Mercurial > libavcodec.hg
view acelp_filters.h @ 7016:b514562e0aef libavcodec
set default mix levels regardless of bitstream id
author | jbr |
---|---|
date | Sat, 07 Jun 2008 22:29:45 +0000 |
parents | 94465a2c3b34 |
children | 2b763a495c07 |
line wrap: on
line source
/* * 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 FFMPEG_ACELP_FILTERS_H #define FFMPEG_ACELP_FILTERS_H #include <stdint.h> /** * low-pass FIR (Finite Impulse Response) filter coefficients * * A similar filter is named b30 in G.729. * * G.729 specification says: * b30 is based on Hamming windowed sinc functions, truncated at +/-29 and * padded with zeros at +/-30 b30[30]=0. * The filter has a cut-off frequency (-3 dB) at 3600 Hz in the oversampled * domain. * * After some analysis, I found this approximation: * * PI * x * Hamm(x,N) = 0.53836-0.46164*cos(--------) * N-1 * --- * 2 * * PI * x * Hamm'(x,k) = Hamm(x - k, 2*k+1) = 0.53836 + 0.46164*cos(--------) * k * * sin(PI * x) * Sinc(x) = ----------- (normalized sinc function) * PI * x * * h(t,B) = 2 * B * Sinc(2 * B * t) (impulse response of sinc low-pass filter) * * b(k,B, n) = Hamm'(n, k) * h(n, B) * * * 3600 * B = ---- * 8000 * * 3600 - cut-off frequency * 8000 - sampling rate * k - filter order * * ff_acelp_interp_filter[6*i+j] = b(10, 3600/8000, i+j/6) * * The filter assumes the following order of fractions (X - integer delay): * * 1/3 precision: X 1/3 2/3 X 1/3 2/3 X * 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 * * The filter can be used for 1/3 precision, too, by * passing 2*pitch_delay_frac as third parameter to the interpolation routine. * */ extern const int16_t ff_acelp_interp_filter[61]; /** * \brief Generic interpolation routine * \param out [out] buffer for interpolated data * \param in input data * \param filter_coeffs interpolation filter coefficients (0.15) * \param precision filter is able to interpolate with 1/precision precision of pitch delay * \param pitch_delay_frac pitch delay, fractional part [0..precision-1] * \param filter_length filter length * \param length length of speech data to process * * filter_coeffs contains coefficients of the positive half of the symmetric * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient. * See ff_acelp_interp_filter fot example. * */ void ff_acelp_interpolate( int16_t* out, const int16_t* in, const int16_t* filter_coeffs, int precision, int pitch_delay_frac, int filter_length, int length); /** * \brief Circularly convolve fixed vector with a phase dispersion impulse * response filter (D.6.2 of G.729 and 6.1.5 of AMR). * \param fc_out vector with filter applied * \param fc_in source vector * \param filter phase filter coefficients * * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] } * * \note fc_in and fc_out should not overlap! */ void ff_acelp_convolve_circ( int16_t* fc_out, const int16_t* fc_in, const int16_t* filter, int subframe_size); /** * \brief LP synthesis filter * \param out [out] pointer to output buffer * \param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) * \param in input signal * \param buffer_length amount of data to process * \param filter_length filter length (11 for 10th order LP filter) * \param stop_on_overflow 1 - return immediately if overflow occurs * 0 - ignore overflows * * \return 1 if overflow occurred, 0 - otherwise * * \note Output buffer must contain 10 samples of past * speech data before pointer. * * Routine applies 1/A(z) filter to given speech data. */ int ff_acelp_lp_synthesis_filter( int16_t *out, const int16_t* filter_coeffs, const int16_t* in, int buffer_length, int filter_length, int stop_on_overflow); /** * \brief Calculates coefficients of weighted A(z/weight) filter. * \param out [out] weighted A(z/weight) result * filter (-0x8000 <= (3.12) < 0x8000) * \param in source filter (-0x8000 <= (3.12) < 0x8000) * \param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000) * \param filter_length filter length (11 for 10th order LP filter) * * out[i]=weight_pow[i]*in[i] , i=0..9 */ void ff_acelp_weighted_filter( int16_t *out, const int16_t* in, const int16_t *weight_pow, int filter_length); /** * \brief high-pass filtering and upscaling (4.2.5 of G.729) * \param out [out] output buffer for filtered speech data * \param hpf_f [in/out] 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 100Hz * * \note Two items before the top of the out 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); #endif /* FFMPEG_ACELP_FILTERS_H */