FFmpeg
af_adenorm.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
7  * version 2.1 of the License, or (at your option) any later version.
8  *
9  * FFmpeg is distributed in the hope that it will be useful,
10  * but WITHOUT ANY WARRANTY; without even the implied warranty of
11  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12  * Lesser General Public License for more details.
13  *
14  * You should have received a copy of the GNU Lesser General Public
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 "libavutil/avassert.h"
21 #include "libavutil/opt.h"
22 #include "audio.h"
23 #include "avfilter.h"
24 #include "filters.h"
25 
26 enum FilterType {
32 };
33 
34 typedef struct ADenormContext {
35  const AVClass *class;
36 
37  double level;
38  double level_db;
39  int type;
41 
43  const void *src, int nb_samples);
45 
46 static void dc_denorm_fltp(AVFilterContext *ctx, void *dstp,
47  const void *srcp, int nb_samples)
48 {
49  ADenormContext *s = ctx->priv;
50  const float *src = (const float *)srcp;
51  float *dst = (float *)dstp;
52  const float dc = s->level;
53 
54  for (int n = 0; n < nb_samples; n++) {
55  dst[n] = src[n] + dc;
56  }
57 }
58 
59 static void dc_denorm_dblp(AVFilterContext *ctx, void *dstp,
60  const void *srcp, int nb_samples)
61 {
62  ADenormContext *s = ctx->priv;
63  const double *src = (const double *)srcp;
64  double *dst = (double *)dstp;
65  const double dc = s->level;
66 
67  for (int n = 0; n < nb_samples; n++) {
68  dst[n] = src[n] + dc;
69  }
70 }
71 
72 static void ac_denorm_fltp(AVFilterContext *ctx, void *dstp,
73  const void *srcp, int nb_samples)
74 {
75  ADenormContext *s = ctx->priv;
76  const float *src = (const float *)srcp;
77  float *dst = (float *)dstp;
78  const float dc = s->level;
79  const int64_t N = s->in_samples;
80 
81  for (int n = 0; n < nb_samples; n++) {
82  dst[n] = src[n] + dc * (((N + n) & 1) ? -1.f : 1.f);
83  }
84 }
85 
86 static void ac_denorm_dblp(AVFilterContext *ctx, void *dstp,
87  const void *srcp, int nb_samples)
88 {
89  ADenormContext *s = ctx->priv;
90  const double *src = (const double *)srcp;
91  double *dst = (double *)dstp;
92  const double dc = s->level;
93  const int64_t N = s->in_samples;
94 
95  for (int n = 0; n < nb_samples; n++) {
96  dst[n] = src[n] + dc * (((N + n) & 1) ? -1. : 1.);
97  }
98 }
99 
100 static void sq_denorm_fltp(AVFilterContext *ctx, void *dstp,
101  const void *srcp, int nb_samples)
102 {
103  ADenormContext *s = ctx->priv;
104  const float *src = (const float *)srcp;
105  float *dst = (float *)dstp;
106  const float dc = s->level;
107  const int64_t N = s->in_samples;
108 
109  for (int n = 0; n < nb_samples; n++) {
110  dst[n] = src[n] + dc * ((((N + n) >> 8) & 1) ? -1.f : 1.f);
111  }
112 }
113 
114 static void sq_denorm_dblp(AVFilterContext *ctx, void *dstp,
115  const void *srcp, int nb_samples)
116 {
117  ADenormContext *s = ctx->priv;
118  const double *src = (const double *)srcp;
119  double *dst = (double *)dstp;
120  const double dc = s->level;
121  const int64_t N = s->in_samples;
122 
123  for (int n = 0; n < nb_samples; n++) {
124  dst[n] = src[n] + dc * ((((N + n) >> 8) & 1) ? -1. : 1.);
125  }
126 }
127 
128 static void ps_denorm_fltp(AVFilterContext *ctx, void *dstp,
129  const void *srcp, int nb_samples)
