FFmpeg  4.0
af_surround.c
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1 /*
2  * Copyright (c) 2017 Paul B Mahol
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 
21 #include "libavutil/audio_fifo.h"
22 #include "libavutil/channel_layout.h"
23 #include "libavutil/opt.h"
24 #include "libavcodec/avfft.h"
25 #include "avfilter.h"
26 #include "audio.h"
27 #include "formats.h"
28 
29 typedef struct AudioSurroundContext {
30  const AVClass *class;
31 
32  char *out_channel_layout_str;
33  char *in_channel_layout_str;
34 
35  float level_in;
36  float level_out;
37  float fc_in;
38  float fc_out;
39  float lfe_in;
40  float lfe_out;
41 
42  float *input_levels;
43  float *output_levels;
44  int output_lfe;
45  int lowcutf;
46  int highcutf;
47 
48  float lowcut;
49  float highcut;
50 
51  uint64_t out_channel_layout;
52  uint64_t in_channel_layout;
53  int nb_in_channels;
54  int nb_out_channels;
55 
56  AVFrame *input;
57  AVFrame *output;
58  AVFrame *overlap_buffer;
59 
60  int buf_size;
61  int hop_size;
62  AVAudioFifo *fifo;
63  RDFTContext **rdft, **irdft;
64  float *window_func_lut;
65 
66  int64_t pts;
67 
68  void (*filter)(AVFilterContext *ctx);
69  void (*upmix_stereo)(AVFilterContext *ctx,
70  float l_phase,
71  float r_phase,
72  float c_phase,
73  float mag_total,
74  float x, float y,
75  int n);
76  void (*upmix_2_1)(AVFilterContext *ctx,
77  float l_phase,
78  float r_phase,
79  float c_phase,
80  float mag_total,
81  float lfe_im,
82  float lfe_re,
83  float x, float y,
84  int n);
85  void (*upmix_3_0)(AVFilterContext *ctx,
86  float l_phase,
87  float r_phase,
88  float c_mag,
89  float c_phase,
90  float mag_total,
91  float x, float y,
92  int n);
93  void (*upmix_5_0)(AVFilterContext *ctx,
94  float c_re, float c_im,
95  float mag_totall, float mag_totalr,
96  float fl_phase, float fr_phase,
97  float bl_phase, float br_phase,
98  float sl_phase, float sr_phase,
99  float xl, float yl,
100  float xr, float yr,
101  int n);
102  void (*upmix_5_1)(AVFilterContext *ctx,
103  float c_re, float c_im,
104  float lfe_re, float lfe_im,
105  float mag_totall, float mag_totalr,
106  float fl_phase, float fr_phase,
107  float bl_phase, float br_phase,
108  float sl_phase, float sr_phase,
109  float xl, float yl,
110  float xr, float yr,
111  int n);
112 } AudioSurroundContext;
113 
114 static int query_formats(AVFilterContext *ctx)
115 {
116  AudioSurroundContext *s = ctx->priv;
117  AVFilterFormats *formats = NULL;
118  AVFilterChannelLayouts *layouts = NULL;
119  int ret;
120 
121  ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP);
122  if (ret)
123  return ret;
124  ret = ff_set_common_formats(ctx, formats);
125  if (ret)
126  return ret;
127 
128  layouts = NULL;
129  ret = ff_add_channel_layout(&layouts, s->out_channel_layout);
130  if (ret)
131  return ret;
132 
133  ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);
134  if (ret)
135  return ret;
136 
137  layouts = NULL;
138  ret = ff_add_channel_layout(&layouts, s->in_channel_layout);
139  if (ret)
140  return ret;
141 
142  ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts);
143  if (ret)
144  return ret;
145 
146  formats = ff_all_samplerates();
147  if (!formats)
148  return AVERROR(ENOMEM);
149  return ff_set_common_samplerates(ctx, formats);
150 }
151 
152 static int config_input(AVFilterLink *inlink)
153 {
154  AVFilterContext *ctx = inlink->dst;
155  AudioSurroundContext *s = ctx->priv;
156  int ch;
157 
158  s->rdft = av_calloc(inlink->channels, sizeof(*s->rdft));
159  if (!s->rdft)
160  return AVERROR(ENOMEM);
161 
162  for (ch = 0; ch < inlink->channels; ch++) {
163  s->rdft[ch] = av_rdft_init(ff_log2(s->buf_size), DFT_R2C);
164  if (!s->rdft[ch])
165  return AVERROR(ENOMEM);
166  }
167  s->nb_in_channels = inlink->channels;
168  s->input_levels = av_malloc_array(s->nb_in_channels, sizeof(*s->input_levels));
169  if (!s->input_levels)
170  return AVERROR(ENOMEM);
171  for (ch = 0; ch < s->nb_in_channels; ch++)
172  s->input_levels[ch] = s->level_in;
173  ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_FRONT_CENTER);
174  if (ch >= 0)
175  s->input_levels[ch] *= s->fc_in;
176  ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_LOW_FREQUENCY);
177  if (ch >= 0)
178  s->input_levels[ch] *= s->lfe_in;
179 
180  s->input = ff_get_audio_buffer(inlink, s->buf_size * 2);
181  if (!s->input)
182  return AVERROR(ENOMEM);
183 
184  s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->buf_size);
185  if (!s->fifo)
186  return AVERROR(ENOMEM);
187 
188  s->lowcut = 1.f * s->lowcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2);
189  s->highcut = 1.f * s->highcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2);
190 
191  return 0;
192 }
193 
194 static int config_output(AVFilterLink *outlink)
195 {
196  AVFilterContext *ctx = outlink->src;
197  AudioSurroundContext *s = ctx->priv;
198  int ch;
199 
200  s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
201  if (!s->irdft)
202  return AVERROR(ENOMEM);
203 
204  for (ch = 0; ch < outlink->channels; ch++) {
205  s->irdft[ch] = av_rdft_init(ff_log2(s->buf_size), IDFT_C2R);
206  if (!s->irdft[ch])
207  return AVERROR(ENOMEM);
208  }
209  s->nb_out_channels = outlink->channels;
210  s->output_levels = av_malloc_array(s->nb_out_channels, sizeof(*s->output_levels));
211  if (!s->output_levels)
212  return AVERROR(ENOMEM);
213  for (ch = 0; ch < s->nb_out_channels; ch++)
214  s->output_levels[ch] = s->level_out;
215  ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_FRONT_CENTER);
216  if (ch >= 0)
217  s->output_levels[ch] *= s->fc_out;
218  ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_LOW_FREQUENCY);
219  if (ch >= 0)
220  s->output_levels[ch] *= s->lfe_out;
221 
222  s->output = ff_get_audio_buffer(outlink, s->buf_size * 2);
223  s->overlap_buffer = ff_get_audio_buffer(outlink, s->buf_size * 2);
224  if (!s->overlap_buffer || !s->output)
225  return AVERROR(ENOMEM);
226 
227  return 0;
228 }
229 
230 static void stereo_position(float a, float p, float *x, float *y)
231 {
232  *x = av_clipf(a+FFMAX(0, sinf(p-M_PI_2))*FFDIFFSIGN(a,0), -1, 1);
233  *y = av_clipf(cosf(a*M_PI_2+M_PI)*cosf(M_PI_2-p/M_PI)*M_LN10+1, -1, 1);
234 }
235 
236 static inline void get_lfe(int output_lfe, int n, float lowcut, float highcut,
237  float *lfe_mag, float *mag_total)
238 {
239  if (output_lfe && n < highcut) {
240  *lfe_mag = n < lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(lowcut-n)/(lowcut-highcut)));
241  *lfe_mag *= *mag_total;
242  *mag_total -= *lfe_mag;
243  } else {
244  *lfe_mag = 0.f;
245  }
246 }
247 
248 static void upmix_1_0(AVFilterContext *ctx,
249  float l_phase,
250  float r_phase,
251  float c_phase,
252  float mag_total,
253  float x, float y,
254  int n)
255 {
256  AudioSurroundContext *s = ctx->priv;
257  float mag, *dst;
258 
259  dst = (float *)s->output->extended_data[0];
260 
261  mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
262 
263  dst[2 * n ] = mag * cosf(c_phase);
264  dst[2 * n + 1] = mag * sinf(c_phase);
265 }
266 
267 static void upmix_stereo(AVFilterContext *ctx,
268  float l_phase,
269  float r_phase,
270  float c_phase,
271  float mag_total,
272  float x, float y,
273  int n)
274 {
275  AudioSurroundContext *s = ctx->priv;
276  float l_mag, r_mag, *dstl, *dstr;
277 
278  dstl = (float *)s->output->extended_data[0];
279  dstr = (float *)s->output->extended_data[1];
280 
281  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
282  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
283 
284  dstl[2 * n ] = l_mag * cosf(l_phase);
285  dstl[2 * n + 1] = l_mag * sinf(l_phase);
286 
