FFmpeg
af_adynamicequalizer.c
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1 /*
2  * This file is part of FFmpeg.
3  *
4  * FFmpeg is free software; you can redistribute it and/or
5  * modify it under the terms of the GNU Lesser General Public
6  * License as published by the Free Software Foundation; either
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8  *
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12  * Lesser General Public License for more details.
13  *
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15  * License along with FFmpeg; if not, write to the Free Software
16  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17  */
18 
19 #include <float.h>
20 
21 #include "libavutil/opt.h"
22 #include "avfilter.h"
23 #include "audio.h"
24 #include "formats.h"
25 
27  const AVClass *class;
28 
29  double threshold;
30  double dfrequency;
31  double dqfactor;
32  double tfrequency;
33  double tqfactor;
34  double ratio;
35  double range;
36  double makeup;
37  double attack;
38  double release;
39  double attack_coef;
40  double release_coef;
41  int mode;
42  int direction;
43  int type;
44 
47 
49 {
50  AVFilterContext *ctx = inlink->dst;
52 
53  s->state = ff_get_audio_buffer(inlink, 8);
54  if (!s->state)
55  return AVERROR(ENOMEM);
56 
57  for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
58  double *state = (double *)s->state->extended_data[ch];
59 
60  state[4] = 1.;
61  }
62 
63  return 0;
64 }
65 
66 static double get_svf(double in, double *m, double *a, double *b)
67 {
68  const double v0 = in;
69  const double v3 = v0 - b[1];
70  const double v1 = a[0] * b[0] + a[1] * v3;
71  const double v2 = b[1] + a[1] * b[0] + a[2] * v3;
72 
73  b[0] = 2. * v1 - b[0];
74  b[1] = 2. * v2 - b[1];
75 
76  return m[0] * v0 + m[1] * v1 + m[2] * v2;
77 }
78 
79 typedef struct ThreadData {
80  AVFrame *in, *out;
81 } ThreadData;
82 
83 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
84 {
86  ThreadData *td = arg;
87  AVFrame *in = td->in;
88  AVFrame *out = td->out;
89  const double sample_rate = in->sample_rate;
90  const double makeup = s->makeup;
91  const double ratio = s->ratio;
92  const double range = s->range;
93  const double dfrequency = fmin(s->dfrequency, sample_rate * 0.5);
94  const double tfrequency = fmin(s->tfrequency, sample_rate * 0.5);
95  const double threshold = s->threshold;
96  const double release = s->release_coef;
97  const double irelease = 1. - release;
98  const double attack = s->attack_coef;
99  const double iattack = 1. - attack;
100  const double dqfactor = s->dqfactor;
101  const double tqfactor = s->tqfactor;
102  const double fg = tan(M_PI * tfrequency / sample_rate);
103  const double dg = tan(M_PI * dfrequency / sample_rate);
104  const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
105  const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
106  const int direction = s->direction;
107  const int mode = s->mode;
108  const int type = s->type;
109  double da[3], dm[3];
110 
111  {
112  double k = 1. / dqfactor;
113 
114  da[0] = 1. / (1. + dg * (dg + k));
115  da[1] = dg * da[0];
116  da[2] = dg * da[1];
117 
118  dm[0] = 0.;
119  dm[1] = k;
120  dm[2] = 0.;
121  }
122 
123  for (int ch = start; ch < end; ch++) {
124  const double *src = (const double *)in->extended_data[ch];
125  double *dst = (double *)out->extended_data[ch];
126  double *state = (double *)s->state->extended_data[ch];
127 
128  for (int n = 0; n < out->nb_samples; n++) {
129  double detect, gain, v, listen;
130  double fa[3], fm[3];
131  double k, g;
132 
133  detect = listen = get_svf(src[n], dm, da, state);
134  detect = fabs(detect);
135 
136  if (direction == 0 && mode == 0 && detect < threshold)
137  detect = 1. / av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
