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Update real input 1024 fft

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pull/6/merge
boblark 4 years ago
parent 7b8cda0bde
commit bc7ad85336
3 changed files with 344 additions and 110 deletions
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  1. 132
      analyze_fft1024_F32.cpp
  2. 100
      analyze_fft1024_F32.h
  3. 222
      examples/TestFFT1024/TestFFT1024.ino

132
analyze_fft1024_F32.cpp
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@ -1,5 +1,6 @@
/* analyze_fft1024_F32.cpp Converted from Teensy I16 Audio Library
* This version uses float F32 inputs. See comments at analyze_fft1024_F32.h
* Converted to use half-length FFT 17 March 2021 RSL
*
* Conversion parts copyright (c) Bob Larkin 2021
*
@ -33,36 +34,32 @@
#include "analyze_fft1024_F32.h"
// Move audio data from an audio_block_f32_t to the FFT instance buffer.
// This is for 128 numbers per block
static void copy_to_fft_buffer(void *destination, const void *source)
{
// This is for 128 numbers per block, only.
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 < 128; i++) {
*dst++ = *src++; // real sample
*dst++ = 0.0f; // 0 for Imag
*dst++ = *src++; // real sample for half-length FFT
}
}
}
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
static void apply_window_to_fft_buffer(void *buffer, const void *window) {
float *buf = (float *)buffer;
const float *win = (float *)window;
for (int i=0; i < 1024; i++)
buf[2*i] *= *win++;
}
for(int i=0; i<NFFT; i++)
buf[i] *= *win++;
}
void AudioAnalyzeFFT1024_F32::update(void) {
audio_block_f32_t *block;
uint32_t tt;
float outDC=0.0f;
float magsq=0.0f;
block = AudioStream_F32::receiveReadOnly_f32();
if (!block) return;
// tt=micros();
switch (state) {
case 0:
blocklist[0] = block;
@ -82,6 +79,55 @@ void AudioAnalyzeFFT1024_F32::update(void) {
break;
case 4:
blocklist[4] = block;
// Now the post FT processing for using half-length transform
// FFT was in state==7, but it loops around to here. Does some
// zero data at startup that is harmless.
count++; // Next do non-coherent averaging
for(int i=0; i<NFFT_D2; i++) {
if(i>0) {
float rns = 0.5f*(fft_buffer[2*i] + fft_buffer[NFFT-2*i]);
float ins = 0.5f*(fft_buffer[2*i+1] + fft_buffer[NFFT-2*i+1]);
float rnd = 0.5f*(fft_buffer[2*i] - fft_buffer[NFFT-2*i]);
float ind = 0.5f*(fft_buffer[2*i+1] - fft_buffer[NFFT-2*i+1]);
float xr = rns + cosN[i]*ins - sinN[i]*rnd;
float xi = ind - sinN[i]*ins - cosN[i]*rnd;
magsq = xr*xr + xi*xi;
}
else {
magsq = outDC*outDC; // Do the DC term
}
if(count==1) // Starting new average
sumsq[i] = magsq;
else if (count<=nAverage) // Adding on to average
sumsq[i] += magsq;
}
if (count >= nAverage) { // Average is finished
// Set outputflag false here to minimize reads of output[] data
// when it is being updated.
outputflag = false;
count = 0;
float inAf = 1.0f/(float)nAverage;
float kMaxDB = 20.0*log10f((float)NFFT_D2); // 54.1854 for 1024
for(int i=0; i<NFFT_D2; i++) {
if(outputType==FFT_RMS)
output[i] = sqrtf(inAf*sumsq[i]);
else if(outputType==FFT_POWER)
output[i] = inAf*sumsq[i];
else if(outputType==FFT_DBFS) {
if(sumsq[i]>0.0f)
output[i] = 10.0f*log10f(inAf*sumsq[i]) - kMaxDB; // 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 NFFT_D2
outputflag = true;
} // End of average is finished
state = 5;
break;
case 5:
@ -96,55 +142,32 @@ void AudioAnalyzeFFT1024_F32::update(void) {
blocklist[7] = block;
// We have 4 previous blocks pointed to by blocklist[]:
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+0x080, blocklist[1]->data);
copy_to_fft_buffer(fft_buffer+0x100, blocklist[2]->data);
copy_to_fft_buffer(fft_buffer+0x180, blocklist[3]->data);
// and 4 new blocks, just gathered:
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);
copy_to_fft_buffer(fft_buffer+0x200, blocklist[4]->data);
copy_to_fft_buffer(fft_buffer+0x280, blocklist[5]->data);
copy_to_fft_buffer(fft_buffer+0x300, blocklist[6]->data);
copy_to_fft_buffer(fft_buffer+0x380, blocklist[7]->data);
if (pWin)
apply_window_to_fft_buffer(fft_buffer, window);
outDC = 0.0f;
for(int i=0; i<NFFT; i++)
outDC += fft_buffer[i];
outDC /= ((float)NFFT);
#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);
// For T3.x go back to old (deprecated) style (check radix2/radix4)<<<
arm_cfft_radix2_f32(&fft_inst, fft_buffer);
#endif
count++; // Next do non-coherent averaging
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(count==1) // Starting new average
sumsq[i] = magsq;
else if (count<=nAverage) // Adding on to average
sumsq[i] += magsq;
}
if (count >= nAverage) { // Average is finished
count = 0;
float inAf = 1.0f/(float)nAverage;
for(int i=0; i<512; i++) {
if(outputType==FFT_RMS)
output[i] = sqrtf(inAf*sumsq[i]);
else if(outputType==FFT_POWER)
output[i] = inAf*sumsq[i];
else if(outputType==FFT_DBFS) {
if(sumsq[i]>0.0f)
output[i] = 10.0f*log10f(inAf*sumsq[i])-54.1854f; // 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;
} // End of average is finished
// FFT output is now in fft_buffer. Pick up processing at state==4.
