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