//
// This file is part of Dire Wolf, an amateur radio packet TNC.
//
// Copyright (C) 2011, 2014, 2015 John Langner, WB2OSZ
//
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, either version 2 of the License, or
// (at your option) any later version.
//
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License for more details.
//
// You should have received a copy of the GNU General Public License
// along with this program. If not, see <http://www.gnu.org/licenses/>.
//
/*------------------------------------------------------------------
*
* Module: gen_tone.c
*
* Purpose: Convert bits to AFSK for writing to .WAV sound file
* or a sound device.
*
*
*---------------------------------------------------------------*/
#include <stdio.h>
#include <math.h>
#include <unistd.h>
#include <string.h>
#include <stdlib.h>
#include <assert.h>
#include "direwolf.h"
#include "audio.h"
#include "gen_tone.h"
#include "textcolor.h"
#include "fsk_demod_state.h" /* for MAX_FILTER_SIZE which might be overly generous for here. */
/* but safe if we use same size as for receive. */
#include "dsp.h"
// Properties of the digitized sound stream & modem.
static struct audio_s *save_audio_config_p;
/*
* 8 bit samples are unsigned bytes in range of 0 .. 255.
*
* 16 bit samples are signed short in range of -32768 .. +32767.
*/
/* Constants after initialization. */
#define TICKS_PER_CYCLE ( 256.0 * 256.0 * 256.0 * 256.0 )
static int ticks_per_sample[MAX_CHANS]; /* Same for both channels of same soundcard */
/* because they have same sample rate */
/* but less confusing to have for each channel. */
static int ticks_per_bit[MAX_CHANS];
static int f1_change_per_sample[MAX_CHANS];
static int f2_change_per_sample[MAX_CHANS];
static short sine_table[256];
/* Accumulators. */
static unsigned int tone_phase[MAX_CHANS]; // Phase accumulator for tone generation.
// Upper bits are used as index into sine table.
static int bit_len_acc[MAX_CHANS]; // To accumulate fractional samples per bit.
static int lfsr[MAX_CHANS]; // Shift register for scrambler.
/*
* The K9NG/G3RUH output originally took a very simple and lazy approach.
* We simply generated a square wave with + or - the desired amplitude.
* This has a couple undesirable properties.
*
* - Transmitting a square wave would splatter into adjacent
* channels of the transmitter doesn't limit the bandwidth.
*
* - The usual sample rate of 44100 is not a multiple of the
* baud rate so jitter would be added to the zero crossings.
*
* Starting in version 1.2, we try to overcome these issues by using
* a higher sample rate, low pass filtering, and down sampling.
*
* What sort of low pass filter would be appropriate? Intuitively,
* we would expect a cutoff frequency somewhere between baud/2 and baud.
* The current values were found with a small amount of trial and
* error for best results. Future improvement is certainly possible.
*/
/*
* For low pass filtering of 9600 baud data.
*/
/* Add sample to buffer and shift the rest down. */
// TODO: Can we have one copy of these in dsp.h?
static inline void push_sample (float val, float *buff, int size)
{
memmove(buff+1,buff,(size-1)*sizeof(float));
buff[0] = val;
}
/* FIR filter kernel. */
static inline float convolve (const float *data, const float *filter, int filter_size)
{
float sum = 0;
int j;
for (j=0; j<filter_size; j++) {
sum += filter[j] * data[j];
}
return (sum);
}
static int lp_filter_size[MAX_CHANS];
static float raw[MAX_CHANS][MAX_FILTER_SIZE] __attribute__((aligned(16)));
static float lp_filter[MAX_CHANS][MAX_FILTER_SIZE] __attribute__((aligned(16)));
static int resample[MAX_CHANS];
#define UPSAMPLE 2
/*------------------------------------------------------------------
*
* Name: gen_tone_init
*
* Purpose: Initialize for AFSK tone generation which might
* be used for RTTY or amateur packet radio.
*
* Inputs: audio_config_p - Pointer to modem parameter structure, modem_s.
*
* The fields we care about are:
*
* samples_per_sec
* baud
* mark_freq
* space_freq
* samples_per_sec
*
* amp - Signal amplitude on scale of 0 .. 100.
*
* Returns: 0 for success.
* -1 for failure.
*
* Description: Calculate various constants for use by the direct digital synthesis
* audio tone generation.
*
*----------------------------------------------------------------*/
static int amp16bit; /* for 9600 baud */
int gen_tone_init (struct audio_s *audio_config_p, int amp)
{
int j;
int chan = 0;
/*
* Save away modem parameters for later use.
