ffmpeg/libavcodec/acelp_vectors.c

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/*
* adaptive and fixed codebook vector operations for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <inttypes.h>
#include "libavutil/common.h"
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "acelp_vectors.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_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,
};
const float ff_pow_0_7[10] = {
0.700000, 0.490000, 0.343000, 0.240100, 0.168070,
0.117649, 0.082354, 0.057648, 0.040354, 0.028248
};
const float ff_pow_0_75[10] = {
0.750000, 0.562500, 0.421875, 0.316406, 0.237305,
0.177979, 0.133484, 0.100113, 0.075085, 0.056314
};
const float ff_pow_0_55[10] = {
0.550000, 0.302500, 0.166375, 0.091506, 0.050328,
0.027681, 0.015224, 0.008373, 0.004605, 0.002533
};
const float ff_b60_sinc[61] = {
0.898529 , 0.865051 , 0.769257 , 0.624054 , 0.448639 , 0.265289 ,
0.0959167 , -0.0412598 , -0.134338 , -0.178986 , -0.178528 , -0.142609 ,
-0.0849304 , -0.0205078 , 0.0369568 , 0.0773926 , 0.0955200 , 0.0912781 ,
0.0689392 , 0.0357056 , 0. , -0.0305481 , -0.0504150 , -0.0570068 ,
-0.0508423 , -0.0350037 , -0.0141602 , 0.00665283, 0.0230713 , 0.0323486 ,
0.0335388 , 0.0275879 , 0.0167847 , 0.00411987, -0.00747681, -0.0156860 ,
-0.0193481 , -0.0183716 , -0.0137634 , -0.00704956, 0. , 0.00582886 ,
0.00939941, 0.0103760 , 0.00903320, 0.00604248, 0.00238037, -0.00109863 ,
-0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
0.00103760, 0.00222778, 0.00277710, 0.00271606, 0.00213623, 0.00115967 ,
0.
};
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_decode_10_pulses_35bits(const int16_t *fixed_index,
AMRFixed *fixed_sparse,
const uint8_t *gray_decode,
int half_pulse_count, int bits)
{
int i;
int mask = (1 << bits) - 1;
fixed_sparse->no_repeat_mask = 0;
fixed_sparse->n = 2 * half_pulse_count;
for (i = 0; i < half_pulse_count; i++) {
const int pos1 = gray_decode[fixed_index[2*i+1] & mask] + i;
const int pos2 = gray_decode[fixed_index[2*i ] & mask] + i;
const float sign = (fixed_index[2*i+1] & (1 << bits)) ? -1.0 : 1.0;
fixed_sparse->x[2*i+1] = pos1;
fixed_sparse->x[2*i ] = pos2;
fixed_sparse->y[2*i+1] = sign;
fixed_sparse->y[2*i ] = pos2 < pos1 ? -sign : sign;
}
}
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_adaptive_gain_control(float *out, const float *in, float speech_energ,
int size, float alpha, float *gain_mem)
{
int i;
float postfilter_energ = avpriv_scalarproduct_float_c(in, in, 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;
out[i] = in[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 = avpriv_scalarproduct_float_c(in, in, n);
if (scalefactor)
scalefactor = sqrt(sum_of_squares / scalefactor);
for (i = 0; i < n; i++)
out[i] = in[i] * scalefactor;
}
void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
{
int i;
for (i=0; i < in->n; i++) {
int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
float y = in->y[i] * scale;
do {
out[x] += y;
y *= in->pitch_fac;
x += in->pitch_lag;
} while (x < size && repeats);
}
}
void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
{
int i;
for (i=0; i < in->n; i++) {
int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
do {
out[x] = 0.0;
x += in->pitch_lag;
} while (x < size && repeats);
}
}