FFmpeg
mpegaudiodsp_template.c
Go to the documentation of this file.
1 /*
2  * Copyright (c) 2001, 2002 Fabrice Bellard
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 <stdint.h>
22 
23 #include "libavutil/attributes.h"
24 #include "libavutil/mem.h"
25 #include "dct32.h"
26 #include "mathops.h"
27 #include "mpegaudiodsp.h"
28 #include "mpegaudio.h"
29 
30 #if USE_FLOATS
31 #define RENAME(n) n##_float
32 
33 static inline float round_sample(float *sum)
34 {
35  float sum1=*sum;
36  *sum = 0;
37  return sum1;
38 }
39 
40 #define MACS(rt, ra, rb) rt+=(ra)*(rb)
41 #define MULS(ra, rb) ((ra)*(rb))
42 #define MULH3(x, y, s) ((s)*(y)*(x))
43 #define MLSS(rt, ra, rb) rt-=(ra)*(rb)
44 #define MULLx(x, y, s) ((y)*(x))
45 #define FIXHR(x) ((float)(x))
46 #define FIXR(x) ((float)(x))
47 #define SHR(a,b) ((a)*(1.0f/(1<<(b))))
48 
49 #else
50 
51 #define RENAME(n) n##_fixed
52 #define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
53 
54 static inline int round_sample(int64_t *sum)
55 {
56  int sum1;
57  sum1 = (int)((*sum) >> OUT_SHIFT);
58  *sum &= (1<<OUT_SHIFT)-1;
59  return av_clip_int16(sum1);
60 }
61 
62 # define MULS(ra, rb) MUL64(ra, rb)
63 # define MACS(rt, ra, rb) MAC64(rt, ra, rb)
64 # define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
65 # define MULH3(x, y, s) MULH((s)*(x), y)
66 # define MULLx(x, y, s) MULL((int)(x),(y),s)
67 # define SHR(a,b) (((int)(a))>>(b))
68 # define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
69 # define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
70 #endif
71 
72 /** Window for MDCT. Actually only the elements in [0,17] and
73  [MDCT_BUF_SIZE/2, MDCT_BUF_SIZE/2 + 17] are actually used. The rest
74  is just to preserve alignment for SIMD implementations.
75 */
76 DECLARE_ALIGNED(16, INTFLOAT, RENAME(ff_mdct_win))[8][MDCT_BUF_SIZE];
77 
78 DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
79 
80 #define SUM8(op, sum, w, p) \
81 { \
82  op(sum, (w)[0 * 64], (p)[0 * 64]); \
83  op(sum, (w)[1 * 64], (p)[1 * 64]); \
84  op(sum, (w)[2 * 64], (p)[2 * 64]); \
85  op(sum, (w)[3 * 64], (p)[3 * 64]); \
86  op(sum, (w)[4 * 64], (p)[4 * 64]); \
87  op(sum, (w)[5 * 64], (p)[5 * 64]); \
88  op(sum, (w)[6 * 64], (p)[6 * 64]); \
89  op(sum, (w)[7 * 64], (p)[7 * 64]); \
90 }
91 
92 #define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
93 { \
94  INTFLOAT tmp;\
95  tmp = p[0 * 64];\
96  op1(sum1, (w1)[0 * 64], tmp);\
97  op2(sum2, (w2)[0 * 64], tmp);\
98  tmp = p[1 * 64];\
99  op1(sum1, (w1)[1 * 64], tmp);\
100  op2(sum2, (w2)[1 * 64], tmp);\
101  tmp = p[2 * 64];\
102  op1(sum1, (w1)[2 * 64], tmp);\
103  op2(sum2, (w2)[2 * 64], tmp);\
104  tmp = p[3 * 64];\
105  op1(sum1, (w1)[3 * 64], tmp);\
106  op2(sum2, (w2)[3 * 64], tmp);\
107  tmp = p[4 * 64];\
108  op1(sum1, (w1)[4 * 64], tmp);\
109  op2(sum2, (w2)[4 * 64], tmp);\
110  tmp = p[5 * 64];\
111  op1(sum1, (w1)[5 * 64], tmp);\
112  op2(sum2, (w2)[5 * 64], tmp);\
113  tmp = p[6 * 64];\
114  op1(sum1, (w1)[6 * 64], tmp);\
115  op2(sum2, (w2)[6 * 64], tmp);\
116  tmp = p[7 * 64];\
