GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_volumedetect.c Lines: 0 60 0.0 %
Date: 2020-08-14 10:39:37 Branches: 0 40 0.0 %

Line Branch Exec Source
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/*
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 * Copyright (c) 2012 Nicolas George
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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 License
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 * 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
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 * GNU 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 License
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 * along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
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 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
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#include "libavutil/channel_layout.h"
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#include "libavutil/avassert.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct VolDetectContext {
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    /**
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     * Number of samples at each PCM value.
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     * histogram[0x8000 + i] is the number of samples at value i.
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     * The extra element is there for symmetry.
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     */
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    uint64_t histogram[0x10001];
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} VolDetectContext;
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static int query_formats(AVFilterContext *ctx)
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{
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    static const enum AVSampleFormat sample_fmts[] = {
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        AV_SAMPLE_FMT_S16,
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        AV_SAMPLE_FMT_S16P,
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        AV_SAMPLE_FMT_NONE
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    };
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    AVFilterFormats *formats;
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    AVFilterChannelLayouts *layouts;
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    int ret;
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    if (!(formats = ff_make_format_list(sample_fmts)))
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        return AVERROR(ENOMEM);
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    layouts = ff_all_channel_counts();
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    if (!layouts)
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        return AVERROR(ENOMEM);
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    ret = ff_set_common_channel_layouts(ctx, layouts);
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    if (ret < 0)
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        return ret;
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    return ff_set_common_formats(ctx, formats);
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *samples)
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{
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    AVFilterContext *ctx = inlink->dst;
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    VolDetectContext *vd = ctx->priv;
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    int nb_samples  = samples->nb_samples;
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    int nb_channels = samples->channels;
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    int nb_planes   = nb_channels;
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    int plane, i;
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    int16_t *pcm;
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    if (!av_sample_fmt_is_planar(samples->format)) {
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        nb_samples *= nb_channels;
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        nb_planes = 1;
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    }
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    for (plane = 0; plane < nb_planes; plane++) {
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        pcm = (int16_t *)samples->extended_data[plane];
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        for (i = 0; i < nb_samples; i++)
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            vd->histogram[pcm[i] + 0x8000]++;
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    }
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    return ff_filter_frame(inlink->dst->outputs[0], samples);
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}
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#define MAX_DB 91
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static inline double logdb(uint64_t v)
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{
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    double d = v / (double)(0x8000 * 0x8000);
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    if (!v)
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        return MAX_DB;
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    return -log10(d) * 10;
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}
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static void print_stats(AVFilterContext *ctx)
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{
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    VolDetectContext *vd = ctx->priv;
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    int i, max_volume, shift;
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    uint64_t nb_samples = 0, power = 0, nb_samples_shift = 0, sum = 0;
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    uint64_t histdb[MAX_DB + 1] = { 0 };
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    for (i = 0; i < 0x10000; i++)
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        nb_samples += vd->histogram[i];
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    av_log(ctx, AV_LOG_INFO, "n_samples: %"PRId64"\n", nb_samples);
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    if (!nb_samples)
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        return;
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    /* If nb_samples > 1<<34, there is a risk of overflow in the
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       multiplication or the sum: shift all histogram values to avoid that.
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       The total number of samples must be recomputed to avoid rounding
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       errors. */
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    shift = av_log2(nb_samples >> 33);
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    for (i = 0; i < 0x10000; i++) {
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        nb_samples_shift += vd->histogram[i] >> shift;
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        power += (i - 0x8000) * (i - 0x8000) * (vd->histogram[i] >> shift);
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    }
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    if (!nb_samples_shift)
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        return;
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    power = (power + nb_samples_shift / 2) / nb_samples_shift;
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    av_assert0(power <= 0x8000 * 0x8000);
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    av_log(ctx, AV_LOG_INFO, "mean_volume: %.1f dB\n", -logdb(power));
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    max_volume = 0x8000;
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    while (max_volume > 0 && !vd->histogram[0x8000 + max_volume] &&
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                             !vd->histogram[0x8000 - max_volume])
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        max_volume--;
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    av_log(ctx, AV_LOG_INFO, "max_volume: %.1f dB\n", -logdb(max_volume * max_volume));
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    for (i = 0; i < 0x10000; i++)
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        histdb[(int)logdb((i - 0x8000) * (i - 0x8000))] += vd->histogram[i];
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    for (i = 0; i <= MAX_DB && !histdb[i]; i++);
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    for (; i <= MAX_DB && sum < nb_samples / 1000; i++) {
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        av_log(ctx, AV_LOG_INFO, "histogram_%ddb: %"PRId64"\n", i, histdb[i]);
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        sum += histdb[i];
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    }
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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    print_stats(ctx);
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}
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static const AVFilterPad volumedetect_inputs[] = {
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    {
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        .name         = "default",
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        .type         = AVMEDIA_TYPE_AUDIO,
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        .filter_frame = filter_frame,
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    },
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    { NULL }
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};
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static const AVFilterPad volumedetect_outputs[] = {
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    {
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        .name = "default",
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        .type = AVMEDIA_TYPE_AUDIO,
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    },
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    { NULL }
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};
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AVFilter ff_af_volumedetect = {
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    .name          = "volumedetect",
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    .description   = NULL_IF_CONFIG_SMALL("Detect audio volume."),
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    .priv_size     = sizeof(VolDetectContext),
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    .query_formats = query_formats,
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    .uninit        = uninit,
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    .inputs        = volumedetect_inputs,
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    .outputs       = volumedetect_outputs,
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};