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libavfilter
af_asetrate.c
Go to the documentation of this file.
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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
{
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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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ASetRateContext
*sr = ctx->
priv
;
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int
sample_rates
[] = { sr->
sample_rate
, -1 };
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ff_formats_ref
(
ff_make_format_list
(sample_rates),
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&ctx->
outputs
[0]->
in_samplerates
);
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return
0;
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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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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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};
Generated on Sun Mar 8 2015 02:35:05 for FFmpeg by
1.8.2