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author | Aurelien Jacobs <aurel@gnuage.org> | 2017-12-17 19:59:30 +0100 |
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committer | Aurelien Jacobs <aurel@gnuage.org> | 2018-03-07 22:26:53 +0100 |
commit | ff4600d95471a653073a961ec77f32e2f946684a (patch) | |
tree | 403810f4aeb160c06803c31a10c6e02c65b86fb8 /libavcodec/sbcdsp.c | |
parent | 2e08de08159df2079f1db2a7d8fe66e2ad2238d5 (diff) | |
download | ffmpeg-ff4600d95471a653073a961ec77f32e2f946684a.tar.gz |
sbc: implement SBC encoder (low-complexity subband codec)
This was originally based on libsbc, and was fully integrated into ffmpeg.
Diffstat (limited to 'libavcodec/sbcdsp.c')
-rw-r--r-- | libavcodec/sbcdsp.c | 382 |
1 files changed, 382 insertions, 0 deletions
diff --git a/libavcodec/sbcdsp.c b/libavcodec/sbcdsp.c new file mode 100644 index 0000000000..e155387f0d --- /dev/null +++ b/libavcodec/sbcdsp.c @@ -0,0 +1,382 @@ +/* + * 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; +} |