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/opt.h"
23 #include "libavutil/tx.h"
24 
25 #include "audio.h"
26 #include "avfilter.h"
27 #include "filters.h"
28 #include "internal.h"
29 
30 #define NBANDS 17
31 #define M 15
32 
33 typedef struct EqParameter {
34  float lower, upper, gain;
35 } EqParameter;
36 
37 typedef struct SuperEqualizerContext {
38  const AVClass *class;
39 
41 
42  float gains[NBANDS + 1];
43 
44  float fact[M + 1];
45  float aa;
46  float iza;
47  float *ires, *irest;
50 
55 
56 static const float bands[] = {
57  65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
58  1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
59 };
60 
61 static float izero(SuperEqualizerContext *s, float x)
62 {
63  float ret = 1;
64  int m;
65 
66  for (m = 1; m <= M; m++) {
67  float t;
68 
69  t = pow(x / 2, m) / s->fact[m];
70  ret += t*t;
71  }
72 
73  return ret;
74 }
75 
76 static float hn_lpf(int n, float f, float fs)
77 {
78  float t = 1 / fs;
79  float omega = 2 * M_PI * f;
80 
81  if (n * omega * t == 0)
82  return 2 * f * t;
83  return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
84 }
85 
86 static float hn_imp(int n)
87 {
88  return n == 0 ? 1.f : 0.f;
89 }
90 
91 static float hn(int n, EqParameter *param, float fs)
92 {
93  float ret, lhn;
94  int i;
95 
96  lhn = hn_lpf(n, param[0].upper, fs);
97  ret = param[0].gain*lhn;
98 
99  for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
100  float lhn2 = hn_lpf(n, param[i].upper, fs);
101  ret += param[i].gain * (lhn2 - lhn);
102  lhn = lhn2;
103  }
104 
105  ret += param[i].gain * (hn_imp(n) - lhn);
106 
107  return ret;
108 }
109 
110 static float alpha(float a)
111 {
112  if (a <= 21)
113  return 0;
114  if (a <= 50)
115  return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
116  return .1102f * (a - 8.7f);
117 }
118 
119 static float win(SuperEqualizerContext *s, float n, int N)
120 {
121  return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
122 }
123 
124 static void process_param(float *bc, EqParameter *param, float fs)
125 {
126  int i;
127 
128  for (i = 0; i <= NBANDS; i++) {
129  param[i].lower = i == 0 ? 0 : bands[i - 1];
130  param[i].upper = i == NBANDS ? fs : bands[i];
131  param[i].gain = bc[i];
132  }
133 }
134 
135 static int equ_init(SuperEqualizerContext *s, int wb)
136 {
137  float scale = 1.f, iscale = 1.f;
138  int i, j, ret;
139 
140  ret = av_tx_init(&s->rdft, &s->tx_fn, AV_TX_FLOAT_RDFT, 0, 1 << wb, &scale, 0);
141  if (ret < 0)
142  return ret;
143 
144  ret = av_tx_init(&s->irdft, &s->itx_fn, AV_TX_FLOAT_RDFT, 1, 1 << wb, &iscale, 0);
145  if (ret < 0)
146  return ret;
147 
148  s->aa = 96;
149  s->winlen = (1 << (wb-1))-1;
150  s->tabsize = 1 << wb;
151 
152  s->ires = av_calloc(s->tabsize + 2, sizeof(float));
153  s->irest = av_calloc(s->tabsize, sizeof(float));
154  s->fsamples = av_calloc(s->tabsize, sizeof(float));
155  s->fsamples_out = av_calloc(s->tabsize + 2, sizeof(float));
156  if (!s->ires || !s->irest || !s->fsamples || !s->fsamples_out)
157  return AVERROR(ENOMEM);
158 
159  for (i = 0; i <= M; i++) {
160  s->fact[i] = 1;
161  for (j = 1; j <= i; j++)
162  s->fact[i] *= j;
163  }
164 
165  s->iza = izero(s, alpha(s->aa));
166 
167  return 0;
168 }
169 
170 static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
171 {
172  const int winlen = s->winlen;
173  const int tabsize = s->tabsize;
174  int i;
175 
176  if (fs <= 0)
177  return;
178 
179  process_param(lbc, param, fs);
180  for (i = 0; i < winlen; i++)
181  s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
182  for (; i < tabsize; i++)
183  s->irest[i] = 0;
184 
185  s->tx_fn(s->rdft, s->ires, s->irest, sizeof(float));
186 }
187 
189 {
190  AVFilterContext *ctx = inlink->dst;
191  SuperEqualizerContext *s = ctx->priv;
192  AVFilterLink *outlink = ctx->outputs[0];
193  const float *ires = s->ires;
194  float *fsamples_out = s->fsamples_out;
195  float *fsamples = s->fsamples;
196  int ch, i;
197 
198  AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
199  float *src, *dst, *ptr;
200 
201  if (!out) {
202  av_frame_free(&in);
203  return AVERROR(ENOMEM);
204  }
205 
206  for (ch = 0; ch < in->ch_layout.nb_channels; ch++) {
207  ptr = (float *)out->extended_data[ch];
208  dst = (float *)s->out->extended_data[ch];
209  src = (float *)in->extended_data[ch];
210 
211  for (i = 0; i < in->nb_samples; i++)
212  fsamples[i] = src[i];
213  for (; i < s->tabsize; i++)
214  fsamples[i] = 0;
215 
216  s->tx_fn(s->rdft, fsamples_out, fsamples, sizeof(float));
217 
218  for (i = 0; i <= s->tabsize / 2; i++) {
219  float re, im;
220 
221  re = ires[i*2 ] * fsamples_out[i*2] - ires[i*2+1] * fsamples_out[i*2+1];
