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avfilter: add sidechaingate filter

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
pull/162/merge
Paul B Mahol 9 years ago
parent acc2347cf4
commit bd5afecdcb
6 changed files with 232 additions and 4 deletions
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  1. 1
      Changelog
  2. 61
      doc/filters.texi
  3. 1
      libavfilter/Makefile
  4. 170
      libavfilter/af_agate.c
  5. 1
      libavfilter/allfilters.c
  6. 2
      libavfilter/version.h

1
Changelog
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@ -38,6 +38,7 @@ version <next>:
- acompressor filter
- support encoding 16-bit RLE SGI images
- apulsator filter
- sidechaingate audio filter
version 2.8:

61
doc/filters.texi
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@ -2597,6 +2597,67 @@ ffmpeg -i main.flac -i sidechain.flac -filter_complex "[1:a]asplit=2[sc][mix];[0
@end example
@end itemize
@section sidechaingate
A sidechain gate acts like a normal (wideband) gate but has the ability to
filter the detected signal before sending it to the gain reduction stage.
Normally a gate uses the full range signal to detect a level above the
threshold.
For example: If you cut all lower frequencies from your sidechain signal
the gate will decrease the volume of your track only if not enough highs
appear. With this technique you are able to reduce the resonation of a
natural drum or remove "rumbling" of muted strokes from a heavily distorted
guitar.
It needs two input streams and returns one output stream.
First input stream will be processed depending on second stream signal.
The filter accepts the following options:
@table @option
@item level_in
Set input level before filtering.
Default is 1. Allowed range is from 0.015625 to 64.
@item range
Set the level of gain reduction when the signal is below the threshold.
Default is 0.06125. Allowed range is from 0 to 1.
@item threshold
If a signal rises above this level the gain reduction is released.
Default is 0.125. Allowed range is from 0 to 1.
@item ratio
Set a ratio about which the signal is reduced.
Default is 2. Allowed range is from 1 to 9000.
@item attack
Amount of milliseconds the signal has to rise above the threshold before gain
reduction stops.
Default is 20 milliseconds. Allowed range is from 0.01 to 9000.
@item release
Amount of milliseconds the signal has to fall below the threshold before the
reduction is increased again. Default is 250 milliseconds.
Allowed range is from 0.01 to 9000.
@item makeup
Set amount of amplification of signal after processing.
Default is 1. Allowed range is from 1 to 64.
@item knee
Curve the sharp knee around the threshold to enter gain reduction more softly.
Default is 2.828427125. Allowed range is from 1 to 8.
@item detection
Choose if exact signal should be taken for detection or an RMS like one.
Default is peak. Can be peak or rms.
@item link
Choose if the average level between all channels or the louder channel affects
the reduction.
Default is average. Can be average or maximum.
@end table
@section silencedetect
Detect silence in an audio stream.

1
libavfilter/Makefile
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@ -83,6 +83,7 @@ OBJS-$(CONFIG_REPLAYGAIN_FILTER) += af_replaygain.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_RUBBERBAND_FILTER) += af_rubberband.o
OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER) += af_sidechaincompress.o
OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o

