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OpenAudio_ArduinoLibrary/RadioFMDetector_F32.h

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/*
* RadioFMDetector_F32
* 22 March 2020 Bob Larkin
* With much credit to:
* Chip Audette (OpenAudio) Feb 2017
* Building from AudioFilterFIR from Teensy Audio Library
* (AudioFilterFIR credited to Pete (El Supremo))
* and of course, to PJRC for the Teensy and Teensy Audio Library
*
* Copyright (c) 2020 Bob Larkin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
/* This consists of a single input at some frequency, such as 10 to 20 kHz and
* an output, such as 0 to 5 kHz. The output level is linearly dependent on the
* frequency of the input sine wave frequency, i.e., an it is an FM detector.
* The input needs to be band limited below the lower frequency side of the
* input, typically 10 kHz. This is not part of this block.
*
* NOTE: Due to the sample frequencies we are working with, like 44.1 kHz, this
* detector cannot handle full FM broadcast bandwidths. It is suitable for
* NBFM as used in communications, marine radio, ham radio, etc.
*
* The output can be FIR filtered using default parameters,
* or using coefficients from an array. A separate single pole de-emphasis filer
* is included that again can be programmed.
*
* Internally, the detector uses a pair of mixers (multipliers) that generate the
* in-phase and quadrature inputs to a atan2 type of phase detector. These
* mixers have two output signals at the difference (desired) and sum (undesired)
* frequencies. The high frequency sum signal can be filtered (For a 15 kHz center,
* with an input band of 10 to 20 kHz the sum signal will be from 25 to 35 kHz that
* wraps around the 22 kHz half-sample point to produce 19 to 9 kHz. This needs to
* be removed before the atan2. A pair of FIR filters, using FIR_IQ_Coeffs
* are used. These are again programmable and default to a 29-tap LPF with
* a 5 kHz cutoff.
*
* Status: Tested static, tested with FM modulated Fluke 6061B.
* An input of about 60 microvolts to the SGTL5000 gave 12 dB SINAD.
* The output sounded good. Tested T3.6 and T4.0. No known bugs
*
* Output: Float, sensitivity is 2*pi*(f - fCenter)*sample_rate_Hz
* For 44117Hz samplerate, this is 0.000142421 per Hz
*
* Accuracy: The function used is precise. However, the approximations, such
* fastAtan2, slightly limit the accuracy. A 200 point sample of a
* 14 kHz input had an average error of 0.03 Hz
* and a standard deviation of 0.81 Hz.
* The largest errors in this sample were about +/- 1.7 Hz. This is
* with the default filters.
*
* Functions:
* frequency(float fCenter ) sets the center frequency in Hz, default 15000.
*
* filterOut(float *firCoeffs, uint nFIR, float Kdem) sets output filtering where:
* float32_t* firCoeffs is an array of coefficients
* uint nFIR is the number of coefficients
* float32_t Kdem is the de-emphasis frequency factor, where
* Kdem = 1/(0.5+(tau*fsample)) and tau is the de-emphasis
* time constant, typically 0.0005 second and fsample is
* the sample frequency, typically 44117.
*
* filterIQ(float *fir_IQ_Coeffs, uint nFIR_IQ) sets output filtering where:
* float32_t* fir_IQ_Coeffs is an array of coefficients
* uint nFIR_IQ is the number of coefficients, max 60
*
* setSampleRate_Hz(float32_t _sampleRate_Hz) allows dynamic changing of
* the sample rate (experimental as of May 2020).
*
* returnInitializeFMError() Returns the initialization errors.
* B0001 (value 1) is an error in the IQ FIR Coefficients or quantity.
* B0010 (value 2) is an error in the Output FIR Coefficients or quantity.
* B0100 (value 4) is an error in the de-emphasis constant
* B1000 (value 8) is center frequency above half-sample frequency.
* All for debug.
*
* showError(uint16_t e) Turns error printing in the update function on (e=1)
* or off (e=0). For debug only.
*
* Time: For T3.6, an update of a 128 sample block, 430 microseconds, or
* 3.4 microseconds per data point.
* For T4.0, 97 microseconds, or 0.76 microseconds per data point.
*
* Error checking: See functions setSampleRate_Hz() and returnInitializeFMError()
* above.