287  dstr[2 * n ] = r_mag * cosf(r_phase);
288  dstr[2 * n + 1] = r_mag * sinf(r_phase);
289 }
290 
291 static void upmix_2_1(AVFilterContext *ctx,
292  float l_phase,
293  float r_phase,
294  float c_phase,
295  float mag_total,
296  float x, float y,
297  int n)
298 {
299  AudioSurroundContext *s = ctx->priv;
300  float lfe_mag, l_mag, r_mag, *dstl, *dstr, *dstlfe;
301 
302  dstl = (float *)s->output->extended_data[0];
303  dstr = (float *)s->output->extended_data[1];
304  dstlfe = (float *)s->output->extended_data[2];
305 
306  get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
307 
308  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
309  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
310 
311  dstl[2 * n ] = l_mag * cosf(l_phase);
312  dstl[2 * n + 1] = l_mag * sinf(l_phase);
313 
314  dstr[2 * n ] = r_mag * cosf(r_phase);
315  dstr[2 * n + 1] = r_mag * sinf(r_phase);
316 
317  dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
318  dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
319 }
320 
321 static void upmix_3_0(AVFilterContext *ctx,
322  float l_phase,
323  float r_phase,
324  float c_phase,
325  float mag_total,
326  float x, float y,
327  int n)
328 {
329  AudioSurroundContext *s = ctx->priv;
330  float l_mag, r_mag, c_mag, *dstc, *dstl, *dstr;
331 
332  dstl = (float *)s->output->extended_data[0];
333  dstr = (float *)s->output->extended_data[1];
334  dstc = (float *)s->output->extended_data[2];
335 
336  c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
337  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
338  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
339 
340  dstl[2 * n ] = l_mag * cosf(l_phase);
341  dstl[2 * n + 1] = l_mag * sinf(l_phase);
342 
343  dstr[2 * n ] = r_mag * cosf(r_phase);
344  dstr[2 * n + 1] = r_mag * sinf(r_phase);
345 
346  dstc[2 * n ] = c_mag * cosf(c_phase);
347  dstc[2 * n + 1] = c_mag * sinf(c_phase);
348 }
349 
350 static void upmix_3_1(AVFilterContext *ctx,
351  float l_phase,
352  float r_phase,
353  float c_phase,
354  float mag_total,
355  float x, float y,
356  int n)
357 {
358  AudioSurroundContext *s = ctx->priv;
359  float lfe_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstlfe;
360 
361  dstl = (float *)s->output->extended_data[0];
362  dstr = (float *)s->output->extended_data[1];
363  dstc = (float *)s->output->extended_data[2];
364  dstlfe = (float *)s->output->extended_data[3];
365 
366  get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
367 
368  c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
369  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
370  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
371 
372  dstl[2 * n ] = l_mag * cosf(l_phase);
373  dstl[2 * n + 1] = l_mag * sinf(l_phase);
374 
375  dstr[2 * n ] = r_mag * cosf(r_phase);
376  dstr[2 * n + 1] = r_mag * sinf(r_phase);
377 
378  dstc[2 * n ] = c_mag * cosf(c_phase);
379  dstc[2 * n + 1] = c_mag * sinf(c_phase);
380 
381  dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
382  dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
383 }
384 
385 static void upmix_3_1_surround(AVFilterContext *ctx,
386  float l_phase,
387  float r_phase,
388  float c_phase,
389  float c_mag,
390  float mag_total,
391  float x, float y,
392  int n)
393 {
394  AudioSurroundContext *s = ctx->priv;
395  float lfe_mag, l_mag, r_mag, *dstc, *dstl, *dstr, *dstlfe;
396 
397  dstl = (float *)s->output->extended_data[0];
398  dstr = (float *)s->output->extended_data[1];
399  dstc = (float *)s->output->extended_data[2];
400  dstlfe = (float *)s->output->extended_data[3];
401 
402  get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &c_mag);
403 
404  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
405  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
406 
407  dstl[2 * n ] = l_mag * cosf(l_phase);
408  dstl[2 * n + 1] = l_mag * sinf(l_phase);
409 
410  dstr[2 * n ] = r_mag * cosf(r_phase);
411  dstr[2 * n + 1] = r_mag * sinf(r_phase);
412 
413  dstc[2 * n ] = c_mag * cosf(c_phase);
414  dstc[2 * n + 1] = c_mag * sinf(c_phase);
415 
416  dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
417  dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
418 }
419 
420 static void upmix_4_0(AVFilterContext *ctx,
421  float l_phase,
422  float r_phase,
423  float c_phase,
424  float mag_total,
425  float x, float y,
426  int n)
427 {
428  float b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb;
429  AudioSurroundContext *s = ctx->priv;
430 
431  dstl = (float *)s->output->extended_data[0];
432  dstr = (float *)s->output->extended_data[1];
433  dstc = (float *)s->output->extended_data[2];
434  dstb = (float *)s->output->extended_data[3];
435 
436  c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
437  b_mag = sqrtf(1.f - fabsf(x)) * ((1.f - y) * .5f) * mag_total;
438  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
439  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
440 
441  dstl[2 * n ] = l_mag * cosf(l_phase);
442  dstl[2 * n + 1] = l_mag * sinf(l_phase);
443 
444  dstr[2 * n ] = r_mag * cosf(r_phase);
445  dstr[2 * n + 1] = r_mag * sinf(r_phase);
446 
447  dstc[2 * n ] = c_mag * cosf(c_phase);
448  dstc[2 * n + 1] = c_mag * sinf(c_phase);
449 
450  dstb[2 * n ] = b_mag * cosf(c_phase);
451  dstb[2 * n + 1] = b_mag * sinf(c_phase);
452 }
453 
454 static void upmix_4_1(AVFilterContext *ctx,
455  float l_phase,
456  float r_phase,
457  float c_phase,
458  float mag_total,
459  float x, float y,
460  int n)
461 {
462  float lfe_mag, b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb, *dstlfe;
463  AudioSurroundContext *s = ctx->priv;
464 
465  dstl = (float *)s->output->extended_data[0];
466  dstr = (float *)s->output->extended_data[1];
467  dstc = (float *)s->output->extended_data[2];
468  dstlfe = (float *)s->output->extended_data[3];
469  dstb = (float *)s->output->extended_data[4];
470 
471  get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
472 
473  dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
474  dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
475 
476  c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
477  b_mag = sqrtf(1.f - fabsf(x)) * ((1.f - y) * .5f) * mag_total;
478  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
479  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
480 
481  dstl[2 * n ] = l_mag * cosf(l_phase);
482  dstl[2 * n + 1] = l_mag * sinf(l_phase);
483 
484  dstr[2 * n ] = r_mag * cosf(r_phase);
485  dstr[2 * n + 1] = r_mag * sinf(r_phase);
486 
487  dstc[2 * n ] = c_mag * cosf(c_phase);
488  dstc[2 * n + 1] = c_mag * sinf(c_phase);
489 
490  dstb[2 * n ] = b_mag * cosf(c_phase);
491  dstb[2 * n + 1] = b_mag * sinf(c_phase);
492 }
493 
494 static void upmix_5_0_back(AVFilterContext *ctx,
495  float l_phase,
496  float r_phase,
497  float c_phase,
498  float mag_total,
499  float x, float y,
500  int n)
501 {
502  float l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs;
503  AudioSurroundContext *s = ctx->priv;
504 
505  dstl = (float *)s->output->extended_data[0];
506  dstr = (float *)s->output->extended_data[1];
507  dstc = (float *)s->output->extended_data[2];
508  dstls = (float *)s->output->extended_data[3];
509  dstrs = (float *)s->output->extended_data[4];
510 
511  c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
512  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
513  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
514  ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
515  rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
516 
517  dstl[2 * n ] = l_mag * cosf(l_phase);
518  dstl[2 * n + 1] = l_mag * sinf(l_phase);
