Split off celp_filters.[ch] from acelp_filters.[ch] for the QCELP decoder.
patch by Kenan Gillet, kenan.gillet gmail com Originally committed as revision 15680 to svn://svn.ffmpeg.org/ffmpeg/trunk
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@ -156,7 +156,7 @@ OBJS-$(CONFIG_QDRAW_DECODER) += qdrw.o
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OBJS-$(CONFIG_QPEG_DECODER) += qpeg.o
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OBJS-$(CONFIG_QTRLE_DECODER) += qtrle.o
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OBJS-$(CONFIG_QTRLE_ENCODER) += qtrleenc.o
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OBJS-$(CONFIG_RA_144_DECODER) += ra144.o acelp_filters.o
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OBJS-$(CONFIG_RA_144_DECODER) += ra144.o celp_filters.o
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OBJS-$(CONFIG_RA_288_DECODER) += ra288.o
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OBJS-$(CONFIG_RAWVIDEO_DECODER) += rawdec.o
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OBJS-$(CONFIG_RAWVIDEO_ENCODER) += rawenc.o
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@ -81,65 +81,6 @@ void ff_acelp_interpolate(
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}
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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 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_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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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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void ff_acelp_high_pass_filter(
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int16_t* out,
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@ -60,50 +60,6 @@ void ff_acelp_interpolate(
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int filter_length,
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int length);
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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_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 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_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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int rounder);
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/**
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* high-pass filtering and upscaling (4.2.5 of G.729).
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86
libavcodec/celp_filters.c
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86
libavcodec/celp_filters.c
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@ -0,0 +1,86 @@
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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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72
libavcodec/celp_filters.h
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72
libavcodec/celp_filters.h
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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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@ -25,7 +25,7 @@
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#include "avcodec.h"
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#include "bitstream.h"
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#include "ra144.h"
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#include "acelp_filters.h"
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#include "celp_filters.h"
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#define NBLOCKS 4 ///< number of subblocks within a block
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#define BLOCKSIZE 40 ///< subblock size in 16-bit words
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@ -201,8 +201,8 @@ static void do_output_subblock(RA144Context *ractx, const uint16_t *lpc_coefs,
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memcpy(ractx->curr_sblock, ractx->curr_sblock + 40,
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10*sizeof(*ractx->curr_sblock));
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if (ff_acelp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs,
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block, BLOCKSIZE, 10, 1, 0xfff))
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if (ff_celp_lp_synthesis_filter(ractx->curr_sblock + 10, lpc_coefs,
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block, BLOCKSIZE, 10, 1, 0xfff))
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memset(ractx->curr_sblock, 0, 50*sizeof(*ractx->curr_sblock));
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}
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