FFmpeg
dcadsp.c
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1 /*
2  * Copyright (C) 2016 foo86
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/mem.h"
22 
23 #include "dcadsp.h"
24 #include "dcamath.h"
25 
26 static void decode_hf_c(int32_t **dst,
27  const int32_t *vq_index,
28  const int8_t hf_vq[1024][32],
29  int32_t scale_factors[32][2],
30  ptrdiff_t sb_start, ptrdiff_t sb_end,
31  ptrdiff_t ofs, ptrdiff_t len)
32 {
33  int i, j;
34 
35  for (i = sb_start; i < sb_end; i++) {
36  const int8_t *coeff = hf_vq[vq_index[i]];
37  int32_t scale = scale_factors[i][0];
38  for (j = 0; j < len; j++)
39  dst[i][j + ofs] = clip23(coeff[j] * scale + (1 << 3) >> 4);
40  }
41 }
42 
43 static void decode_joint_c(int32_t **dst, int32_t **src,
44  const int32_t *scale_factors,
45  ptrdiff_t sb_start, ptrdiff_t sb_end,
46  ptrdiff_t ofs, ptrdiff_t len)
47 {
48  int i, j;
49 
50  for (i = sb_start; i < sb_end; i++) {
51  int32_t scale = scale_factors[i];
52  for (j = 0; j < len; j++)
53  dst[i][j + ofs] = clip23(mul17(src[i][j + ofs], scale));
54  }
55 }
56 
57 static void lfe_fir_float_c(float *pcm_samples, int32_t *lfe_samples,
58  const float *filter_coeff, ptrdiff_t npcmblocks,
59  int dec_select)
60 {
61  // Select decimation factor
62  int factor = 64 << dec_select;
63  int ncoeffs = 8 >> dec_select;
64  int nlfesamples = npcmblocks >> (dec_select + 1);
65  int i, j, k;
66 
67  for (i = 0; i < nlfesamples; i++) {
68  // One decimated sample generates 64 or 128 interpolated ones
69  for (j = 0; j < factor / 2; j++) {
70  float a = 0;
71  float b = 0;
72 
73  for (k = 0; k < ncoeffs; k++) {
74  a += filter_coeff[ j * ncoeffs + k] * lfe_samples[-k];
75  b += filter_coeff[255 - j * ncoeffs - k] * lfe_samples[-k];
76  }
77 
78  pcm_samples[ j] = a;
79  pcm_samples[factor / 2 + j] = b;
80  }
81 
82  lfe_samples++;
83  pcm_samples += factor;
84  }
85 }
86 
87 static void lfe_fir0_float_c(float *pcm_samples, int32_t *lfe_samples,
88  const float *filter_coeff, ptrdiff_t npcmblocks)
89 {
90  lfe_fir_float_c(pcm_samples, lfe_samples, filter_coeff, npcmblocks, 0);
91 }
92 
93 static void lfe_fir1_float_c(float *pcm_samples, int32_t *lfe_samples,
94  const float *filter_coeff, ptrdiff_t npcmblocks)
95 {
96  lfe_fir_float_c(pcm_samples, lfe_samples, filter_coeff, npcmblocks, 1);
97 }
98 
99 static void lfe_x96_float_c(float *dst, const float *src,
100  float *hist, ptrdiff_t len)
101 {
102  float prev = *hist;
103  int i;
104 
105  for (i = 0; i < len; i++) {
106  float a = 0.25f * src[i] + 0.75f * prev;
107  float b = 0.75f * src[i] + 0.25f * prev;
108  prev = src[i];
109  *dst++ = a;
110  *dst++ = b;
111  }
112 
113  *hist = prev;
114 }
115 
117  FFTContext *imdct,
118  float *pcm_samples,
119  int32_t **subband_samples_lo,
120  int32_t **subband_samples_hi,
121  float *hist1, int *offset, float *hist2,
122  const float *filter_coeff, ptrdiff_t npcmblocks,
123  float scale)
124 {
125  LOCAL_ALIGNED_32(float, input, [32]);
126  int i, j;
127 
128  for (j = 0; j < npcmblocks; j++) {
129  // Load in one sample from each subband
130  for (i = 0; i < 32; i++) {
131  if ((i - 1) & 2)
132  input[i] = -subband_samples_lo[i][j];
133  else
134  input[i] = subband_samples_lo[i][j];
135  }
136 
137  // One subband sample generates 32 interpolated ones
138  synth->synth_filter_float(imdct, hist1, offset,
