FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_anlms.c
Date: 2024-02-29 09:57:37
Exec Total Coverage
Lines: 0 82 0.0%
Functions: 0 5 0.0%
Branches: 0 65 0.0%

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1 /*
2 * Copyright (c) 2019 Paul B Mahol
3 *
4 * This file is part of FFmpeg.
5 *
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
10 *
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #include "libavutil/channel_layout.h"
22 #include "libavutil/common.h"
23 #include "libavutil/float_dsp.h"
24 #include "libavutil/opt.h"
25
26 #include "audio.h"
27 #include "avfilter.h"
28 #include "filters.h"
29 #include "formats.h"
30 #include "internal.h"
31
32 enum OutModes {
33 IN_MODE,
34 DESIRED_MODE,
35 OUT_MODE,
36 NOISE_MODE,
37 ERROR_MODE,
38 NB_OMODES
39 };
40
41 typedef struct AudioNLMSContext {
42 const AVClass *class;
43
44 int order;
45 float mu;
46 float eps;
47 float leakage;
48 int output_mode;
49 int precision;
50
51 int kernel_size;
52 AVFrame *offset;
53 AVFrame *delay;
54 AVFrame *coeffs;
55 AVFrame *tmp;
56
57 AVFrame *frame[2];
58
59 int anlmf;
60
61 int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
62
63 AVFloatDSPContext *fdsp;
64 } AudioNLMSContext;
65
66 #define OFFSET(x) offsetof(AudioNLMSContext, x)
67 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
68 #define AT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
69
70 static const AVOption anlms_options[] = {
71 { "order", "set the filter order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=256}, 1, INT16_MAX, A },
72 { "mu", "set the filter mu", OFFSET(mu), AV_OPT_TYPE_FLOAT, {.dbl=0.75}, 0, 2, AT },
73 { "eps", "set the filter eps", OFFSET(eps), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 1, AT },
74 { "leakage", "set the filter leakage", OFFSET(leakage), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, AT },
75 { "out_mode", "set output mode", OFFSET(output_mode), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_OMODES-1, AT, .unit = "mode" },
76 { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AT, .unit = "mode" },
77 { "d", "desired", 0, AV_OPT_TYPE_CONST, {.i64=DESIRED_MODE}, 0, 0, AT, .unit = "mode" },
78 { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AT, .unit = "mode" },
79 { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE}, 0, 0, AT, .unit = "mode" },
80 { "e", "error", 0, AV_OPT_TYPE_CONST, {.i64=ERROR_MODE}, 0, 0, AT, .unit = "mode" },
81 { "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, .unit = "precision" },
82 { "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "precision" },
83 { "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "precision" },
84 { "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, .unit = "precision" },
85 { NULL }
86 };
87
88 AVFILTER_DEFINE_CLASS_EXT(anlms, "anlm(f|s)", anlms_options);
89
90 static int query_formats(AVFilterContext *ctx)
91 {
92 AudioNLMSContext *s = ctx->priv;
93 static const enum AVSampleFormat sample_fmts[3][3] = {
94 { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
95 { AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
96 { AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
97 };
98 int ret;
99
100 if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
101 return ret;
102
103 if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
104 return ret;
105
106 return ff_set_common_all_samplerates(ctx);
107 }
108
109 static int activate(AVFilterContext *ctx)
110 {
111 AudioNLMSContext *s = ctx->priv;
112 int i, ret, status;
113 int nb_samples;
114 int64_t pts;
115
116 FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
117
118 nb_samples = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]),
119 ff_inlink_queued_samples(ctx->inputs[1]));
120 for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) {
121 if (s->frame[i])
122 continue;
123
124 if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) {
125 ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frame[i]);
126 if (ret < 0)
127 return ret;
128 }
129 }
130
131 if (s->frame[0] && s->frame[1]) {
