2 * various filters for ACELP-based codecs
4 * Copyright (c) 2008 Vladimir Voroshilov
6 * This file is part of Libav.
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9 * modify it under the terms of the GNU Lesser General Public
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23 #ifndef AVCODEC_ACELP_FILTERS_H
24 #define AVCODEC_ACELP_FILTERS_H
29 * low-pass Finite Impulse Response filter coefficients.
31 * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
32 * the coefficients are scaled by 2^15.
33 * This array only contains the right half of the filter.
34 * This filter is likely identical to the one used in G.729, though this
35 * could not be determined from the original comments with certainty.
37 extern const int16_t ff_acelp_interp_filter[61];
40 * Generic FIR interpolation routine.
41 * @param[out] out buffer for interpolated data
42 * @param in input data
43 * @param filter_coeffs interpolation filter coefficients (0.15)
44 * @param precision sub sample factor, that is the precision of the position
45 * @param frac_pos fractional part of position [0..precision-1]
46 * @param filter_length filter length
47 * @param length length of output
49 * filter_coeffs contains coefficients of the right half of the symmetric
50 * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
51 * See ff_acelp_interp_filter for an example.
53 void ff_acelp_interpolate(int16_t* out, const int16_t* in,
54 const int16_t* filter_coeffs, int precision,
55 int frac_pos, int filter_length, int length);
58 * Floating point version of ff_acelp_interpolate()
60 void ff_acelp_interpolatef(float *out, const float *in,
61 const float *filter_coeffs, int precision,
62 int frac_pos, int filter_length, int length);
66 * high-pass filtering and upscaling (4.2.5 of G.729).
67 * @param[out] out output buffer for filtered speech data
68 * @param[in,out] hpf_f past filtered data from previous (2 items long)
69 * frames (-0x20000000 <= (14.13) < 0x20000000)
70 * @param in speech data to process
71 * @param length input data size
73 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
74 * 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
76 * The filter has a cut-off frequency of 1/80 of the sampling freq
78 * @note Two items before the top of the out buffer must contain two items from the
79 * tail of the previous subframe.
81 * @remark It is safe to pass the same array in in and out parameters.
83 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
84 * but constants differs in 5th sign after comma). Fortunately in
85 * fixed-point all coefficients are the same as in G.729. Thus this
86 * routine can be used for the fixed-point AMR decoder, too.
88 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
89 const int16_t* in, int length);
92 * Apply an order 2 rational transfer function in-place.
94 * @param out output buffer for filtered speech samples
95 * @param in input buffer containing speech data (may be the same as out)
96 * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
97 * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
98 * @param gain scale factor for final output
99 * @param mem intermediate values used by filter (should be 0 initially)
100 * @param n number of samples
102 void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
103 const float zero_coeffs[2],
104 const float pole_coeffs[2],
106 float mem[2], int n);
109 * Apply tilt compensation filter, 1 - tilt * z-1.
111 * @param mem pointer to the filter's state (one single float)
112 * @param tilt tilt factor
113 * @param samples array where the filter is applied
114 * @param size the size of the samples array
116 void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
119 #endif /* AVCODEC_ACELP_FILTERS_H */