nest-open-source / nest-android-app / ffmpeg / 8e1cc55b733ae723de9931f20a13731bed8162f5 / . / libavcodec / celp_filters.h

/* | |

* various filters for CELP-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_CELP_FILTERS_H | |

#define AVCODEC_CELP_FILTERS_H | |

#include <stdint.h> | |

typedef struct CELPFContext { | |

/** | |

* 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 (*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 (should be a multiple of eight) | |

* @param filter_length filter length (10 for 10th order LP filter; | |

* should be a multiple of two) | |

* | |

* @note Output buffer must contain filter_length samples of past | |

* speech data before pointer. | |

* | |

* Routine applies A(z) filter to given speech data. | |

*/ | |

void (*celp_lp_zero_synthesis_filterf)(float *out, const float *filter_coeffs, | |

const float *in, int buffer_length, | |

int filter_length); | |

}CELPFContext; | |

/** | |

* Initialize CELPFContext. | |

*/ | |

void ff_celp_filter_init(CELPFContext *c); | |

void ff_celp_filter_init_mips(CELPFContext *c); | |

/** | |

* 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 */ |