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4cb6964244
The vector dequantization has a test in a loop preventing effective SIMD
implementation. By moving it out of the loop, this loop can be DSPized.
Therefore, modify the current DSP implementation. In particular, the
DSP implementation no longer has to handle null loop sizes.
The decode_hf implementations have following timings:
For x86 Arrandale:
C SSE SSE2 SSE4
win32: 260 162 119 104
win64: 242 N/A 89 72
The arm NEON optimizations follow in a later patch as external asm. The
now unused check for the y modifier in arm inline asm is removed from
configure.
113 lines
3.7 KiB
C
113 lines
3.7 KiB
C
/*
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* Copyright (c) 2004 Gildas Bazin
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* Copyright (c) 2010 Mans Rullgard <mans@mansr.com>
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*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with Libav; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "config.h"
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#include "libavutil/attributes.h"
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#include "libavutil/intreadwrite.h"
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#include "dcadsp.h"
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static void decode_hf_c(float dst[DCA_SUBBANDS][8],
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const int32_t vq_num[DCA_SUBBANDS],
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const int8_t hf_vq[1024][32], intptr_t vq_offset,
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int32_t scale[DCA_SUBBANDS][2],
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intptr_t start, intptr_t end)
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{
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int i, l;
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for (l = start; l < end; l++) {
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/* 1 vector -> 32 samples but we only need the 8 samples
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* for this subsubframe. */
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const int8_t *ptr = &hf_vq[vq_num[l]][vq_offset];
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float fscale = scale[l][0] * (1 / 16.0);
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for (i = 0; i < 8; i++)
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dst[l][i] = ptr[i] * fscale;
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}
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}
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static inline void
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dca_lfe_fir(float *out, const float *in, const float *coefs,
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int decifactor)
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{
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float *out2 = out + 2 * decifactor - 1;
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int num_coeffs = 256 / decifactor;
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int j, k;
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/* One decimated sample generates 2*decifactor interpolated ones */
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for (k = 0; k < decifactor; k++) {
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float v0 = 0.0;
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float v1 = 0.0;
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for (j = 0; j < num_coeffs; j++, coefs++) {
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v0 += in[-j] * *coefs;
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v1 += in[j + 1 - num_coeffs] * *coefs;
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}
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*out++ = v0;
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*out2-- = v1;
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}
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}
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static void dca_qmf_32_subbands(float samples_in[32][8], int sb_act,
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SynthFilterContext *synth, FFTContext *imdct,
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float synth_buf_ptr[512],
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int *synth_buf_offset, float synth_buf2[32],
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const float window[512], float *samples_out,
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float raXin[32], float scale)
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{
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int i;
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int subindex;
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for (i = sb_act; i < 32; i++)
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raXin[i] = 0.0;
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/* Reconstructed channel sample index */
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for (subindex = 0; subindex < 8; subindex++) {
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/* Load in one sample from each subband and clear inactive subbands */
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for (i = 0; i < sb_act; i++) {
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unsigned sign = (i - 1) & 2;
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uint32_t v = AV_RN32A(&samples_in[i][subindex]) ^ sign << 30;
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AV_WN32A(&raXin[i], v);
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}
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synth->synth_filter_float(imdct, synth_buf_ptr, synth_buf_offset,
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synth_buf2, window, samples_out, raXin, scale);
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samples_out += 32;
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}
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}
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static void dca_lfe_fir0_c(float *out, const float *in, const float *coefs)
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{
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dca_lfe_fir(out, in, coefs, 32);
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}
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static void dca_lfe_fir1_c(float *out, const float *in, const float *coefs)
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{
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dca_lfe_fir(out, in, coefs, 64);
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}
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av_cold void ff_dcadsp_init(DCADSPContext *s)
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{
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s->lfe_fir[0] = dca_lfe_fir0_c;
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s->lfe_fir[1] = dca_lfe_fir1_c;
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s->qmf_32_subbands = dca_qmf_32_subbands;
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s->decode_hf = decode_hf_c;
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if (ARCH_ARM) ff_dcadsp_init_arm(s);
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if (ARCH_X86) ff_dcadsp_init_x86(s);
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}
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