mirror of https://github.com/FFmpeg/FFmpeg.git
patch by Kenan Gillet, kenan.gillet gmail com Originally committed as revision 15680 to svn://svn.ffmpeg.org/ffmpeg/trunkpull/126/head
Kenan Gillet
16 years ago
committed by
Diego Biurrun
parent
60c25a4beb
commit
4599d22c0c
6 changed files with 162 additions and 107 deletions
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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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#include <inttypes.h> |
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#include "avcodec.h" |
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#include "celp_filters.h" |
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void ff_celp_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 len) |
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{ |
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int i, k; |
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memset(fc_out, 0, len * sizeof(int16_t)); |
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/* Since there are few pulses over an entire subframe (i.e. almost
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all fc_in[i] are zero) it is faster to loop over fc_in first. */ |
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for(i=0; i<len; i++) |
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{ |
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if(fc_in[i]) |
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{ |
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for(k=0; k<i; k++) |
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fc_out[k] += (fc_in[i] * filter[len + k - i]) >> 15; |
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for(k=i; k<len; k++) |
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fc_out[k] += (fc_in[i] * filter[ k - i]) >> 15; |
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} |
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} |
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} |
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int ff_celp_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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int rounder) |
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{ |
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int i,n; |
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// These two lines are to avoid a -1 subtraction in the main loop
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filter_length++; |
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filter_coeffs--; |
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for(n=0; n<buffer_length; n++) |
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{ |
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int sum = rounder; |
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for(i=1; i<filter_length; i++) |
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sum -= filter_coeffs[i] * out[n-i]; |
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sum = (sum >> 12) + in[n]; |
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if(sum + 0x8000 > 0xFFFFU) |
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{ |
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if(stop_on_overflow) |
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return 1; |
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sum = (sum >> 31) ^ 32767; |
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} |
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out[n] = sum; |
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} |
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return 0; |
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} |
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@ -0,0 +1,72 @@ |
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/*
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* various filters for CELP-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 AVCODEC_CELP_FILTERS_H |
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#define AVCODEC_CELP_FILTERS_H |
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#include <stdint.h> |
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/**
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* 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_celp_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 len); |
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/**
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* 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 (10 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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* @param rounder the amount to add for rounding (usually 0x800 or 0xfff) |
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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_celp_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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int rounder); |
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#endif /* AVCODEC_CELP_FILTERS_H */ |
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