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dsd-fme/src/dsd_symbol.c

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/*
* Copyright (C) 2010 DSD Author
* GPG Key ID: 0x3F1D7FD0 (74EF 430D F7F2 0A48 FCE6 F630 FAA2 635D 3F1D 7FD0)
*
* Permission to use, copy, modify, and/or distribute this software for any
* purpose with or without fee is hereby granted, provided that the above
* copyright notice and this permission notice appear in all copies.
*
* THE SOFTWARE IS PROVIDED "AS IS" AND ISC DISCLAIMS ALL WARRANTIES WITH
* REGARD TO THIS SOFTWARE INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY
* AND FITNESS. IN NO EVENT SHALL ISC BE LIABLE FOR ANY SPECIAL, DIRECT,
* INDIRECT, OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES WHATSOEVER RESULTING FROM
* LOSS OF USE, DATA OR PROFITS, WHETHER IN AN ACTION OF CONTRACT, NEGLIGENCE
* OR OTHER TORTIOUS ACTION, ARISING OUT OF OR IN CONNECTION WITH THE USE OR
* PERFORMANCE OF THIS SOFTWARE.
*/
#include "dsd.h"
int
getSymbol (dsd_opts * opts, dsd_state * state, int have_sync)
{
short sample;
int i, sum, symbol, count;
ssize_t result;
sum = 0;
count = 0;
sample = 0; //init sample with a value of 0...see if this was causing issues with raw audio monitoring
for (i = 0; i < state->samplesPerSymbol; i++) //right HERE
{
// timing control
if ((i == 0) && (have_sync == 0))
{
if (state->samplesPerSymbol == 20)
{
if ((state->jitter >= 7) && (state->jitter <= 10))
{
i--;
}
else if ((state->jitter >= 11) && (state->jitter <= 14))
{
i++;
}
}
else if (state->rf_mod == 1)
{
if ((state->jitter >= 0) && (state->jitter < state->symbolCenter))
{
i++; // fall back
}
else if ((state->jitter > state->symbolCenter) && (state->jitter < 10))
{
i--; // catch up
}
}
else if (state->rf_mod == 2)
{
if ((state->jitter >= state->symbolCenter - 1) && (state->jitter <= state->symbolCenter))
{
i--;
}
else if ((state->jitter >= state->symbolCenter + 1) && (state->jitter <= state->symbolCenter + 2))
{
i++;
}
}
else if (state->rf_mod == 0)
{
if ((state->jitter > 0) && (state->jitter <= state->symbolCenter))
{
i--; // catch up
}
else if ((state->jitter > state->symbolCenter) && (state->jitter < state->samplesPerSymbol))
{
i++; // fall back
}
}
state->jitter = -1;
}
// Read the new sample from the input
if(opts->audio_in_type == 0) //pulse audio
{
pa_simple_read(opts->pulse_digi_dev_in, &sample, 2, NULL );
}
else if (opts->audio_in_type == 5) //OSS
{
read (opts->audio_in_fd, &sample, 2);
}
//stdin only, wav files moving to new number
else if (opts->audio_in_type == 1) //won't work in windows, needs posix pipe (mintty)
{
result = sf_read_short(opts->audio_in_file, &sample, 1);
if(result == 0)
{
sf_close(opts->audio_in_file);
cleanupAndExit (opts, state);
}
}
//wav files, same but using seperate value so we can still manipulate ncurses menu
//since we can not worry about getch/stdin conflict
else if (opts->audio_in_type == 2)
{
result = sf_read_short(opts->audio_in_file, &sample, 1);
if(result == 0)
{
sf_close(opts->audio_in_file);
fprintf (stderr, "\nEnd of %s\n", opts->audio_in_dev);
//open pulse input if we are pulse output AND using ncurses terminal
if (opts->audio_out_type == 0 && opts->use_ncurses_terminal == 1)
{
opts->audio_in_type = 0; //set input type
openPulseInput(opts); //open pulse input
}
