/* * 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; }