view acelp_filters.h @ 7744:7477cbdacb20 libavcodec

Fix lossless jpeg encoder to comply to spec and store full redundant residuals, Note this does not change RGB32 as we need to check this against some decoder that supports it.

/*
 * various filters for ACELP-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 FFMPEG_ACELP_FILTERS_H
#define FFMPEG_ACELP_FILTERS_H

#include <stdint.h>

/**
 * low-pass Finite Impulse Response filter coefficients.
 *
 * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
 * the coefficients are scaled by 2^15.
 * This array only contains the right half of the filter.
 * This filter is likely identical to the one used in G.729, though this
 * could not be determined from the original comments with certainity.
 */
extern const int16_t ff_acelp_interp_filter[61];

/**
 * Generic FIR interpolation routine.
 * @param out [out] buffer for interpolated data
 * @param in input data
 * @param filter_coeffs interpolation filter coefficients (0.15)
 * @param precision sub sample factor, that is the precision of the position
 * @param frac_pos fractional part of position [0..precision-1]
 * @param filter_length filter length
 * @param length length of output
 *
 * filter_coeffs contains coefficients of the right half of the symmetric
 * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
 * See ff_acelp_interp_filter for an example.
 *
 */
void ff_acelp_interpolate(
        int16_t* out,
        const int16_t* in,
        const int16_t* filter_coeffs,
        int precision,
        int frac_pos,
        int filter_length,
        int length);

/**
 * 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_acelp_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 10 samples of past
 *       speech data before pointer.
 *
 * Routine applies 1/A(z) filter to given speech data.
 */
int ff_acelp_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);


/**
 * high-pass filtering and upscaling (4.2.5 of G.729).
 * @param out [out] output buffer for filtered speech data
 * @param hpf_f [in/out] past filtered data from previous (2 items long)
 *                       frames (-0x20000000 <= (14.13) < 0x20000000)
 * @param in speech data to process
 * @param length input data size
 *
 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
 *          1.9330735 * out[i-1] - 0.93589199 * out[i-2]
 *
 * 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
 *       tail of the previous subframe.
 *
 * @remark It is safe to pass the same array in in and out parameters.
 *
 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
 *         but constants differs in 5th sign after comma). Fortunately in
 *         fixed-point all coefficients are the same as in G.729. Thus this
 *         routine can be used for the fixed-point AMR decoder, too.
 */
void ff_acelp_high_pass_filter(
        int16_t* out,
        int hpf_f[2],
        const int16_t* in,
        int length);

#endif /* FFMPEG_ACELP_FILTERS_H */