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Diffstat (limited to 'apps/codecs/libwmavoice/celp_filters.h')
-rw-r--r-- | apps/codecs/libwmavoice/celp_filters.h | 119 |
1 files changed, 0 insertions, 119 deletions
diff --git a/apps/codecs/libwmavoice/celp_filters.h b/apps/codecs/libwmavoice/celp_filters.h deleted file mode 100644 index 145e3d3346..0000000000 --- a/apps/codecs/libwmavoice/celp_filters.h +++ /dev/null @@ -1,119 +0,0 @@ -/* - * 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 */ |