FFmpeg  4.4.4
af_stereotools.c
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1 /*
2  * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen
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 
22 #include "libavutil/opt.h"
23 #include "avfilter.h"
24 #include "audio.h"
25 #include "formats.h"
26 
27 typedef struct StereoToolsContext {
28  const AVClass *class;
29 
30  int softclip;
31  int mute_l;
32  int mute_r;
33  int phase_l;
34  int phase_r;
35  int mode;
36  int bmode_in;
37  int bmode_out;
38  double slev;
39  double sbal;
40  double mlev;
41  double mpan;
42  double phase;
43  double base;
44  double delay;
45  double balance_in;
46  double balance_out;
49  double sc_level;
51  double level_in;
52  double level_out;
53 
54  double *buffer;
55  int length;
56  int pos;
58 
59 #define OFFSET(x) offsetof(StereoToolsContext, x)
60 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
61 
62 static const AVOption stereotools_options[] = {
63  { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
64  { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
65  { "balance_in", "set balance in", OFFSET(balance_in), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
66  { "balance_out", "set balance out", OFFSET(balance_out), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
67  { "softclip", "enable softclip", OFFSET(softclip), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
68  { "mutel", "mute L", OFFSET(mute_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
69  { "muter", "mute R", OFFSET(mute_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
70  { "phasel", "phase L", OFFSET(phase_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
71  { "phaser", "phase R", OFFSET(phase_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
72  { "mode", "set stereo mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 10, A, "mode" },
73  { "lr>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" },
74  { "lr>ms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" },
75  { "ms>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "mode" },
76  { "lr>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "mode" },
77  { "lr>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, A, "mode" },
78  { "lr>l+r", 0, 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, A, "mode" },
79  { "lr>rl", 0, 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, A, "mode" },
80  { "ms>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, A, "mode" },
81  { "ms>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, A, "mode" },
82  { "ms>rl", 0, 0, AV_OPT_TYPE_CONST, {.i64=9}, 0, 0, A, "mode" },
83  { "lr>l-r", 0, 0, AV_OPT_TYPE_CONST, {.i64=10}, 0, 0, A, "mode" },
84  { "slev", "set side level", OFFSET(slev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
85  { "sbal", "set side balance", OFFSET(sbal), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
86  { "mlev", "set middle level", OFFSET(mlev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
87  { "mpan", "set middle pan", OFFSET(mpan), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
88  { "base", "set stereo base", OFFSET(base), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
89  { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -20, 20, A },
90  { "sclevel", "set S/C level", OFFSET(sc_level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 100, A },
91  { "phase", "set stereo phase", OFFSET(phase), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 360, A },
92  { "bmode_in", "set balance in mode", OFFSET(bmode_in), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
93  { "balance", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "bmode" },
94  { "amplitude", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "bmode" },
95  { "power", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "bmode" },
96  { "bmode_out", "set balance out mode", OFFSET(bmode_out), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
97  { NULL }
98 };
99 
101 
103 {
106  int ret;
107 
108  if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
109  (ret = ff_set_common_formats (ctx , formats )) < 0 ||
111  (ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
112  return ret;
113 
116 }
117 
118 static int config_input(AVFilterLink *inlink)
119 {
120  AVFilterContext *ctx = inlink->dst;
121  StereoToolsContext *s = ctx->priv;
122 
123  s->length = FFALIGN(inlink->sample_rate / 10, 2);
124  if (!s->buffer)
125  s->buffer = av_calloc(s->length, sizeof(*s->buffer));
126  if (!s->buffer)
127  return AVERROR(ENOMEM);
128 
