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Teensy_SDR.ino
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Teensy_SDR.ino
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/* Teensy SDR
* version 2016_02_22
* on my new laptop
* DD4WH mods included
*
* Experiment: freq_conv einbauen --> ok!
* AM switchoff (passthrough) in other modes --> ok! (sideband suppression superb)
* now only 47% processor load in SSB modes !!!
*
* ***************************************************
*
* this script only works with modified mixer.h in Audio library
* allow for negative gain !
*
*****************************************************
* processor load:
* SSB: 50% / 53 % (mem 12 / 19)
* AM: 63% / 66 % (mem 14 / 21)
* after elimination of biquad 3 & 4
* SSB: 45% / 47% (mem 14 / 19)
* AM: 57% / 60% (mem 14 / 19)
*
New features by DD4WH
+ spectrum display now with 44.1kHz bandwidth, uses complex FFT256IQ library by kpc
+ spectrum display with pixels instead of bars, LPF implemented for smooth appearance
+ added AM demodulation
+ added pseudo stereo AM
+ added DSB demodulation
+ added passband tuning, meaning easy adjustment of different bandwidths for upper and lower sideband (in DSB and StereoAM mode)
+ added automatic snap mode, which automatically searches the carrier in DSB mode when in 100Hz tuning step Tuning mode --> uses three-point-quadratic interpolation (thanks vladn)
+ added quadrature oscillators in the I and Q paths in order to translate IQ to baseband for demodulation, so that all filters work in baseband now (thanks Clint KA7OEI)
+ Hilbert and FIR filter bandwidths 1.8kHz, 2.6kHz, 3.6kHz, 4.4kHz, 6.5kHz, 8kHz, 11kHz --> all combined with adjustable IIR
+ more IIR filter bandwidths 1.8kHz, 2.3kHz, 2.6kHz, 3.1kHz, 3.6kHz, 4.0kHz, 4.4kHz, 5.4kHz, 6.5kHz, 8kHz, 11kHz
+ filter "switching" done automatically by smooth adjusting bandwidth with encoder (IIR and FIR/Hilberts automatically switched)
+ biquad 1: fixed bandwidths with postIIR filter (8th order IIR --> 4 cascaded biquad stages, inverted Chebychev)
+ (manually adjustable notch filter (8th order IIR biquad filter))
+ calibrated graphical S-meter implemented, S-value independent of analog gain ("RF gain") setting and digital volume control
+ tone control: bass & treble (postAudioprocessor)
+ recorder included, records audio from the SDR radio to RAW-files on the SD-card - up to 255 files, ca. 8MB per minute, playback also possible
+ tune step for LW & MW 9kHz, all other bands 5kHz
+ Real Time Clock integrated (backed up by battery)
+ date adjust integrated
+ frequency limits 12kHz - 36MHz implemented by using the new Si5351 library by Jason --> now SAQ on 17.1kHz is receivable ;-)
+ new 2015 version of Jason´s Si5351 library included: now tunes down to 12kHz, but below that Si5351 crashes, requires new boot
+ also new Si5351 library needed correction_factor for frequencies
+ Menu addition: adjust correction_constant for Si5351
+ Menu addition: adjust correction_factor for Si5351
+ Menu addition: Save settings to EEPROM
+ Menu addition: Load settings from EEPROM
+ bands array now also holds mode & USB bandwidth & LSB bandwidth for each band (and saved by EEPROM)
+ band is now global variable and is saved in EEPROM --> after load switches to last band saved
+ LPF automatic relay switching --> my hardware has 5 switchable elliptic Low Pass Filters: these are really needed for serious receive to supress 3rd, 5th, 7th and higher order harmonics
+ now five buttons (two additional ones)
+ added BAND down function, so buttons 1 and 2 now function as BAND DOWN and BAND UP
+ manual calibration of I + Q amplitude & phase --> phase adjustment mixes a little amount of I into Q or vice versa before passing the Hilbert Filters
+ button 5: Menu "RF gain" = manually adjust analog codec gain = "RF gain" in front of ADC (lineInLevel adjustable from 0-15 --> 0dB to 22.5dB in steps of 1.5dB)
+ Menu addition: choose among eleven different FFT Window Functions
+ added to EEPROM load and save: IQ amplitude & IQ phase calibrations, RF gain, mode, bandwidth USB, bandwidth LSB, FFT Window Function, LPF coeff for spectrum display
+ separate variables & voids: mode & audio bandwidth
+ show voltage Vcc (Menu2 entry) and check every 5 secs and warn if < 11 volts
+ larger bandwiths for IIR filters in AM mode to allow for sideband-selected AM reception
+ Freq_conv.cpp mit receiveWritable
+ very narrow manually adjustable notch filter with graphical indicator
+ frequency display: update only those frequency digits that have changed during tuning
+ time display: update only digits that have changed
Todo
+ sideband selected AM by shifting IF with an offset --> implemented in menu passband tuning
+ decimate and interpolate for biquad1 & biquad2 --> is it worth it? We already have reduced processor load to 47% . . .
