path: root/libavcodec/acelp_filters.h (plain)
blob: fe86cb20974238a011004ed871ba66278cb875c0

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
53	}ACELPFContext;
54
55	/**
56	* Initialize ACELPFContext.
57	*/
58	void ff_acelp_filter_init(ACELPFContext *c);
59	void ff_acelp_filter_init_mips(ACELPFContext *c);
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	void ff_acelp_interpolate(int16_t* out, const int16_t* in,
87	const int16_t* filter_coeffs, int precision,
88	int frac_pos, int filter_length, int length);
89
90	/**
91	* Floating point version of ff_acelp_interpolate()
92	*/
93	void ff_acelp_interpolatef(float *out, const float *in,
94	const float *filter_coeffs, int precision,
95	int frac_pos, int filter_length, int length);
96
97
98	/**
99	* high-pass filtering and upscaling (4.2.5 of G.729).
100	* @param[out] out output buffer for filtered speech data
101	* @param[in,out] hpf_f past filtered data from previous (2 items long)
102	* frames (-0x20000000 <= (14.13) < 0x20000000)
103	* @param in speech data to process
104	* @param length input data size
105	*
106	* out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
107	* 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
108	*
109	* The filter has a cut-off frequency of 1/80 of the sampling freq
110	*
111	* @note Two items before the top of the in buffer must contain two items from the
112	* tail of the previous subframe.
113	*
114	* @remark It is safe to pass the same array in in and out parameters.
115	*
116	* @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
117	* but constants differs in 5th sign after comma). Fortunately in
118	* fixed-point all coefficients are the same as in G.729. Thus this
119	* routine can be used for the fixed-point AMR decoder, too.
120	*/
121	void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
122	const int16_t* in, int length);
123
124	/**
125	* Apply an order 2 rational transfer function in-place.
126	*
127	* @param out output buffer for filtered speech samples
128	* @param in input buffer containing speech data (may be the same as out)
129	* @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
130	* @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
131	* @param gain scale factor for final output
132	* @param mem intermediate values used by filter (should be 0 initially)
133	* @param n number of samples
134	*/
135	void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
136	const float zero_coeffs[2],
137	const float pole_coeffs[2],
138	float gain,
139	float mem[2], int n);
140
141	/**
142	* Apply tilt compensation filter, 1 - tilt * z-1.
143	*
144	* @param mem pointer to the filter's state (one single float)
145	* @param tilt tilt factor
146	* @param samples array where the filter is applied
147	* @param size the size of the samples array
148	*/
149	void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
150
151
152	#endif /* AVCODEC_ACELP_FILTERS_H */
153