*/
#ifndef _radioFMDetector_f32_h
#define _radioFMDetector_f32_h
#include "mathDSP_F32.h"
#include "AudioStream_F32.h"
#include "arm_math.h"
#define MAX_FIR_IQ_COEFFS 100
#define MAX_FIR_OUT_COEFFS 120
#define TEST_TIME_FM 0
class RadioFMDetector_F32 : public AudioStream_F32 {
//GUI: inputs:1, outputs:2 //this line used for automatic generation of GUI node
//GUI: shortName: FMDetector
public:
// Default block size and sample rate:
RadioFMDetector_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {
initializeFM();
}
// Option of AudioSettings_F32 change to block size and/or sample rate:
RadioFMDetector_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) {
sampleRate_Hz = settings.sample_rate_Hz;
block_size = settings.audio_block_samples;
initializeFM();
}
// Provide for changing input center frequency, in Hz
void frequency(float32_t _fCenter) {
fCenter = _fCenter;
phaseIncrement = 512.0f * fCenter / sampleRate_Hz;
}
// Provide for user FIR for I and Q signals to user supplied array
void filterIQ(float32_t* _fir_IQ_Coeffs, int _nFIR_IQ) {
if( fir_IQ_Coeffs==NULL || nFIR_IQ<4 || nFIR_IQ>MAX_FIR_IQ_COEFFS ) {
initializeFMErrors |= 1;
return;
}
fir_IQ_Coeffs = _fir_IQ_Coeffs;
nFIR_IQ = _nFIR_IQ;
initializeFM();
}
// Provide for changing to user FIR for detector output, (and user de-emphasis)
void filterOut(float32_t *_fir_Out_Coeffs, int _nFIR_Out, float32_t _Kdem) {
if( _fir_Out_Coeffs==NULL || _nFIR_Out<4 || _nFIR_Out>MAX_FIR_OUT_COEFFS) {
initializeFMErrors |= 2;
return;
}
if( _Kdem<0.0001 || _Kdem>1.0 ) {
initializeFMErrors |= 4;
return;
}
fir_Out_Coeffs = _fir_Out_Coeffs;
nFIR_Out = _nFIR_Out;
Kdem = _Kdem;
OneMinusKdem = 1.0f - Kdem;
initializeFM();
}
// Provide for changing to user supplied BiQuad for Squelch input.
void setSquelchFilter(float* _sqCoeffs) {
if( _sqCoeffs==NULL)
pCfSq = coeffSqIn; // Default filter
else
pCfSq = _sqCoeffs;
initializeFM();
}
// The squelch level reads nominally 0.0 to 1.0 where
float getSquelchLevel (void) {
return squelchLevel;
}
// The squelch threshold is nominally 0.7 where
// 0.0 always lets audio through.
void setSquelchThreshold (float _sqTh) {
squelchThreshold = _sqTh;
}
void setSquelchDecay (float _sqDcy) {
gamma = _sqDcy;
alpha = 0.5f*(1.0f - gamma);
}
/* void setSampleRate_Hz(float32_t _sampleRate_Hz) {
if (fCenter > _sampleRate_Hz/2.0f) { // Check freq range
initializeFMErrors |= 8;
return;
}
sampleRate_Hz = _sampleRate_Hz;
// update phase increment for new frequency
phaseIncrement = 512.0f * fCenter / sampleRate_Hz;
} // NOT SUPPORTED - No Dynamic changes
*/
void showError(uint16_t e) {
errorPrintFM = e;
}
uint16_t returnInitializeFMError(void) {
return initializeFMErrors;
}
void update(void);
private:
// One input data pointer
audio_block_f32_t *inputQueueArray_f32[1];
float32_t fCenter = 15000.0f;
float32_t phaseS = 0.0f;
float32_t phaseS_C = 128.00f;
float32_t phaseIncrement = 512.0f*15000.0f/AUDIO_SAMPLE_RATE_EXACT;
float32_t sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT;
uint16_t block_size = AUDIO_BLOCK_SAMPLES;
// De-emphasis constant
float32_t Kdem = 0.045334f;
float32_t OneMinusKdem = 0.954666f;
// Save last data point of atan2 for differentiator
float32_t diffLast = 0.0f;
// Save last data point for next update of de-emphasis filter
float32_t dLast = 0.0f;
// Control error printing in update(), normally off
uint16_t errorPrintFM = 0;
// Monitor constructor errors
uint16_t initializeFMErrors = 0;
uint16_t nFIR_IQ = 29;
float32_t* fir_IQ_Coeffs = fir_IQ29;
uint16_t nFIR_Out = 39;
float32_t* fir_Out_Coeffs = fir_Out39;
/* Info - The structure from arm_biquad_casd_df1_inst_f32 consists of
* uint32_t numStages;
* const float32_t *pCoeffs; //Points to the array of coefficients, length 5*numStages.
