/* * various filters for ACELP-based codecs * * Copyright (c) 2008 Vladimir Voroshilov * * This file is part of Libav. * * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #ifndef AVCODEC_ACELP_FILTERS_H #define AVCODEC_ACELP_FILTERS_H #include <stdint.h> /** * low-pass Finite Impulse Response filter coefficients. * * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq, * the coefficients are scaled by 2^15. * This array only contains the right half of the filter. * This filter is likely identical to the one used in G.729, though this * could not be determined from the original comments with certainity. */ extern const int16_t ff_acelp_interp_filter[61]; /** * Generic FIR interpolation routine. * @param[out] out buffer for interpolated data * @param in input data * @param filter_coeffs interpolation filter coefficients (0.15) * @param precision sub sample factor, that is the precision of the position * @param frac_pos fractional part of position [0..precision-1] * @param filter_length filter length * @param length length of output * * filter_coeffs contains coefficients of the right half of the symmetric * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient. * See ff_acelp_interp_filter for an example. * */ void ff_acelp_interpolate(int16_t* out, const int16_t* in, const int16_t* filter_coeffs, int precision, int frac_pos, int filter_length, int length); /** * Floating point version of ff_acelp_interpolate() */ void ff_acelp_interpolatef(float *out, const float *in, const float *filter_coeffs, int precision, int frac_pos, int filter_length, int length); /** * high-pass filtering and upscaling (4.2.5 of G.729). * @param[out] out output buffer for filtered speech data * @param[in,out] hpf_f past filtered data from previous (2 items long) * frames (-0x20000000 <= (14.13) < 0x20000000) * @param in speech data to process * @param length input data size * * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] + * 1.9330735 * out[i-1] - 0.93589199 * out[i-2] * * The filter has a cut-off frequency of 1/80 of the sampling freq * * @note Two items before the top of the out buffer must contain two items from the * tail of the previous subframe. * * @remark It is safe to pass the same array in in and out parameters. * * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula, * but constants differs in 5th sign after comma). Fortunately in * fixed-point all coefficients are the same as in G.729. Thus this * routine can be used for the fixed-point AMR decoder, too. */ void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2], const int16_t* in, int length); /** * Apply an order 2 rational transfer function in-place. * * @param out output buffer for filtered speech samples * @param in input buffer containing speech data (may be the same as out) * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator * @param gain scale factor for final output * @param mem intermediate values used by filter (should be 0 initially) * @param n number of samples */ void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, const float zero_coeffs[2], const float pole_coeffs[2], float gain, float mem[2], int n); /** * Apply tilt compensation filter, 1 - tilt * z-1. * * @param mem pointer to the filter's state (one single float) * @param tilt tilt factor * @param samples array where the filter is applied * @param size the size of the samples array */ void ff_tilt_compensation(float *mem, float tilt, float *samples, int size); #endif /* AVCODEC_ACELP_FILTERS_H */