[FFmpeg-devel] [PATCH] avfilter: add Affine Projection adaptive audio filter

[Paul B Mahol <onemda@gmail.com>]

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From 6c355f79e9c21a11e5e1266da7936a4ac2dc07ac Mon Sep 17 00:00:00 2001
From: Paul B Mahol <onemda@gmail.com>
Date: Sun, 30 Apr 2023 17:06:00 +0200
Subject: [PATCH] avfilter: add Affine Projection adaptive audio filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 doc/filters.texi         |  43 ++++
 libavfilter/Makefile     |   1 +
 libavfilter/af_aap.c     | 451 +++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c |   1 +
 4 files changed, 496 insertions(+)
 create mode 100644 libavfilter/af_aap.c

diff --git a/doc/filters.texi b/doc/filters.texi
index 5268b2003c..3c9d32aa76 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -418,6 +418,49 @@ build.
 
 Below is a description of the currently available audio filters.
 
+@section aap
+Apply Affine Projection algorithm to the first audio stream using the second audio stream.
+
+This adaptive filter is used to estimate unknown audio based on multiple input audio samples.
+Affine projection algorithm can make trade-offs between computation complexity with convergence speed.
+
+A description of the accepted options follows.
+
+@table @option
+@item order
+Set the filter order.
+
+@item projection
+Set the projection order.
+
+@item mu
+Set the filter mu.
+
+@item delta
+Set the coefficient to initialize internal covariance matrix.
+
+@item out_mode
+Set the filter output samples. It accepts the following values:
+@table @option
+@item i
+Pass the 1st input.
+
+@item d
+Pass the 2nd input.
+
+@item o
+Pass difference between desired, 2nd input and error signal estimate.
+
+@item n
+Pass difference between input, 1st input and error signal estimate.
+
+@item e
+Pass error signal estimated samples.
+
+Default value is @var{o}.
+@end table
+@end table
+
 @section acompressor
 
 A compressor is mainly used to reduce the dynamic range of a signal.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 5e5068b564..0da62540f8 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -35,6 +35,7 @@ OBJS-$(CONFIG_DNN)                           += dnn_filter_common.o
 include $(SRC_PATH)/libavfilter/dnn/Makefile
 
