216 lines
6.3 KiB
C
216 lines
6.3 KiB
C
/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#undef ZERO
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#undef ONE
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#undef ftype
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#undef SAMPLE_FORMAT
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#if DEPTH == 32
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#define SAMPLE_FORMAT float
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#define ftype float
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#define ONE 1.f
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#define ZERO 0.f
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#else
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#define SAMPLE_FORMAT double
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#define ftype double
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#define ONE 1.0
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#define ZERO 0.0
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#endif
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#define fn3(a,b) a##_##b
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#define fn2(a,b) fn3(a,b)
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#define fn(a) fn2(a, SAMPLE_FORMAT)
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static ftype fn(fir_sample)(AudioAPContext *s, ftype sample, ftype *delay,
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ftype *coeffs, ftype *tmp, int *offset)
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{
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const int order = s->order;
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ftype output;
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delay[*offset] = sample;
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memcpy(tmp, coeffs + order - *offset, order * sizeof(ftype));
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#if DEPTH == 32
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output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size);
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#else
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output = s->fdsp->scalarproduct_double(delay, tmp, s->kernel_size);
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#endif
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if (--(*offset) < 0)
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*offset = order - 1;
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return output;
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}
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static int fn(lup_decompose)(ftype **MA, const int N, const ftype tol, int *P)
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{
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for (int i = 0; i <= N; i++)
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P[i] = i;
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for (int i = 0; i < N; i++) {
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ftype maxA = ZERO;
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int imax = i;
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for (int k = i; k < N; k++) {
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ftype absA = fabs(MA[k][i]);
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if (absA > maxA) {
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maxA = absA;
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imax = k;
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}
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}
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if (maxA < tol)
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return 0;
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if (imax != i) {
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FFSWAP(int, P[i], P[imax]);
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FFSWAP(ftype *, MA[i], MA[imax]);
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P[N]++;
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}
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for (int j = i + 1; j < N; j++) {
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MA[j][i] /= MA[i][i];
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for (int k = i + 1; k < N; k++)
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MA[j][k] -= MA[j][i] * MA[i][k];
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}
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}
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return 1;
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}
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static void fn(lup_invert)(ftype *const *MA, const int *P, const int N, ftype **IA)
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{
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for (int j = 0; j < N; j++) {
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for (int i = 0; i < N; i++) {
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IA[i][j] = P[i] == j ? ONE : ZERO;
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for (int k = 0; k < i; k++)
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IA[i][j] -= MA[i][k] * IA[k][j];
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}
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for (int i = N - 1; i >= 0; i--) {
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for (int k = i + 1; k < N; k++)
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IA[i][j] -= MA[i][k] * IA[k][j];
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IA[i][j] /= MA[i][i];
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}
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}
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}
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static ftype fn(process_sample)(AudioAPContext *s, ftype input, ftype desired, int ch)
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{
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ftype *dcoeffs = (ftype *)s->dcoeffs->extended_data[ch];
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ftype *coeffs = (ftype *)s->coeffs->extended_data[ch];
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ftype *delay = (ftype *)s->delay->extended_data[ch];
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ftype **itmpmp = (ftype **)&s->itmpmp[s->projection * ch];
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ftype **tmpmp = (ftype **)&s->tmpmp[s->projection * ch];
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ftype *tmpm = (ftype *)s->tmpm->extended_data[ch];
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ftype *tmp = (ftype *)s->tmp->extended_data[ch];
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ftype *e = (ftype *)s->e->extended_data[ch];
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ftype *x = (ftype *)s->x->extended_data[ch];
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ftype *w = (ftype *)s->w->extended_data[ch];
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int *p = (int *)s->p->extended_data[ch];
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int *offset = (int *)s->offset->extended_data[ch];
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const int projection = s->projection;
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const ftype delta = s->delta;
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const int order = s->order;
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const int length = projection + order;
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const ftype mu = s->mu;
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const ftype tol = 0.00001f;
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ftype output;
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x[offset[2] + length] = x[offset[2]] = input;
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delay[offset[0] + order] = input;
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output = fn(fir_sample)(s, input, delay, coeffs, tmp, offset);
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e[offset[1]] = e[offset[1] + projection] = desired - output;
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for (int i = 0; i < projection; i++) {
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const int iprojection = i * projection;
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for (int j = i; j < projection; j++) {
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ftype sum = ZERO;
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for (int k = 0; k < order; k++)
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sum += x[offset[2] + i + k] * x[offset[2] + j + k];
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tmpm[iprojection + j] = sum;
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if (i != j)
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tmpm[j * projection + i] = sum;
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}
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tmpm[iprojection + i] += delta;
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}
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fn(lup_decompose)(tmpmp, projection, tol, p);
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fn(lup_invert)(tmpmp, p, projection, itmpmp);
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for (int i = 0; i < projection; i++) {
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ftype sum = ZERO;
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for (int j = 0; j < projection; j++)
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sum += itmpmp[i][j] * e[j + offset[1]];
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w[i] = sum;
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}
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for (int i = 0; i < order; i++) {
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ftype sum = ZERO;
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for (int j = 0; j < projection; j++)
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sum += x[offset[2] + i + j] * w[j];
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dcoeffs[i] = sum;
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}
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for (int i = 0; i < order; i++)
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coeffs[i] = coeffs[i + order] = coeffs[i] + mu * dcoeffs[i];
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if (--offset[1] < 0)
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offset[1] = projection - 1;
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if (--offset[2] < 0)
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offset[2] = length - 1;
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switch (s->output_mode) {
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case IN_MODE: output = input; break;
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case DESIRED_MODE: output = desired; break;
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case OUT_MODE: output = desired - output; break;
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case NOISE_MODE: output = input - output; break;
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case ERROR_MODE: break;
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}
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return output;
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}
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static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
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{
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AudioAPContext *s = ctx->priv;
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AVFrame *out = arg;
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const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
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const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
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for (int c = start; c < end; c++) {
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const ftype *input = (const ftype *)s->frame[0]->extended_data[c];
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const ftype *desired = (const ftype *)s->frame[1]->extended_data[c];
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ftype *output = (ftype *)out->extended_data[c];
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for (int n = 0; n < out->nb_samples; n++) {
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output[n] = fn(process_sample)(s, input[n], desired[n], c);
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if (ctx->is_disabled)
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output[n] = input[n];
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}
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}
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return 0;
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}
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