/* * audio encoder psychoacoustic model * Copyright (C) 2008 Konstantin Shishkov * * This file is part of Libav. * * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include <string.h> #include "avcodec.h" #include "psymodel.h" #include "iirfilter.h" #include "libavutil/mem.h" extern const FFPsyModel ff_aac_psy_model; av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens, const uint8_t **bands, const int* num_bands, int num_groups, const uint8_t *group_map) { int i, j, k = 0; ctx->avctx = avctx; ctx->ch = av_mallocz(sizeof(ctx->ch[0]) * avctx->channels * 2); ctx->group = av_mallocz(sizeof(ctx->group[0]) * num_groups); ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens); ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens); memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens); memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens); /* assign channels to groups (with virtual channels for coupling) */ for (i = 0; i < num_groups; i++) { /* NOTE: Add 1 to handle the AAC chan_config without modification. * This has the side effect of allowing an array of 0s to map * to one channel per group. */ ctx->group[i].num_ch = group_map[i] + 1; for (j = 0; j < ctx->group[i].num_ch * 2; j++) ctx->group[i].ch[j] = &ctx->ch[k++]; } switch (ctx->avctx->codec_id) { case AV_CODEC_ID_AAC: ctx->model = &ff_aac_psy_model; break; } if (ctx->model->init) return ctx->model->init(ctx); return 0; } FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel) { int i = 0, ch = 0; while (ch <= channel) ch += ctx->group[i++].num_ch; return &ctx->group[i-1]; } av_cold void ff_psy_end(FFPsyContext *ctx) { if (ctx->model->end) ctx->model->end(ctx); av_freep(&ctx->bands); av_freep(&ctx->num_bands); av_freep(&ctx->group); av_freep(&ctx->ch); } typedef struct FFPsyPreprocessContext{ AVCodecContext *avctx; float stereo_att; struct FFIIRFilterCoeffs *fcoeffs; struct FFIIRFilterState **fstate; }FFPsyPreprocessContext; #define FILT_ORDER 4 av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx) { FFPsyPreprocessContext *ctx; int i; float cutoff_coeff = 0; ctx = av_mallocz(sizeof(FFPsyPreprocessContext)); ctx->avctx = avctx; if (avctx->cutoff > 0) cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate; if (cutoff_coeff) ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS, FILT_ORDER, cutoff_coeff, 0.0, 0.0); if (ctx->fcoeffs) { ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels); for (i = 0; i < avctx->channels; i++) ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER); } return ctx; } void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels) { int ch; int frame_size = ctx->avctx->frame_size; if (ctx->fstate) { for (ch = 0; ch < channels; ch++) ff_iir_filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size, &audio[ch][frame_size], 1, &audio[ch][frame_size], 1); } } av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx) { int i; ff_iir_filter_free_coeffs(ctx->fcoeffs); if (ctx->fstate) for (i = 0; i < ctx->avctx->channels; i++) ff_iir_filter_free_state(ctx->fstate[i]); av_freep(&ctx->fstate); av_free(ctx); }