diff --git a/libavcodec/acelp_filters.c b/libavcodec/acelp_filters.c new file mode 100644 index 0000000000..b98c6f741e --- /dev/null +++ b/libavcodec/acelp_filters.c @@ -0,0 +1,125 @@ +/* + * 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 + */ + +#include + +#include "avcodec.h" +#include "acelp_filters.h" +#define FRAC_BITS 13 +#include "mathops.h" + +void ff_acelp_convolve_circ( + int16_t* fc_out, + const int16_t* fc_in, + const int16_t* filter, + int subframe_size) +{ + int i, k; + + memset(fc_out, 0, subframe_size * sizeof(int16_t)); + + /* Since there are few pulses over entire subframe (i.e. almost all + fc_in[i] are zero, in case of G.729D the buffer contains two non-zero + samples before the call to ff_acelp_enhance_harmonics, and (due to + pitch_delay bounded to [20; 143]) a maximum four non-zero samples + for a total of 40 after the call to it), it is faster to swap two loops + and process non-zero samples only. This will reduce the number of + multiplications from 40*40 to 4*40 for G.729D */ + for(i=0; i> 15; + + for(k=i; k> 15; + } + } +} + +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) +{ + int i,n; + + for(n=0; n> 12) + in[n]; + + /* Check for overflow */ + if(sum + 0x8000 > 0xFFFFU) + { + if(stop_on_overflow) + return 1; + sum = (sum >> 31) ^ 32767; + } + out[n] = sum; + } + + return 0; +} + +void ff_acelp_weighted_filter( + int16_t *out, + const int16_t* in, + const int16_t *weight_pow, + int filter_length) +{ + int n; + for(n=0; n> 15; /* (3.12) = (0.15) * (3.12) with rounding */ +} + +void ff_acelp_high_pass_filter( + int16_t* out, + int hpf_f[2], + const int16_t* in, + int length) +{ + int i; + int tmp; + + for(i=0; i> 12); /* (15.0) = 2 * (13.13) = (14.13) */ + + hpf_f[1] = hpf_f[0]; + hpf_f[0] = tmp; + } +} diff --git a/libavcodec/acelp_filters.h b/libavcodec/acelp_filters.h new file mode 100644 index 0000000000..155a847a77 --- /dev/null +++ b/libavcodec/acelp_filters.h @@ -0,0 +1,113 @@ +/* + * 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 + +/** + * \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