mirror of https://github.com/wb2osz/direwolf.git
502 lines
13 KiB
C
502 lines
13 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, 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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* 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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#include "fsk_demod_state.h" /* for MAX_FILTER_SIZE which might be overly generous for here. */
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/* but safe if we use same size as for receive. */
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#include "dsp.h"
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// Properties of the digitized sound stream & modem.
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static struct audio_s *save_audio_config_p;
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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[MAX_CHANS]; /* Same for both channels of same soundcard */
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/* because they have same sample rate */
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/* but less confusing to have for each channel. */
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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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* The K9NG/G3RUH output originally took a very simple and lazy approach.
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* We simply generated a square wave with + or - the desired amplitude.
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* This has a couple undesirable properties.
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*
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* - Transmitting a square wave would splatter into adjacent
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* channels of the transmitter doesn't limit the bandwidth.
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*
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* - The usual sample rate of 44100 is not a multiple of the
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* baud rate so jitter would be added to the zero crossings.
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*
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* Starting in version 1.2, we try to overcome these issues by using
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* a higher sample rate, low pass filtering, and down sampling.
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*
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* What sort of low pass filter would be appropriate? Intuitively,
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* we would expect a cutoff frequency somewhere between baud/2 and baud.
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* The current values were found with a small amount of trial and
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* error for best results. Future improvement is certainly possible.
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*/
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/*
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* For low pass filtering of 9600 baud data.
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*/
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/* Add sample to buffer and shift the rest down. */
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// TODO: Can we have one copy of these in dsp.h?
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static inline void push_sample (float val, float *buff, int size)
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{
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memmove(buff+1,buff,(size-1)*sizeof(float));
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buff[0] = val;
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}
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/* FIR filter kernel. */
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static inline float convolve (const float *data, const float *filter, int filter_size)
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{
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float sum = 0;
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int j;
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for (j=0; j<filter_size; j++) {
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sum += filter[j] * data[j];
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}
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return (sum);
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}
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static int lp_filter_size[MAX_CHANS];
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static float raw[MAX_CHANS][MAX_FILTER_SIZE] __attribute__((aligned(16)));
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static float lp_filter[MAX_CHANS][MAX_FILTER_SIZE] __attribute__((aligned(16)));
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static int resample[MAX_CHANS];
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#define UPSAMPLE 2
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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: audio_config_p - 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 *audio_config_p, 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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save_audio_config_p = audio_config_p;
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amp16bit = (32767 * amp) / 100;
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for (chan = 0; chan < MAX_CHANS; chan++) {
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if (audio_config_p->achan[chan].valid) {
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int a = ACHAN2ADEV(chan);
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ticks_per_sample[chan] = (int) ((TICKS_PER_CYCLE / (double)audio_config_p->adev[a].samples_per_sec ) + 0.5);
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ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / (double)audio_config_p->achan[chan].baud ) + 0.5);
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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);
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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);
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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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}
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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 must fit in range of -32768 .. +32767. */
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if (s < -32768) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("gen_tone_init: Excessive amplitude is being clipped.\n");
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s = -32768;
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}
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else if (s > 32767) {
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text_color_set(DW_COLOR_ERROR);
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dw_printf ("gen_tone_init: Excessive amplitude is being clipped.\n");
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s = 32767;
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}
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sine_table[j] = s;
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}
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/*
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* Low pass filter for 9600 baud.