130 {
131  ADenormContext *s = ctx->priv;
132  const float *src = (const float *)srcp;
133  float *dst = (float *)dstp;
134  const float dc = s->level;
135  const int64_t N = s->in_samples;
136 
137  for (int n = 0; n < nb_samples; n++) {
138  dst[n] = src[n] + dc * (((N + n) & 255) ? 0.f : 1.f);
139  }
140 }
141 
142 static void ps_denorm_dblp(AVFilterContext *ctx, void *dstp,
143  const void *srcp, int nb_samples)
144 {
145  ADenormContext *s = ctx->priv;
146  const double *src = (const double *)srcp;
147  double *dst = (double *)dstp;
148  const double dc = s->level;
149  const int64_t N = s->in_samples;
150 
151  for (int n = 0; n < nb_samples; n++) {
152  dst[n] = src[n] + dc * (((N + n) & 255) ? 0. : 1.);
153  }
154 }
155 
156 static int config_output(AVFilterLink *outlink)
157 {
158  AVFilterContext *ctx = outlink->src;
159  ADenormContext *s = ctx->priv;
160 
161  switch (outlink->format) {
162  case AV_SAMPLE_FMT_FLTP:
163  s->filter[DC_TYPE] = dc_denorm_fltp;
164  s->filter[AC_TYPE] = ac_denorm_fltp;
165  s->filter[SQ_TYPE] = sq_denorm_fltp;
166  s->filter[PS_TYPE] = ps_denorm_fltp;
167  break;
168  case AV_SAMPLE_FMT_DBLP:
169  s->filter[DC_TYPE] = dc_denorm_dblp;
170  s->filter[AC_TYPE] = ac_denorm_dblp;
171  s->filter[SQ_TYPE] = sq_denorm_dblp;
172  s->filter[PS_TYPE] = ps_denorm_dblp;
173  break;
174  default:
175  av_assert0(0);
176  }
177 
178  return 0;
179 }
180 
181 typedef struct ThreadData {
182  AVFrame *in, *out;
183 } ThreadData;
184 
185 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
186 {
187  ADenormContext *s = ctx->priv;
188  ThreadData *td = arg;
189  AVFrame *out = td->out;
190  AVFrame *in = td->in;
191  const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
192  const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
193 
194  for (int ch = start; ch < end; ch++) {
195  s->filter[s->type](ctx, out->extended_data[ch],
196  in->extended_data[ch],
197  in->nb_samples);
198  }
199 
200  return 0;
201 }
202 
204 {
205  AVFilterContext *ctx = inlink->dst;
206  ADenormContext *s = ctx->priv;
207  AVFilterLink *outlink = ctx->outputs[0];
208  ThreadData td;
209  AVFrame *out;
210 
211  if (av_frame_is_writable(in)) {
212  out = in;
213  } else {
214  out = ff_get_audio_buffer(outlink, in->nb_samples);
215  if (!out) {
216  av_frame_free(&in);
217  return AVERROR(ENOMEM);
218  }
220  }
221 
222  s->level = exp(s->level_db / 20. * M_LN10);
223  td.in = in; td.out = out;
225  FFMIN(inlink->ch_layout.nb_channels, ff_filter_get_nb_threads(ctx)));
226 
227  s->in_samples += in->nb_samples;
228 
229  if (out != in)
230  av_frame_free(&in);
231  return ff_filter_frame(outlink, out);
232 }
233 
234 static const AVFilterPad adenorm_inputs[] = {
235  {
236  .name = "default",
237  .type = AVMEDIA_TYPE_AUDIO,
238  .filter_frame = filter_frame,
239  },
240 };
241 
242 static const AVFilterPad adenorm_outputs[] = {
243  {
244  .name = "default",
245  .type = AVMEDIA_TYPE_AUDIO,
246  .config_props = config_output,
247  },
248 };
249 
250 #define OFFSET(x) offsetof(ADenormContext, x)
251 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
252 
253 static const AVOption adenorm_options[] = {
254  { "level", "set level", OFFSET(level_db), AV_OPT_TYPE_DOUBLE, {.dbl=-351}, -451, -90, FLAGS },
