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
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
|
/*
* various filters 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
*/
#ifndef FFMPEG_ACELP_FILTERS_H
#define FFMPEG_ACELP_FILTERS_H
#include <stdint.h>
/**
* low-pass FIR (Finite Impulse Response) filter coefficients
*
* A similar filter is named b30 in G.729.
*
* G.729 specification says:
* b30 is based on Hamming windowed sinc functions, truncated at +/-29 and
* padded with zeros at +/-30 b30[30]=0.
* The filter has a cut-off frequency (-3 dB) at 3600 Hz in the oversampled
* domain.
*
* After some analysis, I found this approximation:
*
* PI * x
* Hamm(x,N) = 0.53836-0.46164*cos(--------)
* N-1
* ---
* 2
*
* PI * x
* Hamm'(x,k) = Hamm(x - k, 2*k+1) = 0.53836 + 0.46164*cos(--------)
* k
*
* sin(PI * x)
* Sinc(x) = ----------- (normalized sinc function)
* PI * x
*
* h(t,B) = 2 * B * Sinc(2 * B * t) (impulse response of sinc low-pass filter)
*
* b(k,B, n) = Hamm'(n, k) * h(n, B)
*
*
* 3600
* B = ----
* 8000
*
* 3600 - cut-off frequency
* 8000 - sampling rate
* k - filter order
*
* ff_acelp_interp_filter[6*i+j] = b(10, 3600/8000, i+j/6)
*
* The filter assumes the following order of fractions (X - integer delay):
*
* 1/3 precision: X 1/3 2/3 X 1/3 2/3 X
* 1/6 precision: X 1/6 2/6 3/6 4/6 5/6 X 1/6 2/6 3/6 4/6 5/6 X
*
* The filter can be used for 1/3 precision, too, by
* passing 2*pitch_delay_frac as third parameter to the interpolation routine.
*
*/
extern const int16_t ff_acelp_interp_filter[61];
/**
* \brief Generic interpolation routine
* \param out [out] buffer for interpolated data
* \param in input data
* \param filter_coeffs interpolation filter coefficients (0.15)
* \param precision filter is able to interpolate with 1/precision precision of pitch delay
* \param pitch_delay_frac pitch delay, fractional part [0..precision-1]
* \param filter_length filter length
* \param length length of speech data to process
*
* filter_coeffs contains coefficients of the positive half of the symmetric
* interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
* See ff_acelp_interp_filter fot example.
*
*/
void ff_acelp_interpolate(
int16_t* out,
const int16_t* in,
const int16_t* filter_coeffs,
int precision,
int pitch_delay_frac,
int filter_length,
int length);
/**
* \brief 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_acelp_convolve_circ(
int16_t* fc_out,
const int16_t* fc_in,
const int16_t* filter,
int subframe_size);
/**
* \brief 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 (11 for 10th order LP filter)
* \param stop_on_overflow 1 - return immediately if overflow occurs
* 0 - ignore overflows
*
* \return 1 if overflow occurred, 0 - otherwise
*
* \note Output buffer must contain 10 samples of past
* speech data before pointer.
*
* Routine applies 1/A(z) filter to given speech data.
*/
int ff_acelp_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);
/**
* \brief Calculates coefficients of weighted A(z/weight) filter.
* \param out [out] weighted A(z/weight) result
* filter (-0x8000 <= (3.12) < 0x8000)
* \param in source filter (-0x8000 <= (3.12) < 0x8000)
* \param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000)
* \param filter_length filter length (11 for 10th order LP filter)
*
* out[i]=weight_pow[i]*in[i] , i=0..9
*/
void ff_acelp_weighted_filter(
int16_t *out,
const int16_t* in,
const int16_t *weight_pow,
int filter_length);
/**
* \brief high-pass filtering and upscaling (4.2.5 of G.729)
* \param out [out] output buffer for filtered speech data
* \param hpf_f [in/out] past filtered data from previous (2 items long)
* frames (-0x20000000 <= (14.13) < 0x20000000)
* \param in speech data to process
* \param length input data size
*
* out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
* 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
*
* The filter has a cut-off frequency of 100Hz
*
* \note Two items before the top of the out buffer must contain two items from the
* tail of the previous subframe.
*
* \remark It is safe to pass the same array in in and out parameters.
*
* \remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
* but constants differs in 5th sign after comma). Fortunately in
* fixed-point all coefficients are the same as in G.729. Thus this
* routine can be used for the fixed-point AMR decoder, too.
*/
void ff_acelp_high_pass_filter(
int16_t* out,
int hpf_f[2],
const int16_t* in,
int length);
#endif /* FFMPEG_ACELP_FILTERS_H */
|