* float32_t *pState; //Points to the array of state variables, length 4*numStages. */
arm_biquad_casd_df1_inst_f32 iirSqIn_inst;
// Default 2 stage Squelch input BiQuad filter, 3000 Hz, 4000 Hz both Q=5
// The -6 dB points are 2680 and 4420 Hz
// The -20 dB points are 2300 and 5300 Hz
float coeffSqIn[10] = {
0.0398031529f, 0.0f, -0.0398031529f, 1.74762569f, -0.92039369f,
0.0511929547f, 0.0f, -0.0511929547f, 1.59770204f, -0.89761409f};
float* pCfSq = coeffSqIn;
float stateSqIn[8];
float squelchThreshold = 0.7f;
float squelchLevel = 1.0f;
float gamma = 0.99;
float alpha = 0.5f*(1.0f - gamma);
#if TEST_TIME_FM
elapsedMicros tElapse;
int32_t iitt = 999000; // count up to a million during startup
#endif
// ARM CMSIS FIR filter instances and State vectors, sized for max, max
arm_fir_instance_f32 FMDet_I_inst;
float32_t State_I_F32[AUDIO_BLOCK_SAMPLES + MAX_FIR_IQ_COEFFS]; // 228
arm_fir_instance_f32 FMDet_Q_inst;
float32_t State_Q_F32[AUDIO_BLOCK_SAMPLES + MAX_FIR_IQ_COEFFS]; // 248
arm_fir_instance_f32 FMDet_Out_inst;
float32_t State_Out_F32[AUDIO_BLOCK_SAMPLES + MAX_FIR_OUT_COEFFS];
// Initialize the FM Detector, part of setting up and changing parameters
void initializeFM(void) {
if (fir_IQ_Coeffs && nFIR_IQ <= MAX_FIR_IQ_COEFFS) {
/* the instance setup call
* void arm_fir_init_f32(
* arm_fir_instance_f32* S, points to instance of floating-point FIR filter structure.
* uint16_t numTaps, Number of filter coefficients in the filter.
* float32_t* pCoeffs, points to the filter coefficients buffer.
* float32_t* pState, points to the state buffer.
* uint32_t blockSize) Number of samples that are processed per call.
*/
arm_fir_init_f32(&FMDet_I_inst, nFIR_IQ, (float32_t*)fir_IQ_Coeffs, &State_I_F32[0], (uint32_t)block_size);
arm_fir_init_f32(&FMDet_Q_inst, nFIR_IQ, (float32_t*)fir_IQ_Coeffs, &State_Q_F32[0], (uint32_t)block_size);
}
else initializeFMErrors |= B0001;
if (fir_Out_Coeffs && nFIR_Out <= MAX_FIR_OUT_COEFFS) {
arm_fir_init_f32(&FMDet_Out_inst, nFIR_Out, (float32_t*)fir_Out_Coeffs, &State_Out_F32[0], (uint32_t)block_size);
}
else initializeFMErrors |= B0010;
dLast = 0.0;
// Initialize squelch Input BPF BiQuad instance (ARM DSP Math Library)
// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html
arm_biquad_cascade_df1_init_f32(&iirSqIn_inst, 2, pCfSq, &stateSqIn[0]);
}
/* FIR filter designed with http://t-filter.appspot.com
* fs = 44100 Hz, < 5kHz ripple 0.29 dB, >9 kHz, -62 dB, 29 taps
*/
float32_t fir_IQ29[29] = {
-0.000970689f, -0.004690292f, -0.008256345f, -0.007565650f,
0.001524420f, 0.015435011f, 0.021920240f, 0.008211937f,
-0.024286413f, -0.052184700f, -0.040532507f, 0.031248107f,
0.146902412f, 0.255179564f, 0.299445269f, 0.255179564f,
0.146902412f, 0.031248107f, -0.040532507f, -0.052184700f,
-0.024286413f, 0.008211937f, 0.021920240f, 0.015435011f,
0.001524420f, -0.007565650f, -0.008256345f, -0.004690292f,
-0.000970689f};
/* FIR filter designed with http://t-filter.appspot.com
* fs = 44100 Hz, < 3kHz ripple 0.36 dB, >6 kHz, -60 dB, 39 taps
* Corrected to give DC gain = 1.00
*/
float32_t fir_Out39[39] = {
-0.0008908477f, -0.0008401274f, -0.0001837353f, 0.0017556005f,
0.0049353322f, 0.0084952916f, 0.0107668722f, 0.0097441685f,
0.0039877576f, -0.0063455016f, -0.0188069300f, -0.0287453055f,
-0.0303831521f, -0.0186809770f, 0.0085931270f, 0.0493875744f,
0.0971742012f, 0.1423015880f, 0.1745838382f, 0.1863024485f,
0.1745838382f, 0.1423015880f, 0.0971742012f, 0.0493875744f,
0.0085931270f, -0.0186809770f, -0.0303831521f, -0.0287453055f,
-0.0188069300f, -0.0063455016f, 0.0039877576f, 0.0097441685f,
0.0107668722f, 0.0084952916f, 0.0049353322f, 0.0017556005f,
-0.0001837353f, -0.0008401274f, -0.0008908477f };
};
#endif