[FFmpeg-devel] [PATCH] avfilter: add anlms filter
Paul B Mahol
onemda at gmail.com
Fri Oct 4 11:04:46 EEST 2019
Signed-off-by: Paul B Mahol <onemda at gmail.com>
---
doc/filters.texi | 23 +++
libavfilter/Makefile | 1 +
libavfilter/af_anlms.c | 308 +++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
4 files changed, 333 insertions(+)
create mode 100644 libavfilter/af_anlms.c
diff --git a/doc/filters.texi b/doc/filters.texi
index fbc3a404dd..49884fe16e 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -1814,6 +1814,29 @@ Change output mode.
Syntax for the command is : "i", "o" or "n" string.
@end table
+ at section anlms
+Apply Normalized Least-Mean-Squares algorithm to first audio stream using second audio stream.
+
+This is adaptive filter 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,
+2nd input audio stream and the actual signal, 1st input audio stream).
+
+A description of the accepted options follows.
+
+ at table @option
+ at item order
+Set filter order.
+
+ at item mu
+Set filter mu.
+
+ at item eps
+Set the filter eps.
+
+ at item leakage
+Set the filter leakage.
+ at end table
+
@section anull
Pass the audio source unchanged to the output.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 182fe9df4b..16bb8cd965 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -63,6 +63,7 @@ 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-$(CONFIG_ANLMS_FILTER) += af_anlms.o
OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o
OBJS-$(CONFIG_APAD_FILTER) += af_apad.o
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o
diff --git a/libavfilter/af_anlms.c b/libavfilter/af_anlms.c
new file mode 100644
index 0000000000..571c0313e1
--- /dev/null
+++ b/libavfilter/af_anlms.c
@@ -0,0 +1,308 @@
+/*
+ * Copyright (c) 2019 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
+ */
+
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "filters.h"
+#include "internal.h"
+
+typedef struct AudioNLMSContext {
+ const AVClass *class;
+
+ int order;
+ float mu;
+ float eps;
+ float leakage;
+
+ int kernel_size;
+ AVFrame *offset;
+ AVFrame *delay;
+ AVFrame *coeffs;
+ AVFrame *tmp;
+
+ AVFrame *frame[2];
+
+ AVFloatDSPContext *fdsp;
+} AudioNLMSContext;
+
+#define OFFSET(x) offsetof(AudioNLMSContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption anlms_options[] = {
+ { "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=256}, 1, INT16_MAX, A },
+ { "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0, 1, A },
+ { "eps", "set the filter eps", OFFSET(eps), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, A },
+ { "leakage", "set the filter leakage", OFFSET(leakage), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, A },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(anlms);
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_FLTP,
+ 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 float fir_sample(AudioNLMSContext *s, float sample, float *delay,
+ float *coeffs, float *tmp, int *offset)
+{
+ const int order = s->order;
+ float output;
+
+ delay[*offset] = sample;
+
+ memcpy(tmp, coeffs + order - *offset, order * sizeof(float));
+
+ output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
+
+ if (--(*offset) < 0)
+ *offset = order - 1;
+
+ return output;
+}
+
+static float process_sample(AudioNLMSContext *s, float input, float desired,
+ float *delay, float *coeffs, float *tmp, int *offsetp)
+{
+ const int order = s->order;
+ const float leakage = s->leakage;
+ const float mu = s->mu;
+ const float a = 1.f - leakage * mu;
+ float sum, y, e, norm, b;
+ int offset = *offsetp;
+
+ delay[offset + order] = input;
+
+ y = fir_sample(s, input, delay, coeffs, tmp, offsetp);
+ e = desired - y;
+
+ sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
+
+ norm = s->eps + sum;
+ b = mu * e / norm;
+
+ memcpy(tmp, delay + offset, order * sizeof(float));
+
+ s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
+
+ s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
+
+ memcpy(coeffs + order, coeffs, order * sizeof(float));
+
+ return y;
+}
+
