From cda21a1e12c8dbc5dbf39556cec26b3e255184ec Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Sun, 16 Apr 2023 18:53:07 +0200
Subject: [PATCH] avfilter: add arls filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
libavfilter/Makefile | 1 +
libavfilter/af_arls.c | 353 +++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
3 files changed, 355 insertions(+)
create mode 100644 libavfilter/af_arls.c
@@ -90,6 +90,7 @@ OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o
OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o
OBJS-$(CONFIG_AREVERSE_FILTER) += f_reverse.o
+OBJS-$(CONFIG_ARLS_FILTER) += af_arls.o
OBJS-$(CONFIG_ARNNDN_FILTER) += af_arnndn.o
OBJS-$(CONFIG_ASDR_FILTER) += af_asdr.o
OBJS-$(CONFIG_ASEGMENT_FILTER) += f_segment.o
new file mode 100644
@@ -0,0 +1,353 @@
+/*
+ * Copyright (c) 2023 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/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"
+
+enum OutModes {
+ IN_MODE,
+ DESIRED_MODE,
+ OUT_MODE,
+ NOISE_MODE,
+ NB_OMODES
+};
+
+typedef struct AudioRLSContext {
+ const AVClass *class;
+
+ int order;
+ float lambda;
+ float delta;
+ int output_mode;
+
+ int kernel_size;
+ AVFrame *offset;
+ AVFrame *delay;
+ AVFrame *coeffs;
+ AVFrame *p, *dp;
+ AVFrame *gains;
+ AVFrame *u, *tmp;
+
+ AVFrame *frame[2];
+
+ AVFloatDSPContext *fdsp;
+} AudioRLSContext;
+
+#define OFFSET(x) offsetof(AudioRLSContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+#define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
+
+static const AVOption arls_options[] = {
+ { "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=16}, 1, INT16_MAX, A },
+ { "lambda", "set the filter lambda", OFFSET(lambda), AV_OPT_TYPE_FLOAT, {.dbl=1.f}, 0, 1, AT },
+ { "delta", "set the filter delta", OFFSET(delta), AV_OPT_TYPE_FLOAT, {.dbl=2.f}, 0, INT16_MAX, A },
+ { "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, "mode" },
+ { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, "mode" },
+ { "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, "mode" },
+ { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, "mode" },
+ { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, "mode" },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(arls);
+
+static float fir_sample(AudioRLSContext *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(AudioRLSContext *s, float input, float desired, int ch)
+{
+ float *coeffs = (float *)s->coeffs->extended_data[ch];
+ float *delay = (float *)s->delay->extended_data[ch];
+ float *gains = (float *)s->gains->extended_data[ch];
+ float *tmp = (float *)s->tmp->extended_data[ch];
+ float *u = (float *)s->u->extended_data[ch];
+ float *p = (float *)s->p->extended_data[ch];
+ float *dp = (float *)s->dp->extended_data[ch];
+ int *offsetp = (int *)s->offset->extended_data[ch];
+ const int kernel_size = s->kernel_size;
+ const int order = s->order;
+ const float lambda = s->lambda;
+ int offset = *offsetp;
+ float g = lambda;
+ float output, e;
+
+ delay[offset + order] = input;
+
+ output = fir_sample(s, input, delay, coeffs, tmp, offsetp);
+ e = desired - output;
+
+ for (int i = 0, pos = offset; i < order; i++, pos++) {
+ const int ikernel_size = i * kernel_size;
+
+ u[i] = 0.f;
+ for (int k = 0, pos = offset; k < order; k++, pos++)
+ u[i] += p[ikernel_size + k] * delay[pos];
+
+ g += u[i] * delay[pos];
+ }
+
+ g = 1.f / g;
+
+ for (int i = 0; i < order; i++) {
+ const int ikernel_size = i * kernel_size;
+
+ gains[i] = u[i] * g;
+ coeffs[i] = coeffs[order + i] = coeffs[i] + gains[i] * e;
+ tmp[i] = 0.f;
+ for (int k = 0, pos = offset; k < order; k++, pos++)
+ tmp[i] += p[ikernel_size + k] * delay[pos];
+ }
+
+ for (int i = 0; i < order; i++) {
+ const int ikernel_size = i * kernel_size;
