From ce4e8fb2920144782f5f78ed75a97081eaa7c775 Mon Sep 17 00:00:00 2001 From: boblark Date: Tue, 2 Mar 2021 15:37:43 -0800 Subject: [PATCH] add analyze_fft256_iq_F32 and examples/TestFFT256iq --- analyze_fft256_iq_F32.cpp | 261 +++++++++++++ analyze_fft256_iq_F32.h | 491 +++++++++++++++++++++++++ examples/TestFFT256iq/TestFFT256iq.ino | 51 +++ 3 files changed, 803 insertions(+) create mode 100644 analyze_fft256_iq_F32.cpp create mode 100644 analyze_fft256_iq_F32.h create mode 100644 examples/TestFFT256iq/TestFFT256iq.ino diff --git a/analyze_fft256_iq_F32.cpp b/analyze_fft256_iq_F32.cpp new file mode 100644 index 0000000..1789038 --- /dev/null +++ b/analyze_fft256_iq_F32.cpp @@ -0,0 +1,261 @@ +/* analyze_fft_iq_F32.cpp + * + * 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_fft256_iq_F32.h" + +// 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; + for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) { + *dst++ = *srcI++; // real sample, interleave + //*dst++ = 0.0f; + *dst++ = *srcQ++; // imag + //*dst++ = 0.0f; + } + } + +static void apply_window_to_fft_buffer1(void *fft_buffer, const void *window) { + float *buf = (float *)fft_buffer; // 0th entry is real (do window) 1th is imag + const float *win = (float *)window; + for (int i=0; i < 256; i++) { + buf[2*i] *= *win++; // real + buf[2*i + 1] *= *win++; // imag + } + } + +void AudioAnalyzeFFT256_IQ_F32::update(void) { + audio_block_f32_t *block_i,*block_q; + + 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 + + // prevblock_i and _q are pointers to the IQ data collected last update() + if (!prevblock_i || !prevblock_q) { // Startup + prevblock_i = block_i; + prevblock_q = block_q; + return; // Nothing to release + } + // FFT is 256 and blocks are 128, so we need 2 blocks. We still do + // this every 128 samples to get 50% overlap on FFT data to roughly + // compensate for windowing. + // ( dest, i-source, q-source ) + copy_to_fft_buffer1(fft_buffer, prevblock_i->data, prevblock_q->data); + copy_to_fft_buffer1(fft_buffer+256, block_i->data, block_q->data); + if (pWin) + apply_window_to_fft_buffer1(fft_buffer, window); + arm_cfft_radix4_f32(&fft_inst, fft_buffer); // Finally the FFT + + count++; + for (int i=0; i < 256; i++) { + float ss = fft_buffer[2*i]*fft_buffer[2*i] + fft_buffer[2*i+1]*fft_buffer[2*i+1]; + if(count==1) // Starting new average + sumsq[i] = ss; + else if (count <= nAverage) // Adding on to average + sumsq[i] += ss; + } + + if (count >= nAverage) { // Average is finished + count = 0; + float inAf = 1.0f/(float)nAverage; + for (int i=0; i < 256; i++) { + int ii = 255 - (i ^ 128); + if(outputType==FFT_RMS) + output[ii] = sqrtf(inAf*sumsq[ii]); + else if(outputType==FFT_POWER) + output[ii] = inAf*sumsq[ii]; + else if(outputType==FFT_DBFS) + output[ii] = 10.0f*log10f(inAf*sumsq[ii])-42.1442f; // Scaled to FS sine wave + else + output[ii] = 0.0f; + } + } + outputflag = true; + release(prevblock_i); // Release the 2 blocks that were block_i + release(prevblock_q); // and block_q on last time through update() + prevblock_i = block_i; // We will use these 2 blocks on next update() + prevblock_q = block_q; // Just change pointers +} + +#if 0 +============================================================== + +============================================================== +/* analyze_fft1024_F32.cpp Converted from Teensy I16 Audio Library + * This version uses float F32 