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178 lines
8.4 KiB
178 lines
8.4 KiB
/*


* AudioFilterEqualizer_F32


*


* Created: Bob Larkin W7PUA 8 May 2020


*


* This is a direct translation of the receiver audio equalizer written


* by this author for the opensource DSP10 radio in 1999. See


* http://www.janbob.com/electron/dsp10/dsp10.htm and


* http://www.janbob.com/electron/dsp10/uhf3_35a.zip


*


* Credit and thanks to PJRC, Paul Stoffregen and Teensy products for the audio


* system and library that this is built upon as well as the float32


* work of Chip Audette embodied in the OpenAudio_ArduinoLibrary. Many thanks


* for the library structures and wonderful Teensy products.


*


* This equalizer is specified by an array of 'nBands' frequency bands


* each of of arbitrary frequency span. The first band always starts at


* 0.0 Hz, and that value is not entered. Each band is specified by the upper


* frequency limit to the band.


* The last band always ends at half of the sample frequency, which for 44117 Hz


* sample frequency would be 22058.5. Each band is specified by its upper


* frequency in an .INO supplied array feq[]. The dB level of that band is


* specified by a value, in dB, arranged in an .INO supplied array


* aeq[]. Thus a trivial bass/treble control might look like:


* nBands = 3;


* feq[] = {300.0, 1500.0, 22058.5};


* float32_t bass = 2.5; // in dB, relative to anything


* float32_t treble = 6.0;


* aeq[] = {bass, 0.0, treble};


*


* It may be obvious that this equalizer is a more general case of the common


* functions such as lowpass, bandpass, notch, etc. For instance, a pair


* of band pass filters would look like:


* nBands = 5;


* feq[] = {500.0, 700.0, 2000.0, 2200.0, 22058.5};


* aeq[] = {100.0, 0.0, 100.0, 2.0, 100.0};


* where we added 2 dB of gain to the 2200 to 2400 Hz filter, relative to the 500


* to 700 Hz band.


*


* An octave band equalizer is made by starting at some low frequency, say 40 Hz for the


* first band. The lowest frequency band will be from 0.0 Hz up to that first frequency.


* Next multiply the first frequency by 2, creating in our example, a band from 40.0


* to 80 Hz. This is continued until the last frequency is about 22058 Hz.


* This works out to require 10 bands, as follows:


* nBands = 10;


* feq[] = { 40.0, 80.0, 160.0, 320.0, 640.0, 1280.0, 2560.0, 5120.0, 10240.0, 22058.5};


* aeq[] = { 5.0, 4.0, 2.0, 3.0, 4.0, 1.0, 3.0, 6.0, 3.0, 0.5 };


*


* For a "half octave" equalizer, multiply each upper band limit by the square root of 2 = 1.414


* to get the next band limit. For that case, feq[] would start with a sequence


* like 40, 56.56, 80.00, 113.1, 160.0, ... for a total of about 20 bands.


*


* How well all of this is achieved depends on the number of FIR coefficients


* being used. In the Teensy 3.6 / 4.0 the resourses allow a hefty number,


* say 201, of coefficients to be used without stealing all the processor time


* (see Timing, below). The coefficient and FIR memory is sized for a maximum of


* 250 coefficients, but can be recompiled for bigger with the define FIR_MAX_COEFFS.


* To simplify calculations, the number of FIR coefficients should be odd. If not


* odd, the number will be reduced by one, quietly.


*


* If you try to make the bands too narrow for the number of FIR coeffficients,


* the approximation to the desired curve becomes poor. This can all be evaluated


* by the function getResponse(nPoints, pResponse) which fills an .INOsupplied array


* pResponse[nPoints] with the frequency response of the equalizer in dB. The nPoints


* are spread evenly between 0.0 and half of the sample frequency.


*


* Initialization is a 2step process. This makes it practical to change equalizer


* levels onthefly. The constructor starts up with a 4tap FIR setup for direct


* pass through. Then the setup() in the .INO can specify the equalizer.


