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OpenAudio_ArduinoLibrary/RadioFMDiscriminator_F32.h

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/*
* RadioFMDiscriminator_F32
* 25 April 2022 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) 2022 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., it is an FM discriminator.
*
* NOTE: Due to the sample frequencies we are working with, like 44.1 kHz, this
* discriminator 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 discriminator uses a pair of single pole BPF that
*
* Status:
*
*
* 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).
*
* Time: For T4.0, 45 microseconds for a block of 128 data points.
*
*/
#ifndef _radioFMDiscriminator_f32_h
#define _radioFMDiscriminator_f32_h
//#include "mathDSP_F32.h"
#include "AudioStream_F32.h"
//#include "arm_math.h"
#define LPF_NONE 0
#define LPF_FIR 1
#define LPF_IIR 2
#define TEST_TIME_FM 0
class RadioFMDiscriminator_F32 : public AudioStream_F32 {
//GUI: inputs:1, outputs:2 //this line used for automatic generation of GUI node
//GUI: shortName: FMDiscriminator
public:
// Default block size and sample rate:
RadioFMDiscriminator_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {
}
// Option of AudioSettings_F32 change to block size and/or sample rate:
RadioFMDiscriminator_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) {
sampleRate_Hz = settings.sample_rate_Hz;
block_size = settings.audio_block_samples;
}
// This sets the parameters of the discriminator. The output LPF, if any,
// must precede this function.
void initializeFMDiscriminator(float32_t _f1, float32_t _f2, float32_t _q1, float32_t _q2) {
f1 = _f1; f2 = _f2;
q1 = _q1; q2 = _q2;
// Design the 2 single pole filters:
setBandpass(coeff_f1BPF, f1, q1);
setBandpass(coeff_f2BPF, f2, q2);
// Initialize BiQuad instances for BPF's (ARM DSP Math Library)
// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html
// arm_biquad_cascade_df1_init_f32(&biquad_inst, numStagesUsed, &coeff32[0], &StateF32[0])
arm_biquad_cascade_df1_init_f32(&f1BPF_inst, 1, &coeff_f1BPF[0], &state_f1BPF[0]);
arm_biquad_cascade_df1_init_f32(&f2BPF_inst, 1, &coeff_f2BPF[0], &state_f2BPF[0]);
/* The FIR instance setup call
* void arm_fir_init_f32(
* arm_fir_instance_f32* S, points to instance of floating-point FIR filter struct
* 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.
*/
if (fir_Out_Coeffs && outputFilterType == LPF_FIR)
{
arm_fir_init_f32(&FMDet_Out_inst, nFIR_Out, &fir_Out_Coeffs[0],
&State_FIR_Out[0], (uint32_t)block_size);
}
else
{
;
}
// Initialize squelch Input BPF BiQuad instance
arm_biquad_cascade_df1_init_f32(&iirSqIn_inst, 2, pCfSq, &stateSqIn[0]);
}
// Provide for changing to user FIR for discriminator output, (and user de-emphasis)
// This should precede setting discriminator parameters
void filterOutFIR(float32_t *_fir_Out_Coeffs, int _nFIR_Out, float32_t *_State_FIR_Out, float32_t _Kdem) {
if(_fir_Out_Coeffs==NULL)
{
outputFilterType = LPF_NONE;
return;
}
if( _Kdem<0.0001 || _Kdem>1.0 ) {
return;
}
outputFilterType = LPF_FIR;
fir_Out_Coeffs = _fir_Out_Coeffs;
nFIR_Out = _nFIR_Out;
State_FIR_Out = _State_FIR_Out;
Kdem = _Kdem;
OneMinusKdem = 1.0f - Kdem;
}
// This should precede setting discriminator parameters, if used
void filterOutIIR(float32_t _frequency, float32_t _q, float32_t _Kdem) {
if( _frequency < 0.0001f)
{
outputFilterType = LPF_NONE;
return;
}
outputFilterType = LPF_IIR;
setLowpass(coeff_outLPF, _frequency, _q);
arm_biquad_cascade_df1_init_f32(&outLPF_inst, 1, &coeff_outLPF[0], &state_outLPF[0]);
if( _Kdem<0.0001 || _Kdem>1.0 ) {
return;
}
Kdem = _Kdem;
OneMinusKdem = 1.0f - Kdem;
}
// Provide for changing to user supplied BiQuad for Squelch input.
