mirror of https://github.com/wb2osz/direwolf.git
749 lines
21 KiB
C
749 lines
21 KiB
C
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//
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// This file is part of Dire Wolf, an amateur radio packet TNC.
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//
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// Copyright (C) 2011, 2012, 2013, 2014, 2015 John Langner, WB2OSZ
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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 as published by
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// the Free Software Foundation, either version 2 of the License, or
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// (at your option) any later version.
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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, see <http://www.gnu.org/licenses/>.
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//
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/*-------------------------------------------------------------------
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*
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* Name: atest.c
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*
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* Purpose: Test fixture for the AFSK demodulator.
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*
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* Inputs: Takes audio from a .WAV file insted of the audio device.
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*
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* Description: This can be used to test the AFSK demodulator under
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* controlled and reproducable conditions for tweaking.
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*
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* For example
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*
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* (1) Download WA8LMF's TNC Test CD image file from
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* http://wa8lmf.net/TNCtest/index.htm
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*
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* (2) Burn a physical CD.
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*
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* (3) "Rip" the desired tracks with Windows Media Player.
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* Select .WAV file format.
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*
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* "Track 2" is used for most tests because that is more
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* realistic for most people using the speaker output.
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*
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*
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* Without ONE_CHAN defined:
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*
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* Notice that the number of packets decoded, as reported by
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* this test program, will be twice the number expected because
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* we are decoding the left and right audio channels separately.
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*
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*
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* With ONE_CHAN defined:
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*
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* Only process one channel.
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*
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*--------------------------------------------------------------------*/
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// #define X 1
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#include <stdio.h>
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#include <unistd.h>
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#include <stdlib.h>
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#include <assert.h>
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#include <string.h>
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#include <time.h>
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#include <getopt.h>
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#define ATEST_C 1
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#include "audio.h"
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#include "demod.h"
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#include "multi_modem.h"
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#include "textcolor.h"
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#include "ax25_pad.h"
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#include "hdlc_rec2.h"
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#include "dlq.h"
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#include "ptt.h"
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#if 0 /* Typical but not flexible enough. */
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struct wav_header { /* .WAV file header. */
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char riff[4]; /* "RIFF" */
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int filesize; /* file length - 8 */
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char wave[4]; /* "WAVE" */
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char fmt[4]; /* "fmt " */
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int fmtsize; /* 16. */
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short wformattag; /* 1 for PCM. */
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short nchannels; /* 1 for mono, 2 for stereo. */
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int nsamplespersec; /* sampling freq, Hz. */
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int navgbytespersec; /* = nblockalign*nsamplespersec. */
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short nblockalign; /* = wbitspersample/8 * nchannels. */
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short wbitspersample; /* 16 or 8. */
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char data[4]; /* "data" */
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int datasize; /* number of bytes following. */
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} ;
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#endif
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/* 8 bit samples are unsigned bytes */
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/* in range of 0 .. 255. */
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/* 16 bit samples are signed short */
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/* in range of -32768 .. +32767. */
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static struct {
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char riff[4]; /* "RIFF" */
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int filesize; /* file length - 8 */
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char wave[4]; /* "WAVE" */
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} header;
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static struct {
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char id[4]; /* "LIST" or "fmt " */
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int datasize;
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} chunk;
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static struct {
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short wformattag; /* 1 for PCM. */
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short nchannels; /* 1 for mono, 2 for stereo. */
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int nsamplespersec; /* sampling freq, Hz. */
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int navgbytespersec; /* = nblockalign*nsamplespersec. */
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short nblockalign; /* = wbitspersample/8 * nchannels. */
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short wbitspersample; /* 16 or 8. */
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char extras[4];
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} format;
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static struct {
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char data[4]; /* "data" */
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int datasize;
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} wav_data;
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static FILE *fp;
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static int e_o_f;
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static int packets_decoded = 0;
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static int decimate = 0; /* Reduce that sampling rate if set. */
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/* 1 = normal, 2 = half, etc. */
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static struct audio_s my_audio_config;
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static int error_if_less_than = -1; /* Exit with error status if this minimum not reached. */
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/* Can be used to check that performance has not decreased. */
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static int error_if_greater_than = -1; /* Exit with error status if this maximum exceeded. */
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/* Can be used to check that duplicate removal is not broken. */
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//#define EXPERIMENT_G 1
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//#define EXPERIMENT_H 1
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#if defined(EXPERIMENT_G) || defined(EXPERIMENT_H)
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static int count[MAX_SUBCHANS];
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#if EXPERIMENT_H
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extern float space_gain[MAX_SUBCHANS];
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#endif
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#endif
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static void usage (void);
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static int decode_only = 0; /* Set to 0 or 1 to decode only one channel. 2 for both. */
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int main (int argc, char *argv[])
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{
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int err;
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int c;
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int channel;
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time_t start_time;
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#if defined(EXPERIMENT_G) || defined(EXPERIMENT_H)
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int j;
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for (j=0; j<MAX_SUBCHANS; j++) {
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count[j] = 0;
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}
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#endif
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text_color_init(1);
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text_color_set(DW_COLOR_INFO);
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/*
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* First apply defaults.
