Mercurial > libavcodec.hg
view acelp_filters.h @ 6776:4b9f28275b14 libavcodec
Document AVCodecContext channels field.
patch by Stefano Sabatini, stefano.sabatini-lala poste it
author | diego |
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date | Sun, 11 May 2008 11:15:18 +0000 |
parents | 1f02f929b9ff |
children | b03cd8d29d60 |
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/* * 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