[FFmpeg-cvslog] avfilter: add audio limiter filter
Paul B Mahol
git at videolan.org
Sun Sep 13 21:42:57 CEST 2015
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Sat Sep 5 19:12:58 2015 +0000| [39c61d84594780fc3f329806ab864a3734c84527] | committer: Paul B Mahol
avfilter: add audio limiter filter
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=39c61d84594780fc3f329806ab864a3734c84527
---
Changelog | 1 +
doc/filters.texi | 35 +++++
libavfilter/Makefile | 1 +
libavfilter/af_alimiter.c | 361 +++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/version.h | 2 +-
6 files changed, 400 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index 24d1f05..77607e5 100644
--- a/Changelog
+++ b/Changelog
@@ -5,6 +5,7 @@ version <next>:
- DXV decoding
- extrastereo filter
- ocr filter
+- alimiter filter
version 2.8:
diff --git a/doc/filters.texi b/doc/filters.texi
index 447caf5..4958374 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -641,6 +641,41 @@ Force the output to either unsigned 8-bit or signed 16-bit stereo
aformat=sample_fmts=u8|s16:channel_layouts=stereo
@end example
+ at section alimiter
+
+The limiter prevents input signal from raising over a desired threshold.
+This limiter uses lookahead technology to prevent your signal from distorting.
+It means that there is a small delay after signal is processed. Keep in mind
+that the delay it produces is the attack time you set.
+
+The filter accepts the following options:
+
+ at table @option
+ at item limit
+Don't let signals above this level pass the limiter. The removed amplitude is
+added automatically. Default is 1.
+
+ at item attack
+The limiter will reach its attenuation level in this amount of time in
+milliseconds. Default is 5 milliseconds.
+
+ at item release
+Come back from limiting to attenuation 1.0 in this amount of milliseconds.
+Default is 50 milliseconds.
+
+ at item asc
+When gain reduction is always needed ASC takes care of releasing to an
+average reduction level rather than reaching a reduction of 0 in the release
+time.
+
+ at item asc_level
+Select how much the release time is affected by ASC, 0 means nearly no changes
+in release time while 1 produces higher release times.
+ at end table
+
+Depending on picked setting it is recommended to upsample input 2x or 4x times
+with @ref{aresample} before applying this filter.
+
@section allpass
Apply a two-pole all-pass filter with central frequency (in Hz)
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index d3e282f..b0a2a7a 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -30,6 +30,7 @@ OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
+OBJS-$(CONFIG_ALIMITER_FILTER) += af_alimiter.o
OBJS-$(CONFIG_ALLPASS_FILTER) += af_biquads.o
OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o
OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
diff --git a/libavfilter/af_alimiter.c b/libavfilter/af_alimiter.c
new file mode 100644
index 0000000..37839de
--- /dev/null
+++ b/libavfilter/af_alimiter.c
@@ -0,0 +1,361 @@
+/*
+ * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
+ * Copyright (c) 2015 Paul B Mahol
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * Lookahead limiter filter
+ */
+
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/opt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "internal.h"
+
+typedef struct AudioLimiterContext {
+ const AVClass *class;
+
+ double limit;
+ double attack;
+ double release;
+ double att;
+ int auto_release;
+ double asc;
+ int asc_c;
+ int asc_pos;
+ double asc_coeff;
+
+ double *buffer;
+ int buffer_size;
+ int pos;
+ int *nextpos;
+ double *nextdelta;
+
+ double delta;
+ int nextiter;
+ int nextlen;
+ int asc_changed;
+} AudioLimiterContext;
+
+#define OFFSET(x) offsetof(AudioLimiterContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM
+#define F AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption alimiter_options[] = {