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*/
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memset (&my_audio_config, 0, sizeof(my_audio_config));
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my_audio_config.adev[0].num_channels = DEFAULT_NUM_CHANNELS;
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my_audio_config.adev[0].samples_per_sec = DEFAULT_SAMPLES_PER_SEC;
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my_audio_config.adev[0].bits_per_sample = DEFAULT_BITS_PER_SAMPLE;
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// Results v0.9:
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//
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// fix_bits = 0 971 packets, 69 sec
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// fix_bits = SINGLE 990 64
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// fix_bits = DOUBLE 992 65
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// fix_bits = TRIPLE 992 67
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// fix_bits = TWO_SEP 1004 476
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// Essentially no difference in time for those with order N time.
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// Time increases greatly for the one with order N^2 time.
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// Results: version 1.1, decoder C, my_audio_config.fix_bits = RETRY_MAX - 1
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//
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// 971 NONE
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// +19 SINGLE
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// +2 DOUBLE
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// +12 TWO_SEP
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// +3 REMOVE_MANY
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// ----
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// 1007 Total in 1008 sec. More than twice as long as earlier version.
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// Results: version 1.1, decoders ABC, my_audio_config.fix_bits = RETRY_MAX - 1
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//
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// 976 NONE
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// +21 SINGLE
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// +1 DOUBLE
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// +22 TWO_SEP
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// +1 MANY
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// +3 REMOVE_MANY
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// ----
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// 1024 Total in 2042 sec.
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// About 34 minutes of CPU time for a roughly 40 minute CD.
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// Many computers wouldn't be able to keep up.
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// The SINGLE and TWO_SEP techniques are the most effective.
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// Should we reorder the enum values so that TWO_SEP
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// comes after SINGLE? That way "FIX_BITS 2" would
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// use the two most productive techniques and not waste
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// time on the others. People with plenty of CPU power
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// to spare can still specify larger numbers for the other
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// techniques with less return on investment.
