0,0 → 1,169 |
/* |
* 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> |
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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); |
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#endif /* AVCODEC_CELP_FILTERS_H */ |