mirror of
https://github.com/pineappleEA/pineapple-src.git
synced 2024-12-10 21:18:25 -05:00
388 lines
14 KiB
C
Executable File
388 lines
14 KiB
C
Executable File
/*
|
|
* Bluetooth low-complexity, subband codec (SBC)
|
|
*
|
|
* Copyright (C) 2017 Aurelien Jacobs <aurel@gnuage.org>
|
|
* Copyright (C) 2012-2013 Intel Corporation
|
|
* Copyright (C) 2008-2010 Nokia Corporation
|
|
* Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
|
|
* Copyright (C) 2004-2005 Henryk Ploetz <henryk@ploetzli.ch>
|
|
* Copyright (C) 2005-2006 Brad Midgley <bmidgley@xmission.com>
|
|
*
|
|
* This file is part of FFmpeg.
|
|
*
|
|
* FFmpeg is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 of the License, or (at your option) any later version.
|
|
*
|
|
* FFmpeg is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser General Public
|
|
* License along with FFmpeg; if not, write to the Free Software
|
|
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
|
|
*/
|
|
|
|
/**
|
|
* @file
|
|
* SBC basic "building bricks"
|
|
*/
|
|
|
|
#include <stdint.h>
|
|
#include <limits.h>
|
|
#include <string.h>
|
|
#include "libavutil/common.h"
|
|
#include "libavutil/intmath.h"
|
|
#include "libavutil/intreadwrite.h"
|
|
#include "sbc.h"
|
|
#include "sbcdsp.h"
|
|
#include "sbcdsp_data.h"
|
|
|
|
/*
|
|
* A reference C code of analysis filter with SIMD-friendly tables
|
|
* reordering and code layout. This code can be used to develop platform
|
|
* specific SIMD optimizations. Also it may be used as some kind of test
|
|
* for compiler autovectorization capabilities (who knows, if the compiler
|
|
* is very good at this stuff, hand optimized assembly may be not strictly
|
|
* needed for some platform).
|
|
*
|
|
* Note: It is also possible to make a simple variant of analysis filter,
|
|
* which needs only a single constants table without taking care about
|
|
* even/odd cases. This simple variant of filter can be implemented without
|
|
* input data permutation. The only thing that would be lost is the
|
|
* possibility to use pairwise SIMD multiplications. But for some simple
|
|
* CPU cores without SIMD extensions it can be useful. If anybody is
|
|
* interested in implementing such variant of a filter, sourcecode from
|
|
* bluez versions 4.26/4.27 can be used as a reference and the history of
|
|
* the changes in git repository done around that time may be worth checking.
|
|
*/
|
|
|
|
static av_always_inline void sbc_analyze_simd(const int16_t *in, int32_t *out,
|
|
const int16_t *consts,
|
|
unsigned subbands)
|
|
{
|
|
int32_t t1[8];
|
|
int16_t t2[8];
|
|
int i, j, hop = 0;
|
|
|
|
/* rounding coefficient */
|
|
for (i = 0; i < subbands; i++)
|
|
t1[i] = 1 << (SBC_PROTO_FIXED_SCALE - 1);
|
|
|
|
/* low pass polyphase filter */
|
|
for (hop = 0; hop < 10*subbands; hop += 2*subbands)
|
|
for (i = 0; i < 2*subbands; i++)
|
|
t1[i >> 1] += in[hop + i] * consts[hop + i];
|
|
|
|
/* scaling */
|
|
for (i = 0; i < subbands; i++)
|
|
t2[i] = t1[i] >> SBC_PROTO_FIXED_SCALE;
