forked from Mirrors/freeswitch
314ae8b6f3
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@11535 d0543943-73ff-0310-b7d9-9358b9ac24b2
410 lines
13 KiB
C
410 lines
13 KiB
C
/*
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* SpanDSP - a series of DSP components for telephony
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*
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* modem_echo_tests.c
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*
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* Written by Steve Underwood <steveu@coppice.org>
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*
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* Copyright (C) 2004 Steve Underwood
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*
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* All rights reserved.
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License version 2, as
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* published by the Free Software Foundation.
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*
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* This program 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
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
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*
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* $Id: modem_echo_tests.c,v 1.31 2008/11/30 10:17:31 steveu Exp $
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*/
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/*! \page modem_echo_can_tests_page Line echo cancellation for modems tests
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\section modem_echo_can_tests_page_sec_1 What does it do?
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Currently the echo cancellation tests only provide simple exercising of the
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cancellor in the way it might be used for line echo cancellation. The test code
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is in echotests.c.
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The goal is to test the echo cancellor again the G.16X specs. Clearly, that also
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means the goal for the cancellor itself is to comply with those specs. Right
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now, the only aspect of these tests implemented is the line impulse response
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models in g168tests.c.
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\section modem_echo_can_tests_page_sec_2 How does it work?
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The current test consists of feeding a wave file of real speech to the echo
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cancellor as the transmit signal. A very simple model of a telephone line is
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used to simulate a simple echo from the transmit signal. A second wave file of
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real speech is also used to simulate a signal received form the far end of the
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line. This is gated so it is only placed for one second every 10 seconds,
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simulating the double talk condition. The resulting echo cancelled signal can
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either be store in a file for further analysis, or played back as the data is
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processed.
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A number of modified versions of this test have been performed. The signal level
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of the two speech sources has been varied. Several simple models of the
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telephone line have been used. Although the current cancellor design has known
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limitations, it seems stable for all these test conditions. No instability has
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been observed in the current version due to arithmetic overflow when the speech
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is very loud (with earlier versions, well, ....:) ). The lack of saturating
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arithmetic in general purpose CPUs is a huge disadvantage here, as software
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saturation logic would cause a major slow down. Floating point would be good,
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but is not usable in the Linux kernel. Anyway, the bottom line seems to be the
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current design is genuinely useful, if imperfect.
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\section modem_echo_can_tests_page_sec_2 How do I use it?
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Build the tests with the command "./build". Currently there is no proper make
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setup, or way to build individual tests. "./build" will built all the tests
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which currently exist for the DSP functions. The echo cancellation test assumes
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there are two wave files containing mono, 16 bit signed PCM speech data, sampled
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at 8kHz. These should be called local_sound.wav and far_sound.wav. A third wave
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file will be produced. This very crudely starts with the first 256 bytes from
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the local_sound.wav file, followed by the results of the echo cancellation. The
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resulting audio is also played to the /dev/dsp device. A printf near the end of
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echo_tests.c is commented out with a \#if. If this is enabled, detailed
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information about the results of the echo cancellation will be written to
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stdout. By saving this into a file, Grace (recommended), GnuPlot, or some other
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plotting package may be used to graphically display the functioning of the
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cancellor.
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*/
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#if defined(HAVE_CONFIG_H)
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#include "config.h"
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#endif
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#if defined(HAVE_FL_FL_H) && defined(HAVE_FL_FL_CARTESIAN_H) && defined(HAVE_FL_FL_AUDIO_METER_H)
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#define ENABLE_GUI
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#endif
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#include <stdlib.h>
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#include <stdio.h>
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#include <fcntl.h>
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#include <string.h>
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#include <time.h>
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#include <audiofile.h>
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#if defined(HAVE_MATH_H)
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#define GEN_CONST
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#endif
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//#if defined(WITH_SPANDSP_INTERNALS)
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#define SPANDSP_EXPOSE_INTERNAL_STRUCTURES
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//#endif
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#include "spandsp.h"
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#include "spandsp/g168models.h"
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#include "spandsp-sim.h"
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#if defined(ENABLE_GUI)
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#include "echo_monitor.h"
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#endif
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#if !defined(NULL)
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#define NULL (void *) 0
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#endif
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typedef struct
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{
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const char *name;
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int max;
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int cur;
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AFfilehandle handle;
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int16_t signal[8000];
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} signal_source_t;
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signal_source_t local_css;
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fir32_state_t line_model;
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AFfilehandle resulthandle;
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int16_t residue_sound[8000];
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int residue_cur = 0;
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int do_codec_munge = TRUE;
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int use_gui = FALSE;
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static const int16_t tone_1khz[] = {10362, 7327, 0, -7327, -10362, -7327, 0, 7327};
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static inline void put_residue(int16_t tx, int16_t residue)