// This should precede setting discriminator parameters, if used
void setSquelchFilter(float* _sqCoeffs) {
if( _sqCoeffs==NULL)
pCfSq = coeffSqIn; // Default filter
else
pCfSq = _sqCoeffs;
}
// 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);
}
// This should precede setting discriminator parameters, if used
void setSampleRate_Hz(float32_t _sampleRate_Hz) {
sampleRate_Hz = _sampleRate_Hz;
}
virtual void update(void);
private:
// One input data pointer
audio_block_f32_t *inputQueueArray_f32[1];
float32_t sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT;
uint16_t block_size = AUDIO_BLOCK_SAMPLES;
/* A pair of single pole BPF for the discriminator:
* 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.
*/
float f1, q1, f2, q2;
arm_biquad_casd_df1_inst_f32 f1BPF_inst;
float coeff_f1BPF[5];
float state_f1BPF[4];
arm_biquad_casd_df1_inst_f32 f2BPF_inst;
float coeff_f2BPF[5];
float state_f2BPF[4];
// De-emphasis constant
float32_t Kdem = 0.045334f;
float32_t OneMinusKdem = 0.954666f;
// Save last data point for next update of de-emphasis filter
float32_t dLast = -1.0f;
// The output FIR LPF (optional)
int outputFilterType = LPF_NONE;
// ARM CMSIS FIR filter instances and State vectors
arm_fir_instance_f32 FMDet_Out_inst;
float32_t *State_FIR_Out; // 128+nFIR_Out
uint16_t nFIR_Out;
float32_t* fir_Out_Coeffs = NULL;
float32_t discrOut = 0.0f;
// Output IIR Biquad alternative
arm_biquad_casd_df1_inst_f32 outLPF_inst;
float coeff_outLPF[5];
float state_outLPF[4];
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
#if 0
/* Info Only, an example FIR filter, include this in INO to use.
* 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
// Unity gain BPF Biquad, CMSIS format (not Matlab)
void setBandpass(float32_t* pCoeff, float32_t frequency, float32_t q) {
float32_t w0 = 2.0f*3.141592654f*frequency/sampleRate_Hz;
float32_t alpha = sin(w0)/(2.0f*q);
float32_t scale = 1.0f/(1.0f + alpha);
/* b0 */ *(pCoeff+0) = alpha*scale;
/* b1 */ *(pCoeff+1) = 0.0f;
/* b2 */ *(pCoeff+2) = (-alpha)*scale;
/* a1 */ *(pCoeff+3) = -(-2.0f*cos(w0))*scale;
/* a2 */ *(pCoeff+4) = -(1.0f - alpha)*scale;
}
// Unity gain LPF, CMSIS format
void setLowpass(float32_t* pCoeff, float32_t frequency, float32_t q) {
float32_t w0 = frequency*(2.0f*3.141592654f / sampleRate_Hz);
float32_t alpha = sin(w0) / ((double)q*2.0f);
float32_t cosW0 = cos(w0);
float32_t scale = 1.0f/(1.0f+alpha); // which is equal to 1.0f / a0
/* b0 */ *(pCoeff+0) = ((1.0f - cosW0) / 2.0f)*scale;
/* b1 */ *(pCoeff+1) = (1.0f - cosW0)*scale;
/* b2 */ *(pCoeff+2) = *(pCoeff+0);
/* a1 */ *(pCoeff+3) = -(-2.0f*cosW0)*scale;
/* a2 */ *(pCoeff+4) = -(1.0f - alpha)*scale;
}
};
#endif