FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_flanger.c
Date: 2022-12-05 20:26:17
Exec Total Coverage
Lines: 0 72 0.0%
Functions: 0 4 0.0%
Branches: 0 18 0.0%

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1 /*
2 * Copyright (c) 2006 Rob Sykes <robs@users.sourceforge.net>
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/avstring.h"
22 #include "libavutil/opt.h"
23 #include "libavutil/samplefmt.h"
24 #include "avfilter.h"
25 #include "audio.h"
26 #include "internal.h"
27 #include "generate_wave_table.h"
28
29 #define INTERPOLATION_LINEAR 0
30 #define INTERPOLATION_QUADRATIC 1
31
32 typedef struct FlangerContext {
33 const AVClass *class;
34 double delay_min;
35 double delay_depth;
36 double feedback_gain;
37 double delay_gain;
38 double speed;
39 int wave_shape;
40 double channel_phase;
41 int interpolation;
42 double in_gain;
43 int max_samples;
44 uint8_t **delay_buffer;
45 int delay_buf_pos;
46 double *delay_last;
47 float *lfo;
48 int lfo_length;
49 int lfo_pos;
50 } FlangerContext;
51
52 #define OFFSET(x) offsetof(FlangerContext, x)
53 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
54
55 static const AVOption flanger_options[] = {
56 { "delay", "base delay in milliseconds", OFFSET(delay_min), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 30, A },
57 { "depth", "added swept delay in milliseconds", OFFSET(delay_depth), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0, 10, A },
58 { "regen", "percentage regeneration (delayed signal feedback)", OFFSET(feedback_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -95, 95, A },
59 { "width", "percentage of delayed signal mixed with original", OFFSET(delay_gain), AV_OPT_TYPE_DOUBLE, {.dbl=71}, 0, 100, A },
60 { "speed", "sweeps per second (Hz)", OFFSET(speed), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.1, 10, A },
61 { "shape", "swept wave shape", OFFSET(wave_shape), AV_OPT_TYPE_INT, {.i64=WAVE_SIN}, WAVE_SIN, WAVE_NB-1, A, "type" },
62 { "triangular", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
63 { "t", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_TRI}, 0, 0, A, "type" },
64 { "sinusoidal", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
65 { "s", NULL, 0, AV_OPT_TYPE_CONST, {.i64=WAVE_SIN}, 0, 0, A, "type" },
66 { "phase", "swept wave percentage phase-shift for multi-channel", OFFSET(channel_phase), AV_OPT_TYPE_DOUBLE, {.dbl=25}, 0, 100, A },
67 { "interp", "delay-line interpolation", OFFSET(interpolation), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "itype" },
68 { "linear", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_LINEAR}, 0, 0, A, "itype" },
69 { "quadratic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=INTERPOLATION_QUADRATIC}, 0, 0, A, "itype" },
70 { NULL }
71 };
72
73 AVFILTER_DEFINE_CLASS(flanger);
74
75 static av_cold int init(AVFilterContext *ctx)
76 {
77 FlangerContext *s = ctx->priv;
78
79 s->feedback_gain /= 100;
80 s->delay_gain /= 100;
81 s->channel_phase /= 100;
82 s->delay_min /= 1000;
83 s->delay_depth /= 1000;
84 s->in_gain = 1 / (1 + s->delay_gain);
85 s->delay_gain /= 1 + s->delay_gain;
86 s->delay_gain *= 1 - fabs(s->feedback_gain);
87
88 return 0;
89 }
90
91 static int config_input(AVFilterLink *inlink)
92 {
93 AVFilterContext *ctx = inlink->dst;
94 FlangerContext *s = ctx->priv;
95
96 s->max_samples = (s->delay_min + s->delay_depth) * inlink->sample_rate + 2.5;
97 s->lfo_length = inlink->sample_rate / s->speed;
98 s->delay_last = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->delay_last));
