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 "internal.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;
40  int64_t in_samples;
41 
42  void (*filter)(AVFilterContext *ctx, void *dst,
43  const void *src, int nb_samples);
45 
47 {
50  static const enum AVSampleFormat sample_fmts[] = {
53  };
54  int ret;
55 
57  if (!formats)
58  return AVERROR(ENOMEM);
60  if (ret < 0)
61  return ret;
62 
64  if (!layouts)
65  return AVERROR(ENOMEM);
66 
68  if (ret < 0)
69  return ret;
70 
73 }
74 
75 static void dc_denorm_fltp(AVFilterContext *ctx, void *dstp,
76  const void *srcp, int nb_samples)
77 {
78  ADenormContext *s = ctx->priv;
79  const float *src = (const float *)srcp;
80  float *dst = (float *)dstp;
81  const float dc = s->level;
82 
83  for (int n = 0; n < nb_samples; n++) {
84  dst[n] = src[n] + dc;
85  }
86 }
87 
88 static void dc_denorm_dblp(AVFilterContext *ctx, void *dstp,
89  const void *srcp, int nb_samples)
90 {
91  ADenormContext *s = ctx->priv;
92  const double *src = (const double *)srcp;
93  double *dst = (double *)dstp;
94  const double dc = s->level;
95 
96  for (int n = 0; n < nb_samples; n++) {
97  dst[n] = src[n] + dc;
98  }
99 }
100 
101 static void ac_denorm_fltp(AVFilterContext *ctx, void *dstp,
102  const void *srcp, int nb_samples)
103 {
104  ADenormContext *s = ctx->priv;
105  const float *src = (const float *)srcp;
106  float *dst = (float *)dstp;
107  const float dc = s->level;
108  const int64_t N = s->in_samples;
109 
110  for (int n = 0; n < nb_samples; n++) {
111  dst[n] = src[n] + dc * (((N + n) & 1) ? -1.f : 1.f);
112  }
113 }
114 
115 static void ac_denorm_dblp(AVFilterContext *ctx, void *dstp,
116  const void *srcp, int nb_samples)
117 {
118  ADenormContext *s = ctx->priv;
119  const double *src = (const double *)srcp;
120  double *dst = (double *)dstp;
121  const double dc = s->level;
122  const int64_t N = s->in_samples;
123 
124  for (int n = 0; n < nb_samples; n++) {
125  dst[n] = src[n] + dc * (((N + n) & 1) ? -1. : 1.);
126  }
127 }
128 
129 static void sq_denorm_fltp(AVFilterContext *ctx, void *dstp,
130  const void *srcp, int nb_samples)
131 {
132  ADenormContext *s = ctx->priv;
133  const float *src = (const float *)srcp;
134  float *dst = (float *)dstp;
135  const float dc = s->level;
136  const int64_t N = s->in_samples;
137 
138  for (int n = 0; n < nb_samples; n++) {
139  dst[n] = src[n] + dc * ((((N + n) >> 8) & 1) ? -1.f : 1.f);
140  }
141 }
142 
143 static void sq_denorm_dblp(AVFilterContext *ctx, void *dstp,
144  const void *srcp, int nb_samples)
145 {
146  ADenormContext *s = ctx->priv;
147  const double *src = (const double *)srcp;
148  double *dst = (double *)dstp;
149  const double dc = s->level;
150  const int64_t N = s->in_samples;
151 
152  for (int n = 0; n < nb_samples; n++) {
153  dst[n] = src[n] + dc * ((((N + n) >> 8) & 1) ? -1. : 1.);
154  }
155 }
156 
157 static void ps_denorm_fltp(AVFilterContext *ctx, void *dstp,
158  const void *srcp, int nb_samples)
159 {
160  ADenormContext *s = ctx->priv;
161  const float *src = (const float *)srcp;
162  float *dst = (float *)dstp;
163  const float dc = s->level;
164  const int64_t N = s->in_samples;
165 
166  for (int n = 0; n < nb_samples; n++) {
167  dst[n] = src[n] + dc * (((N + n) & 255) ? 0.f : 1.f);
168  }
169 }
170 
171 static void ps_denorm_dblp(AVFilterContext *ctx, void *dstp,
172  const void *srcp, int nb_samples)
173 {
174  ADenormContext *s = ctx->priv;
175  const double *src = (const double *)srcp;
176  double *dst = (double *)dstp;
177  const double dc = s->level;
178  const int64_t N = s->in_samples;
179 
180  for (int n = 0; n < nb_samples; n++) {
181  dst[n] = src[n] + dc * (((N + n) & 255) ? 0. : 1.);
182  }
183 }
184 
185 static int config_output(AVFilterLink *outlink)
186 {
187  AVFilterContext *ctx = outlink->src;
188  ADenormContext *s = ctx->priv;
189 
190  switch (s->type) {
191  case DC_TYPE:
192  switch (outlink->format) {
193  case AV_SAMPLE_FMT_FLTP: s->filter = dc_denorm_fltp; break;
194  case AV_SAMPLE_FMT_DBLP: s->filter = dc_denorm_dblp; break;
195  }
196  break;
197  case AC_TYPE:
198  switch (outlink->format) {
199  case AV_SAMPLE_FMT_FLTP: s->filter = ac_denorm_fltp; break;
200  case AV_SAMPLE_FMT_DBLP: s->filter = ac_denorm_dblp; break;
201  }
202  break;
203  case SQ_TYPE:
204  switch (outlink->format) {
205  case AV_SAMPLE_FMT_FLTP: s->filter = sq_denorm_fltp; break;
206  case AV_SAMPLE_FMT_DBLP: s->filter = sq_denorm_dblp; break;
