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203 lines
8.0 KiB
203 lines
8.0 KiB
/**
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******************************************************************************
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* @file AudioFilterConvolution_F32.cpp
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* @author Giuseppe Callipo - IK8YFW - ik8yfw@libero.it
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* @version V2.0.0
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* @date 06-02-2021
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* @brief F32 Filter Convolution
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*
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******************************************************************************
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******************************************************************************
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This software is based on the AudioFilterConvolution routine
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Written by Brian Millier on Mar 2017
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https://circuitcellar.com/research-design-hub/fancy-filtering-with-the-teensy-3-6/
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and modified by Giuseppe Callipo - ik8yfw.
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Modifications:
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1) Class refactoring, change some methods visibility;
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2) Filter coefficients calculation included into class;
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3) Change the class for running in both with F32
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OpenAudio_ArduinoLibrary for Teensy;
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4) Added initFilter method for single anf fast initialization and on
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the fly reinititializzation;
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5) Optimize it to use as output audio filter on SDR receiver.
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6) Optimize the time execution
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*******************************************************************/
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/* Additional Notes from Bob
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* Object creations is required. See the OpenAudio_ArduinoLibrary
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* Design Tool for object declarations along with
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* automatic generatin of code. As an example this could produce
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* the following needed global code
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* AudioFilterConvolution_F32 FilterConv(audio_settings);
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* AudioConnection_F32 patchCord1(FilterConv,0,Output_i2s,0);
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* AudioConnection_F32 patchCord2(FilterConv,0,Output_i2s,1);
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*
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* There are three class functions:
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* void initFilter(float32_t fc, float32_t Astop,
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* int type, float32_t dfc);
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* void passThrough(int stat);
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* float32_t* getCoeffPtr(void);
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*
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* initFilter() is used to design the "mask" function that sets the filter
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* response. All filters use the Kaiser window that is characterized by
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* a programable first sidelobe level and decreasing sidelobes as the
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* frequency departs from the pass band. For many applications this is an
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* excellent response. The response type is set by the integer "type." The
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* options are:
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* type=LOWPASS Low pass with fc cutoff frequency and dfc not used.
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* type=HIGHPASS High pass with fc cutoff frequency and dfc not used.
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* type=BANDPASS Band pass with fc center frequency and dfc pass band width.
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* type=BANDREJECT Band reject with fc center frequency and dfc reject band width.
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* type=HILBERT Hilbert transform. *** Not Currently Available ***
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*
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* Astop is a value in dB that approximates the first sidelobe level
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* going into the stop band. This is a feature of the Kaiser window that
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* allows trading off first sidelobe levels against the speed of
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* transition from the passband to the stop band(s). Values in the 25
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* to 70 dB range work well.
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*
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* Two examples of initFilter():
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* // IK8YFW CW - Centered at 800Hz, ( 40 db x oct ), 2=BPF, width = 1200Hz
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* FilterConv.initFilter((float32_t)800, 40, 2, 1200.0);
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*
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* // IK8YFWSSB - Centered at 1500Hz, ( 60 db x oct ), 2=BPF, width = 3000Hz
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* FilterConv.initFilter((float32_t)1500, 60, 2, 3000.0);
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*
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* The band edges of filters here are specified by their -6 dB points.
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*
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* passThrough(int stat) allows data for this filter object to be passed through
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* unchanged with stat=1. The dfault is stat=0.
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*
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* getCoeffPtr() returns a pointer to the coefficient array. To use this, compute
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* the coefficients of a 512 tap FIR filter with the desired response. Then
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* load the 512 float32_t buffer with the coefficients. Disabling the audio
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* path may be needed to prevent "pop" noises.
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*
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* An alternate way to specify
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*
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* This class is compatible with, and included in, OpenAudio_ArduinoLibrary_F32.
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* If you are using the include OpenAudio_ArduinoLibrary.h, this class's
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* include file will be swept in.
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*
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* Only block_size = 128 is supported.
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* Sample rate can be changed.
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*
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* Speed of execution is the force behind the convolution filter form.
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* Measured 128 sample in update() is 139 microseconds (T4.x).
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* Comparison with a conventional FIR from this library, the
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* AudioFilterFIRGeneral_F32, showed that a 512 tap FIR gave
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* essentially the same response but was somewhat slower at
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* 225 microseconds per 128 update. Also, note that this form of the
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* computation uses about 44 kB of data memory where the direct FIR
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* uses about 10 kB.