519 
520  dstr[2 * n ] = r_mag * cosf(r_phase);
521  dstr[2 * n + 1] = r_mag * sinf(r_phase);
522 
523  dstc[2 * n ] = c_mag * cosf(c_phase);
524  dstc[2 * n + 1] = c_mag * sinf(c_phase);
525 
526  dstls[2 * n ] = ls_mag * cosf(l_phase);
527  dstls[2 * n + 1] = ls_mag * sinf(l_phase);
528 
529  dstrs[2 * n ] = rs_mag * cosf(r_phase);
530  dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
531 }
532 
533 static void upmix_5_1_back(AVFilterContext *ctx,
534  float l_phase,
535  float r_phase,
536  float c_phase,
537  float mag_total,
538  float x, float y,
539  int n)
540 {
541  float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlfe;
542  AudioSurroundContext *s = ctx->priv;
543 
544  dstl = (float *)s->output->extended_data[0];
545  dstr = (float *)s->output->extended_data[1];
546  dstc = (float *)s->output->extended_data[2];
547  dstlfe = (float *)s->output->extended_data[3];
548  dstls = (float *)s->output->extended_data[4];
549  dstrs = (float *)s->output->extended_data[5];
550 
551  get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
552 
553  c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
554  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
555  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
556  ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
557  rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
558 
559  dstl[2 * n ] = l_mag * cosf(l_phase);
560  dstl[2 * n + 1] = l_mag * sinf(l_phase);
561 
562  dstr[2 * n ] = r_mag * cosf(r_phase);
563  dstr[2 * n + 1] = r_mag * sinf(r_phase);
564 
565  dstc[2 * n ] = c_mag * cosf(c_phase);
566  dstc[2 * n + 1] = c_mag * sinf(c_phase);
567 
568  dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
569  dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
570 
571  dstls[2 * n ] = ls_mag * cosf(l_phase);
572  dstls[2 * n + 1] = ls_mag * sinf(l_phase);
573 
574  dstrs[2 * n ] = rs_mag * cosf(r_phase);
575  dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
576 }
577 
578 static void upmix_5_1_back_surround(AVFilterContext *ctx,
579  float l_phase,
580  float r_phase,
581  float c_phase,
582  float c_mag,
583  float mag_total,
584  float x, float y,
585  int n)
586 {
587  AudioSurroundContext *s = ctx->priv;
588  float lfe_mag, l_mag, r_mag, *dstc, *dstl, *dstr, *dstlfe;
589  float ls_mag, rs_mag, *dstls, *dstrs;
590 
591  dstl = (float *)s->output->extended_data[0];
592  dstr = (float *)s->output->extended_data[1];
593  dstc = (float *)s->output->extended_data[2];
594  dstlfe = (float *)s->output->extended_data[3];
595  dstls = (float *)s->output->extended_data[4];
596  dstrs = (float *)s->output->extended_data[5];
597 
598  get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &c_mag);
599 
600  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
601  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
602  ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
603  rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
604 
605  dstl[2 * n ] = l_mag * cosf(l_phase);
606  dstl[2 * n + 1] = l_mag * sinf(l_phase);
607 
608  dstr[2 * n ] = r_mag * cosf(r_phase);
609  dstr[2 * n + 1] = r_mag * sinf(r_phase);
610 
611  dstc[2 * n ] = c_mag * cosf(c_phase);
612  dstc[2 * n + 1] = c_mag * sinf(c_phase);
613 
614  dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
615  dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
616 
617  dstls[2 * n ] = ls_mag * cosf(l_phase);
618  dstls[2 * n + 1] = ls_mag * sinf(l_phase);
619 
620  dstrs[2 * n ] = rs_mag * cosf(r_phase);
621  dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
622 }
623 
624 static void upmix_5_1_back_2_1(AVFilterContext *ctx,
625  float l_phase,
626  float r_phase,
627  float c_phase,
628  float mag_total,
629  float lfe_re,
630  float lfe_im,
631  float x, float y,
632  int n)
633 {
634  AudioSurroundContext *s = ctx->priv;
635  float c_mag, l_mag, r_mag, *dstc, *dstl, *dstr, *dstlfe;
636  float ls_mag, rs_mag, *dstls, *dstrs;
637 
638  dstl = (float *)s->output->extended_data[0];
639  dstr = (float *)s->output->extended_data[1];
640  dstc = (float *)s->output->extended_data[2];
641  dstlfe = (float *)s->output->extended_data[3];
642  dstls = (float *)s->output->extended_data[4];
643  dstrs = (float *)s->output->extended_data[5];
644 
645  c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
646  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
647  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
648  ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
649  rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
650 
651  dstl[2 * n ] = l_mag * cosf(l_phase);
652  dstl[2 * n + 1] = l_mag * sinf(l_phase);
653 
654  dstr[2 * n ] = r_mag * cosf(r_phase);
655  dstr[2 * n + 1] = r_mag * sinf(r_phase);
656 
657  dstc[2 * n ] = c_mag * cosf(c_phase);
658  dstc[2 * n + 1] = c_mag * sinf(c_phase);
659 
660  dstlfe[2 * n ] = lfe_re;
661  dstlfe[2 * n + 1] = lfe_im;
662 
663  dstls[2 * n ] = ls_mag * cosf(l_phase);
664  dstls[2 * n + 1] = ls_mag * sinf(l_phase);
665 
666  dstrs[2 * n ] = rs_mag * cosf(r_phase);
667  dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
668 }
669 
670 static void upmix_7_0(AVFilterContext *ctx,
671  float l_phase,
672  float r_phase,
673  float c_phase,
674  float mag_total,
675  float x, float y,
676  int n)
677 {
678  float l_mag, r_mag, ls_mag, rs_mag, c_mag, lb_mag, rb_mag;
679  float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb;
680  AudioSurroundContext *s = ctx->priv;
681 
682  dstl = (float *)s->output->extended_data[0];
683  dstr = (float *)s->output->extended_data[1];
684  dstc = (float *)s->output->extended_data[2];
685  dstlb = (float *)s->output->extended_data[3];
686  dstrb = (float *)s->output->extended_data[4];
687  dstls = (float *)s->output->extended_data[5];
688  dstrs = (float *)s->output->extended_data[6];
689 
690  c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
691  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
692  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
693  lb_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
694  rb_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
695  ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - fabsf(y)) * mag_total;
696  rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - fabsf(y)) * mag_total;
697 
698  dstl[2 * n ] = l_mag * cosf(l_phase);
699  dstl[2 * n + 1] = l_mag * sinf(l_phase);
700 
701  dstr[2 * n ] = r_mag * cosf(r_phase);
702  dstr[2 * n + 1] = r_mag * sinf(r_phase);
703 
704  dstc[2 * n ] = c_mag * cosf(c_phase);
705  dstc[2 * n + 1] = c_mag * sinf(c_phase);
706 
707  dstlb[2 * n ] = lb_mag * cosf(l_phase);
708  dstlb[2 * n + 1] = lb_mag * sinf(l_phase);
709 
710  dstrb[2 * n ] = rb_mag * cosf(r_phase);
711  dstrb[2 * n + 1] = rb_mag * sinf(r_phase);
712 
713  dstls[2 * n ] = ls_mag * cosf(l_phase);
714  dstls[2 * n + 1] = ls_mag * sinf(l_phase);
715 
716  dstrs[2 * n ] = rs_mag * cosf(r_phase);
717  dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
718 }
719 
720 static void upmix_7_1(AVFilterContext *ctx,
721  float l_phase,
722  float r_phase,
723  float c_phase,
724  float mag_total,
725  float x, float y,
726  int n)
727 {
728  float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_mag, lb_mag, rb_mag;
729  float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe;
730  AudioSurroundContext *s = ctx->priv;
731 
732  dstl = (float *)s->output->extended_data[0];
733  dstr = (float *)s->output->extended_data[1];
734  dstc = (float *)s->output->extended_data[2];
735  dstlfe = (float *)s->output->extended_data[3];
736  dstlb = (float *)s->output->extended_data[4];
