freeswitch/libs/spandsp/src/gsm0610_preprocess.c

151 lines
4.2 KiB
C

/*
* SpanDSP - a series of DSP components for telephony
*
* gsm0610_preprocess.c - GSM 06.10 full rate speech codec.
*
* Written by Steve Underwood <steveu@coppice.org>
*
* Copyright (C) 2006 Steve Underwood
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU Lesser General Public License version 2.1,
* as published by the Free Software Foundation.
*
* This program 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 this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* This code is based on the widely used GSM 06.10 code available from
* http://kbs.cs.tu-berlin.de/~jutta/toast.html
*/
/*! \file */
#if defined(HAVE_CONFIG_H)
#include "config.h"
#endif
#include <assert.h>
#include <inttypes.h>
#if defined(HAVE_TGMATH_H)
#include <tgmath.h>
#endif
#if defined(HAVE_MATH_H)
#include <math.h>
#endif
#include "floating_fudge.h"
#include <stdlib.h>
#include "spandsp/telephony.h"
#include "spandsp/fast_convert.h"
#include "spandsp/bitstream.h"
#include "spandsp/saturated.h"
#include "spandsp/gsm0610.h"
#include "gsm0610_local.h"
/*
4.2.0 .. 4.2.3 PREPROCESSING SECTION
After A-law to linear conversion (or directly from the
A to D converter) the following scaling is assumed for
input to the RPE-LTP algorithm:
in: 0.1.....................12
S.v.v.v.v.v.v.v.v.v.v.v.v.*.*.*
Where S is the sign bit, v a valid bit, and * a "don't care" bit.
The original signal is called sop[..]
out: 0.1................... 12
S.S.v.v.v.v.v.v.v.v.v.v.v.v.0.0
*/
void gsm0610_preprocess(gsm0610_state_t *s, const int16_t amp[GSM0610_FRAME_LEN], int16_t so[GSM0610_FRAME_LEN])
{
int16_t z1;
int16_t mp;
int16_t s1;
int16_t msp;
int16_t SO;
int32_t L_z2;
int32_t L_s2;
int32_t L_temp;
#if !defined(__GNUC__)
int16_t lsp;
#endif
int k;
z1 = s->z1;
L_z2 = s->L_z2;
mp = s->mp;
for (k = 0; k < GSM0610_FRAME_LEN; k++)
{
/* 4.2.1 Downscaling of the input signal */
SO = (amp[k] >> 1) & ~3;
/* This is supposed to have been downscaled by previous routine. */
assert(SO >= -0x4000);
assert(SO <= 0x3FFC);
/* 4.2.2 Offset compensation */
/* This part implements a high-pass filter and requires extended
arithmetic precision for the recursive part of this filter.
The input of this procedure is the array so[0...159] and the
output the array sof[0...159].
*/
/* Compute the non-recursive part */
s1 = SO - z1;
z1 = SO;
assert(s1 != INT16_MIN);
/* Compute the recursive part */
L_s2 = s1;
L_s2 <<= 15;
/* Perform a 31 by 16 bits multiplication */
#if defined(__GNUC__)
L_z2 = ((int64_t) L_z2*32735 + 0x4000) >> 15;
/* Alternate (ANSI) version of below line does slightly different rounding:
* L_temp = L_z2 >> 9;
* L_temp += L_temp >> 5;
* L_temp = (++L_temp) >> 1;
* L_z2 = L_z2 - L_temp;
*/
L_z2 = sat_add32(L_z2, L_s2);
#else
/* This does L_z2 = L_z2 * 0x7FD5/0x8000 + L_s2 */
msp = (int16_t) (L_z2 >> 15);
lsp = (int16_t) (L_z2 - ((int32_t) msp << 15));
L_s2 += gsm_mult_r(lsp, 32735);
L_temp = (int32_t) msp*32735;
L_z2 = sat_add32(L_temp, L_s2);
#endif
/* Compute sof[k] with rounding */
L_temp = sat_add32(L_z2, 16384);
/* 4.2.3 Preemphasis */
msp = gsm_mult_r(mp, -28180);
mp = (int16_t) (L_temp >> 15);
so[k] = sat_add16(mp, msp);
}
/*endfor*/
s->z1 = z1;
s->L_z2 = L_z2;
s->mp = mp;
}
/*- End of function --------------------------------------------------------*/
/*- End of file ------------------------------------------------------------*/