139  hist2, filter_coeff,
140  pcm_samples, input, scale);
141  pcm_samples += 32;
142  }
143 }
144 
146  FFTContext *imdct,
147  float *pcm_samples,
148  int32_t **subband_samples_lo,
149  int32_t **subband_samples_hi,
150  float *hist1, int *offset, float *hist2,
151  const float *filter_coeff, ptrdiff_t npcmblocks,
152  float scale)
153 {
154  LOCAL_ALIGNED_32(float, input, [64]);
155  int i, j;
156 
157  if (!subband_samples_hi)
158  memset(&input[32], 0, sizeof(input[0]) * 32);
159 
160  for (j = 0; j < npcmblocks; j++) {
161  // Load in one sample from each subband
162  if (subband_samples_hi) {
163  // Full 64 subbands, first 32 are residual coded
164  for (i = 0; i < 32; i++) {
165  if ((i - 1) & 2)
166  input[i] = -subband_samples_lo[i][j] - subband_samples_hi[i][j];
167  else
168  input[i] = subband_samples_lo[i][j] + subband_samples_hi[i][j];
169  }
170  for (i = 32; i < 64; i++) {
171  if ((i - 1) & 2)
172  input[i] = -subband_samples_hi[i][j];
173  else
174  input[i] = subband_samples_hi[i][j];
175  }
176  } else {
177  // Only first 32 subbands
178  for (i = 0; i < 32; i++) {
179  if ((i - 1) & 2)
180  input[i] = -subband_samples_lo[i][j];
181  else
182  input[i] = subband_samples_lo[i][j];
183  }
184  }
185 
186  // One subband sample generates 64 interpolated ones
187  synth->synth_filter_float_64(imdct, hist1, offset,
188  hist2, filter_coeff,
189  pcm_samples, input, scale);
190  pcm_samples += 64;
191  }
192 }
193 
194 static void lfe_fir_fixed_c(int32_t *pcm_samples, int32_t *lfe_samples,
195  const int32_t *filter_coeff, ptrdiff_t npcmblocks)
196 {
197  // Select decimation factor
198  int nlfesamples = npcmblocks >> 1;
199  int i, j, k;
200 
201  for (i = 0; i < nlfesamples; i++) {
202  // One decimated sample generates 64 interpolated ones
203  for (j = 0; j < 32; j++) {
204  int64_t a = 0;
205  int64_t b = 0;
206 
207  for (k = 0; k < 8; k++) {
208  a += (int64_t)filter_coeff[ j * 8 + k] * lfe_samples[-k];
209  b += (int64_t)filter_coeff[255 - j * 8 - k] * lfe_samples[-k];
210  }
211 
212  pcm_samples[ j] = clip23(norm23(a));
213  pcm_samples[32 + j] = clip23(norm23(b));
214  }
215 
216  lfe_samples++;
217  pcm_samples += 64;
218  }
219 }
220 
221 static void lfe_x96_fixed_c(int32_t *dst, const int32_t *src,
222  int32_t *hist, ptrdiff_t len)
223 {
224  int32_t prev = *hist;
225  int i;
226 
227  for (i = 0; i < len; i++) {
228  int64_t a = INT64_C(2097471) * src[i] + INT64_C(6291137) * prev;
229  int64_t b = INT64_C(6291137) * src[i] + INT64_C(2097471) * prev;
230  prev = src[i];
231  *dst++ = clip23(norm23(a));
232  *dst++ = clip23(norm23(b));
233  }
234 
235  *hist = prev;
236 }
237 
239  DCADCTContext *imdct,
240  int32_t *pcm_samples,
241  int32_t **subband_samples_lo,
242  int32_t **subband_samples_hi,
243  int32_t *hist1, int *offset, int32_t *hist2,
244  const int32_t *filter_coeff, ptrdiff_t npcmblocks)
245 {
247  int i, j;
248 
249  for (j = 0; j < npcmblocks; j++) {
250  // Load in one sample from each subband
251  for (i = 0; i < 32; i++)
252  input[i] = subband_samples_lo[i][j];
253 
254  // One subband sample generates 32 interpolated ones
255  synth->synth_filter_fixed(imdct, hist1, offset,
256  hist2, filter_coeff,
257  pcm_samples, input);
258  pcm_samples += 32;
259  }
260 }
261 
263  DCADCTContext *imdct,
264  int32_t *pcm_samples,