//else cleanup and exit
else
{
cleanupAndExit(opts, state);
}
}
}
else if (opts->audio_in_type == 3)
{
#ifdef USE_RTLSDR
// Read demodulated stream here
if (get_rtlsdr_sample(&sample, opts, state) < 0)
cleanupAndExit(opts, state);
//update root means square power level
opts->rtl_rms = rtl_return_rms();
sample *= opts->rtl_volume_multiplier;
#endif
}
//tcp socket input from SDR++ -- now with 1 retry if connection is broken
else if (opts->audio_in_type == 8)
{
#ifdef AERO_BUILD
result = sf_read_short(opts->tcp_file_in, &sample, 1);
if(result == 0) {
fprintf (stderr, "\nConnection to TCP Server Interrupted. Trying again in 3 seconds.\n");
sample = 0;
sf_close(opts->tcp_file_in); //close current connection on this end
sleep (3); //halt all processing and wait 3 seconds
//attempt to reconnect to socket
opts->tcp_sockfd = 0;
opts->tcp_sockfd = Connect(opts->tcp_hostname, opts->tcp_portno);
if (opts->tcp_sockfd != 0)
{
//reset audio input stream
opts->audio_in_file_info = calloc(1, sizeof(SF_INFO));
opts->audio_in_file_info->samplerate=opts->wav_sample_rate;
opts->audio_in_file_info->channels=1;
opts->audio_in_file_info->seekable=0;
opts->audio_in_file_info->format=SF_FORMAT_RAW|SF_FORMAT_PCM_16|SF_ENDIAN_LITTLE;
opts->tcp_file_in = sf_open_fd(opts->tcp_sockfd, SFM_READ, opts->audio_in_file_info, 0);
if(opts->tcp_file_in == NULL)
{
fprintf(stderr, "Error, couldn't Reconnect to TCP with libsndfile: %s\n", sf_strerror(NULL));
}
else fprintf (stderr, "TCP Socket Reconnected Successfully.\n");
}
else fprintf (stderr, "TCP Socket Connection Error.\n");
//now retry reading sample
result = sf_read_short(opts->tcp_file_in, &sample, 1);
if (result == 0) {
sf_close(opts->tcp_file_in);
opts->audio_in_type = 0; //set input type
opts->tcp_sockfd = 0; //added this line so we will know if it connected when using ncurses terminal keyboard shortcut
//openPulseInput(opts); //open pulse inpput
sample = 0; //zero sample on bad result, keep the ball rolling
//open pulse input if we are pulse output AND using ncurses terminal
if (opts->audio_out_type == 0 && opts->use_ncurses_terminal == 1)
{
fprintf (stderr, "Connection to TCP Server Disconnected.\n");
fprintf (stderr, "Opening Pulse Audio Input.\n");
opts->audio_in_type = 0; //set input type
openPulseInput(opts); //open pulse input
}
//else cleanup and exit
else
{
fprintf (stderr, "Connection to TCP Server Disconnected.\n");
fprintf (stderr, "Closing DSD-FME.\n");
cleanupAndExit(opts, state);
}
}
}
#else
result = sf_read_short(opts->tcp_file_in, &sample, 1);
if(result == 0) {
TCP_RETRY:
if (exitflag == 1) cleanupAndExit(opts, state); //needed to break the loop on ctrl+c
fprintf (stderr, "\nConnection to TCP Server Interrupted. Trying again in 3 seconds.\n");
sample = 0;
sf_close(opts->tcp_file_in); //close current connection on this end
sleep (3); //halt all processing and wait 3 seconds
//attempt to reconnect to socket
opts->tcp_sockfd = 0;
opts->tcp_sockfd = Connect(opts->tcp_hostname, opts->tcp_portno);
if (opts->tcp_sockfd != 0)
{
//reset audio input stream
opts->audio_in_file_info = calloc(1, sizeof(SF_INFO));
opts->audio_in_file_info->samplerate=opts->wav_sample_rate;
opts->audio_in_file_info->channels=1;
opts->audio_in_file_info->seekable=0;
opts->audio_in_file_info->format=SF_FORMAT_RAW|SF_FORMAT_PCM_16|SF_ENDIAN_LITTLE;