inputs. See comments at analyze_fft1024_F32.h + * + * Conversion parts copyright (c) Bob Larkin 2021 + * + * 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_F32.h" +// #include "utility/dspinst.h" + +// Move audio data from an audio_block_f32_t to the FFT instance buffer. +static void copy_to_fft_buffer(void *destination, const void *source) +{ + const float *src = (const float *)source; + float *dst = (float *)destination; + + for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) { + *dst++ = *src++; // real sample + *dst++ = 0.0f; // 0 for Imag + } +} + +static void apply_window_to_fft_buffer(void *buffer, const void *window) +{ + float *buf = (float *)buffer; // 0th entry is real (do window) 1th is imag + const float *win = (float *)window; + + for (int i=0; i < 1024; i++) + buf[2*i] *= *win++; +} + +void AudioAnalyzeFFT1024_F32::update(void) +{ + audio_block_f32_t *block; + block = receiveReadOnly_f32(); + if (!block) return; + +// What all does 7EM cover?? +#if defined(__ARM_ARCH_7EM__) + switch (state) { + case 0: + blocklist[0] = block; + state = 1; + break; + case 1: + blocklist[1] = block; + state = 2; + break; + case 2: + blocklist[2] = block; + state = 3; + break; + case 3: + blocklist[3] = block; + state = 4; + break; + case 4: + blocklist[4] = block; + state = 5; + break; + case 5: + blocklist[5] = block; + state = 6; + break; + case 6: + blocklist[6] = block; + state = 7; + break; + case 7: + blocklist[7] = block; + copy_to_fft_buffer(fft_buffer+0x000, blocklist[0]->data); + copy_to_fft_buffer(fft_buffer+0x100, blocklist[1]->data); + copy_to_fft_buffer(fft_buffer+0x200, blocklist[2]->data); + copy_to_fft_buffer(fft_buffer+0x300, blocklist[3]->data); + copy_to_fft_buffer(fft_buffer+0x400, blocklist[4]->data); + copy_to_fft_buffer(fft_buffer+0x500, blocklist[5]->data); + copy_to_fft_buffer(fft_buffer+0x600, blocklist[6]->data); + copy_to_fft_buffer(fft_buffer+0x700, blocklist[7]->data); + + if (pWin) + apply_window_to_fft_buffer(fft_buffer, window); + + arm_cfft_radix4_f32(&fft_inst, fft_buffer); + + for (int i=0; i < 512; i++) { + float magsq = fft_buffer[2*i]*fft_buffer[2*i] + fft_buffer[2*i+1]*fft_buffer[2*i+1]; + if(outputType==FFT_RMS) + output[i] = sqrtf(magsq); + else if(outputType==FFT_POWER) + output[i] = magsq; + else if(outputType==FFT_DBFS) + output[i] = 10.0f*log10f(magsq)-54.1854f; // Scaled to FS sine wave + else + output[i] = 0.0f; + } + outputflag = true; + release(blocklist[0]); + release(blocklist[1]); + release(blocklist[2]); + release(blocklist[3]); + blocklist[0] = blocklist[4]; + blocklist[1] = blocklist[5]; + blocklist[2] = blocklist[6]; + blocklist[3] = blocklist[7]; + state = 4; + break; + } +#else + release(block); +#endif +} +#endif diff --git a/analyze_fft256_iq_F32.h b/analyze_fft256_iq_F32.h new file mode 100644 index 0000000..11916d0 --- /dev/null +++ b/analyze_fft256_iq_F32.h @@ -0,0 +1,491 @@ +/* analyze_fft_iq_F32.h + * + * 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. + */ + +/* Does complex input FFT of 1024 points. Output is not audio, and is magnitude + * only. Multiple 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 setOutputType(int _type) + * + * Timing, max is longest update() time: + * 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_fft256iq_h_ +#define analyze_fft256iq_h_ + +//#include "AudioStream.h" +//#include "arm_math.h" + +#include "Arduino.h" +#include "AudioStream_F32.h" +#include "arm_math.h" +#include "mathDSP_F32.h" + +#define FFT_RMS 0 +#define FFT_POWER 1 +#define FFT_DBFS 2 + +#define NO_WINDOW 0 +#define AudioWindowNone 0 +#define AudioWindowHanning256 1 +#define AudioWindowKaiser256 2 +#define AudioWindowBlackmanHarris256 3 + +class AudioAnalyzeFFT256_IQ_F32 : public AudioStream_F32 { +//GUI: inputs:2, outputs:4 //this line used for automatic generation of GUI node +//GUI: shortName:AnalyzeFFT256IQ +public: + AudioAnalyzeFFT256_IQ_F32() : AudioStream_F32(2, inputQueueArray) { // NEEDS SETTINGS etc <<<<<<<< + arm_cfft_radix4_init_f32(&fft_inst, 256, 0, 1); + useHanningWindow(); + } + + bool available() { + if (outputflag == true) { + outputflag = false; + return true; + } + return false; + } + + float read(unsigned int binNumber) { + if (binNumber>511 || 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 > 511) return 0.0; + if (binLast > 511) binLast = 511; + uint32_t sum = 0; + do { + sum += output[binFirst++]; + } while (binFirst <= binLast); + return (float)sum * (1.0 / 16384.0); + } + + int windowFunction(int wNum) { + if(wNum == AudioWindowKaiser256) + 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 == AudioWindowKaiser256) { + if(_kdb<20.0f) + kd = 20.0f; + else + kd = _kdb; + useKaiserWindow(kd); + } + else if (wNum == AudioWindowBlackmanHarris256) + 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) { + 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<256; i++) + *p++ = *pwin++; // Copy for the FFT + } + + // Output RMS (default) Power or dBFS + void setOutputType(int _type) { + outputType = _type; + } + + virtual void update(void); + +private: + float output[256]; + float window[256]; + float *pWin = window; + float fft_buffer[512]; + float sumsq[256]; // Avoid re-use of output[] + uint8_t state = 0; + bool outputflag = false; + audio_block_f32_t *inputQueueArray[2]; + audio_block_f32_t *prevblock_i,*prevblock_q; + arm_cfft_radix4_instance_f32 fft_inst; + int outputType = FFT_RMS; //Same type as I16 version init + int count = 0; + int nAverage = 1; + + // The Hann window is a good all-around window + void useHanningWindow(void) { + for (int i=0; i < 256; i++) { + // 2*PI/255 = 0.0246399424 + window[i] = 0.5*(1.0 - cosf(0.0246399424*(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 < 256; i++) { + float kx = 0.0246399424; // 2*PI/255 + 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<128; n++) { + xn2 = 0.5f+(float32_t)n; + // 4/(1023^2)=0.00000382215877f + // xn2 = 0.00000382215877f*xn2*xn2; + // 4/(255^2)=0.000061514802f + xn2 = 0.000061514802f*xn2*xn2; + window[127 - n]=kbes*(mathEqualizer.i0f(beta*sqrtf(1.0-xn2))); + window[128 + n] = window[255 - n]; + } + } + }; +#endif + + +#if 0 +//================================================== + +//==================================================== +/* analyze_fft1024_F32.h Converted from Teensy I16 Audio Library + * + * 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. + */ + +/* Moved directly I16 to F32. Bob Larkin 16 Feb 2021 + * Does real input FFT of 1024 points. Output is not audio, and is magnitude + * only. Multiple 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 setOutputType(int _type) + * + * Timing, max is longest update() time: + * T3.6 Windowed, RMS out, 1016 uSec max + * T3.6 Windowed, Power Out, 975 uSec max + * T3.6 Windowed, dBFS out, 1591 uSec max + * No Window saves 60 uSec on T3.6 