987903826b
Consistently apply this rule: the guard name is obtained from the filename by stripping the leading "lib", converting '/' and '.' to '_' and uppercasing the resulting name. Guard names in the root directory have to be prefixed by "FFMPEG_". Originally committed as revision 15120 to svn://svn.ffmpeg.org/ffmpeg/trunk
138 lines
4.7 KiB
C
138 lines
4.7 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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/**
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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 certainity.
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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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*/
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void ff_acelp_interpolate(
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int16_t* out,
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const int16_t* in,
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const int16_t* filter_coeffs,
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int precision,
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int frac_pos,
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int filter_length,
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int length);
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/**
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* Circularly convolve fixed vector with a phase dispersion impulse
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* response filter (D.6.2 of G.729 and 6.1.5 of AMR).
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* @param fc_out vector with filter applied
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* @param fc_in source vector
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* @param filter phase filter coefficients
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*
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* fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
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*
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* \note fc_in and fc_out should not overlap!
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*/
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void ff_acelp_convolve_circ(
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int16_t* fc_out,
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const int16_t* fc_in,
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const int16_t* filter,
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int len);
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/**
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* LP synthesis filter.
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* @param out [out] pointer to output buffer
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* @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
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* @param in input signal
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* @param buffer_length amount of data to process
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* @param filter_length filter length (10 for 10th order LP filter)
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* @param stop_on_overflow 1 - return immediately if overflow occurs
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* 0 - ignore overflows
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* @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
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*
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* @return 1 if overflow occurred, 0 - otherwise
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*
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* @note Output buffer must contain 10 samples of past
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* speech data before pointer.
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*
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* Routine applies 1/A(z) filter to given speech data.
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*/
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int ff_acelp_lp_synthesis_filter(
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int16_t *out,
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const int16_t* filter_coeffs,
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const int16_t* in,
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int buffer_length,
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int filter_length,
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int stop_on_overflow,
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int rounder);
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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 hpf_f [in/out] 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 out 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(
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int16_t* out,
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int hpf_f[2],
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const int16_t* in,
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int length);
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#endif /* AVCODEC_ACELP_FILTERS_H */
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