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aebf07075f
The DCA core decoder converts integer coefficients read from the bitstream to floats just after reading them (along with dequantization). All the other steps of the audio reconstruction are done with floats which makes the output for the DTS lossless extension (XLL) actually lossy. This patch changes the DCA core to work with integer coefficients until QMF. At this point the integer coefficients are converted to floats. The coefficients for the LFE channel (lfe_data) are not touched. This is the first step for the really lossless XLL decoding.
67 lines
2.1 KiB
C
67 lines
2.1 KiB
C
/*
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* Format Conversion Utils
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* Copyright (c) 2000, 2001 Fabrice Bellard
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* Copyright (c) 2002-2004 Michael Niedermayer <michaelni@gmx.at>
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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 "avcodec.h"
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#include "fmtconvert.h"
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#include "libavutil/common.h"
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static void int32_to_float_fmul_scalar_c(float *dst, const int32_t *src,
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float mul, int len)
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{
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int i;
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for(i=0; i<len; i++)
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dst[i] = src[i] * mul;
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}
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static void int32_to_float_c(float *dst, const int32_t *src, intptr_t len)
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{
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int i;
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for (i = 0; i < len; i++)
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dst[i] = (float)src[i];
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}
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static void int32_to_float_fmul_array8_c(FmtConvertContext *c, float *dst,
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const int32_t *src, const float *mul,
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int len)
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{
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int i;
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for (i = 0; i < len; i += 8)
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c->int32_to_float_fmul_scalar(&dst[i], &src[i], *mul++, 8);
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}
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av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
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{
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c->int32_to_float = int32_to_float_c;
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c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_c;
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c->int32_to_float_fmul_array8 = int32_to_float_fmul_array8_c;
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if (ARCH_AARCH64)
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ff_fmt_convert_init_aarch64(c, avctx);
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if (ARCH_ARM)
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ff_fmt_convert_init_arm(c, avctx);
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if (ARCH_PPC)
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ff_fmt_convert_init_ppc(c, avctx);
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if (ARCH_X86)
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ff_fmt_convert_init_x86(c, avctx);
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}
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