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for (channel=0; channel<MAX_CHANS; channel++) {
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my_audio_config.achan[channel].modem_type = MODEM_AFSK;
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my_audio_config.achan[channel].mark_freq = DEFAULT_MARK_FREQ;
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my_audio_config.achan[channel].space_freq = DEFAULT_SPACE_FREQ;
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my_audio_config.achan[channel].baud = DEFAULT_BAUD;
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strlcpy (my_audio_config.achan[channel].profiles, "E", sizeof(my_audio_config.achan[channel].profiles));
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my_audio_config.achan[channel].num_freq = 1;
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my_audio_config.achan[channel].offset = 0;
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my_audio_config.achan[channel].fix_bits = RETRY_NONE;
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my_audio_config.achan[channel].sanity_test = SANITY_APRS;
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//my_audio_config.achan[channel].sanity_test = SANITY_AX25;
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//my_audio_config.achan[channel].sanity_test = SANITY_NONE;
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my_audio_config.achan[channel].passall = 0;
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//my_audio_config.achan[channel].passall = 1;
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}
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while (1) {
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//int this_option_optind = optind ? optind : 1;
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int option_index = 0;
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static struct option long_options[] = {
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{"future1", 1, 0, 0},
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{"future2", 0, 0, 0},
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{"future3", 1, 0, 'c'},
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{0, 0, 0, 0}
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};
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/* ':' following option character means arg is required. */
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c = getopt_long(argc, argv, "B:P:D:F:L:G:012",
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long_options, &option_index);
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if (c == -1)
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break;
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switch (c) {
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case 'B': /* -B for data Bit rate */
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/* 300 implies 1600/1800 AFSK. */
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/* 1200 implies 1200/2200 AFSK. */
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/* 9600 implies scrambled. */
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my_audio_config.achan[0].baud = atoi(optarg);
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dw_printf ("Data rate set to %d bits / second.\n", my_audio_config.achan[0].baud);
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if (my_audio_config.achan[0].baud < 100 || my_audio_config.achan[0].baud > 10000) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("Use a more reasonable bit rate in range of 100 - 10000.\n");
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exit (EXIT_FAILURE);
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}
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if (my_audio_config.achan[0].baud < 600) {
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my_audio_config.achan[0].modem_type = MODEM_AFSK;
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my_audio_config.achan[0].mark_freq = 1600;
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my_audio_config.achan[0].space_freq = 1800;
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strlcpy (my_audio_config.achan[0].profiles, "D", sizeof(my_audio_config.achan[0].profiles));
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}
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else if (my_audio_config.achan[0].baud > 2400) {
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my_audio_config.achan[0].modem_type = MODEM_SCRAMBLE;
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my_audio_config.achan[0].mark_freq = 0;
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my_audio_config.achan[0].space_freq = 0;
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strlcpy (my_audio_config.achan[0].profiles, " ", sizeof(my_audio_config.achan[0].profiles)); // avoid getting default later.
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dw_printf ("Using scrambled baseband signal rather than AFSK.\n");
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}
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else {
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my_audio_config.achan[0].modem_type = MODEM_AFSK;
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my_audio_config.achan[0].mark_freq = 1200;
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my_audio_config.achan[0].space_freq = 2200;
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}
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break;
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case 'P': /* -P for modem profile. */
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dw_printf ("Demodulator profile set to \"%s\"\n", optarg);
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strlcpy (my_audio_config.achan[0].profiles, optarg, sizeof(my_audio_config.achan[0].profiles));
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break;
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case 'D': /* -D reduce sampling rate for lower CPU usage. */
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decimate = atoi(optarg);
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dw_printf ("Divide audio sample rate by %d\n", decimate);
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if (decimate < 1 || decimate > 8) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("Unreasonable value for -D.\n");
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exit (1);
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}
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dw_printf ("Divide audio sample rate by %d\n", decimate);
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my_audio_config.achan[0].decimate = decimate;
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break;
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case 'F': /* -D set "fix bits" level. */
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my_audio_config.achan[0].fix_bits = atoi(optarg);
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if (my_audio_config.achan[0].fix_bits < RETRY_NONE || my_audio_config.achan[0].fix_bits >= RETRY_MAX) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("Invalid Fix Bits level.\n");
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exit (1);
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}
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break;
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case 'L': /* -L error if less than this number decoded. */
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error_if_less_than = atoi(optarg);
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break;
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case 'G': /* -G error if greater than this number decoded. */
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error_if_greater_than = atoi(optarg);
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break;
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case '0': /* channel 0, left from stereo */
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decode_only = 0;
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break;
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case '1': /* channel 1, right from stereo */
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decode_only = 1;
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break;
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case '2': /* decode both from stereo */
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decode_only = 2;
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break;
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case '?':
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/* Unknown option message was already printed. */
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usage ();
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break;
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default:
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/* Should not be here. */
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text_color_set(DW_COLOR_ERROR);
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dw_printf("?? getopt returned character code 0%o ??\n", c);
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usage ();
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}
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}
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memcpy (&my_audio_config.achan[1], &my_audio_config.achan[0], sizeof(my_audio_config.achan[0]));
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if (optind >= argc) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("Specify .WAV file name on command line.\n");
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usage ();
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}
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fp = fopen(argv[optind], "rb");
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if (fp == NULL) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("Couldn't open file for read: %s\n", argv[optind]);
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//perror ("more info?");
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exit (1);
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}
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start_time = time(NULL);
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/*
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* Read the file header.
