FFmpeg
sbcdsp.c
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1 /*
2  * Bluetooth low-complexity, subband codec (SBC)
3  *
4  * Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
5  * Copyright (C) 2012-2013 Intel Corporation
6  * Copyright (C) 2008-2010 Nokia Corporation
7  * Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
8  * Copyright (C) 2004-2005 Henryk Ploetz <henryk@ploetzli.ch>
9  * Copyright (C) 2005-2006 Brad Midgley <bmidgley@xmission.com>
10  *
11  * This file is part of FFmpeg.
12  *
13  * FFmpeg is free software; you can redistribute it and/or
14  * modify it under the terms of the GNU Lesser General Public
15  * License as published by the Free Software Foundation; either
16  * version 2.1 of the License, or (at your option) any later version.
17  *
18  * FFmpeg is distributed in the hope that it will be useful,
19  * but WITHOUT ANY WARRANTY; without even the implied warranty of
20  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
21  * Lesser General Public License for more details.
22  *
23  * You should have received a copy of the GNU Lesser General Public
24  * License along with FFmpeg; if not, write to the Free Software
25  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
26  */
27 
28 /**
29  * @file
30  * SBC basic "building bricks"
31  */
32 
33 #include <stdint.h>
34 #include <limits.h>
35 #include <string.h>
36 #include "libavutil/common.h"
37 #include "libavutil/intmath.h"
38 #include "libavutil/intreadwrite.h"
39 #include "sbc.h"
40 #include "sbcdsp.h"
41 #include "sbcdsp_data.h"
42 
43 /*
44  * A reference C code of analysis filter with SIMD-friendly tables
45  * reordering and code layout. This code can be used to develop platform
46  * specific SIMD optimizations. Also it may be used as some kind of test
47  * for compiler autovectorization capabilities (who knows, if the compiler
48  * is very good at this stuff, hand optimized assembly may be not strictly
49  * needed for some platform).
50  *
51  * Note: It is also possible to make a simple variant of analysis filter,
52  * which needs only a single constants table without taking care about
53  * even/odd cases. This simple variant of filter can be implemented without
54  * input data permutation. The only thing that would be lost is the
55  * possibility to use pairwise SIMD multiplications. But for some simple
56  * CPU cores without SIMD extensions it can be useful. If anybody is
57  * interested in implementing such variant of a filter, sourcecode from
58  * bluez versions 4.26/4.27 can be used as a reference and the history of
59  * the changes in git repository done around that time may be worth checking.
60  */
61 
62 static av_always_inline void sbc_analyze_simd(const int16_t *in, int32_t *out,
63  const int16_t *consts,
64  unsigned subbands)
65 {
66  int32_t t1[8];
67  int16_t t2[8];
68  int i, j, hop = 0;
69 
70  /* rounding coefficient */
71  for (i = 0; i < subbands; i++)
72  t1[i] = 1 << (SBC_PROTO_FIXED_SCALE - 1);
73 
74  /* low pass polyphase filter */
75  for (hop = 0; hop < 10*subbands; hop += 2*subbands)
76  for (i = 0; i < 2*subbands; i++)
77  t1[i >> 1] += in[hop + i] * consts[hop + i];
78 
79  /* scaling */
80  for (i = 0; i < subbands; i++)
81  t2[i] = t1[i] >> SBC_PROTO_FIXED_SCALE;
82 
83  memset(t1, 0, sizeof(t1));
84 
85  /* do the cos transform */
86  for (i = 0; i < subbands/2; i++)
87  for (j = 0; j < 2*subbands; j++)
88  t1[j>>1] += t2[i * 2 + (j&1)] * consts[10*subbands + i*2*subbands + j];
89 
