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304 lines
12 KiB
304 lines
12 KiB
/*
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* RadioFMDiscriminator_F32
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* 25 April 2022 Bob Larkin
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* With much credit to:
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* Chip Audette (OpenAudio) Feb 2017
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* Building from AudioFilterFIR from Teensy Audio Library
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* (AudioFilterFIR credited to Pete (El Supremo))
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* and of course, to PJRC for the Teensy and Teensy Audio Library
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*
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* Copyright (c) 2022 Bob Larkin
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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/* This consists of a single input at some frequency, such as 10 to 20 kHz and
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* an output, such as 0 to 5 kHz. The output level is linearly dependent on the
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* frequency of the input sine wave frequency, i.e., it is an FM discriminator.
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*
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* NOTE: Due to the sample frequencies we are working with, like 44.1 kHz, this
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* discriminator cannot handle full FM broadcast bandwidths. It is suitable for
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* NBFM as used in communications, marine radio, ham radio, etc.
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*
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* The output can be FIR filtered using default parameters,
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* or using coefficients from an array. A separate single pole de-emphasis filer
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* is included that again can be programmed.
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*
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* Internally, the discriminator uses a pair of single pole BPF that
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*
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* Status:
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*
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*
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* Functions:
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* frequency(float fCenter ) sets the center frequency in Hz, default 15000.
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*
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* filterOut(float *firCoeffs, uint nFIR, float Kdem) sets output filtering where:
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* float32_t* firCoeffs is an array of coefficients
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* uint nFIR is the number of coefficients
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* float32_t Kdem is the de-emphasis frequency factor, where
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* Kdem = 1/(0.5+(tau*fsample)) and tau is the de-emphasis
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* time constant, typically 0.0005 second and fsample is
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* the sample frequency, typically 44117.
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*
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* filterIQ(float *fir_IQ_Coeffs, uint nFIR_IQ) sets output filtering where:
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* float32_t* fir_IQ_Coeffs is an array of coefficients
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* uint nFIR_IQ is the number of coefficients, max 60
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*
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* setSampleRate_Hz(float32_t _sampleRate_Hz) allows dynamic changing of
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* the sample rate (experimental as of May 2020).
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*
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* Time: For T4.0, 45 microseconds for a block of 128 data points.
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*
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*/
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#ifndef _radioFMDiscriminator_f32_h
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#define _radioFMDiscriminator_f32_h
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//#include "mathDSP_F32.h"
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#include "AudioStream_F32.h"
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//#include "arm_math.h"
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#define LPF_NONE 0
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#define LPF_FIR 1
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#define LPF_IIR 2
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#define TEST_TIME_FM 0
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class RadioFMDiscriminator_F32 : public AudioStream_F32 {
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//GUI: inputs:1, outputs:2 //this line used for automatic generation of GUI node
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//GUI: shortName: FMDiscriminator
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public:
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// Default block size and sample rate:
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RadioFMDiscriminator_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {
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}
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// Option of AudioSettings_F32 change to block size and/or sample rate:
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RadioFMDiscriminator_F32(const AudioSettings_F32 &settings) : AudioStream_F32(1, inputQueueArray_f32) {
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sampleRate_Hz = settings.sample_rate_Hz;
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block_size = settings.audio_block_samples;
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}
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// This sets the parameters of the discriminator. The output LPF, if any,
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// must precede this function.
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void initializeFMDiscriminator(float32_t _f1, float32_t _f2, float32_t _q1, float32_t _q2) {
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f1 = _f1; f2 = _f2;
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q1 = _q1; q2 = _q2;
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// Design the 2 single pole filters:
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setBandpass(coeff_f1BPF, f1, q1);
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setBandpass(coeff_f2BPF, f2, q2);
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// Initialize BiQuad instances for BPF's (ARM DSP Math Library)
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// https://www.keil.com/pack/doc/CMSIS/DSP/html/group__BiquadCascadeDF1.html
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// arm_biquad_cascade_df1_init_f32(&biquad_inst, numStagesUsed, &coeff32[0], &StateF32[0])
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arm_biquad_cascade_df1_init_f32(&f1BPF_inst, 1, &coeff_f1BPF[0], &state_f1BPF[0]);
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arm_biquad_cascade_df1_init_f32(&f2BPF_inst, 1, &coeff_f2BPF[0], &state_f2BPF[0]);
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/* The FIR instance setup call
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* void arm_fir_init_f32(
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* arm_fir_instance_f32* S, points to instance of floating-point FIR filter struct
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* uint16_t numTaps, Number of filter coefficients in the filter.
