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diff --git a/apps/codecs/libwmavoice/celp_filters.h b/apps/codecs/libwmavoice/celp_filters.h
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-/*
- * 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);
-
-/**
- * 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 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). 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 */