for any output. + * T4.0 Windowed, RMS Out, 149 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_fft256iq_F32_h_ +#define analyze_fft256iq_F32_h_ + +#include "Arduino.h" +#include "AudioStream_F32.h" +#include "arm_math.h" +#include "mathDSP_F32.h" + +#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_F32 : public AudioStream_F32 { +//GUI: inputs:1, outputs:0 //this line used for automatic generation of GUI node +//GUI: shortName:AnalyzeFFT1024 +public: + AudioAnalyzeFFT1024_F32() : AudioStream_F32(1, inputQueueArray) { + arm_cfft_radix4_init_f32(&fft_inst, 1024, 0, 1); + useHanningWindow(); // Revisit this for more flexibility <<<<< + } + + bool available() { + if (outputflag == true) { + outputflag = false; + return true; + } + return false; + } + + float read(unsigned int binNumber) { + if (binNumber>511 || 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 > 511) return 0.0; + if (binLast > 511) binLast = 511; + uint32_t sum = 0; + do { + sum += output[binFirst++]; + } while (binFirst <= binLast); + return (float)sum * (1.0 / 16384.0); + } + + 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 512 data points occur. + float* getData(void) { + 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++; + } + + // Output RMS (default) Power or dBFS + void setOutputType(int _type) { + outputType = _type; + } + + virtual void update(void); + +private: + float output[512]; + float window[1024]; + float *pWin = window; + audio_block_f32_t *blocklist[8]; + float fft_buffer[2048]; + uint8_t state = 0; + bool outputflag = false; + audio_block_f32_t *inputQueueArray[1]; + arm_cfft_radix4_instance_f32 fft_inst; + int outputType = FFT_RMS; //Same type as I16 version init + + // 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 +#endif diff --git a/examples/TestFFT256iq/TestFFT256iq.ino b/examples/TestFFT256iq/TestFFT256iq.ino new file mode 100644 index 0000000..a7ffd83 --- /dev/null +++ b/examples/TestFFT256iq/TestFFT256iq.ino @@ -0,0 +1,51 @@ + +// TestFFT256iq.ino + +#include "OpenAudio_ArduinoLibrary.h" +#include "AudioStream_F32.h" +#include +#include +#include +#include +#include + +// GUItool: begin automatically generated code +AudioSynthSineCosine_F32 sine_cos1; //xy=76,532 +AudioAnalyzeFFT256_IQ_F32 FFT256iq1; //xy=243,532 +AudioOutputI2S_F32 audioOutI2S1; //xy=246,591 +AudioConnection_F32 patchCord1(sine_cos1, 0, FFT256iq1, 0); +AudioConnection_F32 patchCord2(sine_cos1, 1, FFT256iq1, 1); +//AudioControlSGTL5000 sgtl5000_1; +// GUItool: end automatically generated code + +void setup(void) { + float* pPwr; + + Serial.begin(9600); + delay(1000); + AudioMemory_F32(20); + Serial.println("FFT256IQ Test"); +// sgtl5000_1.enable(); //start the audio board + // sgtl5000_1.inputSelect(AUDIO_INPUT_LINEIN); // or AUDIO_INPUT_MIC + + sine_cos1.amplitude(0.5); // Initialize Waveform Generator + // bin spacing = 44117.648/256 = 172.335 172.3 * 4 = 689.335 Hz (T3.6) + // Half bin higher is 775.3 for testing windows + //sine_cos1.frequency(689.34f); + sine_cos1.frequency(1723.35f); + + FFT256iq1.setOutputType(FFT_DBFS); + FFT256iq1.windowFunction(AudioWindowBlackmanHarris256); + //float* pw = FFT256iq1.getWindow(); // Print window + //for (int i=0; i<256; i++) Serial.println(pw[i], 4); + + delay(1000); + if( FFT256iq1.available() ) + pPwr = FFT256iq1.getData(); + + for(int i=0; i<256; i++) + Serial.println(*(pPwr + i), 8 ); + } + +void loop(void) { + }