mirror of https://github.com/FFmpeg/FFmpeg.git
Paul B Mahol
2 years ago
parent
cc86343b96
commit
f66536cc58
7 changed files with 620 additions and 1 deletions
@ -0,0 +1,227 @@ |
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/*
|
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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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#if DEPTH == 64 |
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static double scalarproduct_double(const double *v1, const double *v2, int len) |
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{ |
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double p = 0.0; |
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for (int i = 0; i < len; i++) |
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p += v1[i] * v2[i]; |
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return p; |
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} |
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#endif |
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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 = 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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@ -0,0 +1,332 @@ |
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/*
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* Copyright (c) 2023 Paul B Mahol |
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* |
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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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#include "libavutil/channel_layout.h" |
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#include "libavutil/common.h" |
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#include "libavutil/float_dsp.h" |
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#include "libavutil/opt.h" |
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#include "audio.h" |
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#include "avfilter.h" |
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#include "formats.h" |
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#include "filters.h" |
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#include "internal.h" |
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enum OutModes { |
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IN_MODE, |
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DESIRED_MODE, |
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OUT_MODE, |
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NOISE_MODE, |
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ERROR_MODE, |
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NB_OMODES |
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}; |
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typedef struct AudioAPContext { |
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const AVClass *class; |
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int order; |
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int projection; |
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float mu; |
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float delta; |
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int output_mode; |
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int precision; |
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int kernel_size; |
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AVFrame *offset; |
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AVFrame *delay; |
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AVFrame *coeffs; |
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AVFrame *e; |
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AVFrame *p; |
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AVFrame *x; |
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AVFrame *w; |
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AVFrame *dcoeffs; |
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AVFrame *tmp; |
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AVFrame *tmpm; |
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AVFrame *itmpm; |
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void **tmpmp; |
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void **itmpmp; |
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AVFrame *frame[2]; |
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int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs); |
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AVFloatDSPContext *fdsp; |
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} AudioAPContext; |
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#define OFFSET(x) offsetof(AudioAPContext, x) |
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
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#define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM |
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static const AVOption aap_options[] = { |
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{ "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=16}, 1, INT16_MAX, A }, |
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{ "projection", "set the filter projection", OFFSET(projection), AV_OPT_TYPE_INT, {.i64=2}, 1, 256, A }, |
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{ "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.0001},0,1, AT }, |
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{ "delta", "set the filter delta", OFFSET(delta), AV_OPT_TYPE_FLOAT, {.dbl=0.001},0, 1, AT }, |
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{ "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, "mode" }, |
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{ "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, "mode" }, |
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{ "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, "mode" }, |
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{ "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, "mode" }, |
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{ "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, "mode" }, |
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{ "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, "mode" }, |
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{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "precision" }, |
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{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "precision" }, |
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{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "precision" }, |
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{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "precision" }, |
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{ NULL } |
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}; |
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AVFILTER_DEFINE_CLASS(aap); |
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static int query_formats(AVFilterContext *ctx) |
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{ |
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AudioAPContext *s = ctx->priv; |
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static const enum AVSampleFormat sample_fmts[3][3] = { |
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }, |
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{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }, |
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{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE }, |
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}; |
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int ret; |
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if ((ret = ff_set_common_all_channel_counts(ctx)) < 0) |
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return ret; |
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if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0) |
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return ret; |
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return ff_set_common_all_samplerates(ctx); |
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} |
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static int activate(AVFilterContext *ctx) |
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{ |
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AudioAPContext *s = ctx->priv; |
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int i, ret, status; |
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int nb_samples; |
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int64_t pts; |
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FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx); |
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nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), |
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ff_inlink_queued_samples(ctx->inputs[1])); |
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for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) { |
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if (s->frame[i]) |