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{
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int outframes;
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residue_sound[residue_cur++] = tx;
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residue_sound[residue_cur++] = residue;
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if (residue_cur >= 8000)
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{
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residue_cur >>= 1;
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outframes = afWriteFrames(resulthandle,
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AF_DEFAULT_TRACK,
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residue_sound,
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residue_cur);
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if (outframes != residue_cur)
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{
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fprintf(stderr, " Error writing residue sound\n");
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exit(2);
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}
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residue_cur = 0;
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}
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}
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/*- End of function --------------------------------------------------------*/
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static void signal_load(signal_source_t *sig, const char *name)
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{
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sig->name = name;
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if ((sig->handle = afOpenFile_telephony_read(sig->name, 1)) == AF_NULL_FILEHANDLE)
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{
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fprintf(stderr, " Cannot open sound file '%s'\n", sig->name);
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exit(2);
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}
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sig->max = afReadFrames(sig->handle, AF_DEFAULT_TRACK, sig->signal, 8000);
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if (sig->max < 0)
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{
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fprintf(stderr, " Error reading sound file '%s'\n", sig->name);
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exit(2);
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}
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}
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/*- End of function --------------------------------------------------------*/
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static void signal_free(signal_source_t *sig)
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{
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if (afCloseFile(sig->handle) != 0)
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{
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fprintf(stderr, " Cannot close sound file '%s'\n", sig->name);
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exit(2);
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}
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}
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/*- End of function --------------------------------------------------------*/
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static void signal_restart(signal_source_t *sig)
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{
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sig->cur = 0;
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}
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/*- End of function --------------------------------------------------------*/
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static int16_t signal_amp(signal_source_t *sig)
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{
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int16_t tx;
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tx = sig->signal[sig->cur++];
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if (sig->cur >= sig->max)
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sig->cur = 0;
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return tx;
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}
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/*- End of function --------------------------------------------------------*/
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static inline int16_t codec_munger(int16_t amp)
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{
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if (do_codec_munge)
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return alaw_to_linear(linear_to_alaw(amp));
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return amp;
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}
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/*- End of function --------------------------------------------------------*/
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static void channel_model_create(int model)
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{
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static const int32_t *line_models[] =
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{
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line_model_d2_coeffs,
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line_model_d3_coeffs,
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line_model_d4_coeffs,
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line_model_d5_coeffs,
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line_model_d6_coeffs,
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line_model_d7_coeffs,
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line_model_d8_coeffs,
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line_model_d9_coeffs
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};
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static int line_model_sizes[] =
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{
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sizeof(line_model_d2_coeffs)/sizeof(int32_t),
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sizeof(line_model_d3_coeffs)/sizeof(int32_t),
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sizeof(line_model_d4_coeffs)/sizeof(int32_t),
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sizeof(line_model_d5_coeffs)/sizeof(int32_t),
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sizeof(line_model_d6_coeffs)/sizeof(int32_t),
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sizeof(line_model_d7_coeffs)/sizeof(int32_t),
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sizeof(line_model_d8_coeffs)/sizeof(int32_t),
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sizeof(line_model_d9_coeffs)/sizeof(int32_t)
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};
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fir32_create(&line_model, line_models[model], line_model_sizes[model]);
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}
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/*- End of function --------------------------------------------------------*/
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static int16_t channel_model(int16_t local, int16_t far)
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{
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int16_t echo;
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int16_t rx;
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/* Channel modelling is merely simulating the effects of A-law distortion
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and using one of the echo models from G.168 */
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/* The local tx signal will have gone through an A-law munging before
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it reached the line's analogue area where the echo occurs. */
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echo = fir32(&line_model, codec_munger(local/8));
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/* The far end signal will have been through an A-law munging, although
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this should not affect things. */
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rx = echo + codec_munger(far);
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/* This mixed echo and far end signal will have been through an A-law munging when it came back into
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the digital network. */
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rx = codec_munger(rx);
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return rx;
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}
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/*- End of function --------------------------------------------------------*/
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int main(int argc, char *argv[])
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{
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modem_echo_can_state_t *ctx;
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//awgn_state_t local_noise_source;
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awgn_state_t far_noise_source;
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int i;
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int clean;
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int16_t rx;
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int16_t tx;
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int local_cur;
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int far_cur;
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int result_cur;
