forked from len0rd/rockbox
Moved id3.c, mpeg.c and replaygain.c from firmware/ to apps/. This is the first step in separating the generic metadata code and the id3-specific code.
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@18760 a1c6a512-1295-4272-9138-f99709370657
This commit is contained in:
parent
98fa3913f9
commit
e76c69f3e4
5 changed files with 6 additions and 4 deletions
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@ -7,6 +7,7 @@ abrepeat.c
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bookmark.c
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debug_menu.c
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filetypes.c
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id3.c
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language.c
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main.c
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menu.c
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@ -17,6 +18,9 @@ menus/theme_menu.c
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menus/eq_menu.c
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buffering.c
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voice_thread.c
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replaygain.c
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#else /* !SWCODEC */
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mpeg.c
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#endif
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menus/main_menu.c
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menus/playback_menu.c
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1353
apps/id3.c
Normal file
1353
apps/id3.c
Normal file
File diff suppressed because it is too large
Load diff
2873
apps/mpeg.c
Normal file
2873
apps/mpeg.c
Normal file
File diff suppressed because it is too large
Load diff
457
apps/replaygain.c
Normal file
457
apps/replaygain.c
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@ -0,0 +1,457 @@
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/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2005 Magnus Holmgren
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include <ctype.h>
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#include <inttypes.h>
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#include <math.h>
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#include <stdbool.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <system.h>
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#include "id3.h"
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#include "debug.h"
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#include "replaygain.h"
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/* The fixed point math routines (with the exception of fp_atof) are based
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* on oMathFP by Dan Carter (http://orbisstudios.com).
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*/
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/* 12 bits of precision gives fairly accurate result, but still allows a
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* compact implementation. The math code supports up to 13...
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*/
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#define FP_BITS (12)
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#define FP_MASK ((1 << FP_BITS) - 1)
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#define FP_ONE (1 << FP_BITS)
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#define FP_TWO (2 << FP_BITS)
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#define FP_HALF (1 << (FP_BITS - 1))
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#define FP_LN2 ( 45426 >> (16 - FP_BITS))
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#define FP_LN2_INV ( 94548 >> (16 - FP_BITS))
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#define FP_EXP_ZERO ( 10922 >> (16 - FP_BITS))
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#define FP_EXP_ONE ( -182 >> (16 - FP_BITS))
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#define FP_EXP_TWO ( 4 >> (16 - FP_BITS))
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#define FP_INF (0x7fffffff)
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#define FP_LN10 (150902 >> (16 - FP_BITS))
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#define FP_MAX_DIGITS (4)
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#define FP_MAX_DIGITS_INT (10000)
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#define FP_FAST_MUL_DIV
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#ifdef FP_FAST_MUL_DIV
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/* These macros can easily overflow, but they are good enough for our uses,
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* and saves some code.
