Patch attached.
From db6a900bbe6a20afc44d3bc7b5416a2093d0c95f Mon Sep 17 00:00:00 2001
From: Paul B Mahol <one...@gmail.com>
Date: Wed, 25 Nov 2015 11:36:45 +0100
Subject: [PATCH] avfilter: add audio compressor filter
Signed-off-by: Paul B Mahol <one...@gmail.com>
---
doc/filters.texi | 72 +++++++++++++++++
libavfilter/Makefile | 1 +
libavfilter/af_sidechaincompress.c | 157 ++++++++++++++++++++++++++++++-------
libavfilter/allfilters.c | 1 +
4 files changed, 202 insertions(+), 29 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index eedd02f..7b3773e 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -318,6 +318,78 @@ build.
Below is a description of the currently available audio filters.
+@section acompressor
+
+A compressor is mainly used to reduce the dynamic of a signal.
+Especially modern music is mostly compressed at a high ratio to
+improve the overall loudness. It's done to get the highest attention
+of a listener, "fatten" the sound and bring more "power" to the track.
+If a signal is compressed too much it may sound dull or "dead"
+afterwards or it may start to "pump" (which could be a powerful effect
+but can also destroy a track completely).
+The right compression is the key to reach a professional sound and is
+the high art of mixing and mastering. Because of it's complex settings
+it may take a long time to get the right feeling for this kind of effect.
+
+Compression is done by detecting the volume above a chosen level
+@code{threshold} and divide it by the factor set with @code{ratio}.
+So if you set the threshold to -12dB and your signal reaches -6dB a ratio
+of 2:1 will result in a signal at -9dB. Because an exact manipulation of
+the signal would cause distrotion of the waveform the reduction can be
+levelled over the time. This is done by setting "Attack" and "Release".
+@code{attack} determines how long the signal has to rise above the threshold
+before any reduction will occur and @code{release} sets the time the signal
+has to fall below the threshold to reduce the reduction again. Shorter signals
+than the chosen attack time will be left untouched.
+The overall reduction of the signal can be made up afterwards with the
+@code{makeup} setting. So compressing the peaks of a signal about 6dB and
+rising the makeup to this level results in a signal two times louder than the
+source. To gain a softer entry in the compression the @code{knee} flattens the
+hard edge at the threshold in the range of the chosen decibels.
+
+The filter accepts the following options:
+
+@table @option
+@item threshold
+If a signal of second stream raises above this level it will affect the gain
+reduction of first stream.
+By default is 0.125. Range is between 0.00097563 and 1.
+
+@item ratio
+Set a ratio about which the signal is reduced. 1:2 means that if the level
+raised 4dB above the threshold, it will be only 2dB above after the reduction.
+Default is 2. Range is between 1 and 20.
+
+@item attack
+Amount of milliseconds the signal has to rise above the threshold before gain
+reduction starts. Default is 20. Range is between 0.01 and 2000.
+
+@item release
+Amount of milliseconds the signal has to fall below the threshold before
+reduction is decreased again. Default is 250. Range is between 0.01 and 9000.
+
+@item makeup
+Set the amount by how much signal will be amplified after processing.
+Default is 2. Range is from 1 and 64.
+
+@item knee
+Curve the sharp knee around the threshold to enter gain reduction more softly.
+Default is 2.82843. Range is between 1 and 8.
+
+@item link
+Choose if the @code{average} level between all channels of input stream
+or the louder(@code{maximum}) channel of input stream affects the
+reduction. Default is @code{average}.
+
+@item detection
+Should the exact signal be taken in case of @code{peak} or an RMS one in case
+of @code{rms}. Default is @code{rms} which is mainly smoother.
+
+@item mix
+How much to use compressed signal in output. Default is 1.
+Range is between 0 and 1.