+ LMS noise reduction, does it work even without decimation / interpolation ?
+ VFO A and B
+ switch between local time & UTC
+ Menu point: reset to factory settings
+ memories for stations with name & time read from txt.file on SD card
* simple software define radio using the Softrock transceiver
* the Teensy audio shield is used to capture and generate 16 bit audio
* audio processing is done by the Teensy 3.1
* simple UI runs on a 160x128 color TFT display - AdaFruit or Banggood knockoff which has a different LCD controller
* Copyright (C) 2014, 2015 Rich Heslip rheslip@hotmail.com
* History:
* 4/14 initial version by R Heslip VE3MKC
* 6/14 Loftur E. Jónasson TF3LJ/VE2LJX - filter improvements, inclusion of Metro, software AGC module, optimized audio processing, UI changes
* 1/15 RH - added encoder and SI5351 tuning library by Jason Milldrum <milldrum@gmail.com>
* - added HW AGC option which uses codec AGC module
* - added experimental waterfall display for CW
* 3/15 RH - updated code to Teensyduino 1.21 and audio lib 1.02
* - added a lot of #defines to neaten up the code
* - added another summer at output - switches audio routing at runtime, provides a nice way to adjust I/Q balance and do AGC/ALC
* - added CW I/Q oscillators for CW transmit mode
* - added SSB and CW transmit
* - restructured the code to facilitate TX/RX switching
* Todo:
* clean up some more of the hard coded HW and UI stuff
*
* 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 3 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/>.
*/
#define ver "v 2016.02.22"
#include <Time.h>
#include <TimeLib.h>
#include <Metro.h>
#include <Audio.h>
#include <Wire.h>
#include <SD.h>
#include <Encoder.h>
#include <EEPROM.h>
#include <si5351.h>
#include <Bounce.h>
//#include <Bounce2.h>
#include <Adafruit_GFX.h> // LCD Core graphics library
//#include <Adafruit_QDTech.h>// 1.8" TFT Module using Samsung S6D02A1 chip
#include <Adafruit_ST7735.h> // Hardware-specific library
#include <SPI.h>
#include "analyze_fft256iq.h" // new complex FFT library for full bandwidth spectrum from I & Q --> 44.1kHz
#include "AM_demod.h"
#include "freq_conv.h"
#include "filters.h"
#include "display.h"
#include "iir_18.h"
#include "iir_23.h"
#include "iir_26.h"
#include "iir_31.h"
#include "iir_36.h"
#include "iir_40.h"
#include "iir_44.h"
#include "iir_54.h"
#include "iir_65.h"
#include "iir_80.h"
#include "iir_110.h"
void sinewave(void);
extern void agc(void); // RX agc function
extern void setup_display(void);
extern void init_display(void);
//extern void show_spectrum(void); // spectrum display draw
extern void show_spectrum(float line_gain, float LPFcoeff); // spectrum display draw
extern void show_waterfall(void); // waterfall display
//extern void show_bandwidth(int filterwidth); // show filter bandwidth
extern void show_radiomode(String mode); // show filter bandwidth
extern void show_band(String bandname); // show band
extern void show_frequency(long int freq); // show frequency
extern void show_tunestep (String S);
extern void show_bandwidth (int M, long int FU, long int FL);
extern void show_notch (int notchF, int MODE);
time_t getTeensy3Time()
{
return Teensy3Clock.get();
}
#define Schedule_length 3
//#define DEBUG
//#define DEBUG_TIMING // for timing execution - monitor HW pin
// SW configuration defines
// don't use more than one AGC!
//#define SW_AGC // define for Loftur's SW AGC - this has to be tuned carefully for your particular implementation
// codec hardware AGC works but it looks at the entire input bandwidth
// ie codec HW AGC works on the strongest signal, not necessarily what you are listening to
// it should work well for ALC (mic input) though
//#define HW_AGC // define for codec AGC
// #define CW_WATERFALL // define for experimental CW waterfall - needs faster update rate
#define AUDIO_STATS // shows audio library CPU utilization etc on serial console
// band selection stuff
struct band {
long freq; // frequency in Hz
// unsigned long freq; // frequency in Hz
String name; // name of band
int mode; // sideband & bandwidth
int bandwidthU;
int bandwidthL;
int RFgain;
};
// Estebans defs
//static int16_t IF_FREQ = 5515; // IF Oscillator frequency
int16_t IF_FREQ = 5515; // IF Oscillator frequency
float s_old = 0;
//static float32_t Osc_Q_buffer_i_f[AUDIO_BLOCK_SAMPLES];
//static float32_t Osc_I_buffer_i_f[AUDIO_BLOCK_SAMPLES];
float32_t Osc_Q_buffer_i_f[AUDIO_BLOCK_SAMPLES];
float32_t Osc_I_buffer_i_f[AUDIO_BLOCK_SAMPLES];
q15_t Osc_Q_buffer_i[AUDIO_BLOCK_SAMPLES];
q15_t Osc_I_buffer_i[AUDIO_BLOCK_SAMPLES];
//static bool flag = 0;
bool flag = 0;
bool freqconv_sw=1; // 1 = frequency conversion on; 0 = freqconv off
bool AM_pass = 1; // 1 = Am demodulation on; 0 = off
q15_t last_dc_level = 0;
//#define IF_FREQ 5515 //11029 //5515 // IF Oscillator frequency, already defined above!!