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* float32_t* pCoeffs, points to the filter coefficients buffer.
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* float32_t* pState, points to the state buffer.
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* uint32_t blockSize) Number of samples that are processed per call.
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*/
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if (fir_Out_Coeffs && outputFilterType == LPF_FIR)
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{
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arm_fir_init_f32(&FMDet_Out_inst, nFIR_Out, &fir_Out_Coeffs[0],
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&State_FIR_Out[0], (uint32_t)block_size);
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}
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else
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{
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;
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}
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// Initialize squelch Input BPF BiQuad instance
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arm_biquad_cascade_df1_init_f32(&iirSqIn_inst, 2, pCfSq, &stateSqIn[0]);
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}
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// Provide for changing to user FIR for discriminator output, (and user de-emphasis)
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// This should precede setting discriminator parameters
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void filterOutFIR(float32_t *_fir_Out_Coeffs, int _nFIR_Out, float32_t *_State_FIR_Out, float32_t _Kdem) {
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if(_fir_Out_Coeffs==NULL)
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{
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outputFilterType = LPF_NONE;
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return;
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}
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if( _Kdem<0.0001 || _Kdem>1.0 ) {
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return;
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}
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outputFilterType = LPF_FIR;
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fir_Out_Coeffs = _fir_Out_Coeffs;
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nFIR_Out = _nFIR_Out;
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State_FIR_Out = _State_FIR_Out;
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Kdem = _Kdem;
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OneMinusKdem = 1.0f - Kdem;
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}
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// This should precede setting discriminator parameters, if used
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void filterOutIIR(float32_t _frequency, float32_t _q, float32_t _Kdem) {
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if( _frequency < 0.0001f)
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{
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outputFilterType = LPF_NONE;
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return;
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}
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outputFilterType = LPF_IIR;
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setLowpass(coeff_outLPF, _frequency, _q);
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arm_biquad_cascade_df1_init_f32(&outLPF_inst, 1, &coeff_outLPF[0], &state_outLPF[0]);
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if( _Kdem<0.0001 || _Kdem>1.0 ) {
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return;
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}
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Kdem = _Kdem;
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OneMinusKdem = 1.0f - Kdem;
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}
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// Provide for changing to user supplied BiQuad for Squelch input.
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// This should precede setting discriminator parameters, if used
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void setSquelchFilter(float* _sqCoeffs) {
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if( _sqCoeffs==NULL)
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pCfSq = coeffSqIn; // Default filter
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else
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pCfSq = _sqCoeffs;
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}
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// The squelch level reads nominally 0.0 to 1.0 where
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float getSquelchLevel (void) {
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return squelchLevel;
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}
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// The squelch threshold is nominally 0.7 where
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// 0.0 always lets audio through.
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void setSquelchThreshold (float _sqTh) {
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squelchThreshold = _sqTh;
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}
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void setSquelchDecay (float _sqDcy) {
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gamma = _sqDcy;
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alpha = 0.5f*(1.0f - gamma);
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}
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// This should precede setting discriminator parameters, if used
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void setSampleRate_Hz(float32_t _sampleRate_Hz) {
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sampleRate_Hz = _sampleRate_Hz;
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}
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virtual void update(void);
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private:
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// One input data pointer
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audio_block_f32_t *inputQueueArray_f32[1];
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float32_t sampleRate_Hz = AUDIO_SAMPLE_RATE_EXACT;
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uint16_t block_size = AUDIO_BLOCK_SAMPLES;
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/* A pair of single pole BPF for the discriminator:
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* Info - The structure from arm_biquad_casd_df1_inst_f32 consists of
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* uint32_t numStages;
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* const float32_t *pCoeffs; //Points to the array of coefficients, length 5*numStages.
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* float32_t *pState; //Points to the array of state variables, length 4*numStages.
