/* * Bluetooth low-complexity, subband codec (SBC) * * Copyright (C) 2017 Aurelien Jacobs * Copyright (C) 2012-2013 Intel Corporation * Copyright (C) 2008-2010 Nokia Corporation * Copyright (C) 2004-2010 Marcel Holtmann * Copyright (C) 2004-2005 Henryk Ploetz * Copyright (C) 2005-2006 Brad Midgley * * 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 #include #include #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); }