737  dstrb = (float *)s->output->extended_data[5];
738  dstls = (float *)s->output->extended_data[6];
739  dstrs = (float *)s->output->extended_data[7];
740 
741  get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
742 
743  c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
744  l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
745  r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
746  lb_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
747  rb_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
748  ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - fabsf(y)) * mag_total;
749  rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - fabsf(y)) * mag_total;
750 
751  dstl[2 * n ] = l_mag * cosf(l_phase);
752  dstl[2 * n + 1] = l_mag * sinf(l_phase);
753 
754  dstr[2 * n ] = r_mag * cosf(r_phase);
755  dstr[2 * n + 1] = r_mag * sinf(r_phase);
756 
757  dstc[2 * n ] = c_mag * cosf(c_phase);
758  dstc[2 * n + 1] = c_mag * sinf(c_phase);
759 
760  dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
761  dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
762 
763  dstlb[2 * n ] = lb_mag * cosf(l_phase);
764  dstlb[2 * n + 1] = lb_mag * sinf(l_phase);
765 
766  dstrb[2 * n ] = rb_mag * cosf(r_phase);
767  dstrb[2 * n + 1] = rb_mag * sinf(r_phase);
768 
769  dstls[2 * n ] = ls_mag * cosf(l_phase);
770  dstls[2 * n + 1] = ls_mag * sinf(l_phase);
771 
772  dstrs[2 * n ] = rs_mag * cosf(r_phase);
773  dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
774 }
775 
776 static void upmix_7_1_5_0_side(AVFilterContext *ctx,
777  float c_re, float c_im,
778  float mag_totall, float mag_totalr,
779  float fl_phase, float fr_phase,
780  float bl_phase, float br_phase,
781  float sl_phase, float sr_phase,
782  float xl, float yl,
783  float xr, float yr,
784  int n)
785 {
786  float fl_mag, fr_mag, ls_mag, rs_mag, lb_mag, rb_mag;
787  float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe;
788  float lfe_mag, c_phase, mag_total = (mag_totall + mag_totalr) * 0.5;
789  AudioSurroundContext *s = ctx->priv;
790 
791  dstl = (float *)s->output->extended_data[0];
792  dstr = (float *)s->output->extended_data[1];
793  dstc = (float *)s->output->extended_data[2];
794  dstlfe = (float *)s->output->extended_data[3];
795  dstlb = (float *)s->output->extended_data[4];
796  dstrb = (float *)s->output->extended_data[5];
797  dstls = (float *)s->output->extended_data[6];
798  dstrs = (float *)s->output->extended_data[7];
799 
800  c_phase = atan2f(c_im, c_re);
801 
802  get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
803 
804  fl_mag = sqrtf(.5f * (xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
805  fr_mag = sqrtf(.5f * (xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
806  lb_mag = sqrtf(.5f * (-xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
807  rb_mag = sqrtf(.5f * (-xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
808  ls_mag = sqrtf(1.f - fabsf(xl)) * ((yl + 1.f) * .5f) * mag_totall;
809  rs_mag = sqrtf(1.f - fabsf(xr)) * ((yr + 1.f) * .5f) * mag_totalr;
810 
811  dstl[2 * n ] = fl_mag * cosf(fl_phase);
812  dstl[2 * n + 1] = fl_mag * sinf(fl_phase);
813 
814  dstr[2 * n ] = fr_mag * cosf(fr_phase);
815  dstr[2 * n + 1] = fr_mag * sinf(fr_phase);
816 
817  dstc[2 * n ] = c_re;
818  dstc[2 * n + 1] = c_im;
819 
820  dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
821  dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
822 
823  dstlb[2 * n ] = lb_mag * cosf(bl_phase);
824  dstlb[2 * n + 1] = lb_mag * sinf(bl_phase);
825 
826  dstrb[2 * n ] = rb_mag * cosf(br_phase);
827  dstrb[2 * n + 1] = rb_mag * sinf(br_phase);
828 
829  dstls[2 * n ] = ls_mag * cosf(sl_phase);
830  dstls[2 * n + 1] = ls_mag * sinf(sl_phase);
831 
832  dstrs[2 * n ] = rs_mag * cosf(sr_phase);
833  dstrs[2 * n + 1] = rs_mag * sinf(sr_phase);
834 }
835 
836 static void upmix_7_1_5_1(AVFilterContext *ctx,
837  float c_re, float c_im,
838  float lfe_re, float lfe_im,
839  float mag_totall, float mag_totalr,
840  float fl_phase, float fr_phase,
841  float bl_phase, float br_phase,
842  float sl_phase, float sr_phase,
843  float xl, float yl,
844  float xr, float yr,
845  int n)
846 {
847  float fl_mag, fr_mag, ls_mag, rs_mag, lb_mag, rb_mag;
848  float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe;
849  AudioSurroundContext *s = ctx->priv;
850 
851  dstl = (float *)s->output->extended_data[0];
852  dstr = (float *)s->output->extended_data[1];
853  dstc = (float *)s->output->extended_data[2];
854  dstlfe = (float *)s->output->extended_data[3];
855  dstlb = (float *)s->output->extended_data[4];
856  dstrb = (float *)s->output->extended_data[5];
857  dstls = (float *)s->output->extended_data[6];
858  dstrs = (float *)s->output->extended_data[7];
859 
860  fl_mag = sqrtf(.5f * (xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
861  fr_mag = sqrtf(.5f * (xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
862  lb_mag = sqrtf(.5f * (-xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
863  rb_mag = sqrtf(.5f * (-xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
864  ls_mag = sqrtf(1.f - fabsf(xl)) * ((yl + 1.f) * .5f) * mag_totall;
865  rs_mag = sqrtf(1.f - fabsf(xr)) * ((yr + 1.f) * .5f) * mag_totalr;
866 
867  dstl[2 * n ] = fl_mag * cosf(fl_phase);
868  dstl[2 * n + 1] = fl_mag * sinf(fl_phase);
869 
870  dstr[2 * n ] = fr_mag * cosf(fr_phase);
871  dstr[2 * n + 1] = fr_mag * sinf(fr_phase);
872 
873  dstc[2 * n ] = c_re;
874  dstc[2 * n + 1] = c_im;
875 
876  dstlfe[2 * n ] = lfe_re;
877  dstlfe[2 * n + 1] = lfe_im;
878 
879  dstlb[2 * n ] = lb_mag * cosf(bl_phase);
880  dstlb[2 * n + 1] = lb_mag * sinf(bl_phase);
881 
882  dstrb[2 * n ] = rb_mag * cosf(br_phase);
883  dstrb[2 * n + 1] = rb_mag * sinf(br_phase);
884 
885  dstls[2 * n ] = ls_mag * cosf(sl_phase);
886  dstls[2 * n + 1] = ls_mag * sinf(sl_phase);
887 
888  dstrs[2 * n ] = rs_mag * cosf(sr_phase);
889  dstrs[2 * n + 1] = rs_mag * sinf(sr_phase);
890 }
891 
892 static void filter_stereo(AVFilterContext *ctx)
893 {
894  AudioSurroundContext *s = ctx->priv;
895  float *srcl, *srcr;
896  int n;
897 
898  srcl = (float *)s->input->extended_data[0];
899  srcr = (float *)s->input->extended_data[1];
900 
901  for (n = 0; n < s->buf_size; n++) {
902  float l_re = srcl[2 * n], r_re = srcr[2 * n];
903  float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
904  float c_phase = atan2f(l_im + r_im, l_re + r_re);
905  float l_mag = hypotf(l_re, l_im);
906  float r_mag = hypotf(r_re, r_im);
907  float l_phase = atan2f(l_im, l_re);
908  float r_phase = atan2f(r_im, r_re);
909  float phase_dif = fabsf(l_phase - r_phase);
910  float mag_dif = (l_mag - r_mag) / (l_mag + r_mag);
911  float mag_total = hypotf(l_mag, r_mag);
912  float x, y;
913 
914  if (phase_dif > M_PI)
915  phase_dif = 2 * M_PI - phase_dif;
916 
917  stereo_position(mag_dif, phase_dif, &x, &y);
918 
919  s->upmix_stereo(ctx, l_phase, r_phase, c_phase, mag_total, x, y, n);
920  }
921 }
922 
923 static void filter_surround(AVFilterContext *ctx)
924 {
925  AudioSurroundContext *s = ctx->priv;
926  float *srcl, *srcr, *srcc;
927  int n;
928 
929  srcl = (float *)s->input->extended_data[0];
930  srcr = (float *)s->input->extended_data[1];
931  srcc = (float *)s->input->extended_data[2];
932 
933  for (n = 0; n < s->buf_size; n++) {
934  float l_re = srcl[2 * n], r_re = srcr[2 * n];
935  float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
936  float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
937  float c_mag = hypotf(c_re, c_im);
938  float c_phase = atan2f(c_im, c_re);
939  float l_mag = hypotf(l_re, l_im);
940  float r_mag = hypotf(r_re, r_im);
