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 
65 
66 #define OFFSET(x) offsetof(AudioCrossoverContext, x)
67 #define AF AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
68 
69 static const AVOption acrossover_options[] = {
70  { "split", "set split frequencies", OFFSET(splits_str), AV_OPT_TYPE_STRING, {.str="500"}, 0, 0, AF },
71  { "order", "set order", OFFSET(order), AV_OPT_TYPE_INT, {.i64=1}, 0, 2, AF, "m" },
72  { "2nd", "2nd order", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, AF, "m" },
73  { "4th", "4th order", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, AF, "m" },
74  { "8th", "8th order", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, AF, "m" },
75  { NULL }
76 };
77 
78 AVFILTER_DEFINE_CLASS(acrossover);
79 
81 {
83  char *p, *arg, *saveptr = NULL;
84  int i, ret = 0;
85 
86  s->splits = av_calloc(MAX_SPLITS, sizeof(*s->splits));
87  if (!s->splits)
88  return AVERROR(ENOMEM);
89 
90  p = s->splits_str;
91  for (i = 0; i < MAX_SPLITS; i++) {
92  float freq;
93 
94  if (!(arg = av_strtok(p, " |", &saveptr)))
95  break;
96 
97  p = NULL;
98 
99  av_sscanf(arg, "%f", &freq);
100  if (freq <= 0) {
101  av_log(ctx, AV_LOG_ERROR, "Frequency %f must be positive number.\n", freq);
102  return AVERROR(EINVAL);
103  }
104 
105  if (i > 0 && freq <= s->splits[i-1]) {
106  av_log(ctx, AV_LOG_ERROR, "Frequency %f must be in increasing order.\n", freq);
107  return AVERROR(EINVAL);
108  }
109 
110  s->splits[i] = freq;
111  }
112 
113  s->nb_splits = i;
114 
115  for (i = 0; i <= s->nb_splits; i++) {
116  AVFilterPad pad = { 0 };
117  char *name;
118 
119  pad.type = AVMEDIA_TYPE_AUDIO;
120  name = av_asprintf("out%d", ctx->nb_outputs);
121  if (!name)
122  return AVERROR(ENOMEM);
123  pad.name = name;
124 
125  if ((ret = ff_insert_outpad(ctx, i, &pad)) < 0) {
126  av_freep(&pad.name);
127  return ret;
128  }
129  }
130 
131  return ret;
132 }
133 
134 static void set_lp(BiquadContext *b, float fc, float q, float sr)
135 {
136  double omega = (2.0 * M_PI * fc / sr);
137  double sn = sin(omega);
138  double cs = cos(omega);
139  double alpha = (sn / (2 * q));
140  double inv = (1.0 / (1.0 + alpha));
141 
142  b->a2 = b->a0 = (inv * (1.0 - cs) * 0.5);
143  b->a1 = b->a0 + b->a0;
144  b->b1 = -2. * cs * inv;
145  b->b2 = (1. - alpha) * inv;
146 }
147 
148 static void set_hp(BiquadContext *b, float fc, float q, float sr)
149 {
150  double omega = 2 * M_PI * fc / sr;
151  double sn = sin(omega);
152  double cs = cos(omega);
153  double alpha = sn / (2 * q);
154  double inv = 1.0 / (1.0 + alpha);
155 
156  b->a0 = inv * (1. + cs) / 2.;
157  b->a1 = -2. * b->a0;
158  b->a2 = b->a0;
159  b->b1 = -2. * cs * inv;
160  b->b2 = (1. - alpha) * inv;
161 }
162 
164 {
165  AVFilterContext *ctx = inlink->dst;
166  AudioCrossoverContext *s = ctx->priv;
167  int ch, band, sample_rate = inlink->sample_rate;
168  double q;
169 
170  s->xover = av_calloc(inlink->channels, sizeof(*s->xover));
171  if (!s->xover)
172  return AVERROR(ENOMEM);
173 
174  switch (s->order) {
175  case 0:
176  q = 0.5;
177  s->filter_count = 1;
178  break;
179  case 1:
180  q = M_SQRT1_2;
181  s->filter_count = 2;
182  break;
183  case 2:
184  q = 0.54;
185  s->filter_count = 4;
186  break;
187  }
188 
189  for (ch = 0; ch < inlink->channels; ch++) {
190  for (band = 0; band <= s->nb_splits; band++) {
191  set_lp(&s->xover[ch].lp[band][0], s->splits[band], q, sample_rate);
192  set_hp(&s->xover[ch].hp[band][0], s->splits[band], q, sample_rate);
193 
194  if (s->order > 1) {
195  set_lp(&s->xover[ch].lp[band][1], s->splits[band], 1.34, sample_rate);
196  set_hp(&s->xover[ch].hp[band][1], s->splits[band], 1.34, sample_rate);
197  set_lp(&s->xover[ch].lp[band][2], s->splits[band], q, sample_rate);
198  set_hp(&s->xover[ch].hp[band][2], s->splits[band], q, sample_rate);
199  set_lp(&s->xover[ch].lp[band][3], s->splits[band], 1.34, sample_rate);
200  set_hp(&s->xover[ch].hp[band][3], s->splits[band], 1.34, sample_rate);
201  } else {
202  set_lp(&s->xover[ch].lp[band][1], s->splits[band], q, sample_rate);
203  set_hp(&s->xover[ch].hp[band][1], s->splits[band], q, sample_rate);
204  }
205  }
206  }
207 
208  return 0;
209 }
210 
212 {
215  static const enum AVSampleFormat sample_fmts[] = {
218  };
219  int ret;
220 
221  layouts = ff_all_channel_counts();
222  if (!layouts)
223  return AVERROR(ENOMEM);
224  ret = ff_set_common_channel_layouts(ctx, layouts);
225  if (ret < 0)
226  return ret;
227 
228  formats = ff_make_format_list(sample_fmts);
229  if (!formats)
230  return AVERROR(ENOMEM);
231  ret = ff_set_common_formats(ctx, formats);
232  if (ret < 0)
233  return ret;
234 
235  formats = ff_all_samplerates();
236  if (!formats)
237  return AVERROR(ENOMEM);
238  return ff_set_common_samplerates(ctx, formats);
239 }
240 
241 static double biquad_process(BiquadContext *b, double in)
242 {
243  double out = in * b->a0 + b->i1 * b->a1 + b->i2 * b->a2 - b->o1 * b->b1 - b->o2 * b->b2;
244 
245  b->i2 = b->i1;
246  b->o2 = b->o1;
247  b->i1 = in;
248  b->o1 = out;
249 
250  return out;
251 }
252 
254 {
255  AVFilterContext *ctx = inlink->dst;
