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lavfi: add asetrate filter.

tags/n2.0
Nicolas George 12 years ago
parent
commit
b57c23f8c8
6 changed files with 135 additions and 1 deletions
  1. +1
    -0
      Changelog
  2. +12
    -0
      doc/filters.texi
  3. +1
    -0
      libavfilter/Makefile
  4. +119
    -0
      libavfilter/af_asetrate.c
  5. +1
    -0
      libavfilter/allfilters.c
  6. +1
    -1
      libavfilter/version.h

+ 1
- 0
Changelog View File

@@ -28,6 +28,7 @@ version <next>:
- The matroska demuxer can now output proper verbatim ASS packets. It will
become the default at the next libavformat major bump.
- decent native animated GIF encoding
- asetrate filter


version 1.2:


+ 12
- 0
doc/filters.texi View File

@@ -900,6 +900,18 @@ disable padding for the last frame, use:
asetnsamples=n=1234:p=0
@end example

@section asetrate

Set the sample rate without altering the PCM data.
This will result in a change of speed and pitch.

The filter accepts the following options:

@table @option
@item sample_rate, r
Set the output sample rate. Default is 44100 Hz.
@end table

@section ashowinfo

Show a line containing various information for each input audio frame.


+ 1
- 0
libavfilter/Makefile View File

@@ -64,6 +64,7 @@ OBJS-$(CONFIG_ASELECT_FILTER) += f_select.o
OBJS-$(CONFIG_ASENDCMD_FILTER) += f_sendcmd.o
OBJS-$(CONFIG_ASETNSAMPLES_FILTER) += af_asetnsamples.o
OBJS-$(CONFIG_ASETPTS_FILTER) += f_setpts.o
OBJS-$(CONFIG_ASETRATE_FILTER) += af_asetrate.o
OBJS-$(CONFIG_ASETTB_FILTER) += f_settb.o
OBJS-$(CONFIG_ASHOWINFO_FILTER) += af_ashowinfo.o
OBJS-$(CONFIG_ASPLIT_FILTER) += split.o


+ 119
- 0
libavfilter/af_asetrate.c View File

@@ -0,0 +1,119 @@
/*
* 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 {
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 };

ff_formats_ref(ff_make_format_list(sample_rates),
&ctx->outputs[0]->in_samplerates);
return 0;
}

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,
},
{ NULL }
};

static const AVFilterPad asetrate_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_props,
},
{ NULL }
};

AVFilter avfilter_af_asetrate = {
.name = "asetrate",
.description = NULL_IF_CONFIG_SMALL("Change the sample rate without "
"altering the data."),
.query_formats = query_formats,
.priv_size = sizeof(ASetRateContext),
.inputs = asetrate_inputs,
.outputs = asetrate_outputs,
.priv_class = &asetrate_class,
};

+ 1
- 0
libavfilter/allfilters.c View File

@@ -62,6 +62,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(ASENDCMD, asendcmd, af);
REGISTER_FILTER(ASETNSAMPLES, asetnsamples, af);
REGISTER_FILTER(ASETPTS, asetpts, af);
REGISTER_FILTER(ASETRATE, asetrate, af);
REGISTER_FILTER(ASETTB, asettb, af);
REGISTER_FILTER(ASHOWINFO, ashowinfo, af);
REGISTER_FILTER(ASPLIT, asplit, af);


+ 1
- 1
libavfilter/version.h View File

@@ -29,7 +29,7 @@
#include "libavutil/avutil.h"

#define LIBAVFILTER_VERSION_MAJOR 3
#define LIBAVFILTER_VERSION_MINOR 58
#define LIBAVFILTER_VERSION_MINOR 59
#define LIBAVFILTER_VERSION_MICRO 100

#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \


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