941  float l_phase = atan2f(l_im, l_re);
942  float r_phase = atan2f(r_im, r_re);
943  float phase_dif = fabsf(l_phase - r_phase);
944  float mag_dif = (l_mag - r_mag) / (l_mag + r_mag);
945  float mag_total = hypotf(l_mag, r_mag);
946  float x, y;
947 
948  if (phase_dif > M_PI)
949  phase_dif = 2 * M_PI - phase_dif;
950 
951  stereo_position(mag_dif, phase_dif, &x, &y);
952 
953  s->upmix_3_0(ctx, l_phase, r_phase, c_phase, c_mag, mag_total, x, y, n);
954  }
955 }
956 
957 static void filter_2_1(AVFilterContext *ctx)
958 {
959  AudioSurroundContext *s = ctx->priv;
960  float *srcl, *srcr, *srclfe;
961  int n;
962 
963  srcl = (float *)s->input->extended_data[0];
964  srcr = (float *)s->input->extended_data[1];
965  srclfe = (float *)s->input->extended_data[2];
966 
967  for (n = 0; n < s->buf_size; n++) {
968  float l_re = srcl[2 * n], r_re = srcr[2 * n];
969  float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
970  float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
971  float c_phase = atan2f(l_im + r_im, l_re + r_re);
972  float l_mag = hypotf(l_re, l_im);
973  float r_mag = hypotf(r_re, r_im);
974  float l_phase = atan2f(l_im, l_re);
975  float r_phase = atan2f(r_im, r_re);
976  float phase_dif = fabsf(l_phase - r_phase);
977  float mag_dif = (l_mag - r_mag) / (l_mag + r_mag);
978  float mag_total = hypotf(l_mag, r_mag);
979  float x, y;
980 
981  if (phase_dif > M_PI)
982  phase_dif = 2 * M_PI - phase_dif;
983 
984  stereo_position(mag_dif, phase_dif, &x, &y);
985 
986  s->upmix_2_1(ctx, l_phase, r_phase, c_phase, mag_total, lfe_re, lfe_im, x, y, n);
987  }
988 }
989 
990 static void filter_5_0_side(AVFilterContext *ctx)
991 {
992  AudioSurroundContext *s = ctx->priv;
993  float *srcl, *srcr, *srcc, *srcsl, *srcsr;
994  int n;
995 
996  srcl = (float *)s->input->extended_data[0];
997  srcr = (float *)s->input->extended_data[1];
998  srcc = (float *)s->input->extended_data[2];
999  srcsl = (float *)s->input->extended_data[3];
1000  srcsr = (float *)s->input->extended_data[4];
1001 
1002  for (n = 0; n < s->buf_size; n++) {
1003  float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
1004  float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
1005  float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
1006  float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1];
1007  float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1];
1008  float fl_mag = hypotf(fl_re, fl_im);
1009  float fr_mag = hypotf(fr_re, fr_im);
1010  float fl_phase = atan2f(fl_im, fl_re);
1011  float fr_phase = atan2f(fr_im, fr_re);
1012  float sl_mag = hypotf(sl_re, sl_im);
1013  float sr_mag = hypotf(sr_re, sr_im);
1014  float sl_phase = atan2f(sl_im, sl_re);
1015  float sr_phase = atan2f(sr_im, sr_re);
1016  float phase_difl = fabsf(fl_phase - sl_phase);
1017  float phase_difr = fabsf(fr_phase - sr_phase);
1018  float mag_difl = (fl_mag - sl_mag) / (fl_mag + sl_mag);
1019  float mag_difr = (fr_mag - sr_mag) / (fr_mag + sr_mag);
1020  float mag_totall = hypotf(fl_mag, sl_mag);
1021  float mag_totalr = hypotf(fr_mag, sr_mag);
1022  float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re);
1023  float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re);
1024  float xl, yl;
1025  float xr, yr;
1026 
1027  if (phase_difl > M_PI)
1028  phase_difl = 2 * M_PI - phase_difl;
1029 
1030  if (phase_difr > M_PI)
1031  phase_difr = 2 * M_PI - phase_difr;
1032 
1033  stereo_position(mag_difl, phase_difl, &xl, &yl);
1034  stereo_position(mag_difr, phase_difr, &xr, &yr);
1035 
1036  s->upmix_5_0(ctx, c_re, c_im,
1037  mag_totall, mag_totalr,
1038  fl_phase, fr_phase,
1039  bl_phase, br_phase,
1040  sl_phase, sr_phase,
1041  xl, yl, xr, yr, n);
1042  }
1043 }
1044 
1045 static void filter_5_1_side(AVFilterContext *ctx)
1046 {
1047  AudioSurroundContext *s = ctx->priv;
1048  float *srcl, *srcr, *srcc, *srclfe, *srcsl, *srcsr;
1049  int n;
1050 
1051  srcl = (float *)s->input->extended_data[0];
1052  srcr = (float *)s->input->extended_data[1];
1053  srcc = (float *)s->input->extended_data[2];
1054  srclfe = (float *)s->input->extended_data[3];
1055  srcsl = (float *)s->input->extended_data[4];
1056  srcsr = (float *)s->input->extended_data[5];
1057 
1058  for (n = 0; n < s->buf_size; n++) {
1059  float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
1060  float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
1061  float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
1062  float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
1063  float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1];
1064  float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1];
1065  float fl_mag = hypotf(fl_re, fl_im);
1066  float fr_mag = hypotf(fr_re, fr_im);
1067  float fl_phase = atan2f(fl_im, fl_re);
1068  float fr_phase = atan2f(fr_im, fr_re);
1069  float sl_mag = hypotf(sl_re, sl_im);
1070  float sr_mag = hypotf(sr_re, sr_im);
1071  float sl_phase = atan2f(sl_im, sl_re);
1072  float sr_phase = atan2f(sr_im, sr_re);
1073  float phase_difl = fabsf(fl_phase - sl_phase);
1074  float phase_difr = fabsf(fr_phase - sr_phase);
1075  float mag_difl = (fl_mag - sl_mag) / (fl_mag + sl_mag);
1076  float mag_difr = (fr_mag - sr_mag) / (fr_mag + sr_mag);
1077  float mag_totall = hypotf(fl_mag, sl_mag);
1078  float mag_totalr = hypotf(fr_mag, sr_mag);
1079  float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re);
1080  float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re);
1081  float xl, yl;
1082  float xr, yr;
1083 
1084  if (phase_difl > M_PI)
1085  phase_difl = 2 * M_PI - phase_difl;
1086 
1087  if (phase_difr > M_PI)
1088  phase_difr = 2 * M_PI - phase_difr;
1089 
1090  stereo_position(mag_difl, phase_difl, &xl, &yl);
1091  stereo_position(mag_difr, phase_difr, &xr, &yr);
1092 
1093  s->upmix_5_1(ctx, c_re, c_im, lfe_re, lfe_im,
1094  mag_totall, mag_totalr,
1095  fl_phase, fr_phase,
1096  bl_phase, br_phase,
1097  sl_phase, sr_phase,
1098  xl, yl, xr, yr, n);
1099  }
1100 }
1101 
1102 static void filter_5_1_back(AVFilterContext *ctx)
1103 {
1104  AudioSurroundContext *s = ctx->priv;
1105  float *srcl, *srcr, *srcc, *srclfe, *srcbl, *srcbr;
1106  int n;
1107 
1108  srcl = (float *)s->input->extended_data[0];
1109  srcr = (float *)s->input->extended_data[1];
1110  srcc = (float *)s->input->extended_data[2];
1111  srclfe = (float *)s->input->extended_data[3];
1112  srcbl = (float *)s->input->extended_data[4];
1113  srcbr = (float *)s->input->extended_data[5];
1114 
1115  for (n = 0; n < s->buf_size; n++) {
1116  float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
1117  float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
1118  float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
1119  float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
1120  float bl_re = srcbl[2 * n], bl_im = srcbl[2 * n + 1];
1121  float br_re = srcbr[2 * n], br_im = srcbr[2 * n + 1];
1122  float fl_mag = hypotf(fl_re, fl_im);
1123  float fr_mag = hypotf(fr_re, fr_im);
1124  float fl_phase = atan2f(fl_im, fl_re);
1125  float fr_phase = atan2f(fr_im, fr_re);
1126  float bl_mag = hypotf(bl_re, bl_im);
1127  float br_mag = hypotf(br_re, br_im);
1128  float bl_phase = atan2f(bl_im, bl_re);
1129  float br_phase = atan2f(br_im, br_re);
1130  float phase_difl = fabsf(fl_phase - bl_phase);
1131  float phase_difr = fabsf(fr_phase - br_phase);
1132  float mag_difl = (fl_mag - bl_mag) / (fl_mag + bl_mag);
1133  float mag_difr = (fr_mag - br_mag) / (fr_mag + br_mag);
1134  float mag_totall = hypotf(fl_mag, bl_mag);
1135  float mag_totalr = hypotf(fr_mag, br_mag);
1136  float sl_phase = atan2f(fl_im + bl_im, fl_re + bl_re);