256  AudioCrossoverContext *s = ctx->priv;
257  AVFrame *frames[MAX_BANDS] = { NULL };
258  int i, f, ch, band, ret = 0;
259 
260  for (i = 0; i < ctx->nb_outputs; i++) {
261  frames[i] = ff_get_audio_buffer(ctx->outputs[i], in->nb_samples);
262 
263  if (!frames[i]) {
264  ret = AVERROR(ENOMEM);
265  break;
266  }
267 
268  frames[i]->pts = in->pts;
269  }
270 
271  if (ret < 0)
272  goto fail;
273 
274  for (ch = 0; ch < inlink->channels; ch++) {
275  const double *src = (const double *)in->extended_data[ch];
276  CrossoverChannel *xover = &s->xover[ch];
277 
278  for (band = 0; band < ctx->nb_outputs; band++) {
279  double *dst = (double *)frames[band]->extended_data[ch];
280 
281  for (i = 0; i < in->nb_samples; i++) {
282  dst[i] = src[i];
283 
284  for (f = 0; f < s->filter_count; f++) {
285  if (band + 1 < ctx->nb_outputs) {
286  BiquadContext *lp = &xover->lp[band][f];
287  dst[i] = biquad_process(lp, dst[i]);
288  }
289 
290  if (band - 1 >= 0) {
291  BiquadContext *hp = &xover->hp[band - 1][f];
292  dst[i] = biquad_process(hp, dst[i]);
293  }
294  }
295  }
296  }
297  }
298 
299  for (i = 0; i < ctx->nb_outputs; i++) {
300  ret = ff_filter_frame(ctx->outputs[i], frames[i]);
301  if (ret < 0)
302  break;
303  }
304 
305 fail:
306  av_frame_free(&in);
307 
308  return ret;
309 }
310 
312 {
313  AudioCrossoverContext *s = ctx->priv;
314  int i;
315 
316  av_freep(&s->splits);
317 
318  for (i = 0; i < ctx->nb_outputs; i++)
319  av_freep(&ctx->output_pads[i].name);
320 }
321 
322 static const AVFilterPad inputs[] = {
323  {
324  .name = "default",
325  .type = AVMEDIA_TYPE_AUDIO,
326  .filter_frame = filter_frame,
327  .config_props = config_input,
328  },
329  { NULL }
330 };
331 
333  .name = "acrossover",
334  .description = NULL_IF_CONFIG_SMALL("Split audio into per-bands streams."),
335  .priv_size = sizeof(AudioCrossoverContext),
336  .priv_class = &acrossover_class,
337  .init = init,
338  .uninit = uninit,
340  .inputs = inputs,
341  .outputs = NULL,
343 };
#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:549
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
static av_cold int init(AVFilterContext *ctx)
Definition: af_acrossover.c:80
AVOption.
Definition: opt.h:246
static int config_input(AVFilterLink *inlink)
static const AVFilterPad inputs[]
Main libavfilter public API header.
#define M_SQRT1_2
Definition: mathematics.h:58
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){int planes=out->planar?out->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
double, planar
Definition: samplefmt.h:70
static void set_lp(BiquadContext *b, float fc, float q, float sr)
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:65
#define src
Definition: vp8dsp.c:254
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:244
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
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:1080
AVFilterPad * output_pads
array of output pads
Definition: avfilter.h:349
#define av_cold
Definition: attributes.h:82
AVOptions.
#define f(width, name)
Definition: cbs_vp9.c:255
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:388
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
Definition: avfilter.h:111
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:54
#define fc(width, name, range_min, range_max)
Definition: cbs_av1.c:551
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
#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:568
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:202
unsigned nb_outputs
number of output pads
Definition: avfilter.h:351
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
const char * arg
Definition: jacosubdec.c:66
#define fail()
Definition: checkasm.h:121
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)
#define b
Definition: input.c:41
audio channel layout utility functions
static int query_formats(AVFilterContext *ctx)
AVFormatContext * ctx
Definition: movenc.c:48
#define s(width, name)
Definition: cbs_vp9.c:257
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:85
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:144
const char * name
Filter name.
Definition: avfilter.h:148
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
#define flags(name, subs,...)
Definition: cbs_av1.c:561
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:184
static av_cold void uninit(AVFilterContext *ctx)
static void set_hp(BiquadContext *b, float fc, float q, float sr)
#define AF
Definition: af_acrossover.c:67
A list of supported formats for one end of a filter link.
Definition: formats.h:64
#define OFFSET(x)
Definition: af_acrossover.c:66
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
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:410
static const AVOption acrossover_options[]
Definition: af_acrossover.c:69
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
Definition: internal.h:285
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
simple arithmetic expression evaluator
const char * name
Definition: opengl_enc.c:102
#define MAX_BANDS
Definition: af_acrossover.c:39