222  im = ires[i*2+1] * fsamples_out[i*2] + ires[i*2 ] * fsamples_out[i*2+1];
223 
224  fsamples_out[i*2 ] = re;
225  fsamples_out[i*2+1] = im;
226  }
227 
228  s->itx_fn(s->irdft, fsamples, fsamples_out, sizeof(AVComplexFloat));
229 
230  for (i = 0; i < s->winlen; i++)
231  dst[i] += fsamples[i] / s->tabsize;
232  for (i = s->winlen; i < s->tabsize; i++)
233  dst[i] = fsamples[i] / s->tabsize;
234  for (i = 0; i < out->nb_samples; i++)
235  ptr[i] = dst[i];
236  for (i = 0; i < s->winlen; i++)
237  dst[i] = dst[i+s->winlen];
238  }
239 
240  out->pts = in->pts;
241  av_frame_free(&in);
242 
243  return ff_filter_frame(outlink, out);
244 }
245 
247 {
248  AVFilterLink *inlink = ctx->inputs[0];
249  AVFilterLink *outlink = ctx->outputs[0];
250  SuperEqualizerContext *s = ctx->priv;
251  AVFrame *in = NULL;
252  int ret;
253 
255 
256  ret = ff_inlink_consume_samples(inlink, s->winlen, s->winlen, &in);
257  if (ret < 0)
258  return ret;
259  if (ret > 0)
260  return filter_frame(inlink, in);
261 
264 
265  return FFERROR_NOT_READY;
266 }
267 
269 {
270  SuperEqualizerContext *s = ctx->priv;
271 
272  return equ_init(s, 14);
273 }
274 
276 {
277  AVFilterContext *ctx = inlink->dst;
278  SuperEqualizerContext *s = ctx->priv;
279 
280  s->out = ff_get_audio_buffer(inlink, s->tabsize);
281  if (!s->out)
282  return AVERROR(ENOMEM);
283 
284  return 0;
285 }
286 
287 static int config_output(AVFilterLink *outlink)
288 {
289  AVFilterContext *ctx = outlink->src;
290  SuperEqualizerContext *s = ctx->priv;
291 
292  make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
293 
294  return 0;
295 }
296 
298 {
299  SuperEqualizerContext *s = ctx->priv;
300 
301  av_frame_free(&s->out);
302  av_freep(&s->irest);
303  av_freep(&s->ires);
304  av_freep(&s->fsamples);
305  av_freep(&s->fsamples_out);
306  av_tx_uninit(&s->rdft);
307  av_tx_uninit(&s->irdft);
308 }
309 
311  {
312  .name = "default",
313  .type = AVMEDIA_TYPE_AUDIO,
314  .config_props = config_input,
315  },
316 };
317 
319  {
320  .name = "default",
321  .type = AVMEDIA_TYPE_AUDIO,
322  .config_props = config_output,
323  },
324 };
325 
326 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
327 #define OFFSET(x) offsetof(SuperEqualizerContext, x)
328 
330  { "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
331  { "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
332  { "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
333  { "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
334  { "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
335  { "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
336  { "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
337  { "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
338  { "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
339  { "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
340  { "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
341  { "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
342  { "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
343  { "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
344  { "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
345  { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
346  { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
347  { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
348  { NULL }
349 };
350 
351 AVFILTER_DEFINE_CLASS(superequalizer);
352 
354  .name = "superequalizer",
355  .description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
356  .priv_size = sizeof(SuperEqualizerContext),
357  .priv_class = &superequalizer_class,
358  .init = init,
359  .activate = activate,
360  .uninit = uninit,
364 };
hn_lpf
static float hn_lpf(int n, float f, float fs)
Definition: af_superequalizer.c:76
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
SuperEqualizerContext::fsamples
float * fsamples
Definition: af_superequalizer.c:48
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:297
superequalizer_options
static const AVOption superequalizer_options[]
Definition: af_superequalizer.c:329
SuperEqualizerContext::in
AVFrame * in
Definition: af_superequalizer.c:51
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: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
FFERROR_NOT_READY
return FFERROR_NOT_READY
Definition: filter_design.txt:204
alpha
static float alpha(float a)
Definition: af_superequalizer.c:110
AVTXContext
Definition: tx_priv.h:228
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
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
im
float im
Definition: fft.c:79
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:325
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:432
AVOption
AVOption.