 # audio filters
+OBJS-$(CONFIG_AAP_FILTER)                    += af_aap.o
 OBJS-$(CONFIG_ABENCH_FILTER)                 += f_bench.o
 OBJS-$(CONFIG_ACOMPRESSOR_FILTER)            += af_sidechaincompress.o
 OBJS-$(CONFIG_ACONTRAST_FILTER)              += af_acontrast.o
diff --git a/libavfilter/af_aap.c b/libavfilter/af_aap.c
new file mode 100644
index 0000000000..978a853137
--- /dev/null
+++ b/libavfilter/af_aap.c
@@ -0,0 +1,451 @@
+/*
+ * 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/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"
+
+enum OutModes {
+    IN_MODE,
+    DESIRED_MODE,
+    OUT_MODE,
+    NOISE_MODE,
+    ERROR_MODE,
+    NB_OMODES
+};
+
+typedef struct AudioAPContext {
+    const AVClass *class;
+
+    int order;
+    int projection;
+    float mu;
+    float delta;
+    int output_mode;
+
+    int kernel_size;
+    AVFrame *offset;
+    AVFrame *delay;
+    AVFrame *coeffs;
+    AVFrame *e;
+    AVFrame *p;
+    AVFrame *x;
+    AVFrame *w;
+    AVFrame *dcoeffs;
+    AVFrame *tmp;
+    AVFrame *tmpm;
+    AVFrame *itmpm;
+
+    float **tmpmp;
+    float **itmpmp;
+
+    AVFrame *frame[2];
+
+    AVFloatDSPContext *fdsp;
+} AudioAPContext;
+
+#define OFFSET(x) offsetof(AudioAPContext, 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 aap_options[] = {
+    { "order",      "set the filter order",      OFFSET(order),       AV_OPT_TYPE_INT,   {.i64=16},   1, INT16_MAX, A },
+    { "projection", "set the filter projection", OFFSET(projection),  AV_OPT_TYPE_INT,   {.i64=2},    1, 256, A },
+    { "mu",         "set the filter mu",         OFFSET(mu),          AV_OPT_TYPE_FLOAT, {.dbl=0.0001},0,1, AT },
+    { "delta",      "set the filter delta",      OFFSET(delta),       AV_OPT_TYPE_FLOAT, {.dbl=0.001},0, 1, AT },
+    { "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" },
+    {  "e", "error",   0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE},   0, 0, AT, "mode" },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aap);
+
+static float fir_sample(AudioAPContext *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 int lup_decompose(float **MA, const int N, const float tol, int *P)
+{
+    for (int i = 0; i <= N; i++)
+        P[i] = i;
+
+    for (int i = 0; i < N; i++) {
+        float maxA = 0.f;
+        int imax = i;
+
+        for (int k = i; k < N; k++) {
+            float absA = fabs(MA[k][i]);
+            if (absA > maxA) {
+                maxA = absA;
+                imax = k;
+            }
+        }
+
+        if (maxA < tol)
+            return 0;
+
+        if (imax != i) {
+            FFSWAP(int, P[i], P[imax]);
+            FFSWAP(float *, MA[i], MA[imax]);
+            P[N]++;
+        }
+
+        for (int j = i + 1; j < N; j++) {
+            MA[j][i] /= MA[i][i];
+
+            for (int k = i + 1; k < N; k++)
+                MA[j][k] -= MA[j][i] * MA[i][k];
+        }
+    }
+
+    return 1;
+}
+
+static void lup_invert(float *const *MA, const int *P, const int N, float **IA)
+{
+    for (int j = 0; j < N; j++) {
+        for (int i = 0; i < N; i++) {
+            IA[i][j] = P[i] == j ? 1.f : 0.f;
+
+            for (int k = 0; k < i; k++)
+                IA[i][j] -= MA[i][k] * IA[k][j];
+        }
+
+        for (int i = N - 1; i >= 0; i--) {
+            for (int k = i + 1; k < N; k++)
+                IA[i][j] -= MA[i][k] * IA[k][j];
+
+            IA[i][j] /= MA[i][i];
+        }
+    }
+}
+
+static float process_sample(AudioAPContext *s, float input, float desired, int ch)
+{
+    float *dcoeffs = (float *)s->dcoeffs->extended_data[ch];
+    float *coeffs = (float *)s->coeffs->extended_data[ch];
+    float *delay = (float *)s->delay->extended_data[ch];
+    float **itmpmp = &s->itmpmp[s->projection * ch];
+    float **tmpmp = &s->tmpmp[s->projection * ch];
+    float *tmpm = (float *)s->tmpm->extended_data[ch];
+    float *tmp = (float *)s->tmp->extended_data[ch];
+    float *e = (float *)s->e->extended_data[ch];
+    float *x = (float *)s->x->extended_data[ch];
+    float *w = (float *)s->w->extended_data[ch];
+    int *p = (int *)s->p->extended_data[ch];
+    int *offset = (int *)s->offset->extended_data[ch];
+    const int projection = s->projection;
+    const float delta = s->delta;
+    const int order = s->order;
+    const int length = projection + order;
+    const float mu = s->mu;
+    const float tol = 0.00001f;
+    float output;
+
+    x[offset[2] + length] = x[offset[2]] = input;
+    delay[offset[0] + order] = input;
+
+    output = fir_sample(s, input, delay, coeffs, tmp, offset);
+    e[offset[1]] = e[offset[1] + projection] = desired - output;
+
+    for (int i = 0; i < projection; i++) {
+        const int iprojection = i * projection;
+
+        for (int j = i; j < projection; j++) {
