1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
|
/*
* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.h"
#include "audio.h"
enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES };
enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS };
typedef struct SimpleLFO {
double phase;
double freq;
double offset;
double amount;
double pwidth;
int mode;
int srate;
} SimpleLFO;
typedef struct AudioPulsatorContext {
const AVClass *class;
int mode;
double level_in;
double level_out;
double amount;
double offset_l;
double offset_r;
double pwidth;
double bpm;
double hertz;
int ms;
int timing;
SimpleLFO lfoL, lfoR;
} AudioPulsatorContext;
#define OFFSET(x) offsetof(AudioPulsatorContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption apulsator_options[] = {
{ "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
{ "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, },
{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=SINE}, SINE, NB_MODES-1, FLAGS, .unit = "mode" },
{ "sine", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SINE}, 0, 0, FLAGS, .unit = "mode" },
{ "triangle", NULL, 0, AV_OPT_TYPE_CONST, {.i64=TRIANGLE},0, 0, FLAGS, .unit = "mode" },
{ "square", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQUARE}, 0, 0, FLAGS, .unit = "mode" },
{ "sawup", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWUP}, 0, 0, FLAGS, .unit = "mode" },
{ "sawdown", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWDOWN}, 0, 0, FLAGS, .unit = "mode" },
{ "amount", "set modulation", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, FLAGS },
{ "offset_l", "set offset L", OFFSET(offset_l), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, FLAGS },
{ "offset_r", "set offset R", OFFSET(offset_r), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS },
{ "width", "set pulse width", OFFSET(pwidth), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 2, FLAGS },
{ "timing", "set timing", OFFSET(timing), AV_OPT_TYPE_INT, {.i64=2}, 0, NB_TIMINGS-1, FLAGS, .unit = "timing" },
{ "bpm", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_BPM}, 0, 0, FLAGS, .unit = "timing" },
{ "ms", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_MS}, 0, 0, FLAGS, .unit = "timing" },
{ "hz", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_HZ}, 0, 0, FLAGS, .unit = "timing" },
{ "bpm", "set BPM", OFFSET(bpm), AV_OPT_TYPE_DOUBLE, {.dbl=120}, 30, 300, FLAGS },
{ "ms", "set ms", OFFSET(ms), AV_OPT_TYPE_INT, {.i64=500}, 10, 2000, FLAGS },
{ "hz", "set frequency", OFFSET(hertz), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0.01, 100, FLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(apulsator);
static void lfo_advance(SimpleLFO *lfo, unsigned count)
{
lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate);
if (lfo->phase >= 1)
lfo->phase = fmod(lfo->phase, 1);
}
static double lfo_get_value(SimpleLFO *lfo)
{
double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset);
double val;
if (phs > 1)
phs = fmod(phs, 1.);
switch (lfo->mode) {
case SINE:
val = sin(phs * 2 * M_PI);
break;
case TRIANGLE:
if (phs > 0.75)
val = (phs - 0.75) * 4 - 1;
else if (phs > 0.25)
val = -4 * phs + 2;
else
val = phs * 4;
break;
case SQUARE:
val = phs < 0.5 ? -1 : +1;
break;
case SAWUP:
val = phs * 2 - 1;
break;
case SAWDOWN:
val = 1 - phs * 2;
break;
default: av_assert0(0);
}
return val * lfo->amount;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioPulsatorContext *s = ctx->priv;
const double *src = (const double *)in->data[0];
const int nb_samples = in->nb_samples;
const double level_out = s->level_out;
const double level_in = s->level_in;
const double amount = s->amount;
AVFrame *out;
double *dst;
int n;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(inlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
dst = (double *)out->data[0];
for (n = 0; n < nb_samples; n++) {
double outL;
double outR;
double inL = src[0] * level_in;
double inR = src[1] * level_in;
double procL = inL;
double procR = inR;
procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2;
procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2;
outL = procL + inL * (1 - amount);
outR = procR + inR * (1 - amount);
outL *= level_out;
outR *= level_out;
dst[0] = outL;
dst[1] = outR;
lfo_advance(&s->lfoL, 1);
lfo_advance(&s->lfoR, 1);
dst += 2;
src += 2;
}
if (in != out)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterChannelLayouts *layout = NULL;
AVFilterFormats *formats = NULL;
int ret;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
(ret = ff_set_common_formats (ctx , formats )) < 0 ||
(ret = ff_add_channel_layout (&layout , &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) < 0 ||
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioPulsatorContext *s = ctx->priv;
double freq;
switch (s->timing) {
case UNIT_BPM: freq = s->bpm / 60; break;
case UNIT_MS: freq = 1 / (s->ms / 1000.); break;
case UNIT_HZ: freq = s->hertz; break;
default: av_assert0(0);
}
s->lfoL.freq = freq;
s->lfoR.freq = freq;
s->lfoL.mode = s->mode;
s->lfoR.mode = s->mode;
s->lfoL.offset = s->offset_l;
s->lfoR.offset = s->offset_r;
s->lfoL.srate = inlink->sample_rate;
s->lfoR.srate = inlink->sample_rate;
s->lfoL.amount = s->amount;
s->lfoR.amount = s->amount;
s->lfoL.pwidth = s->pwidth;
s->lfoR.pwidth = s->pwidth;
return 0;
}
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
.filter_frame = filter_frame,
},
};
const AVFilter ff_af_apulsator = {
.name = "apulsator",
.description = NULL_IF_CONFIG_SMALL("Audio pulsator."),
.priv_size = sizeof(AudioPulsatorContext),
.priv_class = &apulsator_class,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_QUERY_FUNC(query_formats),
};
|
