mirror of https://github.com/wb2osz/direwolf.git
Speed up 9600 demodulator.
This commit is contained in:
wb2osz
parent
049614d16c
commit
9b9744ba15
85
src/demod.c
85
src/demod.c
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@ -1,7 +1,7 @@
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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, 2016, 2019 John Langner, WB2OSZ
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// Copyright (C) 2011, 2012, 2013, 2014, 2015, 2016, 2019, 2021 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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@ -60,10 +60,6 @@
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static struct audio_s *save_audio_config_p;
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// TODO: temp experiment.
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static int zerostuff = 1; // temp experiment.
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// Current state of all the decoders.
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@ -676,12 +672,6 @@ int demod_init (struct audio_s *pa)
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strlcpy (save_audio_config_p->achan[chan].profiles, "+", sizeof(save_audio_config_p->achan[chan].profiles));
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}
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#ifdef TUNE_ZEROSTUFF
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zerostuff = TUNE_ZEROSTUFF;
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#endif
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/*
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* We need a minimum number of audio samples per bit time for good performance.
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* Easier to check here because demod_9600_init might have an adjusted sample rate.
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@ -696,26 +686,32 @@ int demod_init (struct audio_s *pa)
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if (save_audio_config_p->achan[chan].upsample == 0) {
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if (ratio < 5) {
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if (ratio < 4) {
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// example: 44100 / 9600 is 4.59
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// Big improvement with x2.
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// x4 seems to work the best.
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// The other parameters are not as touchy.
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// Might reduce on ARM if it takes too much CPU power.
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// This is extreme.
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// No one should be using a sample rate this low but
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// amazingly a recording with 22050 rate can be decoded.
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// 3 and 4 are the same. Need more tests.
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save_audio_config_p->achan[chan].upsample = 4;
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}
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else if (ratio < 5) {
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// example: 44100 / 9600 is 4.59
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// 3 is slightly better than 2 or 4.
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save_audio_config_p->achan[chan].upsample = 3;
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}
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else if (ratio < 10) {
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// 48000 / 9600 is 5.00
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// Need more research. Treat like above for now.
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// example: 48000 / 9600 = 5
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// 3 is slightly better than 2 or 4.
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save_audio_config_p->achan[chan].upsample = 4;
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save_audio_config_p->achan[chan].upsample = 3;
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}
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else if (ratio < 15) {
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// ...
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// ... guessing
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save_audio_config_p->achan[chan].upsample = 2;
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}
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@ -786,7 +782,8 @@ int demod_init (struct audio_s *pa)
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}
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demod_9600_init (save_audio_config_p->achan[chan].modem_type,
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save_audio_config_p->achan[chan].upsample * save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec,
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save_audio_config_p->adev[ACHAN2ADEV(chan)].samples_per_sec,
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save_audio_config_p->achan[chan].upsample,
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save_audio_config_p->achan[chan].baud, D);
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if (strchr(save_audio_config_p->achan[chan].profiles, '+') != NULL) {
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@ -924,7 +921,7 @@ __attribute__((hot))
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void demod_process_sample (int chan, int subchan, int sam)
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{
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float fsam;
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int k;
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//int k;
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struct demodulator_state_s *D;
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@ -1016,47 +1013,7 @@ void demod_process_sample (int chan, int subchan, int sam)
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case MODEM_AIS:
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default:
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if (zerostuff) {
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/* Literature says this is better if followed */
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/* by appropriate low pass filter. */
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/* So far, both are same in tests with different */
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/* optimal low pass filter parameters. */
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for (k=1; k<save_audio_config_p->achan[chan].upsample; k++) {
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demod_9600_process_sample (chan, 0, D);
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}
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demod_9600_process_sample (chan, sam * save_audio_config_p->achan[chan].upsample, D);
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}
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else {
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/* Linear interpolation. */
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static int prev_sam;
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switch (save_audio_config_p->achan[chan].upsample) {
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case 1:
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demod_9600_process_sample (chan, sam, D);
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break;