117  op1(sum1, (w1)[7 * 64], tmp);\
118  op2(sum2, (w2)[7 * 64], tmp);\
119 }
120 
121 void RENAME(ff_mpadsp_apply_window)(MPA_INT *synth_buf, MPA_INT *window,
122  int *dither_state, OUT_INT *samples,
123  ptrdiff_t incr)
124 {
125  register const MPA_INT *w, *w2, *p;
126  int j;
127  OUT_INT *samples2;
128 #if USE_FLOATS
129  float sum, sum2;
130 #else
131  int64_t sum, sum2;
132 #endif
133 
134  /* copy to avoid wrap */
135  memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
136 
137  samples2 = samples + 31 * incr;
138  w = window;
139  w2 = window + 31;
140 
141  sum = *dither_state;
142  p = synth_buf + 16;
143  SUM8(MACS, sum, w, p);
144  p = synth_buf + 48;
145  SUM8(MLSS, sum, w + 32, p);
146  *samples = round_sample(&sum);
147  samples += incr;
148  w++;
149 
150  /* we calculate two samples at the same time to avoid one memory
151  access per two sample */
152  for(j=1;j<16;j++) {
153  sum2 = 0;
154  p = synth_buf + 16 + j;
155  SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
156  p = synth_buf + 48 - j;
157  SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
158 
159  *samples = round_sample(&sum);
160  samples += incr;
161  sum += sum2;
162  *samples2 = round_sample(&sum);
163  samples2 -= incr;
164  w++;
165  w2--;
166  }
167 
168  p = synth_buf + 32;
169  SUM8(MLSS, sum, w + 32, p);
170  *samples = round_sample(&sum);
171  *dither_state= sum;
172 }
173 
174 /* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
175  32 samples. */
176 void RENAME(ff_mpa_synth_filter)(MPADSPContext *s, MPA_INT *synth_buf_ptr,
177  int *synth_buf_offset,
178  MPA_INT *window, int *dither_state,
179  OUT_INT *samples, ptrdiff_t incr,
180  MPA_INT *sb_samples)
181 {
182  MPA_INT *synth_buf;
183  int offset;
184 
185  offset = *synth_buf_offset;
186  synth_buf = synth_buf_ptr + offset;
187 
188  s->RENAME(dct32)(synth_buf, sb_samples);
189  s->RENAME(apply_window)(synth_buf, window, dither_state, samples, incr);
190 
191  offset = (offset - 32) & 511;
192  *synth_buf_offset = offset;
193 }
194 
195 av_cold void RENAME(ff_mpa_synth_init)(MPA_INT *window)
196 {
197  int i, j;
198 
199  /* max = 18760, max sum over all 16 coefs : 44736 */
200  for(i=0;i<257;i++) {
201  INTFLOAT v;
202  v = ff_mpa_enwindow[i];
203 #if USE_FLOATS
204  v *= 1.0 / (1LL<<(16 + FRAC_BITS));
205 #endif
206  window[i] = v;
207  if ((i & 63) != 0)
208  v = -v;
209  if (i != 0)
210  window[512 - i] = v;
211  }
212 
213 
214  // Needed for avoiding shuffles in ASM implementations
215  for(i=0; i < 8; i++)
216  for(j=0; j < 16; j++)
217  window[512+16*i+j] = window[64*i+32-j];
218 
219  for(i=0; i < 8; i++)
220  for(j=0; j < 16; j++)
221  window[512+128+16*i+j] = window[64*i+48-j];
222 }
223 
224 av_cold void RENAME(ff_init_mpadsp_tabs)(void)
225 {
226  int i, j;
227  /* compute mdct windows */
228  for (i = 0; i < 36; i++) {
229  for (j = 0; j < 4; j++) {
230  double d;
231 
232  if (j == 2 && i % 3 != 1)
233  continue;
234 
235  d = sin(M_PI * (i + 0.5) / 36.0);
236  if (j == 1) {
237  if (i >= 30) d = 0;
238  else if (i >= 24) d = sin(M_PI * (i - 18 + 0.5) / 12.0);