Definition: opt.h:251
SuperEqualizerContext
Definition: af_superequalizer.c:37
SuperEqualizerContext::params
EqParameter params[NBANDS+1]
Definition: af_superequalizer.c:40
AF
#define AF
Definition: af_superequalizer.c:326
AVComplexFloat
Definition: tx.h:27
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:175
ff_af_superequalizer
const AVFilter ff_af_superequalizer
Definition: af_superequalizer.c:353
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:311
make_fir
static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
Definition: af_superequalizer.c:170
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:47
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:883
win
static float win(SuperEqualizerContext *s, float n, int N)
Definition: af_superequalizer.c:119
SuperEqualizerContext::ires
float * ires
Definition: af_superequalizer.c:47
SuperEqualizerContext::tx_fn
av_tx_fn tx_fn
Definition: af_superequalizer.c:53
AVFrame::ch_layout
AVChannelLayout ch_layout
Channel layout of the audio data.
Definition: frame.h:709
scale
static av_always_inline float scale(float x, float s)
Definition: vf_v360.c:1389
SuperEqualizerContext::fact
float fact[M+1]
Definition: af_superequalizer.c:44
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:49
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:127
SuperEqualizerContext::winlen
int winlen
Definition: af_superequalizer.c:49
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_superequalizer.c:268
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_superequalizer.c:287
NBANDS
#define NBANDS
Definition: af_superequalizer.c:30
s
#define s(width, name)
Definition: cbs_vp9.c:256
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:124
filters.h
ctx
AVFormatContext * ctx
Definition: movenc.c:48
bands
static const float bands[]
Definition: af_superequalizer.c:56
superequalizer_outputs
static const AVFilterPad superequalizer_outputs[]
Definition: af_superequalizer.c:318
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: internal.h:190
izero
static float izero(SuperEqualizerContext *s, float x)
Definition: af_superequalizer.c:61
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:1423
NULL
#define NULL
Definition: coverity.c:32
fs
#define fs(width, name, subs,...)
Definition: cbs_vp9.c:258
OFFSET
#define OFFSET(x)
Definition: af_superequalizer.c:327
SuperEqualizerContext::out
AVFrame * out
Definition: af_superequalizer.c:51
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:122
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
hn
static float hn(int n, EqParameter *param, float fs)
Definition: af_superequalizer.c:91
EqParameter::gain
float gain
Definition: af_superequalizer.c:34
for
for(k=2;k<=8;++k)
Definition: h264pred_template.c:425
SuperEqualizerContext::tabsize
int tabsize
Definition: af_superequalizer.c:49
EqParameter::lower
float lower
Definition: af_superequalizer.c:34
equ_init
static int equ_init(SuperEqualizerContext *s, int wb)
Definition: af_superequalizer.c:135
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:46
FF_FILTER_FORWARD_WANTED
FF_FILTER_FORWARD_WANTED(outlink, inlink)
N
#define N
Definition: af_mcompand.c:53
superequalizer_inputs
static const AVFilterPad superequalizer_inputs[]
Definition: af_superequalizer.c:310
M_PI
#define M_PI
Definition: mathematics.h:52
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:294
SuperEqualizerContext::rdft
AVTXContext * rdft
Definition: af_superequalizer.c:52
internal.h
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:228
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:405
EqParameter::upper
float upper
Definition: af_superequalizer.c:34
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
M
#define M
Definition: af_superequalizer.c:31
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:386
SuperEqualizerContext::gains
float gains[NBANDS+1]
Definition: af_superequalizer.c:42
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:55
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Definition: mem.c:272
SuperEqualizerContext::itx_fn
av_tx_fn itx_fn
Definition: af_superequalizer.c:53
AVFilter
Filter definition.
Definition: avfilter.h:171
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:275
SuperEqualizerContext::aa
float aa
Definition: af_superequalizer.c:45
avfilter.h
SuperEqualizerContext::fsamples_out
float * fsamples_out
Definition: af_superequalizer.c:48
AVFilterContext
An instance of a filter.
Definition: avfilter.h:415
SuperEqualizerContext::irdft
AVTXContext * irdft
Definition: af_superequalizer.c:52
audio.h
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_superequalizer.c:188
FF_FILTER_FORWARD_STATUS
FF_FILTER_FORWARD_STATUS(inlink, outlink)
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:191
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:33
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(superequalizer)
hn_imp
static float hn_imp(int n)
Definition: af_superequalizer.c:86
activate
static int activate(AVFilterContext *ctx)
Definition: af_superequalizer.c:246
tx.h
re
float re
Definition: fft.c:79