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@ -248,244 +248,3 @@ private: |
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} |
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}; |
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#endif |
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#if 0 |
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//==================================================
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//====================================================
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/* analyze_fft1024_F32.h Converted from Teensy I16 Audio Library
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* |
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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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/* Moved directly I16 to F32. Bob Larkin 16 Feb 2021
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* Does real input FFT of 1024 points. Output is not audio, and is magnitude |
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* only. Multiple 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 setOutputType(int _type) |
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* |
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* Timing, max is longest update() time: |
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* T3.6 Windowed, RMS out, 1016 uSec max |
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* T3.6 Windowed, Power Out, 975 uSec max |
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* T3.6 Windowed, dBFS out, 1591 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, 149 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_fft256iq_F32_h_ |
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#define analyze_fft256iq_F32_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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#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_F32 : public AudioStream_F32 { |
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//GUI: inputs:1, outputs:0 //this line used for automatic generation of GUI node
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//GUI: shortName:AnalyzeFFT1024
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public: |
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AudioAnalyzeFFT1024_F32() : AudioStream_F32(1, inputQueueArray) { |
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arm_cfft_radix4_init_f32(&fft_inst, 1024, 0, 1); |
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useHanningWindow(); // Revisit this for more flexibility <<<<<
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} |
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bool available() { |
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if (outputflag == true) { |
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outputflag = false; |
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return true; |
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} |
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return false; |
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} |
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float read(unsigned int binNumber) { |
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if (binNumber>511 || 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 > 511) return 0.0; |
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if (binLast > 511) binLast = 511; |
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uint32_t 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 (float)sum * (1.0 / 16384.0); |
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} |
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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 512 data points occur.
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float* getData(void) { |
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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++; |
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} |
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// Output RMS (default) Power or dBFS
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void setOutputType(int _type) { |
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outputType = _type; |
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} |
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virtual void update(void); |
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private: |
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float output[512]; |
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float window[1024]; |
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float *pWin = window; |
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audio_block_f32_t *blocklist[8]; |
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float fft_buffer[2048]; |
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uint8_t state = 0; |
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bool outputflag = false; |
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audio_block_f32_t *inputQueueArray[1]; |
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arm_cfft_radix4_instance_f32 fft_inst; |
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int outputType = FFT_RMS; //Same type as I16 version init
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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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#endif |
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boblark
4 years ago