FFmpeg
af_anlmdn.c
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1 /*
2  * Copyright (c) 2019 Paul B Mahol
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include <float.h>
22 
23 #include "libavutil/avassert.h"
24 #include "libavutil/audio_fifo.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/opt.h"
27 #include "avfilter.h"
28 #include "audio.h"
29 #include "formats.h"
30 
31 #include "af_anlmdndsp.h"
32 
33 #define WEIGHT_LUT_NBITS 20
34 #define WEIGHT_LUT_SIZE (1<<WEIGHT_LUT_NBITS)
35 
36 #define SQR(x) ((x) * (x))
37 
38 typedef struct AudioNLMeansContext {
39  const AVClass *class;
40 
41  float a;
42  int64_t pd;
43  int64_t rd;
44  float m;
45  int om;
46 
49 
50  int K;
51  int S;
52  int N;
53  int H;
54 
55  int offset;
58 
59  int64_t pts;
60 
62  int eof_left;
63 
66 
67 enum OutModes {
72 };
73 
74 #define OFFSET(x) offsetof(AudioNLMeansContext, x)
75 #define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
76 
77 static const AVOption anlmdn_options[] = {
78  { "strength", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, AFT },
79  { "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, AFT },
80  { "patch", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
81  { "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AFT },
82  { "research", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
83  { "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AFT },
84  { "output", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, "mode" },
85  { "o", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, "mode" },
86  { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AFT, "mode" },
87  { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AFT, "mode" },
88  { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE},0, 0, AFT, "mode" },
89  { "smooth", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 15, AFT },
90  { "m", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 15, AFT },
91  { NULL }
92 };
93 
94 AVFILTER_DEFINE_CLASS(anlmdn);
95 
96 static float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
97 {
98  float distance = 0.;
99 
100  for (int k = -K; k <= K; k++)
101  distance += SQR(f1[k] - f2[k]);
102 
103  return distance;
104 }
105 
106 static void compute_cache_c(float *cache, const float *f,
107  ptrdiff_t S, ptrdiff_t K,
108  ptrdiff_t i, ptrdiff_t jj)
109 {
110  int v = 0;
111 
112  for (int j = jj; j < jj + S; j++, v++)
113  cache[v] += -SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
114 }
115 
117 {
120 
121  if (ARCH_X86)
122  ff_anlmdn_init_x86(dsp);
123 }
124 
126 {
127  AudioNLMeansContext *s = ctx->priv;
128  AVFilterLink *outlink = ctx->outputs[0];
129  int newK, newS, newH, newN;
130  AVFrame *new_in, *new_cache;
131 
132  newK = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
133  newS = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
134 
135  newH = newK * 2 + 1;
136  newN = newH + (newK + newS) * 2;
137 
138  av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", newK, newS, newH, newN);
139 
140  if (!s->cache || s->cache->nb_samples < newS * 2) {
141  new_cache = ff_get_audio_buffer(outlink, newS * 2);
142  if (new_cache) {
143  av_frame_free(&s->cache);
144  s->cache = new_cache;
145  } else {
146  return AVERROR(ENOMEM);
147  }
148  }
149  if (!s->cache)
150  return AVERROR(ENOMEM);
151 
152  s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
153  for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
154  float w = -i / s->pdiff_lut_scale;
155 
156  s->weight_lut[i] = expf(w);
157  }
158 
159  if (!s->in || s->in->nb_samples < newN) {
160  new_in = ff_get_audio_buffer(outlink, newN);
161  if (new_in) {
162  av_frame_free(&s->in);
163  s->in = new_in;
164  } else {
165  return AVERROR(ENOMEM);
166  }
167  }
168  if (!s->in)
169  return AVERROR(ENOMEM);
170 
171  s->K = newK;
172  s->S = newS;
173  s->H = newH;
174  s->N = newN;
175 
176  return 0;
177 }
178 
179 static int config_output(AVFilterLink *outlink)
180 {
181  AVFilterContext *ctx = outlink->src;
182  AudioNLMeansContext *s = ctx->priv;
183  int ret;
184 
185  s->eof_left = -1;
186  s->pts = AV_NOPTS_VALUE;
187 
