forked from FFmpeg/FFmpeg
-
Notifications
You must be signed in to change notification settings - Fork 0
/
Copy pathaf_adynamicequalizer.c
286 lines (252 loc) · 11.2 KB
/
af_adynamicequalizer.c
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
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
/*
* 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 <float.h>
#include "libavutil/ffmath.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
enum DetectionModes {
DET_UNSET = 0,
DET_DISABLED,
DET_OFF,
DET_ON,
DET_ADAPTIVE,
NB_DMODES,
};
enum FilterModes {
LISTEN = -1,
CUT_BELOW,
CUT_ABOVE,
BOOST_BELOW,
BOOST_ABOVE,
NB_FMODES,
};
typedef struct ChannelContext {
double fa_double[3], fm_double[3];
double dstate_double[2];
double fstate_double[2];
double tstate_double[2];
double lin_gain_double;
double detect_double;
double threshold_log_double;
double new_threshold_log_double;
double log_sum_double;
double sum_double;
float fa_float[3], fm_float[3];
float dstate_float[2];
float fstate_float[2];
float tstate_float[2];
float lin_gain_float;
float detect_float;
float threshold_log_float;
float new_threshold_log_float;
float log_sum_float;
float sum_float;
void *dqueue;
void *queue;
int position;
int size;
int front;
int back;
int detection;
int init;
} ChannelContext;
typedef struct AudioDynamicEqualizerContext {
const AVClass *class;
double threshold;
double threshold_log;
double dfrequency;
double dqfactor;
double tfrequency;
double tqfactor;
double ratio;
double range;
double makeup;
double dattack;
double drelease;
double dattack_coef;
double drelease_coef;
double gattack_coef;
double grelease_coef;
int mode;
int detection;
int tftype;
int dftype;
int precision;
int format;
int nb_channels;
int (*filter_prepare)(AVFilterContext *ctx);
int (*filter_channels)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
double da_double[3], dm_double[3];
float da_float[3], dm_float[3];
ChannelContext *cc;
} AudioDynamicEqualizerContext;
static int query_formats(AVFilterContext *ctx)
{
AudioDynamicEqualizerContext *s = ctx->priv;
static const enum AVSampleFormat sample_fmts[3][3] = {
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE },
{ AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_NONE },
};
int ret;
if ((ret = ff_set_common_all_channel_counts(ctx)) < 0)
return ret;
if ((ret = ff_set_common_formats_from_list(ctx, sample_fmts[s->precision])) < 0)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static double get_coef(double x, double sr)
{
return 1.0 - exp(-1.0 / (0.001 * x * sr));
}
typedef struct ThreadData {
AVFrame *in, *out;
} ThreadData;
#define DEPTH 32
#include "adynamicequalizer_template.c"
#undef DEPTH
#define DEPTH 64
#include "adynamicequalizer_template.c"
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioDynamicEqualizerContext *s = ctx->priv;
s->format = inlink->format;
s->cc = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->cc));
if (!s->cc)
return AVERROR(ENOMEM);
s->nb_channels = inlink->ch_layout.nb_channels;
switch (s->format) {
case AV_SAMPLE_FMT_DBLP:
s->filter_prepare = filter_prepare_double;
s->filter_channels = filter_channels_double;
break;
case AV_SAMPLE_FMT_FLTP:
s->filter_prepare = filter_prepare_float;
s->filter_channels = filter_channels_float;
break;
}
for (int ch = 0; ch < s->nb_channels; ch++) {
ChannelContext *cc = &s->cc[ch];
cc->queue = av_calloc(inlink->sample_rate, sizeof(double));
cc->dqueue = av_calloc(inlink->sample_rate, sizeof(double));
if (!cc->queue || !cc->dqueue)
return AVERROR(ENOMEM);
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioDynamicEqualizerContext *s = ctx->priv;
ThreadData td;
AVFrame *out;
if (av_frame_is_writable(in)) {
out = in;
} else {
out = ff_get_audio_buffer(outlink, in->nb_samples);
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
}
td.in = in;
td.out = out;
s->filter_prepare(ctx);
ff_filter_execute(ctx, s->filter_channels, &td, NULL,
FFMIN(outlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
if (out != in)
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioDynamicEqualizerContext *s = ctx->priv;
for (int ch = 0; ch < s->nb_channels; ch++) {