129  s->inv_atan_shape = 1.0 / atan(s->sc_level);
130  s->phase_cos_coef = cos(s->phase / 180 * M_PI);
131  s->phase_sin_coef = sin(s->phase / 180 * M_PI);
132 
133  return 0;
134 }
135 
136 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
137 {
138  AVFilterContext *ctx = inlink->dst;
139  AVFilterLink *outlink = ctx->outputs[0];
140  StereoToolsContext *s = ctx->priv;
141  const double *src = (const double *)in->data[0];
142  const double sb = s->base < 0 ? s->base * 0.5 : s->base;
143  const double sbal = 1 + s->sbal;
144  const double mpan = 1 + s->mpan;
145  const double slev = s->slev;
146  const double mlev = s->mlev;
147  const double balance_in = s->balance_in;
148  const double balance_out = s->balance_out;
149  const double level_in = s->level_in;
150  const double level_out = s->level_out;
151  const double sc_level = s->sc_level;
152  const double delay = s->delay;
153  const int length = s->length;
154  const int mute_l = s->mute_l;
155  const int mute_r = s->mute_r;
156  const int phase_l = s->phase_l;
157  const int phase_r = s->phase_r;
158  double *buffer = s->buffer;
159  AVFrame *out;
160  double *dst;
161  int nbuf = inlink->sample_rate * (fabs(delay) / 1000.);
162  int n;
163 
164  nbuf -= nbuf % 2;
165  if (av_frame_is_writable(in)) {
166  out = in;
167  } else {
168  out = ff_get_audio_buffer(outlink, in->nb_samples);
169  if (!out) {
170  av_frame_free(&in);
171  return AVERROR(ENOMEM);
172  }
174  }
175  dst = (double *)out->data[0];
176 
177  for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
178  double L = src[0], R = src[1], l, r, m, S, gl, gr, gd;
179 
180  L *= level_in;
181  R *= level_in;
182 
183  gl = 1. - FFMAX(0., balance_in);
184  gr = 1. + FFMIN(0., balance_in);
185  switch (s->bmode_in) {
186  case 1:
187  gd = gl - gr;
188  gl = 1. + gd;
189  gr = 1. - gd;
190  break;
191  case 2:
192  if (balance_in < 0.) {
193  gr = FFMAX(0.5, gr);
194  gl = 1. / gr;
195  } else if (balance_in > 0.) {
196  gl = FFMAX(0.5, gl);
197  gr = 1. / gl;
198  }
199  break;
200  }
201  L *= gl;
202  R *= gr;
203 
204  if (s->softclip) {
205  R = s->inv_atan_shape * atan(R * sc_level);
206  L = s->inv_atan_shape * atan(L * sc_level);
207  }
208 
209  switch (s->mode) {
210  case 0:
211  m = (L + R) * 0.5;
212  S = (L - R) * 0.5;
213  l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
214  r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
215  L = l;
216  R = r;
217  break;
218  case 1:
219  l = L * FFMIN(1., 2. - sbal);
220  r = R * FFMIN(1., sbal);
221  L = 0.5 * (l + r) * mlev;
222  R = 0.5 * (l - r) * slev;
223  break;
224  case 2:
225  l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
226  r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
227  L = l;
228  R = r;
229  break;
230  case 3:
231  R = L;
232  break;
233  case 4:
234  L = R;
235  break;
236  case 5:
237  L = (L + R) * 0.5;
238  R = L;
239  break;
240  case 6:
241  l = L;
242  L = R;
243  R = l;
244  m = (L + R) * 0.5;
245  S = (L - R) * 0.5;
246  l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
247  r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
248  L = l;
249  R = r;
250  break;
251  case 7:
252  l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
253  L = l;
254  R = l;
255  break;
256  case 8:
257  r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
258  L = r;
259  R = r;
260  break;
261  case 9:
262  l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
263  r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
264  L = r;
265  R = l;
266  break;
267  case 10:
268  L = (L - R) * 0.5;
269  R = L;
270  break;
271  }
272 
273  L *= 1. - mute_l;
274  R *= 1. - mute_r;
275 
276  L *= (2. * (1. - phase_l)) - 1.;
277  R *= (2. * (1. - phase_r)) - 1.;
278 
279  buffer[s->pos ] = L;
280  buffer[s->pos+1] = R;
281 
282  if (delay > 0.) {
283  R = buffer[(s->pos - (int)nbuf + 1 + length) % length];
284  } else if (delay < 0.) {
285  L = buffer[(s->pos - (int)nbuf + length) % length];
286  }
287 
288  l = L + sb * L - sb * R;
289  r = R + sb * R - sb * L;
290 
291  L = l;
292  R = r;
293 
294  l = L * s->phase_cos_coef - R * s->phase_sin_coef;
295  r = L * s->phase_sin_coef + R * s->phase_cos_coef;
296 
297  L = l;
298  R = r;
299 
300  s->pos = (s->pos + 2) % s->length;
301 
302  gl = 1. - FFMAX(0., balance_out);
303  gr = 1. + FFMIN(0., balance_out);
304  switch (s->bmode_out) {
305  case 1:
306  gd = gl - gr;
307  gl = 1. + gd;
308  gr = 1. - gd;
309  break;
310  case 2:
311  if (balance_out < 0.) {
312  gr = FFMAX(0.5, gr);
313  gl = 1. / gr;
314  } else if (balance_out > 0.) {
315  gl = FFMAX(0.5, gl);