255  { "type", "set type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=DC_TYPE}, 0, NB_TYPES-1, FLAGS, .unit = "type" },
256  { "dc", NULL, 0, AV_OPT_TYPE_CONST, {.i64=DC_TYPE}, 0, 0, FLAGS, .unit = "type"},
257  { "ac", NULL, 0, AV_OPT_TYPE_CONST, {.i64=AC_TYPE}, 0, 0, FLAGS, .unit = "type"},
258  { "square",NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQ_TYPE}, 0, 0, FLAGS, .unit = "type"},
259  { "pulse", NULL, 0, AV_OPT_TYPE_CONST, {.i64=PS_TYPE}, 0, 0, FLAGS, .unit = "type"},
260  { NULL }
261 };
262 
263 AVFILTER_DEFINE_CLASS(adenorm);
264 
266  .name = "adenorm",
267  .description = NULL_IF_CONFIG_SMALL("Remedy denormals by adding extremely low-level noise."),
268  .priv_size = sizeof(ADenormContext),
272  .priv_class = &adenorm_class,
273  .process_command = ff_filter_process_command,
276 };
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:98
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:66
ac_denorm_fltp
static void ac_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:72
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
PS_TYPE
@ PS_TYPE
Definition: af_adenorm.c:30
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1061
adenorm_outputs
static const AVFilterPad adenorm_outputs[]
Definition: af_adenorm.c:242
ps_denorm_dblp
static void ps_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:142
int64_t
long long int64_t
Definition: coverity.c:34
ps_denorm_fltp
static void ps_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:128
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:162
FILTER_INPUTS
#define FILTER_INPUTS(array)
Definition: filters.h:262
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:389
AVOption
AVOption.
Definition: opt.h:429
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:205
ThreadData::out
AVFrame * out
Definition: af_adeclick.c:526
AVChannelLayout::nb_channels
int nb_channels
Number of channels in this layout.
Definition: channel_layout.h:327
ThreadData::in
AVFrame * in
Definition: af_adecorrelate.c:155
ac_denorm_dblp
static void ac_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:86
ADenormContext::in_samples
int64_t in_samples
Definition: af_adenorm.c:40
NB_TYPES
@ NB_TYPES
Definition: af_adenorm.c:31
AVFrame::ch_layout
AVChannelLayout ch_layout
Channel layout of the audio data.
Definition: frame.h:790
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
AVFilterPad
A filter pad used for either input or output.
Definition: filters.h:38
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_adenorm.c:156
dc_denorm_fltp
static void dc_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:46
FilterType
FilterType
Definition: af_adenorm.c:26
avassert.h
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_adenorm.c:203
OFFSET
#define OFFSET(x)
Definition: af_adenorm.c:250
FILTER_SAMPLEFMTS
#define FILTER_SAMPLEFMTS(...)
Definition: filters.h:250
FLAGS
#define FLAGS
Definition: af_adenorm.c:251
s
#define s(width, name)
Definition: cbs_vp9.c:198
sq_denorm_fltp
static void sq_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:100
AV_OPT_TYPE_DOUBLE
@ AV_OPT_TYPE_DOUBLE
Underlying C type is double.
Definition: opt.h:267
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
av_assert0
#define av_assert0(cond)
assert() equivalent, that is always enabled.
