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OpenAudio_ArduinoLibrary/AudioFilter90Deg_F32.h

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6.3 KiB

/*
* AudioFilter90Deg_F32.h
* 22 March 2020 Bob Larkin
* Parts are based on Open Audio FIR filter by Chip Audette:
*
* Chip Audette (OpenAudio) Feb 2017
* - Building from AudioFilterFIR from Teensy Audio Library
* (AudioFilterFIR credited to Pete (El Supremo))
* 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, development funding 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 two uncoupled paths that almost have the same amplitude gain
* but differ in phase by exactly 90 degrees. See AudioFilter90Deg_F32.cpp
* The number of coefficients is an odd number for the FIR Hilbert transform
* as that produces an easily achievable integer sample period delay. In
* float, the ARM FIR library routine will handle odd numbers.\, so no zero padding
* is needed.
*
* No default Hilbert Transform is provided, as it is highly application dependent.
* The number of coefficients is an odd number with a maximum of 250. The Iowa
* Hills program can design a Hilbert Transform filter. Use begin(*pCoeff, nCoeff)
* in the .INO to initialize this block.
*
* Status: Tested T3.6 and T4.0. No known bugs.
* Functions:
* begin(*pCoeff, nCoeff); Initializes this block, with:
* pCoeff = pointer to array of F32 Hilbert Transform coefficients
* nCoeff = uint16_t number of Hilbert transform coefficients
* showError(e); Turns error printing in update() on (e=1) and off (e=0). For debug.
* Examples:
* ReceiverPart1.ino
* ReceiverPart2.ino
* Time: Depends on size of Hilbert FIR. Time for main body of update() including
* Hilbert FIR and compensating delay, 128 data block, running on Teensy 3.6 is:
* 19 tap Hilbert (including 0's) 74 microseconds
* 121 tap Hilbert (including 0's) 324 microseconds
* 251 tap Hilbert (including 0's) 646 microseconds
* Same 121 tap Hilbert on T4.0 is 57 microseconds per update()
* Same 251 tap Hilbert on T4.0 is 114 microseconds per update()
*/
#ifndef _filter_90deg_f32_h
#define _filter_90deg_f32_h
#include "AudioStream_F32.h"
#include "arm_math.h"
#define TEST_TIME_90D 1
// Following supports a maximum FIR Hilbert Transform of 251
#define HILBERT_MAX_COEFFS 251
class AudioFilter90Deg_F32 : public AudioStream_F32 {
//GUI: inputs:2, outputs:2 //this line used for automatic generation of GUI node
//GUI: shortName: 90DegPhase
public:
// Option of AudioSettings_F32 change to block size (no sample rate dependent variables here):
AudioFilter90Deg_F32(void) : AudioStream_F32(2, inputQueueArray_f32) {
block_size = AUDIO_BLOCK_SAMPLES;
}
AudioFilter90Deg_F32(const AudioSettings_F32 &settings) : AudioStream_F32(2, inputQueueArray_f32) {
block_size = settings.audio_block_samples;
}
// Initialize the 90Deg by giving it the filter coefficients and number of coefficients
// Then the delay line for the q (Right) channel is initialized
void begin(const float32_t *cp, const int _n_coeffs) {
coeff_p = cp;
n_coeffs = _n_coeffs;
// Initialize FIR instance (ARM DSP Math Library) (for f32 the return is always void)
if (coeff_p!=NULL && n_coeffs<252) {
arm_fir_init_f32(&Ph90Deg_inst, n_coeffs, (float32_t *)coeff_p, &StateF32[0], block_size);
}
else {
coeff_p = NULL; // Stops further FIR filtering for Hilbert
// Serial.println("Hilbert: Missing FIR Coefficients or number > 251");
}
// For the equalizing delay in q, if n_coeffs==19, n_delay=9
// Max of 251 coeffs needs a delay of 125 sample periods.
n_delay = (uint8_t)((n_coeffs-1)/2);
in_index = n_delay;
out_index = 0;
for (uint16_t i=0; i<256; i++){
delayData[i] = 0.0F;
}
} // End of begin()
void showError(uint16_t e) {
errorPrint = e;
}
void update(void);
private:
uint16_t block_size = AUDIO_BLOCK_SAMPLES;
// Two input data pointers
audio_block_f32_t *inputQueueArray_f32[2];
// One output pointer
audio_block_f32_t *blockOut_i;
#if TEST_TIME_90D
// *Temporary* - allows measuring time in microseconds for each part of the update()
elapsedMicros tElapse;
int32_t iitt = 999000; // count up to a million during startup
#endif
// Control error printing in update() 0=No print
uint16_t errorPrint = 0;
//float32_t tmpHil[5]={0.0, 1.0, 0.0, -1.0, 0.0}; coeff_p = &tmpHil[0];
// pointer to current coefficients or NULL
const float32_t *coeff_p = NULL;
uint16_t n_coeffs = 0;
// Variables for the delayed q-channel:
// For the q-channel, we need a delay of ((Ncoeff - 1) / 2) samples. This
// is 9 delay for 19 coefficient FIR. This can be implemented as a simple circular
// buffer if we make the buffer a power of 2 in length and binary-truncate the index.
// Choose 2^8 = 256. For a 251 long Hilbert this wastes 256-128-125 = 3, but
// more for shorter Hilberts.
float32_t delayData[256]; // The circular delay line
uint16_t in_index;
uint16_t out_index;
// And a mask to make the circular buffer limit to a power of 2
uint16_t delayBufferMask = 0X00FF;
uint16_t n_delay;
// ARM DSP Math library filter instance
arm_fir_instance_f32 Ph90Deg_inst;
float32_t StateF32[AUDIO_BLOCK_SAMPLES + HILBERT_MAX_COEFFS];
};
#endif