207  }
208  break;
209  case PS_TYPE:
210  switch (outlink->format) {
211  case AV_SAMPLE_FMT_FLTP: s->filter = ps_denorm_fltp; break;
212  case AV_SAMPLE_FMT_DBLP: s->filter = ps_denorm_dblp; break;
213  }
214  break;
215  default:
216  av_assert0(0);
217  }
218 
219  return 0;
220 }
221 
222 typedef struct ThreadData {
224 } ThreadData;
225 
226 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
227 {
228  ADenormContext *s = ctx->priv;
229  ThreadData *td = arg;
230  AVFrame *out = td->out;
231  AVFrame *in = td->in;
232  const int start = (in->channels * jobnr) / nb_jobs;
233  const int end = (in->channels * (jobnr+1)) / nb_jobs;
234 
235  for (int ch = start; ch < end; ch++) {
236  s->filter(ctx, out->extended_data[ch],
237  in->extended_data[ch],
238  in->nb_samples);
239  }
240 
241  return 0;
242 }
243 
245 {
246  AVFilterContext *ctx = inlink->dst;
247  ADenormContext *s = ctx->priv;
248  AVFilterLink *outlink = ctx->outputs[0];
249  ThreadData td;
250  AVFrame *out;
251 
252  if (av_frame_is_writable(in)) {
253  out = in;
254  } else {
255  out = ff_get_audio_buffer(outlink, in->nb_samples);
256  if (!out) {
257  av_frame_free(&in);
258  return AVERROR(ENOMEM);
259  }
261  }
262 
263  s->level = exp(s->level_db / 20. * M_LN10);
264  td.in = in; td.out = out;
265  ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
267 
268  s->in_samples += in->nb_samples;
269 
270  if (out != in)
271  av_frame_free(&in);
272  return ff_filter_frame(outlink, out);
273 }
274 
275 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
276  char *res, int res_len, int flags)
277 {
278  AVFilterLink *outlink = ctx->outputs[0];
279  int ret;
280 
281  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
282  if (ret < 0)
283  return ret;
284 
285  return config_output(outlink);
286 }
287 
288 static const AVFilterPad adenorm_inputs[] = {
289  {
290  .name = "default",
291  .type = AVMEDIA_TYPE_AUDIO,
292  .filter_frame = filter_frame,
293  },
294  { NULL }
295 };
296 
297 static const AVFilterPad adenorm_outputs[] = {
298  {
299  .name = "default",
300  .type = AVMEDIA_TYPE_AUDIO,
301  .config_props = config_output,
302  },
303  { NULL }
304 };
305 
306 #define OFFSET(x) offsetof(ADenormContext, x)
307 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
308 
309 static const AVOption adenorm_options[] = {
310  { "level", "set level", OFFSET(level_db), AV_OPT_TYPE_DOUBLE, {.dbl=-351}, -451, -90, FLAGS },
311  { "type", "set type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=DC_TYPE}, 0, NB_TYPES-1, FLAGS, "type" },
312  { "dc", NULL, 0, AV_OPT_TYPE_CONST, {.i64=DC_TYPE}, 0, 0, FLAGS, "type"},
313  { "ac", NULL, 0, AV_OPT_TYPE_CONST, {.i64=AC_TYPE}, 0, 0, FLAGS, "type"},
314  { "square",NULL, 0, AV_OPT_TYPE_CONST, {.i64=SQ_TYPE}, 0, 0, FLAGS, "type"},
315  { "pulse", NULL, 0, AV_OPT_TYPE_CONST, {.i64=PS_TYPE}, 0, 0, FLAGS, "type"},
316  { NULL }
317 };
318 
319 AVFILTER_DEFINE_CLASS(adenorm);
320 
322  .name = "adenorm",
323  .description = NULL_IF_CONFIG_SMALL("Remedy denormals by adding extremely low-level noise."),
324  .query_formats = query_formats,
325  .priv_size = sizeof(ADenormContext),
328  .priv_class = &adenorm_class,
332 };
formats
formats
Definition: signature.h:48
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:86
AV_SAMPLE_FMT_FLTP
@ AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:86
td
#define td
Definition: regdef.h:70
ac_denorm_fltp
static void ac_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:101
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_adenorm.c:46
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
ff_make_format_list
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:286
out
FILE * out
Definition: movenc.c:54
PS_TYPE
@ PS_TYPE
Definition: af_adenorm.c:30
process_command
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_adenorm.c:275
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1094
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:926
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
adenorm_outputs
static const AVFilterPad adenorm_outputs[]
Definition: af_adenorm.c:297
ps_denorm_dblp
static void ps_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:171
ps_denorm_fltp
static void ps_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:157
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:204
ff_all_channel_counts
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition.
