avfilter: add Affine Projection adaptive audio filter

Changelog

@@ -6,6 +6,7 @@ version <next>:
 - EVC decoding using external library libxevd
 - EVC encoding using external library libxeve
 - QOA decoder and demuxer
+- aap filter
 
 version 6.1:
 - libaribcaption decoder

doc/filters.texi

@@ -418,6 +418,63 @@ 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
+
+@item precision
+Set which precision to use when processing samples.
+
+@table @option
+@item auto
+Auto pick internal sample format depending on other filters.
+
+@item float
+Always use single-floating point precision sample format.
+
+@item double
+Always use double-floating point precision sample format.
+@end table
+@end table
+
 @section acompressor
 
 A compressor is mainly used to reduce the dynamic range of a signal.

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

libavfilter/aap_template.c

+/*
+ * 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
+ */
+
+#undef ZERO
+#undef ONE
+#undef ftype
+#undef SAMPLE_FORMAT
+#if DEPTH == 32
+#define SAMPLE_FORMAT float
+#define ftype float
+#define ONE 1.f
+#define ZERO 0.f
+#else
+#define SAMPLE_FORMAT double
+#define ftype double
+#define ONE 1.0
+#define ZERO 0.0
+#endif
+
+#define fn3(a,b)   a##_##b
+#define fn2(a,b)   fn3(a,b)
+#define fn(a)      fn2(a, SAMPLE_FORMAT)
+
+#if DEPTH == 64
+static double scalarproduct_double(const double *v1, const double *v2, int len)
+{
+    double p = 0.0;
+
+    for (int i = 0; i < len; i++)
+        p += v1[i] * v2[i];
+
+    return p;
+}
+#endif
+
+static ftype fn(fir_sample)(AudioAPContext *s, ftype sample, ftype *delay,
+                            ftype *coeffs, ftype *tmp, int *offset)
+{
+    const int order = s->order;
+    ftype output;
+
+    delay[*offset] = sample;
+
+    memcpy(tmp, coeffs + order - *offset, order * sizeof(ftype));
+#if DEPTH == 32
+    output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
+#else
+    output = scalarproduct_double(delay, tmp, s->kernel_size);
+#endif
+
+    if (--(*offset) < 0)
+        *offset = order - 1;
+
+    return output;
+}
+
+static int fn(lup_decompose)(ftype **MA, const int N, const ftype tol, int *P)
+{
+    for (int i = 0; i <= N; i++)
+        P[i] = i;
+
+    for (int i = 0; i < N; i++) {
+        ftype maxA = ZERO;
+        int imax = i;
+
+        for (int k = i; k < N; k++) {
+            ftype 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(ftype *, 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 fn(lup_invert)(ftype *const *MA, const int *P, const int N, ftype **IA)
+{
+    for (int j = 0; j < N; j++) {
+        for (int i = 0; i < N; i++) {
+            IA[i][j] = P[i] == j ? ONE : ZERO;
+
+            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 ftype fn(process_sample)(AudioAPContext *s, ftype input, ftype desired, int ch)
+{
+    ftype *dcoeffs = (ftype *)s->dcoeffs->extended_data[ch];
+    ftype *coeffs = (ftype *)s->coeffs->extended_data[ch];
+    ftype *delay = (ftype *)s->delay->extended_data[ch];
+    ftype **itmpmp = (ftype **)&s->itmpmp[s->projection * ch];
+    ftype **tmpmp = (ftype **)&s->tmpmp[s->projection * ch];
+    ftype *tmpm = (ftype *)s->tmpm->extended_data[ch];
+    ftype *tmp = (ftype *)s->tmp->extended_data[ch];
+    ftype *e = (ftype *)s->e->extended_data[ch];
+    ftype *x = (ftype *)s->x->extended_data[ch];
+    ftype *w = (ftype *)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 ftype delta = s->delta;
+    const int order = s->order;
+    const int length = projection + order;
+    const ftype mu = s->mu;
+    const ftype tol = 0.00001f;
+    ftype output;
+
+    x[offset[2] + length] = x[offset[2]] = input;
+    delay[offset[0] + order] = input;
+
+    output = fn(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++) {
+            ftype sum = ZERO;
+            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;
+    }
+
+    fn(lup_decompose)(tmpmp, projection, tol, p);
+    fn(lup_invert)(tmpmp, p, projection, itmpmp);
+
+    for (int i = 0; i < projection; i++) {
+        ftype sum = ZERO;
+        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++) {
+        ftype sum = ZERO;
+        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 fn(filter_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 ftype *input = (const ftype *)s->frame[0]->extended_data[c];
+        const ftype *desired = (const ftype *)s->frame[1]->extended_data[c];
+        ftype *output = (ftype *)out->extended_data[c];
+
+        for (int n = 0; n < out->nb_samples; n++) {
+            output[n] = fn(process_sample)(s, input[n], desired[n], c);
+            if (ctx->is_disabled)
+                output[n] = input[n];
+        }
+    }
+
+    return 0;
+}

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;

libavfilter/version.h

@@ -31,7 +31,7 @@
 
 #include "version_major.h"
 
-#define LIBAVFILTER_VERSION_MINOR  13
+#define LIBAVFILTER_VERSION_MINOR  14
 #define LIBAVFILTER_VERSION_MICRO 100