opts->tcp_file_in = sf_open_fd(opts->tcp_sockfd, SFM_READ, opts->audio_in_file_info, 0);
if(opts->tcp_file_in == NULL)
{
fprintf(stderr, "Error, couldn't Reconnect to TCP with libsndfile: %s\n", sf_strerror(NULL));
}
else fprintf (stderr, "TCP Socket Reconnected Successfully.\n");
}
else
{
fprintf (stderr, "TCP Socket Connection Error.\n");
if (opts->frame_m17 == 1) goto TCP_RETRY; //if using m17 encoder/decoder, just keep looping to keep alive
}
//now retry reading sample
result = sf_read_short(opts->tcp_file_in, &sample, 1);
if (result == 0) {
sf_close(opts->tcp_file_in);
opts->audio_in_type = 0; //set input type
opts->tcp_sockfd = 0; //added this line so we will know if it connected when using ncurses terminal keyboard shortcut
openPulseInput(opts); //open pulse inpput
sample = 0; //zero sample on bad result, keep the ball rolling
fprintf (stderr, "Connection to TCP Server Disconnected.\n");
}
}
#endif
}
//BUG REPORT: 1. DMR Simplex doesn't work with raw wav files. 2. Using the monitor w/ wav file saving may produce undecodable wav files.
//reworked a bit to allow raw audio wav file saving without the monitoring poriton active
if (have_sync == 0)
{
//do an extra checkfor carrier signal so that random raw audio spurts don't play during decoding
// if ( (state->carrier == 1) && ((time(NULL) - state->last_vc_sync_time) < 2)) /This probably doesn't work correctly since we update time check when playing raw audio
// {
// memset (state->analog_out, 0, sizeof(state->analog_out));
// state->analog_sample_counter = 0;
// } //This is the root cause of issue listed above, will evaluate further at a later time for a more elegant solution, or determine if anything is negatively impacted by removing this
//sanity check to prevent an overflow
if (state->analog_sample_counter > 959)
state->analog_sample_counter = 959;
state->analog_out[state->analog_sample_counter++] = sample;
if (state->analog_sample_counter == 960)
{
//get an rms value if not using the rtl built in version
if (opts->audio_in_type != 3 && opts->monitor_input_audio == 1)
opts->rtl_rms = raw_rms(state->analog_out, 960, 1);
//raw wav file saving -- only write when not NXDN, dPMR, or M17 due to noise that can cause tons of false positives when no sync
if (opts->wav_out_raw != NULL && opts->frame_nxdn48 == 0 && opts->frame_nxdn96 == 0 && opts->frame_dpmr == 0 && opts->frame_m17 == 0)
{
sf_write_short(opts->wav_out_raw, state->analog_out, 960);
sf_write_sync (opts->wav_out_raw);
}
//low pass filter
if (opts->use_lpf == 1)
lpf(state, state->analog_out, 960);
//high pass filter
if (opts->use_hpf == 1)
hpf (state, state->analog_out, 960);
//pass band filter
if (opts->use_pbf == 1)
pbf(state, state->analog_out, 960);
//manual gain control
if (opts->audio_gainA > 0.0f)
analog_gain (opts, state, state->analog_out, 960);
//automatic gain control
else
agsm(opts, state, state->analog_out, 960);
//Running RMS after filtering does remove the analog spike from the RMS value
//but noise floor noise will still produce higher values
// if (opts->audio_in_type != 3 && opts->monitor_input_audio == 1)
// opts->rtl_rms = raw_rms(state->analog_out, 960, 1);
//seems to be working now, but RMS values are lower on actual analog signal than on no signal but noise
