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/*
* Copyright (c) 2013 Nicolas George
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public License
* along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/opt.h"
#include "avfilter.h"
#include "internal.h"
typedef struct ASetRateContext {
const AVClass *class;
int sample_rate;
int rescale_pts;
} ASetRateContext;
#define CONTEXT ASetRateContext
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \
{ name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type, \
{ .deffield = def }, min, max, FLAGS, __VA_ARGS__ }
#define OPT_INT(name, field, def, min, max, descr, ...) \
OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)
static const AVOption asetrate_options[] = {
OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
OPT_INT("r", sample_rate, 44100, 1, INT_MAX, "set the sample rate",),
{NULL},
};
AVFILTER_DEFINE_CLASS(asetrate);
static av_cold int query_formats(AVFilterContext *ctx)
{
ASetRateContext *sr = ctx->priv;
int sample_rates[] = { sr->sample_rate, -1 };
return ff_formats_ref(ff_make_format_list(sample_rates),
&ctx->outputs[0]->incfg.samplerates);
}
static av_cold int config_props(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ASetRateContext *sr = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
AVRational intb = ctx->inputs[0]->time_base;
int inrate = inlink->sample_rate;
if (intb.num == 1 && intb.den == inrate) {
outlink->time_base.num = 1;
outlink->time_base.den = outlink->sample_rate;
} else {
outlink->time_base = intb;
sr->rescale_pts = 1;
if (av_q2d(intb) > 1.0 / FFMAX(inrate, outlink->sample_rate))
av_log(ctx, AV_LOG_WARNING, "Time base is inaccurate\n");
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
AVFilterContext *ctx = inlink->dst;
ASetRateContext *sr = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
frame->sample_rate = outlink->sample_rate;
if (sr->rescale_pts)
frame->pts = av_rescale(frame->pts, inlink->sample_rate,
outlink->sample_rate);
return ff_filter_frame(outlink, frame);
}
static const AVFilterPad asetrate_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
};
static const AVFilterPad asetrate_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_props,
},
};
const AVFilter ff_af_asetrate = {
.name = "asetrate",
.description = NULL_IF_CONFIG_SMALL("Change the sample rate without "
"altering the data."),
.priv_size = sizeof(ASetRateContext),
FILTER_INPUTS(asetrate_inputs),
FILTER_OUTPUTS(asetrate_outputs),
FILTER_QUERY_FUNC(query_formats),
.priv_class = &asetrate_class,
};
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