1137  float sr_phase = atan2f(fr_im + br_im, fr_re + br_re);
1138  float xl, yl;
1139  float xr, yr;
1140 
1141  if (phase_difl > M_PI)
1142  phase_difl = 2 * M_PI - phase_difl;
1143 
1144  if (phase_difr > M_PI)
1145  phase_difr = 2 * M_PI - phase_difr;
1146 
1147  stereo_position(mag_difl, phase_difl, &xl, &yl);
1148  stereo_position(mag_difr, phase_difr, &xr, &yr);
1149 
1150  s->upmix_5_1(ctx, c_re, c_im, lfe_re, lfe_im,
1151  mag_totall, mag_totalr,
1152  fl_phase, fr_phase,
1153  bl_phase, br_phase,
1154  sl_phase, sr_phase,
1155  xl, yl, xr, yr, n);
1156  }
1157 }
1158 
1159 static int init(AVFilterContext *ctx)
1160 {
1161  AudioSurroundContext *s = ctx->priv;
1162  float overlap;
1163  int i;
1164 
1165  if (!(s->out_channel_layout = av_get_channel_layout(s->out_channel_layout_str))) {
1166  av_log(ctx, AV_LOG_ERROR, "Error parsing output channel layout '%s'.\n",
1167  s->out_channel_layout_str);
1168  return AVERROR(EINVAL);
1169  }
1170 
1171  if (!(s->in_channel_layout = av_get_channel_layout(s->in_channel_layout_str))) {
1172  av_log(ctx, AV_LOG_ERROR, "Error parsing input channel layout '%s'.\n",
1173  s->in_channel_layout_str);
1174  return AVERROR(EINVAL);
1175  }
1176 
1177  if (s->lowcutf >= s->highcutf) {
1178  av_log(ctx, AV_LOG_ERROR, "Low cut-off '%d' should be less than high cut-off '%d'.\n",
1179  s->lowcutf, s->highcutf);
1180  return AVERROR(EINVAL);
1181  }
1182 
1183  switch (s->in_channel_layout) {
1184  case AV_CH_LAYOUT_STEREO:
1185  s->filter = filter_stereo;
1186  switch (s->out_channel_layout) {
1187  case AV_CH_LAYOUT_MONO:
1188  s->upmix_stereo = upmix_1_0;
1189  break;
1190  case AV_CH_LAYOUT_STEREO:
1191  s->upmix_stereo = upmix_stereo;
1192  break;
1193  case AV_CH_LAYOUT_2POINT1:
1194  s->upmix_stereo = upmix_2_1;
1195  break;
1196  case AV_CH_LAYOUT_SURROUND:
1197  s->upmix_stereo = upmix_3_0;
1198  break;
1199  case AV_CH_LAYOUT_3POINT1:
1200  s->upmix_stereo = upmix_3_1;
1201  break;
1202  case AV_CH_LAYOUT_4POINT0:
1203  s->upmix_stereo = upmix_4_0;
1204  break;
1205  case AV_CH_LAYOUT_4POINT1:
1206  s->upmix_stereo = upmix_4_1;
1207  break;
1208  case AV_CH_LAYOUT_5POINT0_BACK:
1209  s->upmix_stereo = upmix_5_0_back;
1210  break;
1211  case AV_CH_LAYOUT_5POINT1_BACK:
1212  s->upmix_stereo = upmix_5_1_back;
1213  break;
1214  case AV_CH_LAYOUT_7POINT0:
1215  s->upmix_stereo = upmix_7_0;
1216  break;
1217  case AV_CH_LAYOUT_7POINT1:
1218  s->upmix_stereo = upmix_7_1;
1219  break;
1220  default:
1221  goto fail;
1222  }
1223  break;
1224  case AV_CH_LAYOUT_2POINT1:
1225  s->filter = filter_2_1;
1226  switch (s->out_channel_layout) {
1227  case AV_CH_LAYOUT_5POINT1_BACK:
1228  s->upmix_2_1 = upmix_5_1_back_2_1;
1229  break;
1230  default:
1231  goto fail;
1232  }
1233  break;
1234  case AV_CH_LAYOUT_SURROUND:
1235  s->filter = filter_surround;
1236  switch (s->out_channel_layout) {
1237  case AV_CH_LAYOUT_3POINT1:
1238  s->upmix_3_0 = upmix_3_1_surround;
1239  break;
1240  case AV_CH_LAYOUT_5POINT1_BACK:
1241  s->upmix_3_0 = upmix_5_1_back_surround;
1242  break;
1243  default:
1244  goto fail;
1245  }
1246  break;
1247  case AV_CH_LAYOUT_5POINT0:
1248  s->filter = filter_5_0_side;
1249  switch (s->out_channel_layout) {
1250  case AV_CH_LAYOUT_7POINT1:
1251  s->upmix_5_0 = upmix_7_1_5_0_side;
1252  break;
1253  default:
1254  goto fail;
1255  }
1256  break;
1257  case AV_CH_LAYOUT_5POINT1:
1258  s->filter = filter_5_1_side;
1259  switch (s->out_channel_layout) {
1260  case AV_CH_LAYOUT_7POINT1:
1261  s->upmix_5_1 = upmix_7_1_5_1;
1262  break;
1263  default:
1264  goto fail;
1265  }
1266  break;
1267  case AV_CH_LAYOUT_5POINT1_BACK:
1268  s->filter = filter_5_1_back;
1269  switch (s->out_channel_layout) {
1270  case AV_CH_LAYOUT_7POINT1:
1271  s->upmix_5_1 = upmix_7_1_5_1;
1272  break;
1273  default:
1274  goto fail;
1275  }
1276  break;
1277  default:
1278 fail:
1279  av_log(ctx, AV_LOG_ERROR, "Unsupported upmix: '%s' -> '%s'.\n",
1280  s->in_channel_layout_str, s->out_channel_layout_str);
1281  return AVERROR(EINVAL);
1282  }
1283 
1284  s->buf_size = 4096;
1285  s->pts = AV_NOPTS_VALUE;
1286 
1287  s->window_func_lut = av_calloc(s->buf_size, sizeof(*s->window_func_lut));
1288  if (!s->window_func_lut)
1289  return AVERROR(ENOMEM);
1290 
1291  for (i = 0; i < s->buf_size; i++)
1292  s->window_func_lut[i] = sqrtf(0.5 * (1 - cosf(2 * M_PI * i / s->buf_size)) / s->buf_size);
1293  overlap = .5;
1294  s->hop_size = s->buf_size * (1. - overlap);
1295 
1296  return 0;
1297 }
1298 
1299 static int fft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
1300 {
1301  AudioSurroundContext *s = ctx->priv;
1302  const float level_in = s->input_levels[ch];
1303  float *dst;
1304  int n;
1305 
1306  memset(s->input->extended_data[ch] + s->buf_size * sizeof(float), 0, s->buf_size * sizeof(float));
1307 
1308  dst = (float *)s->input->extended_data[ch];
1309  for (n = 0; n < s->buf_size; n++) {
1310  dst[n] *= s->window_func_lut[n] * level_in;
1311  }
1312 
1313  av_rdft_calc(s->rdft[ch], (float *)s->input->extended_data[ch]);
1314 
1315  return 0;
1316 }
1317 
1318 static int ifft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
1319 {
1320  AudioSurroundContext *s = ctx->priv;
1321  const float level_out = s->output_levels[ch];
1322  AVFrame *out = arg;
1323  float *dst, *ptr;
1324  int n;
1325 
1326  av_rdft_calc(s->irdft[ch], (float *)s->output->extended_data[ch]);
1327 
1328  dst = (float *)s->output->extended_data[ch];
1329  ptr = (float *)s->overlap_buffer->extended_data[ch];
1330 
1331  memmove(s->overlap_buffer->extended_data[ch],
1332  s->overlap_buffer->extended_data[ch] + s->hop_size * sizeof(float),
1333  s->buf_size * sizeof(float));
1334  memset(s->overlap_buffer->extended_data[ch] + s->buf_size * sizeof(float),
1335  0, s->hop_size * sizeof(float));
1336 
1337  for (n = 0; n < s->buf_size; n++) {
1338  ptr[n] += dst[n] * s->window_func_lut[n] * level_out;
1339  }
1340 
1341  ptr = (float *)s->overlap_buffer->extended_data[ch];
1342  dst = (float *)out->extended_data[ch];
1343  memcpy(dst, ptr, s->hop_size * sizeof(float));
1344 
1345  return 0;
1346 }
1347 
1348 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
1349 {
1350  AVFilterContext *ctx = inlink->dst;
1351  AVFilterLink *outlink = ctx->outputs[0];
1352  AudioSurroundContext *s = ctx->priv;
1353  int ret;
1354 
1355  ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
1356  in->nb_samples);
1357  if (ret >= 0 && s->pts == AV_NOPTS_VALUE)
1358  s->pts = in->pts;
1359 
1360  av_frame_free(&in);
1361  if (ret < 0)
1362  return ret;
1363 
1364  while (av_audio_fifo_size(s->fifo) >= s->buf_size) {
1365  AVFrame *out;
1366 
1367  ret = av_audio_fifo_peek(s->fifo, (void **)s->input->extended_data, s->buf_size);
1368  if (ret < 0)
1369  return ret;
1370 
1371  ctx->internal->execute(ctx, fft_channel, NULL, NULL, inlink->channels);
1372 
1373  s->filter(ctx);
1374 
1375  out = ff_get_audio_buffer(outlink, s->hop_size);
1376  if (!out)
1377  return AVERROR(ENOMEM);
1378 
1379  ctx->internal->execute(ctx, ifft_channel, out, NULL, outlink->channels);
1380 
1381  out->pts = s->pts;
1382  if (s->pts != AV_NOPTS_VALUE)
1383  s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
1384  av_audio_fifo_drain(s->fifo, s->hop_size);
1385  ret = ff_filter_frame(outlink, out);
1386  if (ret < 0)
1387  return ret;
1388  }
1389 
1390  return 0;
1391 }
1392 
1393 static int request_frame(AVFilterLink *outlink)
1394 {
1395  AVFilterContext *ctx = outlink->src;
1396  AudioSurroundContext *s = ctx->priv;
1397  int ret = 0;
1398 
1399  ret = ff_request_frame(ctx->inputs[0]);
1400 