+            float sum = 0.f;
+            for (int k = 0; k < order; k++)
+                sum += x[offset[2] + i + k] * x[offset[2] + j + k];
+            tmpm[iprojection + j] = sum;
+            if (i != j)
+                tmpm[j * projection + i] = sum;
+        }
+
+        tmpm[iprojection + i] += delta;
+    }
+
+    lup_decompose(tmpmp, projection, tol, p);
+    lup_invert(tmpmp, p, projection, itmpmp);
+
+    for (int i = 0; i < projection; i++) {
+        float sum = 0.f;
+        for (int j = 0; j < projection; j++)
+            sum += itmpmp[i][j] * e[j + offset[1]];
+        w[i] = sum;
+    }
+
+    for (int i = 0; i < order; i++) {
+        float sum = 0.f;
+        for (int j = 0; j < projection; j++)
+            sum += x[offset[2] + i + j] * w[j];
+        dcoeffs[i] = sum;
+    }
+
+    for (int i = 0; i < order; i++)
+        coeffs[i] = coeffs[i + order] = coeffs[i] + mu * dcoeffs[i];
+
+    if (--offset[1] < 0)
+        offset[1] = projection - 1;
+
+    if (--offset[2] < 0)
+        offset[2] = length - 1;
+
+    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;
+    case ERROR_MODE:                              break;
+    }
+    return output;
+}
+
+static int process_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+    AudioAPContext *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)
+{
+    AudioAPContext *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)
+{
+    const int channels = outlink->ch_layout.nb_channels;
+    AVFilterContext *ctx = outlink->src;
+    AudioAPContext *s = ctx->priv;
+
+    s->kernel_size = FFALIGN(s->order, 16);
+
+    if (!s->offset)
+        s->offset = ff_get_audio_buffer(outlink, 3);
+    if (!s->delay)
+        s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
+    if (!s->dcoeffs)
+        s->dcoeffs = ff_get_audio_buffer(outlink, s->kernel_size);
+    if (!s->coeffs)
+        s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
+    if (!s->e)
+        s->e = ff_get_audio_buffer(outlink, 2 * s->projection);
+    if (!s->p)
+        s->p = ff_get_audio_buffer(outlink, s->projection + 1);
+    if (!s->x)
+        s->x = ff_get_audio_buffer(outlink, 2 * (s->projection + s->order));
+    if (!s->w)
+        s->w = ff_get_audio_buffer(outlink, s->projection);
+    if (!s->tmp)
+        s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
+    if (!s->tmpm)
+        s->tmpm = ff_get_audio_buffer(outlink, s->projection * s->projection);
+    if (!s->itmpm)
+        s->itmpm = ff_get_audio_buffer(outlink, s->projection * s->projection);
+
+    if (!s->tmpmp)
+        s->tmpmp = av_calloc(s->projection * channels, sizeof(*s->tmpmp));
+    if (!s->itmpmp)
+        s->itmpmp = av_calloc(s->projection * channels, sizeof(*s->itmpmp));
+
+    if (!s->offset || !s->delay || !s->dcoeffs || !s->coeffs || !s->tmpmp || !s->itmpmp ||
+        !s->e || !s->p || !s->x || !s->w || !s->tmp || !s->tmpm || !s->itmpm)
+        return AVERROR(ENOMEM);
+
+    for (int ch = 0; ch < channels; ch++) {
+        float *itmpm = (float *)s->itmpm->extended_data[ch];
+        float *tmpm = (float *)s->tmpm->extended_data[ch];
+        float **itmpmp = &s->itmpmp[s->projection * ch];
+        float **tmpmp = &s->tmpmp[s->projection * ch];
+
+        for (int i = 0; i < s->projection; i++) {
+            itmpmp[i] = &itmpm[i * s->projection];
+            tmpmp[i] = &tmpm[i * s->projection];
+        }
+    }
+
+    return 0;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    AudioAPContext *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)
+{
+    AudioAPContext *s = ctx->priv;
+
+    av_freep(&s->fdsp);
+
+    av_frame_free(&s->offset);
+    av_frame_free(&s->delay);
+    av_frame_free(&s->dcoeffs);
+    av_frame_free(&s->coeffs);
+    av_frame_free(&s->e);
+    av_frame_free(&s->p);
+    av_frame_free(&s->w);
+    av_frame_free(&s->x);
+    av_frame_free(&s->tmp);
+    av_frame_free(&s->tmpm);
+    av_frame_free(&s->itmpm);
+
+    av_freep(&s->tmpmp);
+    av_freep(&s->itmpmp);
+}
+
+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_aap = {
+    .name           = "aap",
+    .description    = NULL_IF_CONFIG_SMALL("Apply Affine Projection algorithm to first audio stream."),
+    .priv_size      = sizeof(AudioAPContext),
+    .priv_class     = &aap_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,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 5fbfe9d906..9d66213a62 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -21,6 +21,7 @@
 
 #include "avfilter.h"
 
+extern const AVFilter ff_af_aap;
 extern const AVFilter ff_af_abench;
 extern const AVFilter ff_af_acompressor;
 extern const AVFilter ff_af_acontrast;
-- 
2.42.1


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