mirror of https://git.ffmpeg.org/ffmpeg.git
dca: NEON optimised high freq VQ decoding
Signed-off-by: Mans Rullgard <mans@mansr.com>
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@ -0,0 +1,49 @@
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/*
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* Copyright (c) 2011 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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#ifndef AVCODEC_ARM_DCA_H
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#define AVCODEC_ARM_DCA_H
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#include <stdint.h>
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#include "config.h"
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#if HAVE_NEON && HAVE_INLINE_ASM
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#define int8x8_fmul_int32 int8x8_fmul_int32
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static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
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{
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__asm__ ("vcvt.f32.s32 %2, %2, #4 \n"
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"vld1.8 {d0}, [%1,:64] \n"
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"vmovl.s8 q0, d0 \n"
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"vmovl.s16 q1, d1 \n"
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"vmovl.s16 q0, d0 \n"
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"vcvt.f32.s32 q0, q0 \n"
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"vcvt.f32.s32 q1, q1 \n"
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"vmul.f32 q0, q0, %y2 \n"
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"vmul.f32 q1, q1, %y2 \n"
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"vst1.32 {q0-q1}, [%m0,:128] \n"
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: "=Um"(*(float (*)[8])dst)
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: "r"(src), "x"(scale)
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: "d0", "d1", "d2", "d3");
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}
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#endif
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#endif /* AVCODEC_ARM_DCA_H */
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@ -42,6 +42,10 @@
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#include "dcadsp.h"
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#include "fmtconvert.h"
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#if ARCH_ARM
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# include "arm/dca.h"
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#endif
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//#define TRACE
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#define DCA_PRIM_CHANNELS_MAX (7)
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@ -320,7 +324,7 @@ typedef struct {
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int lfe_scale_factor;
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/* Subband samples history (for ADPCM) */
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float subband_samples_hist[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
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DECLARE_ALIGNED(16, float, subband_samples_hist)[DCA_PRIM_CHANNELS_MAX][DCA_SUBBANDS][4];
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DECLARE_ALIGNED(32, float, subband_fir_hist)[DCA_PRIM_CHANNELS_MAX][512];
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DECLARE_ALIGNED(32, float, subband_fir_noidea)[DCA_PRIM_CHANNELS_MAX][32];
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int hist_index[DCA_PRIM_CHANNELS_MAX];
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@ -1057,6 +1061,16 @@ static int decode_blockcode(int code, int levels, int *values)
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static const uint8_t abits_sizes[7] = { 7, 10, 12, 13, 15, 17, 19 };
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static const uint8_t abits_levels[7] = { 3, 5, 7, 9, 13, 17, 25 };
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#ifndef int8x8_fmul_int32
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static inline void int8x8_fmul_int32(float *dst, const int8_t *src, int scale)
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{
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float fscale = scale / 16.0;
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int i;
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for (i = 0; i < 8; i++)
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dst[i] = src[i] * fscale;
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}
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#endif
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static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
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{
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int k, l;
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@ -1161,19 +1175,16 @@ static int dca_subsubframe(DCAContext * s, int base_channel, int block_index)
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for (l = s->vq_start_subband[k]; l < s->subband_activity[k]; 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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int m;
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int hfvq = s->high_freq_vq[k][l];
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if (!s->debug_flag & 0x01) {
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av_log(s->avctx, AV_LOG_DEBUG, "Stream with high frequencies VQ coding\n");
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s->debug_flag |= 0x01;
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}
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for (m = 0; m < 8; m++) {
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subband_samples[k][l][m] =
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high_freq_vq[s->high_freq_vq[k][l]][subsubframe * 8 +
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m]
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* (float) s->scale_factor[k][l][0] / 16.0;
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}
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int8x8_fmul_int32(subband_samples[k][l],
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&high_freq_vq[hfvq][subsubframe * 8],
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s->scale_factor[k][l][0]);
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}
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}
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@ -4224,7 +4224,7 @@ static const float lossless_quant_d[32] = {
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/* Vector quantization tables */
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static const int8_t high_freq_vq[1024][32] =
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DECLARE_ALIGNED(8, static const int8_t, high_freq_vq)[1024][32] =
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{
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{ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
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0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
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