FFmpeg
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
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, 8, 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  { "slev", "set side level", OFFSET(slev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
83  { "sbal", "set side balance", OFFSET(sbal), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
84  { "mlev", "set middle level", OFFSET(mlev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
85  { "mpan", "set middle pan", OFFSET(mpan), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
86  { "base", "set stereo base", OFFSET(base), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
87  { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -20, 20, A },
88  { "sclevel", "set S/C level", OFFSET(sc_level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 100, A },
89  { "phase", "set stereo phase", OFFSET(phase), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 360, A },
90  { "bmode_in", "set balance in mode", OFFSET(bmode_in), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
91  { "balance", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "bmode" },
92  { "amplitude", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "bmode" },
93  { "power", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "bmode" },
94  { "bmode_out", "set balance out mode", OFFSET(bmode_out), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
95  { NULL }
96 };
97 
98 AVFILTER_DEFINE_CLASS(stereotools);
99 
101 {
104  int ret;
105 
106  if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
107  (ret = ff_set_common_formats (ctx , formats )) < 0 ||
108  (ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
109  (ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
110  return ret;
111 
112  formats = ff_all_samplerates();
113  return ff_set_common_samplerates(ctx, formats);
114 }
115 
117 {
118  AVFilterContext *ctx = inlink->dst;
119  StereoToolsContext *s = ctx->priv;
120 
121  s->length = 2 * inlink->sample_rate * 0.05;
122  if (s->length <= 1 || s->length & 1) {
123  av_log(ctx, AV_LOG_ERROR, "sample rate is too small\n");
124  return AVERROR(EINVAL);
125  }
126  s->buffer = av_calloc(s->length, sizeof(*s->buffer));
127  if (!s->buffer)
128  return AVERROR(ENOMEM);
129 
130  s->inv_atan_shape = 1.0 / atan(s->sc_level);
131  s->phase_cos_coef = cos(s->phase / 180 * M_PI);
132  s->phase_sin_coef = sin(s->phase / 180 * M_PI);
133 
134  return 0;
135 }
136 
138 {
139  AVFilterContext *ctx = inlink->dst;
140  AVFilterLink *outlink = ctx->outputs[0];
141  StereoToolsContext *s = ctx->priv;
142  const double *src = (const double *)in->data[0];
143  const double sb = s->base < 0 ? s->base * 0.5 : s->base;
144  const double sbal = 1 + s->sbal;
145  const double mpan = 1 + s->mpan;
146  const double slev = s->slev;
147  const double mlev = s->mlev;
148  const double balance_in = s->balance_in;
149  const double balance_out = s->balance_out;
150  const double level_in = s->level_in;
151  const double level_out = s->level_out;
152  const double sc_level = s->sc_level;
153  const double delay = s->delay;
154  const int length = s->length;
155  const int mute_l = s->mute_l;
156  const int mute_r = s->mute_r;
157  const int phase_l = s->phase_l;
158  const int phase_r = s->phase_r;
159  double *buffer = s->buffer;
160  AVFrame *out;
161  double *dst;
162  int nbuf = inlink->sample_rate * (fabs(delay) / 1000.);
163  int n;
164 
165  nbuf -= nbuf % 2;
166  if (av_frame_is_writable(in)) {
167  out = in;
168  } else {
169  out = ff_get_audio_buffer(outlink, in->nb_samples);
170  if (!out) {
171  av_frame_free(&in);
172  return AVERROR(ENOMEM);
173  }
175  }
176  dst = (double *)out->data[0];
177 
178  for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
179  double L = src[0], R = src[1], l, r, m, S, gl, gr, gd;
180 
181  L *= level_in;
182  R *= level_in;
183 
184  gl = 1. - FFMAX(0., balance_in);
185  gr = 1. + FFMIN(0., balance_in);
186  switch (s->bmode_in) {
187  case 1:
188  gd = gl - gr;
189  gl = 1. + gd;
190  gr = 1. - gd;
191  break;
192  case 2:
193  if (balance_in < 0.) {
194  gr = FFMAX(0.5, gr);
195  gl = 1. / gr;
196  } else if (balance_in > 0.) {
197  gl = FFMAX(0.5, gl);
198  gr = 1. / gl;
199  }
200  break;
201  }
202  L *= gl;
203  R *= gr;
204 
205  if (s->softclip) {
206  R = s->inv_atan_shape * atan(R * sc_level);
207  L = s->inv_atan_shape * atan(L * sc_level);
208  }
209 
210  switch (s->mode) {
211  case 0:
212  m = (L + R) * 0.5;
213  S = (L - R) * 0.5;
214  l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
215  r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
216  L = l;
217  R = r;
218  break;
219  case 1:
220  l = L * FFMIN(1., 2. - sbal);
221  r = R * FFMIN(1., sbal);
222  L = 0.5 * (l + r) * mlev;
223  R = 0.5 * (l - r) * slev;
224  break;
225  case 2:
226  l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
227  r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
228  L = l;
229  R = r;
230  break;
231  case 3:
232  R = L;
233  break;
234  case 4:
235  L = R;
236  break;
237  case 5:
238  L = (L + R) / 2;
239  R = L;
240  break;
241  case 6:
242  l = L;
243  L = R;
244  R = l;
245  m = (L + R) * 0.5;