+ { "limit", "set limit", OFFSET(limit), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.0625, 1, A|F },
+ { "attack", "set attack", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=5}, 0.1, 80, A|F },
+ { "release", "set release", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 8000, A|F },
+ { "asc", "enable asc", OFFSET(auto_release), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A|F },
+ { "asc_level", "set asc level", OFFSET(asc_coeff), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A|F },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(alimiter);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ AudioLimiterContext *s = ctx->priv;
+
+ s->attack /= 1000.;
+ s->release /= 1000.;
+ s->att = 1.;
+ s->asc_pos = -1;
+ s->asc_coeff = pow(0.5, s->asc_coeff - 0.5) * 2 * -1;
+
+ return 0;
+}
+
+static double get_rdelta(AudioLimiterContext *s, double release, int sample_rate,
+ double peak, double limit, double patt, int asc)
+{
+ double rdelta = (1.0 - patt) / (sample_rate * release);
+
+ if (asc && s->auto_release && s->asc_c > 0) {
+ double a_att = limit / (s->asc_coeff * s->asc) * (double)s->asc_c;
+
+ if (a_att > patt) {
+ double delta = FFMAX((a_att - patt) / (sample_rate * release), rdelta / 10);
+
+ if (delta < rdelta)
+ rdelta = delta;
+ }
+ }
+
+ return rdelta;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioLimiterContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ const double *src = (const double *)in->data[0];
+ const int channels = inlink->channels;
+ const int buffer_size = s->buffer_size;
+ double *dst, *buffer = s->buffer;
+ const double release = s->release;
+ const double limit = s->limit;
+ double *nextdelta = s->nextdelta;
+ int *nextpos = s->nextpos;
+ AVFrame *out;
+ double *buf;
+ int n, c, i;
+
+ if (av_frame_is_writable(in)) {
+ out = in;
+ } else {
+ 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++) {
+ double peak = 0;
+
+ for (c = 0; c < channels; c++) {
+ double sample = src[c];
+
+ buffer[s->pos + c] = sample;
+ peak = FFMAX(peak, fabs(sample));
+ }
+
+ if (s->auto_release && peak > limit) {
+ s->asc += peak;
+ s->asc_c++;
+ }
+
+ if (peak > limit) {
+ double patt = FFMIN(limit / peak, 1.);
+ double rdelta = get_rdelta(s, release, inlink->sample_rate,
+ peak, limit, patt, 0);
+ double delta = (limit / peak - s->att) / buffer_size * channels;
+ int found = 0;
+
+ if (delta < s->delta) {
+ s->delta = delta;
+ nextpos[0] = s->pos;
+ nextpos[1] = -1;
+ nextdelta[0] = rdelta;
+ s->nextlen = 1;
+ s->nextiter= 0;
+ } else {
+ for (i = s->nextiter; i < s->nextiter + s->nextlen; i++) {
+ int j = i % buffer_size;
+ double ppeak, pdelta;
+
+ ppeak = fabs(buffer[nextpos[j]]) > fabs(buffer[nextpos[j] + 1]) ?
+ fabs(buffer[nextpos[j]]) : fabs(buffer[nextpos[j] + 1]);
+ pdelta = (limit / peak - limit / ppeak) / (((buffer_size - nextpos[j] + s->pos) % buffer_size) / channels);
+ if (pdelta < nextdelta[j]) {
+ nextdelta[j] = pdelta;
+ found = 1;
+ break;
+ }
+ }
+ if (found) {
+ s->nextlen = i - s->nextiter + 1;
+ nextpos[(s->nextiter + s->nextlen) % buffer_size] = s->pos;
+ nextdelta[(s->nextiter + s->nextlen) % buffer_size] = rdelta;
+ nextpos[(s->nextiter + s->nextlen + 1) % buffer_size] = -1;
+ s->nextlen++;
+ }
+ }
+ }
+
+ buf = &s->buffer[(s->pos + channels) % buffer_size];
+ peak = 0;
+ for (c = 0; c < channels; c++) {
+ double sample = buf[c];
+
+ peak = FFMAX(peak, fabs(sample));
+ }
+
+ if (s->pos == s->asc_pos && !s->asc_changed)
+ s->asc_pos = -1;
+
+ if (s->auto_release && s->asc_pos == -1 && peak > limit) {
+ s->asc -= peak;
+ s->asc_c--;
+ }
+
+ s->att += s->delta;
+
+ for (c = 0; c < channels; c++)
+ dst[c] = buf[c] * s->att;
+
+ if ((s->pos + channels) % buffer_size == nextpos[s->nextiter]) {
+ if (s->auto_release) {
+ s->delta = get_rdelta(s, release, inlink->sample_rate,
+ peak, limit, s->att, 1);
+ if (s->nextlen > 1) {
+ int pnextpos = nextpos[(s->nextiter + 1) % buffer_size];
+ double ppeak = fabs(buffer[pnextpos]) > fabs(buffer[pnextpos + 1]) ?
