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fc193793c6
* qatar/master: aacdec: Use float instead of int16_t for ltp_state to avoid needless rounding. acelp: Remove unused gray_decode table. dfa: Remove unused variable. configure: Include AVX availability in summary output. configure: use same CPPFLAGS in kFreeBSD as Linux Merged-by: Michael Niedermayer <michaelni@gmx.at>
261 lines
6.7 KiB
C
261 lines
6.7 KiB
C
/*
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* adaptive and fixed codebook vector operations for ACELP-based codecs
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*
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* Copyright (c) 2008 Vladimir Voroshilov
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include <inttypes.h>
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#include "avcodec.h"
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#include "acelp_vectors.h"
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#include "celp_math.h"
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const uint8_t ff_fc_2pulses_9bits_track1[16] =
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{
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1, 3,
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6, 8,
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11, 13,
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16, 18,
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21, 23,
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26, 28,
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31, 33,
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36, 38
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};
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const uint8_t ff_fc_2pulses_9bits_track1_gray[16] =
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{
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1, 3,
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8, 6,
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18, 16,
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11, 13,
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38, 36,
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31, 33,
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21, 23,
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28, 26,
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};
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const uint8_t ff_fc_2pulses_9bits_track2_gray[32] =
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{
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0, 2,
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5, 4,
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12, 10,
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7, 9,
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25, 24,
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20, 22,
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14, 15,
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19, 17,
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36, 31,
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21, 26,
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1, 6,
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16, 11,
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27, 29,
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32, 30,
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39, 37,
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34, 35,
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};
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const uint8_t ff_fc_4pulses_8bits_tracks_13[16] =
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{
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0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75,
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};
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const uint8_t ff_fc_4pulses_8bits_track_4[32] =
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{
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3, 4,
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8, 9,
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13, 14,
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18, 19,
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23, 24,
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28, 29,
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33, 34,
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38, 39,
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43, 44,
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48, 49,
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53, 54,
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58, 59,
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63, 64,
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68, 69,
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73, 74,
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78, 79,
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};
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const float ff_pow_0_7[10] = {
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0.700000, 0.490000, 0.343000, 0.240100, 0.168070,
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0.117649, 0.082354, 0.057648, 0.040354, 0.028248
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};
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const float ff_pow_0_75[10] = {
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0.750000, 0.562500, 0.421875, 0.316406, 0.237305,
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0.177979, 0.133484, 0.100113, 0.075085, 0.056314
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};
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const float ff_pow_0_55[10] = {
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0.550000, 0.302500, 0.166375, 0.091506, 0.050328,
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0.027681, 0.015224, 0.008373, 0.004605, 0.002533
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};
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const float ff_b60_sinc[61] = {
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0.898529 , 0.865051 , 0.769257 , 0.624054 , 0.448639 , 0.265289 ,
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0.0959167 , -0.0412598 , -0.134338 , -0.178986 , -0.178528 , -0.142609 ,
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-0.0849304 , -0.0205078 , 0.0369568 , 0.0773926 , 0.0955200 , 0.0912781 ,
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0.0689392 , 0.0357056 , 0. , -0.0305481 , -0.0504150 , -0.0570068 ,
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-0.0508423 , -0.0350037 , -0.0141602 , 0.00665283, 0.0230713 , 0.0323486 ,
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0.0335388 , 0.0275879 , 0.0167847 , 0.00411987, -0.00747681, -0.0156860 ,
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-0.0193481 , -0.0183716 , -0.0137634 , -0.00704956, 0. , 0.00582886 ,
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0.00939941, 0.0103760 , 0.00903320, 0.00604248, 0.00238037, -0.00109863 ,
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-0.00366211, -0.00497437, -0.00503540, -0.00402832, -0.00241089, -0.000579834,
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0.00103760, 0.00222778, 0.00277710, 0.00271606, 0.00213623, 0.00115967 ,
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0.
