| Line | Branch | Exec | Source |
|---|---|---|---|
| 1 | /* | ||
| 2 | * This file is part of FFmpeg. | ||
| 3 | * | ||
| 4 | * FFmpeg is free software; you can redistribute it and/or | ||
| 5 | * modify it under the terms of the GNU Lesser General Public | ||
| 6 | * License as published by the Free Software Foundation; either | ||
| 7 | * version 2.1 of the License, or (at your option) any later version. | ||
| 8 | * | ||
| 9 | * FFmpeg is distributed in the hope that it will be useful, | ||
| 10 | * but WITHOUT ANY WARRANTY; without even the implied warranty of | ||
| 11 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | ||
| 12 | * Lesser General Public License for more details. | ||
| 13 | * | ||
| 14 | * You should have received a copy of the GNU Lesser General Public | ||
| 15 | * License along with FFmpeg; if not, write to the Free Software | ||
| 16 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | ||
| 17 | */ | ||
| 18 | |||
| 19 | #undef ONE | ||
| 20 | #undef ftype | ||
| 21 | #undef SAMPLE_FORMAT | ||
| 22 | #if DEPTH == 32 | ||
| 23 | #define SAMPLE_FORMAT float | ||
| 24 | #define ftype float | ||
| 25 | #define ONE 1.f | ||
| 26 | #else | ||
| 27 | #define SAMPLE_FORMAT double | ||
| 28 | #define ftype double | ||
| 29 | #define ONE 1.0 | ||
| 30 | #endif | ||
| 31 | |||
| 32 | #define fn3(a,b) a##_##b | ||
| 33 | #define fn2(a,b) fn3(a,b) | ||
| 34 | #define fn(a) fn2(a, SAMPLE_FORMAT) | ||
| 35 | |||
| 36 | #if DEPTH == 64 | ||
| 37 | ✗ | static double scalarproduct_double(const double *v1, const double *v2, int len) | |
| 38 | { | ||
| 39 | ✗ | double p = 0.0; | |
| 40 | |||
| 41 | ✗ | for (int i = 0; i < len; i++) | |
| 42 | ✗ | p += v1[i] * v2[i]; | |
| 43 | |||
| 44 | ✗ | return p; | |
| 45 | } | ||
| 46 | #endif | ||
| 47 | |||
| 48 | ✗ | static ftype fn(fir_sample)(AudioNLMSContext *s, ftype sample, ftype *delay, | |
| 49 | ftype *coeffs, ftype *tmp, int *offset) | ||
| 50 | { | ||
| 51 | ✗ | const int order = s->order; | |
| 52 | ftype output; | ||
| 53 | |||
| 54 | ✗ | delay[*offset] = sample; | |
| 55 | |||
| 56 | ✗ | memcpy(tmp, coeffs + order - *offset, order * sizeof(ftype)); | |
| 57 | |||
| 58 | #if DEPTH == 32 | ||
| 59 | ✗ | output = s->fdsp->scalarproduct_float(delay, tmp, s->kernel_size); | |
| 60 | #else | ||
| 61 | ✗ | output = scalarproduct_double(delay, tmp, s->kernel_size); | |
| 62 | #endif | ||
| 63 | |||
| 64 | ✗ | if (--(*offset) < 0) | |
| 65 | ✗ | *offset = order - 1; | |
| 66 | |||
| 67 | ✗ | return output; | |
| 68 | } | ||
| 69 | |||
| 70 | ✗ | static ftype fn(process_sample)(AudioNLMSContext *s, ftype input, ftype desired, | |
| 71 | ftype *delay, ftype *coeffs, ftype *tmp, int *offsetp) | ||
| 72 | { | ||
| 73 | ✗ | const int order = s->order; | |
| 74 | ✗ | const ftype leakage = s->leakage; | |
| 75 | ✗ | const ftype mu = s->mu; | |
