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diff --git a/lib/rbcodec/codecs/libwmavoice/celp_filters.h b/lib/rbcodec/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 */