mirror of https://github.com/wb2osz/direwolf.git
335 lines
7.6 KiB
C
335 lines
7.6 KiB
C
//
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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 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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* Module: gen_tone.c
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*
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* Purpose: Convert bits to AFSK for writing to .WAV sound file
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* or a sound device.
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*
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*
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*---------------------------------------------------------------*/
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#include <stdio.h>
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#include <math.h>
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#include <unistd.h>
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#include <string.h>
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#include <stdlib.h>
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#include <assert.h>
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#include "direwolf.h"
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#include "audio.h"
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#include "gen_tone.h"
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#include "textcolor.h"
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// Properties of the digitized sound stream & modem.
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static struct audio_s modem;
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/*
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* 8 bit samples are unsigned bytes in range of 0 .. 255.
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*
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* 16 bit samples are signed short in range of -32768 .. +32767.
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*/
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/* Constants after initialization. */
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#define TICKS_PER_CYCLE ( 256.0 * 256.0 * 256.0 * 256.0 )
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static int ticks_per_sample; /* same for all channels. */
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static int ticks_per_bit[MAX_CHANS];
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static int f1_change_per_sample[MAX_CHANS];
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static int f2_change_per_sample[MAX_CHANS];
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static short sine_table[256];
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/* Accumulators. */
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static unsigned int tone_phase[MAX_CHANS]; // Phase accumulator for tone generation.
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// Upper bits are used as index into sine table.
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static int bit_len_acc[MAX_CHANS]; // To accumulate fractional samples per bit.
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static int lfsr[MAX_CHANS]; // Shift register for scrambler.
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/*------------------------------------------------------------------
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*
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* Name: gen_tone_init
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*
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* Purpose: Initialize for AFSK tone generation which might
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* be used for RTTY or amateur packet radio.
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*
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* Inputs: pp - Pointer to modem parameter structure, modem_s.
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*
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* The fields we care about are:
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*
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* samples_per_sec
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* baud
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* mark_freq
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* space_freq
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* samples_per_sec
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*
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* amp - Signal amplitude on scale of 0 .. 100.
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*
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* Returns: 0 for success.
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* -1 for failure.
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*
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* Description: Calculate various constants for use by the direct digital synthesis
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* audio tone generation.
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*
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*----------------------------------------------------------------*/
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static int amp16bit; /* for 9600 baud */
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int gen_tone_init (struct audio_s *pp, int amp)
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{
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int j;
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int chan = 0;
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/*
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* Save away modem parameters for later use.
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*/
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memcpy (&modem, pp, sizeof(struct audio_s));
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amp16bit = (32767 * amp) / 100;
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ticks_per_sample = (int) ((TICKS_PER_CYCLE / (double)modem.samples_per_sec ) + 0.5);
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for (chan = 0; chan < modem.num_channels; chan++) {
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ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / (double)modem.baud[chan] ) + 0.5);
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f1_change_per_sample[chan] = (int) (((double)modem.mark_freq[chan] * TICKS_PER_CYCLE / (double)modem.samples_per_sec ) + 0.5);
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f2_change_per_sample[chan] = (int) (((double)modem.space_freq[chan] * TICKS_PER_CYCLE / (double)modem.samples_per_sec ) + 0.5);
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tone_phase[chan] = 0;
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bit_len_acc[chan] = 0;
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lfsr[chan] = 0;
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}
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for (j=0; j<256; j++) {
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double a;
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int s;
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a = ((double)(j) / 256.0) * (2 * M_PI);
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s = (int) (sin(a) * 32767 * amp / 100.0);
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/* 16 bit sound sample is in range of -32768 .. +32767. */
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assert (s >= -32768 && s <= 32767);
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sine_table[j] = s;
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}
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return (0);
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} /* end gen_tone_init */
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/*-------------------------------------------------------------------
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*
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* Name: gen_tone_put_bit
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*
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* Purpose: Generate tone of proper duration for one data bit.
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*
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* Inputs: chan - Audio channel, 0 = first.
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*
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* dat - 0 for f1, 1 for f2.
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*
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* -1 inserts half bit to test data
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* recovery PLL.
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*
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* Assumption: fp is open to a file for write.