+@end table
+
@section acrossfade
Apply cross fade from one input audio stream to another input audio stream.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index c896374..e31bdaa 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -23,6 +23,7 @@ OBJS = allfilters.o \
transform.o \
video.o \
+OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o
OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
diff --git a/libavfilter/af_sidechaincompress.c b/libavfilter/af_sidechaincompress.c
index 25f3fd1..1dce1c0 100644
--- a/libavfilter/af_sidechaincompress.c
+++ b/libavfilter/af_sidechaincompress.c
@@ -21,7 +21,7 @@
/**
* @file
- * Sidechain compressor filter
+ * Audio (Sidechain) Compressor filter
*/
#include "libavutil/avassert.h"
@@ -61,7 +61,7 @@ typedef struct SidechainCompressContext {
#define A AV_OPT_FLAG_AUDIO_PARAM
#define F AV_OPT_FLAG_FILTERING_PARAM
-static const AVOption sidechaincompress_options[] = {
+static const AVOption options[] = {
{ "threshold", "set threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0.125}, 0.000976563, 1, A|F },
{ "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 20, A|F },
{ "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 2000, A|F },
@@ -78,6 +78,7 @@ static const AVOption sidechaincompress_options[] = {
{ NULL }
};
+#define sidechaincompress_options options
AVFILTER_DEFINE_CLASS(sidechaincompress);
static av_cold int init(AVFilterContext *ctx)
@@ -126,33 +127,24 @@ static double output_gain(double lin_slope, double ratio, double thres,
return exp(gain - slope);
}
-static int filter_frame(AVFilterLink *link, AVFrame *frame)
+static int compressor_config_output(AVFilterLink *outlink)
{
- AVFilterContext *ctx = link->dst;
+ AVFilterContext *ctx = outlink->src;
SidechainCompressContext *s = ctx->priv;
- AVFilterLink *sclink = ctx->inputs[1];
- AVFilterLink *outlink = ctx->outputs[0];
- const double makeup = s->makeup;
- const double mix = s->mix;
- const double *scsrc;
- double *sample;
- int nb_samples;
- int ret, i, c;
- for (i = 0; i < 2; i++)
- if (link == ctx->inputs[i])
- break;
- av_assert0(i < 2 && !s->input_frame[i]);
- s->input_frame[i] = frame;
-
- if (!s->input_frame[0] || !s->input_frame[1])
- return 0;
+ s->attack_coeff = FFMIN(1., 1. / (s->attack * outlink->sample_rate / 4000.));
+ s->release_coeff = FFMIN(1., 1. / (s->release * outlink->sample_rate / 4000.));
- nb_samples = FFMIN(s->input_frame[0]->nb_samples,
- s->input_frame[1]->nb_samples);
+ return 0;
+}
- sample = (double *)s->input_frame[0]->data[0];
- scsrc = (const double *)s->input_frame[1]->data[0];
+static void compressor(SidechainCompressContext *s,
+ double *sample, const double *scsrc, int nb_samples,
+ AVFilterLink *inlink, AVFilterLink *sclink)
+{
+ const double makeup = s->makeup;
+ const double mix = s->mix;
+ int i, c;
for (i = 0; i < nb_samples; i++) {
double abs_sample, gain = 1.0;
@@ -179,13 +171,42 @@ static int filter_frame(AVFilterLink *link, AVFrame *frame)
s->knee_start, s->knee_stop,
s->compressed_knee_stop, s->detection);
- for (c = 0; c < outlink->channels; c++)
+ for (c = 0; c < inlink->channels; c++)
sample[c] *= (gain * makeup * mix + (1. - mix));
- sample += outlink->channels;
+ sample += inlink->channels;
scsrc += sclink->channels;
}
+}
+
+#if CONFIG_SIDECHAINCOMPRESS_FILTER
+static int filter_frame(AVFilterLink *link, AVFrame *frame)
+{
+ AVFilterContext *ctx = link->dst;
+ SidechainCompressContext *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] = frame;
+
+ 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];
+ compressor(s, sample, scsrc, nb_samples,
+ ctx->inputs[0], ctx->inputs[1]);
ret = ff_filter_frame(outlink, s->input_frame[0]);
s->input_frame[0] = NULL;
@@ -253,7 +274,6 @@ static int query_formats(AVFilterContext *ctx)
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
- SidechainCompressContext *s = ctx->priv;
if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
av_log(ctx, AV_LOG_ERROR,
@@ -268,8 +288,7 @@ static int config_output(AVFilterLink *outlink)
outlink->channel_layout = ctx->inputs[0]->channel_layout;
outlink->channels = ctx->inputs[0]->channels;
- s->attack_coeff = FFMIN(1., 1. / (s->attack * outlink->sample_rate / 4000.));
- s->release_coeff = FFMIN(1., 1. / (s->release * outlink->sample_rate / 4000.));
+ compressor_config_output(outlink);
return 0;
}
@@ -310,3 +329,83 @@ AVFilter ff_af_sidechaincompress = {
.inputs = sidechaincompress_inputs,
.outputs = sidechaincompress_outputs,
};
+#endif /* CONFIG_SIDECHAINCOMPRESS_FILTER */
+
+#if CONFIG_ACOMPRESSOR_FILTER
+static int acompressor_filter_frame(AVFilterLink *inlink, AVFrame *frame)
+{
+ AVFilterContext *ctx = inlink->dst;
+ SidechainCompressContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ double *sample;
+
+ sample = (double *)frame->data[0];
+ compressor(s, sample, sample, frame->nb_samples,
+ inlink, inlink);
+
+ return ff_filter_frame(outlink, frame);
+}
+
+static int acompressor_query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_DBL,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+#define acompressor_options options
+AVFILTER_DEFINE_CLASS(acompressor);
+
+static const AVFilterPad acompressor_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = acompressor_filter_frame,
+ .needs_writable = 1,
+ },
+ { NULL }
+};
+
+static const AVFilterPad acompressor_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = compressor_config_output,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_acompressor = {
+ .name = "acompressor",
+ .description = NULL_IF_CONFIG_SMALL("Audio compressor."),
+ .priv_size = sizeof(SidechainCompressContext),
+ .priv_class = &acompressor_class,
+ .init = init,
+ .query_formats = acompressor_query_formats,
+ .inputs = acompressor_inputs,
+ .outputs = acompressor_outputs,
+};
+#endif /* CONFIG_ACOMPRESSOR_FILTER */
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index a3f6e62..ccd3f35 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -45,6 +45,7 @@ void avfilter_register_all(void)
return;
initialized = 1;
+ REGISTER_FILTER(ACOMPRESSOR, acompressor, af);
REGISTER_FILTER(ACROSSFADE, acrossfade, af);
REGISTER_FILTER(ADELAY, adelay, af);
REGISTER_FILTER(AECHO, aecho, af);
--
1.9.1
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