long calibration_factor = 10000000 ;// 10002285;
long calibration_constant = 3500;
unsigned long long hilfsf;
#define F_MAX 36000000 + IF_FREQ
#define F_MIN 12000 + IF_FREQ
#define BAND_LW 0 // these can be in any order but indexes need to be sequential for band switching code
#define BAND_MW 1
#define BAND_120M 2
#define BAND_90M 3
#define BAND_75M 4 // these can be in any order but indexes need to be sequential for band switching code
#define BAND_60M 5
#define BAND_49M 6
#define BAND_41M 7
#define BAND_31M 8
#define BAND_25M 9
#define BAND_22M 10
#define BAND_19M 11
#define BAND_16M 12
#define BAND_15M 13
#define BAND_13M 14
#define BAND_11M 15
#define FIRST_BAND BAND_LW
#define LAST_BAND BAND_15M
#define NUM_BANDS 16
// radio operation mode defines used for filter selections etc
#define modeAM 0
#define modeUSB 1
#define modeLSB 2
#define modeDSB 3
#define modeStereoAM 4
#define modeSAM 5
#define firstmode modeAM
#define lastmode modeStereoAM
#define startmode modeAM
#define MAX_BANDWIDTH 11000
// f, band, mode, bandwidth, RFgain
struct band bands[NUM_BANDS] = {
153000-IF_FREQ,"LW", modeAM, 3600,3600,0,
531000-IF_FREQ,"MW", modeAM, 3600,3600,0,
2485000-IF_FREQ,"120M", modeAM, 3600,3600,0,
3500000-IF_FREQ,"90M", modeLSB, 3600,3600,6,
3905000-IF_FREQ,"75M", modeAM, 3600,3600,4,
4750000-IF_FREQ,"60M", modeAM, 3600,3600,7,
5910000-IF_FREQ,"49M", modeAM, 3600,3600,0,
7120000-IF_FREQ,"41M", modeAM, 3600,3600,0,
9420000-IF_FREQ,"31M", modeAM, 3600,3600,0,
11735000-IF_FREQ,"25M", modeAM, 3600,3600,0,
13570000-IF_FREQ,"22M", modeAM, 3600,3600,0,
15140000-IF_FREQ,"19M", modeAM, 3600,3600,0,
17480000-IF_FREQ,"16M", modeAM, 3600,3600,0,
18900000-IF_FREQ,"15M", modeAM, 3600,3600,0,
21450000-IF_FREQ,"13M", modeAM, 3600,3600,0,
25670000-IF_FREQ,"11M", modeAM, 3600,3600,0
};
#define STARTUP_BAND BAND_LW //
int band = STARTUP_BAND;
//SPI connections for 1.8" display
const int8_t sclk = 5;
const int8_t mosi = 4;
const int8_t cs = 2;
const int8_t dc = 3;
const int8_t rst = 1;
const int8_t MenuSW = 20; // button 4 = menu switch: encoder tune or encoder BW adjust
const int8_t ModeSW = 26; // button 3 = USB/LSB & bandwidth, ehemals pin 20
const int8_t BandDOWNSW = 21; // button 1 = band selector down
const int8_t BandUPSW = 24; // button 2 = band selector up
const int8_t TuneSW = 8; // Optical Encoder pushbutton
const int8_t Button5 = 25; // Menu2 switch
const int8_t Band1 = 29; // band selection pins for LPF relays, used with 2N7000: HIGH means LPF is activated
const int8_t Band2 = 30; // always use only one LPF with HIGH, all others have to be LOW
const int8_t Band3 = 31;
const int8_t Band4 = 32;
const int8_t Band5 = 33; // attention, special pin, only use as output pin!
