Subversion Repositories Kolibri OS

/*

 * 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 AVCODEC_ACELP_FILTERS_H

#define AVCODEC_ACELP_FILTERS_H

#include 

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.

 *

 * 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 certainty.

 */

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);

/**

 * Floating point version of ff_acelp_interpolate()

 */

void ff_acelp_interpolatef(float *out, const float *in,

                           const float *filter_coeffs, int precision,

                           int frac_pos, int filter_length, int length);

/**

 * high-pass filtering and upscaling (4.2.5 of G.729).

 * @param[out]     out   output buffer for filtered speech data

 * @param[in,out]  hpf_f 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 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.

 *

 * @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);

/**

 * 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

 */

void ff_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);

/**

 * Apply tilt compensation filter, 1 - tilt * z-1.

 *

 * @param mem pointer to the filter's state (one single float)

 * @param tilt tilt factor

 * @param samples array where the filter is applied

 * @param size the size of the samples array

 */

void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);

#endif /* AVCODEC_ACELP_FILTERS_H */