+static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+ AudioNLMSContext *s = ctx->priv;
+ AVFrame *out = arg;
+ const int start = (out->channels * jobnr) / nb_jobs;
+ const int end = (out->channels * (jobnr+1)) / nb_jobs;
+
+ for (int c = start; c < end; c++) {
+ const float *input = (const float *)s->frame[0]->extended_data[c];
+ const float *desired = (const float *)s->frame[1]->extended_data[c];
+ float *delay = (float *)s->delay->extended_data[c];
+ float *coeffs = (float *)s->coeffs->extended_data[c];
+ float *tmp = (float *)s->tmp->extended_data[c];
+ int *offset = (int *)s->offset->extended_data[c];
+ float *output = (float *)out->extended_data[c];
+
+ for (int n = 0; n < out->nb_samples; n++)
+ output[n] = process_sample(s, input[n], desired[n], delay, coeffs, tmp, offset);
+ }
+
+ return 0;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+ AudioNLMSContext *s = ctx->priv;
+ int i, ret, status;
+ int nb_samples;
+ int64_t pts;
+
+ FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
+
+ nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
+ ff_inlink_queued_samples(ctx->inputs[1]));
+ for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
+ if (s->frame[i])
+ continue;
+
+ if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
+ ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]);
+ if (ret < 0)
+ return ret;
+ }
+ }
+
+ if (s->frame[0] && s->frame[1]) {
+ AVFrame *out;
+
+ out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples);
+ if (!out) {
+ av_frame_free(&s->frame[0]);
+ av_frame_free(&s->frame[1]);
+ return AVERROR(ENOMEM);
+ }
+
+ ctx->internal->execute(ctx, process_channels, out, NULL, FFMIN(ctx->outputs[0]->channels,
+ ff_filter_get_nb_threads(ctx)));
+
+ out->pts = s->frame[0]->pts;
+
+ av_frame_free(&s->frame[0]);
+ av_frame_free(&s->frame[1]);
+
+ ret = ff_filter_frame(ctx->outputs[0], out);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (!nb_samples) {
+ for (i = 0; i < 2; i++) {
+ if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
+ ff_outlink_set_status(ctx->outputs[0], status, pts);
+ return 0;
+ }
+ }
+ }
+
+ if (ff_outlink_frame_wanted(ctx->outputs[0])) {
+ for (i = 0; i < 2; i++) {
+ if (ff_inlink_queued_samples(ctx->inputs[i]) > 0)
+ continue;
+ ff_inlink_request_frame(ctx->inputs[i]);
+ return 0;
+ }
+ }
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioNLMSContext *s = ctx->priv;
+
+ s->kernel_size = FFALIGN(s->order, 16);
+
+ if (!s->offset)
+ s->offset = ff_get_audio_buffer(outlink, 1);
+ if (!s->delay)
+ s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
+ if (!s->coeffs)
+ s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
+ if (!s->tmp)
+ s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
+ if (!s->delay || !s->coeffs || !s->offset || !s->tmp)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ AudioNLMSContext *s = ctx->priv;
+
+ s->fdsp = avpriv_float_dsp_alloc(0);
+ if (!s->fdsp)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioNLMSContext *s = ctx->priv;
+
+ av_freep(&s->fdsp);
+ av_frame_free(&s->delay);
+ av_frame_free(&s->coeffs);
+ av_frame_free(&s->offset);
+ av_frame_free(&s->tmp);
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "input",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ {
+ .name = "desired",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_anlms = {
+ .name = "anlms",
+ .description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Squares algorithm to first audio stream."),
+ .priv_size = sizeof(AudioNLMSContext),
+ .priv_class = &anlms_class,
+ .init = init,
+ .uninit = uninit,
+ .activate = activate,
+ .query_formats = query_formats,
+ .inputs = inputs,
+ .outputs = outputs,
+ .flags = AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 1a26129069..4f8b3039ed 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -56,6 +56,7 @@ extern AVFilter ff_af_amix;
extern AVFilter ff_af_amultiply;
extern AVFilter ff_af_anequalizer;
extern AVFilter ff_af_anlmdn;
+extern AVFilter ff_af_anlms;
extern AVFilter ff_af_anull;
extern AVFilter ff_af_apad;
extern AVFilter ff_af_aperms;
--
2.17.1
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