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OpenAudio_ArduinoLibrary/analyze_fft256_iq_F32.cpp

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/* analyze_fft256_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.
* See analyze_fft256_iq_F32. for more info.
*
* 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_buffer0(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++ = *srcQ++; // imag
}
}
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;
int ii;
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_buffer0(fft_buffer, prevblock_i->data, prevblock_q->data);
copy_to_fft_buffer0(fft_buffer+256, block_i->data, block_q->data);
if (pWin)
apply_window_to_fft_buffer1(fft_buffer, window);
#if defined(__IMXRT1062__)
// Teensyduino core for T4.x supports arm_cfft_f32
// arm_cfft_f32 (const arm_cfft_instance_f32 *S, float32_t *p1, uint8_t ifftFlag, uint8_t bitReverseFlag)
arm_cfft_f32(&Sfft, fft_buffer, 0, 1);
#else
// For T3.x go back to old (deprecated) style
arm_cfft_radix4_f32(&fft_inst, fft_buffer);
#endif
count++;
for (int i = 0; i < 128; i++) {
// From complex FFT the "negative frequencies" are mirrors of the frequencies above fs/2. So, we get
// frequencies from 0 to fs by re-arranging the coefficients. These are powers (not Volts)
// See DD4WH SDR (Note - here and at "if(xAxis & xxxx)" below, we may have redundancy in index changing.
// Leave as is for now.)
float ss0 = fft_buffer[2 * i] * fft_buffer[2 * i] +
fft_buffer[2 * i + 1] * fft_buffer[2 * i + 1];
float ss1 = fft_buffer[2 * (i + 128)] * fft_buffer[2 * (i + 128)] +
fft_buffer[2 * (i + 128) + 1] * fft_buffer[2 * (i + 128) + 1];
if(count==1) { // Starting new average
sumsq[i+128] = ss0;
sumsq[i] = ss1;
}
else if (count <= nAverage) { // Adding on to average
sumsq[i+128] += ss0;
sumsq[i] += ss1;
}
}
if (count >= nAverage) { // Average is finished
count = 0;
float inAf = 1.0f/(float)nAverage;
for (int i=0; i < 256; i++) {
// xAxis, bit 0 left/right; bit 1 low to high
if(xAxis & 0X02)
ii = i;
else
ii = i^128;
if(xAxis & 0X01)
ii = (255 - ii);
if(outputType==FFT_RMS)
output[i] = sqrtf(inAf*sumsq[ii]);
else if(outputType==FFT_POWER)
output[i] = inAf*sumsq[ii];
else if(outputType==FFT_DBFS) {
if(sumsq[i]>0.0f)
output[i] = 10.0f*log10f(inAf*sumsq[ii]) - 42.144f; // Scaled to FS sine wave
else
output[i] = -193.0f; // lsb for 23 bit mantissa
}
else
output[i] = 0.0f;
} // End, set output[i] over all 512
outputflag = true; // moved; rev10mar2021
} // End of average is finished
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
}