265  int32_t **subband_samples_lo,
266  int32_t **subband_samples_hi,
267  int32_t *hist1, int *offset, int32_t *hist2,
268  const int32_t *filter_coeff, ptrdiff_t npcmblocks)
269 {
271  int i, j;
272 
273  if (!subband_samples_hi)
274  memset(&input[32], 0, sizeof(input[0]) * 32);
275 
276  for (j = 0; j < npcmblocks; j++) {
277  // Load in one sample from each subband
278  if (subband_samples_hi) {
279  // Full 64 subbands, first 32 are residual coded
280  for (i = 0; i < 32; i++)
281  input[i] = subband_samples_lo[i][j] + subband_samples_hi[i][j];
282  for (i = 32; i < 64; i++)
283  input[i] = subband_samples_hi[i][j];
284  } else {
285  // Only first 32 subbands
286  for (i = 0; i < 32; i++)
287  input[i] = subband_samples_lo[i][j];
288  }
289 
290  // One subband sample generates 64 interpolated ones
291  synth->synth_filter_fixed_64(imdct, hist1, offset,
292  hist2, filter_coeff,
293  pcm_samples, input);
294  pcm_samples += 64;
295  }
296 }
297 
298 static void decor_c(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)
299 {
300  int i;
301 
302  for (i = 0; i < len; i++)
303  dst[i] += (SUINT)((int)(src[i] * (SUINT)coeff + (1 << 2)) >> 3);
304 }
305 
306 static void dmix_sub_xch_c(int32_t *dst1, int32_t *dst2,
307  const int32_t *src, ptrdiff_t len)
308 {
309  int i;
310 
311  for (i = 0; i < len; i++) {
312  int32_t cs = mul23(src[i], 5931520 /* M_SQRT1_2 * (1 << 23) */);
313  dst1[i] -= cs;
314  dst2[i] -= cs;
315  }
316 }
317 
318 static void dmix_sub_c(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)
319 {
320  int i;
321 
322  for (i = 0; i < len; i++)
323  dst[i] -= (unsigned)mul15(src[i], coeff);
324 }
325 
326 static void dmix_add_c(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)
327 {
328  int i;
329 
330  for (i = 0; i < len; i++)
331  dst[i] += (unsigned)mul15(src[i], coeff);
332 }
333 
334 static void dmix_scale_c(int32_t *dst, int scale, ptrdiff_t len)
335 {
336  int i;
337 
338  for (i = 0; i < len; i++)
339  dst[i] = mul15(dst[i], scale);
340 }
341 
342 static void dmix_scale_inv_c(int32_t *dst, int scale_inv, ptrdiff_t len)
343 {
344  int i;
345 
346  for (i = 0; i < len; i++)
347  dst[i] = mul16(dst[i], scale_inv);
348 }
349 
350 static void filter0(SUINT32 *dst, const int32_t *src, int32_t coeff, ptrdiff_t len)
351 {
352  int i;
353 
354  for (i = 0; i < len; i++)
355  dst[i] -= mul22(src[i], coeff);
356 }
357 
358 static void filter1(SUINT32 *dst, const int32_t *src, int32_t coeff, ptrdiff_t len)
359 {
360  int i;
361 
362  for (i = 0; i < len; i++)
363  dst[i] -= mul23(src[i], coeff);
364 }
365 
367  const int32_t *coeff, ptrdiff_t len)
368 {
369  int i;
370 
371  filter0(src0, src1, coeff[0], len);
372  filter0(src1, src0, coeff[1], len);
373  filter0(src0, src1, coeff[2], len);
374  filter0(src1, src0, coeff[3], len);
375 
376  for (i = 0; i < 8; i++, src0--) {
377  filter1(src0, src1, coeff[i + 4], len);
378  filter1(src1, src0, coeff[i + 12], len);
379  filter1(src0, src1, coeff[i + 4], len);
380  }
381 
382  for (i = 0; i < len; i++) {
383  *dst++ = *src1++;
384  *dst++ = *++src0;
385  }
386 }
387 
388 static void lbr_bank_c(float output[32][4], float **input,
389  const float *coeff, ptrdiff_t ofs, ptrdiff_t len)
390 {
391  float SW0 = coeff[0];
392  float SW1 = coeff[1];
393  float SW2 = coeff[2];
394  float SW3 = coeff[3];
395 
396  float C1 = coeff[4];
