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40e6575aa3
This is more spec-compliant because it does not rely on dead-code elimination by the compiler. Especially MSVC has problems with this, as can be seen in https://ffmpeg.org/pipermail/ffmpeg-devel/2022-May/296373.html or https://ffmpeg.org/pipermail/ffmpeg-devel/2022-May/297022.html This commit does not eliminate every instance where we rely on dead code elimination: It only tackles branching to the initialization of arch-specific dsp code, not e.g. all uses of CONFIG_ and HAVE_ checks. But maybe it is already enough to compile FFmpeg with MSVC with whole-programm-optimizations enabled (if one does not disable too many components). Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
389 lines
14 KiB
C
389 lines
14 KiB
C
/*
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* Bluetooth low-complexity, subband codec (SBC)
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*
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* Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
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* Copyright (C) 2012-2013 Intel Corporation
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* Copyright (C) 2008-2010 Nokia Corporation
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* Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
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* Copyright (C) 2004-2005 Henryk Ploetz <henryk@ploetzli.ch>
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* Copyright (C) 2005-2006 Brad Midgley <bmidgley@xmission.com>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* SBC basic "building bricks"
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*/
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#include <stdint.h>
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#include <limits.h>
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#include <string.h>
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#include "libavutil/common.h"
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#include "libavutil/intmath.h"
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#include "libavutil/intreadwrite.h"
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#include "sbc.h"
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#include "sbcdsp.h"
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#include "sbcdsp_data.h"
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/*
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* A reference C code of analysis filter with SIMD-friendly tables
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* reordering and code layout. This code can be used to develop platform
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* specific SIMD optimizations. Also it may be used as some kind of test
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* for compiler autovectorization capabilities (who knows, if the compiler
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* is very good at this stuff, hand optimized assembly may be not strictly
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* needed for some platform).
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*
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* Note: It is also possible to make a simple variant of analysis filter,
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* which needs only a single constants table without taking care about
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* even/odd cases. This simple variant of filter can be implemented without
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* input data permutation. The only thing that would be lost is the
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* possibility to use pairwise SIMD multiplications. But for some simple
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* CPU cores without SIMD extensions it can be useful. If anybody is
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* interested in implementing such variant of a filter, sourcecode from
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* bluez versions 4.26/4.27 can be used as a reference and the history of
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* the changes in git repository done around that time may be worth checking.
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*/
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static av_always_inline void sbc_analyze_simd(const int16_t *in, int32_t *out,
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const int16_t *consts,
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unsigned subbands)
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{
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int32_t t1[8];
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int16_t t2[8];
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int i, j, hop = 0;
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/* rounding coefficient */
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for (i = 0; i < subbands; i++)
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t1[i] = 1 << (SBC_PROTO_FIXED_SCALE - 1);
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/* low pass polyphase filter */
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for (hop = 0; hop < 10*subbands; hop += 2*subbands)
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for (i = 0; i < 2*subbands; i++)
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t1[i >> 1] += in[hop + i] * consts[hop + i];
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/* scaling */
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for (i = 0; i < subbands; i++)
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t2[i] = t1[i] >> SBC_PROTO_FIXED_SCALE;
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memset(t1, 0, sizeof(t1));
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/* do the cos transform */
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for (i = 0; i < subbands/2; i++)
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for (j = 0; j < 2*subbands; j++)
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t1[j>>1] += t2[i * 2 + (j&1)] * consts[10*subbands + i*2*subbands + j];
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for (i = 0; i < subbands; i++)
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out[i] = t1[i] >> (SBC_COS_TABLE_FIXED_SCALE - SCALE_OUT_BITS);
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}
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static void sbc_analyze_4_simd(const int16_t *in, int32_t *out,
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const int16_t *consts)
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{
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sbc_analyze_simd(in, out, consts, 4);
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}
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static void sbc_analyze_8_simd(const int16_t *in, int32_t *out,
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const int16_t *consts)
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{
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sbc_analyze_simd(in, out, consts, 8);
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}