|
|
|
|
memset(t1, 0, sizeof(t1));
|
|
|
|
/* do the cos transform */
|
|
for (i = 0; i < subbands/2; i++)
|
|
for (j = 0; j < 2*subbands; j++)
|
|
t1[j>>1] += t2[i * 2 + (j&1)] * consts[10*subbands + i*2*subbands + j];
|
|
|
|
for (i = 0; i < subbands; i++)
|
|
out[i] = t1[i] >> (SBC_COS_TABLE_FIXED_SCALE - SCALE_OUT_BITS);
|
|
}
|
|
|
|
static void sbc_analyze_4_simd(const int16_t *in, int32_t *out,
|
|
const int16_t *consts)
|
|
{
|
|
sbc_analyze_simd(in, out, consts, 4);
|
|
}
|
|
|
|
static void sbc_analyze_8_simd(const int16_t *in, int32_t *out,
|
|
const int16_t *consts)
|
|
{
|
|
sbc_analyze_simd(in, out, consts, 8);
|
|
}
|
|
|
|
static inline void sbc_analyze_4b_4s_simd(SBCDSPContext *s,
|
|
int16_t *x, int32_t *out, int out_stride)
|
|
{
|
|
/* Analyze blocks */
|
|
s->sbc_analyze_4(x + 12, out, ff_sbcdsp_analysis_consts_fixed4_simd_odd);
|
|
out += out_stride;
|
|
s->sbc_analyze_4(x + 8, out, ff_sbcdsp_analysis_consts_fixed4_simd_even);
|
|
out += out_stride;
|
|
s->sbc_analyze_4(x + 4, out, ff_sbcdsp_analysis_consts_fixed4_simd_odd);
|
|
out += out_stride;
|
|
s->sbc_analyze_4(x + 0, out, ff_sbcdsp_analysis_consts_fixed4_simd_even);
|
|
}
|
|
|
|
static inline void sbc_analyze_4b_8s_simd(SBCDSPContext *s,
|
|
int16_t *x, int32_t *out, int out_stride)
|
|
{
|
|
/* Analyze blocks */
|
|
s->sbc_analyze_8(x + 24, out, ff_sbcdsp_analysis_consts_fixed8_simd_odd);
|
|
out += out_stride;
|
|
s->sbc_analyze_8(x + 16, out, ff_sbcdsp_analysis_consts_fixed8_simd_even);
|
|
out += out_stride;
|
|
s->sbc_analyze_8(x + 8, out, ff_sbcdsp_analysis_consts_fixed8_simd_odd);
|
|
out += out_stride;
|
|
s->sbc_analyze_8(x + 0, out, ff_sbcdsp_analysis_consts_fixed8_simd_even);
|
|
}
|
|
|
|
static inline void sbc_analyze_1b_8s_simd_even(SBCDSPContext *s,
|
|
int16_t *x, int32_t *out,
|
|
int out_stride);
|
|
|
|
static inline void sbc_analyze_1b_8s_simd_odd(SBCDSPContext *s,
|
|
int16_t *x, int32_t *out,
|
|
int out_stride)
|
|
{
|
|
s->sbc_analyze_8(x, out, ff_sbcdsp_analysis_consts_fixed8_simd_odd);
|
|
s->sbc_analyze_8s = sbc_analyze_1b_8s_simd_even;
|
|
}
|
|
|
|
static inline void sbc_analyze_1b_8s_simd_even(SBCDSPContext *s,
|
|
int16_t *x, int32_t *out,
|
|
int out_stride)
|
|
{
|
|
s->sbc_analyze_8(x, out, ff_sbcdsp_analysis_consts_fixed8_simd_even);
|
|
s->sbc_analyze_8s = sbc_analyze_1b_8s_simd_odd;
|
|
}
|
|
|
|
/*
|
|
* Input data processing functions. The data is endian converted if needed,
|
|
* channels are deintrleaved and audio samples are reordered for use in
|
|
* SIMD-friendly analysis filter function. The results are put into "X"
|
|
* array, getting appended to the previous data (or it is better to say
|
|
* prepended, as the buffer is filled from top to bottom). Old data is
|
|
* discarded when neededed, but availability of (10 * nrof_subbands)
|
|
* contiguous samples is always guaranteed for the input to the analysis
|
|
* filter. This is achieved by copying a sufficient part of old data
|
|
* to the top of the buffer on buffer wraparound.
