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ffmpeg / libavcodec / acelp_filters.h @ 40a655e3

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/*
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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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 * \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 (11 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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 *
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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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/**
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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