188  ret = config_filter(ctx);
189  if (ret < 0)
190  return ret;
191 
192  s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
193  if (!s->fifo)
194  return AVERROR(ENOMEM);
195 
196  ret = av_audio_fifo_write(s->fifo, (void **)s->in->extended_data, s->K + s->S);
197  if (ret < 0)
198  return ret;
199 
200  ff_anlmdn_init(&s->dsp);
201 
202  return 0;
203 }
204 
205 static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
206 {
207  AudioNLMeansContext *s = ctx->priv;
208  AVFrame *out = arg;
209  const int S = s->S;
210  const int K = s->K;
211  const int om = s->om;
212  const float *f = (const float *)(s->in->extended_data[ch]) + K;
213  float *cache = (float *)s->cache->extended_data[ch];
214  const float sw = (65536.f / (4 * K + 2)) / sqrtf(s->a);
215  float *dst = (float *)out->extended_data[ch] + s->offset;
216  const float smooth = s->m;
217 
218  for (int i = S; i < s->H + S; i++) {
219  float P = 0.f, Q = 0.f;
220  int v = 0;
221 
222  if (i == S) {
223  for (int j = i - S; j <= i + S; j++) {
224  if (i == j)
225  continue;
226  cache[v++] = s->dsp.compute_distance_ssd(f + i, f + j, K);
227  }
228  } else {
229  s->dsp.compute_cache(cache, f, S, K, i, i - S);
230  s->dsp.compute_cache(cache + S, f, S, K, i, i + 1);
231  }
232 
233  for (int j = 0; j < 2 * S && !ctx->is_disabled; j++) {
234  const float distance = cache[j];
235  unsigned weight_lut_idx;
236  float w;
237 
238  if (distance < 0.f) {
239  cache[j] = 0.f;
240  continue;
241  }
242  w = distance * sw;
243  if (w >= smooth)
244  continue;
245  weight_lut_idx = w * s->pdiff_lut_scale;
246  av_assert2(weight_lut_idx < WEIGHT_LUT_SIZE);
247  w = s->weight_lut[weight_lut_idx];
248  P += w * f[i - S + j + (j >= S)];
249  Q += w;
250  }
251 
252  P += f[i];
253  Q += 1;
254 
255  switch (om) {
256  case IN_MODE: dst[i - S] = f[i]; break;
257  case OUT_MODE: dst[i - S] = P / Q; break;
258  case NOISE_MODE: dst[i - S] = f[i] - (P / Q); break;
259  }
260  }
261 
262  return 0;
263 }
264 
266 {
267  AVFilterContext *ctx = inlink->dst;
268  AVFilterLink *outlink = ctx->outputs[0];
269  AudioNLMeansContext *s = ctx->priv;
270  AVFrame *out = NULL;
271  int available, wanted, ret;
272 
273  if (s->pts == AV_NOPTS_VALUE)
274  s->pts = in->pts;
275 
276  ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
277  in->nb_samples);
278  av_frame_free(&in);
279 
280  s->offset = 0;
281  available = av_audio_fifo_size(s->fifo);
282  wanted = (available / s->H) * s->H;
283 
284  if (wanted >= s->H && available >= s->N) {
285  out = ff_get_audio_buffer(outlink, wanted);
286  if (!out)
287  return AVERROR(ENOMEM);
288  }
289 
290  while (available >= s->N) {
291  ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, s->N);
292  if (ret < 0)
293  break;
294 
296 
297  av_audio_fifo_drain(s->fifo, s->H);
298 
299  s->offset += s->H;
300  available -= s->H;
301  }
302 
303  if (out) {
304  out->pts = s->pts;
305  out->nb_samples = s->offset;
306  if (s->eof_left >= 0) {
307  out->nb_samples = FFMIN(s->eof_left, s->offset);
308  s->eof_left -= out->nb_samples;
309  }
310  s->pts += av_rescale_q(s->offset, (AVRational){1, outlink->sample_rate}, outlink->time_base);
311 
312  return ff_filter_frame(outlink, out);
313  }
314 
315  return ret;
316 }
317 
318 static int request_frame(AVFilterLink *outlink)
319 {
320  AVFilterContext *ctx = outlink->src;
321  AudioNLMeansContext *s = ctx->priv;
322  int ret;
323 
324  ret = ff_request_frame(ctx->inputs[0]);
325 
326  if (ret == AVERROR_EOF && s->eof_left != 0) {
327  AVFrame *in;
328 
329  if (s->eof_left < 0)
330  s->eof_left = av_audio_fifo_size(s->fifo) - (s->S + s->K);
331  if (s->eof_left <= 0)
332  return AVERROR_EOF;
333  in = ff_get_audio_buffer(outlink, s->H);
334  if (!in)
335  return AVERROR(ENOMEM);
336 
337  return filter_frame(ctx->inputs[0], in);
338  }
339 
340  return ret;
341 }
342 
343 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
344  char *res, int res_len, int flags)
345 {
346  int ret;
347 
348  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
349  if (ret < 0)
350  return ret;
351 
352  ret = config_filter(ctx);
353  if (ret < 0)
354  return ret;
355 
