mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-18 03:19:31 +02:00
504eee37de
Based on AMR SoC code by Robert Swain and Colin McQuillan. Originally committed as revision 20392 to svn://svn.ffmpeg.org/ffmpeg/trunk
120 lines
4.6 KiB
C
120 lines
4.6 KiB
C
/*
|
|
* 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>
|
|
|
|
/**
|
|
* 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 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 1/80 of the sampling freq
|
|
*
|
|
* @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);
|
|
|
|
/**
|
|
* Apply an order 2 rational transfer function in-place.
|
|
*
|
|
* @param samples [in/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 *samples,
|
|
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 */
|