const int8_t VoltCheck = 28; // DAC for voltage check
// definitions for cursor positions display setup
const int8_t pos_x_date = 100;
const int8_t pos_y_date = 119; //91;
const int8_t pos_x_time = 0;
const int8_t pos_y_time = 119;//91;
const int8_t pos_x_station = 0;
const int8_t pos_y_station = 0;
const int8_t pos_x_frequency = 39; //100
const int8_t pos_y_frequency = 59; //119
const int8_t pos_x_menu = 100;
const int8_t pos_y_menu = 119;
#define pos 50 // position of spectrum display, has to be < 124 --> also to be defined in display.cpp
Adafruit_ST7735 tft = Adafruit_ST7735(cs, dc, mosi, sclk, rst); // soft SPI
#define BACKLIGHT 0 // backlight control signal
// Optical Encoder connections
Encoder tune(16, 17);
// clock generator
Si5351 si5351;
#define MASTER_CLK_MULT 4 // QSD frontend requires 4x clock
// various timers
Metro five_sec=Metro(5000); // Set up a 5 second Metro
Metro one_sec = Metro (1000);
Metro ms_100 = Metro(100); // Set up a 100ms Metro
Metro ms_50 = Metro(50); // Set up a 50ms Metro for polling switches
Metro lcd_upd =Metro(4); // Set up a Metro for spectrum display updates
Metro CW_sample =Metro(10); // Set up a Metro for LCD updates
Metro MP3_Smetertimer = Metro(200);
#ifdef CW_WATERFALL
Metro waterfall_upd =Metro(25); // Set up a Metro for waterfall updates
#endif
#define TUNE_STEP1 5000 // shortwave
#define TUNE_STEP2 1 // fine tuning
#define TUNE_STEP3 100 //
#define first_tunehelp 1
#define last_tunehelp 3
int tunehelp = 1;
int tunestep = TUNE_STEP1;
String tune_text="Fast Tune";
// Menus !
#define TUNING 0
#define BWADJUST 1
#define NOTCH 2
#define RECORDING 3
#define CALIBRATIONCONSTANT 4
#define IQADJUST 5
#define IQPHASEADJUST 6
#define SAVETOEEPROM 7
#define LOADFROMEEPROM 8
#define TIMEADJUST 9
#define DATEADJUST 10
#define CALIBRATIONFACTOR 11
#define FFTWINDOW 12
#define LPFSPECTRUM 13
#define BW_step 200
#define first_menu TUNING
#define last_menu LPFSPECTRUM
#define start_menu TUNING
int Menu = TUNING;
String menu_text = "Tuning";
// Menu2 !
#define SNAP 1
#define PASSBAND 2
#define RFGAIN 3
#define BASS 4
#define TREBLE 5
//#define MP3 5
#define VOLTAGE 6
#define VERSION 7
//#define NOTCH 8
#define first_menu2 TUNING
#define last_menu2 VERSION //NOTCH
#define start_menu2 TUNING
int Menu2 = start_menu2;
float bass = 0.4;
float bass_help = bass * 100;
float treble = 0.0;
float treble_help = treble *100;
int notchF = 200; // if 200, notch is off, if >= 400, notch is on
int notchQ = 10;
//bool notchswitch = 0; // 0 = notch filter off; 1 = notchfilter on
//bool notchside = 0; // 0 = LSB; 1 = USB
float LPFcoeff = 0.3; // used for low pass filtering the spectrum display
float LPFcoeffhelp = LPFcoeff * 100;
int passbandBW = 0;
int maximum = 0;
int maxbin = 0;
int Lbin = 0;
int Ubin = 0;
int links = 0;
int rechts = 0;
float delta = 0;
float binBW = AUDIO_SAMPLE_RATE_EXACT / 256.0;
float bin1 = 0;
float bin2 = 0;
float bin3 = 0;
int posbin = 159; // for Rx above IF
// int posbin = 95; // for Rx below IF
int helpmin; // definitions for time and date adjust - Menu
int helphour;
int helpday;
int helpmonth;
int helpyear;
int helpsec;
uint8_t hour10_old;
uint8_t hour1_old;
uint8_t minute10_old;
uint8_t minute1_old;
uint8_t second10_old;
uint8_t second1_old;
bool timeflag = 0;
float Q_gain=1000;
float I_gain=1000;
float I_help = 1.0;
float Q_help = 1.0;
float Q_in_I = 0.000;
float Q_in_I_help = 0;
float I_in_Q = 0.000;
float I_in_Q_help = 0;
float IQcounter = 0;
#define INITIAL_VOLUME 0.8 // 0-1.0 output volume on startup
float vol = INITIAL_VOLUME;
//int line_in_gain = 0; // 0 --> 0 dB, 15 --> 22.3 dB of analog gain in front of the ADC
//int line_in_gain_help;
int Window_FFT = 0;
String FFT_STRING;
float Volt;
int RECORDFLAG = 0;
/* ############################################################
* definitions for MP3 Player & Recorder
* MP3-Player (Frank Bösing) did not work properly, memory issues ??