397  float C2 = coeff[5];
398  float C3 = coeff[6];
399  float C4 = coeff[7];
400 
401  float AL1 = coeff[8];
402  float AL2 = coeff[9];
403 
404  int i;
405 
406  // Short window and 8 point forward MDCT
407  for (i = 0; i < len; i++) {
408  float *src = input[i] + ofs;
409 
410  float a = src[-4] * SW0 - src[-1] * SW3;
411  float b = src[-3] * SW1 - src[-2] * SW2;
412  float c = src[ 2] * SW1 + src[ 1] * SW2;
413  float d = src[ 3] * SW0 + src[ 0] * SW3;
414 
415  output[i][0] = C1 * b - C2 * c + C4 * a - C3 * d;
416  output[i][1] = C1 * d - C2 * a - C4 * b - C3 * c;
417  output[i][2] = C3 * b + C2 * d - C4 * c + C1 * a;
418  output[i][3] = C3 * a - C2 * b + C4 * d - C1 * c;
419  }
420 
421  // Aliasing cancellation for high frequencies
422  for (i = 12; i < len - 1; i++) {
423  float a = output[i ][3] * AL1;
424  float b = output[i+1][0] * AL1;
425  output[i ][3] += b - a;
426  output[i+1][0] -= b + a;
427  a = output[i ][2] * AL2;
428  b = output[i+1][1] * AL2;
429  output[i ][2] += b - a;
430  output[i+1][1] -= b + a;
431  }
432 }
433 
434 static void lfe_iir_c(float *output, const float *input,
435  const float iir[5][4], float hist[5][2],
436  ptrdiff_t factor)
437 {
438  float res, tmp;
439  int i, j, k;
440 
441  for (i = 0; i < 64; i++) {
442  res = *input++;
443 
444  for (j = 0; j < factor; j++) {
445  for (k = 0; k < 5; k++) {
446  tmp = hist[k][0] * iir[k][0] + hist[k][1] * iir[k][1] + res;
447  res = hist[k][0] * iir[k][2] + hist[k][1] * iir[k][3] + tmp;
448 
449  hist[k][0] = hist[k][1];
450  hist[k][1] = tmp;
451  }
452 
453  *output++ = res;
454  res = 0;
455  }
456  }
457 }
458 
460 {
461  s->decode_hf = decode_hf_c;
462  s->decode_joint = decode_joint_c;
463 
464  s->lfe_fir_float[0] = lfe_fir0_float_c;
465  s->lfe_fir_float[1] = lfe_fir1_float_c;
466  s->lfe_x96_float = lfe_x96_float_c;
467  s->sub_qmf_float[0] = sub_qmf32_float_c;
468  s->sub_qmf_float[1] = sub_qmf64_float_c;
469 
470  s->lfe_fir_fixed = lfe_fir_fixed_c;
471  s->lfe_x96_fixed = lfe_x96_fixed_c;
472  s->sub_qmf_fixed[0] = sub_qmf32_fixed_c;
473  s->sub_qmf_fixed[1] = sub_qmf64_fixed_c;
474 
475  s->decor = decor_c;
476 
477  s->dmix_sub_xch = dmix_sub_xch_c;
478  s->dmix_sub = dmix_sub_c;
479  s->dmix_add = dmix_add_c;
480  s->dmix_scale = dmix_scale_c;
481  s->dmix_scale_inv = dmix_scale_inv_c;
482 
483  s->assemble_freq_bands = assemble_freq_bands_c;
484 
485  s->lbr_bank = lbr_bank_c;
486  s->lfe_iir = lfe_iir_c;
487 
488  if (ARCH_X86)
490 }
dcamath.h
mul22
static int32_t mul22(int32_t a, int32_t b)
Definition: dcamath.h:49
lfe_iir_c
static void lfe_iir_c(float *output, const float *input, const float iir[5][4], float hist[5][2], ptrdiff_t factor)
Definition: dcadsp.c:434
mul17
static int32_t mul17(int32_t a, int32_t b)
Definition: dcamath.h:48
C2
#define C2
Definition: mpegaudiodsp_template.c:268
filter1
static void filter1(SUINT32 *dst, const int32_t *src, int32_t coeff, ptrdiff_t len)
Definition: dcadsp.c:358
mul15
static int32_t mul15(int32_t a, int32_t b)
Definition: dcamath.h:46
output
filter_frame For filters that do not use the this method is called when a frame is pushed to the filter s input It can be called at any time except in a reentrant way If the input frame is enough to produce output
Definition: filter_design.txt:225
SynthFilterContext::synth_filter_float
