Mercurial > libavcodec.hg
view psymodel.c @ 12262:dde20597f15e libavcodec
Use "const" qualifier for pointers that point to input data of
audio encoders.
This is purely for clarity/documentation purposes.
author | reimar |
---|---|
date | Sat, 24 Jul 2010 13:59:49 +0000 |
parents | a93946f63075 |
children | 94b578d0af10 |
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/* * audio encoder psychoacoustic model * Copyright (C) 2008 Konstantin Shishkov * * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "avcodec.h" #include "psymodel.h" #include "iirfilter.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) { ctx->avctx = avctx; ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels); 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); switch (ctx->avctx->codec_id) { case CODEC_ID_AAC: ctx->model = &ff_aac_psy_model; break; } if (ctx->model->init) return ctx->model->init(ctx); return 0; } FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx, const int16_t *audio, const int16_t *la, int channel, int prev_type) { return ctx->model->window(ctx, audio, la, channel, prev_type); } void ff_psy_set_band_info(FFPsyContext *ctx, int channel, const float *coeffs, FFPsyWindowInfo *wi) { ctx->model->analyze(ctx, channel, coeffs, wi); } 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->psy_bands); } 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(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, const int16_t *audio, int16_t *dest, int tag, int channels) { int ch, i; if (ctx->fstate) { for (ch = 0; ch < channels; ch++) ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size, audio + ch, ctx->avctx->channels, dest + ch, ctx->avctx->channels); } else { for (ch = 0; ch < channels; ch++) for (i = 0; i < ctx->avctx->frame_size; i++) dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch]; } } 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); }