From ea097196322af263d28aeec26686ed8a5b46b929 Mon Sep 17 00:00:00 2001 From: boblark Date: Sun, 7 Mar 2021 11:17:07 -0800 Subject: [PATCH] add in analyze_fft1024_iq_F32 --- analyze_fft1024_iq_F32.cpp | 187 ++++++++++++++ analyze_fft1024_iq_F32.h | 304 +++++++++++++++++++++++ examples/TestFFT1024iq/TestFFT1024iq.ino | 68 +++++ 3 files changed, 559 insertions(+) create mode 100644 analyze_fft1024_iq_F32.cpp create mode 100644 analyze_fft1024_iq_F32.h create mode 100644 examples/TestFFT1024iq/TestFFT1024iq.ino diff --git a/analyze_fft1024_iq_F32.cpp b/analyze_fft1024_iq_F32.cpp new file mode 100644 index 0000000..10346cc --- /dev/null +++ b/analyze_fft1024_iq_F32.cpp @@ -0,0 +1,187 @@ +/* + * analyze_fft1024_iq_F32.cpp Assembled by Bob Larkin 3 Mar 2021 + * Rev 6 Mar 2021 - Added setXAxis() + * + * Converted to F32 floating point input and also extended + * for complex I and Q inputs + * * Adapted all I/O to be F32 floating point for OpenAudio_ArduinoLibrary + * * Future: Add outputs for I & Q FFT x2 for overlapped FFT + * * Windowing None, Hann, Kaiser and Blackman-Harris. + * + * Conversion Copyright (c) 2021 Bob Larkin + * Same MIT license as PJRC: + * + * Audio Library for Teensy 3.X + * Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com + * + * Development of this audio library was funded by PJRC.COM, LLC by sales of + * Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop + * open source software by purchasing Teensy or other PJRC products. + * + * 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. + */ +#include +#include "analyze_fft1024_iq_F32.h" + +// Note: Suppports block size of 128 only. Very "built in." + +// Move audio data from audio_block_f32_t to the interleaved FFT instance buffer. +static void copy_to_fft_buffer1(void *destination, const void *sourceI, const void *sourceQ) { + const float *srcI = (const float *)sourceI; + const float *srcQ = (const float *)sourceQ; + float *dst = (float *)destination; // part of fft_buffer array. 256 floats per call + for (int i=0; i < 128; i++) { + *dst++ = *srcI++; // real sample, interleave + *dst++ = *srcQ++; // imag + } + } + +static void apply_window_to_fft_buffer1(void *fft_buffer, const void *window) { + float *buf = (float *)fft_buffer; // 0th entry is real (do window) 1st is imag + const float *win = (float *)window; + for (int i=0; i < 1024; i++) { + buf[2*i] *= *win; // real + buf[2*i + 1] *= *win++; // imag + } + } + +void AudioAnalyzeFFT1024_IQ_F32::update(void) { + audio_block_f32_t *block_i,*block_q; + int ii; + + block_i = receiveReadOnly_f32(0); + if (!block_i) return; + block_q = receiveReadOnly_f32(1); + if (!block_q) { + release(block_i); + return; + } + // Here with two new blocks of data + + switch (state) { + case 0: + blocklist_i[0] = block_i; blocklist_q[0] = block_q; + state = 1; + break; + case 1: + blocklist_i[1] = block_i; blocklist_q[1] = block_q; + state = 2; + break; + case 2: + blocklist_i[2] = block_i; blocklist_q[2] = block_q; + state = 3; + break; + case 3: + blocklist_i[3] = block_i; blocklist_q[3] = block_q; + state = 4; + break; + case 4: + blocklist_i[4] = block_i; blocklist_q[4] = block_q; + state = 5; + break; + case 5: + blocklist_i[5] = block_i; blocklist_q[5] = block_q; + state = 6; + break; + case 6: + blocklist_i[6] = block_i; blocklist_q[6] = block_q; + state = 7; + break; + case 7: + blocklist_i[7] = block_i; blocklist_q[7] = block_q; + copy_to_fft_buffer1(fft_buffer+0x000, blocklist_i[0]->data, blocklist_q[0]->data); + copy_to_fft_buffer1(fft_buffer+0x100, blocklist_i[1]->data, blocklist_q[1]->data); + copy_to_fft_buffer1(fft_buffer+0x200, blocklist_i[2]->data, blocklist_q[2]->data); + copy_to_fft_buffer1(fft_buffer+0x300, blocklist_i[3]->data, blocklist_q[3]->data); + copy_to_fft_buffer1(fft_buffer+0x400, blocklist_i[4]->data, blocklist_q[4]->data); + copy_to_fft_buffer1(fft_buffer+0x500, blocklist_i[5]->data, blocklist_q[5]->data); + copy_to_fft_buffer1(fft_buffer+0x600, blocklist_i[6]->data, blocklist_q[6]->data); + copy_to_fft_buffer1(fft_buffer+0x700, blocklist_i[7]->data, blocklist_q[7]->data); + if (pWin) + apply_window_to_fft_buffer1(fft_buffer, window); + +#if defined(__IMXRT1062__) + // Teensyduino core for T4.x supports arm_cfft_f32 + // arm_cfft_f32 (const arm_cfft_instance_f32 *S, float32_t *p1, uint8_t ifftFlag, uint8_t bitReverseFlag) + arm_cfft_f32(&Sfft, fft_buffer, 0, 1); +#else + // For T3.x go back to old (deprecated) style + arm_cfft_radix4_f32(&fft_inst, fft_buffer); +#endif + + count++; + for (int i = 0; i < 512; i++) { + // From complex FFT the "negative frequencies" are mirrors of the frequencies above fs/2. So, we get + // frequencies from 0 to fs by re-arranging the coefficients. These are powers (not Volts) + // See DD4WH SDR + float ss0 = fft_buffer[2 * i] * fft_buffer[2 * i] + + fft_buffer[2 * i + 1] * fft_buffer[2 * i + 1]; + float ss1 = fft_buffer[2 * (i + 512)] * fft_buffer[2 * (i + 512)] + + fft_buffer[2 * (i + 512) + 1] * fft_buffer[2 * (i + 512) + 1]; + + if(count==1) { // Starting new average + sumsq[i+512] = ss0; + sumsq[i] = ss1; + } + else if (count <= nAverage) { // Adding on to average + sumsq[i+512] += ss0; + sumsq[i] += ss1; + } + } + if (count >= nAverage) { // Average is finished + count = 0; + float inAf = 1.0f/(float)nAverage; + for (int i=0; i < 1024; i++) { + // xAxis, bit 0 left/right; bit 1 low to high + if(xAxis & 0X02) + ii = i; + else + ii = i^512; + if(xAxis & 0X01) + ii = (1023 - ii); + + if(outputType==FFT_RMS) + output[i] = sqrtf(inAf*sumsq[ii]); + else if(outputType==FFT_POWER) + output[i] = inAf*sumsq[ii]; + else if(outputType==FFT_DBFS) + output[i] = 10.0f*log10f(inAf*sumsq[ii])-54.1854f; // Scaled to FS sine wave + else + output[i] = 0.0f; + } + } // end of Average is Finished + outputflag = true; + + release(blocklist_i[0]); release(blocklist_q[0]); + release(blocklist_i[1]); release(blocklist_q[1]); + release(blocklist_i[2]); release(blocklist_q[2]); + release(blocklist_i[3]); release(blocklist_q[3]); + + blocklist_i[0] = blocklist_i[4]; + blocklist_i[1] = blocklist_i[5]; + blocklist_i[2] = blocklist_i[6]; + blocklist_i[3] = blocklist_i[7]; + blocklist_q[0] = blocklist_q[4]; + blocklist_q[1] = blocklist_q[5]; + blocklist_q[2] = blocklist_q[6]; + blocklist_q[3] = blocklist_q[7]; + state = 4; + break; // From case 7 + } // End of switch & case 7 + } // End update() diff --git a/analyze_fft1024_iq_F32.h b/analyze_fft1024_iq_F32.h new file mode 100644 index 0000000..392f641 --- /dev/null +++ b/analyze_fft1024_iq_F32.h @@ -0,0 +1,304 @@ +/* + * Analyze_fft1024_iq_F32.h Assembled by Bob Larkin 3 Mar 2021 +* + * Rev 6 Mar 2021 - Added setXAxis() + * Does Fast Fourier Transform of a 1024 point complex (I-Q) input. + * Output is one of three measures of the power in each of the 1024 + * output bins, Power, RMS level or dB relative to a full scale + * sine wave. Windowing of the input data is provided for to reduce + * spreading of the power in the output bins. All inputs are Teensy + * floating point extension (_F32) and all outputs are floating point. + * + * Features include: + * * I and Q inputs are OpenAudio_Arduino Library F32 compatible. + * * FFT output for every 512 inputs to