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304 lines
11 KiB
304 lines
11 KiB
/*
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* Analyze_fft1024_iq_F32.h Assembled by Bob Larkin 3 Mar 2021
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*
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* Rev 6 Mar 2021 - Added setXAxis()
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* Does Fast Fourier Transform of a 1024 point complex (I-Q) input.
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* Output is one of three measures of the power in each of the 1024
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* output bins, Power, RMS level or dB relative to a full scale
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* sine wave. Windowing of the input data is provided for to reduce
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* spreading of the power in the output bins. All inputs are Teensy
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* floating point extension (_F32) and all outputs are floating point.
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*
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* Features include:
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* * I and Q inputs are OpenAudio_Arduino Library F32 compatible.
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* * FFT output for every 512 inputs to overlapped FFTs to
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* compensate for windowing.
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* * Windowing None, Hann, Kaiser and Blackman-Harris.
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* * Multiple bin-sum output to simulate wider bins.
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* * Power averaging of multiple FFT
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* * Soon: F32 audio outputs for I & Q
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*
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* Conversion Copyright (c) 2021 Bob Larkin
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* Same MIT license as PJRC:
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*
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* From original real FFT:
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* Audio Library for Teensy 3.X
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* Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com
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*
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* Development of this audio library was funded by PJRC.COM, LLC by sales of
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* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
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* open source software by purchasing Teensy or other PJRC products.
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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, development funding notice, and this permission
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* notice shall be included in all 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
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* THE SOFTWARE.
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*/
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/* Does complex input FFT of 1024 points. Multiple non-audio (via functions)
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* output formats of RMS (same as I16 version, and default),
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* Power or dBFS (full scale). Output can be bin by bin or a pointer to
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* the output array is available. Several window functions are provided by
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* in-class design, or a custom window can be provided from the INO.
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*
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* Functions (See comments below and #defines above:
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* bool available()
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* float read(unsigned int binNumber)
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* float read(unsigned int binFirst, unsigned int binLast)
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* int windowFunction(int wNum)
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* int windowFunction(int wNum, float _kdb) // Kaiser only
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* float* getData(void)
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* float* getWindow(void)
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* void putWindow(float *pwin)
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* void setNAverage(int NAve) // >=1
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* void setOutputType(int _type)
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* void setXAxis(uint8_t _xAxis) // 0, 1, 2, 3
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*
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* x-Axis direction and offset per setXAxis(xAxis) for sine to I
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* and cosine to Q.
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* If xAxis=0 f=fs/2 in middle, f=0 on right edge
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* If xAxis=1 f=fs/2 in middle, f=0 on left edge
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* If xAxis=2 f=fs/2 on left edge, f=0 in middle
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* If xAxis=3 f=fs/2 on right edgr, f=0 in middle
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* If there is 180 degree phase shift to I or Q these all get reversed.
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*
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* Timing, max is longest update() time: (TBD)
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* T3.6 Windowed, RMS out, - uSec max
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* T3.6 Windowed, Power Out, - uSec max
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* T3.6 Windowed, dBFS out, - uSec max
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* No Window saves 60 uSec on T3.6 for any output.
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* T4.0 Windowed, RMS Out, - uSec
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*
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* Scaling:
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* Full scale for floating point DSP is a nebulous concept. Normally the
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* full scale is -1.0 to +1.0. This is an unscaled FFT and for a sine
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* wave centered in frequency on a bin and of FS amplitude, the power
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* at that center bin will grow by 1024^2/4 = 262144 without windowing.
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* Windowing loss cuts this down. The RMS level can grow to sqrt(262144)
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* or 512. The dBFS has been scaled to make this max value 0 dBFS by
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* removing 54.2 dB. With floating point, the dynamic range is maintained
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* no matter how it is scaled, but this factor needs to be considered
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* when building the INO.
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*/
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#ifndef analyze_fft1024iq_h_
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#define analyze_fft1024iq_h_
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#include "Arduino.h"
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#include "AudioStream_F32.h"
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#include "arm_math.h"
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#include "mathDSP_F32.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 FFT_RMS 0
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#define FFT_POWER 1
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#define FFT_DBFS 2
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#define NO_WINDOW 0
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#define AudioWindowNone 0
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#define AudioWindowHanning1024 1
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#define AudioWindowKaiser1024 2
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#define AudioWindowBlackmanHarris1024 3
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class AudioAnalyzeFFT1024_IQ_F32 : public AudioStream_F32 {
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//GUI: inputs:2, outputs:4 //this line used for automatic generation of GUI node
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//GUI: shortName:AnalyzeFFT1024IQ
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public:
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AudioAnalyzeFFT1024_IQ_F32() : AudioStream_F32(2, inputQueueArray) {
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// __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(__IMXRT1062__)
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// Teensy4 core library has the right files for new FFT
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// arm CMSIS library has predefined structures of type arm_cfft_instance_f32
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Sfft = arm_cfft_sR_f32_len1024; // This is one of the structures
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#else
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arm_cfft_radix4_init_f32(&fft_inst, 1024, 0, 1); // for T3.x
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#endif
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useHanningWindow();
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}
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// There is no varient for "settings," as blocks other than 128 are
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// not supported and, nothing depends on sample rate so we don't need that.
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// Returns true when output data is available.