ChannelContext *cc = &s->cc[ch];
av_freep(&cc->queue);
av_freep(&cc->dqueue);
}
av_freep(&s->cc);
}
#define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption adynamicequalizer_options[] = {
{ "threshold", "set detection threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
{ "dfrequency", "set detection frequency", OFFSET(dfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
{ "dqfactor", "set detection Q factor", OFFSET(dqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
{ "tfrequency", "set target frequency", OFFSET(tfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
{ "tqfactor", "set target Q factor", OFFSET(tqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
{ "attack", "set detection attack duration", OFFSET(dattack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 0.01, 2000, FLAGS },
{ "release","set detection release duration",OFFSET(drelease), AV_OPT_TYPE_DOUBLE, {.dbl=200}, 0.01, 2000, FLAGS },
{ "ratio", "set ratio factor", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 30, FLAGS },
{ "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1000, FLAGS },
{ "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 2000, FLAGS },
{ "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, LISTEN,NB_FMODES-1,FLAGS, .unit = "mode" },
{ "listen", 0, 0, AV_OPT_TYPE_CONST, {.i64=LISTEN}, 0, 0, FLAGS, .unit = "mode" },
{ "cutbelow", 0, 0, AV_OPT_TYPE_CONST, {.i64=CUT_BELOW},0, 0, FLAGS, .unit = "mode" },
{ "cutabove", 0, 0, AV_OPT_TYPE_CONST, {.i64=CUT_ABOVE},0, 0, FLAGS, .unit = "mode" },
{ "boostbelow", 0, 0, AV_OPT_TYPE_CONST, {.i64=BOOST_BELOW},0, 0, FLAGS, .unit = "mode" },
{ "boostabove", 0, 0, AV_OPT_TYPE_CONST, {.i64=BOOST_ABOVE},0, 0, FLAGS, .unit = "mode" },
{ "dftype", "set detection filter type",OFFSET(dftype), AV_OPT_TYPE_INT, {.i64=0}, 0, 3, FLAGS, .unit = "dftype" },
{ "bandpass", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "dftype" },
{ "lowpass", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "dftype" },
{ "highpass", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, .unit = "dftype" },
{ "peak", 0, 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, .unit = "dftype" },
{ "tftype", "set target filter type", OFFSET(tftype), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, FLAGS, .unit = "tftype" },
{ "bell", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, .unit = "tftype" },
{ "lowshelf", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, .unit = "tftype" },
{ "highshelf",0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, .unit = "tftype" },
{ "auto", "set auto threshold", OFFSET(detection), AV_OPT_TYPE_INT, {.i64=DET_OFF},DET_DISABLED,NB_DMODES-1,FLAGS, .unit = "auto" },
{ "disabled", 0, 0, AV_OPT_TYPE_CONST, {.i64=DET_DISABLED}, 0, 0, FLAGS, .unit = "auto" },
{ "off", 0, 0, AV_OPT_TYPE_CONST, {.i64=DET_OFF}, 0, 0, FLAGS, .unit = "auto" },
{ "on", 0, 0, AV_OPT_TYPE_CONST, {.i64=DET_ON}, 0, 0, FLAGS, .unit = "auto" },
{ "adaptive", 0, 0, AV_OPT_TYPE_CONST, {.i64=DET_ADAPTIVE}, 0, 0, FLAGS, .unit = "auto" },
{ "precision", "set processing precision", OFFSET(precision), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, AF, .unit = "precision" },
{ "auto", "set auto processing precision", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, .unit = "precision" },
{ "float", "set single-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, .unit = "precision" },
{ "double","set double-floating point processing precision", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, .unit = "precision" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(adynamicequalizer);
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
};
const AVFilter ff_af_adynamicequalizer = {
.name = "adynamicequalizer",
.description = NULL_IF_CONFIG_SMALL("Apply Dynamic Equalization of input audio."),
.priv_size = sizeof(AudioDynamicEqualizerContext),
.priv_class = &adynamicequalizer_class,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(ff_audio_default_filterpad),
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
AVFILTER_FLAG_SLICE_THREADS,
.process_command = ff_filter_process_command,
};