Signed-off-by: Paul B Mahol <onemda@gmail.com>
Paul B Mahol authored on 2015/11/29 03:50:32... | ... |
@@ -1030,6 +1030,63 @@ It accepts the following values: |
1030 | 1030 |
@end table |
1031 | 1031 |
@end table |
1032 | 1032 |
|
1033 |
+@section apulsator |
|
1034 |
+ |
|
1035 |
+Audio pulsator is something between an autopanner and a tremolo. |
|
1036 |
+But it can produce funny stereo effects as well. Pulsator changes the volume |
|
1037 |
+of the left and right channel based on a LFO (low frequency oscillator) with |
|
1038 |
+different waveforms and shifted phases. |
|
1039 |
+This filter have the ability to define an offset between left and right |
|
1040 |
+channel. An offset of 0 means that both LFO shapes match each other. |
|
1041 |
+The left and right channel are altered equally - a conventional tremolo. |
|
1042 |
+An offset of 50% means that the shape of the right channel is exactly shifted |
|
1043 |
+in phase (or moved backwards about half of the frequency) - pulsator acts as |
|
1044 |
+an autopanner. At 1 both curves match again. Every setting in between moves the |
|
1045 |
+phase shift gapless between all stages and produces some "bypassing" sounds with |
|
1046 |
+sine and triangle waveforms. The more you set the offset near 1 (starting from |
|
1047 |
+the 0.5) the faster the signal passes from the left to the right speaker. |
|
1048 |
+ |
|
1049 |
+The filter accepts the following options: |
|
1050 |
+ |
|
1051 |
+@table @option |
|
1052 |
+@item level_in |
|
1053 |
+Set input gain. By default it is 1. Range is [0.015625 - 64]. |
|
1054 |
+ |
|
1055 |
+@item level_out |
|
1056 |
+Set output gain. By default it is 1. Range is [0.015625 - 64]. |
|
1057 |
+ |
|
1058 |
+@item mode |
|
1059 |
+Set waveform shape the LFO will use. Can be one of: sine, triangle, square, |
|
1060 |
+sawup or sawdown. Default is sine. |
|
1061 |
+ |
|
1062 |
+@item amount |
|
1063 |
+Set modulation. Define how much of original signal is affected by the LFO. |
|
1064 |
+ |
|
1065 |
+@item offset_l |
|
1066 |
+Set left channel offset. Default is 0. Allowed range is [0 - 1]. |
|
1067 |
+ |
|
1068 |
+@item offset_r |
|
1069 |
+Set right channel offset. Default is 0.5. Allowed range is [0 - 1]. |
|
1070 |
+ |
|
1071 |
+@item width |
|
1072 |
+Set pulse width. Default is 1. Allowed range is [0 - 2]. |
|
1073 |
+ |
|
1074 |
+@item timing |
|
1075 |
+Set possible timing mode. Can be one of: bpm, ms or hz. Default is hz. |
|
1076 |
+ |
|
1077 |
+@item bpm |
|
1078 |
+Set bpm. Default is 120. Allowed range is [30 - 300]. Only used if timing |
|
1079 |
+is set to bpm. |
|
1080 |
+ |
|
1081 |
+@item ms |
|
1082 |
+Set ms. Default is 500. Allowed range is [10 - 2000]. Only used if timing |
|
1083 |
+is set to ms. |
|
1084 |
+ |
|
1085 |
+@item hz |
|
1086 |
