annotate celp_filters.h @ 11104:bb877c9cb102 libavcodec

Detect spatial direct MBs partitioned smaller than 16x16 that can be partitioned as 16x16 (except ones changing interlacing relative to the colocated MB). 20 cycles slower during MV generation 175 cycles faster during MC

/*
 * 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 */