142 lines
4.2 KiB
C
142 lines
4.2 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 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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#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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#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)(AudioNLMSContext *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 ftype fn(process_sample)(AudioNLMSContext *s, ftype input, ftype desired,
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ftype *delay, ftype *coeffs, ftype *tmp, int *offsetp)
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{
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const int order = s->order;
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const ftype leakage = s->leakage;
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const ftype mu = s->mu;
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const ftype a = ONE - leakage;
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ftype sum, output, e, norm, b;
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int offset = *offsetp;
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delay[offset + order] = input;
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output = fn(fir_sample)(s, input, delay, coeffs, tmp, offsetp);
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e = desired - output;
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#if DEPTH == 32
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sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size);
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#else
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sum = scalarproduct_double(delay, delay, s->kernel_size);
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#endif
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norm = s->eps + sum;
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b = mu * e / norm;
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if (s->anlmf)
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b *= e * e;
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memcpy(tmp, delay + offset, order * sizeof(ftype));
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#if DEPTH == 32
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s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size);
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s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size);
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#else
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s->fdsp->vector_dmul_scalar(coeffs, coeffs, a, s->kernel_size);
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s->fdsp->vector_dmac_scalar(coeffs, tmp, b, s->kernel_size);
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#endif
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memcpy(coeffs + order, coeffs, order * sizeof(ftype));
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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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AudioNLMSContext *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 *delay = (ftype *)s->delay->extended_data[c];
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ftype *coeffs = (ftype *)s->coeffs->extended_data[c];
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ftype *tmp = (ftype *)s->tmp->extended_data[c];
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int *offset = (int *)s->offset->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], delay, coeffs, tmp, offset);
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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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