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99497b4683
* commit '9a9e2f1c8aa4539a261625145e5c1f46a8106ac2': dsputil: Split audio operations off into a separate context Conflicts: configure libavcodec/takdec.c libavcodec/x86/Makefile libavcodec/x86/dsputil.asm libavcodec/x86/dsputil_init.c libavcodec/x86/dsputil_mmx.c libavcodec/x86/dsputil_x86.h Merged-by: Michael Niedermayer <michaelni@gmx.at>
117 lines
3.8 KiB
C
117 lines
3.8 KiB
C
/*
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* G.729, G729 Annex D postfilter
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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_G729POSTFILTER_H
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#define FFMPEG_G729POSTFILTER_H
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#include <stdint.h>
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#include "audiodsp.h"
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/**
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* tilt compensation factor (G.729, k1>0)
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* 0.2 in Q15
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*/
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#define G729_TILT_FACTOR_PLUS 6554
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/**
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* tilt compensation factor (G.729, k1<0)
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* 0.9 in Q15
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*/
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#define G729_TILT_FACTOR_MINUS 29491
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/* 4.2.2 */
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#define FORMANT_PP_FACTOR_NUM 18022 //0.55 in Q15
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#define FORMANT_PP_FACTOR_DEN 22938 //0.70 in Q15
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/**
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* gain adjustment factor (G.729, 4.2.4)
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* 0.9875 in Q15
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*/
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#define G729_AGC_FACTOR 32358
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#define G729_AGC_FAC1 (32768-G729_AGC_FACTOR)
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/**
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* 1.0 / (1.0 + 0.5) in Q15
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* where 0.5 is the minimum value of
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* weight factor, controlling amount of long-term postfiltering
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*/
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#define MIN_LT_FILT_FACTOR_A 21845
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/**
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* Short interpolation filter length
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*/
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#define SHORT_INT_FILT_LEN 2
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/**
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* Long interpolation filter length
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*/
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#define LONG_INT_FILT_LEN 8
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/**
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* Number of analyzed fractional pitch delays in second stage of long-term
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* postfilter
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*/
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#define ANALYZED_FRAC_DELAYS 7
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/**
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* Amount of past residual signal data stored in buffer
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*/
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#define RES_PREV_DATA_SIZE (PITCH_DELAY_MAX + LONG_INT_FILT_LEN + 1)
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/**
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* \brief Signal postfiltering (4.2)
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* \param dsp initialized DSP context
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* \param ht_prev_data [in/out] (Q12) pointer to variable receiving tilt
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* compensation filter data from previous subframe
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* \param voicing [in/out] (Q0) pointer to variable receiving voicing decision
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* \param lp_filter_coeffs (Q12) LP filter coefficients
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* \param pitch_delay_int integer part of the pitch delay
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* \param residual [in/out] (Q0) residual signal buffer (used in long-term postfilter)
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* \param res_filter_data [in/out] (Q0) speech data of previous subframe
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* \param pos_filter_data [in/out] (Q0) previous speech data for short-term postfilter
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* \param speech [in/out] (Q0) signal buffer
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* \param subframe_size size of subframe
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*
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* Filtering has the following stages:
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* Long-term postfilter (4.2.1)
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* Short-term postfilter (4.2.2).
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* Tilt-compensation (4.2.3)
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*/
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void ff_g729_postfilter(AudioDSPContext *adsp, int16_t* ht_prev_data, int* voicing,
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const int16_t *lp_filter_coeffs, int pitch_delay_int,
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int16_t* residual, int16_t* res_filter_data,
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int16_t* pos_filter_data, int16_t *speech,
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int subframe_size);
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/**
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* \brief Adaptive gain control (4.2.4)
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* \param gain_before (Q0) gain of speech before applying postfilters
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* \param gain_after (Q0) gain of speech after applying postfilters
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* \param speech [in/out] (Q0) signal buffer
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* \param subframe_size length of subframe
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* \param gain_prev (Q12) previous value of gain coefficient
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*
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* \return (Q12) last value of gain coefficient
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*/
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int16_t ff_g729_adaptive_gain_control(int gain_before, int gain_after, int16_t *speech,
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int subframe_size, int16_t gain_prev);
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#endif // FFMPEG_G729POSTFILTER_H
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