+Set frequency in Hz. Default is 2. Allowed range is [0.01 - 100]. Only used |
|
1087 |
+if timing is set to hz. |
|
1088 |
+@end table |
|
1089 |
+ |
|
1033 | 1090 |
@anchor{aresample} |
1034 | 1091 |
@section aresample |
1035 | 1092 |
|
... | ... |
@@ -40,6 +40,7 @@ OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o |
40 | 40 |
OBJS-$(CONFIG_APAD_FILTER) += af_apad.o |
41 | 41 |
OBJS-$(CONFIG_APERMS_FILTER) += f_perms.o |
42 | 42 |
OBJS-$(CONFIG_APHASER_FILTER) += af_aphaser.o generate_wave_table.o |
43 |
+OBJS-$(CONFIG_APULSATOR_FILTER) += af_apulsator.o |
|
43 | 44 |
OBJS-$(CONFIG_AREALTIME_FILTER) += f_realtime.o |
44 | 45 |
OBJS-$(CONFIG_ARESAMPLE_FILTER) += af_aresample.o |
45 | 46 |
OBJS-$(CONFIG_AREVERSE_FILTER) += f_reverse.o |
46 | 47 |
new file mode 100644 |
... | ... |
@@ -0,0 +1,254 @@ |
0 |
+/* |
|
1 |
+ * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others |
|
2 |
+ * |
|
3 |
+ * This file is part of FFmpeg. |
|
4 |
+ * |
|
5 |
+ * FFmpeg is free software; you can redistribute it and/or |
|
6 |
+ * modify it under the terms of the GNU Lesser General Public |
|
7 |
+ * License as published by the Free Software Foundation; either |
|
8 |
+ * version 2.1 of the License, or (at your option) any later version. |
|
9 |
+ * |
|
10 |
+ * FFmpeg is distributed in the hope that it will be useful, |
|
11 |
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of |
|
12 |
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
|
13 |
+ * Lesser General Public License for more details. |
|
14 |
+ * |
|
15 |
+ * You should have received a copy of the GNU Lesser General Public |
|
16 |
+ * License along with FFmpeg; if not, write to the Free Software |
|
17 |
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
|
18 |
+ */ |
|
19 |
+ |
|
20 |
+#include "libavutil/opt.h" |
|
21 |
+#include "avfilter.h" |
|
22 |
+#include "internal.h" |
|
23 |
+#include "audio.h" |
|
24 |
+ |
|
25 |
+enum PulsatorModes { SINE, TRIANGLE, SQUARE, SAWUP, SAWDOWN, NB_MODES }; |
|
26 |
+enum PulsatorTimings { UNIT_BPM, UNIT_MS, UNIT_HZ, NB_TIMINGS }; |
|
27 |
+ |
|
28 |
+typedef struct SimpleLFO { |
|
29 |
+ double phase; |
|
30 |
+ double freq; |
|
31 |
+ double offset; |
|
32 |
+ double amount; |
|
33 |
+ double pwidth; |
|
34 |
+ int mode; |
|
35 |
+ int srate; |
|
36 |
+} SimpleLFO; |
|
37 |
+ |
|
38 |
+typedef struct AudioPulsatorContext { |
|
39 |
+ const AVClass *class; |
|
40 |
+ int mode; |
|
41 |
+ double level_in; |
|
42 |
+ double level_out; |
|
43 |
+ double amount; |
|
44 |
+ double offset_l; |
|
45 |
+ double offset_r; |
|
46 |
+ double pwidth; |
|
47 |
+ double bpm; |
|
48 |
+ double hz; |
|
49 |
+ int ms; |
|
50 |
+ int timing; |
|
51 |
+ |
|
52 |
+ SimpleLFO lfoL, lfoR; |
|
53 |
+} AudioPulsatorContext; |
|
54 |
+ |
|
55 |
+#define OFFSET(x) offsetof(AudioPulsatorContext, x) |