Definition: formats.c:436
AudioConvert::channels
int channels
Definition: audio_convert.c:54
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:324
AVOption
AVOption.
Definition: opt.h:248
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:149
ThreadData::out
AVFrame * out
Definition: af_adeclick.c:502
AVFormatContext::internal
AVFormatInternal * internal
An opaque field for libavformat internal usage.
Definition: avformat.h:1698
ac_denorm_dblp
static void ac_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:115
ADenormContext::in_samples
int64_t in_samples
Definition: af_adenorm.c:40
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:65
NB_TYPES
@ NB_TYPES
Definition: af_adenorm.c:31
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: internal.h:54
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_adenorm.c:185
dc_denorm_fltp
static void dc_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:75
FilterType
FilterType
Definition: af_adenorm.c:26
avassert.h
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_adenorm.c:244
OFFSET
#define OFFSET(x)
Definition: af_adenorm.c:306
inputs
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
Definition: filter_design.txt:243
ff_set_common_formats
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:587
FLAGS
#define FLAGS
Definition: af_adenorm.c:307
s
#define s(width, name)
Definition: cbs_vp9.c:257
sq_denorm_fltp
static void sq_denorm_fltp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:129
AV_OPT_TYPE_DOUBLE
@ AV_OPT_TYPE_DOUBLE
Definition: opt.h:227
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:37
outputs
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
ctx
AVFormatContext * ctx
Definition: movenc.c:48
DC_TYPE
@ DC_TYPE
Definition: af_adenorm.c:27
f
#define f(width, name)
Definition: cbs_vp9.c:255
arg
const char * arg
Definition: jacosubdec.c:66
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
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:659
src
#define src
Definition: vp8dsp.c:255
adenorm_inputs
static const AVFilterPad adenorm_inputs[]
Definition: af_adenorm.c:288
exp
int8_t exp
Definition: eval.c:72
ADenormContext::filter
void(* filter)(AVFilterContext *ctx, void *dst, const void *src, int nb_samples)
Definition: af_adenorm.c:42
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:117
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
av_frame_is_writable
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:595
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
AC_TYPE
@ AC_TYPE
Definition: af_adenorm.c:28
FFMIN
#define FFMIN(a, b)
Definition: common.h:105
N
#define N
Definition: af_mcompand.c:54
internal.h
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:126
in
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Definition: audio_convert.c:326
ADenormContext::level_db
double level_db
Definition: af_adenorm.c:38
adenorm_options
static const AVOption adenorm_options[]
Definition: af_adenorm.c:309
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:802
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
ThreadData
Used for passing data between threads.
Definition: dsddec.c:67
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
AVFilter
Filter definition.
Definition: avfilter.h:145
filter_channels
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_adenorm.c:226
ret
ret
Definition: filter_design.txt:187
ff_af_adenorm
AVFilter ff_af_adenorm
Definition: af_adenorm.c:321
dc_denorm_dblp
static void dc_denorm_dblp(AVFilterContext *ctx, void *dstp, const void *srcp, int nb_samples)
Definition: af_adenorm.c:88
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:421
channel_layout.h
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:70
SQ_TYPE
@ SQ_TYPE
Definition: af_adenorm.c:29
AVFilterContext
An instance of a filter.
Definition: avfilter.h:341
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:117
audio.h
M_LN10
#define M_LN10
Definition: mathematics.h:43
ThreadData::in
AVFrame * in
Definition: af_adenorm.c:223
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:143
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:561
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:575
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
ff_set_common_channel_layouts
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *channel_layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates.
Definition: formats.c:568