FFmpeg
af_acrossover.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 /**
20  * @file
21  * Crossover filter
22  *
23  * Split an audio stream into several bands.
24  */
25 
26 #include "libavutil/attributes.h"
27 #include "libavutil/avstring.h"
29 #include "libavutil/eval.h"
30 #include "libavutil/internal.h"
31 #include "libavutil/opt.h"
32 
33 #include "audio.h"
34 #include "avfilter.h"
35 #include "formats.h"
36 #include "internal.h"
37 
38 #define MAX_SPLITS 16
39 #define MAX_BANDS MAX_SPLITS + 1
40 
41 typedef struct BiquadContext {
42  double a0, a1, a2;
43  double b1, b2;
44  double i1, i2;
45  double o1, o2;
47 
48 typedef struct CrossoverChannel {
52 
53 typedef struct AudioCrossoverContext {
54  const AVClass *class;
55 
56  char *splits_str;
57  int order;
58 
60  int nb_splits;
61  float *splits;
62 
64 
68 
69 #define OFFSET(x) offsetof(AudioCrossoverContext, x)
70 #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
71 
72 static const AVOption acrossover_options[] = {
73  { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
74  { "order", "set order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "m" },
75  { "2nd", "2nd order", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
76  { "4th", "4th order", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
77  { "8th", "8th order", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
78  { NULL }
79 };
80 
81 AVFILTER_DEFINE_CLASS(acrossover);
82 
84 {
86  char *p, *arg, *saveptr = NULL;
87  int i, ret = 0;
88 
89  s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
90  if (!s->splits)
91  return AVERROR(ENOMEM);
92 
93  p = s->splits_str;
94  for (i = 0; i < MAX_SPLITS; i++) {
95  float freq;
96 
97  if (!(arg = av_strtok(p, " |", &saveptr)))
98  break;
99 
100  p = NULL;
101 
102  if (av_sscanf(arg, "%f", &freq) != 1) {
103  av_log(ctx, AV_LOG_ERROR, "Invalid syntax for frequency[%d].\n", i);
104  return AVERROR(EINVAL);
105  }
106  if (freq <= 0) {
107  av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
108  return AVERROR(EINVAL);
109  }
110 
111  if (i > 0 && freq <= s->splits[i-1]) {
112  av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
113  return AVERROR(EINVAL);
114  }
115 
116  s->splits[i] = freq;
117  }
118 
119  s->nb_splits = i;
120 
121  for (i = 0; i <= s->nb_splits; i++) {
122  AVFilterPad pad = { 0 };
123  char *name;
124 
125  pad.type = AVMEDIA_TYPE_AUDIO;
126  name = av_asprintf("out%d", ctx->nb_outputs);
127  if (!name)
128  return AVERROR(ENOMEM);
129  pad.name = name;
130 
131  if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
132  av_freep(&pad.name);
133  return ret;
134  }
135  }
136 
137  return ret;
138 }
139 
140 static void set_lp(BiquadContext *b, double fc, double q, double sr)
141 {
142  double omega = 2.0 * M_PI * fc / sr;
143  double sn = sin(omega);
144  double cs = cos(omega);
145  double alpha = sn / (2. * q);
146  double inv = 1.0 / (1.0 + alpha);
147 
148  b->a0 = (1. - cs) * 0.5 * inv;
149  b->a1 = (1. - cs) * inv;
150  b->a2 = b->a0;
151  b->b1 = -2. * cs * inv;
152  b->b2 = (1. - alpha) * inv;
153 }
154 
155 static void set_hp(BiquadContext *b, double fc, double q, double sr)
156 {
157  double omega = 2 * M_PI * fc / sr;
158  double sn = sin(omega);
159  double cs = cos(omega);
160  double alpha = sn / (2 * q);
161  double inv = 1.0 / (1.0 + alpha);
162 
163  b->a0 = inv * (1. + cs) / 2.;
164  b->a1 = -2. * b->a0;
165  b->a2 = b->a0;
166  b->b1 = -2. * cs * inv;
167  b->b2 = (1. - alpha) * inv;
168 }
169 
171 {
172  AVFilterContext *ctx = inlink->dst;
173  AudioCrossoverContext *s = ctx->priv;
174  int ch, band, sample_rate = inlink->sample_rate;
175  double q;
176 
177  s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
178  if (!s->xover)
179  return AVERROR(ENOMEM);
180 
181  switch (s->order) {
182  case 0:
183  q = 0.5;
184  s->filter_count = 1;
185  break;
186  case 1:
187  q = M_SQRT1_2;
188  s->filter_count = 2;
189  break;
190  case 2:
191  q = 0.54;
192  s->filter_count = 4;
193  break;
194  }
195 
196  for (ch = 0; ch < inlink->channels; ch++) {
197  for (band = 0; band <= s->nb_splits; band++) {
198  set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);
199  set_hp(&s->xover[ch].hp[band][0], s->splits[band], q, sample_rate);
200 
201  if (s->order > 1) {
