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/* analyze_fft_iq_F32.cpp
|
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* |
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* Converted to F32 floating point input and also extended |
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* for complex I and Q inputs |
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* * Adapted all I/O to be F32 floating point for OpenAudio_ArduinoLibrary |
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* * Future: Add outputs for I & Q FFT x2 for overlapped FFT |
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* * Windowing None, Hann, Kaiser and Blackman-Harris. |
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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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* 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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#include <Arduino.h> |
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#include "analyze_fft256_iq_F32.h" |
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// Move audio data from audio_block_f32_t to the interleaved FFT instance buffer.
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static void copy_to_fft_buffer1(void *destination, const void *sourceI, const void *sourceQ) { |
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const float *srcI = (const float *)sourceI; |
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const float *srcQ = (const float *)sourceQ; |
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float *dst = (float *)destination; |
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for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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*dst++ = *srcI++; // real sample, interleave
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//*dst++ = 0.0f;
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*dst++ = *srcQ++; // imag
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//*dst++ = 0.0f;
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} |
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} |
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static void apply_window_to_fft_buffer1(void *fft_buffer, const void *window) { |
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float *buf = (float *)fft_buffer; // 0th entry is real (do window) 1th is imag
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const float *win = (float *)window; |
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for (int i=0; i < 256; i++) { |
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buf[2*i] *= *win++; // real
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buf[2*i + 1] *= *win++; // imag
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} |
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} |
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void AudioAnalyzeFFT256_IQ_F32::update(void) { |
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audio_block_f32_t *block_i,*block_q; |
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block_i = receiveReadOnly_f32(0); |
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if (!block_i) return; |
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block_q = receiveReadOnly_f32(1); |
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if (!block_q) { |
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release(block_i); |
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return; |
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} |
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// Here with two new blocks of data
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// prevblock_i and _q are pointers to the IQ data collected last update()
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if (!prevblock_i || !prevblock_q) { // Startup
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prevblock_i = block_i; |
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prevblock_q = block_q; |
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return; // Nothing to release
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} |
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// FFT is 256 and blocks are 128, so we need 2 blocks. We still do
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// this every 128 samples to get 50% overlap on FFT data to roughly
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// compensate for windowing.
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// ( dest, i-source, q-source )
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copy_to_fft_buffer1(fft_buffer, prevblock_i->data, prevblock_q->data); |
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copy_to_fft_buffer1(fft_buffer+256, block_i->data, block_q->data); |
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if (pWin) |
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apply_window_to_fft_buffer1(fft_buffer, window); |
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arm_cfft_radix4_f32(&fft_inst, fft_buffer); // Finally the FFT
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count++; |
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for (int i=0; i < 256; i++) { |
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float ss = fft_buffer[2*i]*fft_buffer[2*i] + fft_buffer[2*i+1]*fft_buffer[2*i+1]; |
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if(count==1) // Starting new average
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sumsq[i] = ss; |
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else if (count <= nAverage) // Adding on to average
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sumsq[i] += ss; |
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} |
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if (count >= nAverage) { // Average is finished
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count = 0; |
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float inAf = 1.0f/(float)nAverage; |
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for (int i=0; i < 256; i++) { |
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int ii = 255 - (i ^ 128); |
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if(outputType==FFT_RMS) |
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output[ii] = sqrtf(inAf*sumsq[ii]); |
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else if(outputType==FFT_POWER) |
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output[ii] = inAf*sumsq[ii]; |
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else if(outputType==FFT_DBFS) |
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output[ii] = 10.0f*log10f(inAf*sumsq[ii])-42.1442f; // Scaled to FS sine wave
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else |
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output[ii] = 0.0f; |
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} |
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} |
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outputflag = true; |
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release(prevblock_i); // Release the 2 blocks that were block_i
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release(prevblock_q); // and block_q on last time through update()
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prevblock_i = block_i; // We will use these 2 blocks on next update()
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prevblock_q = block_q; // Just change pointers
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} |
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#if 0 |
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============================================================== |
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============================================================== |
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/* analyze_fft1024_F32.cpp Converted from Teensy I16 Audio Library
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* This version uses float F32 inputs. See comments at analyze_fft1024_F32.h |
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* |
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* Conversion parts copyright (c) Bob Larkin 2021 |
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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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#include <Arduino.h> |
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#include "analyze_fft1024_F32.h" |
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// #include "utility/dspinst.h"
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// Move audio data from an audio_block_f32_t to the FFT instance buffer.
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static void copy_to_fft_buffer(void *destination, const void *source) |
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{ |
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const float *src = (const float *)source; |
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float *dst = (float *)destination; |
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for (int i=0; i < AUDIO_BLOCK_SAMPLES; i++) { |
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*dst++ = *src++; // real sample
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*dst++ = 0.0f; // 0 for Imag
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} |
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} |
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static void apply_window_to_fft_buffer(void *buffer, const void *window) |
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{ |
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float *buf = (float *)buffer; // 0th entry is real (do window) 1th is imag
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const float *win = (float *)window; |
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for (int i=0; i < 1024; i++) |
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buf[2*i] *= *win++; |
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} |
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void AudioAnalyzeFFT1024_F32::update(void) |
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{ |
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audio_block_f32_t *block; |
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block = receiveReadOnly_f32(); |
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if (!block) return; |
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// What all does 7EM cover??
