| /* | |
| * various filters for ACELP-based codecs | |
| * | |
| * Copyright (c) 2008 Vladimir Voroshilov | |
| * | |
| * This file is part of FFmpeg. | |
| * | |
| * FFmpeg 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. | |
| * | |
| * FFmpeg 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 FFmpeg; if not, write to the Free Software | |
| * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
| */ | |
| typedef struct ACELPFContext { | |
| /** | |
| * Floating point version of ff_acelp_interpolate() | |
| */ | |
| void (*acelp_interpolatef)(float *out, const float *in, | |
| const float *filter_coeffs, int precision, | |
| int frac_pos, int filter_length, 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 (should be a multiple of eight) | |
| */ | |
| void (*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); | |
| }ACELPFContext; | |
| /** | |
| * Initialize ACELPFContext. | |
| */ | |
| void ff_acelp_filter_init(ACELPFContext *c); | |
| void ff_acelp_filter_init_mips(ACELPFContext *c); | |
| /** | |
| * 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 certainty. | |
| */ | |
| 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 in 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); | |