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static inline void sbc_analyze_4b_4s_simd(SBCDSPContext *s,
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int16_t *x, int32_t *out, int out_stride)
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{
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/* Analyze blocks */
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s->sbc_analyze_4(x + 12, out, ff_sbcdsp_analysis_consts_fixed4_simd_odd);
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out += out_stride;
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s->sbc_analyze_4(x + 8, out, ff_sbcdsp_analysis_consts_fixed4_simd_even);
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out += out_stride;
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s->sbc_analyze_4(x + 4, out, ff_sbcdsp_analysis_consts_fixed4_simd_odd);
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out += out_stride;
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s->sbc_analyze_4(x + 0, out, ff_sbcdsp_analysis_consts_fixed4_simd_even);
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}
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static inline void sbc_analyze_4b_8s_simd(SBCDSPContext *s,
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int16_t *x, int32_t *out, int out_stride)
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{
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/* Analyze blocks */
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s->sbc_analyze_8(x + 24, out, ff_sbcdsp_analysis_consts_fixed8_simd_odd);
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out += out_stride;
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s->sbc_analyze_8(x + 16, out, ff_sbcdsp_analysis_consts_fixed8_simd_even);
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out += out_stride;
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s->sbc_analyze_8(x + 8, out, ff_sbcdsp_analysis_consts_fixed8_simd_odd);
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out += out_stride;
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s->sbc_analyze_8(x + 0, out, ff_sbcdsp_analysis_consts_fixed8_simd_even);
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}
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static inline void sbc_analyze_1b_8s_simd_even(SBCDSPContext *s,
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int16_t *x, int32_t *out,
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int out_stride);
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static inline void sbc_analyze_1b_8s_simd_odd(SBCDSPContext *s,
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int16_t *x, int32_t *out,
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int out_stride)
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{
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s->sbc_analyze_8(x, out, ff_sbcdsp_analysis_consts_fixed8_simd_odd);
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s->sbc_analyze_8s = sbc_analyze_1b_8s_simd_even;
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}
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static inline void sbc_analyze_1b_8s_simd_even(SBCDSPContext *s,
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int16_t *x, int32_t *out,
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int out_stride)
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{
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s->sbc_analyze_8(x, out, ff_sbcdsp_analysis_consts_fixed8_simd_even);
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s->sbc_analyze_8s = sbc_analyze_1b_8s_simd_odd;
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}
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/*
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* Input data processing functions. The data is endian converted if needed,
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* channels are deintrleaved and audio samples are reordered for use in
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* SIMD-friendly analysis filter function. The results are put into "X"
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* array, getting appended to the previous data (or it is better to say
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* prepended, as the buffer is filled from top to bottom). Old data is
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* discarded when neededed, but availability of (10 * nrof_subbands)
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* contiguous samples is always guaranteed for the input to the analysis
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* filter. This is achieved by copying a sufficient part of old data
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* to the top of the buffer on buffer wraparound.
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*/
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static int sbc_enc_process_input_4s(int position, const uint8_t *pcm,
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int16_t X[2][SBC_X_BUFFER_SIZE],
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int nsamples, int nchannels)
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{
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int c;
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/* handle X buffer wraparound */
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if (position < nsamples) {
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for (c = 0; c < nchannels; c++)
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memcpy(&X[c][SBC_X_BUFFER_SIZE - 40], &X[c][position],
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36 * sizeof(int16_t));
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position = SBC_X_BUFFER_SIZE - 40;
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}
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/* copy/permutate audio samples */
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for (; nsamples >= 8; nsamples -= 8, pcm += 16 * nchannels) {
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position -= 8;
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for (c = 0; c < nchannels; c++) {
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int16_t *x = &X[c][position];
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x[0] = AV_RN16(pcm + 14*nchannels + 2*c);
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x[1] = AV_RN16(pcm + 6*nchannels + 2*c);
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x[2] = AV_RN16(pcm + 12*nchannels + 2*c);
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x[3] = AV_RN16(pcm + 8*nchannels + 2*c);
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x[4] = AV_RN16(pcm + 0*nchannels + 2*c);
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x[5] = AV_RN16(pcm + 4*nchannels + 2*c);
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x[6] = AV_RN16(pcm + 2*nchannels + 2*c);
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x[7] = AV_RN16(pcm + 10*nchannels + 2*c);
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}
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}
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return position;
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}