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#if defined(__ARM_ARCH_7EM__) |
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switch (state) { |
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case 0: |
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blocklist[0] = block; |
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state = 1; |
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break; |
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case 1: |
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blocklist[1] = block; |
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state = 2; |
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break; |
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case 2: |
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blocklist[2] = block; |
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state = 3; |
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break; |
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case 3: |
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blocklist[3] = block; |
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state = 4; |
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break; |
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case 4: |
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blocklist[4] = block; |
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state = 5; |
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break; |
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case 5: |
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blocklist[5] = block; |
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state = 6; |
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break; |
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case 6: |
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blocklist[6] = block; |
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state = 7; |
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break; |
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case 7: |
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blocklist[7] = block; |
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copy_to_fft_buffer(fft_buffer+0x000, blocklist[0]->data); |
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copy_to_fft_buffer(fft_buffer+0x100, blocklist[1]->data); |
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copy_to_fft_buffer(fft_buffer+0x200, blocklist[2]->data); |
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copy_to_fft_buffer(fft_buffer+0x300, blocklist[3]->data); |
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copy_to_fft_buffer(fft_buffer+0x400, blocklist[4]->data); |
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copy_to_fft_buffer(fft_buffer+0x500, blocklist[5]->data); |
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copy_to_fft_buffer(fft_buffer+0x600, blocklist[6]->data); |
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copy_to_fft_buffer(fft_buffer+0x700, blocklist[7]->data); |
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if (pWin) |
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apply_window_to_fft_buffer(fft_buffer, window); |
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arm_cfft_radix4_f32(&fft_inst, fft_buffer); |
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for (int i=0; i < 512; i++) { |
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float magsq = fft_buffer[2*i]*fft_buffer[2*i] + fft_buffer[2*i+1]*fft_buffer[2*i+1]; |
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if(outputType==FFT_RMS) |
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output[i] = sqrtf(magsq); |
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else if(outputType==FFT_POWER) |
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output[i] = magsq; |
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else if(outputType==FFT_DBFS) |
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output[i] = 10.0f*log10f(magsq)-54.1854f; // Scaled to FS sine wave
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else |
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output[i] = 0.0f; |
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} |
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outputflag = true; |
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release(blocklist[0]); |
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release(blocklist[1]); |
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release(blocklist[2]); |
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release(blocklist[3]); |
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blocklist[0] = blocklist[4]; |
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blocklist[1] = blocklist[5]; |
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blocklist[2] = blocklist[6]; |
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blocklist[3] = blocklist[7]; |
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state = 4; |
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break; |
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} |
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#else |
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release(block); |
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#endif |
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} |
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#endif |
@ -0,0 +1,491 @@ |
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/* analyze_fft_iq_F32.h
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* |
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* Converted to F32 floating point input and also extended |
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* for complex I and Q inputs |
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* * Adapted all I/O to be F32 floating point for OpenAudio_ArduinoLibrary |
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* * Future: Add outputs for I & Q FFT x2 for overlapped FFT |
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* * Windowing None, Hann, Kaiser and Blackman-Harris. |
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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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* |
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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 |
||||
* 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. 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, - 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_fft256iq_h_ |
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#define analyze_fft256iq_h_ |
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//#include "AudioStream.h"
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//#include "arm_math.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 AudioWindowHanning256 1 |
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#define AudioWindowKaiser256 2 |
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#define AudioWindowBlackmanHarris256 3 |
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class AudioAnalyzeFFT256_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:AnalyzeFFT256IQ
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public: |
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AudioAnalyzeFFT256_IQ_F32() : AudioStream_F32(2, inputQueueArray) { // NEEDS SETTINGS etc <<<<<<<<
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arm_cfft_radix4_init_f32(&fft_inst, 256, 0, 1); |
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useHanningWindow(); |
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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 == AudioWindowKaiser256) |
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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 == AudioWindowKaiser256) { |
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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 == AudioWindowBlackmanHarris256) |
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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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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<256; i++) |
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*p++ = *pwin++; // Copy for the FFT
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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[256]; |
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float window[256]; |
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float *pWin = window; |
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float fft_buffer[512]; |
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float sumsq[256]; // 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 *prevblock_i,*prevblock_q; |
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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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int count = 0; |
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int nAverage = 1; |
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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 < 256; i++) { |
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// 2*PI/255 = 0.0246399424
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window[i] = 0.5*(1.0 - cosf(0.0246399424*(float)i)); |
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} |
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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 < 256; i++) { |
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float kx = 0.0246399424; // 2*PI/255
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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
|
||||
* 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 |
@ -0,0 +1,51 @@ |
||||
|
||||
// TestFFT256iq.ino
|
||||
|
||||
#include "OpenAudio_ArduinoLibrary.h" |
||||
#include "AudioStream_F32.h" |
||||
#include <Audio.h> |
||||
#include <Wire.h> |
||||
#include <SPI.h> |
||||
#include <SD.h> |
||||
#include <SerialFlash.h> |
||||
|
||||
// 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) { |
||||
} |
Loading…
Reference in new issue