90  for (i = 0; i < subbands; i++)
91  out[i] = t1[i] >> (SBC_COS_TABLE_FIXED_SCALE - SCALE_OUT_BITS);
92 }
93 
94 static void sbc_analyze_4_simd(const int16_t *in, int32_t *out,
95  const int16_t *consts)
96 {
97  sbc_analyze_simd(in, out, consts, 4);
98 }
99 
100 static void sbc_analyze_8_simd(const int16_t *in, int32_t *out,
101  const int16_t *consts)
102 {
103  sbc_analyze_simd(in, out, consts, 8);
104 }
105 
106 static inline void sbc_analyze_4b_4s_simd(SBCDSPContext *s,
107  int16_t *x, int32_t *out, int out_stride)
108 {
109  /* Analyze blocks */
110  s->sbc_analyze_4(x + 12, out, ff_sbcdsp_analysis_consts_fixed4_simd_odd);
111  out += out_stride;
112  s->sbc_analyze_4(x + 8, out, ff_sbcdsp_analysis_consts_fixed4_simd_even);
113  out += out_stride;
114  s->sbc_analyze_4(x + 4, out, ff_sbcdsp_analysis_consts_fixed4_simd_odd);
115  out += out_stride;
116  s->sbc_analyze_4(x + 0, out, ff_sbcdsp_analysis_consts_fixed4_simd_even);
117 }
118 
119 static inline void sbc_analyze_4b_8s_simd(SBCDSPContext *s,
120  int16_t *x, int32_t *out, int out_stride)
121 {
122  /* Analyze blocks */
123  s->sbc_analyze_8(x + 24, out, ff_sbcdsp_analysis_consts_fixed8_simd_odd);
124  out += out_stride;
125  s->sbc_analyze_8(x + 16, out, ff_sbcdsp_analysis_consts_fixed8_simd_even);
126  out += out_stride;
127  s->sbc_analyze_8(x + 8, out, ff_sbcdsp_analysis_consts_fixed8_simd_odd);
128  out += out_stride;
129  s->sbc_analyze_8(x + 0, out, ff_sbcdsp_analysis_consts_fixed8_simd_even);
130 }
131 
132 static inline void sbc_analyze_1b_8s_simd_even(SBCDSPContext *s,
133  int16_t *x, int32_t *out,
134  int out_stride);
135 
136 static inline void sbc_analyze_1b_8s_simd_odd(SBCDSPContext *s,
137  int16_t *x, int32_t *out,
138  int out_stride)
139 {
140  s->sbc_analyze_8(x, out, ff_sbcdsp_analysis_consts_fixed8_simd_odd);
141  s->sbc_analyze_8s = sbc_analyze_1b_8s_simd_even;
142 }
143 
144 static inline void sbc_analyze_1b_8s_simd_even(SBCDSPContext *s,
145  int16_t *x, int32_t *out,
146  int out_stride)
147 {
148  s->sbc_analyze_8(x, out, ff_sbcdsp_analysis_consts_fixed8_simd_even);
149  s->sbc_analyze_8s = sbc_analyze_1b_8s_simd_odd;
150 }
151 
152 /*
153  * Input data processing functions. The data is endian converted if needed,
154  * channels are deintrleaved and audio samples are reordered for use in
155  * SIMD-friendly analysis filter function. The results are put into "X"
156  * array, getting appended to the previous data (or it is better to say
157  * prepended, as the buffer is filled from top to bottom). Old data is
158  * discarded when neededed, but availability of (10 * nrof_subbands)
159  * contiguous samples is always guaranteed for the input to the analysis
160  * filter. This is achieved by copying a sufficient part of old data
161  * to the top of the buffer on buffer wraparound.
162  */
163 
164 static int sbc_enc_process_input_4s(int position, const uint8_t *pcm,
165  int16_t X[2][SBC_X_BUFFER_SIZE],
166  int nsamples, int nchannels)
167 {
168  int c;
169 
170  /* handle X buffer wraparound */
171  if (position < nsamples) {
172  for (c = 0; c < nchannels; c++)
173  memcpy(&X[c][SBC_X_BUFFER_SIZE - 40], &X[c][position],
174  36 * sizeof(int16_t));
175  position = SBC_X_BUFFER_SIZE - 40;
176  }
177 
178  /* copy/permutate audio samples */
179  for (; nsamples >= 8; nsamples -= 8, pcm += 16 * nchannels) {
180  position -= 8;
181  for (c = 0; c < nchannels; c++) {
182  int16_t *x = &X[c][position];
183  x[0] = AV_RN16(pcm + 14*nchannels + 2*c);
