i < ctx->avctx->frame_size; i++)
00128                 dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
00129     }
00130 }
00131 
00132 av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
00133 {
00134     int i;
00135     ff_iir_filter_free_coeffs(ctx->fcoeffs);
00136     if (ctx->fstate)
00137         for (i = 0; i < ctx->avctx->channels; i++)
00138             ff_iir_filter_free_state(ctx->fstate[i]);
00139     av_freep(&ctx->fstate);
00140     av_free(ctx);
00141 }
00142 
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