FFmpeg
af_superequalizer.c
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1 /*
2  * Copyright (c) 2002 Naoki Shibata
3  * Copyright (c) 2017 Paul B Mahol
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 #include "libavutil/mem.h"
23 #include "libavutil/opt.h"
24 #include "libavutil/tx.h"
25 
26 #include "audio.h"
27 #include "avfilter.h"
28 #include "filters.h"
29 #include "internal.h"
30 
31 #define NBANDS 17
32 #define M 15
33 
34 typedef struct EqParameter {
35  float lower, upper, gain;
36 } EqParameter;
37 
38 typedef struct SuperEqualizerContext {
39  const AVClass *class;
40 
42 
43  float gains[NBANDS + 1];
44 
45  float fact[M + 1];
46  float aa;
47  float iza;
48  float *ires, *irest;
51 
56 
57 static const float bands[] = {
58  65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
59  1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
60 };
61 
62 static float izero(SuperEqualizerContext *s, float x)
63 {
64  float ret = 1;
65  int m;
66 
67  for (m = 1; m <= M; m++) {
68  float t;
69 
70  t = pow(x / 2, m) / s->fact[m];
71  ret += t*t;
72  }
73 
74  return ret;
75 }
76 
77 static float hn_lpf(int n, float f, float fs)
78 {
79  float t = 1 / fs;
80  float omega = 2 * M_PI * f;
81 
82  if (n * omega * t == 0)
83  return 2 * f * t;
84  return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
85 }
86 
87 static float hn_imp(int n)
88 {
89  return n == 0 ? 1.f : 0.f;
90 }
91 
92 static float hn(int n, EqParameter *param, float fs)
93 {
94  float ret, lhn;
95  int i;
96 
97  lhn = hn_lpf(n, param[0].upper, fs);
98  ret = param[0].gain*lhn;
99 
100  for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
101  float lhn2 = hn_lpf(n, param[i].upper, fs);
102  ret += param[i].gain * (lhn2 - lhn);
103  lhn = lhn2;
104  }
105 
106  ret += param[i].gain * (hn_imp(n) - lhn);
107 
108  return ret;
109 }
110 
111 static float alpha(float a)
112 {
113  if (a <= 21)
114  return 0;
115  if (a <= 50)
116  return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
117  return .1102f * (a - 8.7f);
118 }
119 
120 static float win(SuperEqualizerContext *s, float n, int N)
121 {
122  return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
123 }
124 
125 static void process_param(float *bc, EqParameter *param, float fs)
126 {
127  int i;
128 
129  for (i = 0; i <= NBANDS; i++) {
130  param[i].lower = i == 0 ? 0 : bands[i - 1];
131  param[i].upper = i == NBANDS ? fs : bands[i];
132  param[i].gain = bc[i];
133  }
134 }
135 
136 static int equ_init(SuperEqualizerContext *s, int wb)
137 {
138  float scale = 1.f, iscale = 1.f;
139  int i, j, ret;
140 
141  ret = av_tx_init(&s->rdft, &s->tx_fn, AV_TX_FLOAT_RDFT, 0, 1 << wb, &scale, 0);
142  if (ret < 0)
143  return ret;
144 
145  ret = av_tx_init(&s->irdft, &s->itx_fn, AV_TX_FLOAT_RDFT, 1, 1 << wb, &iscale, 0);
146  if (ret < 0)
147  return ret;
148 
149  s->aa = 96;
150  s->winlen = (1 << (wb-1))-1;
151  s->tabsize = 1 << wb;
152 
153  s->ires = av_calloc(s->tabsize + 2, sizeof(float));
154  s->irest = av_calloc(s->tabsize, sizeof(float));
155  s->fsamples = av_calloc(s->tabsize, sizeof(float));
156  s->fsamples_out = av_calloc(s->tabsize + 2, sizeof(float));
157  if (!s->ires || !s->irest || !s->fsamples || !s->fsamples_out)
158  return AVERROR(ENOMEM);
159 
160  for (i = 0; i <= M; i++) {
161  s->fact[i] = 1;
162  for (j = 1; j <= i; j++)
163  s->fact[i] *= j;
164  }
165 
166  s->iza = izero(s, alpha(s->aa));
167 
168  return 0;
169 }
170 
171 static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
172 {
173  const int winlen = s->winlen;
174  const int tabsize = s->tabsize;
175  int i;
176 
177  if (fs <= 0)
178  return;
179 
180  process_param(lbc, param, fs);
181  for (i = 0; i < winlen; i++)
182  s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
183  for (; i < tabsize; i++)
184  s->irest[i] = 0;