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*/
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#define fp_mul(x, y) (((x) * (y)) >> FP_BITS)
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#define fp_div(x, y) (((x) << FP_BITS) / (y))
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#else
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static long fp_mul(long x, long y)
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{
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long x_neg = 0;
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long y_neg = 0;
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long rc;
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if ((x == 0) || (y == 0))
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{
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return 0;
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}
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if (x < 0)
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{
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x_neg = 1;
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x = -x;
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}
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if (y < 0)
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{
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y_neg = 1;
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y = -y;
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}
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rc = (((x >> FP_BITS) * (y >> FP_BITS)) << FP_BITS)
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+ (((x & FP_MASK) * (y & FP_MASK)) >> FP_BITS)
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+ ((x & FP_MASK) * (y >> FP_BITS))
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+ ((x >> FP_BITS) * (y & FP_MASK));
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if ((x_neg ^ y_neg) == 1)
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{
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rc = -rc;
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}
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return rc;
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}
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static long fp_div(long x, long y)
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{
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long x_neg = 0;
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long y_neg = 0;
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long shifty;
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long rc;
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int msb = 0;
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int lsb = 0;
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if (x == 0)
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{
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return 0;
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}
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if (y == 0)
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{
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return (x < 0) ? -FP_INF : FP_INF;
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}
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if (x < 0)
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{
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x_neg = 1;
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x = -x;
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}
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if (y < 0)
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{
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y_neg = 1;
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y = -y;
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}
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while ((x & (1 << (30 - msb))) == 0)
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{
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msb++;
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}
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while ((y & (1 << lsb)) == 0)
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{
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lsb++;
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}
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shifty = FP_BITS - (msb + lsb);
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rc = ((x << msb) / (y >> lsb));
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if (shifty > 0)
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{
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rc <<= shifty;
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}
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else
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{
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rc >>= -shifty;
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}
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if ((x_neg ^ y_neg) == 1)
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{
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rc = -rc;
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}
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return rc;
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}
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#endif /* FP_FAST_MUL_DIV */
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static long fp_exp(long x)
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{
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long k;
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long z;
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long R;
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long xp;
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if (x == 0)
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{
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return FP_ONE;
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}
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k = (fp_mul(abs(x), FP_LN2_INV) + FP_HALF) & ~FP_MASK;
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if (x < 0)
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{
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k = -k;
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}
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x -= fp_mul(k, FP_LN2);
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z = fp_mul(x, x);
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R = FP_TWO + fp_mul(z, FP_EXP_ZERO + fp_mul(z, FP_EXP_ONE
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+ fp_mul(z, FP_EXP_TWO)));
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xp = FP_ONE + fp_div(fp_mul(FP_TWO, x), R - x);
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if (k < 0)
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{
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k = FP_ONE >> (-k >> FP_BITS);
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}
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else
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{
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k = FP_ONE << (k >> FP_BITS);
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}
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return fp_mul(k, xp);
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}
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static long fp_exp10(long x)
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{
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if (x == 0)
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{
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return FP_ONE;
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}
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return fp_exp(fp_mul(FP_LN10, x));
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}
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static long fp_atof(const char* s, int precision)
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{
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long int_part = 0;
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long int_one = 1 << precision;
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long frac_part = 0;
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long frac_count = 0;
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long frac_max = ((precision * 4) + 12) / 13;
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long frac_max_int = 1;
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long sign = 1;
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bool point = false;
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while ((*s != '\0') && isspace(*s))
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{
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s++;
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}
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if (*s == '-')
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{
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sign = -1;
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s++;
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}
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else if (*s == '+')
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{
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s++;
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}
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while (*s != '\0')
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{
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if (*s == '.')
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{
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if (point)
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{
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break;
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}
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point = true;
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}
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else if (isdigit(*s))
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{
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if (point)
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{
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if (frac_count < frac_max)
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{
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frac_part = frac_part * 10 + (*s - '0');
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frac_count++;
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frac_max_int *= 10;
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}
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}
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else
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{
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int_part = int_part * 10 + (*s - '0');
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}
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}
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else
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{
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break;
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}
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s++;
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}
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while (frac_count < frac_max)
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{
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frac_part *= 10;
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frac_count++;
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frac_max_int *= 10;
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}
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return sign * ((int_part * int_one)
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+ (((int64_t) frac_part * int_one) / frac_max_int));
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}
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static long convert_gain(long gain)
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{
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/* Don't allow unreasonably low or high gain changes.
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* Our math code can't handle it properly anyway. :)
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*/
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if (gain < (-48 * FP_ONE))
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{
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gain = -48 * FP_ONE;
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}
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if (gain > (17 * FP_ONE))
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{
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gain = 17 * FP_ONE;
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}
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gain = fp_exp10(gain / 20) << (24 - FP_BITS);
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return gain;
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}
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/* Get the sample scale factor in Q7.24 format from a gain value. Returns 0
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* for no gain.
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*
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* str Gain in dB as a string. E.g., "-3.45 dB"; the "dB" part is ignored.