AudioStream_F32::release(blocklist[0]);
AudioStream_F32::release(blocklist[1]);
@ -158,5 +181,4 @@ void AudioAnalyzeFFT1024_F32::update(void) {
state = 4;
break;
} // End switch(state)
// Serial.print("uSec: "); Serial.println(micros()-tt);
} // End update()

100
analyze_fft1024_F32.h
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@ -26,7 +26,7 @@
* THE SOFTWARE.
*/
/* Moved directly I16 to F32. Bob Larkin 16 Feb 2021
/* Translated from 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
@ -45,13 +45,13 @@
* void setOutputType(int _type)
* void setNAverage(int nAverage)
*
* Timing, max is longest update() time:
* T3.6 Windowed, Power Out, 975 uSec
* T3.6 Windowed, RMS out, 1016 uSec
* T3.6 Windowed, dBFS out, 1591 uSec
* Timing, max is longest update() time. Comparison is using full complex FFT
* and no load sharing on "states".
* T3.6 Windowed, Power Out, 682 uSec (was 975 w/ 1024 FFT)
* T3.6 Windowed, dBFS out, 834 uSec (was 1591 w/1024 FFT)
* No Window saves 60 uSec on T3.6 for any output.
* T4.0 Windowed, Ave=1, Power Out, 156 uSec
* T4.0 Windowed, Ave=1, dBFS Out, 302 uSec
* T4.0 Windowed, Power Out, 54 uSec (was 156 w/1024 FFT)
* T4.0 Windowed, dBFS Out, 203 uSec (was 302 w/1024 FFT)
* 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
@ -64,6 +64,9 @@
* when building the INO.
*/
// Fixed float/int problem in read(first, last). RSL 3 Mar 21
// Converted to using half-size FFT for real input, with no zero inputs.
// See E. Oran Brigham and many other FFT references. 16 March 2021 RSL
// Moved post-FFT calculations to state 4 to load share. RSL 18 Mar 2021
#ifndef analyze_fft1024_F32_h_
#define analyze_fft1024_F32_h_
@ -76,6 +79,15 @@
#include "arm_const_structs.h"
#endif
// Doing an FFT with NFFT real inputs
#define NFFT 1024
#define NFFT_M1 NFFT-1
#define NFFT_D2 NFFT/2
#define NFFT_D2M1 (NFFT/2)-1
#define NFFT_X2 NFFT*2
#define FFT_PI 3.14159265359f
#define FFT_2PI 6.28318530718f
#define FFT_RMS 0
#define FFT_POWER 1
#define FFT_DBFS 2
@ -96,13 +108,17 @@ public:
#if defined(__IMXRT1062__)
// Teensy4 core library has the right files for new FFT
// arm CMSIS library has predefined structures of type arm_cfft_instance_f32
Sfft = arm_cfft_sR_f32_len1024; // This is one of the structures
Sfft = arm_cfft_sR_f32_len512; // Like this. Changes with size <<<
#else
arm_cfft_radix4_init_f32(&fft_inst, 1024, 0, 1); // for T3.x
arm_cfft_radix2_init_f32(&fft_inst, NFFT_D2, 0, 1); // for T3.x (check radix2/radix4)<<<
#endif
// This is always 128 block size. Any sample rate. No use of "setings"
// Bob: Revisit this to use CMSIS fast fft in place of cfft. faster?
