FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavfilter/af_apulsator.c
Date: 2026-09-24 20:08:24
Exec Total Coverage
Lines: 0 97 0.0%
Functions: 0 5 0.0%
Branches: 0 46 0.0%
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1 /*
2 * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
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/avassert.h"
22 #include "libavutil/channel_layout.h"
23 #include "libavutil/opt.h"
24 #include "avfilter.h"
25 #include "filters.h"
26 #include "formats.h"
27 #include "audio.h"
28
29 enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES };
30 enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS };
31
32 typedef struct SimpleLFO {
33 double phase;
34 double freq;
35 double offset;
36 double amount;
37 double pwidth;
38 int mode;
39 int srate;
40 } SimpleLFO;
41
42 typedef struct AudioPulsatorContext {
43 const AVClass *class;
44 int mode;
45 double level_in;
46 double level_out;
47 double amount;
48 double offset_l;
49 double offset_r;
50 double pwidth;
51 double bpm;
52 double hertz;
53 int ms;
54 int timing;
55
56 SimpleLFO lfoL, lfoR;
57 } AudioPulsatorContext;
58
59 #define OFFSET(x) offsetof(AudioPulsatorContext, x)
60 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
61
62 static const AVOption apulsator_options[] = {
63 { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
64 { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
65 { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=SINE}, SINE, NB_MODES-1, FLAGS, .unit = "mode" },
66 { "sine", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SINE}, 0, 0, FLAGS, .unit = "mode" },
67 { "triangle", NULL, 0, AV_OPT_TYPE_CONST, {.i64=TRIANGLE},0, 0, FLAGS, .unit = "mode" },
68 { "square", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQUARE}, 0, 0, FLAGS, .unit = "mode" },
69 { "sawup", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWUP}, 0, 0, FLAGS, .unit = "mode" },
70 { "sawdown", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWDOWN}, 0, 0, FLAGS, .unit = "mode" },
71 { "amount", "set modulation", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, FLAGS },
72 { "offset_l", "set offset L", OFFSET(offset_l), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, FLAGS },
73 { "offset_r", "set offset R", OFFSET(offset_r), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS },
74 { "width", "set pulse width", OFFSET(pwidth), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 2, FLAGS },
75 { "timing", "set timing", OFFSET(timing), AV_OPT_TYPE_INT, {.i64=2}, 0, NB_TIMINGS-1, FLAGS, .unit = "timing" },
76 { "bpm", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_BPM}, 0, 0, FLAGS, .unit = "timing" },
77 { "ms", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_MS}, 0, 0, FLAGS, .unit = "timing" },
78 { "hz", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_HZ}, 0, 0, FLAGS, .unit = "timing" },
79 { "bpm", "set BPM", OFFSET(bpm), AV_OPT_TYPE_DOUBLE, {.dbl=120}, 30, 300, FLAGS },
80 { "ms", "set ms", OFFSET(ms), AV_OPT_TYPE_INT, {.i64=500}, 10, 2000, FLAGS },
81 { "hz", "set frequency", OFFSET(hertz), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0.01, 100, FLAGS },
82 { NULL }
83 };
84
85 AVFILTER_DEFINE_CLASS(apulsator);
86
87 ✗ static void lfo_advance(SimpleLFO *lfo, unsigned count)
88 {
89 ✗ lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate);
90 ✗ if (lfo->phase >= 1)
91 ✗ lfo->phase = fmod(lfo->phase, 1);
92 ✗ }
93
94 ✗ static double lfo_get_value(SimpleLFO *lfo)
95 {
96 ✗ double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
97 double val;
98
99 ✗ if (phs > 1)
100 ✗ phs = fmod(phs, 1.);
101
102 ✗ switch (lfo->mode) {
103 ✗ case SINE:
104 ✗ val = sin(phs * 2 * M_PI);
105 ✗ break;
106 ✗ case TRIANGLE:
107 ✗ if (phs > 0.75)
108 ✗ val = (phs - 0.75) * 4 - 1;
109 ✗ else if (phs > 0.25)