overlapped FFTs to + * compensate for windowing. + * * Windowing None, Hann, Kaiser and Blackman-Harris. + * * Multiple bin-sum output to simulate wider bins. + * * Power averaging of multiple FFT + * * Soon: F32 audio outputs for I & Q + * + * Conversion Copyright (c) 2021 Bob Larkin + * Same MIT license as PJRC: + * + * From original real FFT: + * Audio Library for Teensy 3.X + * Copyright (c) 2014, Paul Stoffregen, paul@pjrc.com + * + * Development of this audio library was funded by PJRC.COM, LLC by sales of + * Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop + * open source software by purchasing Teensy or other PJRC products. + * + * 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. + */ + +/* Does complex input FFT of 1024 points. Multiple non-audio (via functions) + * output formats of RMS (same as I16 version, and default), + * Power or dBFS (full scale). Output can be bin by bin or a pointer to + * the output array is available. Several window functions are provided by + * in-class design, or a custom window can be provided from the INO. + * + * Functions (See comments below and #defines above: + * bool available() + * float read(unsigned int binNumber) + * float read(unsigned int binFirst, unsigned int binLast) + * int windowFunction(int wNum) + * int windowFunction(int wNum, float _kdb) // Kaiser only + * float* getData(void) + * float* getWindow(void) + * void putWindow(float *pwin) + * void setNAverage(int NAve) // >=1 + * void setOutputType(int _type) + * void setXAxis(uint8_t _xAxis) // 0, 1, 2, 3 + * + * x-Axis direction and offset per setXAxis(xAxis) for sine to I + * and cosine to Q. + * If xAxis=0 f=fs/2 in middle, f=0 on right edge + * If xAxis=1 f=fs/2 in middle, f=0 on left edge + * If xAxis=2 f=fs/2 on left edge, f=0 in middle + * If xAxis=3 f=fs/2 on right edgr, f=0 in middle + * If there is 180 degree phase shift to I or Q these all get reversed. + * + * Timing, max is longest update() time: (TBD) + * T3.6 Windowed, RMS out, - uSec max + * T3.6 Windowed, Power Out, - uSec max + * T3.6 Windowed, dBFS out, - uSec max + * No Window saves 60 uSec on T3.6 for any output. + * T4.0 Windowed, RMS Out, - uSec + * + * Scaling: + * Full scale for floating point DSP is a nebulous concept. Normally the + * full scale is -1.0 to +1.0. This is an unscaled FFT and for a sine + * wave centered in frequency on a bin and of FS amplitude, the power + * at that center bin will grow by 1024^2/4 = 262144 without windowing. + * Windowing loss cuts this down. The RMS level can grow to sqrt(262144) + * or 512. The dBFS has been scaled to make this max value 0 dBFS by + * removing 54.2 dB. With floating point, the dynamic range is maintained + * no matter how it is scaled, but this factor needs to be considered + * when building the INO. + */ + +#ifndef analyze_fft1024iq_h_ +#define analyze_fft1024iq_h_ + +#include "Arduino.h" +#include "AudioStream_F32.h" +#include "arm_math.h" +#include "mathDSP_F32.h" +#if defined(__IMXRT1062__) +#include "arm_const_structs.h" +#endif + +#define FFT_RMS 0 +#define FFT_POWER 1 +#define FFT_DBFS 2 + +#define NO_WINDOW 0 +#define AudioWindowNone 0 +#define AudioWindowHanning1024 1 +#define AudioWindowKaiser1024 2 +#define AudioWindowBlackmanHarris1024 3 + +class AudioAnalyzeFFT1024_IQ_F32 : public AudioStream_F32 { +//GUI: inputs:2, outputs:4 //this line used for automatic generation of GUI node +//GUI: shortName:AnalyzeFFT1024IQ +public: + AudioAnalyzeFFT1024_IQ_F32() : AudioStream_F32(2, inputQueueArray) { + // __MK20DX128__ T_LC; __MKL26Z64__ T3.0; __MK20DX256__T3.1 and T3.2 + // __MK64FX512__) T3.5; __MK66FX1M0__ T3.6; __IMXRT1062__ T4.0 and T4.1 +#if defined(__IMXRT1062__) + // Teensy4 core library has the right files for new FFT + // arm CMSIS library has predefined structures of type arm_cfft_instance_f32 + Sfft = arm_cfft_sR_f32_len1024; // This is one of the structures +#else + arm_cfft_radix4_init_f32(&fft_inst, 1024, 0, 1); // for T3.x +#endif + useHanningWindow(); + } + // There is no varient for "settings," as blocks other than 128 are + // not supported and, nothing depends on sample rate so we don't need that. + + // Returns true when output data is available. + bool available() { + if (outputflag == true) { + outputflag = false; // No double returns + return true; + } + return false; + } + + // Returns a single bin output + float read(unsigned int binNumber) { + if (binNumber>1023 || binNumber<0) return 0.0; + return output[binNumber]; + } + + // Return sum of several bins. Normally use with power output. + // This produces the equivalent of bigger bins. + float read(unsigned int binFirst, unsigned int binLast) { + if (binFirst > binLast) { + unsigned int tmp = binLast; + binLast = binFirst; + binFirst = tmp; + } + if (binFirst > 1023) return 0.0; + if (binLast > 1023) binLast = 1023; + float sum = 0; + do { + sum += output[binFirst++]; + } while (binFirst <= binLast); + return sum; + } + + // Sets None, Hann, or Blackman-Harris window with no parameter + int windowFunction(int wNum) { + if(wNum == AudioWindowKaiser1024) + return -1; // Kaiser needs the kdb + windowFunction(wNum, 0.0f); + return 0; + } + + int windowFunction(int wNum, float _kdb) { + float kd; + pWin = window; + if(wNum == NO_WINDOW) + pWin = NULL; + else if (wNum == AudioWindowKaiser1024) { + if(_kdb<20.0f) + kd = 20.0f; + else + kd = _kdb; + useKaiserWindow(kd); + } + else if (wNum == AudioWindowBlackmanHarris1024) + useBHWindow(); + else + useHanningWindow(); // Default + return 0; + } + + // Fast pointer transfer. Be aware that the data will go away + // after the next 256 data points occur. + float* getData(void) { + // available() sets outputflag false + return output; + } + + // You can use this to design windows + float* getWindow(void) { + return window; + } + + // Bring custom window from the INO + void putWindow(float *pwin) { + float *p = window; + for(int i=0; i<1024; i++) + *p++ = *pwin++; // Copy for the FFT + } + + // Number of FFT averaged in the output + void setNAverage(int _nAverage) { + nAverage = _nAverage; + } + + // Output RMS (default), power or dBFS (FFT_RMS, FFT_POWER, FFT_DBFS) + void setOutputType(int _type) { + outputType = _type; + } + + // xAxis, bit 0 left/right; bit 1 low to high; default 0X03 + void setXAxis(uint8_t _xAxis) { + xAxis = _xAxis; + } + + virtual void update(void); + +private: + float output[1024]; + float window[1024]; + float *pWin = window; + float fft_buffer[2048]; + float sumsq[1024]; // Avoid re-use of output[] + uint8_t state = 0; + bool outputflag = false; + audio_block_f32_t *inputQueueArray[2]; + audio_block_f32_t *blocklist_i[8]; + audio_block_f32_t *blocklist_q[8]; + +#if defined(__IMXRT1062__) + // For T4.x + // const static arm_cfft_instance_f32 arm_cfft_sR_f32_len1024; + arm_cfft_instance_f32 Sfft; +#else + arm_cfft_radix4_instance_f32 fft_inst; +#endif + + int outputType = FFT_RMS; //Same type as I16 version init + int count = 0; + int nAverage = 1; + uint8_t xAxis = 0x03; + + // The Hann window is a good all-around window + void useHanningWindow(void) { + for (int i=0; i < 1024; i++) { + // 2*PI/1023 = 0.006141921 + window[i] = 0.5*(1.0 - cosf(0.006141921f*(float)i)); + } + } + + // Blackman-Harris produces a first sidelobe more than 90 dB down. + // The price is a bandwidth of about 2 bins. Very useful at times. + void useBHWindow(void) { + for (int i=0; i < 1024; i++) { + float kx = 0.006141921; // 2*PI/1023 + int ix = (float) i; + window[i] = 0.35875 - + 0.48829*cosf( kx*ix) + + 0.14128*cosf(2.0f*kx*ix) - + 0.01168*cosf(3.0f*kx*ix); + } + } + + /* The windowing function here is that of James Kaiser. This has a number + * of desirable features. The sidelobes drop off as the frequency away from a transition. + * Also, the tradeoff of sidelobe level versus cutoff rate is variable. + * Here we specify it in terms of kdb, the highest sidelobe, in dB, next to a sharp cutoff. For + * calculating the windowing vector, we need a parameter beta, found as follows: + */ + void useKaiserWindow(float kdb) { + float32_t beta, kbes, xn2; + mathDSP_F32 mathEqualizer; // For Bessel function + + if (kdb < 20.0f) + beta = 0.0; + else + beta = -2.17+0.17153*kdb-0.0002841*kdb*kdb; // Within a dB or so + + // Note: i0f is the fp zero'th order modified Bessel function (see mathDSP_F32.h) + kbes = 1.0f / mathEqualizer.i0f(beta); // An additional derived parameter used in loop + for (int n=0; n<512; n++) { + xn2 = 0.5f+(float32_t)n; + // 4/(1023^2)=0.00000382215877f + xn2 = 0.00000382215877f*xn2*xn2; + window[511 - n]=kbes*(mathEqualizer.i0f(beta*sqrtf(1.0-xn2))); + window[512 + n] = window[511 - n]; + } + } + }; +#endif diff --git a/examples/TestFFT1024iq/TestFFT1024iq.ino b/examples/TestFFT1024iq/TestFFT1024iq.ino new file mode 100644 index 0000000..9ca422f --- /dev/null +++ b/examples/TestFFT1024iq/TestFFT1024iq.ino @@ -0,0 +1,68 @@ + +// TestFFT1024iq.ino for Teensy 3.x, 4.x +// Bob Larkin 3 March 2021// Rev to include xAxis control. 6 Mar 2021 + +// Generate Sin and Cosine pair and input to IQ FFT. +// Serial Print out powers of all 1024 bins in +// dB relative to Sine Wave Full Scale + +// Public Domain + +#include "OpenAudio_ArduinoLibrary.h" +#include "AudioStream_F32.h" + +// GUItool: begin automatically generated code +AudioSynthSineCosine_F32 sine_cos1; //xy=76,532 +AudioAnalyzeFFT1024_IQ_F32 FFT1024iq1; //xy=243,532 +AudioOutputI2S_F32 audioOutI2S1; //xy=246,591 +AudioConnection_F32 patchCord1(sine_cos1, 0, FFT1024iq1, 0); +AudioConnection_F32 patchCord2(sine_cos1, 1, FFT1024iq1, 1); +// GUItool: end automatically generated code + +void setup(void) { + float* pPwr; + + Serial.begin(9600); + delay(1000); + AudioMemory_F32(50); + Serial.println("FFT1024IQ Test"); + + sine_cos1.amplitude(1.0f); // Initialize Waveform Generator + + // Pick T3.6 bin center + //sine_cos1.frequency(689.33); + + // or pick T4.x bin center + sine_cos1.frequency(689.0625f); + + // or pick any old frequency + //sine_cos1.frequency(7100.0); + + // elect the output format + FFT1024iq1.setOutputType(FFT_DBFS); + + // Select the wndow function + //FFT1024iq1.windowFunction(AudioWindowNone); + //FFT1024iq1.windowFunction(AudioWindowHanning1024); + //FFT1024iq1.windowFunction(AudioWindowKaiser1024, 55.0f); + FFT1024iq1.windowFunction(AudioWindowBlackmanHarris1024); + + // Uncomment to Serial print window function + //float* pw = FFT1024iq1.getWindow(); // Print window + //for (int i=0; i<512; i++) Serial.println(pw[i], 4); + + // xAxis, bit 0 left/right; bit 1 low to high; default 0X03 + FFT1024iq1.setXAxis(0X03); + + delay(1000); + // Print output, once + if( FFT1024iq1.available() ) { + pPwr = FFT1024iq1.getData(); + for(int i=0; i<1024; i++) + Serial.println(*(pPwr + i), 8 ); + } + Serial.println(""); + } + +void loop(void) { + }