1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
|
/*
* various filters for CELP-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
*/
#ifndef AVCODEC_CELP_FILTERS_H
#define AVCODEC_CELP_FILTERS_H
#include <stdint.h>
/**
* Circularly convolve fixed vector with a phase dispersion impulse
* response filter (D.6.2 of G.729 and 6.1.5 of AMR).
* @param fc_out vector with filter applied
* @param fc_in source vector
* @param filter phase filter coefficients
*
* fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
*
* \note fc_in and fc_out should not overlap!
*/
void ff_celp_convolve_circ(
int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int len);
/**
* LP synthesis filter.
* @param out [out] pointer to output buffer
* @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
* @param in input signal
* @param buffer_length amount of data to process
* @param filter_length filter length (10 for 10th order LP filter)
* @param stop_on_overflow 1 - return immediately if overflow occurs
* 0 - ignore overflows
* @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
*
* @return 1 if overflow occurred, 0 - otherwise
*
* @note Output buffer must contain filter_length samples of past
* speech data before pointer.
*
* Routine applies 1/A(z) filter to given speech data.
*/
int ff_celp_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 rounder);
/**
* LP synthesis filter.
* @param out [out] pointer to output buffer
* - the array out[-filter_length, -1] must
* contain the previous result of this filter
* @param filter_coeffs filter coefficients.
* @param in input signal
* @param buffer_length amount of data to process
* @param filter_length filter length (10 for 10th order LP filter)
*
* @note Output buffer must contain filter_length samples of past
* speech data before pointer.
*
* Routine applies 1/A(z) filter to given speech data.
*/
void ff_celp_lp_synthesis_filterf(
float *out,
const float* filter_coeffs,
const float* in,
int buffer_length,
int filter_length);
#endif /* AVCODEC_CELP_FILTERS_H */
|