239  else if (i >= 18) d = 1;
240  } else if (j == 3) {
241  if (i < 6) d = 0;
242  else if (i < 12) d = sin(M_PI * (i - 6 + 0.5) / 12.0);
243  else if (i < 18) d = 1;
244  }
245  //merge last stage of imdct into the window coefficients
246  d *= 0.5 * IMDCT_SCALAR / cos(M_PI * (2 * i + 19) / 72);
247 
248  if (j == 2)
249  RENAME(ff_mdct_win)[j][i/3] = FIXHR((d / (1<<5)));
250  else {
251  int idx = i < 18 ? i : i + (MDCT_BUF_SIZE/2 - 18);
252  RENAME(ff_mdct_win)[j][idx] = FIXHR((d / (1<<5)));
253  }
254  }
255  }
256 
257  /* NOTE: we do frequency inversion adter the MDCT by changing
258  the sign of the right window coefs */
259  for (j = 0; j < 4; j++) {
260  for (i = 0; i < MDCT_BUF_SIZE; i += 2) {
261  RENAME(ff_mdct_win)[j + 4][i ] = RENAME(ff_mdct_win)[j][i ];
262  RENAME(ff_mdct_win)[j + 4][i + 1] = -RENAME(ff_mdct_win)[j][i + 1];
263  }
264  }
265 }
266 /* cos(pi*i/18) */
267 #define C1 FIXHR(0.98480775301220805936/2)
268 #define C2 FIXHR(0.93969262078590838405/2)
269 #define C3 FIXHR(0.86602540378443864676/2)
270 #define C4 FIXHR(0.76604444311897803520/2)
271 #define C5 FIXHR(0.64278760968653932632/2)
272 #define C6 FIXHR(0.5/2)
273 #define C7 FIXHR(0.34202014332566873304/2)
274 #define C8 FIXHR(0.17364817766693034885/2)
275 
276 /* 0.5 / cos(pi*(2*i+1)/36) */
277 static const INTFLOAT icos36[9] = {
278  FIXR(0.50190991877167369479),
279  FIXR(0.51763809020504152469), //0
280  FIXR(0.55168895948124587824),
281  FIXR(0.61038729438072803416),
282  FIXR(0.70710678118654752439), //1
283  FIXR(0.87172339781054900991),
284  FIXR(1.18310079157624925896),
285  FIXR(1.93185165257813657349), //2
286  FIXR(5.73685662283492756461),
287 };
288 
289 /* 0.5 / cos(pi*(2*i+1)/36) */
290 static const INTFLOAT icos36h[9] = {
291  FIXHR(0.50190991877167369479/2),
292  FIXHR(0.51763809020504152469/2), //0
293  FIXHR(0.55168895948124587824/2),
294  FIXHR(0.61038729438072803416/2),
295  FIXHR(0.70710678118654752439/2), //1
296  FIXHR(0.87172339781054900991/2),
297  FIXHR(1.18310079157624925896/4),
298  FIXHR(1.93185165257813657349/4), //2
299 // FIXHR(5.73685662283492756461),
300 };
301 
302 /* using Lee like decomposition followed by hand coded 9 points DCT */
303 static void imdct36(INTFLOAT *out, INTFLOAT *buf, SUINTFLOAT *in, INTFLOAT *win)
304 {
305  int i, j;
306  SUINTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
307  SUINTFLOAT tmp[18], *tmp1, *in1;
308 
309  for (i = 17; i >= 1; i--)
310  in[i] += in[i-1];
311  for (i = 17; i >= 3; i -= 2)
312  in[i] += in[i-2];
313 
314  for (j = 0; j < 2; j++) {
315  tmp1 = tmp + j;
316  in1 = in + j;
317 
318  t2 = in1[2*4] + in1[2*8] - in1[2*2];
319 
320  t3 = in1[2*0] + SHR(in1[2*6],1);
321  t1 = in1[2*0] - in1[2*6];
322  tmp1[ 6] = t1 - SHR(t2,1);
323  tmp1[16] = t1 + t2;
324 
325  t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
326  t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
327  t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
328 
329  tmp1[10] = t3 - t0 - t2;
330  tmp1[ 2] = t3 + t0 + t1;
