FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/sbcdsp.c
Date: 2021-09-26 18:22:30
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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 }
388