316  gr = 1. / gl;
317  }
318  break;
319  }
320  L *= gl;
321  R *= gr;
322 
323 
324  L *= level_out;
325  R *= level_out;
326 
327  if (ctx->is_disabled) {
328  dst[0] = src[0];
329  dst[1] = src[1];
330  } else {
331  dst[0] = L;
332  dst[1] = R;
333  }
334  }
335 
336  if (out != in)
337  av_frame_free(&in);
338  return ff_filter_frame(outlink, out);
339 }
340 
341 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
342  char *res, int res_len, int flags)
343 {
344  int ret;
345 
346  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
347  if (ret < 0)
348  return ret;
349 
350  return config_input(ctx->inputs[0]);
351 }
352 
354 {
355  StereoToolsContext *s = ctx->priv;
356 
357  av_freep(&s->buffer);
358 }
359 
360 static const AVFilterPad inputs[] = {
361  {
362  .name = "default",
363  .type = AVMEDIA_TYPE_AUDIO,
364  .filter_frame = filter_frame,
365  .config_props = config_input,
366  },
367  { NULL }
368 };
369 
370 static const AVFilterPad outputs[] = {
371  {
372  .name = "default",
373  .type = AVMEDIA_TYPE_AUDIO,
374  },
375  { NULL }
376 };
377 
379  .name = "stereotools",
380  .description = NULL_IF_CONFIG_SMALL("Apply various stereo tools."),
381  .query_formats = query_formats,
382  .priv_size = sizeof(StereoToolsContext),
383  .priv_class = &stereotools_class,
384  .uninit = uninit,
385  .inputs = inputs,
386  .outputs = outputs,
389 };
AVFilter ff_af_stereotools
AVFILTER_DEFINE_CLASS(stereotools)
static int query_formats(AVFilterContext *ctx)
static int config_input(AVFilterLink *inlink)
static const AVFilterPad inputs[]
static const AVOption stereotools_options[]
static const AVFilterPad outputs[]
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
#define A
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
static av_cold void uninit(AVFilterContext *ctx)
#define OFFSET(x)
#define L(x)
Definition: vp56_arith.h:36
#define av_cold
Definition: attributes.h:88
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
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
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1096
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
Main libavfilter public API header.
#define flags(name, subs,...)
Definition: cbs_av1.c:561
#define s(width, name)
Definition: cbs_vp9.c:257
uint64_t layout
audio channel layout utility functions
#define FFMIN(a, b)
Definition: common.h:105
#define FFMAX(a, b)
Definition: common.h:103
#define NULL
Definition: coverity.c:32
static __device__ float fabs(float a)
Definition: cuda_runtime.h:182
mode
Use these values in ebur128_init (or'ed).
Definition: ebur128.h:83
int
#define S(s, c, i)
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:338
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
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:332
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:575
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
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:421
@ AV_OPT_TYPE_CONST
Definition: opt.h:234
@ AV_OPT_TYPE_INT
Definition: opt.h:225
@ AV_OPT_TYPE_DOUBLE
Definition: opt.h:227
@ AV_OPT_TYPE_BOOL
Definition: opt.h:242
#define AV_CH_LAYOUT_STEREO
#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:134
#define AVERROR(e)
Definition: error.h:43
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:658
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:245
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AV_SAMPLE_FMT_DBL
double
Definition: samplefmt.h:64
for(j=16;j >0;--j)
#define R
Definition: huffyuvdsp.h:34
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define FFALIGN(x, a)
Definition: macros.h:48
#define M_PI
Definition: mathematics.h:52
AVOptions.
static char buffer[20]
Definition: seek.c:32
formats
Definition: signature.h:48
Describe the class of an AVClass context structure.
Definition: log.h:67
A list of supported channel layouts.
Definition: formats.h:86
An instance of a filter.
Definition: avfilter.h:341
A list of supported formats for one end of a filter link.
Definition: formats.h:65
A filter pad used for either input or output.
Definition: internal.h:54
const char * name
Pad name.
Definition: internal.h:60
Filter definition.
Definition: avfilter.h:145
const char * name
Filter name.
Definition: avfilter.h:149
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
AVOption.
Definition: opt.h:248
#define av_freep(p)
#define src
Definition: vp8dsp.c:255
FILE * out
Definition: movenc.c:54
AVFormatContext * ctx
Definition: movenc.c:48
const char * r
Definition: vf_curves.c:116
uint8_t base
Definition: vp3data.h:141