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continue; |
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if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) { |
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ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]); |
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if (ret < 0) |
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return ret; |
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} |
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} |
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if (s->frame[0] && s->frame[1]) { |
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AVFrame *out; |
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out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples); |
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if (!out) { |
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av_frame_free(&s->frame[0]); |
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av_frame_free(&s->frame[1]); |
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return AVERROR(ENOMEM); |
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} |
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ff_filter_execute(ctx, s->filter_channels, out, NULL, |
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FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx))); |
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out->pts = s->frame[0]->pts; |
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out->duration = s->frame[0]->duration; |
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av_frame_free(&s->frame[0]); |
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av_frame_free(&s->frame[1]); |
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ret = ff_filter_frame(ctx->outputs[0], out); |
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if (ret < 0) |
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return ret; |
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} |
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if (!nb_samples) { |
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for (i = 0; i < 2; i++) { |
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if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) { |
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ff_outlink_set_status(ctx->outputs[0], status, pts); |
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return 0; |
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} |
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} |
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} |
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if (ff_outlink_frame_wanted(ctx->outputs[0])) { |
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for (i = 0; i < 2; i++) { |
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if (s->frame[i] || ff_inlink_queued_samples(ctx->inputs[i]) > 0) |
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continue; |
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ff_inlink_request_frame(ctx->inputs[i]); |
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return 0; |
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} |
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} |
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return 0; |
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} |
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#define DEPTH 32 |
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#include "aap_template.c" |
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#undef DEPTH |
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#define DEPTH 64 |
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#include "aap_template.c" |
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static int config_output(AVFilterLink *outlink) |
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{ |
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const int channels = outlink->ch_layout.nb_channels; |
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AVFilterContext *ctx = outlink->src; |
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AudioAPContext *s = ctx->priv; |
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s->kernel_size = FFALIGN(s->order, 16); |
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if (!s->offset) |
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s->offset = ff_get_audio_buffer(outlink, 3); |
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if (!s->delay) |
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s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size); |
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if (!s->dcoeffs) |
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s->dcoeffs = ff_get_audio_buffer(outlink, s->kernel_size); |
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if (!s->coeffs) |
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s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size); |
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if (!s->e) |
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s->e = ff_get_audio_buffer(outlink, 2 * s->projection); |
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if (!s->p) |
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s->p = ff_get_audio_buffer(outlink, s->projection + 1); |
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if (!s->x) |
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s->x = ff_get_audio_buffer(outlink, 2 * (s->projection + s->order)); |
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if (!s->w) |
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s->w = ff_get_audio_buffer(outlink, s->projection); |
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if (!s->tmp) |
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s->tmp = ff_get_audio_buffer(outlink, s->kernel_size); |
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if (!s->tmpm) |
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s->tmpm = ff_get_audio_buffer(outlink, s->projection * s->projection); |
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if (!s->itmpm) |
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s->itmpm = ff_get_audio_buffer(outlink, s->projection * s->projection); |
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if (!s->tmpmp) |
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s->tmpmp = av_calloc(s->projection * channels, sizeof(*s->tmpmp)); |
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if (!s->itmpmp) |
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s->itmpmp = av_calloc(s->projection * channels, sizeof(*s->itmpmp)); |
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if (!s->offset || !s->delay || !s->dcoeffs || !s->coeffs || !s->tmpmp || !s->itmpmp || |
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!s->e || !s->p || !s->x || !s->w || !s->tmp || !s->tmpm || !s->itmpm) |
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return AVERROR(ENOMEM); |
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switch (outlink->format) { |
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case AV_SAMPLE_FMT_DBLP: |
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for (int ch = 0; ch < channels; ch++) { |
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double *itmpm = (double *)s->itmpm->extended_data[ch]; |
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double *tmpm = (double *)s->tmpm->extended_data[ch]; |
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double **itmpmp = (double **)&s->itmpmp[s->projection * ch]; |
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double **tmpmp = (double **)&s->tmpmp[s->projection * ch]; |
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for (int i = 0; i < s->projection; i++) { |
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itmpmp[i] = &itmpm[i * s->projection]; |
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tmpmp[i] = &tmpm[i * s->projection]; |
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} |
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} |
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s->filter_channels = filter_channels_double; |
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break; |
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case AV_SAMPLE_FMT_FLTP: |
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for (int ch = 0; ch < channels; ch++) { |
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float *itmpm = (float *)s->itmpm->extended_data[ch]; |
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float *tmpm = (float *)s->tmpm->extended_data[ch]; |
||||
float **itmpmp = (float **)&s->itmpmp[s->projection * ch]; |
||||
float **tmpmp = (float **)&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]; |
||||
} |
||||
} |
||||
|
||||
s->filter_channels = filter_channels_float; |
||||
break; |
||||
} |
||||
|
||||
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_QUERY_FUNC(query_formats), |
||||
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL | |
||||
AVFILTER_FLAG_SLICE_THREADS, |
||||
.process_command = ff_filter_process_command, |
||||
}; |
||||
Loading…
Reference in new issue