GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_asetrate.c Lines: 22 28 78.6 %
Date: 2020-09-25 23:16:12 Branches: 3 10 30.0 %

Line Branch Exec Source
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/*
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 * Copyright (c) 2013 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/opt.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct ASetRateContext {
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    const AVClass *class;
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    int sample_rate;
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    int rescale_pts;
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} ASetRateContext;
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#define CONTEXT ASetRateContext
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#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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#define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \
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    { name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type,          \
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      { .deffield = def }, min, max, FLAGS, __VA_ARGS__ }
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#define OPT_INT(name, field, def, min, max, descr, ...) \
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    OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)
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static const AVOption asetrate_options[] = {
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    OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
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    OPT_INT("r",           sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
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    {NULL},
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};
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AVFILTER_DEFINE_CLASS(asetrate);
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static av_cold int query_formats(AVFilterContext *ctx)
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{
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1
    ASetRateContext *sr = ctx->priv;
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    int sample_rates[] = { sr->sample_rate, -1 };
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    return ff_formats_ref(ff_make_format_list(sample_rates),
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                          &ctx->outputs[0]->incfg.samplerates);
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}
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static av_cold int config_props(AVFilterLink *outlink)
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{
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    AVFilterContext *ctx = outlink->src;
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    ASetRateContext *sr = ctx->priv;
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    AVFilterLink *inlink = ctx->inputs[0];
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    AVRational intb = ctx->inputs[0]->time_base;
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    int inrate = inlink->sample_rate;
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    if (intb.num == 1 && intb.den == inrate) {
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        outlink->time_base.num = 1;
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        outlink->time_base.den = outlink->sample_rate;
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    } else {
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        outlink->time_base = intb;
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        sr->rescale_pts = 1;
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        if (av_q2d(intb) > 1.0 / FFMAX(inrate, outlink->sample_rate))
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            av_log(ctx, AV_LOG_WARNING, "Time base is inaccurate\n");
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    }
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1
    return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
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{
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    AVFilterContext *ctx = inlink->dst;
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    ASetRateContext *sr = ctx->priv;
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    AVFilterLink *outlink = ctx->outputs[0];
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    frame->sample_rate = outlink->sample_rate;
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    if (sr->rescale_pts)
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        frame->pts = av_rescale(frame->pts, inlink->sample_rate,
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                                           outlink->sample_rate);
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    return ff_filter_frame(outlink, frame);
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}
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static const AVFilterPad asetrate_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 asetrate_outputs[] = {
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    {
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        .name         = "default",
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        .type         = AVMEDIA_TYPE_AUDIO,
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        .config_props = config_props,
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    },
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    { NULL }
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};
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AVFilter ff_af_asetrate = {
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    .name          = "asetrate",
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    .description   = NULL_IF_CONFIG_SMALL("Change the sample rate without "
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                                          "altering the data."),
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    .query_formats = query_formats,
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    .priv_size     = sizeof(ASetRateContext),
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    .inputs        = asetrate_inputs,
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    .outputs       = asetrate_outputs,
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    .priv_class    = &asetrate_class,
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};