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* Doesn't handle all possible cases but good enough for our purposes.
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*/
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err= fread (&header, (size_t)12, (size_t)1, fp);
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(void)(err);
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if (strncmp(header.riff, "RIFF", 4) != 0 || strncmp(header.wave, "WAVE", 4) != 0) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("This is not a .WAV format file.\n");
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exit (EXIT_FAILURE);
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}
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err = fread (&chunk, (size_t)8, (size_t)1, fp);
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if (strncmp(chunk.id, "LIST", 4) == 0) {
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err = fseek (fp, (long)chunk.datasize, SEEK_CUR);
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err = fread (&chunk, (size_t)8, (size_t)1, fp);
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}
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if (strncmp(chunk.id, "fmt ", 4) != 0) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("WAV file error: Found \"%4.4s\" where \"fmt \" was expected.\n", chunk.id);
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exit(1);
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}
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if (chunk.datasize != 16 && chunk.datasize != 18) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("WAV file error: Need fmt chunk datasize of 16 or 18. Found %d.\n", chunk.datasize);
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exit(1);
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}
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err = fread (&format, (size_t)chunk.datasize, (size_t)1, fp);
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err = fread (&wav_data, (size_t)8, (size_t)1, fp);
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if (strncmp(wav_data.data, "data", 4) != 0) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("WAV file error: Found \"%4.4s\" where \"data\" was expected.\n", wav_data.data);
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exit(1);
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}
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// TODO: Should have proper message, not abort.
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assert (format.nchannels == 1 || format.nchannels == 2);
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assert (format.wbitspersample == 8 || format.wbitspersample == 16);
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my_audio_config.adev[0].samples_per_sec = format.nsamplespersec;
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my_audio_config.adev[0].bits_per_sample = format.wbitspersample;
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my_audio_config.adev[0].num_channels = format.nchannels;
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my_audio_config.achan[0].valid = 1;
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if (format.nchannels == 2) my_audio_config.achan[1].valid = 1;
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text_color_set(DW_COLOR_INFO);
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dw_printf ("%d samples per second\n", my_audio_config.adev[0].samples_per_sec);
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dw_printf ("%d bits per sample\n", my_audio_config.adev[0].bits_per_sample);
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dw_printf ("%d audio channels\n", my_audio_config.adev[0].num_channels);
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dw_printf ("%d audio bytes in file\n", (int)(wav_data.datasize));
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dw_printf ("Fix Bits level = %d\n", my_audio_config.achan[0].fix_bits);
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/*
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* Initialize the AFSK demodulator and HDLC decoder.
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*/
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multi_modem_init (&my_audio_config);
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e_o_f = 0;
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while ( ! e_o_f)
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{
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int audio_sample;
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int c;
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for (c=0; c<my_audio_config.adev[0].num_channels; c++)
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{
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/* This reads either 1 or 2 bytes depending on */
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/* bits per sample. */
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audio_sample = demod_get_sample (ACHAN2ADEV(c));
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if (audio_sample >= 256 * 256)
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e_o_f = 1;
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if (decode_only == 0 && c != 0) continue;
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if (decode_only == 1 && c != 1) continue;
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multi_modem_process_sample(c,audio_sample);
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}
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/* When a complete frame is accumulated, */
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/* process_rec_frame, below, is called. */
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}
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text_color_set(DW_COLOR_INFO);
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dw_printf ("\n\n");
|
|
|
|
#if EXPERIMENT_G
|
|
|
|
for (j=0; j<MAX_SUBCHANS; j++) {
|
|
float db = 20.0 * log10f(space_gain[j]);
|
|
dw_printf ("%+.1f dB, %d\n", db, count[j]);
|
|
}
|
|
#endif
|
|
#if EXPERIMENT_H
|
|
|
|
for (j=0; j<MAX_SUBCHANS; j++) {
|
|
dw_printf ("%d\n", count[j]);
|
|
}
|
|
#endif
|
|
dw_printf ("%d packets decoded in %d seconds.\n", packets_decoded, (int)(time(NULL) - start_time));
|
|
|
|
if (error_if_less_than != -1 && packets_decoded < error_if_less_than) {
|
|
text_color_set(DW_COLOR_ERROR);
|
|
dw_printf ("\n * * * TEST FAILED: number decoded is less than %d * * * \n", error_if_less_than);
|
|
exit (1);
|
|
}
|
|
if (error_if_greater_than != -1 && packets_decoded > error_if_greater_than) {
|
|
text_color_set(DW_COLOR_ERROR);
|
|
dw_printf ("\n * * * TEST FAILED: number decoded is greater than %d * * * \n", error_if_greater_than);
|
|
exit (1);
|
|
}
|
|
|
|
exit (0);
|
|
}
|
|
|
|
|
|
/*
|
|
* Simulate sample from the audio device.
