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ffmpeg / libavcodec / psymodel.c @ 78e65cd7

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/*
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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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FFPsyWindowInfo ff_psy_suggest_window(FFPsyContext *ctx,
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                                      const int16_t *audio, const int16_t *la,
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                                      int channel, int prev_type)
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{
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    return ctx->model->window(ctx, audio, la, channel, prev_type);
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}
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void ff_psy_set_band_info(FFPsyContext *ctx, int channel,
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                          const float *coeffs, FFPsyWindowInfo *wi)
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{
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    ctx->model->analyze(ctx, channel, coeffs, wi);
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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;
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    ctx = av_mallocz(sizeof(FFPsyPreprocessContext));
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    ctx->avctx = avctx;
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    if(avctx->flags & CODEC_FLAG_QSCALE)
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        cutoff_coeff = 1.0f / av_clip(1 + avctx->global_quality / FF_QUALITY_SCALE, 1, 8);
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    else
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        cutoff_coeff = avctx->bit_rate / (4.0f * avctx->sample_rate * avctx->channels);
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    ctx->fcoeffs = ff_iir_filter_init_coeffs(FF_FILTER_TYPE_BUTTERWORTH, FF_FILTER_MODE_LOWPASS,
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                                           FILT_ORDER, 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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        }
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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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}
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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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}
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