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FFmpeg/libavcodec/acelp_filters.c
Vladimir Voroshilov 542c064d1b Update comment to version, negotiated with Diego, and
fix missing period (not latest revision of patch
was wrongly committed).

Originally committed as revision 13113 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-05-11 10:07:11 +00:00

125 lines
3.5 KiB
C

/*
* various filters for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <inttypes.h>
#include "avcodec.h"
#include "acelp_filters.h"
#define FRAC_BITS 13
#include "mathops.h"
void ff_acelp_convolve_circ(
int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int subframe_size)
{
int i, k;
memset(fc_out, 0, subframe_size * sizeof(int16_t));
/* Since there are few pulses over an entire subframe (i.e. almost
all fc_in[i] are zero) it is faster to swap two loops and process
non-zero samples only. In the case of G.729D the buffer contains
two non-zero samples before the call to ff_acelp_enhance_harmonics
and, due to pitch_delay being bounded by [20; 143], a maximum
of four non-zero samples for a total of 40 after the call. */
for(i=0; i<subframe_size; i++)
{
if(fc_in[i])
{
for(k=0; k<i; k++)
fc_out[k] += (fc_in[i] * filter[subframe_size + k - i]) >> 15;
for(k=i; k<subframe_size; k++)
fc_out[k] += (fc_in[i] * filter[k - i]) >> 15;
}
}
}
int ff_acelp_lp_synthesis_filter(
int16_t *out,
const int16_t* filter_coeffs,
const int16_t* in,
int buffer_length,
int filter_length,
int stop_on_overflow)
{
int i,n;
for(n=0; n<buffer_length; n++)
{
int sum = 0x800;
for(i=1; i<filter_length; i++)
sum -= filter_coeffs[i] * out[n-i];
sum = (sum >> 12) + in[n];
/* Check for overflow */
if(sum + 0x8000 > 0xFFFFU)
{
if(stop_on_overflow)
return 1;
sum = (sum >> 31) ^ 32767;
}
out[n] = sum;
}
return 0;
}
void ff_acelp_weighted_filter(
int16_t *out,
const int16_t* in,
const int16_t *weight_pow,
int filter_length)
{
int n;
for(n=0; n<filter_length; n++)
out[n] = (in[n] * weight_pow[n] + 0x4000) >> 15; /* (3.12) = (0.15) * (3.12) with rounding */
}
void ff_acelp_high_pass_filter(
int16_t* out,
int hpf_f[2],
const int16_t* in,
int length)
{
int i;
int tmp;
for(i=0; i<length; i++)
{
tmp = MULL(hpf_f[0], 15836); /* (14.13) = (13.13) * (1.13) */
tmp += MULL(hpf_f[1], -7667); /* (13.13) = (13.13) * (0.13) */
tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]); /* (14.13) = (0.13) * (14.0) */
/* Multiplication by 2 with rounding can cause short type
overflow, thus clipping is required. */
out[i] = av_clip_int16((tmp + 0x800) >> 12); /* (15.0) = 2 * (13.13) = (14.13) */
hpf_f[1] = hpf_f[0];
hpf_f[0] = tmp;
}
}