mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-18 03:19:31 +02:00
117 lines
3.7 KiB
C
117 lines
3.7 KiB
C
/*
|
|
* 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>
|
|
|
|
/**
|
|
* 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);
|
|
|
|
/**
|
|
* 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 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 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)
|
|
*
|
|
* @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 */
|