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 {
82  AudioCrossoverContext *s = ctx->priv;
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 
222  if (!layouts)
223  return AVERROR(ENOMEM);
225  if (ret < 0)
226  return ret;
227 
229  if (!formats)
230  return AVERROR(ENOMEM);
232  if (ret < 0)
233  return ret;
234 
236  if (!formats)
237  return AVERROR(ENOMEM);
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 };
formats
formats
Definition: signature.h:48
BiquadContext::a2
double a2
Definition: af_acrossover.c:42
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
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:85
OFFSET
#define OFFSET(x)
Definition: af_acrossover.c:66
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:283
out
FILE * out
Definition: movenc.c:54
ff_set_common_channel_layouts
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
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
sample_fmts
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:686
CrossoverChannel
Definition: af_acrossover.c:48
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
ch
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
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_asprintf
char * av_asprintf(const char *fmt,...)
Definition: avstring.c:113
av_frame_free
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
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:410
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
name
const char * name
Definition: avisynth_c.h:867
AVOption
AVOption.
Definition: opt.h:246
b
#define b
Definition: input.c:41
MAX_SPLITS
#define MAX_SPLITS
Definition: af_acrossover.c:38
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(acrossover)
fc
#define fc(width, name, range_min, range_max)
Definition: cbs_av1.c:555
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:148
CrossoverChannel::lp
BiquadContext lp[MAX_BANDS][4]
Definition: af_acrossover.c:49
sample_rate
sample_rate
Definition: ffmpeg_filter.c:191
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:64
formats.h
BiquadContext
Definition: af_acrossover.c:41
AF
#define AF
Definition: af_acrossover.c:67
acrossover_options
static const AVOption acrossover_options[]
Definition: af_acrossover.c:69
BiquadContext::i1
double i1
Definition: af_acrossover.c:44
fail
#define fail()
Definition: checkasm.h:120
frames
if it could not because there are no more frames
Definition: filter_design.txt:266
set_hp
static void set_hp(BiquadContext *b, float fc, float q, float sr)
Definition: af_acrossover.c:148
AudioCrossoverContext
Definition: af_acrossover.c:53
src
#define src
Definition: vp8dsp.c:254
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
av_cold
#define av_cold
Definition: attributes.h:84
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:568
s
#define s(width, name)
Definition: cbs_vp9.c:257
AudioCrossoverContext::xover
CrossoverChannel * xover
Definition: af_acrossover.c:63
AVMEDIA_TYPE_AUDIO
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
av_strtok
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
BiquadContext::i2
double i2
Definition: af_acrossover.c:44
outputs
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
ctx
AVFormatContext * ctx
Definition: movenc.c:48
f
#define f(width, name)
Definition: cbs_vp9.c:255
arg
const char * arg
Definition: jacosubdec.c:66
BiquadContext::o1
double o1
Definition: af_acrossover.c:45
av_sscanf
int av_sscanf(const char *string, const char *format,...)
See libc sscanf manual for more information.
Definition: avsscanf.c:962
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
NULL
#define NULL
Definition: coverity.c:32
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_acrossover.c:253
AudioCrossoverContext::nb_splits
int nb_splits
Definition: af_acrossover.c:60
CrossoverChannel::hp
BiquadContext hp[MAX_BANDS][4]
Definition: af_acrossover.c:50
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
AVFILTER_FLAG_DYNAMIC_OUTPUTS
#define AVFILTER_FLAG_DYNAMIC_OUTPUTS
The number of the filter outputs is not determined just by AVFilter.outputs.
Definition: avfilter.h:111
eval.h
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_acrossover.c:211
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:188
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_acrossover.c:311
BiquadContext::b1
double b1
Definition: af_acrossover.c:43
AV_SAMPLE_FMT_NONE
@ AV_SAMPLE_FMT_NONE
Definition: samplefmt.h:59
BiquadContext::a0
double a0
Definition: af_acrossover.c:42
AudioCrossoverContext::splits
float * splits
Definition: af_acrossover.c:61
attributes.h
AudioCrossoverContext::order
int order
Definition: af_acrossover.c:57
M_PI
#define M_PI
Definition: mathematics.h:52
ff_af_acrossover
AVFilter ff_af_acrossover
Definition: af_acrossover.c:332
biquad_process
static double biquad_process(BiquadContext *b, double in)
Definition: af_acrossover.c:241
internal.h
set_lp
static void set_lp(BiquadContext *b, float fc, float q, float sr)
Definition: af_acrossover.c:134
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
AudioCrossoverContext::filter_count
int filter_count
Definition: af_acrossover.c:59
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
internal.h
AVSampleFormat
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
BiquadContext::a1
double a1
Definition: af_acrossover.c:42
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
inputs
static const AVFilterPad inputs[]
Definition: af_acrossover.c:322
AVFilter
Filter definition.
Definition: avfilter.h:144
ret
ret
Definition: filter_design.txt:187
AVFilterPad::type
enum AVMediaType type
AVFilterPad type.
Definition: internal.h:65
M_SQRT1_2
#define M_SQRT1_2
Definition: mathematics.h:58
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
init
static av_cold int init(AVFilterContext *ctx)
Definition: af_acrossover.c:80
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
channel_layout.h
BiquadContext::b2
double b2
Definition: af_acrossover.c:43
ff_insert_outpad
static int ff_insert_outpad(AVFilterContext *f, unsigned index, AVFilterPad *p)
Insert a new output pad for the filter.
Definition: internal.h:285
AV_OPT_TYPE_INT
@ AV_OPT_TYPE_INT
Definition: opt.h:223
avfilter.h
MAX_BANDS
#define MAX_BANDS
Definition: af_acrossover.c:39
AV_SAMPLE_FMT_DBLP
@ AV_SAMPLE_FMT_DBLP
double, planar
Definition: samplefmt.h:70
AVFilterContext
An instance of a filter.
Definition: avfilter.h:338
audio.h
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_acrossover.c:163
alpha
static const int16_t alpha[]
Definition: ilbcdata.h:55
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
BiquadContext::o2
double o2
Definition: af_acrossover.c:45
flags
#define flags(name, subs,...)
Definition: cbs_av1.c:565
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
avstring.h
AV_OPT_TYPE_STRING
@ AV_OPT_TYPE_STRING
Definition: opt.h:227
AV_OPT_TYPE_CONST
@ AV_OPT_TYPE_CONST
Definition: opt.h:232
AudioCrossoverContext::splits_str
char * splits_str
Definition: af_acrossover.c:56