Definition: avassert.h:40
filters.h
ctx
AVFormatContext * ctx
Definition: movenc.c:49
DC_TYPE
@ DC_TYPE
Definition: af_adenorm.c:27
FILTER_OUTPUTS
#define FILTER_OUTPUTS(array)
Definition: filters.h:263
arg
const char * arg
Definition: jacosubdec.c:67
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:75
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:725
adenorm_inputs
static const AVFilterPad adenorm_inputs[]
Definition: af_adenorm.c:234
exp
int8_t exp
Definition: eval.c:73
f
f
Definition: af_crystalizer.c:122
ADenormContext
Definition: af_adenorm.c:34
dc
Tag MUST be and< 10hcoeff half pel interpolation filter coefficients, hcoeff[0] are the 2 middle coefficients[1] are the next outer ones and so on, resulting in a filter like:...eff[2], hcoeff[1], hcoeff[0], hcoeff[0], hcoeff[1], hcoeff[2] ... the sign of the coefficients is not explicitly stored but alternates after each coeff and coeff[0] is positive, so ...,+,-,+,-,+,+,-,+,-,+,... hcoeff[0] is not explicitly stored but found by subtracting the sum of all stored coefficients with signs from 32 hcoeff[0]=32 - hcoeff[1] - hcoeff[2] - ... a good choice for hcoeff and htaps is htaps=6 hcoeff={40,-10, 2} an alternative which requires more computations at both encoder and decoder side and may or may not be better is htaps=8 hcoeff={42,-14, 6,-2}ref_frames minimum of the number of available reference frames and max_ref_frames for example the first frame after a key frame always has ref_frames=1spatial_decomposition_type wavelet type 0 is a 9/7 symmetric compact integer wavelet 1 is a 5/3 symmetric compact integer wavelet others are reserved stored as delta from last, last is reset to 0 if always_reset||keyframeqlog quality(logarithmic quantizer scale) stored as delta from last, last is reset to 0 if always_reset||keyframemv_scale stored as delta from last, last is reset to 0 if always_reset||keyframe FIXME check that everything works fine if this changes between framesqbias dequantization bias stored as delta from last, last is reset to 0 if always_reset||keyframeblock_max_depth maximum depth of the block tree stored as delta from last, last is reset to 0 if always_reset||keyframequant_table quantization tableHighlevel bitstream structure:==============================--------------------------------------------|Header|--------------------------------------------|------------------------------------|||Block0||||split?||||yes no||||......... intra?||||:Block01 :yes no||||:Block02 :....... ..........||||:Block03 ::y DC ::ref index:||||:Block04 ::cb DC ::motion x :||||......... :cr DC ::motion y :||||....... ..........|||------------------------------------||------------------------------------|||Block1|||...|--------------------------------------------|------------ ------------ ------------|||Y subbands||Cb subbands||Cr subbands||||--- ---||--- ---||--- ---|||||LL0||HL0||||LL0||HL0||||LL0||HL0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||LH0||HH0||||LH0||HH0||||LH0||HH0|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HL1||LH1||||HL1||LH1||||HL1||LH1|||||--- ---||--- ---||--- ---||||--- ---||--- ---||--- ---|||||HH1||HL2||||HH1||HL2||||HH1||HL2|||||...||...||...|||------------ ------------ ------------|--------------------------------------------Decoding process:=================------------|||Subbands|------------||||------------|Intra DC||||LL0 subband prediction ------------|\ Dequantization ------------------- \||Reference frames|\ IDWT|------- -------|Motion \|||Frame 0||Frame 1||Compensation . OBMC v -------|------- -------|--------------. \------> Frame n output Frame Frame<----------------------------------/|...|------------------- Range Coder:============Binary Range Coder:------------------- The implemented range coder is an adapted version based upon "Range encoding: an algorithm for removing redundancy from a digitised message." by G. N. N. Martin. The symbols encoded by the Snow range coder are bits(0|1). The associated probabilities are not fix but change depending on the symbol mix seen so far. bit seen|new state ---------+----------------------------------------------- 0|256 - state_transition_table[256 - old_state];1|state_transition_table[old_state];state_transition_table={ 0, 0, 0, 0, 0, 0, 0, 0, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 37, 38, 39, 40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56, 56, 57, 58, 59, 60, 61, 62, 63, 64, 65, 