annotate celp_filters.h @ 9473:e38284cd69dc libavcodec

Use memcpy instead of the very inefficient bytecopy where both are correct (i.e. no overlap of src and dst is possible).

/*
 * 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);

/**
 * 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)
 *
 * @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);

#endif /* AVCODEC_CELP_FILTERS_H */