138  else if (direction == 0 && mode == 1 && detect < threshold)
139  detect = av_clipd(1. + makeup + (threshold - detect) * ratio, 1., range);
140  else if (direction == 1 && mode == 0 && detect > threshold)
141  detect = 1. / av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
142  else if (direction == 1 && mode == 1 && detect > threshold)
143  detect = av_clipd(1. + makeup + (detect - threshold) * ratio, 1., range);
144  else
145  detect = 1.;
146 
147  if (detect < state[4]) {
148  detect = iattack * detect + attack * state[4];
149  } else {
150  detect = irelease * detect + release * state[4];
151  }
152 
153  if (state[4] != detect || n == 0) {
154  state[4] = gain = detect;
155 
156  switch (type) {
157  case 0:
158  k = 1. / (tqfactor * gain);
159 
160  fa[0] = 1. / (1. + fg * (fg + k));
161  fa[1] = fg * fa[0];
162  fa[2] = fg * fa[1];
163 
164  fm[0] = 1.;
165  fm[1] = k * (gain * gain - 1.);
166  fm[2] = 0.;
167  break;
168  case 1:
169  k = 1. / tqfactor;
170  g = fg / sqrt(gain);
171 
172  fa[0] = 1. / (1. + g * (g + k));
173  fa[1] = g * fa[0];
174  fa[2] = g * fa[1];
175 
176  fm[0] = 1.;
177  fm[1] = k * (gain - 1.);
178  fm[2] = gain * gain - 1.;
179  break;
180  case 2:
181  k = 1. / tqfactor;
182  g = fg / sqrt(gain);
183 
184  fa[0] = 1. / (1. + g * (g + k));
185  fa[1] = g * fa[0];
186  fa[2] = g * fa[1];
187 
188  fm[0] = gain * gain;
189  fm[1] = k * (1. - gain) * gain;
190  fm[2] = 1. - gain * gain;
191  break;
192  }
193  }
194 
195  v = get_svf(src[n], fm, fa, &state[2]);
196  v = mode == -1 ? listen : v;
197  dst[n] = ctx->is_disabled ? src[n] : v;
198  }
199  }
200 
201  return 0;
202 }
203 
204 static double get_coef(double x, double sr)
205 {
206  return exp(-1000. / (x * sr));
207 }
208 
210 {
211  AVFilterContext *ctx = inlink->dst;
212  AVFilterLink *outlink = ctx->outputs[0];
214  ThreadData td;
215  AVFrame *out;
216 
217  if (av_frame_is_writable(in)) {
218  out = in;
219  } else {
220  out = ff_get_audio_buffer(outlink, in->nb_samples);
221  if (!out) {
222  av_frame_free(&in);
223  return AVERROR(ENOMEM);
224  }
226  }
227 
228  s->attack_coef = get_coef(s->attack, in->sample_rate);
229  s->release_coef = get_coef(s->release, in->sample_rate);
230 
231  td.in = in;
232  td.out = out;
235 
236  if (out != in)
237  av_frame_free(&in);
238  return ff_filter_frame(outlink, out);
239 }
240 
242 {
244 
245  av_frame_free(&s->state);
246 }
247 
248 #define OFFSET(x) offsetof(AudioDynamicEqualizerContext, x)
249 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
250 
252  { "threshold", "set detection threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
253  { "dfrequency", "set detection frequency", OFFSET(dfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
254  { "dqfactor", "set detection Q factor", OFFSET(dqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
255  { "tfrequency", "set target frequency", OFFSET(tfrequency), AV_OPT_TYPE_DOUBLE, {.dbl=1000}, 2, 1000000, FLAGS },
256  { "tqfactor", "set target Q factor", OFFSET(tqfactor), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.001, 1000, FLAGS },
257  { "attack", "set attack duration", OFFSET(attack), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 2000, FLAGS },
258  { "release", "set release duration", OFFSET(release), AV_OPT_TYPE_DOUBLE, {.dbl=200}, 1, 2000, FLAGS },
259  { "ratio", "set ratio factor", OFFSET(ratio), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 30, FLAGS },
260  { "makeup", "set makeup gain", OFFSET(makeup), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 100, FLAGS },
261  { "range", "set max gain", OFFSET(range), AV_OPT_TYPE_DOUBLE, {.dbl=50}, 1, 200, FLAGS },
262  { "mode", "set mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, -1, 1, FLAGS, "mode" },