|
56 |
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
|
57 |
+ |
|
58 |
+static const AVOption apulsator_options[] = { |
|
59 |
+ { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, }, |
|
60 |
+ { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, FLAGS, }, |
|
61 |
+ { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=SINE}, SINE, NB_MODES-1, FLAGS, "mode" }, |
|
62 |
+ { "sine", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SINE}, 0, 0, FLAGS, "mode" }, |
|
63 |
+ { "triangle", NULL, 0, AV_OPT_TYPE_CONST, {.i64=TRIANGLE},0, 0, FLAGS, "mode" }, |
|
64 |
+ { "square", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQUARE}, 0, 0, FLAGS, "mode" }, |
|
65 |
+ { "sawup", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWUP}, 0, 0, FLAGS, "mode" }, |
|
66 |
+ { "sawdown", NULL, 0, AV_OPT_TYPE_CONST, {.i64=SAWDOWN}, 0, 0, FLAGS, "mode" }, |
|
67 |
+ { "amount", "set modulation", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, FLAGS }, |
|
68 |
+ { "offset_l", "set offset L", OFFSET(offset_l), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, FLAGS }, |
|
69 |
+ { "offset_r", "set offset R", OFFSET(offset_r), AV_OPT_TYPE_DOUBLE, {.dbl=.5}, 0, 1, FLAGS }, |
|
70 |
+ { "width", "set pulse width", OFFSET(pwidth), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 2, FLAGS }, |
|
71 |
+ { "timing", "set timing", OFFSET(timing), AV_OPT_TYPE_INT, {.i64=2}, 0, NB_TIMINGS-1, FLAGS, "timing" }, |
|
72 |
+ { "bpm", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_BPM}, 0, 0, FLAGS, "timing" }, |
|
73 |
+ { "ms", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_MS}, 0, 0, FLAGS, "timing" }, |
|
74 |
+ { "hz", NULL, 0, AV_OPT_TYPE_CONST, {.i64=UNIT_HZ}, 0, 0, FLAGS, "timing" }, |
|
75 |
+ { "bpm", "set BPM", OFFSET(bpm), AV_OPT_TYPE_DOUBLE, {.dbl=120}, 30, 300, FLAGS }, |
|
76 |
+ { "ms", "set ms", OFFSET(ms), AV_OPT_TYPE_INT, {.i64=500}, 10, 2000, FLAGS }, |
|
77 |
+ { "hz", "set frequency", OFFSET(hz), AV_OPT_TYPE_DOUBLE, {.dbl=2}, 0.01, 100, FLAGS }, |
|
78 |
+ { NULL } |
|
79 |
+}; |
|
80 |
+ |
|
81 |
+AVFILTER_DEFINE_CLASS(apulsator); |
|
82 |
+ |
|
83 |
+static void lfo_advance(SimpleLFO *lfo, unsigned count) |
|
84 |
+{ |
|
85 |
+ lfo->phase = fabs(lfo->phase + count * lfo->freq / lfo->srate); |
|
86 |
+ if (lfo->phase >= 1) |
|
87 |
+ lfo->phase = fmod(lfo->phase, 1); |
|
88 |
+} |
|
89 |
+ |
|
90 |
+static double lfo_get_value(SimpleLFO *lfo) |
|
91 |
+{ |
|
92 |
+ double phs = FFMIN(100, lfo->phase / FFMIN(1.99, FFMAX(0.01, lfo->pwidth)) + lfo->offset); |
|
93 |
+ double val; |
|
94 |
+ |
|
95 |
+ if (phs > 1) |
|
96 |
+ phs = fmod(phs, 1.); |
|
97 |
+ |
|
98 |
+ switch (lfo->mode) { |
|
99 |
+ case SINE: |
|
100 |
+ val = sin(phs * 2 * M_PI); |
|
101 |
+ break; |
|
102 |
+ case TRIANGLE: |
|
103 |
+ if (phs > 0.75) |