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*/
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for (chan = 0; chan < MAX_CHANS; chan++) {
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if (audio_config_p->achan[chan].valid &&
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(audio_config_p->achan[chan].modem_type == MODEM_SCRAMBLE
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|| audio_config_p->achan[chan].modem_type == MODEM_BASEBAND)) {
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int a = ACHAN2ADEV(chan);
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int samples_per_sec; /* Might be scaled up! */
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int baud;
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/* These numbers were by trial and error. Need more investigation here. */
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float filter_len_bits = 88 * 9600.0 / (44100.0 * 2.0);
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/* Filter length in number of data bits. */
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/* Currently 9.58 */
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float lpf_baud = 0.8; /* Lowpass cutoff freq as fraction of baud rate */
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float fc; /* Cutoff frequency as fraction of sampling frequency. */
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samples_per_sec = audio_config_p->adev[a].samples_per_sec * UPSAMPLE;
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baud = audio_config_p->achan[chan].baud;
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ticks_per_sample[chan] = (int) ((TICKS_PER_CYCLE / (double)samples_per_sec ) + 0.5);
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ticks_per_bit[chan] = (int) ((TICKS_PER_CYCLE / (double)baud ) + 0.5);
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lp_filter_size[chan] = (int) (( filter_len_bits * (float)samples_per_sec / baud) + 0.5);
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if (lp_filter_size[chan] < 10) {
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text_color_set(DW_COLOR_DEBUG);
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dw_printf ("gen_tone_init: unexpected, chan %d, lp_filter_size %d < 10\n", chan, lp_filter_size[chan]);
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lp_filter_size[chan] = 10;
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}
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else if (lp_filter_size[chan] > MAX_FILTER_SIZE) {
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text_color_set(DW_COLOR_DEBUG);
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dw_printf ("gen_tone_init: unexpected, chan %d, lp_filter_size %d > %d\n", chan, lp_filter_size[chan], MAX_FILTER_SIZE);
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lp_filter_size[chan] = MAX_FILTER_SIZE;
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}
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fc = (float)baud * lpf_baud / (float)samples_per_sec;
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//text_color_set(DW_COLOR_DEBUG);
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//dw_printf ("gen_tone_init: chan %d, call gen_lowpass(fc=%.2f, , size=%d, )\n", chan, fc, lp_filter_size[chan]);
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gen_lowpass (fc, lp_filter[chan], lp_filter_size[chan], BP_WINDOW_HAMMING);
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}
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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 a = ACHAN2ADEV(chan); /* device for channel. */
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assert (save_audio_config_p->achan[chan].valid);
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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 (save_audio_config_p->achan[chan].modem_type == MODEM_SCRAMBLE) {
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int x;
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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 (save_audio_config_p->achan[chan].modem_type == MODEM_AFSK) {
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int sam;
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tone_phase[chan] += dat ? f2_change_per_sample[chan] : f1_change_per_sample[chan];
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sam = sine_table[(tone_phase[chan] >> 24) & 0xff];
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gen_tone_put_sample (chan, a, sam);
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}
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else {
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float fsam = dat ? amp16bit : (-amp16bit);
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/* version 1.2 - added a low pass filter instead of square wave out. */
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push_sample (fsam, raw[chan], lp_filter_size[chan]);
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resample[chan]++;
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if (resample[chan] >= UPSAMPLE) {
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int sam;
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sam = (int) convolve (raw[chan], lp_filter[chan], lp_filter_size[chan]);
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resample[chan] = 0;
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gen_tone_put_sample (chan, a, sam);
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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[chan];
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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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void gen_tone_put_sample (int chan, int a, int sam) {
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/* Ship out an audio sample. */
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assert (save_audio_config_p->adev[a].num_channels == 1 || save_audio_config_p->adev[a].num_channels == 2);
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/* Generalize to allow 8 bits someday? */
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assert (save_audio_config_p->adev[a].bits_per_sample == 16);
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// TODO: Should print message telling user to reduce output level.
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if (sam < -32767) sam = -32767;
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else if (sam > 32767) sam = 32767;
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if (save_audio_config_p->adev[a].num_channels == 1) {
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/* Mono */
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audio_put (a, sam & 0xff);
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audio_put (a, (sam >> 8) & 0xff);
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}
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else {
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if (chan == ADEVFIRSTCHAN(a)) {
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/* Stereo, left channel. */
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audio_put (a, sam & 0xff);
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audio_put (a, (sam >> 8) & 0xff);
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audio_put (a, 0);
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audio_put (a, 0);
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}
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else {
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/* Stereo, right channel. */
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audio_put (a, 0);
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audio_put (a, 0);
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audio_put (a, sam & 0xff);
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audio_put (a, (sam >> 8) & 0xff);
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}
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}
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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 my_audio_config;
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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 (&my_audio_config, 0, sizeof(my_audio_config));
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strlcpy (my_audio_config.adev[0].adevice_in, DEFAULT_ADEVICE, sizeof(my_audio_config.adev[0].adevice_in));
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strlcpy (my_audio_config.adev[0].adevice_out, DEFAULT_ADEVICE, sizeof(my_audio_config.adev[0].adevice_out));
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audio_open (&my_audio_config);
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gen_tone_init (&my_audio_config, 100);
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for (r=0; r<2; r++) {
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for (n=0; n<my_audio_config.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<my_audio_config.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 (&my_audio_config, 0, sizeof(my_audio_config));
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strlcpy (my_audio_config.adev[0].adevice_in, DEFAULT_ADEVICE, sizeof(my_audio_config.adev[0].adevice_in));
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strlcpy (my_audio_config.adev[0].adevice_out, DEFAULT_ADEVICE, , sizeof(my_audio_config.adev[0].adevice_out));
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my_audio_config.adev[0].num_channels = 2;
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audio_open (&my_audio_config);
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gen_tone_init (&my_audio_config, 100);
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for (r=0; r<4; r++) {
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for (n=0; n<my_audio_config.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<my_audio_config.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<my_audio_config.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<my_audio_config.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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