185 
186  s->tx_fn(s->rdft, s->ires, s->irest, sizeof(float));
187 }
188 
190 {
191  AVFilterContext *ctx = inlink->dst;
192  SuperEqualizerContext *s = ctx->priv;
193  AVFilterLink *outlink = ctx->outputs[0];
194  const float *ires = s->ires;
195  float *fsamples_out = s->fsamples_out;
196  float *fsamples = s->fsamples;
197  int ch, i;
198 
199  AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
200  float *src, *dst, *ptr;
201 
202  if (!out) {
203  av_frame_free(&in);
204  return AVERROR(ENOMEM);
205  }
206 
207  for (ch = 0; ch < in->ch_layout.nb_channels; ch++) {
208  ptr = (float *)out->extended_data[ch];
209  dst = (float *)s->out->extended_data[ch];
210  src = (float *)in->extended_data[ch];
211 
212  for (i = 0; i < in->nb_samples; i++)
213  fsamples[i] = src[i];
214  for (; i < s->tabsize; i++)
215  fsamples[i] = 0;
216 
217  s->tx_fn(s->rdft, fsamples_out, fsamples, sizeof(float));
218 
219  for (i = 0; i <= s->tabsize / 2; i++) {
220  float re, im;
221 
222  re = ires[i*2 ] * fsamples_out[i*2] - ires[i*2+1] * fsamples_out[i*2+1];
223  im = ires[i*2+1] * fsamples_out[i*2] + ires[i*2 ] * fsamples_out[i*2+1];
224 
225  fsamples_out[i*2 ] = re;
226  fsamples_out[i*2+1] = im;
227  }
228 
229  s->itx_fn(s->irdft, fsamples, fsamples_out, sizeof(AVComplexFloat));
230 
231  for (i = 0; i < s->winlen; i++)
232  dst[i] += fsamples[i] / s->tabsize;
233  for (i = s->winlen; i < s->tabsize; i++)
234  dst[i] = fsamples[i] / s->tabsize;
235  for (i = 0; i < out->nb_samples; i++)
236  ptr[i] = dst[i];
237  for (i = 0; i < s->winlen; i++)
238  dst[i] = dst[i+s->winlen];
239  }
240 
241  out->pts = in->pts;
242  av_frame_free(&in);
243 
244  return ff_filter_frame(outlink, out);
245 }
246 
248 {
249  AVFilterLink *inlink = ctx->inputs[0];
250  AVFilterLink *outlink = ctx->outputs[0];
251  SuperEqualizerContext *s = ctx->priv;
252  AVFrame *in = NULL;
253  int ret;
254 
256 
257  ret = ff_inlink_consume_samples(inlink, s->winlen, s->winlen, &in);
258  if (ret < 0)
259  return ret;
260  if (ret > 0)
261  return filter_frame(inlink, in);
262 
265 
266  return FFERROR_NOT_READY;
267 }
268 
270 {
271  SuperEqualizerContext *s = ctx->priv;
272 
273  return equ_init(s, 14);
274 }
275 
277 {
278  AVFilterContext *ctx = inlink->dst;
279  SuperEqualizerContext *s = ctx->priv;
280 
281  s->out = ff_get_audio_buffer(inlink, s->tabsize);
282  if (!s->out)
283  return AVERROR(ENOMEM);
284 
285  return 0;
286 }
287 
288 static int config_output(AVFilterLink *outlink)
289 {
290  AVFilterContext *ctx = outlink->src;
291  SuperEqualizerContext *s = ctx->priv;
292 
293  make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
294 
295  return 0;
296 }
297 
299 {
300  SuperEqualizerContext *s = ctx->priv;
301 
302  av_frame_free(&s->out);
303  av_freep(&s->irest);
304  av_freep(&s->ires);
305  av_freep(&s->fsamples);
306  av_freep(&s->fsamples_out);
307  av_tx_uninit(&s->rdft);
308  av_tx_uninit(&s->irdft);
309 }
310 
312  {
313  .name = "default",
314  .type = AVMEDIA_TYPE_AUDIO,
315  .config_props = config_input,
316  },
317 };
318 
320  {
321  .name = "default",
322  .type = AVMEDIA_TYPE_AUDIO,
323  .config_props = config_output,
324  },
325 };
326 
327 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
328 #define OFFSET(x) offsetof(SuperEqualizerContext, x)
329 
331  { "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
332  { "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
333  { "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
334  { "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
335  { "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
336  { "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
337  { "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
338  { "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
