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FFmpeg/libavcodec/acelp_vectors.h

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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
*/
#ifndef AVCODEC_ACELP_VECTORS_H
#define AVCODEC_ACELP_VECTORS_H
#include <stdint.h>
/** Sparse representation for the algebraic codebook (fixed) vector */
typedef struct {
int n;
int x[10];
float y[10];
int no_repeat_mask;
int pitch_lag;
float pitch_fac;
} AMRFixed;
/**
* Track|Pulse| Positions
* -------------------------------------------------------------------------
* 1 | 0 | 0, 5, 10, 15, 20, 25, 30, 35, 40, 45, 50, 55, 60, 65, 70, 75
* -------------------------------------------------------------------------
* 2 | 1 | 1, 6, 11, 16, 21, 26, 31, 36, 41, 46, 51, 56, 61, 66, 71, 76
* -------------------------------------------------------------------------
* 3 | 2 | 2, 7, 12, 17, 22, 27, 32, 37, 42, 47, 52, 57, 62, 67, 72, 77
* -------------------------------------------------------------------------
*
* Table contains only first the pulse indexes.
*
* Used in G.729 @@8k, G.729 @@4.4k, AMR @@7.95k, AMR @@7.40k
*/
extern const uint8_t ff_fc_4pulses_8bits_tracks_13[16];
/**
* Track|Pulse| Positions
* -------------------------------------------------------------------------
* 4 | 3 | 3, 8, 13, 18, 23, 28, 33, 38, 43, 48, 53, 58, 63, 68, 73, 78
* | | 4, 9, 14, 19, 24, 29, 34, 39, 44, 49, 54, 59, 64, 69, 74, 79
* -------------------------------------------------------------------------
*
* @remark Track in the table should be read top-to-bottom, left-to-right.
*
* Used in G.729 @@8k, G.729 @@4.4k, AMR @@7.95k, AMR @@7.40k
*/
extern const uint8_t ff_fc_4pulses_8bits_track_4[32];
/**
* Track|Pulse| Positions
* -----------------------------------------
* 1 | 0 | 1, 6, 11, 16, 21, 26, 31, 36
* | | 3, 8, 13, 18, 23, 28, 33, 38
* -----------------------------------------
*
* @remark Track in the table should be read top-to-bottom, left-to-right.
*
* @note (EE) Reference G.729D code also uses gray decoding for each
* pulse index before looking up the value in the table.
*
* Used in G.729 @@6.4k (with gray coding), AMR @@5.9k (without gray coding)
*/
extern const uint8_t ff_fc_2pulses_9bits_track1[16];
extern const uint8_t ff_fc_2pulses_9bits_track1_gray[16];
/**
* b60 hamming windowed sinc function coefficients
*/
extern const float ff_b60_sinc[61];
/**
* Table of pow(0.7,n)
*/
extern const float ff_pow_0_7[10];
/**
* Table of pow(0.75,n)
*/
extern const float ff_pow_0_75[10];
/**
* Table of pow(0.55,n)
*/
extern const float ff_pow_0_55[10];
/**
* Decode fixed-codebook vector (3.8 and D.5.8 of G.729, 5.7.1 of AMR).
* @param[out] fc_v decoded fixed codebook vector (2.13)
* @param tab1 table used for first pulse_count pulses
* @param tab2 table used for last pulse
* @param pulse_indexes fixed codebook indexes
* @param pulse_signs signs of the excitation pulses (0 bit value
* means negative sign)
* @param bits number of bits per one pulse index
* @param pulse_count number of pulses decoded using first table
* @param bits length of one pulse index in bits
*
* Used in G.729 @@8k, G.729 @@4.4k, G.729 @@6.4k, AMR @@7.95k, AMR @@7.40k
*/
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);
/**
* Decode the algebraic codebook index to pulse positions and signs and
* construct the algebraic codebook vector for MODE_12k2.
*
* @note: The positions and signs are explicitly coded in MODE_12k2.
*
* @param fixed_index positions of the ten pulses
* @param fixed_sparse pointer to the algebraic codebook vector
* @param gray_decode gray decoding table
* @param half_pulse_count number of couples of pulses
* @param bits length of one pulse index in bits
*/
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);
/**
* weighted sum of two vectors with rounding.
* @param[out] out result of addition
* @param in_a first vector
* @param in_b second vector
* @param weight_coeff_a first vector weight coefficient
* @param weight_coeff_a second vector weight coefficient
* @param rounder this value will be added to the sum of the two vectors
* @param shift result will be shifted to right by this value
* @param length vectors length
*
* @note It is safe to pass the same buffer for out and in_a or in_b.
*
* out[i] = (in_a[i]*weight_a + in_b[i]*weight_b + rounder) >> shift
*/
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);
/**
* float implementation of weighted sum of two vectors.
* @param[out] out result of addition
* @param in_a first vector
* @param in_b second vector
* @param weight_coeff_a first vector weight coefficient
* @param weight_coeff_a second vector weight coefficient
* @param length vectors length
*
* @note It is safe to pass the same buffer for out and in_a or in_b.
*/
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);
/**
* Adaptive gain control (as used in AMR postfiltering)
*
* @param out output buffer for filtered speech data
* @param in the input speech buffer (may be the same as out)
* @param speech_energ input energy
* @param size the input buffer size
* @param alpha exponential filter factor
* @param gain_mem a pointer to the filter memory (single float of size)
*/
void ff_adaptive_gain_control(float *out, const float *in, float speech_energ,
int size, float alpha, float *gain_mem);
/**
* Set the sum of squares of a signal by scaling
*
* @param out output samples
* @param in input samples
* @param sum_of_squares new sum of squares
* @param n number of samples
*
* @note If the input is zero (or its energy underflows), the output is zero.
* This is the behavior of AGC in the AMR reference decoder. The QCELP
* reference decoder seems to have undefined behavior.
*
* TIA/EIA/IS-733 2.4.8.3-2/3/4/5, 2.4.8.6
* 3GPP TS 26.090 6.1 (6)
*/
void ff_scale_vector_to_given_sum_of_squares(float *out, const float *in,
float sum_of_squares, const int n);
/**
* Add fixed vector to an array from a sparse representation
*
* @param out fixed vector with pitch sharpening
* @param in sparse fixed vector
* @param scale number to multiply the fixed vector by
* @param size the output vector size
*/
void ff_set_fixed_vector(float *out, const AMRFixed *in, float scale, int size);
/**
* Clear array values set by set_fixed_vector
*
* @param out fixed vector to be cleared
* @param in sparse fixed vector
* @param size the output vector size
*/
void ff_clear_fixed_vector(float *out, const AMRFixed *in, int size);
#endif /* AVCODEC_ACELP_VECTORS_H */
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