170
libavfilter/af_agate.c
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@ -18,6 +18,12 @@
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* Audio (Sidechain) Gate filter
*/
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
@ -46,12 +52,14 @@ typedef struct AudioGateContext {
double lin_slope;
double attack_coeff;
double release_coeff;
AVFrame *input_frame[2];
} AudioGateContext;
#define OFFSET(x) offsetof(AudioGateContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption agate_options[] = {
static const AVOption options[] = {
{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
{ "range", "set max gain reduction", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=0.06125}, 0, 1, A },
{ "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0, 1, A },
@ -69,6 +77,7 @@ static const AVOption agate_options[] = {
{ NULL }
};
#define agate_options options
AVFILTER_DEFINE_CLASS(agate);
static int query_formats(AVFilterContext *ctx)
@ -97,7 +106,7 @@ static int query_formats(AVFilterContext *ctx)
return ff_set_common_samplerates(ctx, formats);
}
static int config_input(AVFilterLink *inlink)
static int agate_config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioGateContext *s = ctx->priv;
@ -221,7 +230,7 @@ static const AVFilterPad inputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
.config_props = agate_config_input,
},
{ NULL }
};
@ -243,3 +252,158 @@ AVFilter ff_af_agate = {
.inputs = inputs,
.outputs = outputs,
};
#if CONFIG_SIDECHAINGATE_FILTER
#define sidechaingate_options options
AVFILTER_DEFINE_CLASS(sidechaingate);
static int scfilter_frame(AVFilterLink *link, AVFrame *in)
{
AVFilterContext *ctx = link->dst;
AudioGateContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
const double *scsrc;
double *sample;
int nb_samples;
int ret, i;
for (i = 0; i < 2; i++)
if (link == ctx->inputs[i])
break;
av_assert0(i < 2 && !s->input_frame[i]);
s->input_frame[i] = in;
if (!s->input_frame[0] || !s->input_frame[1])
return 0;
nb_samples = FFMIN(s->input_frame[0]->nb_samples,
s->input_frame[1]->nb_samples);
sample = (double *)s->input_frame[0]->data[0];
scsrc = (const double *)s->input_frame[1]->data[0];
gate(s, sample, sample, scsrc, nb_samples,
ctx->inputs[0], ctx->inputs[1]);
ret = ff_filter_frame(outlink, s->input_frame[0]);
s->input_frame[0] = NULL;
av_frame_free(&s->input_frame[1]);
return ret;
}
static int screquest_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioGateContext *s = ctx->priv;
int i, ret;
/* get a frame on each input */
for (i = 0; i < 2; i++) {
AVFilterLink *inlink = ctx->inputs[i];
if (!s->input_frame[i] &&
(ret = ff_request_frame(inlink)) < 0)
return ret;
/* request the same number of samples on all inputs */
if (i == 0)
ctx->inputs[1]->request_samples = s->input_frame[0]->nb_samples;
}
return 0;
}
static int scquery_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts = NULL;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE
};
int ret, i;
if (!ctx->inputs[0]->in_channel_layouts ||
!ctx->inputs[0]->in_channel_layouts->nb_channel_layouts) {
av_log(ctx, AV_LOG_WARNING,
"No channel layout for input 1\n");
return AVERROR(EAGAIN);
}
if ((ret = ff_add_channel_layout(&layouts, ctx->inputs[0]->in_channel_layouts->channel_layouts[0])) < 0 ||
(ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts)) < 0)
return ret;
for (i = 0; i < 2; i++) {
layouts = ff_all_channel_counts();
if ((ret = ff_channel_layouts_ref(layouts, &ctx->inputs[i]->out_channel_layouts)) < 0)
return ret;
}
formats = ff_make_format_list(sample_fmts);
if ((ret = ff_set_common_formats(ctx, formats)) < 0)
return ret;
formats = ff_all_samplerates();
return ff_set_common_samplerates(ctx, formats);
}
static int scconfig_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
av_log(ctx, AV_LOG_ERROR,
"Inputs must have the same sample rate "
"%d for in0 vs %d for in1\n",
ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
return AVERROR(EINVAL);
}
outlink->sample_rate = ctx->inputs[0]->sample_rate;
outlink->time_base = ctx->inputs[0]->time_base;
outlink->channel_layout = ctx->inputs[0]->channel_layout;
outlink->channels = ctx->inputs[0]->channels;
agate_config_input(ctx->inputs[0]);
return 0;
}
static const AVFilterPad sidechaingate_inputs[] = {
{
.name = "main",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = scfilter_frame,
.needs_writable = 1,
.needs_fifo = 1,
},{
.name = "sidechain",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = scfilter_frame,
.needs_fifo = 1,
},
{ NULL }
};
static const AVFilterPad sidechaingate_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = scconfig_output,
.request_frame = screquest_frame,
},
{ NULL }
};
AVFilter ff_af_sidechaingate = {
.name = "sidechaingate",
.description = NULL_IF_CONFIG_SMALL("Audio sidechain gate."),
.priv_size = sizeof(AudioGateContext),
.priv_class = &sidechaingate_class,
.query_formats = scquery_formats,
.inputs = sidechaingate_inputs,
.outputs = sidechaingate_outputs,
};
#endif /* CONFIG_SIDECHAINGATE_FILTER */

1
libavfilter/allfilters.c
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@ -105,6 +105,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(RESAMPLE, resample, af);
REGISTER_FILTER(RUBBERBAND, rubberband, af);
REGISTER_FILTER(SIDECHAINCOMPRESS, sidechaincompress, af);
REGISTER_FILTER(SIDECHAINGATE, sidechaingate, af);
REGISTER_FILTER(SILENCEDETECT, silencedetect, af);
REGISTER_FILTER(SILENCEREMOVE, silenceremove, af);
REGISTER_FILTER(STEREOTOOLS, stereotools, af);

2
libavfilter/version.h
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@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
#define LIBAVFILTER_VERSION_MINOR 18
#define LIBAVFILTER_VERSION_MINOR 19
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \

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