263  { "listen", 0, 0, AV_OPT_TYPE_CONST, {.i64=-1}, 0, 0, FLAGS, "mode" },
264  { "cut", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
265  { "boost", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
266  { "tftype", "set target filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, FLAGS, "type" },
267  { "bell", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
268  { "lowshelf", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
269  { "highshelf",0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
270  { "direction", "set direction", OFFSET(direction), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "direction" },
271  { "downward", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "direction" },
272  { "upward", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "direction" },
273  { NULL }
274 };
275 
276 AVFILTER_DEFINE_CLASS(adynamicequalizer);
277 
278 static const AVFilterPad inputs[] = {
279  {
280  .name = "default",
281  .type = AVMEDIA_TYPE_AUDIO,
282  .filter_frame = filter_frame,
283  .config_props = config_input,
284  },
285 };
286 
287 static const AVFilterPad outputs[] = {
288  {
289  .name = "default",
290  .type = AVMEDIA_TYPE_AUDIO,
291  },
292 };
293 
295  .name = "adynamicequalizer",
296  .description = NULL_IF_CONFIG_SMALL("Apply Dynamic Equalization of input audio."),
297  .priv_size = sizeof(AudioDynamicEqualizerContext),
298  .priv_class = &adynamicequalizer_class,
299  .uninit = uninit,
305  .process_command = ff_filter_process_command,
306 };
ff_get_audio_buffer
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:100
td
#define td
Definition: regdef.h:70
AudioDynamicEqualizerContext::dfrequency
double dfrequency
Definition: af_adynamicequalizer.c:30
AVERROR
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later. That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another. Frame references ownership and permissions
opt.h
AudioDynamicEqualizerContext::range
double range
Definition: af_adynamicequalizer.c:35
out
FILE * out
Definition: movenc.c:54
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1009
FILTER_SINGLE_SAMPLEFMT
#define FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
Definition: internal.h:183
inlink
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
Definition: filter_design.txt:212
ff_af_adynamicequalizer
const AVFilter ff_af_adynamicequalizer
Definition: af_adynamicequalizer.c:294
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:116
AudioDynamicEqualizerContext::makeup
double makeup
Definition: af_adynamicequalizer.c:36
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:325
AudioDynamicEqualizerContext::type
int type
Definition: af_adynamicequalizer.c:43
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_adynamicequalizer.c:48
AVOption
AVOption.
Definition: opt.h:251
b
#define b
Definition: input.c:41
FLAGS
#define FLAGS
Definition: af_adynamicequalizer.c:249
float.h
filter_channels
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_adynamicequalizer.c:83
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:175
AudioDynamicEqualizerContext::mode
int mode
Definition: af_adynamicequalizer.c:41
AudioDynamicEqualizerContext::state
AVFrame * state
Definition: af_adynamicequalizer.c:45
ThreadData::out
AVFrame * out
Definition: af_adeclick.c:473
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:311
ThreadData::in
AVFrame * in
Definition: af_adecorrelate.c:154
get_coef
static double get_coef(double x, double sr)
Definition: af_adynamicequalizer.c:204
sample_rate
sample_rate
Definition: ffmpeg_filter.c:156
formats.h
v0
#define v0
Definition: regdef.h:26
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_adynamicequalizer.c:209
AVFrame::ch_layout
AVChannelLayout ch_layout
Channel layout of the audio data.
