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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 FFmpeg.
00006  *
00007  * FFmpeg 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  * FFmpeg 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 FFmpeg; 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,
00029                         int num_lens,
00030                         const uint8_t **bands, const int* num_bands)
00031 {
00032     ctx->avctx = avctx;
00033     ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels);
00034     ctx->bands     = av_malloc (sizeof(ctx->bands[0])     * num_lens);
00035     ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
00036     memcpy(ctx->bands,     bands,     sizeof(ctx->bands[0])     *  num_lens);
00037     memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) *  num_lens);
00038     switch (ctx->avctx->codec_id) {
00039     case CODEC_ID_AAC:
00040         ctx->model = &ff_aac_psy_model;
00041         break;
00042     }
00043     if (ctx->model->init)
00044         return ctx->model->init(ctx);
00045     return 0;
00046 }
00047 
00048 av_cold void ff_psy_end(FFPsyContext *ctx)
00049 {
00050     if (ctx->model->end)
00051         ctx->model->end(ctx);
00052     av_freep(&ctx->bands);
00053     av_freep(&ctx->num_bands);
00054     av_freep(&ctx->psy_bands);
00055 }
00056 
00057 typedef struct FFPsyPreprocessContext{
00058     AVCodecContext *avctx;
00059     float stereo_att;
00060     struct FFIIRFilterCoeffs *fcoeffs;
00061     struct FFIIRFilterState **fstate;
00062 }FFPsyPreprocessContext;
00063 
00064 #define FILT_ORDER 4
00065 
00066 av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
00067 {
00068     FFPsyPreprocessContext *ctx;
00069     int i;
00070     float cutoff_coeff = 0;
00071     ctx        = av_mallocz(sizeof(FFPsyPreprocessContext));
00072     ctx->avctx = avctx;
00073 
00074     if (avctx->cutoff > 0)
00075         cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
00076 
00077     if (cutoff_coeff)
00078     ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
00079                                              FF_FILTER_MODE_LOWPASS, FILT_ORDER,
00080                                              cutoff_coeff, 0.0, 0.0);
00081     if (ctx->fcoeffs) {
00082         ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
00083         for (i = 0; i < avctx->channels; i++)
00084             ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
00085     }
00086     return ctx;
00087 }
00088 
00089 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
00090                        const int16_t *audio, int16_t *dest,
00091                        int tag, int channels)
00092 {
00093     int ch, i;
00094     if (ctx->fstate) {
00095         for (ch = 0; ch < channels; ch++)
00096             ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
00097                           audio + ch, ctx->avctx->channels,
00098                           dest  + ch, ctx->avctx->channels);
00099     } else {
00100         for (ch = 0; ch < channels; ch++)
00101             for (i = 0; i < ctx->avctx->frame_size; i++)
00102                 dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
00103     }
00104 }
00105 
00106 av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
00107 {
00108     int i;
00109     ff_iir_filter_free_coeffs(ctx->fcoeffs);
00110     if (ctx->fstate)
00111         for (i = 0; i < ctx->avctx->channels; i++)
00112             ff_iir_filter_free_state(ctx->fstate[i]);
00113     av_freep(&ctx->fstate);
00114     av_free(ctx);
00115 }
00116 

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