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case 2:
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demod_9600_process_sample (chan, (prev_sam + sam) / 2, D);
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demod_9600_process_sample (chan, sam, D);
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break;
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case 3:
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demod_9600_process_sample (chan, (2 * prev_sam + sam) / 3, D);
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demod_9600_process_sample (chan, (prev_sam + 2 * sam) / 3, D);
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demod_9600_process_sample (chan, sam, D);
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break;
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case 4:
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demod_9600_process_sample (chan, (3 * prev_sam + sam) / 4, D);
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demod_9600_process_sample (chan, (prev_sam + sam) / 2, D);
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demod_9600_process_sample (chan, (prev_sam + 3 * sam) / 4, D);
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demod_9600_process_sample (chan, sam, D);
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break;
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default:
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assert (0);
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break;
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}
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prev_sam = sam;
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}
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demod_9600_process_sample (chan, sam, save_audio_config_p->achan[chan].upsample, D);
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break;
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} /* switch modem_type */
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206
src/demod_9600.c
206
src/demod_9600.c
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@ -1,7 +1,7 @@
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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, 2015, 2019 John Langner, WB2OSZ
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// Copyright (C) 2011, 2012, 2013, 2015, 2019, 2021 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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@ -25,7 +25,8 @@
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*
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* Module: demod_9600.c
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*
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* Purpose: Demodulator for scrambled baseband encoding.
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* Purpose: Demodulator for baseband signal.
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* This is used for AX.25 (with scrambling) and IL2P without.
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*
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* Input: Audio samples from either a file or the "sound card."
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*
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@ -45,12 +46,14 @@
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#include <ctype.h>
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// Fine tuning for different demodulator types.
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// Don't remove this section. It is here for a reason.
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#define DCD_THRESH_ON 32 // Hysteresis: Can miss 0 out of 32 for detecting lock.
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// This is best for actual on-the-air signals.
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// Still too many brief false matches.
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#define DCD_THRESH_OFF 8 // Might want a little more fine tuning.
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#define DCD_GOOD_WIDTH 1024 // No more than 1024!!!
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#define DCD_THRESH_ON 32 // Hysteresis: Can miss 0 out of 32 for detecting lock.
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// This is best for actual on-the-air signals.
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// Still too many brief false matches.
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#define DCD_THRESH_OFF 8 // Might want a little more fine tuning.
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#define DCD_GOOD_WIDTH 1024 // No more than 1024!!!
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#include "fsk_demod_state.h" // Values above override defaults.
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#include "tune.h"
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@ -125,9 +128,12 @@ static inline float agc (float in, float fast_attack, float slow_decay, float *p
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*
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* Inputs: modem_type - Determines whether scrambling is used.
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*
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* samples_per_sec - Number of samples per second.
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* Might be upsampled in hopes of
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* reducing the PLL jitter.
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* samples_per_sec - Number of samples per second for audio.
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*
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* upsample - Factor to upsample the incoming stream.
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* After a lot of experimentation, I discovered that
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* it works better if the data is upsampled.
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* This reduces the jitter for PLL syncronization.
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*
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* baud - Data rate in bits per second.
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*
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@ -137,10 +143,13 @@ static inline float agc (float in, float fast_attack, float slow_decay, float *p
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*
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*----------------------------------------------------------------*/
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void demod_9600_init (enum modem_t modem_type, int samples_per_sec, int baud, struct demodulator_state_s *D)
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void demod_9600_init (enum modem_t modem_type, int original_sample_rate, int upsample, int baud, struct demodulator_state_s *D)
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{
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float fc;
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int j;
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if (upsample < 1) upsample = 1;
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if (upsample > 4) upsample = 4;
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memset (D, 0, sizeof(struct demodulator_state_s));
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D->modem_type = modem_type;
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@ -155,12 +164,13 @@ void demod_9600_init (enum modem_t modem_type, int samples_per_sec, int baud, st
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// case 'L': // upsample x4 with filtering.
