ffmpeg/libavcodec/acelp_vectors.c

191 lines
4.2 KiB
C

/*
* adaptive and fixed codebook vector operations 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_vectors.h"
#include "celp_math.h"
const uint8_t ff_fc_2pulses_9bits_track1[16] =
{
1, 3,
6, 8,
11, 13,
16, 18,
21, 23,
26, 28,
31, 33,
36, 38
};
const uint8_t ff_fc_2pulses_9bits_track1_gray[16] =
{
1, 3,
8, 6,
18, 16,
11, 13,
38, 36,
31, 33,
21, 23,
28, 26,
};
const uint8_t ff_fc_2pulses_9bits_track2_gray[32] =
{
0, 2,
5, 4,
12, 10,
7, 9,
25, 24,
20, 22,
14, 15,
19, 17,
36, 31,
21, 26,
1, 6,
16, 11,
27, 29,
32, 30,
39, 37,
34, 35,
};
const uint8_t ff_fc_4pulses_8bits_tracks_13[16] =
{
0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75,
};
const uint8_t ff_fc_4pulses_8bits_track_4[32] =
{
3, 4,
8, 9,
13, 14,
18, 19,
23, 24,
28, 29,
33, 34,
38, 39,
43, 44,
48, 49,
53, 54,
58, 59,
63, 64,
68, 69,
73, 74,
78, 79,
};
#if 0
static uint8_t gray_decode[32] =
{
0, 1, 3, 2, 7, 6, 4, 5,
15, 14, 12, 13, 8, 9, 11, 10,
31, 30, 28, 29, 24, 25, 27, 26,
16, 17, 19, 18, 23, 22, 20, 21
};
#endif
void ff_acelp_fc_pulse_per_track(
int16_t* fc_v,
const uint8_t *tab1,
const uint8_t *tab2,
int pulse_indexes,
int pulse_signs,
int pulse_count,
int bits)
{
int mask = (1 << bits) - 1;
int i;
for(i=0; i<pulse_count; i++)
{
fc_v[i + tab1[pulse_indexes & mask]] +=
(pulse_signs & 1) ? 8191 : -8192; // +/-1 in (2.13)
pulse_indexes >>= bits;
pulse_signs >>= 1;
}
fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192;
}
void ff_acelp_weighted_vector_sum(
int16_t* out,
const int16_t *in_a,
const int16_t *in_b,
int16_t weight_coeff_a,
int16_t weight_coeff_b,
int16_t rounder,
int shift,
int length)
{
int i;
// Clipping required here; breaks OVERFLOW test.
for(i=0; i<length; i++)
out[i] = av_clip_int16((
in_a[i] * weight_coeff_a +
in_b[i] * weight_coeff_b +
rounder) >> shift);
}
void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
float weight_coeff_a, float weight_coeff_b, int length)
{
int i;
for(i=0; i<length; i++)
out[i] = weight_coeff_a * in_a[i]
+ weight_coeff_b * in_b[i];
}
void ff_adaptative_gain_control(float *buf_out, float speech_energ,
int size, float alpha, float *gain_mem)
{
int i;
float postfilter_energ = ff_dot_productf(buf_out, buf_out, size);
float gain_scale_factor = 1.0;
float mem = *gain_mem;
if (postfilter_energ)
gain_scale_factor = sqrt(speech_energ / postfilter_energ);
gain_scale_factor *= 1.0 - alpha;
for (i = 0; i < size; i++) {
mem = alpha * mem + gain_scale_factor;
buf_out[i] *= mem;
}
*gain_mem = mem;
}
void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in,
float sum_of_squares, const int n)
{
int i;
float scalefactor = ff_dot_productf(in, in, n);
if (scalefactor)
scalefactor = sqrt(sum_of_squares / scalefactor);
for (i = 0; i < n; i++)
out[i] = in[i] * scalefactor;
}