/* * 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 */ #ifndef FFMPEG_ACELP_FILTERS_H #define FFMPEG_ACELP_FILTERS_H #include /** * 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); /** * Circularly convolve fixed vector with a phase dispersion impulse * response filter (D.6.2 of G.729 and 6.1.5 of AMR). * @param fc_out vector with filter applied * @param fc_in source vector * @param filter phase filter coefficients * * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] } * * \note fc_in and fc_out should not overlap! */ void ff_acelp_convolve_circ( int16_t* fc_out, const int16_t* fc_in, const int16_t* filter, int subframe_size); /** * LP synthesis filter. * @param out [out] pointer to output buffer * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) * @param in input signal * @param buffer_length amount of data to process * @param filter_length filter length (10 for 10th order LP filter) * @param stop_on_overflow 1 - return immediately if overflow occurs * 0 - ignore overflows * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) * * @return 1 if overflow occurred, 0 - otherwise * * @note Output buffer must contain 10 samples of past * speech data before pointer. * * Routine applies 1/A(z) filter to given speech data. */ int ff_acelp_lp_synthesis_filter( int16_t *out, const int16_t* filter_coeffs, const int16_t* in, int buffer_length, int filter_length, int stop_on_overflow, int rounder); /** * Calculates coefficients of weighted A(z/weight) filter. * @param out [out] weighted A(z/weight) result * filter (-0x8000 <= (3.12) < 0x8000) * @param in source filter (-0x8000 <= (3.12) < 0x8000) * @param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000) * @param filter_length filter length (11 for 10th order LP filter) * * out[i]=weight_pow[i]*in[i] , i=0..9 */ void ff_acelp_weighted_filter( int16_t *out, const int16_t* in, const int16_t *weight_pow, int filter_length); /** * high-pass filtering and upscaling (4.2.5 of G.729). * @param out [out] output buffer for filtered speech data * @param hpf_f [in/out] 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 100Hz * * @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); #endif /* FFMPEG_ACELP_FILTERS_H */