* it worked as standalone without SDR radio and with 1024FFT
* ###########################################################
*/
int audiomem = 0;
int Rec_Mode = 0; // Recorder: 0 = stopped, 1 = recording, 2 = playing
File frec; // audio is recorded to this file first
int playfile = 0;
//int recfile = -1;
// buttons in RECORD Mode
Bounce bouncer1 = Bounce(BandUPSW, 50); //play
Bounce bouncer3= Bounce(BandDOWNSW, 50); //record
Bounce bouncer2 = Bounce(ModeSW, 50);//stop & switch to radio
Bounce bouncer4 = Bounce(MenuSW, 50);//previous track
Bounce bouncer5 = Bounce(Button5, 50); //next track
/* ############################################################
* end of definitions for MP3 Player
*
* ###########################################################
*/
// audio definitions
// Create the Audio components. These should be created in the
// order data flows, inputs/sources -> processing -> outputs
//
AudioInputI2S audioinput; // Audio Shield: mic or line-in
AudioMixer4 audio_in_I;
AudioMixer4 audio_in_Q;
AudioEffectFreqConv FreqConv;
AMDemod AM;
// FIR filters
AudioFilterFIR FIR_I;
AudioFilterFIR FIR_Q;
// IIR filters (= biquads)
AudioFilterBiquad biquad1;
AudioFilterBiquad biquad2;
AudioFilterBiquad biquad3;
AudioFilterBiquad biquad4;
AudioMixer4 Summer; // Summer (add inputs)
AudioAnalyzeFFT256IQ myFFT; // Spectrum Display complex FFT
/*
AudioSynthWaveform cosIF; // Local Oscillator
AudioSynthWaveform sinIF; // Local Oscillator
//AudioAnalyzePeak MP3_peak; // does not seem to work: bug internally in AudioAnalyzePeak ???
AudioEffectMultiply multiply_I_cos; // Mixer (multiply inputs)
AudioEffectMultiply multiply_Q_sin; // Mixer (multiply inputs)
AudioEffectMultiply multiply_I_sin; // Mixer (multiply inputs)
AudioEffectMultiply multiply_Q_cos; // Mixer (multiply inputs)
*/
AudioRecordQueue recorder; // to record audio from the radio to SD card
AudioMixer4 recmix;
//AudioMixer4 add_substract_I; // belongs to the complex NCO
//AudioMixer4 add_substract_Q; // belongs toe the complex NCO
AudioMixer4 USB;
AudioMixer4 LSB;
AudioMixer4 mixleft;
AudioMixer4 mixright;
AudioOutputI2S audioOutput; // Audio Shield: headphones & line-out
//AudioPlaySdWav playWAV1;
//AudioPlaySdMp3 playMp31;
//AudioPlaySdAac playAac1;
AudioPlaySdRaw playSd; // playback recorded files in RAW format
AudioControlSGTL5000 audioShield; // Create an object to control the audio shield.
//---------------------------------------------------------------------------------------------------------
// Create Audio connections to build a software defined Radio Receiver
//
// audio input and IQ amplitude and phase correction
AudioConnection c5(audioinput, 0, audio_in_I, 0);
AudioConnection c6(audioinput, 1, audio_in_Q, 0);
AudioConnection c7(audio_in_I, 0, audio_in_Q, 1);
AudioConnection c8(audio_in_Q, 0, audio_in_I, 1);
// complex FFT for spectrum display
AudioConnection c23(audioinput, 0, myFFT, 0);
AudioConnection c24(audioinput, 1, myFFT, 1);
AudioConnection c3(audio_in_I, 0, FreqConv, 0);
AudioConnection c4(audio_in_Q, 0, FreqConv, 1);
/*
// Quadrature oscillator for frequency translation of the I and Q into baseband
AudioConnection c3(audio_in_I, 0, multiply_I_cos, 0);
AudioConnection c4(audio_in_Q, 0, multiply_Q_cos, 0);
AudioConnection c9(audio_in_I, 0, multiply_I_sin, 0);
AudioConnection c10(audio_in_Q, 0, multiply_Q_sin, 0);
AudioConnection c11(cosIF, 0, multiply_I_cos, 1);
AudioConnection c12(cosIF, 0, multiply_Q_cos, 1);
AudioConnection c13(sinIF, 0, multiply_Q_sin, 1);
AudioConnection c14(sinIF, 0, multiply_I_sin, 1);
AudioConnection c15(multiply_I_cos, 0, add_substract_I, 0);
AudioConnection c16(multiply_Q_sin, 0, add_substract_I, 1);
AudioConnection c17(multiply_I_sin, 0, add_substract_Q, 0);
AudioConnection c18(multiply_Q_cos, 0, add_substract_Q, 1);
