FFmpeg coverage

Line	Branch	Exec	Source
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, sbcdsp_analysis_consts_fixed4_simd_odd);
111		✗	out += out_stride;
112		✗	s->sbc_analyze_4(x + 8, out, sbcdsp_analysis_consts_fixed4_simd_even);
113		✗	out += out_stride;
114		✗	s->sbc_analyze_4(x + 4, out, sbcdsp_analysis_consts_fixed4_simd_odd);
115		✗	out += out_stride;
116		✗	s->sbc_analyze_4(x + 0, out, 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, sbcdsp_analysis_consts_fixed8_simd_odd);
124		✗	out += out_stride;
125		✗	s->sbc_analyze_8(x + 16, out, sbcdsp_analysis_consts_fixed8_simd_even);
126		✗	out += out_stride;
127		✗	s->sbc_analyze_8(x + 8, out, sbcdsp_analysis_consts_fixed8_simd_odd);
128		✗	out += out_stride;
129		✗	s->sbc_analyze_8(x + 0, out, 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, 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, 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			#elif ARCH_X86
386		✗	ff_sbcdsp_init_x86(s);
387			#endif
388		✗	}
389