* The newEqualizer() function has several parameters, the number of equalizer bands,


* the frequencies of the bands, and the sidelobe level. All of these can be changed


* dynamically. This function can be changed dynamically, but it may be desireable to


* mute the audio during the change to prevent clicks.


*


* This 16bit integer version adjusts the maximum coefficient size to scale16 in the calls


* to both equalizerNew() and getResponse(). Broadband equalizers can work with fullscale


* 32767.0f sorts of levels, where narrow band filtering may need smaller values to


* prevent overload. Experiment and check carefully. Use lower values if there are doubts.


*


* For a passthrough function, something like this (which can be intermixed with fancy equalizers):


* float32_t fBand[] = {10000.0f, 22058.5f};


* float32_t dbBand[] = {0.0f, 0.0f};


* equalize1.equalizerNew(2, &fBand[0], &dbBand[0], 4, &equalizeCoeffs[0], 30.0f, 32767.0f);


*


* Measured timing of update() for a 128 sample block, Teensy 3.6:


* Fixed time 13 microseconds


* Per FIR Coefficient time 2.5 microseconds


* Total for 199 FIR Coefficients = 505 microseconds (17.4% of 44117 Hz available time)


*


* Per FIR Coefficient, Teensy 4.0, 0.44 microseconds


*


* Copyright (c) 2020 Bob Larkin


* Any snippets of code from PJRC or Chip Audette used here brings with it


* the associated license.


*


* In addition, work here is covered by MIT LIcense:


*


* 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.


*/




#ifndef _filter_equalizer_f32_h


#define _filter_equalizer_f32_h




#include "Arduino.h"


#include "AudioStream_F32.h"


#include "arm_math.h"


#include "mathDSP_F32.h"




#ifndef MF_PI


#define MF_PI 3.1415926f


#endif




// Temporary timing test


#define TEST_TIME_EQ 0




#define EQUALIZER_MAX_COEFFS 251




#define ERR_EQ_BANDS 1


#define ERR_EQ_SIDELOBES 2


#define ERR_EQ_NFIR 3




class AudioFilterEqualizer_F32 : public AudioStream_F32


{


//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node


//GUI: shortName:filter_Equalizer


public:


AudioFilterEqualizer_F32(void): AudioStream_F32(1,inputQueueArray) {


// Initialize FIR instance (ARM DSP Math Library) with default simple passthrough FIR


arm_fir_init_f32(&fir_inst, nFIR, (float32_t *)cf32f, &StateF32[0], (uint32_t)block_size);


}


AudioFilterEqualizer_F32(const AudioSettings_F32 &settings): AudioStream_F32(1,inputQueueArray) {


block_size = settings.audio_block_samples;


sample_rate_Hz = settings.sample_rate_Hz;


arm_fir_init_f32(&fir_inst, nFIR, (float32_t *)cf32f, &StateF32[0], (uint32_t)block_size);


}




uint16_t equalizerNew(uint16_t _nBands, float32_t *feq, float32_t *adb,


uint16_t _nFIR, float32_t *_cf32f, float32_t kdb);


void getResponse(uint16_t nFreq, float32_t *rdb);


void update(void);




private:


audio_block_f32_t *inputQueueArray[1];


uint16_t block_size = AUDIO_BLOCK_SAMPLES;


float32_t firStart[4] = {0.0, 1.0, 0.0, 0.0}; // Initialize to passthrough


float32_t* cf32f = firStart; // pointer to current coefficients


uint16_t nFIR = 4; // Number of coefficients


uint16_t nBands = 2;


float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE;




// *Temporary*  TEST_TIME allows measuring time in microseconds for each part of the update()


#if TEST_TIME_EQ


elapsedMicros tElapse;


int32_t iitt = 999000; // count up to a million during startup


#endif


// ARM DSP Math library filter instance


arm_fir_instance_f32 fir_inst;


float32_t StateF32[AUDIO_BLOCK_SAMPLES + EQUALIZER_MAX_COEFFS]; // max, max


};


#endif