356  return 0;
357 }
358 
360 {
361  AudioNLMeansContext *s = ctx->priv;
362 
363  av_audio_fifo_free(s->fifo);
364  av_frame_free(&s->in);
365  av_frame_free(&s->cache);
366 }
367 
368 static const AVFilterPad inputs[] = {
369  {
370  .name = "default",
371  .type = AVMEDIA_TYPE_AUDIO,
372  .filter_frame = filter_frame,
373  },
374 };
375 
376 static const AVFilterPad outputs[] = {
377  {
378  .name = "default",
379  .type = AVMEDIA_TYPE_AUDIO,
380  .config_props = config_output,
381  .request_frame = request_frame,
382  },
383 };
384 
386  .name = "anlmdn",
387  .description = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
388  .priv_size = sizeof(AudioNLMeansContext),
389  .priv_class = &anlmdn_class,
390  .uninit = uninit,
394  .process_command = process_command,
397 };
av_audio_fifo_free
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
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:88
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
ff_anlmdn_init
void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
Definition: af_anlmdn.c:116
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
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_anlmdn.c:265
out
FILE * out
Definition: movenc.c:54
OUT_MODE
@ OUT_MODE
Definition: af_anlmdn.c:69
AudioNLMDNDSPContext::compute_distance_ssd
float(* compute_distance_ssd)(const float *f1, const float *f2, ptrdiff_t K)
Definition: af_anlmdndsp.h:32
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1018
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:57
FILTER_SINGLE_SAMPLEFMT
#define FILTER_SINGLE_SAMPLEFMT(sample_fmt_)
Definition: internal.h:184
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:112
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:303
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:396
w
uint8_t w
Definition: llviddspenc.c:38
af_anlmdndsp.h
AVOption
AVOption.
Definition: opt.h:247
WEIGHT_LUT_SIZE
#define WEIGHT_LUT_SIZE
Definition: af_anlmdn.c:34
request_frame
static int request_frame(AVFilterLink *outlink)
Definition: af_anlmdn.c:318
AV_OPT_TYPE_DURATION
@ AV_OPT_TYPE_DURATION
Definition: opt.h:238
expf
#define expf(x)
Definition: libm.h:283
ff_request_frame
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:420
AudioNLMeansContext::N
int N
Definition: af_anlmdn.c:52
AudioNLMeansContext::S
int S
Definition: af_anlmdn.c:51
float.h
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_anlmdn.c:179
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:153
AudioNLMeansContext::pdiff_lut_scale
float pdiff_lut_scale
Definition: af_anlmdn.c:47
outputs
static const AVFilterPad outputs[]
Definition: af_anlmdn.c:376
AudioNLMeansContext::in
AVFrame * in
Definition: af_anlmdn.c:56
config_filter
static int config_filter(AVFilterContext *ctx)
Definition: af_anlmdn.c:125
formats.h
S
#define S(s, c, i)
Definition: flacdsp_template.c:46
AVAudioFifo
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
av_audio_fifo_drain
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
Definition: audio_fifo.c:201
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:50
AudioNLMeansContext::om
int om
Definition: af_anlmdn.c:45
avassert.h
av_cold
#define av_cold
Definition: attributes.h:90
anlmdn_options
static const AVOption anlmdn_options[]
Definition: af_anlmdn.c:77
NOISE_MODE
@ NOISE_MODE
Definition: af_anlmdn.c:70
s
#define s(width, name)
Definition: cbs_vp9.c:257
AudioNLMeansContext::fifo
AVAudioFifo * fifo
Definition: af_anlmdn.c:61
AudioNLMeansContext::offset
int offset
Definition: af_anlmdn.c:55
AudioNLMeansContext::H
int H
Definition: af_anlmdn.c:53
av_audio_fifo_write
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
AV_LOG_DEBUG
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:201
ctx
AVFormatContext * ctx
Definition: movenc.c:48
SQR
#define SQR(x)
Definition: af_anlmdn.c:36
AudioNLMeansContext
Definition: af_anlmdn.c:38
av_rescale_q
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:141
AudioNLMDNDSPContext
Definition: af_anlmdndsp.h:31
f
#define f(width, name)
Definition: cbs_vp9.c:255
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: internal.h:191
arg
const char * arg