if ( (opts->rtl_rms > opts->rtl_squelch_level) && opts->monitor_input_audio == 1 && state->carrier == 0 ) //added carrier check here in lieu of disabling it above
{
if (opts->audio_out_type == 0)
pa_simple_write(opts->pulse_raw_dev_out, state->analog_out, 960*2, NULL);
if (opts->audio_out_type == 8)
udp_socket_blasterA (opts, state, 960*2, state->analog_out);
//NOTE: Worked okay earlier in Cygwin, so should be fine -- can only operate at 48k1, else slow mode lag
//This one will only operate when OSS 48k1 (when both input and output are OSS audio)
if (opts->audio_out_type == 5 && opts->pulse_digi_rate_out == 48000 && opts->pulse_digi_out_channels == 1)
write (opts->audio_out_fd, state->analog_out, 960*2);
//STDOUT, but only when operating at 48k1 (no go just yet)
// if (opts->audio_out_type == 1 && opts->pulse_digi_rate_out == 48000 && opts->pulse_digi_out_channels == 1)
// write (opts->audio_out_fd, state->analog_out, 960*2);
//OSS 8k1 (no go just yet)
// if (opts->audio_out_type == 2 && opts->pulse_digi_rate_out == 48000 && opts->pulse_digi_out_channels == 1)
// write (opts->audio_out_fd, state->analog_out, 960*2);
//test updating the sync time, so we can hold here while trunking or scanning
state->last_cc_sync_time = time(NULL);
state->last_vc_sync_time = time(NULL);
}
//raw wav file saving -- save the WAV file samples before we apply filtering to them
// if (opts->wav_out_raw != NULL && opts->frame_nxdn48 == 0 && opts->frame_nxdn96 == 0 && opts->frame_dpmr == 0 && opts->frame_m17 == 0)
// {
// sf_write_short(opts->wav_out_raw, state->analog_out, 960);
// sf_write_sync (opts->wav_out_raw);
// }
memset (state->analog_out, 0, sizeof(state->analog_out));
state->analog_sample_counter = 0;
}
}
if (have_sync == 1)
{
//sanity check to prevent an overflow
if (state->analog_sample_counter > 959)
state->analog_sample_counter = 959;
state->analog_out[state->analog_sample_counter++] = sample;
if (state->analog_sample_counter == 960)
{
//raw wav file saving -- file size on this blimps pretty fast 1 min ~= 6 MB; 1 hour ~= 360 MB;
if (opts->wav_out_raw != NULL)
{
sf_write_short(opts->wav_out_raw, state->analog_out, 960);
sf_write_sync (opts->wav_out_raw);
}
//zero out and reset counter
memset (state->analog_out, 0, sizeof(state->analog_out));
state->analog_sample_counter = 0;
}
}
if (opts->use_cosine_filter)
{
if ( (state->lastsynctype >= 10 && state->lastsynctype <= 13) || state->lastsynctype == 32 || state->lastsynctype == 33
|| state->lastsynctype == 34 || state->lastsynctype == 30 || state->lastsynctype == 31)
{
sample = dmr_filter(sample);
}
else if (state->lastsynctype == 8 || state->lastsynctype == 9 ||state->lastsynctype == 16 || state->lastsynctype == 17)
{
sample = m17_filter(sample);
}
else if (
state->lastsynctype == 20 || state->lastsynctype == 21 ||
state->lastsynctype == 22 || state->lastsynctype == 23 ||
state->lastsynctype == 24 || state->lastsynctype == 25 ||
state->lastsynctype == 26 || state->lastsynctype == 27 ||
state->lastsynctype == 28 || state->lastsynctype == 29 ) //||
//state->lastsynctype == 35 || state->lastsynctype == 36) //phase 2 C4FM disc tap input
{
//if(state->samplesPerSymbol == 20)
if(opts->frame_nxdn48 == 1)
{
sample = nxdn_filter(sample);
}
//else if (state->lastsynctype >= 20 && state->lastsynctype <=27) //this the right range?
