GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/psymodel.c Lines: 53 74 71.6 %
Date: 2019-11-22 03:34:36 Branches: 18 46 39.1 %

Line Branch Exec Source
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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 <string.h>
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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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#include "libavutil/mem.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, int num_lens,
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                        const uint8_t **bands, const int* num_bands,
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                        int num_groups, const uint8_t *group_map)
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{
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    int i, j, k = 0;
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    ctx->avctx = avctx;
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    ctx->ch        = av_mallocz_array(sizeof(ctx->ch[0]), avctx->channels * 2);
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    ctx->group     = av_mallocz_array(sizeof(ctx->group[0]), num_groups);
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    ctx->bands     = av_malloc_array (sizeof(ctx->bands[0]),      num_lens);
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    ctx->num_bands = av_malloc_array (sizeof(ctx->num_bands[0]),  num_lens);
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    ctx->cutoff    = avctx->cutoff;
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    if (!ctx->ch || !ctx->group || !ctx->bands || !ctx->num_bands) {
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        ff_psy_end(ctx);
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        return AVERROR(ENOMEM);
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    }
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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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    /* assign channels to groups (with virtual channels for coupling) */
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    for (i = 0; i < num_groups; i++) {
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        /* NOTE: Add 1 to handle the AAC chan_config without modification.
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         *       This has the side effect of allowing an array of 0s to map
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         *       to one channel per group.
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         */
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        ctx->group[i].num_ch = group_map[i] + 1;
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        for (j = 0; j < ctx->group[i].num_ch * 2; j++)
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            ctx->group[i].ch[j]  = &ctx->ch[k++];
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    }
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    switch (ctx->avctx->codec_id) {
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    case AV_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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FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel)
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{
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    int i = 0, ch = 0;
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    while (ch <= channel)
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        ch += ctx->group[i++].num_ch;
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    return &ctx->group[i-1];
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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 && 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->group);
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    av_freep(&ctx->ch);
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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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    struct FFIIRFilterContext fiir;
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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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    if (!ctx)
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        return NULL;
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    ctx->avctx = avctx;
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    /* AAC has its own LP method */
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    if (avctx->codec_id != AV_CODEC_ID_AAC) {
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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 && cutoff_coeff < 0.98)
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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_array(sizeof(ctx->fstate[0]), avctx->channels);
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            if (!ctx->fstate) {
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                av_free(ctx->fcoeffs);
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                av_free(ctx);
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                return NULL;
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            }
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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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    }
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    ff_iir_filter_init(&ctx->fiir);
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    return ctx;
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}
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void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels)
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{
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    int ch;
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    int frame_size = ctx->avctx->frame_size;
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    FFIIRFilterContext *iir = &ctx->fiir;
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    if (ctx->fstate) {
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        for (ch = 0; ch < channels; ch++)
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            iir->filter_flt(ctx->fcoeffs, ctx->fstate[ch], frame_size,
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                            &audio[ch][frame_size], 1, &audio[ch][frame_size], 1);
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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_coeffsp(&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_statep(&ctx->fstate[i]);
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    av_freep(&ctx->fstate);
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    av_free(ctx);
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}