1401  if (ret == AVERROR_EOF && av_audio_fifo_size(s->fifo) > 0 && av_audio_fifo_size(s->fifo) < s->buf_size) {
1402  AVFrame *in;
1403 
1404  in = ff_get_audio_buffer(outlink, s->buf_size - av_audio_fifo_size(s->fifo));
1405  if (!in)
1406  return AVERROR(ENOMEM);
1407  ret = filter_frame(ctx->inputs[0], in);
1408  av_audio_fifo_drain(s->fifo, s->buf_size);
1409  }
1410 
1411  return ret;
1412 }
1413 
1414 static av_cold void uninit(AVFilterContext *ctx)
1415 {
1416  AudioSurroundContext *s = ctx->priv;
1417  int ch;
1418 
1419  av_frame_free(&s->input);
1420  av_frame_free(&s->output);
1421  av_frame_free(&s->overlap_buffer);
1422 
1423  for (ch = 0; ch < s->nb_in_channels; ch++) {
1424  av_rdft_end(s->rdft[ch]);
1425  }
1426  for (ch = 0; ch < s->nb_out_channels; ch++) {
1427  av_rdft_end(s->irdft[ch]);
1428  }
1429  av_freep(&s->input_levels);
1430  av_freep(&s->output_levels);
1431  av_freep(&s->rdft);
1432  av_freep(&s->irdft);
1433  av_audio_fifo_free(s->fifo);
1434  av_freep(&s->window_func_lut);
1435 }
1436 
1437 #define OFFSET(x) offsetof(AudioSurroundContext, x)
1438 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
1439 
1440 static const AVOption surround_options[] = {
1441  { "chl_out", "set output channel layout", OFFSET(out_channel_layout_str), AV_OPT_TYPE_STRING, {.str="5.1"}, 0, 0, FLAGS },
1442  { "chl_in", "set input channel layout", OFFSET(in_channel_layout_str), AV_OPT_TYPE_STRING, {.str="stereo"},0, 0, FLAGS },
1443  { "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
1444  { "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
1445  { "lfe", "output LFE", OFFSET(output_lfe), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, FLAGS },
1446  { "lfe_low", "LFE low cut off", OFFSET(lowcutf), AV_OPT_TYPE_INT, {.i64=128}, 0, 256, FLAGS },
1447  { "lfe_high", "LFE high cut off", OFFSET(highcutf), AV_OPT_TYPE_INT, {.i64=256}, 0, 512, FLAGS },
1448  { "fc_in", "set front center channel input level", OFFSET(fc_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
1449  { "fc_out", "set front center channel output level", OFFSET(fc_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
1450  { "lfe_in", "set lfe channel input level", OFFSET(lfe_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
1451  { "lfe_out", "set lfe channel output level", OFFSET(lfe_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
1452  { NULL }
1453 };
1454 
1455 AVFILTER_DEFINE_CLASS(surround);
1456 
1457 static const AVFilterPad inputs[] = {
1458  {
1459  .name = "default",
1460  .type = AVMEDIA_TYPE_AUDIO,
1461  .filter_frame = filter_frame,
1462  .config_props = config_input,
1463  },
1464  { NULL }
1465 };
1466 
1467 static const AVFilterPad outputs[] = {
1468  {
1469  .name = "default",
1470  .type = AVMEDIA_TYPE_AUDIO,
1471  .request_frame = request_frame,
1472  .config_props = config_output,
1473  },
1474  { NULL }
1475 };
1476 
1477 AVFilter ff_af_surround = {
1478  .name = "surround",
1479  .description = NULL_IF_CONFIG_SMALL("Apply audio surround upmix filter."),
1480  .query_formats = query_formats,
1481  .priv_size = sizeof(AudioSurroundContext),
1482  .priv_class = &surround_class,
1483  .init = init,
1484  .uninit = uninit,
1485  .inputs = inputs,
1486  .outputs = outputs,
1487  .flags = AVFILTER_FLAG_SLICE_THREADS,
1488 };
AV_SAMPLE_FMT_FLTP
float, planar
Definition: samplefmt.h:69
NULL
#define NULL
Definition: coverity.c:32
AV_CH_LAYOUT_7POINT1
#define AV_CH_LAYOUT_7POINT1
Definition: channel_layout.h:107
AV_CH_LAYOUT_4POINT1
#define AV_CH_LAYOUT_4POINT1
Definition: channel_layout.h:92
s
const char * s
Definition: avisynth_c.h:768
av_audio_fifo_alloc
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
config_input
static int config_input(AVFilterLink *inlink)
Definition: af_surround.c:152
stereo_position
static void stereo_position(float a, float p, float *x, float *y)
Definition: af_surround.c:230
AVFrame
This structure describes decoded (raw) audio or video data.
Definition: frame.h:218
AVOption
AVOption.
Definition: opt.h:246
init
static int init(AVFilterContext *ctx)
Definition: af_surround.c:1159
upmix_4_0
static void upmix_4_0(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:420
AV_CH_LAYOUT_SURROUND
#define AV_CH_LAYOUT_SURROUND
Definition: channel_layout.h:89
avfilter.h
Main libavfilter public API header.
AV_OPT_TYPE_INT
Definition: opt.h:223
query_formats
static int query_formats(AVFilterContext *ctx)
Definition: af_surround.c:114
av_audio_fifo_free
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
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 &&HAVE_MMX) 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
OFFSET
#define OFFSET(x)
Definition: af_surround.c:1437
request_frame
static int request_frame(AVFilterLink *outlink)
Definition: af_surround.c:1393
uninit
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_surround.c:1414
AV_CH_LAYOUT_4POINT0
#define AV_CH_LAYOUT_4POINT0
Definition: channel_layout.h:91
AV_CH_LAYOUT_7POINT0
#define AV_CH_LAYOUT_7POINT0
Definition: channel_layout.h:105
AudioSurroundContext::upmix_5_0
void(* upmix_5_0)(AVFilterContext *ctx, float c_re, float c_im, float mag_totall, float mag_totalr, float fl_phase, float fr_phase, float bl_phase, float br_phase, float sl_phase, float sr_phase, float xl, float yl, float xr, float yr, int n)
Definition: af_surround.c:93
AV_CH_LAYOUT_STEREO
#define AV_CH_LAYOUT_STEREO
Definition: channel_layout.h:86
AV_CH_LAYOUT_5POINT0
#define AV_CH_LAYOUT_5POINT0
Definition: channel_layout.h:95
av_calloc
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
AVFilterPad::name
const char * name
Pad name.
Definition: internal.h:60
av_get_channel_layout
uint64_t av_get_channel_layout(const char *name)
Return a channel layout id that matches name, or 0 if no match is found.
Definition: channel_layout.c:139
AVFilterContext::inputs
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
upmix_3_1_surround
static void upmix_3_1_surround(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float c_mag, float mag_total, float x, float y, int n)
Definition: af_surround.c:385
ff_channel_layouts_ref
int ff_channel_layouts_ref(AVFilterChannelLayouts *f, AVFilterChannelLayouts **ref)
Add *ref as a new reference to f.
Definition: formats.c:435
ff_filter_frame
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
AudioSurroundContext::upmix_2_1
void(* upmix_2_1)(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float lfe_im, float lfe_re, float x, float y, int n)
Definition: af_surround.c:76
FLAGS
#define FLAGS
Definition: af_surround.c:1438
av_cold
#define av_cold
Definition: attributes.h:82
opt.h
AVOptions.
upmix_7_1_5_1
static void upmix_7_1_5_1(AVFilterContext *ctx, float c_re, float c_im, float lfe_re, float lfe_im, float mag_totall, float mag_totalr, float fl_phase, float fr_phase, float bl_phase, float br_phase, float sl_phase, float sr_phase, float xl, float yl, float xr, float yr, int n)
Definition: af_surround.c:836
AVFrame::pts
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:311
cosf
#define cosf(x)
Definition: libm.h:78
AV_CH_LOW_FREQUENCY
#define AV_CH_LOW_FREQUENCY
Definition: channel_layout.h:52
AV_OPT_TYPE_STRING
Definition: opt.h:227
filter_stereo
static void filter_stereo(AVFilterContext *ctx)
Definition: af_surround.c:892
flags
static int flags
Definition: log.c:55
atan2f
#define atan2f(y, x)
Definition: libm.h:45
AVERROR_EOF
#define AVERROR_EOF
End of file.
Definition: error.h:55
av_log
#define av_log(a,...)
Definition: tableprint_vlc.h:28
ifft_channel
static int ifft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
Definition: af_surround.c:1318
AVFilterPad
A filter pad used for either input or output.
