GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_apad.c Lines: 43 51 84.3 %
Date: 2020-08-14 10:39:37 Branches: 15 32 46.9 %

Line Branch Exec Source
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/*
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 * Copyright (c) 2012 Michael Niedermayer
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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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/**
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 * @file
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 * audio pad filter.
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 *
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 * Based on af_aresample.c
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 */
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/opt.h"
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#include "libavutil/samplefmt.h"
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#include "libavutil/avassert.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "internal.h"
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typedef struct APadContext {
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    const AVClass *class;
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    int64_t next_pts;
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    int packet_size;
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    int64_t pad_len, pad_len_left;
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    int64_t whole_len, whole_len_left;
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    int64_t pad_dur;
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    int64_t whole_dur;
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} APadContext;
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#define OFFSET(x) offsetof(APadContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption apad_options[] = {
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    { "packet_size", "set silence packet size",                                  OFFSET(packet_size), AV_OPT_TYPE_INT,   { .i64 = 4096 }, 0, INT_MAX, A },
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    { "pad_len",     "set number of samples of silence to add",                  OFFSET(pad_len),     AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, A },
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    { "whole_len",   "set minimum target number of samples in the audio stream", OFFSET(whole_len),   AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, A },
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    { "pad_dur",     "set duration of silence to add",                           OFFSET(pad_dur),     AV_OPT_TYPE_DURATION, { .i64 = 0 }, 0, INT64_MAX, A },
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    { "whole_dur",   "set minimum target duration in the audio stream",          OFFSET(whole_dur),   AV_OPT_TYPE_DURATION, { .i64 = 0 }, 0, INT64_MAX, A },
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    { NULL }
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};
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AVFILTER_DEFINE_CLASS(apad);
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static av_cold int init(AVFilterContext *ctx)
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{
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1
    APadContext *s = ctx->priv;
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1
    s->next_pts = AV_NOPTS_VALUE;
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    if (s->whole_len >= 0 && s->pad_len >= 0) {
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        av_log(ctx, AV_LOG_ERROR, "Both whole and pad length are set, this is not possible\n");
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        return AVERROR(EINVAL);
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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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    APadContext *s = ctx->priv;
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    if (s->whole_len >= 0) {
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        s->whole_len_left = FFMAX(s->whole_len_left - frame->nb_samples, 0);
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        av_log(ctx, AV_LOG_DEBUG,
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               "n_out:%d whole_len_left:%"PRId64"\n", frame->nb_samples, s->whole_len_left);
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    }
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    s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
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    return ff_filter_frame(ctx->outputs[0], frame);
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}
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static int request_frame(AVFilterLink *outlink)
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{
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    AVFilterContext *ctx = outlink->src;
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    APadContext *s = ctx->priv;
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    int ret;
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    ret = ff_request_frame(ctx->inputs[0]);
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    if (ret == AVERROR_EOF && !ctx->is_disabled) {
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2
        int n_out = s->packet_size;
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        AVFrame *outsamplesref;
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        if (s->whole_len >= 0 && s->pad_len < 0) {
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            s->pad_len = s->pad_len_left = s->whole_len_left;
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        }
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        if (s->pad_len >=0 || s->whole_len >= 0) {
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            n_out = FFMIN(n_out, s->pad_len_left);
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            s->pad_len_left -= n_out;
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            av_log(ctx, AV_LOG_DEBUG,
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                   "padding n_out:%d pad_len_left:%"PRId64"\n", n_out, s->pad_len_left);
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        }
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        if (!n_out)
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1
            return AVERROR_EOF;
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1
        outsamplesref = ff_get_audio_buffer(outlink, n_out);
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1
        if (!outsamplesref)
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            return AVERROR(ENOMEM);
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1
        av_assert0(outsamplesref->sample_rate == outlink->sample_rate);
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1
        av_assert0(outsamplesref->nb_samples  == n_out);
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1
        av_samples_set_silence(outsamplesref->extended_data, 0,
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                               n_out,
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                               outsamplesref->channels,
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1
                               outsamplesref->format);
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        outsamplesref->pts = s->next_pts;
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        if (s->next_pts != AV_NOPTS_VALUE)
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1
            s->next_pts += av_rescale_q(n_out, (AVRational){1, outlink->sample_rate}, outlink->time_base);
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        return ff_filter_frame(outlink, outsamplesref);
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    }
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    return ret;
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}
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static int config_output(AVFilterLink *outlink)
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{
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    AVFilterContext *ctx = outlink->src;
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    APadContext *s  = ctx->priv;
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    if (s->pad_dur)
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        s->pad_len = av_rescale(s->pad_dur, outlink->sample_rate, AV_TIME_BASE);
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    if (s->whole_dur)
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        s->whole_len = av_rescale(s->whole_dur, outlink->sample_rate, AV_TIME_BASE);
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    s->pad_len_left   = s->pad_len;
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    s->whole_len_left = s->whole_len;
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    return 0;
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}
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static const AVFilterPad apad_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 apad_outputs[] = {
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    {
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        .name          = "default",
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        .request_frame = request_frame,
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        .config_props  = config_output,
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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_apad = {
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    .name          = "apad",
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    .description   = NULL_IF_CONFIG_SMALL("Pad audio with silence."),
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    .init          = init,
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    .priv_size     = sizeof(APadContext),
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    .inputs        = apad_inputs,
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    .outputs       = apad_outputs,
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    .priv_class    = &apad_class,
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    .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
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};