yading@10: /*
yading@10:  * various filters for ACELP-based codecs
yading@10:  *
yading@10:  * Copyright (c) 2008 Vladimir Voroshilov
yading@10:  *
yading@10:  * This file is part of FFmpeg.
yading@10:  *
yading@10:  * FFmpeg is free software; you can redistribute it and/or
yading@10:  * modify it under the terms of the GNU Lesser General Public
yading@10:  * License as published by the Free Software Foundation; either
yading@10:  * version 2.1 of the License, or (at your option) any later version.
yading@10:  *
yading@10:  * FFmpeg is distributed in the hope that it will be useful,
yading@10:  * but WITHOUT ANY WARRANTY; without even the implied warranty of
yading@10:  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
yading@10:  * Lesser General Public License for more details.
yading@10:  *
yading@10:  * You should have received a copy of the GNU Lesser General Public
yading@10:  * License along with FFmpeg; if not, write to the Free Software
yading@10:  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
yading@10:  */
yading@10: 
yading@10: #ifndef AVCODEC_ACELP_FILTERS_H
yading@10: #define AVCODEC_ACELP_FILTERS_H
yading@10: 
yading@10: #include <stdint.h>
yading@10: 
yading@10: typedef struct ACELPFContext {
yading@10:     /**
yading@10:     * Floating point version of ff_acelp_interpolate()
yading@10:     */
yading@10:     void (*acelp_interpolatef)(float *out, const float *in,
yading@10:                             const float *filter_coeffs, int precision,
yading@10:                             int frac_pos, int filter_length, int length);
yading@10: 
yading@10:     /**
yading@10:      * Apply an order 2 rational transfer function in-place.
yading@10:      *
yading@10:      * @param out output buffer for filtered speech samples
yading@10:      * @param in input buffer containing speech data (may be the same as out)
yading@10:      * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
yading@10:      * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
yading@10:      * @param gain scale factor for final output
yading@10:      * @param mem intermediate values used by filter (should be 0 initially)
yading@10:      * @param n number of samples (should be a multiple of eight)
yading@10:      */
yading@10:     void (*acelp_apply_order_2_transfer_function)(float *out, const float *in,
yading@10:                                                   const float zero_coeffs[2],
yading@10:                                                   const float pole_coeffs[2],
yading@10:                                                   float gain,
yading@10:                                                   float mem[2], int n);
yading@10: 
yading@10: }ACELPFContext;
yading@10: 
yading@10: /**
yading@10:  * Initialize ACELPFContext.
yading@10:  */
yading@10: void ff_acelp_filter_init(ACELPFContext *c);
yading@10: void ff_acelp_filter_init_mips(ACELPFContext *c);
yading@10: 
yading@10: /**
yading@10:  * low-pass Finite Impulse Response filter coefficients.
yading@10:  *
yading@10:  * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
yading@10:  * the coefficients are scaled by 2^15.
yading@10:  * This array only contains the right half of the filter.
yading@10:  * This filter is likely identical to the one used in G.729, though this
yading@10:  * could not be determined from the original comments with certainty.
yading@10:  */
yading@10: extern const int16_t ff_acelp_interp_filter[61];
yading@10: 
yading@10: /**
yading@10:  * Generic FIR interpolation routine.
yading@10:  * @param[out] out buffer for interpolated data
yading@10:  * @param in input data
yading@10:  * @param filter_coeffs interpolation filter coefficients (0.15)
yading@10:  * @param precision sub sample factor, that is the precision of the position
yading@10:  * @param frac_pos fractional part of position [0..precision-1]
yading@10:  * @param filter_length filter length
yading@10:  * @param length length of output
yading@10:  *
yading@10:  * filter_coeffs contains coefficients of the right half of the symmetric
yading@10:  * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
yading@10:  * See ff_acelp_interp_filter for an example.
yading@10:  *
yading@10:  */
yading@10: void ff_acelp_interpolate(int16_t* out, const int16_t* in,
yading@10:                           const int16_t* filter_coeffs, int precision,
yading@10:                           int frac_pos, int filter_length, int length);
yading@10: 
yading@10: /**
yading@10:  * Floating point version of ff_acelp_interpolate()
yading@10:  */
yading@10: void ff_acelp_interpolatef(float *out, const float *in,
yading@10:                            const float *filter_coeffs, int precision,
yading@10:                            int frac_pos, int filter_length, int length);
yading@10: 
yading@10: 
yading@10: /**
yading@10:  * high-pass filtering and upscaling (4.2.5 of G.729).
yading@10:  * @param[out]     out   output buffer for filtered speech data
yading@10:  * @param[in,out]  hpf_f past filtered data from previous (2 items long)
yading@10:  *                       frames (-0x20000000 <= (14.13) < 0x20000000)
yading@10:  * @param in speech data to process
yading@10:  * @param length input data size
yading@10:  *
yading@10:  * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
yading@10:  *          1.9330735 * out[i-1] - 0.93589199 * out[i-2]
yading@10:  *
yading@10:  * The filter has a cut-off frequency of 1/80 of the sampling freq
yading@10:  *
yading@10:  * @note Two items before the top of the in buffer must contain two items from the
yading@10:  *       tail of the previous subframe.
yading@10:  *
yading@10:  * @remark It is safe to pass the same array in in and out parameters.
yading@10:  *
yading@10:  * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
yading@10:  *         but constants differs in 5th sign after comma). Fortunately in
yading@10:  *         fixed-point all coefficients are the same as in G.729. Thus this
yading@10:  *         routine can be used for the fixed-point AMR decoder, too.
yading@10:  */
yading@10: void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
yading@10:                                const int16_t* in, int length);
yading@10: 
yading@10: /**
yading@10:  * Apply an order 2 rational transfer function in-place.
yading@10:  *
yading@10:  * @param out output buffer for filtered speech samples
yading@10:  * @param in input buffer containing speech data (may be the same as out)
yading@10:  * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
yading@10:  * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
yading@10:  * @param gain scale factor for final output
yading@10:  * @param mem intermediate values used by filter (should be 0 initially)
yading@10:  * @param n number of samples
yading@10:  */
yading@10: void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
yading@10:                                               const float zero_coeffs[2],
yading@10:                                               const float pole_coeffs[2],
yading@10:                                               float gain,
yading@10:                                               float mem[2], int n);
yading@10: 
yading@10: /**
yading@10:  * Apply tilt compensation filter, 1 - tilt * z-1.
yading@10:  *
yading@10:  * @param mem pointer to the filter's state (one single float)
yading@10:  * @param tilt tilt factor
yading@10:  * @param samples array where the filter is applied
yading@10:  * @param size the size of the samples array
yading@10:  */
yading@10: void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
yading@10: 
yading@10: 
yading@10: #endif /* AVCODEC_ACELP_FILTERS_H */