*
* Copyright (c) 2008 Vladimir Voroshilov
*
- * This file is part of Libav.
+ * This file is part of FFmpeg.
*
- * Libav is free software; you can redistribute it and/or
+ * 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.
*
- * Libav is distributed in the hope that it will be useful,
+ * 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 Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
+typedef struct ACELPFContext {
+ /**
+ * Floating point version of ff_acelp_interpolate()
+ */
+ void (*acelp_interpolatef)(float *out, const float *in,
+ const float *filter_coeffs, int precision,
+ int frac_pos, int filter_length, int length);
+
+ /**
+ * Apply an order 2 rational transfer function in-place.
+ *
+ * @param out output buffer for filtered speech samples
+ * @param in input buffer containing speech data (may be the same as out)
+ * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
+ * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
+ * @param gain scale factor for final output
+ * @param mem intermediate values used by filter (should be 0 initially)
+ * @param n number of samples (should be a multiple of eight)
+ */
+ void (*acelp_apply_order_2_transfer_function)(float *out, const float *in,
+ const float zero_coeffs[2],
+ const float pole_coeffs[2],
+ float gain,
+ float mem[2], int n);
+
+}ACELPFContext;
+
+/**
+ * Initialize ACELPFContext.
+ */
+void ff_acelp_filter_init(ACELPFContext *c);
+void ff_acelp_filter_init_mips(ACELPFContext *c);
+
/**
* low-pass Finite Impulse Response filter coefficients.
*
*
* The filter has a cut-off frequency of 1/80 of the sampling freq
*
- * @note Two items before the top of the out buffer must contain two items from the
+ * @note Two items before the top of the in buffer must contain two items from the
* tail of the previous subframe.
*
* @remark It is safe to pass the same array in in and out parameters.