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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00

avfilter: add dcshift filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Paul B Mahol 2015-01-28 15:46:58 +00:00
parent a05a737316
commit edf217ebb7
6 changed files with 188 additions and 2 deletions

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@ -22,6 +22,7 @@ version <next>:
- removed libmpcodecs
- Changed default DNxHD colour range in QuickTime .mov derivatives to mpeg range
- ported softpulldown filter from libmpcodecs as repeatfields filter
- dcshift filter
version 2.5:

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@ -917,6 +917,7 @@ audio, the data is treated as if all the planes were concatenated.
A list of Adler-32 checksums for each data plane.
@end table
@anchor{astats}
@section astats
Display time domain statistical information about the audio channels.
@ -1394,6 +1395,24 @@ compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
@end example
@end itemize
@section dcshift
Apply a DC shift to the audio.
This can be useful to remove a DC offset (caused perhaps by a hardware problem
in the recording chain) from the audio. The effect of a DC offset is reduced
headroom and hence volume. The @ref{astats} filter can be used to determine if
a signal has a DC offset.
@table @option
@item shift
Set the DC shift, allowed range is [-1, 1]. It indicates the amount to shift
the audio.
@item limitergain
Optional. It should have a value much less than 1 (e.g. 0.05 or 0.02) and is
used to prevent clipping.
@end table
@section earwax
Make audio easier to listen to on headphones.

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@ -65,6 +65,7 @@ OBJS-$(CONFIG_BS2B_FILTER) += af_bs2b.o
OBJS-$(CONFIG_CHANNELMAP_FILTER) += af_channelmap.o
OBJS-$(CONFIG_CHANNELSPLIT_FILTER) += af_channelsplit.o
OBJS-$(CONFIG_COMPAND_FILTER) += af_compand.o
OBJS-$(CONFIG_DCSHIFT_FILTER) += af_dcshift.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_EBUR128_FILTER) += f_ebur128.o
OBJS-$(CONFIG_EQUALIZER_FILTER) += af_biquads.o

164
libavfilter/af_dcshift.c Normal file
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@ -0,0 +1,164 @@
/*
* Copyright (c) 2000 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
* Copyright (c) 2000 Fabien COELHO <fabien@coelho.net>
*
* 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 "libavutil/samplefmt.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"
typedef struct DCShiftContext {
const AVClass *class;
double dcshift;
double limiterthreshhold;
double limitergain;
} DCShiftContext;
#define OFFSET(x) offsetof(DCShiftContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption dcshift_options[] = {
{ "shift", "set DC shift", OFFSET(dcshift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
{ "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
{ NULL }
};
AVFILTER_DEFINE_CLASS(dcshift);
static av_cold int init(AVFilterContext *ctx)
{
DCShiftContext *s = ctx->priv;
s->limiterthreshhold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterChannelLayouts *layouts;
AVFilterFormats *formats;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
};
layouts = ff_all_channel_layouts();
if (!layouts)
return AVERROR(ENOMEM);
ff_set_common_channel_layouts(ctx, layouts);
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_formats(ctx, formats);
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
ff_set_common_samplerates(ctx, formats);
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out = ff_get_audio_buffer(inlink, in->nb_samples);
DCShiftContext *s = ctx->priv;
int i, j;
double dcshift = s->dcshift;
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
if (s->limitergain > 0) {
for (i = 0; i < inlink->channels; i++) {
const int32_t *src = (int32_t *)in->extended_data[i];
int32_t *dst = (int32_t *)out->extended_data[i];
for (j = 0; j < in->nb_samples; j++) {
double d;
d = src[j];
if (d > s->limiterthreshhold && dcshift > 0) {
d = (d - s->limiterthreshhold) * s->limitergain /
(INT32_MAX - s->limiterthreshhold) +
s->limiterthreshhold + dcshift;
} else if (d < -s->limiterthreshhold && dcshift < 0) {
d = (d + s->limiterthreshhold) * s->limitergain /
(INT32_MAX - s->limiterthreshhold) -
s->limiterthreshhold + dcshift;
} else {
d = dcshift * INT32_MAX + d;
}
dst[j] = av_clipl_int32(d);
}
}
} else {
for (i = 0; i < inlink->channels; i++) {
const int32_t *src = (int32_t *)in->extended_data[i];
int32_t *dst = (int32_t *)out->extended_data[i];
for (j = 0; j < in->nb_samples; j++) {
double d = dcshift * (INT32_MAX + 1.) + src[j];
dst[j] = av_clipl_int32(d);
}
}
}
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static const AVFilterPad dcshift_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
},
{ NULL }
};
static const AVFilterPad dcshift_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
AVFilter ff_af_dcshift = {
.name = "dcshift",
.description = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
.query_formats = query_formats,
.priv_size = sizeof(DCShiftContext),
.priv_class = &dcshift_class,
.init = init,
.inputs = dcshift_inputs,
.outputs = dcshift_outputs,
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
};

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@ -81,6 +81,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(CHANNELMAP, channelmap, af);
REGISTER_FILTER(CHANNELSPLIT, channelsplit, af);
REGISTER_FILTER(COMPAND, compand, af);
REGISTER_FILTER(DCSHIFT, dcshift, af);
REGISTER_FILTER(EARWAX, earwax, af);
REGISTER_FILTER(EBUR128, ebur128, af);
REGISTER_FILTER(EQUALIZER, equalizer, af);

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@ -30,8 +30,8 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 5
#define LIBAVFILTER_VERSION_MINOR 9
#define LIBAVFILTER_VERSION_MICRO 104
#define LIBAVFILTER_VERSION_MINOR 10
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
LIBAVFILTER_VERSION_MINOR, \