2 * various filters for CELP-based codecs
4 * Copyright (c) 2008 Vladimir Voroshilov
6 * This file is part of Libav.
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
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23 #ifndef AVCODEC_CELP_FILTERS_H
24 #define AVCODEC_CELP_FILTERS_H
29 * Circularly convolve fixed vector with a phase dispersion impulse
30 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
31 * @param fc_out vector with filter applied
32 * @param fc_in source vector
33 * @param filter phase filter coefficients
35 * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
37 * \note fc_in and fc_out should not overlap!
39 void ff_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in,
40 const int16_t *filter, int len);
43 * Add an array to a rotated array.
45 * out[k] = in[k] + fac * lagged[k-lag] with wrap-around
47 * @param out result vector
48 * @param in samples to be added unfiltered
49 * @param lagged samples to be rotated, multiplied and added
50 * @param lag lagged vector delay in the range [0, n]
51 * @param fac scalefactor for lagged samples
52 * @param n number of samples
54 void ff_celp_circ_addf(float *out, const float *in,
55 const float *lagged, int lag, float fac, int n);
58 * LP synthesis filter.
59 * @param[out] out pointer to output buffer
60 * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
61 * @param in input signal
62 * @param buffer_length amount of data to process
63 * @param filter_length filter length (10 for 10th order LP filter)
64 * @param stop_on_overflow 1 - return immediately if overflow occurs
65 * 0 - ignore overflows
66 * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
68 * @return 1 if overflow occurred, 0 - otherwise
70 * @note Output buffer must contain filter_length samples of past
71 * speech data before pointer.
73 * Routine applies 1/A(z) filter to given speech data.
75 int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
76 const int16_t *in, int buffer_length,
77 int filter_length, int stop_on_overflow,
81 * LP synthesis filter.
82 * @param[out] out pointer to output buffer
83 * - the array out[-filter_length, -1] must
84 * contain the previous result of this filter
85 * @param filter_coeffs filter coefficients.
86 * @param in input signal
87 * @param buffer_length amount of data to process
88 * @param filter_length filter length (10 for 10th order LP filter). Must be
89 * greater than 4 and even.
91 * @note Output buffer must contain filter_length samples of past
92 * speech data before pointer.
94 * Routine applies 1/A(z) filter to given speech data.
96 void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs,
97 const float *in, int buffer_length,
101 * LP zero synthesis filter.
102 * @param[out] out pointer to output buffer
103 * @param filter_coeffs filter coefficients.
104 * @param in input signal
105 * - the array in[-filter_length, -1] must
106 * contain the previous input of this filter
107 * @param buffer_length amount of data to process
108 * @param filter_length filter length (10 for 10th order LP filter)
110 * @note Output buffer must contain filter_length samples of past
111 * speech data before pointer.
113 * Routine applies A(z) filter to given speech data.
115 void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs,
116 const float *in, int buffer_length,
119 #endif /* AVCODEC_CELP_FILTERS_H */