331  tmp1[14] = t3 + t2 - t1;
332 
333  tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
334  t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
335  t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
336  t0 = MULH3(in1[2*3], C3, 2);
337 
338  t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
339 
340  tmp1[ 0] = t2 + t3 + t0;
341  tmp1[12] = t2 + t1 - t0;
342  tmp1[ 8] = t3 - t1 - t0;
343  }
344 
345  i = 0;
346  for (j = 0; j < 4; j++) {
347  t0 = tmp[i];
348  t1 = tmp[i + 2];
349  s0 = t1 + t0;
350  s2 = t1 - t0;
351 
352  t2 = tmp[i + 1];
353  t3 = tmp[i + 3];
354  s1 = MULH3(t3 + t2, icos36h[ j], 2);
355  s3 = MULLx(t3 - t2, icos36 [8 - j], FRAC_BITS);
356 
357  t0 = s0 + s1;
358  t1 = s0 - s1;
359  out[(9 + j) * SBLIMIT] = MULH3(t1, win[ 9 + j], 1) + buf[4*(9 + j)];
360  out[(8 - j) * SBLIMIT] = MULH3(t1, win[ 8 - j], 1) + buf[4*(8 - j)];
361  buf[4 * ( 9 + j )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 9 + j], 1);
362  buf[4 * ( 8 - j )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 8 - j], 1);
363 
364  t0 = s2 + s3;
365  t1 = s2 - s3;
366  out[(9 + 8 - j) * SBLIMIT] = MULH3(t1, win[ 9 + 8 - j], 1) + buf[4*(9 + 8 - j)];
367  out[ j * SBLIMIT] = MULH3(t1, win[ j], 1) + buf[4*( j)];
368  buf[4 * ( 9 + 8 - j )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 9 + 8 - j], 1);
369  buf[4 * ( j )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + j], 1);
370  i += 4;
371  }
372 
373  s0 = tmp[16];
374  s1 = MULH3(tmp[17], icos36h[4], 2);
375  t0 = s0 + s1;
376  t1 = s0 - s1;
377  out[(9 + 4) * SBLIMIT] = MULH3(t1, win[ 9 + 4], 1) + buf[4*(9 + 4)];
378  out[(8 - 4) * SBLIMIT] = MULH3(t1, win[ 8 - 4], 1) + buf[4*(8 - 4)];
379  buf[4 * ( 9 + 4 )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 9 + 4], 1);
380  buf[4 * ( 8 - 4 )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 8 - 4], 1);
381 }
382 
383 void RENAME(ff_imdct36_blocks)(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in,
384  int count, int switch_point, int block_type)
385 {
386  int j;
387  for (j=0 ; j < count; j++) {
388  /* apply window & overlap with previous buffer */
389 
390  /* select window */
391  int win_idx = (switch_point && j < 2) ? 0 : block_type;
392  INTFLOAT *win = RENAME(ff_mdct_win)[win_idx + (4 & -(j & 1))];
393 
394  imdct36(out, buf, in, win);
395 
396  in += 18;
397  buf += ((j&3) != 3 ? 1 : (72-3));
398  out++;
399  }
400 }
401 
SHR
#define SHR(a, b)
Definition: mpegaudiodsp_template.c:67
C7
#define C7
Definition: mpegaudiodsp_template.c:273
C2
#define C2
Definition: mpegaudiodsp_template.c:268
out
FILE * out
Definition: movenc.c:54
icos36h
static const INTFLOAT icos36h[9]
Definition: mpegaudiodsp_template.c:290
IMDCT_SCALAR
#define IMDCT_SCALAR
Definition: mpegaudio.h:56
MULH3
#define MULH3(x, y, s)
Definition: mpegaudiodsp_template.c:65
tmp
static uint8_t tmp[11]
Definition: aes_ctr.c:26
w
uint8_t w
Definition: llviddspenc.c:38
MPADSPContext
Definition: mpegaudiodsp.h:27
t0
#define t0
Definition: regdef.h:28
C8
#define C8
Definition: mpegaudiodsp_template.c:274
t1
#define t1