Definition: internal.h:54
AV_CH_LAYOUT_5POINT1
#define AV_CH_LAYOUT_5POINT1
Definition: channel_layout.h:96
upmix_7_1
static void upmix_7_1(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:720
av_rescale_q
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
AV_LOG_ERROR
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
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
ff_add_channel_layout
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:343
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
filter_5_1_back
static void filter_5_1_back(AVFilterContext *ctx)
Definition: af_surround.c:1102
AVERROR
#define AVERROR(e)
Definition: error.h:43
surround_options
static const AVOption surround_options[]
Definition: af_surround.c:1440
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
AV_OPT_TYPE_FLOAT
Definition: opt.h:226
upmix_5_1_back
static void upmix_5_1_back(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:533
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:186
AVFilterContext::priv
void * priv
private data for use by the filter
Definition: avfilter.h:353
AVFILTER_FLAG_SLICE_THREADS
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:116
arg
const char * arg
Definition: jacosubdec.c:66
AudioSurroundContext::upmix_3_0
void(* upmix_3_0)(AVFilterContext *ctx, float l_phase, float r_phase, float c_mag, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:85
AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
IDFT_C2R
Definition: avfft.h:73
ff_add_format
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:337
upmix_5_1_back_2_1
static void upmix_5_1_back_2_1(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float lfe_re, float lfe_im, float x, float y, int n)
Definition: af_surround.c:624
FFMAX
#define FFMAX(a, b)
Definition: common.h:94
get_lfe
static void get_lfe(int output_lfe, int n, float lowcut, float highcut, float *lfe_mag, float *mag_total)
Definition: af_surround.c:236
fail
#define fail()
Definition: checkasm.h:116
av_rdft_calc
void av_rdft_calc(RDFTContext *s, FFTSample *data)
AVAudioFifo
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
FFDIFFSIGN
#define FFDIFFSIGN(x, y)
Comparator.
Definition: common.h:92
av_audio_fifo_size
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
AudioSurroundContext::filter
void(* filter)(AVFilterContext *ctx)
Definition: af_surround.c:68
channel_layout.h
audio channel layout utility functions
upmix_4_1
static void upmix_4_1(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:454
config_output
static int config_output(AVFilterLink *outlink)
Definition: af_surround.c:194
AV_CH_LAYOUT_3POINT1
#define AV_CH_LAYOUT_3POINT1
Definition: channel_layout.h:90
AudioSurroundContext::upmix_stereo
void(* upmix_stereo)(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:69
M_PI_2
#define M_PI_2
Definition: mathematics.h:55
fft_channel
static int fft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
Definition: af_surround.c:1299
outputs
static const AVFilterPad outputs[]
Definition: af_surround.c:1467
ctx
AVFormatContext * ctx
Definition: movenc.c:48
AV_OPT_TYPE_BOOL
Definition: opt.h:240
filter_5_0_side
static void filter_5_0_side(AVFilterContext *ctx)
Definition: af_surround.c:990
AudioSurroundContext::out_channel_layout_str
char * out_channel_layout_str
Definition: af_surround.c:32
DFT_R2C
Definition: avfft.h:72
av_rdft_end
void av_rdft_end(RDFTContext *s)
n
int n
Definition: avisynth_c.h:684
upmix_7_1_5_0_side
static void upmix_7_1_5_0_side(AVFilterContext *ctx, float c_re, float c_im, float mag_totall, float mag_totalr, float fl_phase, float fr_phase, float bl_phase, float br_phase, float sl_phase, float sr_phase, float xl, float yl, float xr, float yr, int n)
Definition: af_surround.c:776
av_rdft_init
RDFTContext * av_rdft_init(int nbits, enum RDFTransformType trans)
Set up a real FFT.
AV_CH_FRONT_CENTER
#define AV_CH_FRONT_CENTER
Definition: channel_layout.h:51
AV_CH_LAYOUT_5POINT1_BACK
#define AV_CH_LAYOUT_5POINT1_BACK
Definition: channel_layout.h:98
upmix_1_0
static void upmix_1_0(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:248
AVFilterChannelLayouts
A list of supported channel layouts.
Definition: formats.h:85
ff_af_surround
AVFilter ff_af_surround
Definition: af_surround.c:1477
AudioSurroundContext::rdft
RDFTContext ** rdft
Definition: af_surround.c:63
sinf
#define sinf(x)
Definition: libm.h:419
void
typedef void(RENAME(mix_any_func_type))
Definition: rematrix_template.c:52
ff_log2
#define ff_log2
Definition: intmath.h:50
avfft.h
FFT functions.
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)\", 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:194
AVClass
Describe the class of an AVClass context structure.
Definition: log.h:67
AVFilter
Filter definition.
Definition: avfilter.h:144
AVRational
Rational number (pair of numerator and denominator).
Definition: rational.h:58
AV_CH_LAYOUT_5POINT0_BACK
#define AV_CH_LAYOUT_5POINT0_BACK
Definition: channel_layout.h:97
AVFilter::name
const char * name
Filter name.
Definition: avfilter.h:148
filter_5_1_side
static void filter_5_1_side(AVFilterContext *ctx)
Definition: af_surround.c:1045
AVFilterContext::outputs
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
layouts
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
av_get_channel_layout_channel_index
int av_get_channel_layout_channel_index(uint64_t channel_layout, uint64_t channel)
Get the index of a channel in channel_layout.
Definition: channel_layout.c:233
ff_all_samplerates
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
upmix_7_0
static void upmix_7_0(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:670
AVFilterContext::internal
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
Definition: avfilter.h:378
av_audio_fifo_write
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
av_audio_fifo_drain
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
Definition: audio_fifo.c:201
M_LN10
#define M_LN10
Definition: mathematics.h:43
filter_frame
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_surround.c:1348
upmix_5_1_back_surround
static void upmix_5_1_back_surround(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float c_mag, float mag_total, float x, float y, int n)
Definition: af_surround.c:578
filter_surround
static void filter_surround(AVFilterContext *ctx)
Definition: af_surround.c:923
AV_CH_LAYOUT_2POINT1
#define AV_CH_LAYOUT_2POINT1
Definition: channel_layout.h:87
AudioSurroundContext::irdft
RDFTContext ** irdft
Definition: af_surround.c:63
AVFilterInternal::execute
avfilter_execute_func * execute
Definition: internal.h:155
audio_fifo.h
Audio FIFO Buffer.
upmix_3_1
static void upmix_3_1(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:350
AudioSurroundContext::in_channel_layout
uint64_t in_channel_layout
Definition: af_surround.c:52
AudioSurroundContext::in_channel_layout_str
char * in_channel_layout_str
Definition: af_surround.c:33
AVFilterFormats
A list of supported formats for one end of a filter link.
Definition: formats.h:64
av_audio_fifo_peek
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
Peek data from an AVAudioFifo.
Definition: audio_fifo.c:138
AVFilterContext
An instance of a filter.
Definition: avfilter.h:338
AudioSurroundContext::out_channel_layout
uint64_t out_channel_layout
Definition: af_surround.c:51
out
FILE * out
Definition: movenc.c:54
av_freep
#define av_freep(p)
Definition: tableprint_vlc.h:35
AVFILTER_DEFINE_CLASS
AVFILTER_DEFINE_CLASS(surround)
M_PI
#define M_PI
Definition: mathematics.h:52
av_malloc_array
#define av_malloc_array(a, b)
Definition: tableprint_vlc.h:32
ff_request_frame
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:407
formats
formats
Definition: signature.h:48
AudioSurroundContext::overlap_buffer
AVFrame * overlap_buffer
Definition: af_surround.c:58
AudioSurroundContext::fifo
AVAudioFifo * fifo
Definition: af_surround.c:62
AVFrame::extended_data
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:265
AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_MONO
Definition: channel_layout.h:85
filter_2_1
static void filter_2_1(AVFilterContext *ctx)
Definition: af_surround.c:957
inputs
static const AVFilterPad inputs[]
Definition: af_surround.c:1457
upmix_5_0_back
static void upmix_5_0_back(AVFilterContext *ctx, float l_phase, float r_phase, float c_phase, float mag_total, float x, float y, int n)
Definition: af_surround.c:494
AVFrame::nb_samples
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:284
for
for(j=16;j >0;--j)
Definition: h264pred_template.c:469
ff_set_common_samplerates
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
AV_NOPTS_VALUE
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248
a
a
Definition: h264pred_template.c:468
AudioSurroundContext::upmix_5_1
void(* upmix_5_1)(AVFilterContext *ctx, float c_re, float c_im, float lfe_re, float lfe_im, float mag_totall, float mag_totalr, float fl_phase, float fr_phase, float bl_phase, float br_phase, float sl_phase, float sr_phase, float xl, float yl, float xr, float yr, int n)
Definition: af_surround.c:102
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