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static int sbc_enc_process_input_8s(int position, const uint8_t *pcm,
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int16_t X[2][SBC_X_BUFFER_SIZE],
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int nsamples, int nchannels)
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{
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int c;
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/* handle X buffer wraparound */
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if (position < nsamples) {
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for (c = 0; c < nchannels; c++)
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memcpy(&X[c][SBC_X_BUFFER_SIZE - 72], &X[c][position],
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72 * sizeof(int16_t));
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position = SBC_X_BUFFER_SIZE - 72;
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}
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if (position % 16 == 8) {
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position -= 8;
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nsamples -= 8;
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for (c = 0; c < nchannels; c++) {
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int16_t *x = &X[c][position];
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x[0] = AV_RN16(pcm + 14*nchannels + 2*c);
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x[2] = AV_RN16(pcm + 12*nchannels + 2*c);
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x[3] = AV_RN16(pcm + 0*nchannels + 2*c);
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x[4] = AV_RN16(pcm + 10*nchannels + 2*c);
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x[5] = AV_RN16(pcm + 2*nchannels + 2*c);
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x[6] = AV_RN16(pcm + 8*nchannels + 2*c);
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x[7] = AV_RN16(pcm + 4*nchannels + 2*c);
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x[8] = AV_RN16(pcm + 6*nchannels + 2*c);
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}
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pcm += 16 * nchannels;
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}
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/* copy/permutate audio samples */
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for (; nsamples >= 16; nsamples -= 16, pcm += 32 * nchannels) {
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position -= 16;
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for (c = 0; c < nchannels; c++) {
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int16_t *x = &X[c][position];
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x[0] = AV_RN16(pcm + 30*nchannels + 2*c);
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x[1] = AV_RN16(pcm + 14*nchannels + 2*c);
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x[2] = AV_RN16(pcm + 28*nchannels + 2*c);
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x[3] = AV_RN16(pcm + 16*nchannels + 2*c);
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x[4] = AV_RN16(pcm + 26*nchannels + 2*c);
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x[5] = AV_RN16(pcm + 18*nchannels + 2*c);
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x[6] = AV_RN16(pcm + 24*nchannels + 2*c);
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x[7] = AV_RN16(pcm + 20*nchannels + 2*c);
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x[8] = AV_RN16(pcm + 22*nchannels + 2*c);
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x[9] = AV_RN16(pcm + 6*nchannels + 2*c);
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x[10] = AV_RN16(pcm + 12*nchannels + 2*c);
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x[11] = AV_RN16(pcm + 0*nchannels + 2*c);
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x[12] = AV_RN16(pcm + 10*nchannels + 2*c);
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x[13] = AV_RN16(pcm + 2*nchannels + 2*c);
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x[14] = AV_RN16(pcm + 8*nchannels + 2*c);
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x[15] = AV_RN16(pcm + 4*nchannels + 2*c);
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}
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}
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if (nsamples == 8) {
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position -= 8;
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for (c = 0; c < nchannels; c++) {
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int16_t *x = &X[c][position];
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x[-7] = AV_RN16(pcm + 14*nchannels + 2*c);
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x[1] = AV_RN16(pcm + 6*nchannels + 2*c);
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x[2] = AV_RN16(pcm + 12*nchannels + 2*c);
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x[3] = AV_RN16(pcm + 0*nchannels + 2*c);
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x[4] = AV_RN16(pcm + 10*nchannels + 2*c);
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x[5] = AV_RN16(pcm + 2*nchannels + 2*c);
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x[6] = AV_RN16(pcm + 8*nchannels + 2*c);
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x[7] = AV_RN16(pcm + 4*nchannels + 2*c);
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}
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}
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return position;
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}
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static void sbc_calc_scalefactors(int32_t sb_sample_f[16][2][8],
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uint32_t scale_factor[2][8],
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int blocks, int channels, int subbands)
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{
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int ch, sb, blk;
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for (ch = 0; ch < channels; ch++) {
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for (sb = 0; sb < subbands; sb++) {
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uint32_t x = 1 << SCALE_OUT_BITS;
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for (blk = 0; blk < blocks; blk++) {
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int32_t tmp = FFABS(sb_sample_f[blk][ch][sb]);
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if (tmp != 0)
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x |= tmp - 1;
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}
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scale_factor[ch][sb] = (31 - SCALE_OUT_BITS) - ff_clz(x);
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}
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}
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}
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static int sbc_calc_scalefactors_j(int32_t sb_sample_f[16][2][8],
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uint32_t scale_factor[2][8],
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int blocks, int subbands)
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{
