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OpenAudio_ArduinoLibrary/analyze_fft256_iq_F32.cpp

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/* 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 <Arduino.h>
#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 <Arduino.h>
#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