184  x[1] = AV_RN16(pcm + 6*nchannels + 2*c);
185  x[2] = AV_RN16(pcm + 12*nchannels + 2*c);
186  x[3] = AV_RN16(pcm + 8*nchannels + 2*c);
187  x[4] = AV_RN16(pcm + 0*nchannels + 2*c);
188  x[5] = AV_RN16(pcm + 4*nchannels + 2*c);
189  x[6] = AV_RN16(pcm + 2*nchannels + 2*c);
190  x[7] = AV_RN16(pcm + 10*nchannels + 2*c);
191  }
192  }
193 
194  return position;
195 }
196 
197 static int sbc_enc_process_input_8s(int position, const uint8_t *pcm,
198  int16_t X[2][SBC_X_BUFFER_SIZE],
199  int nsamples, int nchannels)
200 {
201  int c;
202 
203  /* handle X buffer wraparound */
204  if (position < nsamples) {
205  for (c = 0; c < nchannels; c++)
206  memcpy(&X[c][SBC_X_BUFFER_SIZE - 72], &X[c][position],
207  72 * sizeof(int16_t));
208  position = SBC_X_BUFFER_SIZE - 72;
209  }
210 
211  if (position % 16 == 8) {
212  position -= 8;
213  nsamples -= 8;
214  for (c = 0; c < nchannels; c++) {
215  int16_t *x = &X[c][position];
216  x[0] = AV_RN16(pcm + 14*nchannels + 2*c);
217  x[2] = AV_RN16(pcm + 12*nchannels + 2*c);
218  x[3] = AV_RN16(pcm + 0*nchannels + 2*c);
219  x[4] = AV_RN16(pcm + 10*nchannels + 2*c);
220  x[5] = AV_RN16(pcm + 2*nchannels + 2*c);
221  x[6] = AV_RN16(pcm + 8*nchannels + 2*c);
222  x[7] = AV_RN16(pcm + 4*nchannels + 2*c);
223  x[8] = AV_RN16(pcm + 6*nchannels + 2*c);
224  }
225  pcm += 16 * nchannels;
226  }
227 
228  /* copy/permutate audio samples */
229  for (; nsamples >= 16; nsamples -= 16, pcm += 32 * nchannels) {
230  position -= 16;
231  for (c = 0; c < nchannels; c++) {
232  int16_t *x = &X[c][position];
233  x[0] = AV_RN16(pcm + 30*nchannels + 2*c);
234  x[1] = AV_RN16(pcm + 14*nchannels + 2*c);
235  x[2] = AV_RN16(pcm + 28*nchannels + 2*c);
236  x[3] = AV_RN16(pcm + 16*nchannels + 2*c);
237  x[4] = AV_RN16(pcm + 26*nchannels + 2*c);
238  x[5] = AV_RN16(pcm + 18*nchannels + 2*c);
239  x[6] = AV_RN16(pcm + 24*nchannels + 2*c);
240  x[7] = AV_RN16(pcm + 20*nchannels + 2*c);
241  x[8] = AV_RN16(pcm + 22*nchannels + 2*c);
242  x[9] = AV_RN16(pcm + 6*nchannels + 2*c);
243  x[10] = AV_RN16(pcm + 12*nchannels + 2*c);
244  x[11] = AV_RN16(pcm + 0*nchannels + 2*c);
245  x[12] = AV_RN16(pcm + 10*nchannels + 2*c);
246  x[13] = AV_RN16(pcm + 2*nchannels + 2*c);
247  x[14] = AV_RN16(pcm + 8*nchannels + 2*c);
248  x[15] = AV_RN16(pcm + 4*nchannels + 2*c);
249  }
250  }
251 
252  if (nsamples == 8) {
253  position -= 8;
254  for (c = 0; c < nchannels; c++) {
255  int16_t *x = &X[c][position];
256  x[-7] = AV_RN16(pcm + 14*nchannels + 2*c);
257  x[1] = AV_RN16(pcm + 6*nchannels + 2*c);
258  x[2] = AV_RN16(pcm + 12*nchannels + 2*c);
259  x[3] = AV_RN16(pcm + 0*nchannels + 2*c);
260  x[4] = AV_RN16(pcm + 10*nchannels + 2*c);
261  x[5] = AV_RN16(pcm + 2*nchannels + 2*c);
262  x[6] = AV_RN16(pcm + 8*nchannels + 2*c);
263  x[7] = AV_RN16(pcm + 4*nchannels + 2*c);
264  }
265  }
266 
267  return position;
268 }
269 
270 static void sbc_calc_scalefactors(int32_t sb_sample_f[16][2][8],
271  uint32_t scale_factor[2][8],
272  int blocks, int channels, int subbands)
273 {
274  int ch, sb, blk;
275  for (ch = 0; ch < channels; ch++) {
276  for (sb = 0; sb < subbands; sb++) {
277  uint32_t x = 1 << SCALE_OUT_BITS;
278  for (blk = 0; blk < blocks; blk++) {
279  int32_t tmp = FFABS(sb_sample_f[blk][ch][sb]);
280  if (tmp != 0)
281  x |= tmp - 1;
282  }
283  scale_factor[ch][sb] = (31 - SCALE_OUT_BITS) - ff_clz(x);
284  }
285  }
286 }
287 
288 static int sbc_calc_scalefactors_j(int32_t sb_sample_f[16][2][8],
289  uint32_t scale_factor[2][8],