// Main filters: either Hilberts with +45/-45 degrees or standard FIR filters
AudioConnection c19(add_substract_I, 0, FIR_I,0);
AudioConnection c20(add_substract_Q, 0, FIR_Q,0);
//AudioConnection c19(add_substract_Q, 0, FIR_I,0);
//AudioConnection c20(add_substract_I, 0, FIR_Q,0);
*/
AudioConnection c19(FreqConv, 0, FIR_I,0);
AudioConnection c20(FreqConv, 1, FIR_Q,0);
// Standard AM demodulator
AudioConnection c21(FIR_I, 0, AM, 0);
AudioConnection c22(FIR_Q, 0, AM, 1);
AudioConnection c25a(AM, 0, USB, 2);
AudioConnection c25b(AM, 0, LSB, 2);
// connections between I and Q signals and the AM/LSB/USB/DSB switches (mixers)
AudioConnection c26a(FIR_I ,0, USB, 0);
AudioConnection c26b(FIR_I ,0, LSB, 0);
AudioConnection c27a(FIR_Q ,0, USB, 1);
AudioConnection c27b(FIR_Q ,0, LSB, 1);
// IIR filters separately for left and right channel (to enable "stereo AM" = pseudostereo DSB reception)
AudioConnection c28a(USB, 0, biquad1, 0); // right channel
AudioConnection c29a(biquad1, 0, biquad3, 0);
AudioConnection c28b(LSB, 0, biquad2, 0); // left channel
AudioConnection c29b(biquad2, 0, biquad4, 0);
//mp3
//AudioConnection patch1(playMp31,0,mixleft,0);
//AudioConnection patch2(playMp31,1,mixright,0);
//aac
//AudioConnection patch1a(playAac1, 0, mixleft, 1);
//AudioConnection patch2a(playAac1, 1, mixright, 1);
// play RAW files from SD (in mono)
AudioConnection patch3(playSd, 0, mixleft, 3);
AudioConnection patch4(playSd, 0, mixright, 3);
// outputs from the IIR filters go to both mixers (left & right)
AudioConnection c30a(biquad3,0, mixright, 0);
AudioConnection c30b(biquad3,0, mixleft, 0);
AudioConnection c31a(biquad4,0, mixright, 1);
AudioConnection c31b(biquad4,0, mixleft, 1);
// mix left & right channel for recorder
AudioConnection c101 (mixleft, 0, recmix, 1);
AudioConnection c102a (mixright, 0, recmix, 0);
// connect recorder to recmixer
AudioConnection c102b (recmix, 0, recorder, 0);
// stereo output connected to mixright & mixleft
AudioConnection c32(mixright,0, audioOutput, 1);
AudioConnection c33(mixleft,0, audioOutput, 0);
//---------------------------------------------------------------------------------------------------------
void setup()
{
Serial.begin(9600); // debug console
#ifdef DEBUG
while (!Serial) ; // wait for connection
Serial.println("initializing");
#endif
setSyncProvider(getTeensy3Time);
//setTime(19,55,0,3,5,2015);
pinMode(BACKLIGHT, INPUT_PULLUP); // yanks up display BackLight signal
pinMode(ModeSW, INPUT_PULLUP); // USB/LSB switch
pinMode(BandUPSW, INPUT_PULLUP); //
pinMode(BandDOWNSW, INPUT_PULLUP); //
pinMode(TuneSW, INPUT_PULLUP); // tuning rate = high
pinMode(MenuSW, INPUT_PULLUP); //
pinMode(Button5, INPUT_PULLUP); //
pinMode(Band1, OUTPUT); // LPF switches
pinMode(Band2, OUTPUT); //
pinMode(Band3, OUTPUT); //
pinMode(Band4, OUTPUT); //
pinMode(Band5, OUTPUT); //
// Audio connections require memory to work. For more
// detailed information, see the MemoryAndCpuUsage example
if (!digitalRead(BandDOWNSW)) {AudioMemory(21); audiomem = 21;} // this seems to be the figure that is the cause of the sometimes arbitrary loss of mirror rejection!? 32 does not work
else {AudioMemory(90); audiomem = 100;}
// SDR mode needs a max of 21, M4A playback needs a max of 11, recording needs > 70
// Enable the audio shield and set the output volume.
audioShield.enable();
audioShield.volume(INITIAL_VOLUME/2);
audioShield.audioPostProcessorEnable(); // enables the DAP chain of the codec post audio processing before the headphone out
#ifdef HW_AGC
/* COMMENTS FROM Teensy Audio library:
Valid values for dap_avc parameters
maxGain; Maximum gain that can be applied
0 - 0 dB
1 - 6.0 dB
2 - 12 dB
lbiResponse; Integrator Response
0 - 0 mS
1 - 25 mS
2 - 50 mS
3 - 100 mS
hardLimit
0 - Hard limit disabled. AVC Compressor/Expander enabled.