246  S = (L - R) * 0.5;
247  l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
248  r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
249  L = l;
250  R = r;
251  break;
252  case 7:
253  l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
254  L = l;
255  R = l;
256  break;
257  case 8:
258  r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
259  L = r;
260  R = r;
261  break;
262  }
263 
264  L *= 1. - mute_l;
265  R *= 1. - mute_r;
266 
267  L *= (2. * (1. - phase_l)) - 1.;
268  R *= (2. * (1. - phase_r)) - 1.;
269 
270  buffer[s->pos ] = L;
271  buffer[s->pos+1] = R;
272 
273  if (delay > 0.) {
274  R = buffer[(s->pos - (int)nbuf + 1 + length) % length];
275  } else if (delay < 0.) {
276  L = buffer[(s->pos - (int)nbuf + length) % length];
277  }
278 
279  l = L + sb * L - sb * R;
280  r = R + sb * R - sb * L;
281 
282  L = l;
283  R = r;
284 
285  l = L * s->phase_cos_coef - R * s->phase_sin_coef;
286  r = L * s->phase_sin_coef + R * s->phase_cos_coef;
287 
288  L = l;
289  R = r;
290 
291  s->pos = (s->pos + 2) % s->length;
292 
293  gl = 1. - FFMAX(0., balance_out);
294  gr = 1. + FFMIN(0., balance_out);
295  switch (s->bmode_out) {
296  case 1:
297  gd = gl - gr;
298  gl = 1. + gd;
299  gr = 1. - gd;
300  break;
301  case 2:
302  if (balance_out < 0.) {
303  gr = FFMAX(0.5, gr);
304  gl = 1. / gr;
305  } else if (balance_out > 0.) {
306  gl = FFMAX(0.5, gl);
307  gr = 1. / gl;
308  }
309  break;
310  }
311  L *= gl;
312  R *= gr;
313 
314 
315  L *= level_out;
316  R *= level_out;
317 
318  dst[0] = L;
319  dst[1] = R;
320  }
321 
322  if (out != in)
323  av_frame_free(&in);
324  return ff_filter_frame(outlink, out);
325 }
326 
328 {
329  StereoToolsContext *s = ctx->priv;
330 
331  av_freep(&s->buffer);
332 }
333 
334 static const AVFilterPad inputs[] = {
335  {
336  .name = "default",
337  .type = AVMEDIA_TYPE_AUDIO,
338  .filter_frame = filter_frame,
339  .config_props = config_input,
340  },
341  { NULL }
342 };
343 
344 static const AVFilterPad outputs[] = {
345  {
346  .name = "default",
347  .type = AVMEDIA_TYPE_AUDIO,
348  },
349  { NULL }
350 };
351 
353  .name = "stereotools",
354  .description = NULL_IF_CONFIG_SMALL("Apply various stereo tools."),
355  .query_formats = query_formats,
356  .priv_size = sizeof(StereoToolsContext),
357  .priv_class = &stereotools_class,
358  .uninit = uninit,
359  .inputs = inputs,
360  .outputs = outputs,
361 };
static const AVOption stereotools_options[]
#define NULL
Definition: coverity.c:32
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:550
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVOption.
Definition: opt.h:246
Main libavfilter public API header.
#define AV_CH_LAYOUT_STEREO
#define src
Definition: vp8dsp.c:254
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
const char * name
Pad name.
Definition: internal.h:60
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1093
#define av_cold
Definition: attributes.h:82
AVOptions.
#define OFFSET(x)
#define av_log(a,...)
#define A
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
A filter pad used for either input or output.
Definition: internal.h:54
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
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:569
#define R
Definition: huffyuvdsp.h:34
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:342
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
#define S(s, c, i)
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
const char * r
Definition: vf_curves.c:114
void * priv
private data for use by the filter
Definition: avfilter.h:353
GLsizei GLsizei * length
Definition: opengl_enc.c:114
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:336
#define FFMAX(a, b)
Definition: common.h:94
AVFilter ff_af_stereotools
audio channel layout utility functions
static av_cold void uninit(AVFilterContext *ctx)
#define FFMIN(a, b)
Definition: common.h:96
AVFILTER_DEFINE_CLASS(stereotools)
AVFormatContext * ctx
Definition: movenc.c:48
static int query_formats(AVFilterContext *ctx)
#define s(width, name)
Definition: cbs_vp9.c:257
int n
Definition: avisynth_c.h:760
#define L(x)
Definition: vp56_arith.h:36
static const AVFilterPad outputs[]
static int config_input(AVFilterLink *inlink)
A list of supported channel layouts.
Definition: formats.h:85
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
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
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:394
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:309
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
static const AVFilterPad inputs[]
int
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:338
FILE * out
Definition: movenc.c:54
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
formats
Definition: signature.h:48
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
mode
Use these values in ebur128_init (or&#39;ed).
Definition: ebur128.h:83
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
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 layout
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:557
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654
GLuint buffer
Definition: opengl_enc.c:101