|
104 |
+ val = (phs - 0.75) * 4 - 1; |
|
105 |
+ else if (phs > 0.25) |
|
106 |
+ val = -4 * phs + 2; |
|
107 |
+ else |
|
108 |
+ val = phs * 4; |
|
109 |
+ break; |
|
110 |
+ case SQUARE: |
|
111 |
+ val = phs < 0.5 ? -1 : +1; |
|
112 |
+ break; |
|
113 |
+ case SAWUP: |
|
114 |
+ val = phs * 2 - 1; |
|
115 |
+ break; |
|
116 |
+ case SAWDOWN: |
|
117 |
+ val = 1 - phs * 2; |
|
118 |
+ break; |
|
119 |
+ } |
|
120 |
+ |
|
121 |
+ return val * lfo->amount; |
|
122 |
+} |
|
123 |
+ |
|
124 |
+static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
|
125 |
+{ |
|
126 |
+ AVFilterContext *ctx = inlink->dst; |
|
127 |
+ AVFilterLink *outlink = ctx->outputs[0]; |
|
128 |
+ AudioPulsatorContext *s = ctx->priv; |
|
129 |
+ const double *src = (const double *)in->data[0]; |
|
130 |
+ const int nb_samples = in->nb_samples; |
|
131 |
+ const double level_out = s->level_out; |
|
132 |
+ const double level_in = s->level_in; |
|
133 |
+ const double amount = s->amount; |
|
134 |
+ AVFrame *out; |
|
135 |
+ double *dst; |
|
136 |
+ int n; |
|
137 |
+ |
|
138 |
+ if (av_frame_is_writable(in)) { |
|
139 |
+ out = in; |
|
140 |
+ } else { |
|
141 |
+ out = ff_get_audio_buffer(inlink, in->nb_samples); |
|
142 |
+ if (!out) { |
|
143 |
+ av_frame_free(&in); |
|
144 |
+ return AVERROR(ENOMEM); |
|
145 |
+ } |
|
146 |
+ av_frame_copy_props(out, in); |
|
147 |
+ } |
|
148 |
+ dst = (double *)out->data[0]; |
|
149 |
+ |
|
150 |
+ for (n = 0; n < nb_samples; n++) { |
|
151 |
+ double outL; |
|
152 |
+ double outR; |
|
153 |
+ double inL = src[0] * level_in; |
|
154 |
+ double inR = src[1] * level_in; |
|
155 |
+ double procL = inL; |
|
156 |
+ double procR = inR; |
|
157 |
+ |
|
158 |
+ procL *= lfo_get_value(&s->lfoL) * 0.5 + amount / 2; |
|
159 |
+ procR *= lfo_get_value(&s->lfoR) * 0.5 + amount / 2; |
|
160 |
+ |
|
161 |
+ outL = procL + inL * (1 - amount); |
|
162 |
+ outR = procR + inR * (1 - amount); |
|
163 |
+ |
|
164 |
+ outL *= level_out; |
|
165 |
+ outR *= level_out; |
|
166 |
+ |
|
167 |
+ dst[0] = outL; |
|
168 |
+ dst[1] = outR; |
|
169 |
+ |
|
170 |
+ lfo_advance(&s->lfoL, 1); |
|
171 |
+ lfo_advance(&s->lfoR, 1); |
|
172 |
+ |
|
173 |
+ dst += 2; |
|
174 |
+ src += 2; |
|
175 |
+ } |
|
176 |
+ |
|
177 |
+ if (in != out) |
|
178 |
+ av_frame_free(&in); |
|
179 |
+ |
|
180 |
+ return ff_filter_frame(outlink, out); |
|
181 |
+} |
|
182 |
+ |
|
183 |
+static int query_formats(AVFilterContext *ctx) |
|
184 |
+{ |
|
185 |
+ AVFilterChannelLayouts *layout = NULL; |
|
186 |
+ AVFilterFormats *formats = NULL; |
|
187 |
+ int ret; |
|
188 |
+ |
|
189 |
+ if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 || |
|
190 |
+ (ret = ff_set_common_formats (ctx , formats )) < 0 || |
|
191 |
+ (ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 || |
|
192 |