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D->lp_filter_len_bits = 1.0;
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D->lp_filter_len_bits = 1.0; // -U4 = 61 4.59 samples/symbol
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// Works best with odd number in some tests. Even is better in others.
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//D->lp_filter_size = ((int) (0.5f * ( D->lp_filter_len_bits * (float)samples_per_sec / (float)baud ))) * 2 + 1;
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//D->lp_filter_size = ((int) (0.5f * ( D->lp_filter_len_bits * (float)original_sample_rate / (float)baud ))) * 2 + 1;
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D->lp_filter_size = (int) (( D->lp_filter_len_bits * (float)samples_per_sec / baud) + 0.5f);
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// Just round to nearest integer.
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D->lp_filter_size = (int) (( D->lp_filter_len_bits * (float)original_sample_rate / baud) + 0.5f);
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D->lp_window = BP_WINDOW_COSINE;
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@ -185,8 +195,11 @@ void demod_9600_init (enum modem_t modem_type, int samples_per_sec, int baud, st
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dw_printf ("samples per bit = %.1f\n", (double)samples_per_sec / baud);
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#endif
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// PLL needs to use the upsampled rate.
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D->pll_step_per_sample =
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(int) round(TICKS_PER_PLL_CYCLE * (double) baud / (double)samples_per_sec);
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(int) round(TICKS_PER_PLL_CYCLE * (double) baud / (double)(original_sample_rate * upsample));
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#ifdef TUNE_LP_WINDOW
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@ -217,13 +230,87 @@ void demod_9600_init (enum modem_t modem_type, int samples_per_sec, int baud, st
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D->pll_searching_inertia = TUNE_PLL_SEARCHING;
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#endif
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fc = (float)baud * D->lpf_baud / (float)samples_per_sec;
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// Initial filter (before scattering) is based on upsampled rate.
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fc = (float)baud * D->lpf_baud / (float)(original_sample_rate * upsample);
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//dw_printf ("demod_9600_init: call gen_lowpass(fc=%.2f, , size=%d, )\n", fc, D->lp_filter_size);
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gen_lowpass (fc, D->lp_filter, D->lp_filter_size, D->lp_window);
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gen_lowpass (fc, D->u.bb.lp_filter, D->lp_filter_size * upsample, D->lp_window);
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// New in 1.7 -
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// Use a polyphase filter to reduce the CPU load.
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// Originally I used zero stuffing to upsample.
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// Here is the general idea.
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//
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// Suppose the input samples are 1 2 3 4 5 6 7 8 9 ...
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// Filter coefficents are a b c d e f g h i ...
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//
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// With original sampling rate, the filtering would involve multiplying and adding:
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//
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// 1a 2b 3c 4d 5e 6f ...
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//
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// When upsampling by 3, each of these would need to be evaluated
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// for each audio sample:
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//
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// 1a 0b 0c 2d 0e 0f 3g 0h 0i ...
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// 0a 1b 0c 0d 2e 0f 0g 3h 0i ...
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// 0a 0b 1c 0d 0e 2f 0g 0h 3i ...
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//
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// 2/3 of the multiplies are always by a stuffed zero.
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// We can do this more efficiently by removing them.
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//
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// 1a 2d 3g ...
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// 1b 2e 3h ...
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// 1c 2f 3i ...
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//
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// We scatter the original filter across multiple shorter filters.
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// Each input sample cycles around them to produce the upsampled rate.
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//
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// a d g ...
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// b e h ...
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// c f i ...
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//
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// There are countless sources of information DSP but this one is unique
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// in that it is a college course that mentions APRS.
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// https://www2.eecs.berkeley.edu/Courses/EE123
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//
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// Was the effort worthwhile? Times on an RPi 3.
