1 /*

2 * various filters for ACELP-based codecs

3 *

4 * Copyright (c) 2008 Vladimir Voroshilov

5 *

6 * This file is part of Libav.

7 *

8 * Libav 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 * Libav 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 Libav; if not, write to the Free Software

20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA

21 */

23 #ifndef AVCODEC_ACELP_FILTERS_H

24 #define AVCODEC_ACELP_FILTERS_H

26 #include <stdint.h>

28 /**

29 * low-pass Finite Impulse Response filter coefficients.

30 *

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.

36 */

39 /**

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

48 *

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.

52 *

53 */

58 /**

59 * Floating point version of ff_acelp_interpolate()

60 */

66 /**

67 * high-pass filtering and upscaling (4.2.5 of G.729).

68 * @param[out] out output buffer for filtered speech data

69 * @param[in,out] hpf_f past filtered data from previous (2 items long)

70 * frames (-0x20000000 <= (14.13) < 0x20000000)

71 * @param in speech data to process

72 * @param length input data size

73 *

74 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +

75 * 1.9330735 * out[i-1] - 0.93589199 * out[i-2]

76 *

77 * The filter has a cut-off frequency of 1/80 of the sampling freq

78 *

79 * @note Two items before the top of the out buffer must contain two items from the

80 * tail of the previous subframe.

81 *

82 * @remark It is safe to pass the same array in in and out parameters.

83 *

84 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,

85 * but constants differs in 5th sign after comma). Fortunately in

86 * fixed-point all coefficients are the same as in G.729. Thus this

87 * routine can be used for the fixed-point AMR decoder, too.

88 */

92 /**

93 * Apply an order 2 rational transfer function in-place.

94 *

95 * @param out output buffer for filtered speech samples

96 * @param in input buffer containing speech data (may be the same as out)

97 * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator

98 * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator

99 * @param gain scale factor for final output

100 * @param mem intermediate values used by filter (should be 0 initially)

101 * @param n number of samples

102 */

109 /**

110 * Apply tilt compensation filter, 1 - tilt * z-1.

111 *

112 * @param mem pointer to the filter's state (one single float)

113 * @param tilt tilt factor

114 * @param samples array where the filter is applied

115 * @param size the size of the samples array

116 */