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acelp_filters.h
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1 /*
2  * various filters for ACELP-based codecs
3  *
4  * Copyright (c) 2008 Vladimir Voroshilov
5  *
6  * This file is part of FFmpeg.
7  *
8  * FFmpeg is free software; you can redistribute it and/or
9  * modify it under the terms of the GNU Lesser General Public
10  * License as published by the Free Software Foundation; either
11  * version 2.1 of the License, or (at your option) any later version.
12  *
13  * FFmpeg is distributed in the hope that it will be useful,
14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16  * Lesser General Public License for more details.
17  *
18  * You should have received a copy of the GNU Lesser General Public
19  * License along with FFmpeg; if not, write to the Free Software
20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21  */
22 
23 #ifndef AVCODEC_ACELP_FILTERS_H
24 #define AVCODEC_ACELP_FILTERS_H
25 
26 #include <stdint.h>
27 
28 typedef struct ACELPFContext {
29  /**
30  * Floating point version of ff_acelp_interpolate()
31  */
32  void (*acelp_interpolatef)(float *out, const float *in,
33  const float *filter_coeffs, int precision,
34  int frac_pos, int filter_length, int length);
35 
36  /**
37  * Apply an order 2 rational transfer function in-place.
38  *
39  * @param out output buffer for filtered speech samples
40  * @param in input buffer containing speech data (may be the same as out)
41  * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
42  * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
43  * @param gain scale factor for final output
44  * @param mem intermediate values used by filter (should be 0 initially)
45  * @param n number of samples (should be a multiple of eight)
46  */
47  void (*acelp_apply_order_2_transfer_function)(float *out, const float *in,
48  const float zero_coeffs[2],
49  const float pole_coeffs[2],
50  float gain,
51  float mem[2], int n);
52 
54 
55 /**
56  * Initialize ACELPFContext.
57  */
60 
61 /**
62  * low-pass Finite Impulse Response filter coefficients.
63  *
64  * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
65  * the coefficients are scaled by 2^15.
66  * This array only contains the right half of the filter.
67  * This filter is likely identical to the one used in G.729, though this
68  * could not be determined from the original comments with certainty.
69  */
70 extern const int16_t ff_acelp_interp_filter[61];
71 
72 /**
73  * Generic FIR interpolation routine.
74  * @param[out] out buffer for interpolated data
75  * @param in input data
76  * @param filter_coeffs interpolation filter coefficients (0.15)
77  * @param precision sub sample factor, that is the precision of the position
78  * @param frac_pos fractional part of position [0..precision-1]
79  * @param filter_length filter length
80  * @param length length of output
81  *
82  * filter_coeffs contains coefficients of the right half of the symmetric
83  * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
84  * See ff_acelp_interp_filter for an example.
85  *
86  */
87 void ff_acelp_interpolate(int16_t* out, const int16_t* in,
88  const int16_t* filter_coeffs, int precision,
89  int frac_pos, int filter_length, int length);
90 
91 /**
92  * Floating point version of ff_acelp_interpolate()
93  */
94 void ff_acelp_interpolatef(float *out, const float *in,
95  const float *filter_coeffs, int precision,
96  int frac_pos, int filter_length, int length);
97 
98 
99 /**
100  * high-pass filtering and upscaling (4.2.5 of G.729).
101  * @param[out] out output buffer for filtered speech data
102  * @param[in,out] hpf_f past filtered data from previous (2 items long)
103  * frames (-0x20000000 <= (14.13) < 0x20000000)
104  * @param in speech data to process
105  * @param length input data size
106  *
107  * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
108  * 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
109  *
110  * The filter has a cut-off frequency of 1/80 of the sampling freq
111  *
112  * @note Two items before the top of the in buffer must contain two items from the
113  * tail of the previous subframe.
114  *
115  * @remark It is safe to pass the same array in in and out parameters.
116  *
117  * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
118  * but constants differs in 5th sign after comma). Fortunately in
119  * fixed-point all coefficients are the same as in G.729. Thus this
120  * routine can be used for the fixed-point AMR decoder, too.
121  */
122 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
123  const int16_t* in, int length);
124 
125 /**
126  * Apply an order 2 rational transfer function in-place.
127  *
128  * @param out output buffer for filtered speech samples
129  * @param in input buffer containing speech data (may be the same as out)
130  * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
131  * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
132  * @param gain scale factor for final output
133  * @param mem intermediate values used by filter (should be 0 initially)
134  * @param n number of samples
135  */
136 void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
137  const float zero_coeffs[2],
138  const float pole_coeffs[2],
139  float gain,
140  float mem[2], int n);
141 
142 /**
143  * Apply tilt compensation filter, 1 - tilt * z-1.
144  *
145  * @param mem pointer to the filter's state (one single float)
146  * @param tilt tilt factor
147  * @param samples array where the filter is applied
148  * @param size the size of the samples array
149  */
150 void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
151 
152 
153 #endif /* AVCODEC_ACELP_FILTERS_H */