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//
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// command: atest -B9600 ~/walkabout9600[abc]-compressed*.wav
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//
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// These are 3 recordings of a portable system being carried out of
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// range and back in again. It is a real world test for weak signals.
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//
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// options num decoded seconds x realtime
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// 1.6 1.7 1.6 1.7 1.6 1.7
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// --- --- --- --- --- ---
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// -P- 171 172 23.928 17.967 14.9 19.9
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// -P+ 180 180 54.688 48.772 6.5 7.3
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// -P- -F1 177 178 32.686 26.517 10.9 13.5
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//
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// So, it turns out that -P+ doesn't have a dramatic improvement, only
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// around 4%, for drastically increased CPU requirements.
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// Maybe we should turn that off by default, especially for ARM.
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//
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int k = 0;
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for (int i = 0; i < D->lp_filter_size; i++) {
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D->u.bb.lp_polyphase_1[i] = D->u.bb.lp_filter[k++];
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if (upsample >= 2) {
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D->u.bb.lp_polyphase_2[i] = D->u.bb.lp_filter[k++];
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if (upsample >= 3) {
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D->u.bb.lp_polyphase_3[i] = D->u.bb.lp_filter[k++];
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if (upsample >= 4) {
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D->u.bb.lp_polyphase_4[i] = D->u.bb.lp_filter[k++];
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}
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}
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}
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}
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/* Version 1.2: Experiment with different slicing levels. */
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// Really didn't help that much because we should have a symmetrical signal.
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for (j = 0; j < MAX_SUBCHANS; j++) {
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slice_point[j] = 0.02f * (j - 0.5f * (MAX_SUBCHANS-1));
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@ -259,7 +346,7 @@ void demod_9600_init (enum modem_t modem_type, int samples_per_sec, int baud, st
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* been distorted by going thru voice transceivers not
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* intended to pass this sort of "audio" signal.
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*
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* Data is "scrambled" to reduce the amount of DC bias.
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* For G3RUH mode, data is "scrambled" to reduce the amount of DC bias.
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* The data stream must be unscrambled at the receiving end.
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*
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* We also have a digital phase locked loop (PLL)
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@ -276,6 +363,9 @@ void demod_9600_init (enum modem_t modem_type, int samples_per_sec, int baud, st
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* of the function to be called for each bit recovered
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* from the demodulator. For now, it's simply hard-coded.
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*
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* After experimentation, I found that this works better if
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* the original signal is upsampled by 2x or even 4x.
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*
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* References: 9600 Baud Packet Radio Modem Design
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* http://www.amsat.org/amsat/articles/g3ruh/109.html
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*
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@ -290,63 +380,57 @@ void demod_9600_init (enum modem_t modem_type, int samples_per_sec, int baud, st
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inline static void nudge_pll (int chan, int subchan, int slice, float demod_out, struct demodulator_state_s *D);
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__attribute__((hot))
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void demod_9600_process_sample (int chan, int sam, struct demodulator_state_s *D)
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{
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static void process_filtered_sample (int chan, float fsam, struct demodulator_state_s *D);
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__attribute__((hot))
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void demod_9600_process_sample (int chan, int sam, int upsample, struct demodulator_state_s *D)
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{
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float fsam;
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float amp;
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float demod_out;
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#if DEBUG4
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static FILE *demod_log_fp = NULL;
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static int log_file_seq = 0; /* Part of log file name */
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#endif
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int subchan = 0;
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int demod_data; /* Still scrambled. */
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assert (chan >= 0 && chan < MAX_CHANS);
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assert (subchan >= 0 && subchan < MAX_SUBCHANS);
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/*
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* Filters use last 'filter_size' samples.
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*
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* First push the older samples down.
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*
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* Finally, put the most recent at the beginning.
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*
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* Future project? Rather than shifting the samples,
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* it might be faster to add another variable to keep
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* track of the most recent sample and change the
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* indexing in the later loops that multiply and add.