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*
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*--------------------------------------------------------------------*/
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void tone_gen_put_bit (int chan, int dat)
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{
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int cps = dat ? f2_change_per_sample[chan] : f1_change_per_sample[chan];
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unsigned short sam = 0;
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int x;
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if (dat < 0) {
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/* Hack to test receive PLL recovery. */
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bit_len_acc[chan] -= ticks_per_bit[chan];
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dat = 0;
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}
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if (modem.modem_type[chan] == SCRAMBLE) {
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x = (dat ^ (lfsr[chan] >> 16) ^ (lfsr[chan] >> 11)) & 1;
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lfsr[chan] = (lfsr[chan] << 1) | (x & 1);
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dat = x;
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}
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do {
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if (modem.modem_type[chan] == AFSK) {
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tone_phase[chan] += cps;
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sam = sine_table[(tone_phase[chan] >> 24) & 0xff];
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}
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else {
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// TODO: Might want to low pass filter this.
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sam = dat ? amp16bit : (-amp16bit);
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}
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/* Ship out an audio sample. */
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assert (modem.num_channels == 1 || modem.num_channels == 2);
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/* Generalize to allow 8 bits someday? */
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assert (modem.bits_per_sample == 16);
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if (modem.num_channels == 1)
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{
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audio_put (sam & 0xff);
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audio_put ((sam >> 8) & 0xff);
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}
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else if (modem.num_channels == 2)
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{
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if (chan == 1)
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{
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audio_put (0); // silent left
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audio_put (0);
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}
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audio_put (sam & 0xff);
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audio_put ((sam >> 8) & 0xff);
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//byte_count += 2;
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if (chan == 0)
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{
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audio_put (0); // silent right
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audio_put (0);
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}
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}
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/* Enough for the bit time? */
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bit_len_acc[chan] += ticks_per_sample;
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} while (bit_len_acc[chan] < ticks_per_bit[chan]);
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bit_len_acc[chan] -= ticks_per_bit[chan];
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}
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/*-------------------------------------------------------------------
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*
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* Name: main
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*
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* Purpose: Quick test program for above.
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*
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* Description: Compile like this for unit test:
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*
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* gcc -Wall -DMAIN -o gen_tone_test gen_tone.c audio.c textcolor.c
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*
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* gcc -Wall -DMAIN -o gen_tone_test.exe gen_tone.c audio_win.c textcolor.c -lwinmm
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*
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*--------------------------------------------------------------------*/
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#if MAIN
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int main ()
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{
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int n;
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int chan1 = 0;
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int chan2 = 1;
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int r;
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struct audio_s audio_param;
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/* to sound card */
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/* one channel. 2 times: one second of each tone. */
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memset (&audio_param, 0, sizeof(audio_param));
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strcpy (audio_param.adevice_in, DEFAULT_ADEVICE);
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strcpy (audio_param.adevice_out, DEFAULT_ADEVICE);
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audio_open (&audio_param);
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gen_tone_init (&audio_param, 100);
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for (r=0; r<2; r++) {
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for (n=0; n<audio_param.baud[0] * 2 ; n++) {
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tone_gen_put_bit ( chan1, 1 );
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}
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for (n=0; n<audio_param.baud[0] * 2 ; n++) {
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tone_gen_put_bit ( chan1, 0 );
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}
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}
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audio_close();
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/* Now try stereo. */
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memset (&audio_param, 0, sizeof(audio_param));
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strcpy (audio_param.adevice_in, DEFAULT_ADEVICE);
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strcpy (audio_param.adevice_out, DEFAULT_ADEVICE);
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audio_param.num_channels = 2;
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audio_open (&audio_param);
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gen_tone_init (&audio_param, 100);
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for (r=0; r<4; r++) {
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for (n=0; n<audio_param.baud[0] * 2 ; n++) {
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tone_gen_put_bit ( chan1, 1 );
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}
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for (n=0; n<audio_param.baud[0] * 2 ; n++) {
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tone_gen_put_bit ( chan1, 0 );
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}
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for (n=0; n<audio_param.baud[0] * 2 ; n++) {
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tone_gen_put_bit ( chan2, 1 );
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}
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for (n=0; n<audio_param.baud[0] * 2 ; n++) {
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tone_gen_put_bit ( chan2, 0 );
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}
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}
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audio_close();
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return(0);
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}
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#endif
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/* end gen_tone.c */
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