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authorAurelien Jacobs <aurel@gnuage.org>2017-12-17 19:59:30 +0100
committerAurelien Jacobs <aurel@gnuage.org>2018-03-07 22:26:53 +0100
commitff4600d95471a653073a961ec77f32e2f946684a (patch)
tree403810f4aeb160c06803c31a10c6e02c65b86fb8 /libavcodec/sbcdsp.c
parent2e08de08159df2079f1db2a7d8fe66e2ad2238d5 (diff)
downloadffmpeg-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.c382
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;
+}