|
|
*/
|
|
|
|
int audio_get (int a)
|
|
{
|
|
int ch;
|
|
|
|
if (wav_data.datasize <= 0) {
|
|
e_o_f = 1;
|
|
return (-1);
|
|
}
|
|
|
|
ch = getc(fp);
|
|
wav_data.datasize--;
|
|
|
|
if (ch < 0) {
|
|
text_color_set(DW_COLOR_ERROR);
|
|
dw_printf ("Unexpected end of file.\n");
|
|
e_o_f = 1;
|
|
}
|
|
|
|
return (ch);
|
|
}
|
|
|
|
|
|
|
|
/*
|
|
* Rather than queuing up frames with bad FCS,
|
|
* try to fix them immediately.
|
|
*/
|
|
|
|
void rdq_append (rrbb_t rrbb)
|
|
{
|
|
int chan;
|
|
alevel_t alevel;
|
|
int subchan;
|
|
|
|
|
|
chan = rrbb_get_chan(rrbb);
|
|
subchan = rrbb_get_subchan(rrbb);
|
|
alevel = rrbb_get_audio_level(rrbb);
|
|
|
|
hdlc_rec2_try_to_fix_later (rrbb, chan, subchan, alevel);
|
|
rrbb_delete (rrbb);
|
|
}
|
|
|
|
|
|
/*
|
|
* This is called when we have a good frame.
|
|
*/
|
|
|
|
void dlq_append (dlq_type_t type, int chan, int subchan, packet_t pp, alevel_t alevel, retry_t retries, char *spectrum)
|
|
{
|
|
|
|
char stemp[500];
|
|
unsigned char *pinfo;
|
|
int info_len;
|
|
int h;
|
|
char heard[AX25_MAX_ADDR_LEN];
|
|
char alevel_text[AX25_ALEVEL_TO_TEXT_SIZE];
|
|
|
|
packets_decoded++;
|
|
|
|
ax25_format_addrs (pp, stemp);
|
|
|
|
info_len = ax25_get_info (pp, &pinfo);
|
|
|
|
/* Print so we can see what is going on. */
|
|
|
|
//TODO: quiet option - suppress packet printing, only the count at the end.
|
|
|
|
#if 1
|
|
/* Display audio input level. */
|
|
/* Who are we hearing? Original station or digipeater? */
|
|
|
|
if (ax25_get_num_addr(pp) == 0) {
|
|
/* Not AX.25. No station to display below. */
|
|
h = -1;
|
|
strlcpy (heard, "", sizeof(heard));
|
|
}
|
|
else {
|
|
h = ax25_get_heard(pp);
|
|
ax25_get_addr_with_ssid(pp, h, heard);
|
|
}
|
|
|
|
text_color_set(DW_COLOR_DEBUG);
|
|
dw_printf ("\n");
|
|
dw_printf("DECODED[%d] ", packets_decoded );
|
|
if (h != AX25_SOURCE) {
|
|
dw_printf ("Digipeater ");
|
|
}
|
|
ax25_alevel_to_text (alevel, alevel_text);
|
|
|
|
if (my_audio_config.achan[chan].fix_bits == RETRY_NONE && my_audio_config.achan[chan].passall == 0) {
|
|
dw_printf ("%s audio level = %s %s\n", heard, alevel_text, spectrum);
|
|
}
|
|
else {
|
|
dw_printf ("%s audio level = %s [%s] %s\n", heard, alevel_text, retry_text[(int)retries], spectrum);
|
|
}
|
|
|
|
#endif
|
|
|
|
#if defined(EXPERIMENT_G) || defined(EXPERIMENT_H)
|
|
int j;
|
|
|
|
for (j=0; j<MAX_SUBCHANS; j++) {
|
|
if (spectrum[j] == '|') {
|
|
count[j]++;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
|
|
// Display non-APRS packets in a different color.