339  { "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
340  { "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
341  { "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
342  { "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
343  { "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
344  { "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
345  { "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
346  { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
347  { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
348  { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
349  { NULL }
350 };
351 
352 AVFILTER_DEFINE_CLASS(superequalizer);
353 
355  .name = "superequalizer",
356  .description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
357  .priv_size = sizeof(SuperEqualizerContext),
358  .priv_class = &superequalizer_class,
359  .init = init,
360  .activate = activate,
361  .uninit = uninit,
365 };
hn_lpf
static float hn_lpf(int n, float f, float fs)
Definition: af_superequalizer.c:77
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:97
SuperEqualizerContext::fsamples
float * fsamples
Definition: af_superequalizer.c:49
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_superequalizer.c:298
superequalizer_options
static const AVOption superequalizer_options[]
Definition: af_superequalizer.c:330
SuperEqualizerContext::in
AVFrame * in
Definition: af_superequalizer.c:52
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
out
FILE * out
Definition: movenc.c:55
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1015
FFERROR_NOT_READY
return FFERROR_NOT_READY
Definition: filter_design.txt:204
alpha
static float alpha(float a)
Definition: af_superequalizer.c:111
AVTXContext
Definition: tx_priv.h:235
FILTER_SINGLE_SAMPLEFMT
#define FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
Definition: internal.h:175
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
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:160
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:374
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:486
AVOption
AVOption.
Definition: opt.h:357
SuperEqualizerContext
Definition: af_superequalizer.c:38
SuperEqualizerContext::params
EqParameter params[NBANDS+1]
Definition: af_superequalizer.c:41
AF
#define AF
Definition: af_superequalizer.c:327
AVComplexFloat
Definition: tx.h:27
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:170
ff_af_superequalizer
const AVFilter ff_af_superequalizer
Definition: af_superequalizer.c:354
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:321
make_fir
static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
Definition: af_superequalizer.c:171
FF_FILTER_FORWARD_STATUS_BACK
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
Definition: filters.h:199
SuperEqualizerContext::irest
float * irest
Definition: af_superequalizer.c:48
av_tx_init
av_cold int av_tx_init(AVTXContext **ctx, av_tx_fn *tx, enum AVTXType type, int inv, int len, const void *scale, uint64_t flags)
Initialize a transform context with the given configuration (i)MDCTs with an odd length are currently...
Definition: tx.c:903
win
static float win(SuperEqualizerContext *s, float n, int N)
Definition: af_superequalizer.c:120
SuperEqualizerContext::ires
float * ires
Definition: af_superequalizer.c:48
SuperEqualizerContext::tx_fn
av_tx_fn tx_fn
Definition: af_superequalizer.c:54
AVFrame::ch_layout
AVChannelLayout ch_layout
Channel layout of the audio data.
Definition: frame.h:775
SuperEqualizerContext::fact
float fact[M+1]
Definition: af_superequalizer.c:45
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:33
av_cold
#define av_cold
Definition: attributes.h:90
av_tx_fn
void(* av_tx_fn)(AVTXContext *s, void *out, void *in, ptrdiff_t stride)
Function pointer to a function to perform the transform.