| 76 | ✗ | const ftype a = ONE - leakage; | |
| 77 | ftype sum, output, e, norm, b; | ||
| 78 | ✗ | int offset = *offsetp; | |
| 79 | |||
| 80 | ✗ | delay[offset + order] = input; | |
| 81 | |||
| 82 | ✗ | output = fn(fir_sample)(s, input, delay, coeffs, tmp, offsetp); | |
| 83 | ✗ | e = desired - output; | |
| 84 | |||
| 85 | #if DEPTH == 32 | ||
| 86 | ✗ | sum = s->fdsp->scalarproduct_float(delay, delay, s->kernel_size); | |
| 87 | #else | ||
| 88 | ✗ | sum = scalarproduct_double(delay, delay, s->kernel_size); | |
| 89 | #endif | ||
| 90 | ✗ | norm = s->eps + sum; | |
| 91 | ✗ | b = mu * e / norm; | |
| 92 | ✗ | if (s->anlmf) | |
| 93 | ✗ | b *= e * e; | |
| 94 | |||
| 95 | ✗ | memcpy(tmp, delay + offset, order * sizeof(ftype)); | |
| 96 | |||
| 97 | #if DEPTH == 32 | ||
| 98 | ✗ | s->fdsp->vector_fmul_scalar(coeffs, coeffs, a, s->kernel_size); | |
| 99 | ✗ | s->fdsp->vector_fmac_scalar(coeffs, tmp, b, s->kernel_size); | |
| 100 | #else | ||
| 101 | ✗ | s->fdsp->vector_dmul_scalar(coeffs, coeffs, a, s->kernel_size); | |
| 102 | ✗ | s->fdsp->vector_dmac_scalar(coeffs, tmp, b, s->kernel_size); | |
| 103 | #endif | ||
| 104 | |||
| 105 | ✗ | memcpy(coeffs + order, coeffs, order * sizeof(ftype)); | |
| 106 | |||
| 107 | ✗ | switch (s->output_mode) { | |
| 108 | ✗ | case IN_MODE: output = input; break; | |
| 109 | ✗ | case DESIRED_MODE: output = desired; break; | |
| 110 | ✗ | case OUT_MODE: output = desired - output; break; | |
| 111 | ✗ | case NOISE_MODE: output = input - output; break; | |
| 112 | ✗ | case ERROR_MODE: break; | |
| 113 | } | ||
| 114 | ✗ | return output; | |
| 115 | } | ||
| 116 | |||
| 117 | ✗ | static int fn(filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) | |
| 118 | { | ||
| 119 | ✗ | AudioNLMSContext *s = ctx->priv; | |
| 120 | ✗ | AVFrame *out = arg; | |
| 121 | ✗ | const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs; | |
| 122 | ✗ | const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs; | |
| 123 | |||
| 124 | ✗ | for (int c = start; c < end; c++) { | |
| 125 | ✗ | const ftype *input = (const ftype *)s->frame[0]->extended_data[c]; | |
| 126 | ✗ | const ftype *desired = (const ftype *)s->frame[1]->extended_data[c]; | |
| 127 | ✗ | ftype *delay = (ftype *)s->delay->extended_data[c]; | |
| 128 | ✗ | ftype *coeffs = (ftype *)s->coeffs->extended_data[c]; | |
| 129 | ✗ | ftype *tmp = (ftype *)s->tmp->extended_data[c]; | |
| 130 | ✗ | int *offset = (int *)s->offset->extended_data[c]; | |
| 131 | ✗ | ftype *output = (ftype *)out->extended_data[c]; | |
| 132 | |||
| 133 | ✗ | for (int n = 0; n < out->nb_samples; n++) { | |
| 134 | ✗ | output[n] = fn(process_sample)(s, input[n], desired[n], delay, coeffs, tmp, offset); | |
| 135 | ✗ | if (ctx->is_disabled) | |
| 136 | ✗ | output[n] = input[n]; | |
| 137 | } | ||
| 138 | } | ||
| 139 | |||
| 140 | ✗ | return 0; | |
| 141 | } | ||
| 142 |