*/
save_audio_config_p = audio_config_p;
amp16bit = (32767 * amp) / 100;
for (chan = 0; chan < MAX_CHANS; chan++) {
if (audio_config_p->achan[chan].valid) {
int a = ACHAN2ADEV(chan);
ticks_per_sample[chan] = (int) ((TICKS_PER_CYCLE / (double)audio_config_p->adev[a].samples_per_sec ) + 0.5);
ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / (double)audio_config_p->achan[chan].baud ) + 0.5);
f1_change_per_sample[chan] = (int) (((double)audio_config_p->achan[chan].mark_freq * TICKS_PER_CYCLE / (double)audio_config_p->adev[a].samples_per_sec ) + 0.5);
f2_change_per_sample[chan] = (int) (((double)audio_config_p->achan[chan].space_freq * TICKS_PER_CYCLE / (double)audio_config_p->adev[a].samples_per_sec ) + 0.5);
tone_phase[chan] = 0;
bit_len_acc[chan] = 0;
lfsr[chan] = 0;
}
}
for (j=0; j<256; j++) {
double a;
int s;
a = ((double)(j) / 256.0) * (2 * M_PI);
s = (int) (sin(a) * 32767 * amp / 100.0);
/* 16 bit sound sample is in range of -32768 .. +32767. */
assert (s >= -32768 && s <= 32767);
sine_table[j] = s;
}
/*
* Low pass filter for 9600 baud.
*/
for (chan = 0; chan < MAX_CHANS; chan++) {
if (audio_config_p->achan[chan].valid &&
(audio_config_p->achan[chan].modem_type == MODEM_SCRAMBLE
|| audio_config_p->achan[chan].modem_type == MODEM_BASEBAND)) {
int a = ACHAN2ADEV(chan);
int samples_per_sec; /* Might be scaled up! */
int baud;
/* These numbers were by trial and error. Need more investigation here. */
float filter_len_bits = 88 * 9600.0 / (44100.0 * 2.0);
/* Filter length in number of data bits. */
float lpf_baud = 0.8; /* Lowpass cutoff freq as fraction of baud rate */
float fc; /* Cutoff frequency as fraction of sampling frequency. */
samples_per_sec = audio_config_p->adev[a].samples_per_sec * UPSAMPLE;
baud = audio_config_p->achan[chan].baud;
ticks_per_sample[chan] = (int) ((TICKS_PER_CYCLE / (double)samples_per_sec ) + 0.5);
ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / (double)baud ) + 0.5);
lp_filter_size[chan] = (int) (( filter_len_bits * (float)samples_per_sec / baud) + 0.5);
if (lp_filter_size[chan] < 10 || lp_filter_size[chan] > MAX_FILTER_SIZE) {
text_color_set(DW_COLOR_ERROR);
dw_printf ("gen_tone_init: INTERNAL ERROR, chan %d, lp_filter_size %d\n", chan, lp_filter_size[chan]);
lp_filter_size[chan] = MAX_FILTER_SIZE / 2;
}
fc = (float)baud * lpf_baud / (float)samples_per_sec;
//text_color_set(DW_COLOR_DEBUG);
//dw_printf ("gen_tone_init: chan %d, call gen_lowpass(fc=%.2f, , size=%d, )\n", chan, fc, lp_filter_size[chan]);
gen_lowpass (fc, lp_filter[chan], lp_filter_size[chan], BP_WINDOW_HAMMING);
}
}
return (0);
} /* end gen_tone_init */
/*-------------------------------------------------------------------
*
* Name: gen_tone_put_bit
*
* Purpose: Generate tone of proper duration for one data bit.
*
* Inputs: chan - Audio channel, 0 = first.
*
* dat - 0 for f1, 1 for f2.
*
* -1 inserts half bit to test data
* recovery PLL.
*
* Assumption: fp is open to a file for write.