|
|
*/
|
|
|
|
static int sbc_enc_process_input_4s(int position, const uint8_t *pcm,
|
|
int16_t X[2][SBC_X_BUFFER_SIZE],
|
|
int nsamples, int nchannels)
|
|
{
|
|
int c;
|
|
|
|
/* handle X buffer wraparound */
|
|
if (position < nsamples) {
|
|
for (c = 0; c < nchannels; c++)
|
|
memcpy(&X[c][SBC_X_BUFFER_SIZE - 40], &X[c][position],
|
|
36 * sizeof(int16_t));
|
|
position = SBC_X_BUFFER_SIZE - 40;
|
|
}
|
|
|
|
/* copy/permutate audio samples */
|
|
for (; nsamples >= 8; nsamples -= 8, pcm += 16 * nchannels) {
|
|
position -= 8;
|
|
for (c = 0; c < nchannels; c++) {
|
|
int16_t *x = &X[c][position];
|
|
x[0] = AV_RN16(pcm + 14*nchannels + 2*c);
|
|
x[1] = AV_RN16(pcm + 6*nchannels + 2*c);
|
|
x[2] = AV_RN16(pcm + 12*nchannels + 2*c);
|
|
x[3] = AV_RN16(pcm + 8*nchannels + 2*c);
|
|
x[4] = AV_RN16(pcm + 0*nchannels + 2*c);
|
|
x[5] = AV_RN16(pcm + 4*nchannels + 2*c);
|
|
x[6] = AV_RN16(pcm + 2*nchannels + 2*c);
|
|
x[7] = AV_RN16(pcm + 10*nchannels + 2*c);
|
|
}
|
|
}
|
|
|
|
return position;
|
|
}
|
|
|
|
static int sbc_enc_process_input_8s(int position, const uint8_t *pcm,
|
|
int16_t X[2][SBC_X_BUFFER_SIZE],
|
|
int nsamples, int nchannels)
|
|
{
|
|
int c;
|
|
|
|
/* handle X buffer wraparound */
|
|
if (position < nsamples) {
|
|
for (c = 0; c < nchannels; c++)
|
|
memcpy(&X[c][SBC_X_BUFFER_SIZE - 72], &X[c][position],
|
|
72 * sizeof(int16_t));
|
|
position = SBC_X_BUFFER_SIZE - 72;
|
|
}
|
|
|
|
if (position % 16 == 8) {
|
|
position -= 8;
|
|
nsamples -= 8;
|
|
for (c = 0; c < nchannels; c++) {
|
|
int16_t *x = &X[c][position];
|
|
x[0] = AV_RN16(pcm + 14*nchannels + 2*c);
|
|
x[2] = AV_RN16(pcm + 12*nchannels + 2*c);
|
|
x[3] = AV_RN16(pcm + 0*nchannels + 2*c);
|
|
x[4] = AV_RN16(pcm + 10*nchannels + 2*c);
|
|
x[5] = AV_RN16(pcm + 2*nchannels + 2*c);
|
|
x[6] = AV_RN16(pcm + 8*nchannels + 2*c);
|
|
x[7] = AV_RN16(pcm + 4*nchannels + 2*c);
|
|
x[8] = AV_RN16(pcm + 6*nchannels + 2*c);
|
|
}
|
|
pcm += 16 * nchannels;
|
|
}
|
|
|
|
/* copy/permutate audio samples */
|
|
for (; nsamples >= 16; nsamples -= 16, pcm += 32 * nchannels) {
|
|
position -= 16;
|
|
for (c = 0; c < nchannels; c++) {
|
|
int16_t *x = &X[c][position];
|
|
x[0] = AV_RN16(pcm + 30*nchannels + 2*c);
|
|
x[1] = AV_RN16(pcm + 14*nchannels + 2*c);
|
|
x[2] = AV_RN16(pcm + 28*nchannels + 2*c);
|
|
x[3] = AV_RN16(pcm + 16*nchannels + 2*c);
