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542c064d1b
fix missing period (not latest revision of patch was wrongly committed). Originally committed as revision 13113 to svn://svn.ffmpeg.org/ffmpeg/trunk
125 lines
3.5 KiB
C
125 lines
3.5 KiB
C
/*
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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 "acelp_filters.h"
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#define FRAC_BITS 13
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#include "mathops.h"
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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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int i, k;
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memset(fc_out, 0, subframe_size * 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 swap two loops and process
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non-zero samples only. In the case of G.729D the buffer contains
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two non-zero samples before the call to ff_acelp_enhance_harmonics
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and, due to pitch_delay being bounded by [20; 143], a maximum
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of four non-zero samples for a total of 40 after the call. */
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for(i=0; i<subframe_size; 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[subframe_size + k - i]) >> 15;
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for(k=i; k<subframe_size; 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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{
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int i,n;
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for(n=0; n<buffer_length; n++)
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{
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int sum = 0x800;
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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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/* Check for overflow */
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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_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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int n;
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for(n=0; n<filter_length; n++)
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out[n] = (in[n] * weight_pow[n] + 0x4000) >> 15; /* (3.12) = (0.15) * (3.12) with rounding */
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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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{
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int i;
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int tmp;
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for(i=0; i<length; i++)
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{
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tmp = MULL(hpf_f[0], 15836); /* (14.13) = (13.13) * (1.13) */
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tmp += MULL(hpf_f[1], -7667); /* (13.13) = (13.13) * (0.13) */
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tmp += 7699 * (in[i] - 2*in[i-1] + in[i-2]); /* (14.13) = (0.13) * (14.0) */
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/* Multiplication by 2 with rounding can cause short type
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overflow, thus clipping is required. */
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out[i] = av_clip_int16((tmp + 0x800) >> 12); /* (15.0) = 2 * (13.13) = (14.13) */
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hpf_f[1] = hpf_f[0];
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hpf_f[0] = tmp;
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}
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}
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