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*/
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static long get_replaygain(const char* str)
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{
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long gain = 0;
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if (str)
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{
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gain = fp_atof(str, FP_BITS);
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gain = convert_gain(gain);
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}
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return gain;
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}
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/* Get the peak volume in Q7.24 format.
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*
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* str Peak volume. Full scale is specified as "1.0". Returns 0 for no peak.
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*/
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static long get_replaypeak(const char* str)
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{
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long peak = 0;
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if (str)
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{
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peak = fp_atof(str, 24);
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}
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return peak;
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}
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/* Get a sample scale factor in Q7.24 format from a gain value.
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*
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* int_gain Gain in dB, multiplied by 100.
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*/
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long get_replaygain_int(long int_gain)
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{
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return convert_gain(int_gain * FP_ONE / 100);
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}
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/* Parse a ReplayGain tag conforming to the "VorbisGain standard". If a
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* valid tag is found, update mp3entry struct accordingly. Existing values
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* are not overwritten. Returns number of bytes written to buffer.
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*
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* key Name of the tag.
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* value Value of the tag.
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* entry mp3entry struct to update.
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* buffer Where to store the text for gain values (for later display).
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* length Bytes left in buffer.
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*/
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long parse_replaygain(const char* key, const char* value,
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struct mp3entry* entry, char* buffer, int length)
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{
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char **p = NULL;
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if (((strcasecmp(key, "replaygain_track_gain") == 0)
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|| (strcasecmp(key, "rg_radio") == 0)) && !entry->track_gain)
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{
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entry->track_gain = get_replaygain(value);
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p = &(entry->track_gain_string);
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}
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else if (((strcasecmp(key, "replaygain_album_gain") == 0)
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|| (strcasecmp(key, "rg_audiophile") == 0)) && !entry->album_gain)
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{
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entry->album_gain = get_replaygain(value);
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p = &(entry->album_gain_string);
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}
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else if (((strcasecmp(key, "replaygain_track_peak") == 0)
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|| (strcasecmp(key, "rg_peak") == 0)) && !entry->track_peak)
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{
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entry->track_peak = get_replaypeak(value);
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}
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else if ((strcasecmp(key, "replaygain_album_peak") == 0)
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&& !entry->album_peak)
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{
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entry->album_peak = get_replaypeak(value);
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}
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if (p)
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{
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int len = strlen(value);
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len = MIN(len, length - 1);
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/* A few characters just isn't interesting... */
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if (len > 1)
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{
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strncpy(buffer, value, len);
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buffer[len] = 0;
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*p = buffer;
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return len + 1;
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}
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}
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return 0;
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}
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/* Set ReplayGain values from integers. Existing values are not overwritten.
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* Returns number of bytes written to buffer.
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*
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* album If true, set album values, otherwise set track values.
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* gain Gain value in dB, multiplied by 512. 0 for no gain.
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* peak Peak volume in Q7.24 format, where 1.0 is full scale. 0 for no
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* peak volume.
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* buffer Where to store the text for gain values (for later display).
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* length Bytes left in buffer.
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*/
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long parse_replaygain_int(bool album, long gain, long peak,
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struct mp3entry* entry, char* buffer, int length)
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{
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long len = 0;
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if (buffer != NULL)
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{
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len = snprintf(buffer, length, "%d.%02d dB", gain / 512,
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((abs(gain) & 0x01ff) * 100 + 256) / 512);
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len++;
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}
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if (gain != 0)
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{
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gain = convert_gain(gain * FP_ONE / 512);
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}
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if (album)
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{
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entry->album_gain = gain;
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entry->album_gain_string = buffer;
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if (peak)
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{
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entry->album_peak = peak;
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}
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}
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else
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{
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entry->track_gain = gain;
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entry->track_gain_string = buffer;
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if (peak)
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{
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entry->track_peak = peak;
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}
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}
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return len;
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}
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