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*/
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/* Scale to nice number for convenience. */
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/* Consistent with the AFSK demodulator, we'd like to use */
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/* only half of the dynamic range to have some headroom. */
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/* i.e. input range +-16k becomes +-1 here and is */
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/* displayed in the heard line as audio level 100. */
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fsam = sam / 16384.0;
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fsam = (float)sam / 16384.0f;
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#if defined(TUNE_ZEROSTUFF) && TUNE_ZEROSTUFF == 0
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// experiment - no filtering.
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// Low pass filter
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push_sample (fsam, D->u.bb.audio_in, D->lp_filter_size);
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amp = fsam;
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fsam = convolve (D->u.bb.audio_in, D->u.bb.lp_polyphase_1, D->lp_filter_size);
|
||||
process_filtered_sample (chan, fsam, D);
|
||||
if (upsample >= 2) {
|
||||
fsam = convolve (D->u.bb.audio_in, D->u.bb.lp_polyphase_2, D->lp_filter_size);
|
||||
process_filtered_sample (chan, fsam, D);
|
||||
if (upsample >= 3) {
|
||||
fsam = convolve (D->u.bb.audio_in, D->u.bb.lp_polyphase_3, D->lp_filter_size);
|
||||
process_filtered_sample (chan, fsam, D);
|
||||
if (upsample >= 4) {
|
||||
fsam = convolve (D->u.bb.audio_in, D->u.bb.lp_polyphase_4, D->lp_filter_size);
|
||||
process_filtered_sample (chan, fsam, D);
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
#else
|
||||
push_sample (fsam, D->raw_cb, D->lp_filter_size);
|
||||
|
||||
/*
|
||||
* Low pass filter to reduce noise yet pass the data.
|
||||
*/
|
||||
__attribute__((hot))
|
||||
static void process_filtered_sample (int chan, float fsam, struct demodulator_state_s *D)
|
||||
{
|
||||
|
||||
amp = convolve (D->raw_cb, D->lp_filter, D->lp_filter_size);
|
||||
#endif
|
||||
int subchan = 0;
|
||||
|
||||
/*
|
||||
* Version 1.2: Capture the post-filtering amplitude for display.
|
||||
|
|
@ -359,18 +443,18 @@ void demod_9600_process_sample (int chan, int sam, struct demodulator_state_s *D
|
|||
|
||||
// TODO: probably no need for this. Just use D->m_peak, D->m_valley
|
||||
|
||||
if (amp >= D->alevel_mark_peak) {
|
||||
D->alevel_mark_peak = amp * D->quick_attack + D->alevel_mark_peak * (1.0f - D->quick_attack);
|
||||
if (fsam >= D->alevel_mark_peak) {
|
||||
D->alevel_mark_peak = fsam * D->quick_attack + D->alevel_mark_peak * (1.0f - D->quick_attack);
|
||||
}
|
||||
else {
|
||||
D->alevel_mark_peak = amp * D->sluggish_decay + D->alevel_mark_peak * (1.0f - D->sluggish_decay);
|
||||
D->alevel_mark_peak = fsam * D->sluggish_decay + D->alevel_mark_peak * (1.0f - D->sluggish_decay);
|
||||
}
|
||||
|
||||
if (amp <= D->alevel_space_peak) {
|
||||
D->alevel_space_peak = amp * D->quick_attack + D->alevel_space_peak * (1.0f - D->quick_attack);
|
||||
if (fsam <= D->alevel_space_peak) {
|
||||
D->alevel_space_peak = fsam * D->quick_attack + D->alevel_space_peak * (1.0f - D->quick_attack);
|
||||
}
|
||||
else {
|
||||
D->alevel_space_peak = amp * D->sluggish_decay + D->alevel_space_peak * (1.0f - D->sluggish_decay);
|
||||
D->alevel_space_peak = fsam * D->sluggish_decay + D->alevel_space_peak * (1.0f - D->sluggish_decay);
|
||||
}
|
||||
|
||||
/*
|
||||
|
|
@ -381,12 +465,14 @@ void demod_9600_process_sample (int chan, int sam, struct demodulator_state_s *D
|
|||
* This works by looking at the minimum and maximum signal peaks
|
||||
* and scaling the results to be roughly in the -1.0 to +1.0 range.