Definition: tx.h:151
SuperEqualizerContext::winlen
int winlen
Definition: af_superequalizer.c:50
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_superequalizer.c:269
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_superequalizer.c:288
NBANDS
#define NBANDS
Definition: af_superequalizer.c:31
s
#define s(width, name)
Definition: cbs_vp9.c:198
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
process_param
static void process_param(float *bc, EqParameter *param, float fs)
Definition: af_superequalizer.c:125
filters.h
ctx
AVFormatContext * ctx
Definition: movenc.c:49
bands
static const float bands[]
Definition: af_superequalizer.c:57
superequalizer_outputs
static const AVFilterPad superequalizer_outputs[]
Definition: af_superequalizer.c:319
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: internal.h:182
izero
static float izero(SuperEqualizerContext *s, float x)
Definition: af_superequalizer.c:62
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
ff_inlink_consume_samples
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link's FIFO and update the link's stats.
Definition: avfilter.c:1462
NULL
#define NULL
Definition: coverity.c:32
fs
#define fs(width, name, subs,...)
Definition: cbs_vp9.c:200
OFFSET
#define OFFSET(x)
Definition: af_superequalizer.c:328
SuperEqualizerContext::out
AVFrame * out
Definition: af_superequalizer.c:52
sqrtf
static __device__ float sqrtf(float a)
Definition: cuda_runtime.h:184
sinf
#define sinf(x)
Definition: libm.h:419
f
f
Definition: af_crystalizer.c:121
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:94
hn
static float hn(int n, EqParameter *param, float fs)
Definition: af_superequalizer.c:92
EqParameter::gain
float gain
Definition: af_superequalizer.c:35
for
for(k=2;k<=8;++k)
Definition: h264pred_template.c:425
SuperEqualizerContext::tabsize
int tabsize
Definition: af_superequalizer.c:50
EqParameter::lower
float lower
Definition: af_superequalizer.c:35
equ_init
static int equ_init(SuperEqualizerContext *s, int wb)
Definition: af_superequalizer.c:136
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
SuperEqualizerContext::iza
float iza
Definition: af_superequalizer.c:47
FF_FILTER_FORWARD_WANTED
FF_FILTER_FORWARD_WANTED(outlink, inlink)
N
#define N
Definition: af_mcompand.c:54
superequalizer_inputs
static const AVFilterPad superequalizer_inputs[]
Definition: af_superequalizer.c:311
M_PI
#define M_PI
Definition: mathematics.h:67
av_tx_uninit
av_cold void av_tx_uninit(AVTXContext **ctx)
Frees a context and sets *ctx to NULL, does nothing when *ctx == NULL.
Definition: tx.c:295
SuperEqualizerContext::rdft
AVTXContext * rdft
Definition: af_superequalizer.c:53
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:248
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:454
EqParameter::upper
float upper
Definition: af_superequalizer.c:35
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:256
M
#define M
Definition: af_superequalizer.c:32
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:435
SuperEqualizerContext::gains
float gains[NBANDS+1]
Definition: af_superequalizer.c:43
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:39
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:264
SuperEqualizerContext::itx_fn
av_tx_fn itx_fn
Definition: af_superequalizer.c:54
AVFilter
Filter definition.
Definition: avfilter.h:166
ret
ret
Definition: filter_design.txt:187
AV_TX_FLOAT_RDFT
@ AV_TX_FLOAT_RDFT
Real to complex and complex to real DFTs.
Definition: tx.h:90
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_superequalizer.c:276
SuperEqualizerContext::aa
float aa
Definition: af_superequalizer.c:46
avfilter.h
SuperEqualizerContext::fsamples_out
float * fsamples_out
Definition: af_superequalizer.c:49
AVFilterContext
An instance of a filter.
Definition: avfilter.h:407
SuperEqualizerContext::irdft
AVTXContext * irdft
Definition: af_superequalizer.c:53
mem.h
audio.h
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_superequalizer.c:189
scale
static void scale(int *out, const int *in, const int w, const int h, const int shift)
Definition: intra.c:291
FF_FILTER_FORWARD_STATUS
FF_FILTER_FORWARD_STATUS(inlink, outlink)
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:183
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:34
src
INIT_CLIP pixel * src
Definition: h264pred_template.c:418
EqParameter
Definition: af_superequalizer.c:34
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(superequalizer)
hn_imp
static float hn_imp(int n)
Definition: af_superequalizer.c:87
activate
static int activate(AVFilterContext *ctx)
Definition: af_superequalizer.c:247
tx.h