/*
* various filters for CELP-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_CELP_FILTERS_H
#define AVCODEC_CELP_FILTERS_H
#include
/**
* 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 */