From f4f0760d6131d2e1ab830da8f0be005da0372455 Mon Sep 17 00:00:00 2001 From: boblark Date: Tue, 2 Mar 2021 20:23:05 -0800 Subject: [PATCH] removed unrelated code notes --- analyze_fft256_iq_F32.cpp | 144 +---------------------- analyze_fft256_iq_F32.h | 241 -------------------------------------- 2 files changed, 1 insertion(+), 384 deletions(-) diff --git a/analyze_fft256_iq_F32.cpp b/analyze_fft256_iq_F32.cpp index 1789038..16d10ff 100644 --- a/analyze_fft256_iq_F32.cpp +++ b/analyze_fft256_iq_F32.cpp @@ -1,4 +1,4 @@ -/* analyze_fft_iq_F32.cpp +/* analyze_fft256_iq_F32.cpp * * Converted to F32 floating point input and also extended * for complex I and Q inputs @@ -117,145 +117,3 @@ void AudioAnalyzeFFT256_IQ_F32::update(void) { 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 index 11916d0..b5a0574 100644 --- a/analyze_fft256_iq_F32.h +++ b/analyze_fft256_iq_F32.h @@ -248,244 +248,3 @@ private: } }; #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