*
*--------------------------------------------------------------------*/
void tone_gen_put_bit (int chan, int dat)
{
int a = ACHAN2ADEV(chan); /* device for channel. */
assert (save_audio_config_p->achan[chan].valid);
if (dat < 0) {
/* Hack to test receive PLL recovery. */
bit_len_acc[chan] -= ticks_per_bit[chan];
dat = 0;
}
if (save_audio_config_p->achan[chan].modem_type == MODEM_SCRAMBLE) {
int x;
x = (dat ^ (lfsr[chan] >> 16) ^ (lfsr[chan] >> 11)) & 1;
lfsr[chan] = (lfsr[chan] << 1) | (x & 1);
dat = x;
}
do {
if (save_audio_config_p->achan[chan].modem_type == MODEM_AFSK) {
int sam;
tone_phase[chan] += dat ? f2_change_per_sample[chan] : f1_change_per_sample[chan];
sam = sine_table[(tone_phase[chan] >> 24) & 0xff];
gen_tone_put_sample (chan, a, sam);
}
else {
float fsam = dat ? amp16bit : (-amp16bit);
/* version 1.2 - added a low pass filter instead of square wave out. */
push_sample (fsam, raw[chan], lp_filter_size[chan]);
resample[chan]++;
if (resample[chan] >= UPSAMPLE) {
int sam;
sam = (int) convolve (raw[chan], lp_filter[chan], lp_filter_size[chan]);
resample[chan] = 0;
gen_tone_put_sample (chan, a, sam);
}
}
/* Enough for the bit time? */
bit_len_acc[chan] += ticks_per_sample[chan];
} while (bit_len_acc[chan] < ticks_per_bit[chan]);
bit_len_acc[chan] -= ticks_per_bit[chan];
}
void gen_tone_put_sample (int chan, int a, int sam) {
/* Ship out an audio sample. */
assert (save_audio_config_p->adev[a].num_channels == 1 || save_audio_config_p->adev[a].num_channels == 2);
/* Generalize to allow 8 bits someday? */
assert (save_audio_config_p->adev[a].bits_per_sample == 16);
if (sam < -32767) sam = -32767;
else if (sam > 32767) sam = 32767;
if (save_audio_config_p->adev[a].num_channels == 1) {
/* Mono */
audio_put (a, sam & 0xff);
audio_put (a, (sam >> 8) & 0xff);
}
else {
if (chan == ADEVFIRSTCHAN(a)) {
/* Stereo, left channel. */
audio_put (a, sam & 0xff);
audio_put (a, (sam >> 8) & 0xff);
audio_put (a, 0);
audio_put (a, 0);
}
else {
/* Stereo, right channel. */
audio_put (a, 0);
audio_put (a, 0);
audio_put (a, sam & 0xff);
audio_put (a, (sam >> 8) & 0xff);
}
}
}
/*-------------------------------------------------------------------
*
* Name: main
*
* Purpose: Quick test program for above.
*
* Description: Compile like this for unit test:
*
* gcc -Wall -DMAIN -o gen_tone_test gen_tone.c audio.c textcolor.c
*
* gcc -Wall -DMAIN -o gen_tone_test.exe gen_tone.c audio_win.c textcolor.c -lwinmm
*
*--------------------------------------------------------------------*/
#if MAIN
int main ()
{
int n;
int chan1 = 0;
int chan2 = 1;
int r;
struct audio_s my_audio_config;
/* to sound card */
/* one channel. 2 times: one second of each tone. */
memset (&my_audio_config, 0, sizeof(my_audio_config));
strcpy (my_audio_config.adev[0].adevice_in, DEFAULT_ADEVICE);
strcpy (my_audio_config.adev[0].adevice_out, DEFAULT_ADEVICE);
audio_open (&my_audio_config);
gen_tone_init (&my_audio_config, 100);
for (r=0; r<2; r++) {
for (n=0; n<my_audio_config.baud[0] * 2 ; n++) {
tone_gen_put_bit ( chan1, 1 );
}
for (n=0; n<my_audio_config.baud[0] * 2 ; n++) {
tone_gen_put_bit ( chan1, 0 );
}
}
audio_close();
/* Now try stereo. */
memset (&my_audio_config, 0, sizeof(my_audio_config));
strcpy (my_audio_config.adev[0].adevice_in, DEFAULT_ADEVICE);
strcpy (my_audio_config.adev[0].adevice_out, DEFAULT_ADEVICE);
my_audio_config.adev[0].num_channels = 2;
audio_open (&my_audio_config);
gen_tone_init (&my_audio_config, 100);
for (r=0; r<4; r++) {
for (n=0; n<my_audio_config.baud[0] * 2 ; n++) {
tone_gen_put_bit ( chan1, 1 );
}
for (n=0; n<my_audio_config.baud[0] * 2 ; n++) {
tone_gen_put_bit ( chan1, 0 );
}
for (n=0; n<my_audio_config.baud[0] * 2 ; n++) {
tone_gen_put_bit ( chan2, 1 );
}
for (n=0; n<my_audio_config.baud[0] * 2 ; n++) {
tone_gen_put_bit ( chan2, 0 );
}
}
audio_close();
return(0);
}
#endif
/* end gen_tone.c */