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195 lines
6.6 KiB
C
195 lines
6.6 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 FFMPEG_ACELP_FILTERS_H
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#define FFMPEG_ACELP_FILTERS_H
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#include <stdint.h>
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/**
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* low-pass FIR (Finite Impulse Response) filter coefficients
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*
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* A similar filter is named b30 in G.729.
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*
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* G.729 specification says:
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* b30 is based on Hamming windowed sinc functions, truncated at +/-29 and
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* padded with zeros at +/-30 b30[30]=0.
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* The filter has a cut-off frequency (-3 dB) at 3600 Hz in the oversampled
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* domain.
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*
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* After some analysis, I found this approximation:
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*
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* PI * x
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* Hamm(x,N) = 0.53836-0.46164*cos(--------)
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* N-1
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* ---
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* 2
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*
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* PI * x
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* Hamm'(x,k) = Hamm(x - k, 2*k+1) = 0.53836 + 0.46164*cos(--------)
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* k
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*
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* sin(PI * x)
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* Sinc(x) = ----------- (normalized sinc function)
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* PI * x
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*
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* h(t,B) = 2 * B * Sinc(2 * B * t) (impulse response of sinc low-pass filter)
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*
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* b(k,B, n) = Hamm'(n, k) * h(n, B)
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*
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*
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* 3600
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* B = ----
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* 8000
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*
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* 3600 - cut-off frequency
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* 8000 - sampling rate
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* k - filter order
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*
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* ff_acelp_interp_filter[6*i+j] = b(10, 3600/8000, i+j/6)
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*
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* The filter assumes the following order of fractions (X - integer delay):
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*
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* 1/3 precision: X 1/3 2/3 X 1/3 2/3 X
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* 1/6 precision: X 1/6 2/6 3/6 4/6 5/6 X 1/6 2/6 3/6 4/6 5/6 X
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*
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* The filter can be used for 1/3 precision, too, by
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* passing 2*pitch_delay_frac as third parameter to the interpolation routine.
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*
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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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* \brief Generic 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 filter is able to interpolate with 1/precision precision of pitch delay
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* \param pitch_delay_frac pitch delay, fractional part [0..precision-1]
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* \param filter_length filter length
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* \param length length of speech data to process
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*
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* filter_coeffs contains coefficients of the positive 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 fot 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 pitch_delay_frac,
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int filter_length,
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int length);
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/**
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* \brief 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 subframe_size);
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/**
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* \brief 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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* \brief Calculates coefficients of weighted A(z/weight) filter.
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* \param out [out] weighted A(z/weight) result
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* filter (-0x8000 <= (3.12) < 0x8000)
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* \param in source filter (-0x8000 <= (3.12) < 0x8000)
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* \param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000)
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* \param filter_length filter length (11 for 10th order LP filter)
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*
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* out[i]=weight_pow[i]*in[i] , i=0..9
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*/
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void ff_acelp_weighted_filter(
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int16_t *out,
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const int16_t* in,
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const int16_t *weight_pow,
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int filter_length);
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/**
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* \brief 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 100Hz
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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 /* FFMPEG_ACELP_FILTERS_H */
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