132 AVFrame *out;
133
134 out = ff_get_audio_buffer(ctx->outputs[0], s->frame[0]->nb_samples);
135 if (!out) {
136 av_frame_free(&s->frame[0]);
137 av_frame_free(&s->frame[1]);
138 return AVERROR(ENOMEM);
139 }
140
141 ff_filter_execute(ctx, s->filter_channels, out, NULL,
142 FFMIN(ctx->outputs[0]->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
143
144 out->pts = s->frame[0]->pts;
145 out->duration = s->frame[0]->duration;
146
147 av_frame_free(&s->frame[0]);
148 av_frame_free(&s->frame[1]);
149
150 ret = ff_filter_frame(ctx->outputs[0], out);
151 if (ret < 0)
152 return ret;
153 }
154
155 if (!nb_samples) {
156 for (i = 0; i < 2; i++) {
157 if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
158 ff_outlink_set_status(ctx->outputs[0], status, pts);
159 return 0;
160 }
161 }
162 }
163
164 if (ff_outlink_frame_wanted(ctx->outputs[0])) {
165 for (i = 0; i < 2; i++) {
166 if (s->frame[i] || ff_inlink_queued_samples(ctx->inputs[i]) > 0)
167 continue;
168 ff_inlink_request_frame(ctx->inputs[i]);
169 return 0;
170 }
171 }
172 return 0;
173 }
174
175 #define DEPTH 32
176 #include "anlms_template.c"
177
178 #undef DEPTH
179 #define DEPTH 64
180 #include "anlms_template.c"
181
182 static int config_output(AVFilterLink *outlink)
183 {
184 AVFilterContext *ctx = outlink->src;
185 AudioNLMSContext *s = ctx->priv;
186
187 s->anlmf = !strcmp(ctx->filter->name, "anlmf");
188 s->kernel_size = FFALIGN(s->order, 16);
189
190 if (!s->offset)
191 s->offset = ff_get_audio_buffer(outlink, 1);
192 if (!s->delay)
193 s->delay = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
194 if (!s->coeffs)
195 s->coeffs = ff_get_audio_buffer(outlink, 2 * s->kernel_size);
196 if (!s->tmp)
197 s->tmp = ff_get_audio_buffer(outlink, s->kernel_size);
198 if (!s->delay || !s->coeffs || !s->offset || !s->tmp)
199 return AVERROR(ENOMEM);
200
201 switch (outlink->format) {
202 case AV_SAMPLE_FMT_DBLP:
203 s->filter_channels = filter_channels_double;
204 break;
205 case AV_SAMPLE_FMT_FLTP:
206 s->filter_channels = filter_channels_float;
207 break;
208 }
209
210 return 0;
211 }
212
213 static av_cold int init(AVFilterContext *ctx)
214 {
215 AudioNLMSContext *s = ctx->priv;
216
217 s->fdsp = avpriv_float_dsp_alloc(0);
218 if (!s->fdsp)
219 return AVERROR(ENOMEM);
220
221 return 0;
222 }
223
224 static av_cold void uninit(AVFilterContext *ctx)
225 {
226 AudioNLMSContext *s = ctx->priv;
227
228 av_freep(&s->fdsp);
229 av_frame_free(&s->delay);
230 av_frame_free(&s->coeffs);
231 av_frame_free(&s->offset);
232 av_frame_free(&s->tmp);
233 }
234
235 static const AVFilterPad inputs[] = {
236 {
237 .name = "input",
238 .type = AVMEDIA_TYPE_AUDIO,
239 },
240 {
241 .name = "desired",
242 .type = AVMEDIA_TYPE_AUDIO,
243 },
244 };
245
246 static const AVFilterPad outputs[] = {
247 {
248 .name = "default",
249 .type = AVMEDIA_TYPE_AUDIO,
250 .config_props = config_output,
251 },
252 };
253
254 const AVFilter ff_af_anlms = {
255 .name = "anlms",
256 .description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Squares algorithm to first audio stream."),
257 .priv_size = sizeof(AudioNLMSContext),
258 .priv_class = &anlms_class,
259 .init = init,
260 .uninit = uninit,
261 .activate = activate,
262 FILTER_INPUTS(inputs),
263 FILTER_OUTPUTS(outputs),
264 FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_FLTP),
265 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
266 AVFILTER_FLAG_SLICE_THREADS,
267 .process_command = ff_filter_process_command,
268 };
269
270 const AVFilter ff_af_anlmf = {
271 .name = "anlmf",
272 .description = NULL_IF_CONFIG_SMALL("Apply Normalized Least-Mean-Fourth algorithm to first audio stream."),
273 .priv_size = sizeof(AudioNLMSContext),
274 .priv_class = &anlms_class,
275 .init = init,
276 .uninit = uninit,
277 .activate = activate,
278 FILTER_INPUTS(inputs),
279 FILTER_OUTPUTS(outputs),
280 FILTER_QUERY_FUNC(query_formats),
281 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
282 AVFILTER_FLAG_SLICE_THREADS,
283 .process_command = ff_filter_process_command,
284 };
285