void(* synth_filter_float)(FFTContext *imdct, float *synth_buf_ptr, int *synth_buf_offset, float synth_buf2[32], const float window[512], float out[32], const float in[32], float scale)
Definition: synth_filter.h:28
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:26
C4
#define C4
Definition: mpegaudiodec_template.c:455
b
#define b
Definition: input.c:41
dmix_sub_xch_c
static void dmix_sub_xch_c(int32_t *dst1, int32_t *dst2, const int32_t *src, ptrdiff_t len)
Definition: dcadsp.c:306
mul23
static int32_t mul23(int32_t a, int32_t b)
Definition: dcamath.h:50
decode_hf_c
static void decode_hf_c(int32_t **dst, const int32_t *vq_index, const int8_t hf_vq[1024][32], int32_t scale_factors[32][2], ptrdiff_t sb_start, ptrdiff_t sb_end, ptrdiff_t ofs, ptrdiff_t len)
Definition: dcadsp.c:26
lbr_bank_c
static void lbr_bank_c(float output[32][4], float **input, const float *coeff, ptrdiff_t ofs, ptrdiff_t len)
Definition: dcadsp.c:388
SynthFilterContext
Definition: synth_filter.h:27
C1
#define C1
Definition: mpegaudiodsp_template.c:267
SUINT32
#define SUINT32
Definition: dct32_template.c:31
clip23
static int32_t clip23(int32_t a)
Definition: dcamath.h:54
src
#define src
Definition: vp8dsp.c:254
SynthFilterContext::synth_filter_fixed_64
void(* synth_filter_fixed_64)(DCADCTContext *imdct, int32_t *synth_buf_ptr, int *synth_buf_offset, int32_t synth_buf2[64], const int32_t window[1024], int32_t out[64], const int32_t in[64])
Definition: synth_filter.h:41
dmix_sub_c
static void dmix_sub_c(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)
Definition: dcadsp.c:318
av_cold
#define av_cold
Definition: attributes.h:84
sub_qmf32_fixed_c
static void sub_qmf32_fixed_c(SynthFilterContext *synth, DCADCTContext *imdct, int32_t *pcm_samples, int32_t **subband_samples_lo, int32_t **subband_samples_hi, int32_t *hist1, int *offset, int32_t *hist2, const int32_t *filter_coeff, ptrdiff_t npcmblocks)
Definition: dcadsp.c:238
SynthFilterContext::synth_filter_fixed
void(* synth_filter_fixed)(DCADCTContext *imdct, int32_t *synth_buf_ptr, int *synth_buf_offset, int32_t synth_buf2[32], const int32_t window[512], int32_t out[32], const int32_t in[32])
Definition: synth_filter.h:37
decode_joint_c
static void decode_joint_c(int32_t **dst, int32_t **src, const int32_t *scale_factors, ptrdiff_t sb_start, ptrdiff_t sb_end, ptrdiff_t ofs, ptrdiff_t len)
Definition: dcadsp.c:43
s
#define s(width, name)
Definition: cbs_vp9.c:257
dmix_add_c
static void dmix_add_c(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)
Definition: dcadsp.c:326
DCADCTContext
Definition: dcadct.h:26
lfe_fir0_float_c
static void lfe_fir0_float_c(float *pcm_samples, int32_t *lfe_samples, const float *filter_coeff, ptrdiff_t npcmblocks)
Definition: dcadsp.c:87
mul16
static int32_t mul16(int32_t a, int32_t b)
Definition: dcamath.h:47
lfe_fir_float_c
static void lfe_fir_float_c(float *pcm_samples, int32_t *lfe_samples, const float *filter_coeff, ptrdiff_t npcmblocks, int dec_select)
Definition: dcadsp.c:57
assemble_freq_bands_c
static void assemble_freq_bands_c(int32_t *dst, int32_t *src0, int32_t *src1, const int32_t *coeff, ptrdiff_t len)
Definition: dcadsp.c:366
dmix_scale_inv_c
static void dmix_scale_inv_c(int32_t *dst, int scale_inv, ptrdiff_t len)
Definition: dcadsp.c:342
int32_t
int32_t
Definition: audio_convert.c:194
dmix_scale_c