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bool available() {
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if (outputflag == true) {
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outputflag = false; // No double returns
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return true;
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}
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return false;
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}
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// Returns a single bin output
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float read(unsigned int binNumber) {
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if (binNumber>1023 || binNumber<0) return 0.0;
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return output[binNumber];
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}
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// Return sum of several bins. Normally use with power output.
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// This produces the equivalent of bigger bins.
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float read(unsigned int binFirst, unsigned int binLast) {
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if (binFirst > binLast) {
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unsigned int tmp = binLast;
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binLast = binFirst;
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binFirst = tmp;
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}
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if (binFirst > 1023) return 0.0;
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if (binLast > 1023) binLast = 1023;
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float sum = 0;
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do {
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sum += output[binFirst++];
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} while (binFirst <= binLast);
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return sum;
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}
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// Sets None, Hann, or Blackman-Harris window with no parameter
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int windowFunction(int wNum) {
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if(wNum == AudioWindowKaiser1024)
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return -1; // Kaiser needs the kdb
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windowFunction(wNum, 0.0f);
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return 0;
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}
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int windowFunction(int wNum, float _kdb) {
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float kd;
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pWin = window;
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if(wNum == NO_WINDOW)
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pWin = NULL;
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else if (wNum == AudioWindowKaiser1024) {
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if(_kdb<20.0f)
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kd = 20.0f;
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else
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kd = _kdb;
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useKaiserWindow(kd);
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}
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else if (wNum == AudioWindowBlackmanHarris1024)
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useBHWindow();
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else
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useHanningWindow(); // Default
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return 0;
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}
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// Fast pointer transfer. Be aware that the data will go away
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// after the next 256 data points occur.
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float* getData(void) {
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// available() sets outputflag false
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return output;
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}
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// You can use this to design windows
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float* getWindow(void) {
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return window;
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}
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// Bring custom window from the INO
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void putWindow(float *pwin) {
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float *p = window;
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for(int i=0; i<1024; i++)
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*p++ = *pwin++; // Copy for the FFT
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}
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// Number of FFT averaged in the output
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void setNAverage(int _nAverage) {
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nAverage = _nAverage;
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}
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// Output RMS (default), power or dBFS (FFT_RMS, FFT_POWER, FFT_DBFS)
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void setOutputType(int _type) {
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outputType = _type;
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}
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// xAxis, bit 0 left/right; bit 1 low to high; default 0X03
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void setXAxis(uint8_t _xAxis) {
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xAxis = _xAxis;
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}
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virtual void update(void);
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private:
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float output[1024];
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float window[1024];
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float *pWin = window;
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float fft_buffer[2048];
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float sumsq[1024]; // Avoid re-use of output[]
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uint8_t state = 0;
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bool outputflag = false;
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audio_block_f32_t *inputQueueArray[2];
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audio_block_f32_t *blocklist_i[8];
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audio_block_f32_t *blocklist_q[8];
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#if defined(__IMXRT1062__)
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// For T4.x
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// const static arm_cfft_instance_f32 arm_cfft_sR_f32_len1024;
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arm_cfft_instance_f32 Sfft;
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#else
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arm_cfft_radix4_instance_f32 fft_inst;
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#endif
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int outputType = FFT_RMS; //Same type as I16 version init
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int count = 0;
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int nAverage = 1;
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uint8_t xAxis = 0x03;
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// The Hann window is a good all-around window
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void useHanningWindow(void) {
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for (int i=0; i < 1024; i++) {
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// 2*PI/1023 = 0.006141921
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window[i] = 0.5*(1.0 - cosf(0.006141921f*(float)i));
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}
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}
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// Blackman-Harris produces a first sidelobe more than 90 dB down.
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// The price is a bandwidth of about 2 bins. Very useful at times.
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void useBHWindow(void) {
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for (int i=0; i < 1024; i++) {
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float kx = 0.006141921; // 2*PI/1023
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int ix = (float) i;
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window[i] = 0.35875 -
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0.48829*cosf( kx*ix) +
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0.14128*cosf(2.0f*kx*ix) -
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0.01168*cosf(3.0f*kx*ix);
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}
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}
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/* The windowing function here is that of James Kaiser. This has a number
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* of desirable features. The sidelobes drop off as the frequency away from a transition.
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* Also, the tradeoff of sidelobe level versus cutoff rate is variable.
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* Here we specify it in terms of kdb, the highest sidelobe, in dB, next to a sharp cutoff. For
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* calculating the windowing vector, we need a parameter beta, found as follows:
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*/
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void useKaiserWindow(float kdb) {
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float32_t beta, kbes, xn2;
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mathDSP_F32 mathEqualizer; // For Bessel function
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if (kdb < 20.0f)
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beta = 0.0;
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else
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beta = -2.17+0.17153*kdb-0.0002841*kdb*kdb; // Within a dB or so
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// Note: i0f is the fp zero'th order modified Bessel function (see mathDSP_F32.h)
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kbes = 1.0f / mathEqualizer.i0f(beta); // An additional derived parameter used in loop
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for (int n=0; n<512; n++) {
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xn2 = 0.5f+(float32_t)n;
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// 4/(1023^2)=0.00000382215877f
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xn2 = 0.00000382215877f*xn2*xn2;
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window[511 - n]=kbes*(mathEqualizer.i0f(beta*sqrtf(1.0-xn2)));
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window[512 + n] = window[511 - n];
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}
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}
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};
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#endif
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