Signed-off-by: Paul B Mahol <one...@gmail.com>
---
doc/filters.texi | 4 ++--
libavfilter/Makefile | 1 +
libavfilter/af_anlms.c | 22 +++++++++++++++++++++-
libavfilter/allfilters.c | 1 +
4 files changed, 25 insertions(+), 3 deletions(-)
diff --git a/doc/filters.texi b/doc/filters.texi
index da95688c53..05d4b1a56e 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2301,8 +2301,8 @@ Set smooth factor. Default value is @var{11}. Allowed
range is from @var{1} to @
This filter supports the all above options as @ref{commands}.
-@section anlms
-Apply Normalized Least-Mean-Squares algorithm to the first audio stream using
the second audio stream.
+@section anlmf, anlms
+Apply Normalized Least-Mean-(Squares|Fourth) algorithm to the first audio
stream using the second audio stream.
This adaptive filter is used to mimic a desired filter by finding the filter
coefficients that
relate to producing the least mean square of the error signal (difference
between the desired,
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 2fe495df28..e7bde11ec8 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -71,6 +71,7 @@ OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
OBJS-$(CONFIG_AMULTIPLY_FILTER) += af_amultiply.o
OBJS-$(CONFIG_ANEQUALIZER_FILTER) += af_anequalizer.o
OBJS-$(CONFIG_ANLMDN_FILTER) += af_anlmdn.o
+OBJS-$(CONFIG_ANLMF_FILTER) += af_anlms.o
OBJS-$(CONFIG_ANLMS_FILTER) += af_anlms.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
diff --git a/libavfilter/af_anlms.c b/libavfilter/af_anlms.c
index f1f82b491d..f20e3c0e82 100644
--- a/libavfilter/af_anlms.c
+++ b/libavfilter/af_anlms.c
@@ -54,6 +54,8 @@ typedef struct AudioNLMSContext {
AVFrame *frame[2];
+ int anlmf;
+
AVFloatDSPContext *fdsp;
} AudioNLMSContext;
@@ -74,7 +76,7 @@ static const AVOption anlms_options[] = {
{ NULL }
};
-AVFILTER_DEFINE_CLASS(anlms);
+AVFILTER_DEFINE_CLASS_EXT(anlms, "anlm(f|s)", anlms_options);
static int query_formats(AVFilterContext *ctx)
{
@@ -130,6 +132,8 @@ static float process_sample(AudioNLMSContext *s, float
input, float desired,
norm = s->eps + sum;
b = mu * e / norm;
+ if (s->anlmf)
+ b *= 4.f * e * e;
memcpy(tmp, delay + offset, order * sizeof(float));
@@ -241,6 +245,7 @@ static int config_output(AVFilterLink *outlink)
AVFilterContext *ctx = outlink->src;
AudioNLMSContext *s = ctx->priv;
+ s->anlmf = !strcmp(ctx->filter->name, "anlmf");
s->kernel_size = FFALIGN(s->order, 16);
if (!s->offset)
@@ -312,3 +317,18 @@ const AVFilter ff_af_anlms = {
.flags = AVFILTER_FLAG_SLICE_THREADS,
.process_command = ff_filter_process_command,
};
+
+const AVFilter ff_af_anlmf = {
+ .name = "anlmf",
+ .description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Fourth
algorithm to first audio stream."),
+ .priv_size = sizeof(AudioNLMSContext),
+ .priv_class = &anlms_class,
+ .init = init,
+ .uninit = uninit,
+ .activate = activate,
+ FILTER_INPUTS(inputs),
+ FILTER_OUTPUTS(outputs),
+ FILTER_QUERY_FUNC(query_formats),
+ .flags = AVFILTER_FLAG_SLICE_THREADS,
+ .process_command = ff_filter_process_command,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index ec57a2c49c..7281f98d86 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -64,6 +64,7 @@ extern const AVFilter ff_af_amix;
extern const AVFilter ff_af_amultiply;
extern const AVFilter ff_af_anequalizer;
extern const AVFilter ff_af_anlmdn;
+extern const AVFilter ff_af_anlmf;
extern const AVFilter ff_af_anlms;
extern const AVFilter ff_af_anull;
extern const AVFilter ff_af_apad;
--
2.33.0
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