mirror of https://git.ffmpeg.org/ffmpeg.git
avfilter: add anlms filter
This commit is contained in:
Paul B Mahol
parent
a27c0781dd
commit
e37edc70bd
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@ -14,6 +14,7 @@ version <next>:
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- sierpinski video source
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- scroll video filter
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- photosensitivity filter
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- anlms filter
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version 4.2:
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@ -1814,6 +1814,58 @@ Change output mode.
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Syntax for the command is : "i", "o" or "n" string.
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@end table
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@section anlms
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Apply Normalized Least-Mean-Squares algorithm to the first audio stream using the second audio stream.
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This adaptive filter is used to mimic a desired filter by finding the filter coefficients that
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relate to producing the least mean square of the error signal (difference between the desired,
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2nd input audio stream and the actual signal, the 1st input audio stream).
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A description of the accepted options follows.
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@table @option
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@item order
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Set filter order.
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@item mu
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Set filter mu.
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@item eps
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Set the filter eps.
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@item leakage
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Set the filter leakage.
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@item out_mode
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It accepts the following values:
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@table @option
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@item i
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Pass the 1st input.
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@item d
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Pass the 2nd input.
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@item o
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Pass filtered samples.
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@item n
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Pass difference between desired and filtered samples.
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Default value is @var{o}.
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@end table
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@end table
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@subsection Examples
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@itemize
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@item
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One of many usages of this filter is noise reduction, input audio is filtered
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with same samples that are delayed by fixed ammount, one such example for stereo audio is:
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@example
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asplit[a][b],[a]adelay=32S|32S[a],[b][a]anlms=order=128:leakage=0.0005:mu=.5:out_mode=o
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@end example
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@end itemize
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@section anull
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Pass the audio source unchanged to the output.
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@ -63,6 +63,7 @@ OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o
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OBJS-$(CONFIG_AMULTIPLY_FILTER) += af_amultiply.o
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OBJS-$(CONFIG_ANEQUALIZER_FILTER) += af_anequalizer.o
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OBJS-$(CONFIG_ANLMDN_FILTER) += af_anlmdn.o
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OBJS-$(CONFIG_ANLMS_FILTER) += af_anlms.o
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OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
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OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
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OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
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@ -0,0 +1,328 @@
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/*
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* Copyright (c) 2019 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/avassert.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/common.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/opt.h"
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#include "audio.h"
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#include "avfilter.h"
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#include "formats.h"
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#include "filters.h"
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#include "internal.h"
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enum OutModes {
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IN_MODE,
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DESIRED_MODE,
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OUT_MODE,
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NOISE_MODE,
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NB_OMODES
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};
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typedef struct AudioNLMSContext {
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const AVClass *class;
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int order;
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float mu;
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float eps;
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float leakage;
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int output_mode;
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int kernel_size;
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AVFrame *offset;
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AVFrame *delay;
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AVFrame *coeffs;
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AVFrame *tmp;
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AVFrame *frame[2];
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AVFloatDSPContext *fdsp;
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} AudioNLMSContext;
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#define OFFSET(x) offsetof(AudioNLMSContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption anlms_options[] = {
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{ "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=256}, 1, INT16_MAX, A },
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{ "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0, 1, A },
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{ "eps", "set the filter eps", OFFSET(eps), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, A },
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{ "leakage", "set the filter leakage", OFFSET(leakage), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
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{ "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, A, "mode" },
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{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, A, "mode" },
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{ "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, A, "mode" },
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{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, A, "mode" },
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{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, A, "mode" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(anlms);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_make_format_list(sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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if (!formats)
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return AVERROR(ENOMEM);
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return ff_set_common_samplerates(ctx, formats);
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}
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static float fir_sample(AudioNLMSContext *s, float sample, float *delay,
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float *coeffs, float *tmp, int *offset)
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{
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const int order = s->order;
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float output;
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delay[*offset] = sample;
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memcpy(tmp, coeffs + order - *offset, order * sizeof(float));
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output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
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if (--(*offset) < 0)
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*offset = order - 1;
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return output;
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}
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static float process_sample(AudioNLMSContext *s, float input, float desired,
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float *delay, float *coeffs, float *tmp, int *offsetp)
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{
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const int order = s->order;
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const float leakage = s->leakage;
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const float mu = s->mu;
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const float a = 1.f - leakage * mu;
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float sum, output, e, norm, b;
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int offset = *offsetp;
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delay[offset + order] = input;
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output = fir_sample(s, input, delay, coeffs, tmp, offsetp);
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e = desired - output;
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sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
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norm = s->eps + sum;
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b = mu * e / norm;
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memcpy(tmp, delay + offset, order * sizeof(float));
