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/*
* 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;
}
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