1 - Hard limit enabled. The signal is limited to the programmed threshold (signal saturates at the threshold)
threshold
floating point in range 0 to -96 dB
attack
floating point figure is dB/s rate at which gain is increased
decay
floating point figure is dB/s rate at which gain is reduced
*/
audioShield.autoVolumeControl(2,1,0,-30.0,3.0,70.0); // see comments above
audioShield.autoVolumeEnable();
#endif
// initialize the TFT and show signon message etc
SPI.setMOSI(7); // set up HW SPI for use with the audio card - alternate pins
SPI.setSCK(14);
init_display();
setup_display();
/* tft.fillRect(0,pos_y_frequency, 160,16,BLACK);
tft.setCursor(0,pos_y_frequency);
tft.setTextSize(2);
tft.setTextColor(WHITE);
tft.print("memory ");
tft.print(audiomem);
delay(1000);
tft.setTextSize(1);
setup_display();
*/
// If red button switch is pressed on boot, factory settings are loaded as start settings
//delay (1000);
//if (digitalRead(TuneSW))
EEPROMLOAD(); // get saved frequencies, modes and calibration from EEPROM
// set up initial band and frequency
show_band(bands[band].name);
show_tunestep(tune_text);
// here startup message
tft.fillRect(pos_x_frequency,pos_y_frequency, 160-pos_x_frequency,16,BLACK);
tft.setCursor(pos_x_frequency,pos_y_frequency);
tft.setTextSize(2);
tft.setTextColor(WHITE);
tft.print("DD4WH SDR");
tft.setCursor(pos_x_time,pos_y_time);
tft.setTextSize(1);
tft.setTextColor(WHITE);
tft.print (ver);
delay (2000);
tft.fillRect(pos_x_frequency,pos_y_frequency, 160-pos_x_frequency,16,BLACK); // erase for frequency display
tft.fillRect(pos_x_time, pos_y_time, 80, 8, BLACK); // erase for time display
audioShield.eqSelect (2); // Tone Control
audioShield.eqBands (bass, treble); // (float bass, float treble) in % -100 to +100
audioShield.inputSelect(AUDIO_INPUT_LINEIN); // RX mode uses line ins
setup_gain();
setup_mode(bands[band].mode);
setup_RX(bands[band].mode, bands[band].bandwidthU, bands[band].bandwidthL); // set up the audio chain for reception
FFT_WINDOW_SET (Window_FFT);
FFT_STRING = FFT_WINDOW_STRING (Window_FFT);
audioShield.lineInLevel((float)bands[band].RFgain/10, (float)bands[band].RFgain/10);
if (!(SD.begin(10))) {
// stop here, but print a message repetitively
while (1) {
Serial.println("Unable to access the SD card");
delay(500);
}
}
si5351.init(SI5351_CRYSTAL_LOAD_10PF, 27000000);
// si5351.drive_strength(SI5351_CLK0, SI5351_CLK_DRIVE_8MA); // does this work ???
si5351.set_correction(calibration_constant);
si5351.set_pll(SI5351_PLL_FIXED, SI5351_PLLA);
setfreq();
delay(3);
sinewave();
} // end void SETUP
const char* filename (int pointer) {
String NAME = "REC"+String(pointer)+".RAW";
char str[15];
NAME.toCharArray(str, sizeof(NAME));
return str;
}
void controls() {
// print time every second
if (ms_100.check() == 1) { displayClock();
AudioProcessorUsageMaxReset();
AudioMemoryUsageMaxReset();}
bouncer1.update();
bouncer2.update();
bouncer3.update();
bouncer4.update();
bouncer5.update();
if ( bouncer1.fallingEdge()) {
Serial.print("Play Button Press");
if (Rec_Mode == 1) stopRecording();
if (Rec_Mode == 0){
showtrack();
Rec_Mode = 2;
startPlaying();
}
}
if ( bouncer2.fallingEdge()) {
Serial.print("Stop Button Press");
if (Rec_Mode == 1) stopRecording();
if (Rec_Mode == 2) stopPlaying();
Rec_Mode = 0;
Switch_to_SDR();
Menu = TUNING; Menu2 = TUNING;
RECORDFLAG = 0;
}
if ( bouncer3.fallingEdge()) {
Serial.print("Record Button Press");
if (Rec_Mode == 2) {
stopPlaying();
Switch_to_SDR();
}
if (Rec_Mode == 0) startRecording();
}
if ( bouncer4.fallingEdge()) {
Serial.print("previous track Button Press ");
prevtrack();
}
if ( bouncer5.fallingEdge()) {
Serial.print("next track Button Press");
nexttrack();
}
// If we're playing or recording, carry on...
if (Rec_Mode == 1) {
continueRecording();
}
if (Rec_Mode == 2) {
continuePlaying();
}
}
void startRecording() {
Serial.print("startRecording");
/* if (SD.exists("RECORD.RAW")) {
// The SD library writes new data to the end of the
// file, so to start a new recording, the old file
// must be deleted before new data is written.
SD.remove("RECORD.RAW");
} */
// recfile = recfile + 1;
String NAME = "REC"+String(playfile)+".RAW";
char fi[15];
NAME.toCharArray(fi, sizeof(NAME));
if (SD.exists(fi)) {
// The SD library writes new data to the end of the
// file, so to start a new recording, the old file
// must be deleted before new data is written.