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};
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void ff_acelp_fc_pulse_per_track(
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int16_t* fc_v,
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const uint8_t *tab1,
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const uint8_t *tab2,
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int pulse_indexes,
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int pulse_signs,
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int pulse_count,
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int bits)
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{
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int mask = (1 << bits) - 1;
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int i;
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for(i=0; i<pulse_count; i++)
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{
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fc_v[i + tab1[pulse_indexes & mask]] +=
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(pulse_signs & 1) ? 8191 : -8192; // +/-1 in (2.13)
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pulse_indexes >>= bits;
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pulse_signs >>= 1;
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}
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fc_v[tab2[pulse_indexes]] += (pulse_signs & 1) ? 8191 : -8192;
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}
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void ff_decode_10_pulses_35bits(const int16_t *fixed_index,
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AMRFixed *fixed_sparse,
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const uint8_t *gray_decode,
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int half_pulse_count, int bits)
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{
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int i;
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int mask = (1 << bits) - 1;
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fixed_sparse->no_repeat_mask = 0;
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fixed_sparse->n = 2 * half_pulse_count;
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for (i = 0; i < half_pulse_count; i++) {
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const int pos1 = gray_decode[fixed_index[2*i+1] & mask] + i;
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const int pos2 = gray_decode[fixed_index[2*i ] & mask] + i;
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const float sign = (fixed_index[2*i+1] & (1 << bits)) ? -1.0 : 1.0;
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fixed_sparse->x[2*i+1] = pos1;
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fixed_sparse->x[2*i ] = pos2;
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fixed_sparse->y[2*i+1] = sign;
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fixed_sparse->y[2*i ] = pos2 < pos1 ? -sign : sign;
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}
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}
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void ff_acelp_weighted_vector_sum(
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int16_t* out,
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const int16_t *in_a,
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const int16_t *in_b,
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int16_t weight_coeff_a,
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int16_t weight_coeff_b,
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int16_t rounder,
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int shift,
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int length)
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{
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int i;
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// Clipping required here; breaks OVERFLOW test.
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for(i=0; i<length; i++)
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out[i] = av_clip_int16((
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in_a[i] * weight_coeff_a +
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in_b[i] * weight_coeff_b +
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rounder) >> shift);
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}
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void ff_weighted_vector_sumf(float *out, const float *in_a, const float *in_b,
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float weight_coeff_a, float weight_coeff_b, int length)
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{
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int i;
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for(i=0; i<length; i++)
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out[i] = weight_coeff_a * in_a[i]
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+ weight_coeff_b * in_b[i];
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}
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void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
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int size, float alpha, float *gain_mem)
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{
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int i;
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float postfilter_energ = ff_dot_productf(in, in, size);
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float gain_scale_factor = 1.0;
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float mem = *gain_mem;
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if (postfilter_energ)
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gain_scale_factor = sqrt(speech_energ / postfilter_energ);
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gain_scale_factor *= 1.0 - alpha;
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for (i = 0; i < size; i++) {
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mem = alpha * mem + gain_scale_factor;
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out[i] = in[i] * mem;
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}
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*gain_mem = mem;
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}
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void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in,
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float sum_of_squares, const int n)
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{
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int i;
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float scalefactor = ff_dot_productf(in, in, n);
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if (scalefactor)
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scalefactor = sqrt(sum_of_squares / scalefactor);
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for (i = 0; i < n; i++)
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out[i] = in[i] * scalefactor;
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}
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void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size)
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{
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int i;
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for (i=0; i < in->n; i++) {
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int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
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float y = in->y[i] * scale;
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do {
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out[x] += y;
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y *= in->pitch_fac;
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x += in->pitch_lag;
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} while (x < size && repeats);
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}
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}
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void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size)
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{
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int i;
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for (i=0; i < in->n; i++) {
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int x = in->x[i], repeats = !((in->no_repeat_mask >> i) & 1);
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do {
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out[x] = 0.0;
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x += in->pitch_lag;
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} while (x < size && repeats);
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}
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}
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