202  set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);
203  set_hp(&s->xover[ch].hp[band][1], s->splits[band], 1.34, sample_rate);
204  set_lp(&s->xover[ch].lp[band][2], s->splits[band], q, sample_rate);
205  set_hp(&s->xover[ch].hp[band][2], s->splits[band], q, sample_rate);
206  set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);
207  set_hp(&s->xover[ch].hp[band][3], s->splits[band], 1.34, sample_rate);
208  } else {
209  set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);
210  set_hp(&s->xover[ch].hp[band][1], s->splits[band], q, sample_rate);
211  }
212  }
213  }
214 
215  return 0;
216 }
217 
219 {
222  static const enum AVSampleFormat sample_fmts[] = {
225  };
226  int ret;
227 
228  layouts = ff_all_channel_counts();
229  if (!layouts)
230  return AVERROR(ENOMEM);
231  ret = ff_set_common_channel_layouts(ctx, layouts);
232  if (ret < 0)
233  return ret;
234 
235  formats = ff_make_format_list(sample_fmts);
236  if (!formats)
237  return AVERROR(ENOMEM);
238  ret = ff_set_common_formats(ctx, formats);
239  if (ret < 0)
240  return ret;
241 
242  formats = ff_all_samplerates();
243  if (!formats)
244  return AVERROR(ENOMEM);
245  return ff_set_common_samplerates(ctx, formats);
246 }
247 
248 static double biquad_process(BiquadContext *b, double in)
249 {
250  double out = in * b->a0 + b->i1 * b->a1 + b->i2 * b->a2 - b->o1 * b->b1 - b->o2 * b->b2;
251 
252  b->i2 = b->i1;
253  b->o2 = b->o1;
254  b->i1 = in;
255  b->o1 = out;
256 
257  return out;
258 }
259 
260 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
261 {
262  AudioCrossoverContext *s = ctx->priv;
263  AVFrame *in = s->input_frame;
264  AVFrame **frames = s->frames;
265  const int start = (in->channels * jobnr) / nb_jobs;
266  const int end = (in->channels * (jobnr+1)) / nb_jobs;
267  int f, band;
268 
269  for (int ch = start; ch < end; ch++) {
270  const double *src = (const double *)in->extended_data[ch];
271  CrossoverChannel *xover = &s->xover[ch];
272 
273  for (int i = 0; i < in->nb_samples; i++) {
274  double sample = src[i], lo, hi;
275 
276  for (band = 0; band < ctx->nb_outputs; band++) {
277  double *dst = (double *)frames[band]->extended_data[ch];
278 
279  lo = sample;
280  hi = sample;
281  for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
282  BiquadContext *lp = &xover->lp[band][f];
283  lo = biquad_process(lp, lo);
284  }
285 
286  for (f = 0; band + 1 < ctx->nb_outputs && f < s->filter_count; f++) {
287  BiquadContext *hp = &xover->hp[band][f];
288  hi = biquad_process(hp, hi);
289  }
290 
291  dst[i] = lo;
292 
293  sample = hi;
294  }
295  }
296  }
297 
298  return 0;
299 }
300 
302 {
303  AVFilterContext *ctx = inlink->dst;
304  AudioCrossoverContext *s = ctx->priv;
305  AVFrame **frames = s->frames;
306  int i, ret = 0;
307 
308  for (i = 0; i < ctx->nb_outputs; i++) {
309  frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
310 
311  if (!frames[i]) {
312  ret = AVERROR(ENOMEM);
313  break;
314  }
315 
316  frames[i]->pts = in->pts;
317  }
318 
319  if (ret < 0)
320  goto fail;
321 
322  s->input_frame = in;
323  ctx->internal->execute(ctx, filter_channels, NULL, NULL, FFMIN(inlink->channels,
325 
326  for (i = 0; i < ctx->nb_outputs; i++) {
327  ret = ff_filter_frame(ctx->outputs[i], frames[i]);
328  frames[i] = NULL;
329  if (ret < 0)
330  break;
331  }
332 
333 fail:
334  for (i = 0; i < ctx->nb_outputs; i++)
335  av_frame_free(&frames[i]);
336  av_frame_free(&in);
337  s->input_frame = NULL;
338 
339  return ret;
340 }
341 
343 {
344  AudioCrossoverContext *s = ctx->priv;
345  int i;
346 
347  av_freep(&s->splits);
348  av_freep(&s->xover);
349 
350  for (i = 0; i < ctx->nb_outputs; i++)
351  av_freep(&ctx->output_pads[i].name);
352 }
353 
354 static const AVFilterPad inputs[] = {
355  {
356  .name = "default",
357  .type = AVMEDIA_TYPE_AUDIO,
358  .filter_frame = filter_frame,
359  .config_props = config_input,
360  },
361  { NULL }
362 };
363 
365  .name = "acrossover",
366  .description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
367  .priv_size = sizeof(AudioCrossoverContext),
368  .priv_class = &acrossover_class,
369  .init = init,
370  .uninit = uninit,
372  .inputs = inputs,
373  .outputs = NULL,
376 };
#define NULL
Definition: coverity.c:32
This structure describes decoded (raw) audio or video data.