110 ✗ val = -4 * phs + 2;
111 else
112 ✗ val = phs * 4;
113 ✗ break;
114 ✗ case SQUARE:
115 ✗ val = phs < 0.5 ? -1 : +1;
116 ✗ break;
117 ✗ case SAWUP:
118 ✗ val = phs * 2 - 1;
119 ✗ break;
120 ✗ case SAWDOWN:
121 ✗ val = 1 - phs * 2;
122 ✗ break;
123 ✗ default: av_assert0(0);
124 }
125
126 ✗ return val * lfo->amount;
127 }
128
129 ✗ static int filter_frame(AVFilterLink *inlink, AVFrame *in)
130 {
131 ✗ AVFilterContext *ctx = inlink->dst;
132 ✗ AVFilterLink *outlink = ctx->outputs[0];
133 ✗ AudioPulsatorContext *s = ctx->priv;
134 ✗ const double *src = (const double *)in->data[0];
135 ✗ const int nb_samples = in->nb_samples;
136 ✗ const double level_out = s->level_out;
137 ✗ const double level_in = s->level_in;
138 ✗ const double amount = s->amount;
139 AVFrame *out;
140 double *dst;
141 int n;
142
143 ✗ if (av_frame_is_writable(in)) {
144 ✗ out = in;
145 } else {
146 ✗ out = ff_get_audio_buffer(inlink, in->nb_samples);
147 ✗ if (!out) {
148 ✗ av_frame_free(&in);
149 ✗ return AVERROR(ENOMEM);
150 }
151 ✗ av_frame_copy_props(out, in);
152 }
153 ✗ dst = (double *)out->data[0];
154
155 ✗ for (n = 0; n < nb_samples; n++) {
156 double outL;
157 double outR;
158 ✗ double inL = src[0] * level_in;
159 ✗ double inR = src[1] * level_in;
160 ✗ double procL = inL;
161 ✗ double procR = inR;
162
163 ✗ procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2;
164 ✗ procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2;
165
166 ✗ outL = procL + inL * (1 - amount);
167 ✗ outR = procR + inR * (1 - amount);
168
169 ✗ outL *= level_out;
170 ✗ outR *= level_out;
171
172 ✗ dst[0] = outL;
173 ✗ dst[1] = outR;
174
175 ✗ lfo_advance(&s->lfoL, 1);
176 ✗ lfo_advance(&s->lfoR, 1);
177
178 ✗ dst += 2;
179 ✗ src += 2;
180 }
181
182 ✗ if (in != out)
183 ✗ av_frame_free(&in);
184
185 ✗ return ff_filter_frame(outlink, out);
186 }
187
188 ✗ static int query_formats(const AVFilterContext *ctx,
189 AVFilterFormatsConfig **cfg_in,
190 AVFilterFormatsConfig **cfg_out)
191 {
192 static const enum AVSampleFormat formats[] = {
193 AV_SAMPLE_FMT_DBL,
194 AV_SAMPLE_FMT_NONE,
195 };
196 static const AVChannelLayout layouts[] = {
197 AV_CHANNEL_LAYOUT_STEREO,
198 { .nb_channels = 0 },
199 };
200
201 int ret;
202
203 ✗ ret = ff_set_sample_formats_from_list2(ctx, cfg_in, cfg_out, formats);
204 ✗ if (ret < 0)
205 ✗ return ret;
206
207 ✗ ret = ff_set_common_channel_layouts_from_list2(ctx, cfg_in, cfg_out, layouts);
208 ✗ if (ret < 0)
209 ✗ return ret;
210
211 ✗ return 0;
212 }
213
214 ✗ static int config_input(AVFilterLink *inlink)
215 {
216 ✗ AVFilterContext *ctx = inlink->dst;
217 ✗ AudioPulsatorContext *s = ctx->priv;
218 double freq;
219
220 ✗ switch (s->timing) {
221 ✗ case UNIT_BPM: freq = s->bpm / 60; break;
222 ✗ case UNIT_MS: freq = 1 / (s->ms / 1000.); break;
223 ✗ case UNIT_HZ: freq = s->hertz; break;
224 ✗ default: av_assert0(0);
225 }
226
227 ✗ s->lfoL.freq = freq;
228 ✗ s->lfoR.freq = freq;
229 ✗ s->lfoL.mode = s->mode;
230 ✗ s->lfoR.mode = s->mode;
231 ✗ s->lfoL.offset = s->offset_l;
232 ✗ s->lfoR.offset = s->offset_r;
233 ✗ s->lfoL.srate = inlink->sample_rate;
234 ✗ s->lfoR.srate = inlink->sample_rate;
235 ✗ s->lfoL.amount = s->amount;
236 ✗ s->lfoR.amount = s->amount;
237 ✗ s->lfoL.pwidth = s->pwidth;
238 ✗ s->lfoR.pwidth = s->pwidth;
239
240 ✗ return 0;
241 }
242
243 static const AVFilterPad inputs[] = {
244 {
245 .name = "default",
246 .type = AVMEDIA_TYPE_AUDIO,
247 .config_props = config_input,
248 .filter_frame = filter_frame,
249 },
250 };
251
252 const FFFilter ff_af_apulsator = {
253 .p.name = "apulsator",
254 .p.description = NULL_IF_CONFIG_SMALL("Audio pulsator."),
255 .p.priv_class = &apulsator_class,
256 .priv_size = sizeof(AudioPulsatorContext),
257 FILTER_INPUTS(inputs),
258 FILTER_OUTPUTS(ff_audio_default_filterpad),
259 FILTER_QUERY_FUNC2(query_formats),
260 };
261