Definition: regdef.h:29
C5
#define C5
Definition: mpegaudiodsp_template.c:271
win
static float win(SuperEqualizerContext *s, float n, int N)
Definition: af_superequalizer.c:119
window
static SDL_Window * window
Definition: ffplay.c:368
s3
#define s3
Definition: regdef.h:40
SUM8P2
#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p)
Definition: mpegaudiodsp_template.c:92
C1
#define C1
Definition: mpegaudiodsp_template.c:267
MACS
#define MACS(rt, ra, rb)
Definition: mpegaudiodsp_template.c:63
SUM8
#define SUM8(op, sum, w, p)
Definition: mpegaudiodsp_template.c:80
av_cold
#define av_cold
Definition: attributes.h:90
s
#define s(width, name)
Definition: cbs_vp9.c:257
dct32
void dct32(INTFLOAT *out, const INTFLOAT *tab_arg)
Definition: dct32_template.c:126
s1
#define s1
Definition: regdef.h:38
C4
#define C4
Definition: mpegaudiodsp_template.c:270
mathops.h
dct32.h
SBLIMIT
#define SBLIMIT
Definition: mpegaudio.h:44
ff_mpa_enwindow
const int32_t ff_mpa_enwindow[257]
Definition: mpegaudiodsp_data.c:22
C3
#define C3
Definition: mpegaudiodsp_template.c:269
OUT_SHIFT
#define OUT_SHIFT
Definition: mpegaudiodsp_template.c:52
s2
#define s2
Definition: regdef.h:39
FIXHR
#define FIXHR(a)
Definition: mpegaudiodsp_template.c:69
MLSS
#define MLSS(rt, ra, rb)
Definition: mpegaudiodsp_template.c:64
OUT_INT
int16_t OUT_INT
Definition: mpegaudio.h:76
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
attributes.h
M_PI
#define M_PI
Definition: mathematics.h:52
DECLARE_ALIGNED
#define DECLARE_ALIGNED(n, t, v)
Definition: mem.h:112
in
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Definition: audio_convert.c:326
imdct36
static void imdct36(INTFLOAT *out, INTFLOAT *buf, SUINTFLOAT *in, INTFLOAT *win)
Definition: mpegaudiodsp_template.c:303
i
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
t3
#define t3
Definition: regdef.h:31
apply_window
static void(*const apply_window[4])(AVFloatDSPContext *fdsp, SingleChannelElement *sce, const float *audio)
Definition: aacenc.c:193
mpegaudio.h
FIXR
#define FIXR(a)
Definition: mpegaudiodsp_template.c:68
t2
#define t2
Definition: regdef.h:30
FRAC_BITS
#define FRAC_BITS
Definition: g729postfilter.c:33
RENAME
#define RENAME(n)
Definition: mpegaudiodsp_template.c:51
samples
Filter the word “frame” indicates either a video frame or a group of audio samples
Definition: filter_design.txt:8
round_sample
static int round_sample(int64_t *sum)
Definition: mpegaudiodsp_template.c:54
icos36
static const INTFLOAT icos36[9]
Definition: mpegaudiodsp_template.c:277
mpegaudiodsp.h
MPA_INT
int32_t MPA_INT
Definition: mpegaudio.h:75
mem.h
MDCT_BUF_SIZE
#define MDCT_BUF_SIZE
For SSE implementation, MDCT_BUF_SIZE/2 should be 128-bit aligned.
Definition: mpegaudiodsp.h:89
s0
#define s0
Definition: regdef.h:37
SUINTFLOAT
#define SUINTFLOAT
Definition: dct32_template.c:45
int
int
Definition: ffmpeg_filter.c:192
INTFLOAT
float INTFLOAT
Definition: aac_defines.h:86
MULLx
#define MULLx(x, y, s)
Definition: mpegaudiodsp_template.c:66
Generated on Wed Aug 24 2022 21:31:28 for FFmpeg by   doxygen 1.8.17