+ fabs(buffer[pnextpos]) :
+ fabs(buffer[pnextpos + 1]);
+ double pdelta = (limit / ppeak - s->att) /
+ (((buffer_size + pnextpos -
+ ((s->pos + channels) % buffer_size)) %
+ buffer_size) / channels);
+ if (pdelta < s->delta)
+ s->delta = pdelta;
+ }
+ } else {
+ s->delta = nextdelta[s->nextiter];
+ s->att = limit / peak;
+ }
+
+ s->nextlen -= 1;
+ nextpos[s->nextiter] = -1;
+ s->nextiter = (s->nextiter + 1) % buffer_size;
+ }
+
+ if (s->att > 1.) {
+ s->att = 1.;
+ s->delta = 0.;
+ s->nextiter = 0;
+ s->nextlen = 0;
+ nextpos[0] = -1;
+ }
+
+ if (s->att <= 0.) {
+ s->att = 0.0000000000001;
+ s->delta = (1.0 - s->att) / (inlink->sample_rate * release);
+ }
+
+ if (s->att != 1. && (1. - s->att) < 0.0000000000001)
+ s->att = 1.;
+
+ if (s->delta != 0. && fabs(s->delta) < 0.00000000000001)
+ s->delta = 0.;
+
+ for (c = 0; c < channels; c++)
+ dst[c] = av_clipd(dst[c], -limit, limit);
+
+ s->pos = (s->pos + channels) % buffer_size;
+ src += channels;
+ dst += channels;
+ }
+
+ if (in != out)
+ av_frame_free(&in);
+
+ return ff_filter_frame(outlink, out);
+}
+
+static int 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);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioLimiterContext *s = ctx->priv;
+ int obuffer_size;
+
+ obuffer_size = inlink->sample_rate * inlink->channels * 100 / 1000. + inlink->channels;
+ if (obuffer_size < inlink->channels)
+ return AVERROR(EINVAL);
+
+ s->buffer = av_calloc(obuffer_size, sizeof(*s->buffer));
+ s->nextdelta = av_calloc(obuffer_size, sizeof(*s->nextdelta));
+ s->nextpos = av_malloc_array(obuffer_size, sizeof(*s->nextpos));
+ if (!s->buffer || !s->nextdelta || !s->nextpos)
+ return AVERROR(ENOMEM);
+
+ memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
+ s->buffer_size = inlink->sample_rate * s->attack * inlink->channels;
+ s->buffer_size -= s->buffer_size % inlink->channels;
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioLimiterContext *s = ctx->priv;
+
+ av_freep(&s->buffer);
+ av_freep(&s->nextdelta);
+ av_freep(&s->nextpos);
+}
+
+static const AVFilterPad alimiter_inputs[] = {
+ {
+ .name = "main",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ .config_props = config_input,
+ },
+ { NULL }
+};
+
+static const AVFilterPad alimiter_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_alimiter = {
+ .name = "alimiter",
+ .description = NULL_IF_CONFIG_SMALL("Lookahead limiter."),
+ .priv_size = sizeof(AudioLimiterContext),
+ .priv_class = &alimiter_class,
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .inputs = alimiter_inputs,
+ .outputs = alimiter_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 639e473..c78d79b 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -52,6 +52,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(AFADE, afade, af);
REGISTER_FILTER(AFORMAT, aformat, af);
REGISTER_FILTER(AINTERLEAVE, ainterleave, af);
+ REGISTER_FILTER(ALIMITER, alimiter, af);
REGISTER_FILTER(ALLPASS, allpass, af);
REGISTER_FILTER(AMERGE, amerge, af);
REGISTER_FILTER(AMIX, amix, af);
diff --git a/libavfilter/version.h b/libavfilter/version.h
index e07c4c3..e7204bc 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 2
+#define LIBAVFILTER_VERSION_MINOR 3
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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