static void dmix_scale_c(int32_t *dst, int scale, ptrdiff_t len)
Definition: dcadsp.c:334
sub_qmf64_float_c
static void sub_qmf64_float_c(SynthFilterContext *synth, FFTContext *imdct, float *pcm_samples, int32_t **subband_samples_lo, int32_t **subband_samples_hi, float *hist1, int *offset, float *hist2, const float *filter_coeff, ptrdiff_t npcmblocks, float scale)
Definition: dcadsp.c:145
sub_qmf64_fixed_c
static void sub_qmf64_fixed_c(SynthFilterContext *synth, DCADCTContext *imdct, int32_t *pcm_samples, int32_t **subband_samples_lo, int32_t **subband_samples_hi, int32_t *hist1, int *offset, int32_t *hist2, const int32_t *filter_coeff, ptrdiff_t npcmblocks)
Definition: dcadsp.c:262
DCADSPContext
Definition: dcadsp.h:30
c
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
ff_dcadsp_init
av_cold void ff_dcadsp_init(DCADSPContext *s)
Definition: dcadsp.c:459
SynthFilterContext::synth_filter_float_64
void(* synth_filter_float_64)(FFTContext *imdct, float *synth_buf_ptr, int *synth_buf_offset, float synth_buf2[64], const float window[1024], float out[64], const float in[64], float scale)
Definition: synth_filter.h:33
decor_c
static void decor_c(int32_t *dst, const int32_t *src, int coeff, ptrdiff_t len)
Definition: dcadsp.c:298
a
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:41
lfe_x96_float_c
static void lfe_x96_float_c(float *dst, const float *src, float *hist, ptrdiff_t len)
Definition: dcadsp.c:99
offset
it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf offset
Definition: writing_filters.txt:86
input
and forward the test the status of outputs and forward it to the corresponding return FFERROR_NOT_READY If the filters stores internally one or a few frame for some input
Definition: filter_design.txt:172
dcadsp.h
src0
#define src0
Definition: h264pred.c:138
lfe_fir1_float_c
static void lfe_fir1_float_c(float *pcm_samples, int32_t *lfe_samples, const float *filter_coeff, ptrdiff_t npcmblocks)
Definition: dcadsp.c:93
src1
#define src1
Definition: h264pred.c:139
lfe_x96_fixed_c
static void lfe_x96_fixed_c(int32_t *dst, const int32_t *src, int32_t *hist, ptrdiff_t len)
Definition: dcadsp.c:221
FFTContext
Definition: fft.h:88
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
SUINT
#define SUINT
Definition: dct32_template.c:30
sub_qmf32_float_c
static void sub_qmf32_float_c(SynthFilterContext *synth, FFTContext *imdct, float *pcm_samples, int32_t **subband_samples_lo, int32_t **subband_samples_hi, float *hist1, int *offset, float *hist2, const float *filter_coeff, ptrdiff_t npcmblocks, float scale)
Definition: dcadsp.c:116
len
int len
Definition: vorbis_enc_data.h:452
norm23
static int32_t norm23(int64_t a)
Definition: dcamath.h:44
C3
#define C3
Definition: mpegaudiodec_template.c:454
factor
static const int factor[16]
Definition: vf_pp7.c:75
filter0
static void filter0(SUINT32 *dst, const int32_t *src, int32_t coeff, ptrdiff_t len)
Definition: dcadsp.c:350
mem.h
ff_dcadsp_init_x86
av_cold void ff_dcadsp_init_x86(DCADSPContext *s)
Definition: dcadsp_init.c:36
lfe_fir_fixed_c
static void lfe_fir_fixed_c(int32_t *pcm_samples, int32_t *lfe_samples, const int32_t *filter_coeff, ptrdiff_t npcmblocks)
Definition: dcadsp.c:194
coeff
static const double coeff[2][5]
Definition: vf_owdenoise.c:72
LOCAL_ALIGNED_32
#define LOCAL_ALIGNED_32(t, v,...)
Definition: internal.h:137
int
int
Definition: ffmpeg_filter.c:191