Definition: frame.h:308
static av_cold int init(AVFilterContext *ctx)
Definition: af_acrossover.c:83
AVOption.
Definition: opt.h:248
AVFrame * frames[MAX_BANDS]
Definition: af_acrossover.c:66
static int config_input(AVFilterLink *inlink)
static const AVFilterPad inputs[]
Main libavfilter public API header.
#define M_SQRT1_2
Definition: mathematics.h:58
double, planar
Definition: samplefmt.h:70
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:65
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:569
#define sample
if it could not because there are no more frames
Macro definitions for various function/variable attributes.
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:245
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:287
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:1091
AVFilterPad * output_pads
array of output pads
Definition: avfilter.h:352
#define av_cold
Definition: attributes.h:88
AVOptions.
#define f(width, name)
Definition: cbs_vp9.c:255
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:92
BiquadContext lp[MAX_BANDS][4]
Definition: af_acrossover.c:49
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:401
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
Definition: avfilter.h:112
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:54
#define src
Definition: vp8dsp.c:254
#define fc(width, name, range_min, range_max)
Definition: cbs_av1.c:550
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
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:588
CrossoverChannel * xover
Definition: af_acrossover.c:63
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
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
unsigned nb_outputs
number of output pads
Definition: avfilter.h:354
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:153
void * priv
private data for use by the filter
Definition: avfilter.h:356
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:117
const char * arg
Definition: jacosubdec.c:66
#define fail()
Definition: checkasm.h:123
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
Definition: avsscanf.c:962
BiquadContext hp[MAX_BANDS][4]
Definition: af_acrossover.c:50
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:113
common internal API header
static double biquad_process(BiquadContext *b, double in)
int channels
number of audio channels, only used for audio.
Definition: frame.h:614
audio channel layout utility functions
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:800
static void set_hp(BiquadContext *b, double fc, double q, double sr)
#define FFMIN(a, b)
Definition: common.h:96
static int query_formats(AVFilterContext *ctx)
AVFormatContext * ctx
Definition: movenc.c:48
#define s(width, name)
Definition: cbs_vp9.c:257
static void set_lp(BiquadContext *b, double fc, double q, double sr)
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:86
sample_rate
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
static const int16_t alpha[]
Definition: ilbcdata.h:55
AVFILTER_DEFINE_CLASS(acrossover)
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:145
const char * name
Filter name.
Definition: avfilter.h:149
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:353
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:422
#define flags(name, subs,...)
Definition: cbs_av1.c:560
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
Definition: avfilter.h:381
AVFilter ff_af_acrossover
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
char * av_strtok(char *s, const char *delim, char **saveptr)
Split the string into several tokens which can be accessed by successive calls to av_strtok()...
Definition: avstring.c:186
static av_cold void uninit(AVFilterContext *ctx)
avfilter_execute_func * execute
Definition: internal.h:136
#define AF
Definition: af_acrossover.c:70
double b[3]
Definition: af_aiir.c:42
A list of supported formats for one end of a filter link.
Definition: formats.h:65
#define OFFSET(x)
Definition: af_acrossover.c:69
An instance of a filter.
Definition: avfilter.h:341
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:884
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
FILE * out
Definition: movenc.c:54
#define MAX_SPLITS
Definition: af_acrossover.c:38
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
formats
Definition: signature.h:48
internal API functions
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
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:437
static const AVOption acrossover_options[]
Definition: af_acrossover.c:72
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:355
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
Definition: internal.h:248
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:374
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:576
int i
Definition: input.c:407
simple arithmetic expression evaluator
const char * name
Definition: opengl_enc.c:102
#define MAX_BANDS
Definition: af_acrossover.c:39