diff options
Diffstat (limited to 'libavcodec/acelp_filters.h')
-rw-r--r-- | libavcodec/acelp_filters.h | 43 |
1 files changed, 38 insertions, 5 deletions
diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h index b8715d2..56197bc 100644 --- a/libavcodec/acelp_filters.h +++ b/libavcodec/acelp_filters.h @@ -3,20 +3,20 @@ * * 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 */ @@ -25,6 +25,39 @@ #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. * @@ -76,7 +109,7 @@ void ff_acelp_interpolatef(float *out, const float *in, * * 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. |