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*/
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float f1, q1, f2, q2;
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arm_biquad_casd_df1_inst_f32 f1BPF_inst;
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float coeff_f1BPF[5];
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float state_f1BPF[4];
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arm_biquad_casd_df1_inst_f32 f2BPF_inst;
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float coeff_f2BPF[5];
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float state_f2BPF[4];
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// De-emphasis constant
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float32_t Kdem = 0.045334f;
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float32_t OneMinusKdem = 0.954666f;
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// Save last data point for next update of de-emphasis filter
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float32_t dLast = -1.0f;
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// The output FIR LPF (optional)
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int outputFilterType = LPF_NONE;
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// ARM CMSIS FIR filter instances and State vectors
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arm_fir_instance_f32 FMDet_Out_inst;
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float32_t *State_FIR_Out; // 128+nFIR_Out
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uint16_t nFIR_Out;
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float32_t* fir_Out_Coeffs = NULL;
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float32_t discrOut = 0.0f;
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// Output IIR Biquad alternative
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arm_biquad_casd_df1_inst_f32 outLPF_inst;
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float coeff_outLPF[5];
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float state_outLPF[4];
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arm_biquad_casd_df1_inst_f32 iirSqIn_inst;
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// Default 2 stage Squelch input BiQuad filter, 3000 Hz, 4000 Hz both Q=5
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// The -6 dB points are 2680 and 4420 Hz
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// The -20 dB points are 2300 and 5300 Hz
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float coeffSqIn[10] = {
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0.0398031529f, 0.0f, -0.0398031529f, 1.74762569f, -0.92039369f,
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0.0511929547f, 0.0f, -0.0511929547f, 1.59770204f, -0.89761409f};
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float* pCfSq = coeffSqIn;
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float stateSqIn[8];
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float squelchThreshold = 0.7f;
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float squelchLevel = 1.0f;
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float gamma = 0.99;
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float alpha = 0.5f*(1.0f - gamma);
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#if TEST_TIME_FM
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elapsedMicros tElapse;
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int32_t iitt = 999000; // count up to a million during startup
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#endif
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#if 0
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/* Info Only, an example FIR filter, include this in INO to use.
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* FIR filter designed with http://t-filter.appspot.com
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* fs = 44100 Hz, < 3kHz ripple 0.36 dB, >6 kHz, -60 dB, 39 taps
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* Corrected to give DC gain = 1.00
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*/
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float32_t fir_Out39[39] = {
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-0.0008908477f, -0.0008401274f, -0.0001837353f, 0.0017556005f,
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0.0049353322f, 0.0084952916f, 0.0107668722f, 0.0097441685f,
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0.0039877576f, -0.0063455016f, -0.0188069300f, -0.0287453055f,
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-0.0303831521f, -0.0186809770f, 0.0085931270f, 0.0493875744f,
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0.0971742012f, 0.1423015880f, 0.1745838382f, 0.1863024485f,
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0.1745838382f, 0.1423015880f, 0.0971742012f, 0.0493875744f,
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0.0085931270f, -0.0186809770f, -0.0303831521f, -0.0287453055f,
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-0.0188069300f, -0.0063455016f, 0.0039877576f, 0.0097441685f,
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0.0107668722f, 0.0084952916f, 0.0049353322f, 0.0017556005f,
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-0.0001837353f, -0.0008401274f, -0.0008908477f };
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#endif
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// Unity gain BPF Biquad, CMSIS format (not Matlab)
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void setBandpass(float32_t* pCoeff, float32_t frequency, float32_t q) {
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float32_t w0 = 2.0f*3.141592654f*frequency/sampleRate_Hz;
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float32_t alpha = sin(w0)/(2.0f*q);
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float32_t scale = 1.0f/(1.0f + alpha);
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/* b0 */ *(pCoeff+0) = alpha*scale;
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/* b1 */ *(pCoeff+1) = 0.0f;
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/* b2 */ *(pCoeff+2) = (-alpha)*scale;
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/* a1 */ *(pCoeff+3) = -(-2.0f*cos(w0))*scale;
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/* a2 */ *(pCoeff+4) = -(1.0f - alpha)*scale;
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}
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// Unity gain LPF, CMSIS format
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void setLowpass(float32_t* pCoeff, float32_t frequency, float32_t q) {
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float32_t w0 = frequency*(2.0f*3.141592654f / sampleRate_Hz);
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float32_t alpha = sin(w0) / ((double)q*2.0f);
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float32_t cosW0 = cos(w0);
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float32_t scale = 1.0f/(1.0f+alpha); // which is equal to 1.0f / a0
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/* b0 */ *(pCoeff+0) = ((1.0f - cosW0) / 2.0f)*scale;
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/* b1 */ *(pCoeff+1) = (1.0f - cosW0)*scale;
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/* b2 */ *(pCoeff+2) = *(pCoeff+0);
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/* a1 */ *(pCoeff+3) = -(-2.0f*cosW0)*scale;
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/* a2 */ *(pCoeff+4) = -(1.0f - alpha)*scale;
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}
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};
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#endif
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