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https://github.com/FFmpeg/FFmpeg.git
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6841c8c579
* qatar/master: log: Fix an oob array read. cosmetics: trim trailing whitespace in postproc Ban strncpy() it's too easy to misuse. psymodel: Remove wrapper functions. aacenc: Replace loop counters in aac_encode_frame() with more descriptive 'ch' and 'w'. regtest: remove redundant flags in jpg test regtest: use run_ffmpeg in do_image_formats regtest: simplify encoding functions ffmpeg.c: check for interlaced flag in the correct place. Merged-by: Michael Niedermayer <michaelni@gmx.at>
117 lines
3.8 KiB
C
117 lines
3.8 KiB
C
/*
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* audio encoder psychoacoustic model
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* Copyright (C) 2008 Konstantin Shishkov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "avcodec.h"
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#include "psymodel.h"
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#include "iirfilter.h"
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extern const FFPsyModel ff_aac_psy_model;
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av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx,
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int num_lens,
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const uint8_t **bands, const int* num_bands)
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{
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ctx->avctx = avctx;
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ctx->psy_bands = av_mallocz(sizeof(FFPsyBand) * PSY_MAX_BANDS * avctx->channels);
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ctx->bands = av_malloc (sizeof(ctx->bands[0]) * num_lens);
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ctx->num_bands = av_malloc (sizeof(ctx->num_bands[0]) * num_lens);
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memcpy(ctx->bands, bands, sizeof(ctx->bands[0]) * num_lens);
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memcpy(ctx->num_bands, num_bands, sizeof(ctx->num_bands[0]) * num_lens);
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switch (ctx->avctx->codec_id) {
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case CODEC_ID_AAC:
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ctx->model = &ff_aac_psy_model;
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break;
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}
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if (ctx->model->init)
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return ctx->model->init(ctx);
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return 0;
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}
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av_cold void ff_psy_end(FFPsyContext *ctx)
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{
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if (ctx->model->end)
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ctx->model->end(ctx);
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av_freep(&ctx->bands);
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av_freep(&ctx->num_bands);
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av_freep(&ctx->psy_bands);
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}
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typedef struct FFPsyPreprocessContext{
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AVCodecContext *avctx;
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float stereo_att;
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struct FFIIRFilterCoeffs *fcoeffs;
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struct FFIIRFilterState **fstate;
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}FFPsyPreprocessContext;
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#define FILT_ORDER 4
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av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx)
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{
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FFPsyPreprocessContext *ctx;
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int i;
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float cutoff_coeff = 0;
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ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
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ctx->avctx = avctx;
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if (avctx->cutoff > 0)
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cutoff_coeff = 2.0 * avctx->cutoff / avctx->sample_rate;
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if (cutoff_coeff)
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ctx->fcoeffs = ff_iir_filter_init_coeffs(avctx, FF_FILTER_TYPE_BUTTERWORTH,
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FF_FILTER_MODE_LOWPASS, FILT_ORDER,
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cutoff_coeff, 0.0, 0.0);
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if (ctx->fcoeffs) {
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ctx->fstate = av_mallocz(sizeof(ctx->fstate[0]) * avctx->channels);
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for (i = 0; i < avctx->channels; i++)
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ctx->fstate[i] = ff_iir_filter_init_state(FILT_ORDER);
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}
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return ctx;
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}
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void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx,
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const int16_t *audio, int16_t *dest,
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int tag, int channels)
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{
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int ch, i;
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if (ctx->fstate) {
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for (ch = 0; ch < channels; ch++)
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ff_iir_filter(ctx->fcoeffs, ctx->fstate[tag+ch], ctx->avctx->frame_size,
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audio + ch, ctx->avctx->channels,
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dest + ch, ctx->avctx->channels);
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} else {
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for (ch = 0; ch < channels; ch++)
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for (i = 0; i < ctx->avctx->frame_size; i++)
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dest[i*ctx->avctx->channels + ch] = audio[i*ctx->avctx->channels + ch];
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}
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}
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av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx)
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{
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int i;
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ff_iir_filter_free_coeffs(ctx->fcoeffs);
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if (ctx->fstate)
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for (i = 0; i < ctx->avctx->channels; i++)
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ff_iir_filter_free_state(ctx->fstate[i]);
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av_freep(&ctx->fstate);
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av_free(ctx);
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}
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