|
||||
*/
|
||||
float demod_out;
|
||||
int demod_data; /* Still scrambled. */
|
||||
|
||||
demod_out = agc (amp, D->agc_fast_attack, D->agc_slow_decay, &(D->m_peak), &(D->m_valley));
|
||||
demod_out = agc (fsam, D->agc_fast_attack, D->agc_slow_decay, &(D->m_peak), &(D->m_valley));
|
||||
|
||||
// TODO: There is potential for multiple decoders with one filter.
|
||||
|
||||
//dw_printf ("peak=%.2f valley=%.2f amp=%.2f norm=%.2f\n", D->m_peak, D->m_valley, amp, norm);
|
||||
//dw_printf ("peak=%.2f valley=%.2f fsam=%.2f norm=%.2f\n", D->m_peak, D->m_valley, fsam, norm);
|
||||
|
||||
if (D->num_slicers <= 1) {
|
||||
|
||||
|
|
@ -435,7 +521,7 @@ void demod_9600_process_sample (int chan, int sam, struct demodulator_state_s *D
|
|||
|
||||
fprintf (demod_log_fp, "%.3f, %.3f, %.3f, %.3f, %.3f, %d, %.2f\n",
|
||||
fsam + 6,
|
||||
amp + 4,
|
||||
fsam + 4,
|
||||
D->m_peak + 4,
|
||||
D->m_valley + 4,
|
||||
demod_out + 2,
|
||||
|
|
|
|||
|
|
@ -6,9 +6,9 @@
|
|||
#include "fsk_demod_state.h"
|
||||
|
||||
|
||||
void demod_9600_init (enum modem_t modem_type, int samples_per_sec, int baud, struct demodulator_state_s *D);
|
||||
void demod_9600_init (enum modem_t modem_type, int original_sample_rate, int upsample, int baud, struct demodulator_state_s *D);
|
||||
|
||||
void demod_9600_process_sample (int chan, int sam, struct demodulator_state_s *D);
|
||||
void demod_9600_process_sample (int chan, int sam, int upsample, struct demodulator_state_s *D);
|
||||
|
||||
|
||||
|
||||
|
|
|
|||
|
|
@ -301,6 +301,8 @@ struct demodulator_state_s
|
|||
// //
|
||||
//////////////////////////////////////////////////////////////////////////////////
|
||||
|
||||
// TODO: Continue experiments with root raised cosine filter.
|
||||
// Either switch to that or take out all the related stuff.
|
||||
|
||||
struct bb_only_s {
|
||||
|
||||
|
|
@ -314,8 +316,15 @@ struct demodulator_state_s
|
|||
|
||||
float audio_in[MAX_FILTER_SIZE] __attribute__((aligned(16))); // Audio samples in.
|
||||
|
||||
// FIXME: use lp_filter
|
||||
float rrc_filter[MAX_FILTER_SIZE] __attribute__((aligned(16))); // RRC Low pass filter.
|
||||
|
||||
float lp_filter[MAX_FILTER_SIZE] __attribute__((aligned(16))); // Low pass filter.
|
||||
|
||||
// New in 1.7 - Polyphase filter to reduce CPU requirements.
|
||||
|
||||
float lp_polyphase_1[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
float lp_polyphase_2[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
float lp_polyphase_3[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
float lp_polyphase_4[MAX_FILTER_SIZE] __attribute__((aligned(16)));
|
||||
|
||||
float lp_1_iir_param; // very low pass filters to get DC offset.
|
||||
float lp_1_out;
|
||||
|
|
|
|||
Loading…
Reference in New Issue