Definition: frame.h:709
type
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf type
Definition: writing_filters.txt:86
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_adynamicequalizer.c:241
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:49
get_svf
static double get_svf(double in, double *m, double *a, double *b)
Definition: af_adynamicequalizer.c:66
AudioDynamicEqualizerContext::release_coef
double release_coef
Definition: af_adynamicequalizer.c:40
av_cold
#define av_cold
Definition: attributes.h:90
adynamicequalizer_options
static const AVOption adynamicequalizer_options[]
Definition: af_adynamicequalizer.c:251
s
#define s(width, name)
Definition: cbs_vp9.c:256
g
const char * g
Definition: vf_curves.c:127
AV_OPT_TYPE_DOUBLE
@ AV_OPT_TYPE_DOUBLE
Definition: opt.h:227
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
ctx
AVFormatContext * ctx
Definition: movenc.c:48
state
static struct @344 state
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: internal.h:190
arg
const char * arg
Definition: jacosubdec.c:67
AudioDynamicEqualizerContext::tqfactor
double tqfactor
Definition: af_adynamicequalizer.c:33
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
fabs
static __device__ float fabs(float a)
Definition: cuda_runtime.h:182
NULL
#define NULL
Definition: coverity.c:32
av_frame_copy_props
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:603
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(adynamicequalizer)
exp
int8_t exp
Definition: eval.c:72
outputs
static const AVFilterPad outputs[]
Definition: af_adynamicequalizer.c:287
NULL_IF_CONFIG_SMALL
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:115
OFFSET
#define OFFSET(x)
Definition: af_adynamicequalizer.c:248
AVFrame::sample_rate
int sample_rate
Sample rate of the audio data.
Definition: frame.h:502
for
for(k=2;k<=8;++k)
Definition: h264pred_template.c:425
fmin
double fmin(double, double)
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:533
AudioDynamicEqualizerContext
Definition: af_adynamicequalizer.c:26
ff_filter_process_command
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options.
Definition: avfilter.c:873
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
M_PI
#define M_PI
Definition: mathematics.h:52
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:405
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:386
ff_filter_get_nb_threads
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:793
ThreadData
Used for passing data between threads.
Definition: dsddec.c:69
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:55
AudioDynamicEqualizerContext::ratio
double ratio
Definition: af_adynamicequalizer.c:34
AVFilter
Filter definition.
Definition: avfilter.h:171
AudioDynamicEqualizerContext::attack
double attack
Definition: af_adynamicequalizer.c:37
AudioDynamicEqualizerContext::tfrequency
double tfrequency
Definition: af_adynamicequalizer.c:32
mode
mode
Definition: ebur128.h:83
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:225
avfilter.h
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:67
AudioDynamicEqualizerContext::dqfactor
double dqfactor
Definition: af_adynamicequalizer.c:31
AudioDynamicEqualizerContext::release
double release
Definition: af_adynamicequalizer.c:38
AVFilterContext
An instance of a filter.
Definition: avfilter.h:415
AVFILTER_FLAG_SLICE_THREADS
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:127
audio.h
AudioDynamicEqualizerContext::threshold
double threshold
Definition: af_adynamicequalizer.c:29
AudioDynamicEqualizerContext::attack_coef
double attack_coef
Definition: af_adynamicequalizer.c:39
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:191
src
INIT_CLIP pixel * src
Definition: h264pred_template.c:418
AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
Definition: avfilter.h:160
inputs
static const AVFilterPad inputs[]
Definition: af_adynamicequalizer.c:278
ff_filter_execute
static av_always_inline int ff_filter_execute(AVFilterContext *ctx, avfilter_action_func *func, void *arg, int *ret, int nb_jobs)
Definition: internal.h:142
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
av_clipd
av_clipd
Definition: af_crystalizer.c:132
AudioDynamicEqualizerContext::direction
int direction
Definition: af_adynamicequalizer.c:42