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/*
* 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 <stdint.h>
/**
* \brief 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);
/**
* \brief 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 (11 for 10th order LP filter)
* \param stop_on_overflow 1 - return immediately if overflow occurs
* 0 - ignore overflows
*
* \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);
/**
* \brief 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);
/**
* \brief 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 */
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