+
+ for (int k = 0; k < order; k++)
+ dp[ikernel_size + k] = gains[i] * tmp[k];
+ }
+
+ for (int i = 0; i < order; i++) {
+ const int ikernel_size = i * kernel_size;
+
+ for (int k = 0; k < order; k++)
+ p[ikernel_size + k] = (p[ikernel_size + k] - (dp[ikernel_size + k] + dp[kernel_size * k + i]) * 0.5f) * lambda;
+ }
+
+ switch (s->output_mode) {
+ case IN_MODE: output = input; break;
+ case DESIRED_MODE: output = desired; break;
+ case OUT_MODE: output = desired - output; break;
+ case NOISE_MODE: output = input - output; break;
+ }
+ return output;
+}
+
+static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+ AudioRLSContext *s = ctx->priv;
+ AVFrame *out = arg;
+ const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
+ const int end = (out->ch_layout.nb_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 *output = (float *)out->extended_data[c];
+
+ for (int n = 0; n < out->nb_samples; n++) {
+ output[n] = process_sample(s, input[n], desired[n], c);
+ if (ctx->is_disabled)
+ output[n] = input[n];
+ }
+ }
+
+ return 0;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+ AudioRLSContext *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);
+ }
+
+ ff_filter_execute(ctx, process_channels, out, NULL,
+ FFMIN(ctx->outputs[0]->ch_layout.nb_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;
+ AudioRLSContext *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->gains)
+ s->gains = ff_get_audio_buffer(outlink, s->kernel_size);
+ if (!s->p)
+ s->p = ff_get_audio_buffer(outlink, s->kernel_size * s->kernel_size);
+ if (!s->dp)
+ s->dp = ff_get_audio_buffer(outlink, s->kernel_size * s->kernel_size);
+ if (!s->u)
+ s->u = ff_get_audio_buffer(outlink, s->kernel_size);
+ if (!s->tmp)
+ s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
+
+ if (!s->delay || !s->coeffs || !s->p || !s->dp || !s->gains || !s->offset || !s->u || !s->tmp)
+ return AVERROR(ENOMEM);
+
+ for (int ch = 0; ch < s->offset->ch_layout.nb_channels; ch++) {
+ int *dst = (int *)s->offset->extended_data[ch];
+
+ for (int i = 0; i < s->kernel_size; i++)
+ dst[0] = s->kernel_size - 1;
+ }
+
+ for (int ch = 0; ch < s->p->ch_layout.nb_channels; ch++) {
+ float *dst = (float *)s->p->extended_data[ch];
+
+ for (int i = 0; i < s->kernel_size; i++)
+ dst[i * s->kernel_size + i] = s->delta;
+ }
+
+ return 0;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ AudioRLSContext *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)
+{
+ AudioRLSContext *s = ctx->priv;
+
+ av_freep(&s->fdsp);
+ av_frame_free(&s->delay);
+ av_frame_free(&s->coeffs);
+ av_frame_free(&s->gains);
+ av_frame_free(&s->offset);
+ av_frame_free(&s->p);
+ av_frame_free(&s->dp);
+ av_frame_free(&s->u);
+ av_frame_free(&s->tmp);
+}
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "input",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ {
+ .name = "desired",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+};
+
+const AVFilter ff_af_arls = {
+ .name = "arls",
+ .description = NULL_IF_CONFIG_SMALL("Apply Recursive Least Squares algorithm to first audio stream."),
+ .priv_size = sizeof(AudioRLSContext),
+ .priv_class = &arls_class,
+ .init = init,
+ .uninit = uninit,
+ .activate = activate,
+ FILTER_INPUTS(inputs),
+ FILTER_OUTPUTS(outputs),
+ FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
+ AVFILTER_FLAG_SLICE_THREADS,
+ .process_command = ff_filter_process_command,
+};
@@ -77,6 +77,7 @@ extern const AVFilter ff_af_apulsator;
extern const AVFilter ff_af_arealtime;
extern const AVFilter ff_af_aresample;
extern const AVFilter ff_af_areverse;
+extern const AVFilter ff_af_arls;
extern const AVFilter ff_af_arnndn;
extern const AVFilter ff_af_asdr;
extern const AVFilter ff_af_asegment;
--
2.39.1