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int blk, joint = 0;
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int32_t tmp0, tmp1;
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uint32_t x, y;
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/* last subband does not use joint stereo */
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int sb = subbands - 1;
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x = 1 << SCALE_OUT_BITS;
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y = 1 << SCALE_OUT_BITS;
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for (blk = 0; blk < blocks; blk++) {
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tmp0 = FFABS(sb_sample_f[blk][0][sb]);
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tmp1 = FFABS(sb_sample_f[blk][1][sb]);
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if (tmp0 != 0)
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x |= tmp0 - 1;
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if (tmp1 != 0)
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y |= tmp1 - 1;
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}
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scale_factor[0][sb] = (31 - SCALE_OUT_BITS) - ff_clz(x);
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scale_factor[1][sb] = (31 - SCALE_OUT_BITS) - ff_clz(y);
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/* the rest of subbands can use joint stereo */
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while (--sb >= 0) {
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int32_t sb_sample_j[16][2];
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x = 1 << SCALE_OUT_BITS;
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y = 1 << SCALE_OUT_BITS;
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for (blk = 0; blk < blocks; blk++) {
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tmp0 = sb_sample_f[blk][0][sb];
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tmp1 = sb_sample_f[blk][1][sb];
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sb_sample_j[blk][0] = (tmp0 >> 1) + (tmp1 >> 1);
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sb_sample_j[blk][1] = (tmp0 >> 1) - (tmp1 >> 1);
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tmp0 = FFABS(tmp0);
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tmp1 = FFABS(tmp1);
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if (tmp0 != 0)
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x |= tmp0 - 1;
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if (tmp1 != 0)
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y |= tmp1 - 1;
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}
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scale_factor[0][sb] = (31 - SCALE_OUT_BITS) -
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ff_clz(x);
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scale_factor[1][sb] = (31 - SCALE_OUT_BITS) -
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ff_clz(y);
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x = 1 << SCALE_OUT_BITS;
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y = 1 << SCALE_OUT_BITS;
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for (blk = 0; blk < blocks; blk++) {
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tmp0 = FFABS(sb_sample_j[blk][0]);
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tmp1 = FFABS(sb_sample_j[blk][1]);
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if (tmp0 != 0)
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x |= tmp0 - 1;
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if (tmp1 != 0)
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y |= tmp1 - 1;
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}
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x = (31 - SCALE_OUT_BITS) - ff_clz(x);
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y = (31 - SCALE_OUT_BITS) - ff_clz(y);
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/* decide whether to use joint stereo for this subband */
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if ((scale_factor[0][sb] + scale_factor[1][sb]) > x + y) {
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joint |= 1 << (subbands - 1 - sb);
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scale_factor[0][sb] = x;
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scale_factor[1][sb] = y;
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for (blk = 0; blk < blocks; blk++) {
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sb_sample_f[blk][0][sb] = sb_sample_j[blk][0];
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sb_sample_f[blk][1][sb] = sb_sample_j[blk][1];
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}
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}
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}
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/* bitmask with the information about subbands using joint stereo */
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return joint;
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}
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/*
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* Detect CPU features and setup function pointers
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*/
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av_cold void ff_sbcdsp_init(SBCDSPContext *s)
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{
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/* Default implementation for analyze functions */
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s->sbc_analyze_4 = sbc_analyze_4_simd;
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s->sbc_analyze_8 = sbc_analyze_8_simd;
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s->sbc_analyze_4s = sbc_analyze_4b_4s_simd;
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if (s->increment == 1)
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s->sbc_analyze_8s = sbc_analyze_1b_8s_simd_odd;
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else
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s->sbc_analyze_8s = sbc_analyze_4b_8s_simd;
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/* Default implementation for input reordering / deinterleaving */
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s->sbc_enc_process_input_4s = sbc_enc_process_input_4s;
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s->sbc_enc_process_input_8s = sbc_enc_process_input_8s;
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/* Default implementation for scale factors calculation */
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s->sbc_calc_scalefactors = sbc_calc_scalefactors;
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s->sbc_calc_scalefactors_j = sbc_calc_scalefactors_j;
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#if ARCH_ARM
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ff_sbcdsp_init_arm(s);
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#elif ARCH_X86
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ff_sbcdsp_init_x86(s);
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#endif
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}
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