ffmpeg / libavcodec / acelp_filters.c @ 8955a9d7
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/*


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* various filters for ACELPbased 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 021101301 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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const int16_t ff_acelp_interp_filter[61] = { /* (0.15) */ 
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29443, 28346, 25207, 20449, 14701, 8693, 
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3143, 1352, 4402, 5865, 5850, 4673, 
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2783, 672, 1211, 2536, 3130, 2991, 
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2259, 1170, 0, 1001, 1652, 1868, 
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1666, 1147, 464, 218, 756, 1060, 
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1099, 904, 550, 135, 245, 514, 
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634, 602, 451, 231, 0, 191, 
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308, 340, 296, 198, 78, 36, 
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120, 163, 165, 132, 79, 19, 
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34, 73, 91, 89, 70, 38, 
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0,

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}; 
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void ff_acelp_interpolate(int16_t* out, const int16_t* in, 
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const int16_t* filter_coeffs, int precision, 
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int frac_pos, int filter_length, int length) 
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{ 
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int n, i;

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assert(frac_pos >= 0 && frac_pos < precision);

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for (n = 0; n < length; n++) { 
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int idx = 0; 
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int v = 0x4000; 
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for (i = 0; i < filter_length;) { 
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/* The reference G.729 and AMR fixed point code performs clipping after

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each of the two following accumulations.

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Since clipping affects only the synthetic OVERFLOW test without

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causing an int type overflow, it was moved outside the loop. */

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/* R(x):=ac_v[k+x]

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v += R(ni)*ff_acelp_interp_filter(t+6i)

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v += R(n+i+1)*ff_acelp_interp_filter(6t+6i) */

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v += in[n + i] * filter_coeffs[idx + frac_pos]; 
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idx += precision; 
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i++; 
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v += in[n  i] * filter_coeffs[idx  frac_pos]; 
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} 
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if (av_clip_int16(v >> 15) != (v >> 15)) 
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av_log(NULL, AV_LOG_WARNING, "overflow that would need cliping in ff_acelp_interpolate()\n"); 
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out[n] = v >> 15;

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} 
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} 
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void ff_acelp_interpolatef(float *out, const float *in, 
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const float *filter_coeffs, int precision, 
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int frac_pos, int filter_length, int length) 
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{ 
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int n, i;

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for (n = 0; n < length; n++) { 
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int idx = 0; 
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float v = 0; 
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for (i = 0; i < filter_length;) { 
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v += in[n + i] * filter_coeffs[idx + frac_pos]; 
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idx += precision; 
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i++; 
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v += in[n  i] * filter_coeffs[idx  frac_pos]; 
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} 
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out[n] = v; 
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} 
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} 
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void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2], 
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const int16_t* in, 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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tmp = (hpf_f[0]* 15836LL) >> 13; 
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tmp += (hpf_f[1]* 7667LL) >> 13; 
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tmp += 7699 * (in[i]  2*in[i1] + in[i2]); 
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/* With "+0x800" rounding, clipping is needed

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for ALGTHM and SPEECH tests. */

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out[i] = av_clip_int16((tmp + 0x800) >> 12); 
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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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void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, 
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const float zero_coeffs[2], 
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const float pole_coeffs[2], 
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float gain, float mem[2], int n) 
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{ 
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int i;

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float tmp;

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for (i = 0; i < n; i++) { 
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tmp = gain * in[i]  pole_coeffs[0] * mem[0]  pole_coeffs[1] * mem[1]; 
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out[i] = tmp + zero_coeffs[0] * mem[0] + zero_coeffs[1] * mem[1]; 
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mem[1] = mem[0]; 
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mem[0] = tmp;

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} 
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} 
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void ff_tilt_compensation(float *mem, float tilt, float *samples, int size) 
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{ 
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float new_tilt_mem = samples[size  1]; 
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int i;

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for (i = size  1; i > 0; i) 
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samples[i] = tilt * samples[i  1];

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samples[0] = tilt * *mem;

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*mem = new_tilt_mem; 
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} 
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