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*
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* See the example TestConvolutionFilter.ino for more inforation on the
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* use of this class.
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*
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* NOTE: This filter can be run under Teensy 3.5, 3.6, 4.0, 4.1 ONLY
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*
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* Removed #defines that were not needed. Thanks K7MDL. Bob 6 Mar 2022
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* Separated Teensy 3 and 4 parts. Thanks Paul Bob 16 Jan 2023
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*
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* ************************************************************ */
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// Only exists for T3.5 through T4.1:
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#if defined(__MK64FX512__) || defined(__MK66FX1M0__) || defined(__IMXRT1062__)
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#ifndef AudioFilterConvolution_F32_h_
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#define AudioFilterConvolution_F32_h_
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#include <AudioStream_F32.h>
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#include "arm_math.h"
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#include "arm_common_tables.h"
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#if defined(__IMXRT1062__)
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#include "arm_const_structs.h"
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#endif
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#define MAX_NUMCOEF 513
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#define PIH_F32 1.5707963f
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#define LOWPASS 0
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#define HIGHPASS 1
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#define BANDPASS 2
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#define BANDREJECT 3
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#define HILBERT 4
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class AudioFilterConvolution_F32 :
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public AudioStream_F32
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{
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public:
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AudioFilterConvolution_F32(void) :
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AudioStream_F32(1, inputQueueArray_F32)
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{
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fs = AUDIO_SAMPLE_RATE;
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//block_size = 128; // Always
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// INFO: __MK20DX128__ T_LC; __MKL26Z64__ T3.0; __MK20DX256__ T3.1 and T3.2
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// __MK64FX512__) T3.5; __MK66FX1M0__ T3.6; __IMXRT1062__ T4.0 and T4.1
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#if defined(__MK64FX512__) || defined(__MK66FX1M0__)
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arm_cfft_radix4_init_f32(&fft_instFwd, 1024, 0, 1);
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arm_cfft_radix4_init_f32(&fft_instRev, 1024, 1, 1);
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// arm CMSIS library has predefined structures of type arm_cfft_instance_f32
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// arm_cfft_sR_f32_len1024 is one of the structures
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#endif
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};
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AudioFilterConvolution_F32(const AudioSettings_F32 &settings) :
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AudioStream_F32(1, inputQueueArray_F32)
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{
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// Performs the first initialize
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fs = settings.sample_rate_Hz;
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#if defined(__MK64FX512__) || defined(__MK66FX1M0__)
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arm_cfft_radix4_init_f32(&fft_instFwd, 1024, 0, 1);
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arm_cfft_radix4_init_f32(&fft_instRev, 1024, 1, 1);
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#endif
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};
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virtual void update(void);
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void passThrough(int stat);
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void initFilter (void) {impulse(FIR_Coef);}
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void initFilter (float32_t fc, float32_t Astop,
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int type, float32_t dfc);
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float32_t* getCoeffPtr(void) {return &FIR_Coef[0];}
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private:
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float32_t fs;
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audio_block_f32_t *inputQueueArray_F32[1];
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float32_t *sp_L;
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volatile uint8_t state;
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int i;
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int k;
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int l;
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int passThru=0;
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int enabled=0;
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float32_t FIR_Coef[MAX_NUMCOEF];
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const uint32_t FFT_length = 1024;
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float32_t FIR_filter_mask[2048] __attribute__((aligned(4)));
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float32_t buffer[2048] __attribute__((aligned(4)));
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float32_t tbuffer[2048]__attribute__((aligned(4)));
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float32_t FFT_buffer[2048] __attribute__((aligned(4)));
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float32_t iFFT_buffer[2048] __attribute__((aligned(4)));
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float32_t last_sample_buffer_L[512];
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void impulse(float32_t *coefs);
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int aaa = 0;
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float32_t Izero (float32_t x);
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float32_t m_sinc(int m, float32_t fc);
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void calc_FIR_coeffs (float32_t * coeffs, int numCoeffs,
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float32_t fc, float32_t Astop,
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int type, float32_t dfc,
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float32_t Fsamprate);
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#if defined(__MK64FX512__) || defined(__MK66FX1M0__)
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arm_cfft_radix4_instance_f32 fft_instFwd, fft_instRev;
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// #elif defined(__IMXRT1062__)
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// arm_cfft_instance_f32 arm_cfft_sR_f32_len1024 is built into cmsis
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#endif
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};
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// end of read only once
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#endif
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// End T3.5, T3.6 or T4.x
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#endif
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