|
|
|
|
// TODO: display subchannel if appropriate.
|
|
|
|
if (ax25_is_aprs(pp)) {
|
|
text_color_set(DW_COLOR_REC);
|
|
dw_printf ("[%d] ", chan);
|
|
}
|
|
else {
|
|
text_color_set(DW_COLOR_DEBUG);
|
|
dw_printf ("[%d] ", chan);
|
|
}
|
|
|
|
dw_printf ("%s", stemp); /* stations followed by : */
|
|
ax25_safe_print ((char *)pinfo, info_len, 0);
|
|
dw_printf ("\n");
|
|
|
|
ax25_delete (pp);
|
|
|
|
} /* end app_process_rec_packet */
|
|
|
|
|
|
void ptt_set (int ot, int chan, int ptt_signal)
|
|
{
|
|
return;
|
|
}
|
|
|
|
int get_input (int it, int chan)
|
|
{
|
|
return -1;
|
|
}
|
|
|
|
static void usage (void) {
|
|
|
|
text_color_set(DW_COLOR_ERROR);
|
|
|
|
dw_printf ("\n");
|
|
dw_printf ("atest is a test application which decodes AX.25 frames from an audio\n");
|
|
dw_printf ("recording. This provides an easy way to test Dire Wolf decoding\n");
|
|
dw_printf ("performance much quicker than normal real-time. \n");
|
|
dw_printf ("\n");
|
|
dw_printf ("usage:\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" atest [ options ] wav-file-in\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" -B n Bits/second for data. Proper modem automatically selected for speed.\n");
|
|
dw_printf (" 300 baud uses 1600/1800 Hz AFSK.\n");
|
|
dw_printf (" 1200 (default) baud uses 1200/2200 Hz AFSK.\n");
|
|
dw_printf (" 9600 baud uses K9NG/G2RUH standard.\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" -D n Divide audio sample rate by n.\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" -F n Amount of effort to try fixing frames with an invalid CRC. \n");
|
|
dw_printf (" 0 (default) = consider only correct frames. \n");
|
|
dw_printf (" 1 = Try to fix only a single bit. \n");
|
|
dw_printf (" more = Try modifying more bits to get a good CRC.\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" -P m Select the demodulator type such as A, B, C, D (default for 300 baud),\n");
|
|
dw_printf (" E (default for 1200 baud), F, A+, B+, C+, D+, E+, F+.\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" -0 Use channel 0 (left) of stereo audio (default).\n");
|
|
dw_printf (" -1 use channel 1 (right) of stereo audio.\n");
|
|
dw_printf (" -1 decode both channels of stereo audio.\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" wav-file-in is a WAV format audio file.\n");
|
|
dw_printf ("\n");
|
|
dw_printf ("Examples:\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" gen_packets -o test1.wav\n");
|
|
dw_printf (" atest test1.wav\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" gen_packets -B 300 -o test3.wav\n");
|
|
dw_printf (" atest -B 300 test3.wav\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" gen_packets -B 9600 -o test9.wav\n");
|
|
dw_printf (" atest -B 9600 test9.wav\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" This generates and decodes 3 test files with 1200, 300, and 9600\n");
|
|
dw_printf (" bits per second.\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" atest 02_Track_2.wav\n");
|
|
dw_printf (" atest -P C+ 02_Track_2.wav\n");
|
|
dw_printf (" atest -F 1 02_Track_2.wav\n");
|
|
dw_printf (" atest -P C+ -F 1 02_Track_2.wav\n");
|
|
dw_printf ("\n");
|
|
dw_printf (" Try different combinations of options to find the best decoding\n");
|
|
dw_printf (" performance.\n");
|
|
|
|
exit (1);
|
|
}
|
|
|
|
|
|
|
|
/* end atest.c */
|