else if (opts->frame_dpmr == 1)
{
sample = dpmr_filter(sample);
}
else if (state->samplesPerSymbol == 8) //phase 2 cqpsk
{
//sample = dmr_filter(sample); //work on filter later
}
else
{
sample = dmr_filter(sample);
}
}
}
if ((sample > state->max) && (have_sync == 1) && (state->rf_mod == 0))
{
sample = state->max;
}
else if ((sample < state->min) && (have_sync == 1) && (state->rf_mod == 0))
{
sample = state->min;
}
if (sample > state->center)
{
if (state->lastsample < state->center)
{
state->numflips += 1;
}
if (sample > (state->maxref * 1.25))
{
if (state->lastsample < (state->maxref * 1.25))
{
state->numflips += 1;
}
if ((state->jitter < 0) && (state->rf_mod == 1))
{ // first spike out of place
state->jitter = i;
}
if ((opts->symboltiming == 1) && (have_sync == 0) && (state->lastsynctype != -1))
{
fprintf (stderr, "O");
}
}
else
{
if ((opts->symboltiming == 1) && (have_sync == 0) && (state->lastsynctype != -1))
{
fprintf (stderr, "+");
}
if ((state->jitter < 0) && (state->lastsample < state->center) && (state->rf_mod != 1))
{ // first transition edge
state->jitter = i;
}
}
}
else
{ // sample < 0
if (state->lastsample > state->center)
{
state->numflips += 1;
}
if (sample < (state->minref * 1.25))
{
if (state->lastsample > (state->minref * 1.25))
{
state->numflips += 1;
}
if ((state->jitter < 0) && (state->rf_mod == 1))
{ // first spike out of place
state->jitter = i;
}
if ((opts->symboltiming == 1) && (have_sync == 0) && (state->lastsynctype != -1))
{
fprintf (stderr, "X");
}
}
else
{
if ((opts->symboltiming == 1) && (have_sync == 0) && (state->lastsynctype != -1))
{
fprintf (stderr, "-");
}
if ((state->jitter < 0) && (state->lastsample > state->center) && (state->rf_mod != 1))
{ // first transition edge
state->jitter = i;
}
}
}
if (state->samplesPerSymbol == 20) //nxdn 4800 baud 2400 symbol rate
{
// if ((i >= 9) && (i <= 11))
if ((i >= 7) && (i <= 13)) //7, 13 working good on multiple nxdn48, fewer random errors
{
sum += sample;
count++;
}
}
if (state->samplesPerSymbol == 5) //provoice or gfsk
{
if (i == 2)
{
sum += sample;
count++;
}
}
else
{
if (state->rf_mod == 0)
{
// 0: C4FM modulation
//EXPERIMENTAL: manipulate the left edge depending on sync type
//TODO: See if we can manipulate this a bit more based on AFC or similar type function
int l_edge = 2;
int r_edge = 2;
if (state->synctype == 30 || state->synctype == 31) l_edge = 1; //YSF (need to test with new recordings at BW:12000)
else if ((state->lastsynctype >= 10 && state->lastsynctype <= 13) || state->lastsynctype == 32 || state->lastsynctype == 33) l_edge = 1;
else l_edge = 2; //P25 and NXDN96 perform better, DMR doesn't seem to care too much, but may favor 1 and not 2
if ((i >= state->symbolCenter - l_edge) && (i <= state->symbolCenter + r_edge))
{
sum += sample;
count++;
}
//debug: show what was selected for the left edge
// fprintf (stderr, " L:%d", l_edge);
#ifdef TRACE_DSD
if (i == state->symbolCenter - 1) {
state->debug_sample_left_edge = state->debug_sample_index - 1;
}
if (i == state->symbolCenter + 2) {
state->debug_sample_right_edge = state->debug_sample_index - 1;
}
#endif
}
else if (state->rf_mod == 1) //QPSK
{
//going one left, two on the right for QPSK, local system seems to favor that
//with same PPM setting used on C4FM, unsure if that works on other systems, just
//testing things now, not like QPSK works well here anyways
if ((i == state->symbolCenter - 1) || (i == state->symbolCenter + 2)) //1,2
{
sum += sample;
count++;
}
}
else //GFSK
{
// 1: QPSK modulation
// 2: GFSK modulation
// Note: this has been changed to use an additional symbol to the left
// On the p25_raw_unencrypted.flac it is evident that the timing
// comes one sample too late.
// This change makes a significant improvement in the BER, at least for
// this file.