|
|
x[4] = AV_RN16(pcm + 26*nchannels + 2*c);
|
|
x[5] = AV_RN16(pcm + 18*nchannels + 2*c);
|
|
x[6] = AV_RN16(pcm + 24*nchannels + 2*c);
|
|
x[7] = AV_RN16(pcm + 20*nchannels + 2*c);
|
|
x[8] = AV_RN16(pcm + 22*nchannels + 2*c);
|
|
x[9] = AV_RN16(pcm + 6*nchannels + 2*c);
|
|
x[10] = AV_RN16(pcm + 12*nchannels + 2*c);
|
|
x[11] = AV_RN16(pcm + 0*nchannels + 2*c);
|
|
x[12] = AV_RN16(pcm + 10*nchannels + 2*c);
|
|
x[13] = AV_RN16(pcm + 2*nchannels + 2*c);
|
|
x[14] = AV_RN16(pcm + 8*nchannels + 2*c);
|
|
x[15] = AV_RN16(pcm + 4*nchannels + 2*c);
|
|
}
|
|
}
|
|
|
|
if (nsamples == 8) {
|
|
position -= 8;
|
|
for (c = 0; c < nchannels; c++) {
|
|
int16_t *x = &X[c][position];
|
|
x[-7] = AV_RN16(pcm + 14*nchannels + 2*c);
|
|
x[1] = AV_RN16(pcm + 6*nchannels + 2*c);
|
|
x[2] = AV_RN16(pcm + 12*nchannels + 2*c);
|
|
x[3] = AV_RN16(pcm + 0*nchannels + 2*c);
|
|
x[4] = AV_RN16(pcm + 10*nchannels + 2*c);
|
|
x[5] = AV_RN16(pcm + 2*nchannels + 2*c);
|
|
x[6] = AV_RN16(pcm + 8*nchannels + 2*c);
|
|
x[7] = AV_RN16(pcm + 4*nchannels + 2*c);
|
|
}
|
|
}
|
|
|
|
return position;
|
|
}
|
|
|
|
static void sbc_calc_scalefactors(int32_t sb_sample_f[16][2][8],
|
|
uint32_t scale_factor[2][8],
|
|
int blocks, int channels, int subbands)
|
|
{
|
|
int ch, sb, blk;
|
|
for (ch = 0; ch < channels; ch++) {
|
|
for (sb = 0; sb < subbands; sb++) {
|
|
uint32_t x = 1 << SCALE_OUT_BITS;
|
|
for (blk = 0; blk < blocks; blk++) {
|
|
int32_t tmp = FFABS(sb_sample_f[blk][ch][sb]);
|
|
if (tmp != 0)
|
|
x |= tmp - 1;
|
|
}
|
|
scale_factor[ch][sb] = (31 - SCALE_OUT_BITS) - ff_clz(x);
|
|
}
|
|
}
|
|
}
|
|
|
|
static int sbc_calc_scalefactors_j(int32_t sb_sample_f[16][2][8],
|
|
uint32_t scale_factor[2][8],
|
|
int blocks, int subbands)
|
|
{
|
|
int blk, joint = 0;
|
|
int32_t tmp0, tmp1;
|
|
uint32_t x, y;
|
|
|
|
/* last subband does not use joint stereo */
|
|
int sb = subbands - 1;
|
|
x = 1 << SCALE_OUT_BITS;
|
|
y = 1 << SCALE_OUT_BITS;
|
|
for (blk = 0; blk < blocks; blk++) {
|
|
tmp0 = FFABS(sb_sample_f[blk][0][sb]);
|
|
tmp1 = FFABS(sb_sample_f[blk][1][sb]);
|
|
if (tmp0 != 0)
|
|
x |= tmp0 - 1;
|
|
if (tmp1 != 0)
|
|
y |= tmp1 - 1;
|
|
}
|
|
scale_factor[0][sb] = (31 - SCALE_OUT_BITS) - ff_clz(x);
|
|
scale_factor[1][sb] = (31 - SCALE_OUT_BITS) - ff_clz(y);
|
|
|
|
/* the rest of subbands can use joint stereo */
|
|