66, 67, 68, 69, 70, 71, 72, 73, 74, 75, 75, 76, 77, 78, 79, 80, 81, 82, 83, 84, 85, 86, 87, 88, 89, 90, 91, 92, 93, 94, 94, 95, 96, 97, 98, 99, 100, 101, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 114, 115, 116, 117, 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133, 133, 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149, 150, 151, 152, 152, 153, 154, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164, 165, 166, 167, 168, 169, 170, 171, 171, 172, 173, 174, 175, 176, 177, 178, 179, 180, 181, 182, 183, 184, 185, 186, 187, 188, 189, 190, 190, 191, 192, 194, 194, 195, 196, 197, 198, 199, 200, 201, 202, 202, 204, 205, 206, 207, 208, 209, 209, 210, 211, 212, 213, 215, 215, 216, 217, 218, 219, 220, 220, 222, 223, 224, 225, 226, 227, 227, 229, 229, 230, 231, 232, 234, 234, 235, 236, 237, 238, 239, 240, 241, 242, 243, 244, 245, 246, 247, 248, 248, 0, 0, 0, 0, 0, 0, 0};FIXME Range Coding of integers:------------------------- FIXME Neighboring Blocks:===================left and top are set to the respective blocks unless they are outside of the image in which case they are set to the Null block top-left is set to the top left block unless it is outside of the image in which case it is set to the left block if this block has no larger parent block or it is at the left side of its parent block and the top right block is not outside of the image then the top right block is used for top-right else the top-left block is used Null block y, cb, cr are 128 level, ref, mx and my are 0 Motion Vector Prediction:=========================1. the motion vectors of all the neighboring blocks are scaled to compensate for the difference of reference frames scaled_mv=(mv *(256 *(current_reference+1)/(mv.reference+1))+128)> the median of the scaled top and top right vectors is used as motion vector prediction the used motion vector is the sum of the predictor and(mvx_diff, mvy_diff) *mv_scale Intra DC Prediction block[y][x] dc[1]
Definition: snow.txt:400
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
dst
uint8_t ptrdiff_t const uint8_t ptrdiff_t int intptr_t intptr_t int int16_t * dst
Definition: dsp.h:83
ff_af_adenorm
const AVFilter ff_af_adenorm
Definition: af_adenorm.c:265
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:661
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:900
AC_TYPE
@ AC_TYPE
Definition: af_adenorm.c:28
N
#define N
Definition: af_mcompand.c:54
AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
Definition: avfilter.h:182
ADenormContext::level_db
double level_db
Definition: af_adenorm.c:38
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:469
adenorm_options
static const AVOption adenorm_options[]
Definition: af_adenorm.c:253
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:450
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(adenorm)
ADenormContext::level
double level
Definition: af_adenorm.c:37
ff_filter_get_nb_threads
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:840
ThreadData
Used for passing data between threads.
Definition: dsddec.c:71
FFMIN
#define FFMIN(a, b)
Definition: macros.h:49
AVFilterPad::name
const char * name
Pad name.
Definition: filters.h:44
AVFilter
Filter definition.
Definition: avfilter.h:201
filter_channels
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_adenorm.c:185
dc_denorm_dblp
static void dc_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:59
channel_layout.h
ff_filter_execute
int ff_filter_execute(AVFilterContext *ctx, avfilter_action_func *func, void *arg, int *ret, int nb_jobs)
Definition: avfilter.c:1666
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Underlying C type is int.
Definition: opt.h:259
avfilter.h
ADenormContext::filter
void(* filter[NB_TYPES])(AVFilterContext *ctx, void *dst, const void *src, int nb_samples)
Definition: af_adenorm.c:42
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:67
SQ_TYPE
@ SQ_TYPE
Definition: af_adenorm.c:29
AVFilterContext
An instance of a filter.
Definition: avfilter.h:457
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:152
audio.h
M_LN10
#define M_LN10
Definition: mathematics.h:49
ADenormContext::type
int type
Definition: af_adenorm.c:39
sq_denorm_dblp
static void sq_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:114
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Special option type for declaring named constants.
Definition: opt.h:299
src
#define src
Definition: vp8dsp.c:248