/* * various filters for CELP-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 AVCODEC_CELP_FILTERS_H #define AVCODEC_CELP_FILTERS_H #include typedef struct CELPFContext { /** * LP synthesis filter. * @param[out] out pointer to output buffer * - the array out[-filter_length, -1] must * contain the previous result of this filter * @param filter_coeffs filter coefficients. * @param in input signal * @param buffer_length amount of data to process * @param filter_length filter length (10 for 10th order LP filter). Must be * greater than 4 and even. * * @note Output buffer must contain filter_length samples of past * speech data before pointer. * * Routine applies 1/A(z) filter to given speech data. */ void (*celp_lp_synthesis_filterf)(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length); /** * LP zero synthesis filter. * @param[out] out pointer to output buffer * @param filter_coeffs filter coefficients. * @param in input signal * - the array in[-filter_length, -1] must * contain the previous input of this filter * @param buffer_length amount of data to process (should be a multiple of eight) * @param filter_length filter length (10 for 10th order LP filter; * should be a multiple of two) * * @note Output buffer must contain filter_length samples of past * speech data before pointer. * * Routine applies A(z) filter to given speech data. */ void (*celp_lp_zero_synthesis_filterf)(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length); }CELPFContext; /** * Initialize CELPFContext. */ void ff_celp_filter_init(CELPFContext *c); void ff_celp_filter_init_mips(CELPFContext *c); /** * 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_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in, const int16_t *filter, int len); /** * Add an array to a rotated array. * * out[k] = in[k] + fac * lagged[k-lag] with wrap-around * * @param out result vector * @param in samples to be added unfiltered * @param lagged samples to be rotated, multiplied and added * @param lag lagged vector delay in the range [0, n] * @param fac scalefactor for lagged samples * @param n number of samples */ void ff_celp_circ_addf(float *out, const float *in, const float *lagged, int lag, float fac, int n); /** * 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 shift the result is shifted right by this value * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) * * @return 1 if overflow occurred, 0 - otherwise * * @note Output buffer must contain filter_length samples of past * speech data before pointer. * * Routine applies 1/A(z) filter to given speech data. */ int ff_celp_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 shift, int rounder); /** * LP synthesis filter. * @param[out] out pointer to output buffer * - the array out[-filter_length, -1] must * contain the previous result of this filter * @param filter_coeffs filter coefficients. * @param in input signal * @param buffer_length amount of data to process * @param filter_length filter length (10 for 10th order LP filter). Must be * greater than 4 and even. * * @note Output buffer must contain filter_length samples of past * speech data before pointer. * * Routine applies 1/A(z) filter to given speech data. */ void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length); /** * LP zero synthesis filter. * @param[out] out pointer to output buffer * @param filter_coeffs filter coefficients. * @param in input signal * - the array in[-filter_length, -1] must * contain the previous input of this filter * @param buffer_length amount of data to process * @param filter_length filter length (10 for 10th order LP filter) * * @note Output buffer must contain filter_length samples of past * speech data before pointer. * * Routine applies A(z) filter to given speech data. */ void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs, const float *in, int buffer_length, int filter_length); #endif /* AVCODEC_CELP_FILTERS_H */