/*
* RadioFMDetector_F32.cpp
*
* 25 April 2022
* Bob Larkin, in support of the library:
* Chip Audette, OpenAudio, Apr 2017
* -------------------
*
* 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.
*
* See RadioFMDetector_F32.h for usage details
*/
#include "RadioFMDetector_F32.h"
// 513 values of the sine wave in a float array:
#include "sinTable512_f32.h"
// ==== UPDATE ====
void RadioFMDetector_F32::update(void) {
audio_block_f32_t *blockIn, *blockOut=NULL, *blockZero;
static float saveIn = 0.0f; // for squelch
static float saveOut = 0.0f;
uint16_t i, index_sine;
float32_t deltaPhase, a, b, dtemp1, dtemp2;
float32_t v_i[128]; // max size
float32_t v_q[128];
mathDSP_F32 mathDSP1; // Math support functions
#if TEST_TIME_FM
if (iitt++ >1000000) iitt = -10;
uint32_t t1, t2;
t1 = tElapse;
#endif
// Get input to FM Detector block
blockIn = AudioStream_F32::receiveWritable_f32(0);
if (!blockIn) {
return;
}
if (fir_Out_Coeffs == NULL) {
//if(errorPrintFM) Serial.println("FMDET-ERR: No Out FIR Coefficients");
AudioStream_F32::release(blockIn);
return;
}
// Try to get a block for the FM output
blockOut = AudioStream_F32::allocate_f32();
if (!blockOut){ // Didn't have any
//if(errorPrintFM) Serial.println("FMDET-ERR: No Output Memory");
AudioStream_F32::release(blockIn);
return;
}
blockZero = AudioStream_F32::allocate_f32();
for(int kk=0; kk<128; kk++) blockZero->data[kk] = 0.0f;
// Generate sine and cosine of center frequency and double-balance mix
// these with the input signal to produce an intermediate result
// saved as v_i[] and v_q[]
for (i=0; i < block_size; i++) {
phaseS += phaseIncrement;
if (phaseS > 512.0f)
phaseS -= 512.0f;
index_sine = (uint16_t) phaseS;
deltaPhase = phaseS -(float32_t) index_sine;
/* Read two nearest values of input value from the sin table */
a = sinTable512_f32[index_sine];
b = sinTable512_f32[index_sine+1];
// Linear interpolation and multiplying (DBMixer) with input
v_i[i] = blockIn->data[i] * (a + 0.001953125*(b-a)*deltaPhase);
/* Repeat for cosine by adding 90 degrees phase */
index_sine = (index_sine + 128) & 0x01ff;
/* Read two nearest values of input value from the sin table */
a = sinTable512_f32[index_sine];
b = sinTable512_f32[index_sine+1];
/* deltaPhase will be the same as used for sin */
v_q[i] = blockIn->data[i] * (a + 0.001953125*(b-a)*deltaPhase);
}
// Do I FIR and Q FIR. We can borrow blockIn and blockOut at this point
//void arm_fir_f32( const arm_fir_instance_f32* S, float32_t* pSrc, float32_t* pDst, uint32_t blockSize)
arm_fir_f32(&FMDet_I_inst, v_i, blockIn->data, (uint32_t)blockIn->length);
arm_fir_f32(&FMDet_Q_inst, v_q, blockOut->data, (uint32_t)blockOut->length);
// Do ATAN2, differentiation and de-emphasis in single loop
for(i=0; i<block_size; i++) { // y x
dtemp1 = mathDSP1.fastAtan2((float)blockOut->data[i], (float)blockIn->data[i]);
// Apply differentiator by subtracting last value of atan2
if(dtemp1>MF_PI_2 && diffLast<-MF_PI_2) // Probably a wrap around
dtemp2 = dtemp1 - diffLast - MF_TWOPI;
else if(dtemp1<-MF_PI_2 && diffLast >MF_PI_2) // Probably a reverse wrap around
dtemp2 = dtemp1 - diffLast + MF_TWOPI;
else
dtemp2 = dtemp1 - diffLast; // Differentiate
diffLast = dtemp1; // Ready for next time through loop
// Data point is now dtemp2. Apply single pole de-emphasis LPF, in place
dLast = Kdem * dtemp2 + OneMinusKdem * dLast;
blockIn->data[i] = dLast; // and save to an array
}
// Do output FIR filter. Data now in blockIn.
arm_fir_f32(&FMDet_Out_inst, blockIn->data, blockOut->data, (uint32_t)blockIn->length);
// Squelch picks the audio from before the output filter and does a 4-pole BiQuad BPF
// blockIn->data still has the data we need
arm_biquad_cascade_df1_f32(&iirSqIn_inst, blockIn->data,
blockIn->data, blockIn->length);
// Update the Squelch full-wave envelope detector and single pole LPF
float sumNoise = 0.0f;
for(i=0; i<block_size; i++)
sumNoise += fabsf(blockIn->data[i]); // Ave of rectified noise
squelchLevel = alpha*(sumNoise + saveIn) + gamma*saveOut; // 1 pole
saveIn = sumNoise;
saveOut = squelchLevel;
// Add hysteresis here <<<<
// Squelch gate sends audio to output 1 (right) if squelch is open
if(squelchLevel < squelchThreshold)
AudioStream_F32::transmit(blockOut, 1);
else
AudioStream_F32::transmit(blockZero, 1);
// The un-squelch controlled audio for tone decoding, etc., goes to output 0
AudioStream_F32::transmit(blockOut, 0);
AudioStream_F32::release(blockIn); // Give back the blocks
AudioStream_F32::release(blockOut); AudioStream_F32::release(blockZero);
#if TEST_TIME_FM
t2 = tElapse;
if(iitt++ < 0) {Serial.print("At end of FM Det "); Serial.println (t2 - t1); }
t1 = tElapse;
#endif
}