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int line_model_no;
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time_t now;
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power_meter_t power_before;
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power_meter_t power_after;
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float unadapted_output_power;
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float unadapted_echo_power;
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float adapted_output_power;
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float adapted_echo_power;
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#if defined(ENABLE_GUI)
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int16_t amp[2];
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#endif
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line_model_no = 0;
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use_gui = FALSE;
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for (i = 1; i < argc; i++)
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{
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if (strcmp(argv[i], "-g") == 0)
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{
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use_gui = TRUE;
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continue;
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}
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line_model_no = atoi(argv[1]);
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}
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time(&now);
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ctx = modem_echo_can_create(256);
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awgn_init_dbm0(&far_noise_source, 7162534, -50.0f);
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signal_load(&local_css, "sound_c1_8k.wav");
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if ((resulthandle = afOpenFile_telephony_write("modem_echo.wav", 2)) == AF_NULL_FILEHANDLE)
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{
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fprintf(stderr, " Failed to open result file\n");
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exit(2);
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}
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#if defined(ENABLE_GUI)
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if (use_gui)
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start_echo_can_monitor(256);
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#endif
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local_cur = 0;
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far_cur = 0;
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result_cur = 0;
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channel_model_create(line_model_no);
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#if defined(ENABLE_GUI)
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if (use_gui)
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echo_can_monitor_line_model_update(line_model.coeffs, line_model.taps);
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#endif
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modem_echo_can_flush(ctx);
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power_meter_init(&power_before, 5);
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power_meter_init(&power_after, 5);
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/* Measure the echo power before adaption */
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modem_echo_can_adaption_mode(ctx, FALSE);
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for (i = 0; i < 8000*5; i++)
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{
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tx = tone_1khz[i & 7];
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rx = channel_model(tx, 0);
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clean = modem_echo_can_update(ctx, tx, rx);
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power_meter_update(&power_before, rx);
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power_meter_update(&power_after, clean);
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}
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unadapted_output_power = power_meter_current_dbm0(&power_before);
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unadapted_echo_power = power_meter_current_dbm0(&power_after);
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printf("Pre-adaption: output power %10.5fdBm0, echo power %10.5fdBm0\n", unadapted_output_power, unadapted_echo_power);
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/* Converge the canceller */
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signal_restart(&local_css);
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modem_echo_can_adaption_mode(ctx, TRUE);
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for (i = 0; i < 800*2; i++)
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{
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clean = modem_echo_can_update(ctx, 0, 0);
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put_residue(0, clean);
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}
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for (i = 0; i < 8000*50; i++)
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{
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tx = signal_amp(&local_css);
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rx = channel_model(tx, 0);
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clean = modem_echo_can_update(ctx, tx, rx);
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power_meter_update(&power_before, rx);
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power_meter_update(&power_after, clean);
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#if 0
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if (i%800 == 0)
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printf("Powers %10.5fdBm0 %10.5fdBm0\n", power_meter_current_dbm0(&power_before), power_meter_current_dbm0(&power_after));
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#endif
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put_residue(tx, clean);
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#if defined(ENABLE_GUI)
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if (use_gui)
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{
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echo_can_monitor_can_update(ctx->fir_taps16, 256);
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amp[0] = tx;
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echo_can_monitor_line_spectrum_update(amp, 1);
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}
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#endif
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}
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/* Now lets see how well adapted we are */
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modem_echo_can_adaption_mode(ctx, FALSE);
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for (i = 0; i < 8000*5; i++)
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{
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tx = tone_1khz[i & 7];
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rx = channel_model(tx, 0);
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clean = modem_echo_can_update(ctx, tx, rx);
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power_meter_update(&power_before, rx);
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power_meter_update(&power_after, clean);
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}
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adapted_output_power = power_meter_current_dbm0(&power_before);
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adapted_echo_power = power_meter_current_dbm0(&power_after);
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printf("Post-adaption: output power %10.5fdBm0, echo power %10.5fdBm0\n", adapted_output_power, adapted_echo_power);
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if (fabsf(adapted_output_power - unadapted_output_power) > 0.1f
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adapted_echo_power > unadapted_echo_power - 30.0f)
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{
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printf("Tests failed.\n");
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exit(2);
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}
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modem_echo_can_free(ctx);
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signal_free(&local_css);
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if (afCloseFile(resulthandle) != 0)
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{
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fprintf(stderr, " Cannot close speech file '%s'\n", "result_sound.wav");
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exit(2);
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}
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#if defined(ENABLE_GUI)
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if (use_gui)
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echo_can_monitor_wait_to_end();
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#endif
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printf("Tests passed.\n");
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return 0;
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}
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/*- End of function --------------------------------------------------------*/
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/*- End of file ------------------------------------------------------------*/
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