SD.remove(fi);
}
frec = SD.open(fi, FILE_WRITE);
if (frec) {
recorder.begin();
Rec_Mode = 1;
tft.fillRect(0, 104, 160, 10, ST7735_RED); //ST7735_BLACK);
tft.setCursor(0, 105);
tft.print (" Recording !");
showtrack();
}
}
void continueRecording() {
if (recorder.available() >= 2) {
byte buffer[512];
// Fetch 2 blocks from the audio library and copy
// into a 512 byte buffer. The Arduino SD library
// is most efficient when full 512 byte sector size
// writes are used.
memcpy(buffer, recorder.readBuffer(), 256);
recorder.freeBuffer();
memcpy(buffer+256, recorder.readBuffer(), 256);
recorder.freeBuffer();
// write all 512 bytes to the SD card
// elapsedMicros usec = 0;
frec.write(buffer, 512);
// Uncomment these lines to see how long SD writes
// are taking. A pair of audio blocks arrives every
// 5802 microseconds, so hopefully most of the writes
// take well under 5802 us. Some will take more, as
// the SD library also must write to the FAT tables
// and the SD card controller manages media erase and
// wear leveling. The recorder object can buffer
// approximately 301700 us of audio, to allow time
// for occasional high SD card latency, as long as
// the average write time is under 5802 us.
// if (one_sec.check() == 1) {
// Serial.print("SD write, us=");
// }
// Serial.println(usec);
}
}
void stopRecording() {
Serial.print("stopRecording");
recorder.end();
if (Rec_Mode == 1) {
while (recorder.available() > 0) {
frec.write((byte*)recorder.readBuffer(), 256);
recorder.freeBuffer();
}
frec.close();
// playfile = recfile;
}
Rec_Mode = 0;
clearname();
tft.print (" Recording stopped!");
}
void startPlaying() {
String NAME = "REC"+String(playfile)+".RAW";
char fi[15];
NAME.toCharArray(fi, sizeof(NAME));
Serial.print("startPlaying ");
// const char* fi = filename(playfile);
// const char* fi = "REC13.RAW";
// Serial.print ("File: ");Serial.print(playfile);
Serial.println ("Playfile: "); Serial.print (playfile);
Serial.println ("Name: "); Serial.print (filename(playfile));
delay(100);
Switch_to_MP3();
// playSd.play(filename(playfile));
// playSd.play("REC13.RAW");
playSd.play(fi);
Rec_Mode = 2;
tft.fillRect(0, 104, 160, 10, ST7735_GREEN); //ST7735_BLACK);
tft.setCursor(0, 105);
tft.setTextColor(ST7735_BLACK);
tft.print (" Playing !");
tft.setTextColor(ST7735_WHITE);
showtrack();
}
void continuePlaying() {
if (!playSd.isPlaying()) {
playSd.stop();
Rec_Mode = 0;
clearname();
tft.print("End of recording");
}
}
void stopPlaying() {
Serial.print("stopPlaying");
if (Rec_Mode == 2) playSd.stop();
Rec_Mode = 0;
clearname();
tft.print (" Playing stopped");
}
void prevtrack() {
playfile = playfile - 1;
if (playfile < 0) playfile = 0;
if (playfile > 255) playfile = 255;
showtrack();
if (Rec_Mode == 2) {
stopPlaying(); // old track
startPlaying();// new track = old track -1
}
// if (Rec_Mode == 0) startPlaying();
}
void nexttrack() {
playfile = playfile + 1;
if (playfile < 0) playfile = 0;
if (playfile > 255) playfile = 255;
showtrack();
if (Rec_Mode == 2) {
stopPlaying();
startPlaying();
}
}
void showtrack() {
if (Rec_Mode == 1) {
tft.fillRect(pos_x_menu, pos_y_menu - 14, 60, 9, ST7735_RED); // REC
tft.setTextColor(WHITE);
}
if (Rec_Mode == 2) {
tft.fillRect(pos_x_menu, pos_y_menu - 14, 60, 9, ST7735_GREEN); // PLAY
tft.setTextColor(BLACK);
}
if (Rec_Mode == 0) {
tft.fillRect(pos_x_menu, pos_y_menu - 14, 60, 9, BLACK); // STOP
tft.setTextColor(WHITE);
}
tft.setCursor(pos_x_menu, pos_y_menu - 14);
tft.print("Track: "); tft.print(playfile);
// hier printen den track
Serial.print(playfile);
Serial.print(" ");
// Serial.print(recfile); Serial.print(" ");
}
void Switch_to_MP3() {
// set gain values of SDR chain to 0
audioShield.eqBands (0.7, 0.0); // (float bass, float treble) in % -100 to +100
audio_in_I.gain(0, 0); // take values from EEPROM
audio_in_I.gain(1, 0); // take values from EEPROM
audio_in_I.gain(2,0); //not used
audio_in_I.gain(3,0); //not used
audio_in_Q.gain(0, 0); // always 1, only I gain is calibrated
audio_in_Q.gain(1, 0); // take values from EEPROM
audio_in_Q.gain(2,0); //not used
audio_in_Q.gain(3,0); //not used
/* add_substract_I.gain(0, 0);
add_substract_I.gain(1,0); // substract
add_substract_Q.gain(0,0);
add_substract_Q.gain(0,0); // add
*/
// set audio chain to MP3/M4A
mixleft.gain(1, 0); //AAC on
mixright.gain(1,0);
mixleft.gain(0,0); // MP3 on
mixright.gain(0,0);