nest-open-source / manifest_repos / ffmpeg / 62cf23d89db2251203d6207e2261ddc5498167c6 / . / libavcodec / acelp_filters.h

/* | |

* 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 certainty. | |

*/ | |

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 */ |