+ (ret = ff_set_common_channel_layouts (ctx , layout )) < 0) |
|
193 |
+ return ret; |
|
194 |
+ |
|
195 |
+ formats = ff_all_samplerates(); |
|
196 |
+ return ff_set_common_samplerates(ctx, formats); |
|
197 |
+} |
|
198 |
+ |
|
199 |
+static int config_input(AVFilterLink *inlink) |
|
200 |
+{ |
|
201 |
+ AVFilterContext *ctx = inlink->dst; |
|
202 |
+ AudioPulsatorContext *s = ctx->priv; |
|
203 |
+ double freq; |
|
204 |
+ |
|
205 |
+ switch (s->timing) { |
|
206 |
+ case UNIT_BPM: freq = s->bpm / 60; break; |
|
207 |
+ case UNIT_MS: freq = 1 / (s->ms / 1000.); break; |
|
208 |
+ case UNIT_HZ: freq = s->hz; break; |
|
209 |
+ } |
|
210 |
+ |
|
211 |
+ s->lfoL.freq = freq; |
|
212 |
+ s->lfoR.freq = freq; |
|
213 |
+ s->lfoL.mode = s->mode; |
|
214 |
+ s->lfoR.mode = s->mode; |
|
215 |
+ s->lfoL.offset = s->offset_l; |
|
216 |
+ s->lfoR.offset = s->offset_r; |
|
217 |
+ s->lfoL.srate = inlink->sample_rate; |
|
218 |
+ s->lfoR.srate = inlink->sample_rate; |
|
219 |
+ s->lfoL.amount = s->amount; |
|
220 |
+ s->lfoR.amount = s->amount; |
|
221 |
+ s->lfoL.pwidth = s->pwidth; |
|
222 |
+ s->lfoR.pwidth = s->pwidth; |
|
223 |
+ |
|
224 |
+ return 0; |
|
225 |
+} |
|
226 |
+ |
|
227 |
+static const AVFilterPad inputs[] = { |
|
228 |
+ { |
|
229 |
+ .name = "default", |
|
230 |
+ .type = AVMEDIA_TYPE_AUDIO, |
|
231 |
+ .config_props = config_input, |
|
232 |
+ .filter_frame = filter_frame, |
|
233 |
+ }, |
|
234 |
+ { NULL } |
|
235 |
+}; |
|
236 |
+ |
|
237 |
+static const AVFilterPad outputs[] = { |
|
238 |
+ { |
|
239 |
+ .name = "default", |
|
240 |
+ .type = AVMEDIA_TYPE_AUDIO, |
|
241 |
+ }, |
|
242 |
+ { NULL } |
|
243 |
+}; |
|
244 |
+ |
|
245 |
+AVFilter ff_af_apulsator = { |
|
246 |
+ .name = "apulsator", |
|
247 |
+ .description = NULL_IF_CONFIG_SMALL("Audio pulsator."), |
|
248 |
+ .priv_size = sizeof(AudioPulsatorContext), |
|
249 |
+ .priv_class = &apulsator_class, |
|
250 |
+ .query_formats = query_formats, |
|
251 |
+ .inputs = inputs, |
|
252 |
+ .outputs = outputs, |
|
253 |
+}; |
... | ... |
@@ -62,6 +62,7 @@ void avfilter_register_all(void) |
62 | 62 |
REGISTER_FILTER(APAD, apad, af); |
63 | 63 |
REGISTER_FILTER(APERMS, aperms, af); |
64 | 64 |
REGISTER_FILTER(APHASER, aphaser, af); |
65 |
+ REGISTER_FILTER(APULSATOR, apulsator, af); |
|
65 | 66 |
REGISTER_FILTER(AREALTIME, arealtime, af); |
66 | 67 |
REGISTER_FILTER(ARESAMPLE, aresample, af); |
67 | 68 |
REGISTER_FILTER(AREVERSE, areverse, af); |
... | ... |
@@ -30,7 +30,7 @@ |
30 | 30 |
#include "libavutil/version.h" |
31 | 31 |
|
32 | 32 |
#define LIBAVFILTER_VERSION_MAJOR 6 |
33 |
-#define LIBAVFILTER_VERSION_MINOR 17 |
|
33 |
+#define LIBAVFILTER_VERSION_MINOR 18 |
|
34 | 34 |
#define LIBAVFILTER_VERSION_MICRO 100 |
35 | 35 |
|
36 | 36 |
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ |