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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 FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
00049                                       const int16_t *audio, const int16_t *la,
00050                                       int channel, int prev_type)
00051 {
00052     return ctx->model->window(ctx, audio, la, channel, prev_type);
00053 }
00054 
00055 void ff_psy_set_band_info(FFPsyContext *ctx, int channel,
00056                           const float *coeffs, FFPsyWindowInfo *wi)
00057 {
00058     ctx->model->analyze(ctx, channel, coeffs, wi);
00059 }
00060 
00061 av_cold void ff_psy_end(FFPsyContext *ctx)
00062 {
00063     if (ctx->model->end)
00064         ctx->model->end(ctx);
00065     av_freep(&ctx->bands);
00066     av_freep(&ctx->num_bands);
00067     av_freep(&ctx->psy_bands);
00068 }
00069 
00070 typedef struct FFPsyPreprocessContext{
00071     AVCodecContext *avctx;
00072     float stereo_att;
00073     struct FFIIRFilterCoeffs *fcoeffs;
00074     struct FFIIRFilterState **fstate;
00075 }FFPsyPreprocessContext;
00076 
00077 #define FILT_ORDER 4
00078 
00079 av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
00080 {
00081     FFPsyPreprocessContext *ctx;
00082     int i;
00083     float cutoff_coeff = 0;
00084     ctx        = av_mallocz(sizeof(FFPsyPreprocessContext));
00085     ctx->avctx = avctx;
00086 
00087     if (avctx->cutoff > 0)
00088         cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
00089 
00090     if (cutoff_coeff)
00091     ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS,
00092                                              FILT_ORDER, cutoff_coeff, 0.0, 0.0);
00093     if (ctx->fcoeffs) {
00094         ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
00095         for (i = 0; i < avctx->channels; i++)
00096             ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
00097     }
00098     return ctx;
00099 }
00100 
00101 void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
00102                        const int16_t *audio, int16_t *dest,
00103                        int tag, int channels)
00104 {
00105     int ch, i;
00106     if (ctx->fstate) {
00107         for (ch = 0; ch < channels; ch++)
00108             ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
00109                           audio + ch, ctx->avctx->channels,
00110                           dest  + ch, ctx->avctx->channels);
00111     } else {
00112         for (ch = 0; ch < channels; ch++)
00113             for (i = 0; i < ctx->avctx->frame_size; i++)
00114                 dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
00115     }
00116 }
00117 
00118 av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
00119 {
00120     int i;
00121     ff_iir_filter_free_coeffs(ctx->fcoeffs);
00122     if (ctx->fstate)
00123         for (i = 0; i < ctx->avctx->channels; i++)
00124             ff_iir_filter_free_state(ctx->fstate[i]);
00125     av_freep(&ctx->fstate);
00126 }
00127 

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