while (--sb >= 0) {
|
|
int32_t sb_sample_j[16][2];
|
|
x = 1 << SCALE_OUT_BITS;
|
|
y = 1 << SCALE_OUT_BITS;
|
|
for (blk = 0; blk < blocks; blk++) {
|
|
tmp0 = sb_sample_f[blk][0][sb];
|
|
tmp1 = sb_sample_f[blk][1][sb];
|
|
sb_sample_j[blk][0] = (tmp0 >> 1) + (tmp1 >> 1);
|
|
sb_sample_j[blk][1] = (tmp0 >> 1) - (tmp1 >> 1);
|
|
tmp0 = FFABS(tmp0);
|
|
tmp1 = FFABS(tmp1);
|
|
if (tmp0 != 0)
|
|
x |= tmp0 - 1;
|
|
if (tmp1 != 0)
|
|
y |= tmp1 - 1;
|
|
}
|
|
scale_factor[0][sb] = (31 - SCALE_OUT_BITS) -
|
|
ff_clz(x);
|
|
scale_factor[1][sb] = (31 - SCALE_OUT_BITS) -
|
|
ff_clz(y);
|
|
x = 1 << SCALE_OUT_BITS;
|
|
y = 1 << SCALE_OUT_BITS;
|
|
for (blk = 0; blk < blocks; blk++) {
|
|
tmp0 = FFABS(sb_sample_j[blk][0]);
|
|
tmp1 = FFABS(sb_sample_j[blk][1]);
|
|
if (tmp0 != 0)
|
|
x |= tmp0 - 1;
|
|
if (tmp1 != 0)
|
|
y |= tmp1 - 1;
|
|
}
|
|
x = (31 - SCALE_OUT_BITS) - ff_clz(x);
|
|
y = (31 - SCALE_OUT_BITS) - ff_clz(y);
|
|
|
|
/* decide whether to use joint stereo for this subband */
|
|
if ((scale_factor[0][sb] + scale_factor[1][sb]) > x + y) {
|
|
joint |= 1 << (subbands - 1 - sb);
|
|
scale_factor[0][sb] = x;
|
|
scale_factor[1][sb] = y;
|
|
for (blk = 0; blk < blocks; blk++) {
|
|
sb_sample_f[blk][0][sb] = sb_sample_j[blk][0];
|
|
sb_sample_f[blk][1][sb] = sb_sample_j[blk][1];
|
|
}
|
|
}
|
|
}
|
|
|
|
/* bitmask with the information about subbands using joint stereo */
|
|
return joint;
|
|
}
|
|
|
|
/*
|
|
* Detect CPU features and setup function pointers
|
|
*/
|
|
av_cold void ff_sbcdsp_init(SBCDSPContext *s)
|
|
{
|
|
/* Default implementation for analyze functions */
|
|
s->sbc_analyze_4 = sbc_analyze_4_simd;
|
|
s->sbc_analyze_8 = sbc_analyze_8_simd;
|
|
s->sbc_analyze_4s = sbc_analyze_4b_4s_simd;
|
|
if (s->increment == 1)
|
|
s->sbc_analyze_8s = sbc_analyze_1b_8s_simd_odd;
|
|
else
|
|
s->sbc_analyze_8s = sbc_analyze_4b_8s_simd;
|
|
|
|
/* Default implementation for input reordering / deinterleaving */
|
|
s->sbc_enc_process_input_4s = sbc_enc_process_input_4s;
|
|
s->sbc_enc_process_input_8s = sbc_enc_process_input_8s;
|
|
|
|
/* Default implementation for scale factors calculation */
|
|
s->sbc_calc_scalefactors = sbc_calc_scalefactors;
|
|
s->sbc_calc_scalefactors_j = sbc_calc_scalefactors_j;
|
|
|
|
if (ARCH_ARM)
|
|
ff_sbcdsp_init_arm(s);
|
|
if (ARCH_X86)
|
|
ff_sbcdsp_init_x86(s);
|
|
}
|