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s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
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s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
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memcpy(coeffs + order, coeffs, order * sizeof(float));
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switch (s->output_mode) {
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case IN_MODE: output = input; break;
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case DESIRED_MODE: output = desired; break;
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case OUT_MODE: /*output = output;*/ break;
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case NOISE_MODE: output = desired - output; break;
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}
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return output;
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}
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static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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AudioNLMSContext *s = ctx->priv;
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AVFrame *out = arg;
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const int start = (out->channels * jobnr) / nb_jobs;
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const int end = (out->channels * (jobnr+1)) / nb_jobs;
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for (int c = start; c < end; c++) {
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const float *input = (const float *)s->frame[0]->extended_data[c];
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const float *desired = (const float *)s->frame[1]->extended_data[c];
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float *delay = (float *)s->delay->extended_data[c];
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float *coeffs = (float *)s->coeffs->extended_data[c];
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float *tmp = (float *)s->tmp->extended_data[c];
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int *offset = (int *)s->offset->extended_data[c];
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float *output = (float *)out->extended_data[c];
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for (int n = 0; n < out->nb_samples; n++)
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output[n] = process_sample(s, input[n], desired[n], delay, coeffs, tmp, offset);
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}
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return 0;
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}
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static int activate(AVFilterContext *ctx)
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{
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AudioNLMSContext *s = ctx->priv;
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int i, ret, status;
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int nb_samples;
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int64_t pts;
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FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
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nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
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ff_inlink_queued_samples(ctx->inputs[1]));
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for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
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if (s->frame[i])
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continue;
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if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
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ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]);
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if (ret < 0)
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return ret;
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}
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}
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if (s->frame[0] && s->frame[1]) {
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AVFrame *out;
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out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples);
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if (!out) {
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av_frame_free(&s->frame[0]);
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av_frame_free(&s->frame[1]);
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return AVERROR(ENOMEM);
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}
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ctx->internal->execute(ctx, process_channels, out, NULL, FFMIN(ctx->outputs[0]->channels,
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ff_filter_get_nb_threads(ctx)));
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out->pts = s->frame[0]->pts;
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av_frame_free(&s->frame[0]);
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av_frame_free(&s->frame[1]);
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ret = ff_filter_frame(ctx->outputs[0], out);
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if (ret < 0)
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return ret;
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}
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if (!nb_samples) {
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for (i = 0; i < 2; i++) {
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
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ff_outlink_set_status(ctx->outputs[0], status, pts);
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return 0;
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}
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}
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}
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if (ff_outlink_frame_wanted(ctx->outputs[0])) {
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for (i = 0; i < 2; i++) {
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if (ff_inlink_queued_samples(ctx->inputs[i]) > 0)
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continue;
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ff_inlink_request_frame(ctx->inputs[i]);
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return 0;
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}
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}
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return 0;
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}
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static int config_output(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioNLMSContext *s = ctx->priv;
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s->kernel_size = FFALIGN(s->order, 16);
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if (!s->offset)
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s->offset = ff_get_audio_buffer(outlink, 1);
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if (!s->delay)
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s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
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if (!s->coeffs)
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s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
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if (!s->tmp)
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s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
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if (!s->delay || !s->coeffs || !s->offset || !s->tmp)
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return AVERROR(ENOMEM);
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return 0;
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioNLMSContext *s = ctx->priv;
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s->fdsp = avpriv_float_dsp_alloc(0);
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if (!s->fdsp)
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return AVERROR(ENOMEM);
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioNLMSContext *s = ctx->priv;
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av_freep(&s->fdsp);
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av_frame_free(&s->delay);
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av_frame_free(&s->coeffs);
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av_frame_free(&s->offset);
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av_frame_free(&s->tmp);
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "input",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{
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.name = "desired",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.config_props = config_output,
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},
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{ NULL }
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};
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AVFilter ff_af_anlms = {
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.name = "anlms",
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.description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Squares algorithm to first audio stream."),
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.priv_size = sizeof(AudioNLMSContext),
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.priv_class = &anlms_class,
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.init = init,
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.uninit = uninit,
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.activate = activate,
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.query_formats = query_formats,
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.inputs = inputs,
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.outputs = outputs,
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.flags = AVFILTER_FLAG_SLICE_THREADS,
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};
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@ -56,6 +56,7 @@ extern AVFilter ff_af_amix;
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extern AVFilter ff_af_amultiply;
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extern AVFilter ff_af_anequalizer;
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extern AVFilter ff_af_anlmdn;
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extern AVFilter ff_af_anlms;
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extern AVFilter ff_af_anull;
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extern AVFilter ff_af_apad;
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extern AVFilter ff_af_aperms;
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@ -30,7 +30,7 @@
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#include "libavutil/version.h"
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#define LIBAVFILTER_VERSION_MAJOR 7
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#define LIBAVFILTER_VERSION_MINOR 61
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#define LIBAVFILTER_VERSION_MINOR 62
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#define LIBAVFILTER_VERSION_MICRO 100
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