// if ((i == state->symbolCenter) || (i == state->symbolCenter + 1))
//GFSK should ideally be the same on the left and right, I would think
//I have not observed a system that would benefit from this alignment
if ((i == state->symbolCenter - 1) || (i == state->symbolCenter + 1))
{
sum += sample;
count++;
}
//MISC Notes: When using RTL input, NXDN96 at BW:16 NXDN48 at BW:12
//I am beginning to suspect that RTL input is halfing the bandwidth, which
//is my original consclusion back when I was working on it last, unsure
//or the best course on that, could just be a rtl_fm flaw
//P25 may also work better at 16 as well now, hard to tell the difference since both are good
//DMR may not be any different on 12 or 16
//most likely though, this all will just depend on signal stregth more than anything
//as to how much BW you should set
#ifdef TRACE_DSD
//if (i == state->symbolCenter) {
if (i == state->symbolCenter - 1) {
state->debug_sample_left_edge = state->debug_sample_index - 1;
}
if (i == state->symbolCenter + 1) {
state->debug_sample_right_edge = state->debug_sample_index - 1;
}
#endif
}
}
state->lastsample = sample;
}
symbol = (sum / count);
if ((opts->symboltiming == 1) && (have_sync == 0) && (state->lastsynctype != -1))
{
if (state->jitter >= 0)
{
fprintf (stderr, " %i\n", state->jitter);
}
else
{
fprintf (stderr, "\n");
}
}
#ifdef TRACE_DSD
if (state->samplesPerSymbol == 10) {
float left, right;
if (state->debug_label_file == NULL) {
state->debug_label_file = fopen ("pp_label.txt", "w");
}
left = state->debug_sample_left_edge / SAMPLE_RATE_IN;
right = state->debug_sample_right_edge / SAMPLE_RATE_IN;
if (state->debug_prefix != '\0') {
if (state->debug_prefix == 'I') {
fprintf(state->debug_label_file, "%f\t%f\t%c%c %i\n", left, right, state->debug_prefix, state->debug_prefix_2, symbol);
} else {
fprintf(state->debug_label_file, "%f\t%f\t%c %i\n", left, right, state->debug_prefix, symbol);
}
} else {
fprintf(state->debug_label_file, "%f\t%f\t%i\n", left, right, symbol);
}
}
#endif
//test throttle on wav input files
if (opts->audio_in_type == 2)
{
if (state->use_throttle == 1) usleep(.003); //very environment specific, tuning to cygwin
}
//read op25/fme symbol bin files
if (opts->audio_in_type == 4)
{
//use fopen and read in a symbol, check op25 for clues
if(opts->symbolfile == NULL)
{
fprintf(stderr, "Error Opening File %s\n", opts->audio_in_dev); //double check this
return(-1);
}
state->symbolc = fgetc(opts->symbolfile);
//experimental throttle
if (state->use_throttle == 1)
{
// useconds_t stime = state->symbol_throttle;
// usleep(stime);
usleep(.003); //very environment specific, tuning to cygwin
}
//fprintf(stderr, "%d", state->symbolc);
if( feof(opts->symbolfile) )
{
// opts->audio_in_type = 0; //switch to pulse after playback, ncurses terminal can initiate replay if wanted
fclose(opts->symbolfile);
fprintf (stderr, "\nEnd of %s\n", opts->audio_in_dev);
//open pulse input if we are pulse output AND using ncurses terminal
if (opts->audio_out_type == 0 && opts->use_ncurses_terminal == 1)
{
opts->audio_in_type = 0; //set input type
openPulseInput(opts); //open pulse input
}
//else cleanup and exit
else
{
cleanupAndExit(opts, state);
}
}
//assign symbol/dibit values based on modulation type
if (state->rf_mod == 2) //GFSK
{
symbol = state->symbolc;
if (state->symbolc == 0 )
{
symbol = -3; //-1
}
if (state->symbolc == 1 )
{
symbol = -1; //-3
}
}
else //everything else
{
if (state->symbolc == 0)
{
symbol = 1; //-1
}
if (state->symbolc == 1)
{
symbol = 3; //-3
}
if (state->symbolc == 2)
{
symbol = -1; //1
}
if (state->symbolc == 3)
{
symbol = -3; //3
}
}
}
state->symbolcnt++;
return (symbol);
}
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