Definition: jacosubdec.c:67
if
if(ret)
Definition: filter_design.txt:179
AudioNLMeansContext::cache
AVFrame * cache
Definition: af_anlmdn.c:57
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:66
AudioNLMeansContext::dsp
AudioNLMDNDSPContext dsp
Definition: af_anlmdn.c:64
NULL
#define NULL
Definition: coverity.c:32
filter_channel
static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
Definition: af_anlmdn.c:205
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
av_audio_fifo_alloc
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(anlmdn)
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
OutModes
OutModes
Definition: af_afftdn.c:37
ff_anlmdn_init_x86
void ff_anlmdn_init_x86(AudioNLMDNDSPContext *s)
Definition: af_anlmdn_init.c:28
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:117
P
#define P
AudioNLMeansContext::m
float m
Definition: af_anlmdn.c:44
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_anlmdn.c:359
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
compute_distance_ssd_c
static float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
Definition: af_anlmdn.c:96
AFT
#define AFT
Definition: af_anlmdn.c:75
AudioNLMDNDSPContext::compute_cache
void(* compute_cache)(float *cache, const float *f, ptrdiff_t S, ptrdiff_t K, ptrdiff_t i, ptrdiff_t jj)
Definition: af_anlmdndsp.h:33
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:882
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
process_command
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_anlmdn.c:343
AudioNLMeansContext::rd
int64_t rd
Definition: af_anlmdn.c:43
av_audio_fifo_size
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
AV_OPT_TYPE_FLOAT
@ AV_OPT_TYPE_FLOAT
Definition: opt.h:227
AudioNLMeansContext::weight_lut
float weight_lut[WEIGHT_LUT_SIZE]
Definition: af_anlmdn.c:48
AudioNLMeansContext::pts
int64_t pts
Definition: af_anlmdn.c:59
AudioNLMeansContext::pd
int64_t pd
Definition: af_anlmdn.c:42
av_assert2
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:369
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:271
available
if no frame is available
Definition: filter_design.txt:166
AV_TIME_BASE
#define AV_TIME_BASE
Internal time base represented as integer.
Definition: avutil.h:254
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:350
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
audio_fifo.h
IN_MODE
@ IN_MODE
Definition: af_anlmdn.c:68
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:56
av_rescale
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:128
ff_af_anlmdn
const AVFilter ff_af_anlmdn
Definition: af_anlmdn.c:385
smooth
static float smooth(DeshakeOpenCLContext *deshake_ctx, float *gauss_kernel, int length, float max_val, AVFifoBuffer *values)
Definition: vf_deshake_opencl.c:903
AVFilter
Filter definition.
Definition: avfilter.h:149
ret
ret
Definition: filter_design.txt:187
NB_MODES
@ NB_MODES
Definition: af_anlmdn.c:71
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:224
avfilter.h
AVFilterContext
An instance of a filter.
Definition: avfilter.h:386
OFFSET
#define OFFSET(x)
Definition: af_anlmdn.c:74
compute_cache_c
static void compute_cache_c(float *cache, const float *f, ptrdiff_t S, ptrdiff_t K, ptrdiff_t i, ptrdiff_t jj)
Definition: af_anlmdn.c:106
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:121
audio.h
AudioNLMeansContext::a
float a
Definition: af_anlmdn.c:41
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: internal.h:192
distance
static float distance(float x, float y, int band)
Definition: nellymoserenc.c:233
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:138
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:561
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
avstring.h
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:143
av_audio_fifo_peek
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
Peek data from an AVAudioFifo.
Definition: audio_fifo.c:138
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:233
AudioNLMeansContext::eof_left
int eof_left
Definition: af_anlmdn.c:62
AudioNLMeansContext::K
int K
Definition: af_anlmdn.c:50
inputs
static const AVFilterPad inputs[]
Definition: af_anlmdn.c:368