99 s->lfo = av_calloc(s->lfo_length, sizeof(*s->lfo));
100 if (!s->lfo || !s->delay_last)
101 return AVERROR(ENOMEM);
102
103 ff_generate_wave_table(s->wave_shape, AV_SAMPLE_FMT_FLT, s->lfo, s->lfo_length,
104 rint(s->delay_min * inlink->sample_rate),
105 s->max_samples - 2., 3 * M_PI_2);
106
107 return av_samples_alloc_array_and_samples(&s->delay_buffer, NULL,
108 inlink->ch_layout.nb_channels, s->max_samples,
109 inlink->format, 0);
110 }
111
112 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
113 {
114 AVFilterContext *ctx = inlink->dst;
115 FlangerContext *s = ctx->priv;
116 AVFrame *out_frame;
117 int chan, i;
118
119 if (av_frame_is_writable(frame)) {
120 out_frame = frame;
121 } else {
122 out_frame = ff_get_audio_buffer(ctx->outputs[0], frame->nb_samples);
123 if (!out_frame) {
124 av_frame_free(&frame);
125 return AVERROR(ENOMEM);
126 }
127 av_frame_copy_props(out_frame, frame);
128 }
129
130 for (i = 0; i < frame->nb_samples; i++) {
131
132 s->delay_buf_pos = (s->delay_buf_pos + s->max_samples - 1) % s->max_samples;
133
134 for (chan = 0; chan < inlink->ch_layout.nb_channels; chan++) {
135 double *src = (double *)frame->extended_data[chan];
136 double *dst = (double *)out_frame->extended_data[chan];
137 double delayed_0, delayed_1;
138 double delayed;
139 double in, out;
140 int channel_phase = chan * s->lfo_length * s->channel_phase + .5;
141 double delay = s->lfo[(s->lfo_pos + channel_phase) % s->lfo_length];
142 int int_delay = (int)delay;
143 double frac_delay = modf(delay, &delay);
144 double *delay_buffer = (double *)s->delay_buffer[chan];
145
146 in = src[i];
147 delay_buffer[s->delay_buf_pos] = in + s->delay_last[chan] *
148 s->feedback_gain;
149 delayed_0 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
150 delayed_1 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
151
152 if (s->interpolation == INTERPOLATION_LINEAR) {
153 delayed = delayed_0 + (delayed_1 - delayed_0) * frac_delay;
154 } else {
155 double a, b;
156 double delayed_2 = delay_buffer[(s->delay_buf_pos + int_delay++) % s->max_samples];
157 delayed_2 -= delayed_0;
158 delayed_1 -= delayed_0;
159 a = delayed_2 * .5 - delayed_1;
160 b = delayed_1 * 2 - delayed_2 *.5;
161 delayed = delayed_0 + (a * frac_delay + b) * frac_delay;
162 }
163
164 s->delay_last[chan] = delayed;
165 out = in * s->in_gain + delayed * s->delay_gain;
166 dst[i] = out;
167 }
168 s->lfo_pos = (s->lfo_pos + 1) % s->lfo_length;
169 }
170
171 if (frame != out_frame)
172 av_frame_free(&frame);
173
174 return ff_filter_frame(ctx->outputs[0], out_frame);
175 }
176
177 static av_cold void uninit(AVFilterContext *ctx)
178 {
179 FlangerContext *s = ctx->priv;
180
181 av_freep(&s->lfo);
182 av_freep(&s->delay_last);
183
184 if (s->delay_buffer)
185 av_freep(&s->delay_buffer[0]);
186 av_freep(&s->delay_buffer);
187 }
188
189 static const AVFilterPad flanger_inputs[] = {
190 {
191 .name = "default",
192 .type = AVMEDIA_TYPE_AUDIO,
193 .config_props = config_input,
194 .filter_frame = filter_frame,
195 },
196 };
197
198 static const AVFilterPad flanger_outputs[] = {
199 {
200 .name = "default",
201 .type = AVMEDIA_TYPE_AUDIO,
202 },
203 };
204
205 const AVFilter ff_af_flanger = {
206 .name = "flanger",
207 .description = NULL_IF_CONFIG_SMALL("Apply a flanging effect to the audio."),
208 .priv_size = sizeof(FlangerContext),
209 .priv_class = &flanger_class,
210 .init = init,
211 .uninit = uninit,
212 FILTER_INPUTS(flanger_inputs),
213 FILTER_OUTPUTS(flanger_outputs),
214 FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
215 };
216