| /* |
| * 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); |
| } |
