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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 AVCODEC_ACELP_FILTERS_H
#define AVCODEC_ACELP_FILTERS_H
#include <stdint.h>
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 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);
/**
* 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);
#endif /* AVCODEC_ACELP_FILTERS_H */
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