diff --git a/Changelog b/Changelog index 12fe77c708..6200eebf71 100644 --- a/Changelog +++ b/Changelog @@ -10,6 +10,7 @@ version : - stereotools filter - rubberband filter - tremolo filter +- agate filter version 2.8: diff --git a/doc/filters.texi b/doc/filters.texi index 5bbbaf0b78..5b00eca19f 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -641,6 +641,57 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo aformat=sample_fmts=u8|s16:channel_layouts=stereo @end example +@section agate + +A gate is mainly used to reduce lower parts of a signal. This kind of signal +processing reduces disturbing noise between useful signals. + +Gating is done by detecting the volume below a chosen level @var{threshold} +and divide it by the factor set with @var{ratio}. The bottom of the noise +floor is set via @var{range}. Because an exact manipulation of the signal +would cause distortion of the waveform the reduction can be levelled over +time. This is done by setting @var{attack} and @var{release}. + +@var{attack} determines how long the signal has to fall below the threshold +before any reduction will occur and @var{release} sets the time the signal +has to raise above the threshold to reduce the reduction again. +Shorter signals than the chosen attack time will be left untouched. + +@table @option +@item level_in +Set input level before filtering. + +@item range +Set the level of gain reduction when the signal is below the threshold. + +@item threshold +If a signal rises above this level the gain reduction is released. + +@item ratio +Set a ratio about which the signal is reduced. + +@item attack +Amount of milliseconds the signal has to rise above the threshold before gain +reduction stops. + +@item release +Amount of milliseconds the signal has to fall below the threshold before the +reduction is increased again. + +@item makeup +Set amount of amplification of signal after processing. + +@item knee +Curve the sharp knee around the threshold to enter gain reduction more softly. + +@item detection +Choose if exact signal should be taken for detection or an RMS like one. + +@item link +Choose if the average level between all channels or the louder channel affects +the reduction. +@end table + @section alimiter The limiter prevents input signal from raising over a desired threshold. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 8db5da9bc6..be177db4e1 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -29,6 +29,7 @@ OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o +OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o OBJS-$(CONFIG_ALIMITER_FILTER) += af_alimiter.o OBJS-$(CONFIG_ALLPASS_FILTER) += af_biquads.o diff --git a/libavfilter/af_agate.c b/libavfilter/af_agate.c new file mode 100644 index 0000000000..46ee226235 --- /dev/null +++ b/libavfilter/af_agate.c @@ -0,0 +1,237 @@ +/* + * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit + * + * 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/channel_layout.h" +#include "libavutil/opt.h" +#include "avfilter.h" +#include "audio.h" +#include "formats.h" +#include "hermite.h" + +typedef struct AudioGateContext { + const AVClass *class; + + double level_in; + double attack; + double release; + double threshold; + double ratio; + double knee; + double makeup; + double range; + int link; + int detection; + + double thres; + double knee_start; + double lin_knee_stop; + double knee_stop; + double lin_slope; + double attack_coeff; + double release_coeff; +} AudioGateContext; + +#define OFFSET(x) offsetof(AudioGateContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption agate_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 }, + { "ratio", "set ratio", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 1, 9000, A }, + { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 9000, A }, + { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=250}, 0.01, 9000, A }, + { "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 64, A }, + { "knee", "set knee", OFFSET(knee), AV_OPT_TYPE_DOUBLE, {.dbl=2.828427125}, 1, 8, A }, + { "detection", "set detection", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "detection" }, + { "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "detection" }, + { "rms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "detection" }, + { "link", "set link", OFFSET(link), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "link" }, + { "average", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "link" }, + { "maximum", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "link" }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(agate); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layouts; + int ret; + + ff_add_format(&formats, AV_SAMPLE_FMT_DBL); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return 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_all_samplerates(); + if (!formats) + return AVERROR(ENOMEM); + + return ff_set_common_samplerates(ctx, formats); +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + AudioGateContext *s = ctx->priv; + double lin_threshold = s->threshold; + double lin_knee_sqrt = sqrt(s->knee); + double lin_knee_start; + + if (s->detection) + lin_threshold *= lin_threshold; + + s->attack_coeff = FFMIN(1., 1. / (s->attack * inlink->sample_rate / 4000.)); + s->release_coeff = FFMIN(1., 1. / (s->release * inlink->sample_rate / 4000.)); + s->lin_knee_stop = lin_threshold * lin_knee_sqrt; + lin_knee_start = lin_threshold / lin_knee_sqrt; + s->thres = log(lin_threshold); + s->knee_start = log(lin_knee_start); + s->knee_stop = log(s->lin_knee_stop); + + return 0; +} + +// A fake infinity value (because real infinity may break some hosts) +#define FAKE_INFINITY (65536.0 * 65536.0) + +// Check for infinity (with appropriate-ish tolerance) +#define IS_FAKE_INFINITY(value) (fabs(value-FAKE_INFINITY) < 1.0) + +static double output_gain(double lin_slope, double ratio, double thres, + double knee, double knee_start, double knee_stop, + double lin_knee_stop, double range) +{ + if (lin_slope < lin_knee_stop) { + double slope = log(lin_slope); + double tratio = ratio; + double gain = 0.; + double delta = 0.; + + if (IS_FAKE_INFINITY(ratio)) + tratio = 1000.; + gain = (slope - thres) * tratio + thres; + delta = tratio; + + if (knee > 1. && slope > knee_start) { + gain = hermite_interpolation(slope, knee_start, knee_stop, ((knee_start - thres) * tratio + thres), knee_stop, delta, 1.); + } + return FFMAX(range, exp(gain - slope)); + } + + return 1.; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AVFilterLink *outlink = ctx->outputs[0]; + AudioGateContext *s = ctx->priv; + const double *src = (const double *)in->data[0]; + const double makeup = s->makeup; + const double attack_coeff = s->attack_coeff; + const double release_coeff = s->release_coeff; + const double level_in = s->level_in; + AVFrame *out = NULL; + double *dst; + int n, c; + + if (av_frame_is_writable(in)) { + out = in; + } else { + AVFrame *out = ff_get_audio_buffer(inlink, in->nb_samples); + if (!out) { + av_frame_free(&in); + return AVERROR(ENOMEM); + } + av_frame_copy_props(out, in); + } + dst = (double *)out->data[0]; + + for (n = 0; n < in->nb_samples; n++, src += inlink->channels, dst += inlink->channels) { + double abs_sample = FFABS(src[0]), gain = 1.0; + + for (c = 0; c < inlink->channels; c++) + dst[c] = src[c] * level_in; + + if (s->link == 1) { + for (c = 1; c < inlink->channels; c++) + abs_sample = FFMAX(FFABS(src[c]), abs_sample); + } else { + for (c = 1; c < inlink->channels; c++) + abs_sample += FFABS(src[c]); + + abs_sample /= inlink->channels; + } + + if (s->detection) + abs_sample *= abs_sample; + + s->lin_slope += (abs_sample - s->lin_slope) * (abs_sample > s->lin_slope ? attack_coeff : release_coeff); + if (s->lin_slope > 0.0) + gain = output_gain(s->lin_slope, s->ratio, s->thres, + s->knee, s->knee_start, s->knee_stop, + s->lin_knee_stop, s->range); + + for (c = 0; c < inlink->channels; c++) + dst[c] *= gain * makeup; + } + + if (out != in) + av_frame_free(&in); + return ff_filter_frame(outlink, out); +} + +static const AVFilterPad inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_agate = { + .name = "agate", + .description = NULL_IF_CONFIG_SMALL("Audio gate."), + .query_formats = query_formats, + .priv_size = sizeof(AudioGateContext), + .priv_class = &agate_class, + .inputs = inputs, + .outputs = outputs, +}; diff --git a/libavfilter/af_sidechaincompress.c b/libavfilter/af_sidechaincompress.c index 40ffca6dc3..b8a81fc0bb 100644 --- a/libavfilter/af_sidechaincompress.c +++ b/libavfilter/af_sidechaincompress.c @@ -32,6 +32,7 @@ #include "audio.h" #include "avfilter.h" #include "formats.h" +#include "hermite.h" #include "internal.h" typedef struct SidechainCompressContext { @@ -90,29 +91,6 @@ static av_cold int init(AVFilterContext *ctx) return 0; } -static inline double hermite_interpolation(double x, double x0, double x1, - double p0, double p1, - double m0, double m1) -{ - double width = x1 - x0; - double t = (x - x0) / width; - double t2, t3; - double ct0, ct1, ct2, ct3; - - m0 *= width; - m1 *= width; - - t2 = t*t; - t3 = t2*t; - ct0 = p0; - ct1 = m0; - - ct2 = -3 * p0 - 2 * m0 + 3 * p1 - m1; - ct3 = 2 * p0 + m0 - 2 * p1 + m1; - - return ct3 * t3 + ct2 * t2 + ct1 * t + ct0; -} - // A fake infinity value (because real infinity may break some hosts) #define FAKE_INFINITY (65536.0 * 65536.0) diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 292ecde289..7c93c1dcda 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -51,6 +51,7 @@ void avfilter_register_all(void) REGISTER_FILTER(AEVAL, aeval, af); REGISTER_FILTER(AFADE, afade, af); REGISTER_FILTER(AFORMAT, aformat, af); + REGISTER_FILTER(AGATE, agate, af); REGISTER_FILTER(AINTERLEAVE, ainterleave, af); REGISTER_FILTER(ALIMITER, alimiter, af); REGISTER_FILTER(ALLPASS, allpass, af); diff --git a/libavfilter/hermite.h b/libavfilter/hermite.h new file mode 100644 index 0000000000..6142391b2e --- /dev/null +++ b/libavfilter/hermite.h @@ -0,0 +1,40 @@ +/* + * 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 + */ + +inline double hermite_interpolation(double x, double x0, double x1, + double p0, double p1, + double m0, double m1) +{ + double width = x1 - x0; + double t = (x - x0) / width; + double t2, t3; + double ct0, ct1, ct2, ct3; + + m0 *= width; + m1 *= width; + + t2 = t*t; + t3 = t2*t; + ct0 = p0; + ct1 = m0; + + ct2 = -3 * p0 - 2 * m0 + 3 * p1 - m1; + ct3 = 2 * p0 + m0 - 2 * p1 + m1; + + return ct3 * t3 + ct2 * t2 + ct1 * t + ct0; +} diff --git a/libavfilter/version.h b/libavfilter/version.h index 96503770e1..1426cf8100 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,7 +30,7 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 6 -#define LIBAVFILTER_VERSION_MINOR 7 +#define LIBAVFILTER_VERSION_MINOR 8 #define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \