Libav 0.7.1
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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, 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