290  int blocks, int subbands)
291 {
292  int blk, joint = 0;
293  int32_t tmp0, tmp1;
294  uint32_t x, y;
295 
296  /* last subband does not use joint stereo */
297  int sb = subbands - 1;
298  x = 1 << SCALE_OUT_BITS;
299  y = 1 << SCALE_OUT_BITS;
300  for (blk = 0; blk < blocks; blk++) {
301  tmp0 = FFABS(sb_sample_f[blk][0][sb]);
302  tmp1 = FFABS(sb_sample_f[blk][1][sb]);
303  if (tmp0 != 0)
304  x |= tmp0 - 1;
305  if (tmp1 != 0)
306  y |= tmp1 - 1;
307  }
308  scale_factor[0][sb] = (31 - SCALE_OUT_BITS) - ff_clz(x);
309  scale_factor[1][sb] = (31 - SCALE_OUT_BITS) - ff_clz(y);
310 
311  /* the rest of subbands can use joint stereo */
312  while (--sb >= 0) {
313  int32_t sb_sample_j[16][2];
314  x = 1 << SCALE_OUT_BITS;
315  y = 1 << SCALE_OUT_BITS;
316  for (blk = 0; blk < blocks; blk++) {
317  tmp0 = sb_sample_f[blk][0][sb];
318  tmp1 = sb_sample_f[blk][1][sb];
319  sb_sample_j[blk][0] = (tmp0 >> 1) + (tmp1 >> 1);
320  sb_sample_j[blk][1] = (tmp0 >> 1) - (tmp1 >> 1);
321  tmp0 = FFABS(tmp0);
322  tmp1 = FFABS(tmp1);
323  if (tmp0 != 0)
324  x |= tmp0 - 1;
325  if (tmp1 != 0)
326  y |= tmp1 - 1;
327  }
328  scale_factor[0][sb] = (31 - SCALE_OUT_BITS) -
329  ff_clz(x);
330  scale_factor[1][sb] = (31 - SCALE_OUT_BITS) -
331  ff_clz(y);
332  x = 1 << SCALE_OUT_BITS;
333  y = 1 << SCALE_OUT_BITS;
334  for (blk = 0; blk < blocks; blk++) {
335  tmp0 = FFABS(sb_sample_j[blk][0]);
336  tmp1 = FFABS(sb_sample_j[blk][1]);
337  if (tmp0 != 0)
338  x |= tmp0 - 1;
339  if (tmp1 != 0)
340  y |= tmp1 - 1;
341  }
342  x = (31 - SCALE_OUT_BITS) - ff_clz(x);
343  y = (31 - SCALE_OUT_BITS) - ff_clz(y);
344 
345  /* decide whether to use joint stereo for this subband */
346  if ((scale_factor[0][sb] + scale_factor[1][sb]) > x + y) {
347  joint |= 1 << (subbands - 1 - sb);
348  scale_factor[0][sb] = x;
349  scale_factor[1][sb] = y;
350  for (blk = 0; blk < blocks; blk++) {
351  sb_sample_f[blk][0][sb] = sb_sample_j[blk][0];
352  sb_sample_f[blk][1][sb] = sb_sample_j[blk][1];
353  }
354  }
355  }
356 
357  /* bitmask with the information about subbands using joint stereo */
358  return joint;
359 }
360 
361 /*
362  * Detect CPU features and setup function pointers
363  */
364 av_cold void ff_sbcdsp_init(SBCDSPContext *s)
365 {
366  /* Default implementation for analyze functions */
367  s->sbc_analyze_4 = sbc_analyze_4_simd;
368  s->sbc_analyze_8 = sbc_analyze_8_simd;
369  s->sbc_analyze_4s = sbc_analyze_4b_4s_simd;
370  if (s->increment == 1)
371  s->sbc_analyze_8s = sbc_analyze_1b_8s_simd_odd;
372  else
373  s->sbc_analyze_8s = sbc_analyze_4b_8s_simd;
374 
375  /* Default implementation for input reordering / deinterleaving */
376  s->sbc_enc_process_input_4s = sbc_enc_process_input_4s;
377  s->sbc_enc_process_input_8s = sbc_enc_process_input_8s;
378 
379  /* Default implementation for scale factors calculation */
380  s->sbc_calc_scalefactors = sbc_calc_scalefactors;
381  s->sbc_calc_scalefactors_j = sbc_calc_scalefactors_j;
382 
383  if (ARCH_ARM)
384  ff_sbcdsp_init_arm(s);
385  if (ARCH_X86)
386  ff_sbcdsp_init_x86(s);
387 }
out
FILE * out
Definition: movenc.c:54
sbc_enc_process_input_8s
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Definition: sbcdsp.c:197
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Definition: sbcdsp_data.c:139
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Definition: sbcdsp.c:100
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Definition: sbcdsp.c:164
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