// This class is always 128 block size. Any sample rate. No use of "settings"
useHanningWindow();
// Factors for using half size complex FFT
for(int n=0; n<NFFT_D2; n++) {
sinN[n] = sinf(FFT_PI*((float)n)/((float)NFFT_D2));
cosN[n] = cosf(FFT_PI*((float)n)/((float)NFFT_D2));
}
}
// Inform that the output is available for read()
@ -116,7 +132,7 @@ public:
// Output data from a single bin is transferred
float read(unsigned int binNumber) {
if (binNumber>511 || binNumber<0) return 0.0;
if (binNumber>NFFT_D2M1 || binNumber<0) return 0.0;
return output[binNumber];
}
@ -128,8 +144,8 @@ public:
binLast = binFirst;
binFirst = tmp;
}
if (binFirst > 511) return 0.0;
if (binLast > 511) binLast = 511;
if (binFirst > NFFT_D2M1) return 0.0;
if (binLast > NFFT_D2M1) binLast = NFFT_D2M1;
float sum = 0.0f;
do {
sum += output[binFirst++];
@ -138,7 +154,7 @@ public:
}
int windowFunction(int wNum) {
if(wNum == AudioWindowKaiser1024)
if(wNum == AudioWindowKaiser1024) // Changes with size <<<
return -1; // Kaiser needs the kdb
windowFunction(wNum, 0.0f);
return 0;
@ -149,14 +165,14 @@ public:
pWin = window;
if(wNum == NO_WINDOW)
pWin = NULL;
else if (wNum == AudioWindowKaiser1024) {
else if (wNum == AudioWindowKaiser1024) { // Changes with size <<<
if(_kdb<20.0f)
kd = 20.0f;
else
kd = _kdb;
useKaiserWindow(kd);
}
else if (wNum == AudioWindowBlackmanHarris1024)
else if (wNum == AudioWindowBlackmanHarris1024) // Changes with size <<<
useBHWindow();
else
useHanningWindow(); // Default
@ -177,7 +193,7 @@ public:
// Bring custom window from the INO
void putWindow(float *pwin) {
float *p = window;
for(int i=0; i<1024; i++)
for(int i=0; i<NFFT; i++)
*p++ = *pwin++;
}
@ -194,21 +210,27 @@ public:
virtual void update(void);
private:
float output[512];
float sumsq[512];
float window[1024];
float output[NFFT_D2];
float sumsq[NFFT_D2]; // Accumulates averages of outputs
float window[NFFT];
float *pWin = window;
float fft_buffer[NFFT];
// The cosN and sinN would seem to be twidddle factors. Someday
// look at this and see if they can be stolen from arm math/DSP.
float cosN[NFFT_D2];
float sinN[NFFT_D2];
audio_block_f32_t *blocklist[8];
float fft_buffer[2048];
uint8_t state = 0;
bool outputflag = false;
audio_block_f32_t *inputQueueArray[1];
#if defined(__IMXRT1062__)
// For T4.x
// const static arm_cfft_instance_f32 arm_cfft_sR_f32_len1024;
// For T4.x, 512 length for real 1024 input, etc.
// const static arm_cfft_instance_f32 arm_cfft_sR_f32_len512;
arm_cfft_instance_f32 Sfft;
#else
arm_cfft_radix4_instance_f32 fft_inst;
arm_cfft_radix2_instance_f32 fft_inst; // Check radix2/radix4 <<<
#endif
int outputType = FFT_RMS; //Same type as I16 version init
int nAverage = 1;
@ -216,22 +238,22 @@ private:
// 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));
for (int i=0; i < NFFT; i++) {
float kx = FFT_2PI/((float)NFFT_M1); // 0.006141921 for 1024
window[i] = 0.5f*(1.0f - cosf(kx*(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
for (int i=0; i < NFFT; i++) {
float kx = FFT_2PI/((float)NFFT_M1); // 0.006141921 for 1024
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);
window[i] = 0.35875f -
0.48829f*cosf( kx*ix) +
0.14128f*cosf(2.0f*kx*ix) -
0.01168f*cosf(3.0f*kx*ix);
}
}
@ -252,12 +274,12 @@ private:
// 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++) {
for (int n=0; n<NFFT_D2; 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];
float kx = 4.0f/(((float)NFFT_M1) * ((float)NFFT_M1)); // 0.00000382215877f for 1024
xn2 = kx*xn2*xn2;
window[NFFT_D2M1 - n]=kbes*(mathEqualizer.i0f(beta*sqrtf(1.0-xn2)));
window[NFFT_D2 + n] = window[NFFT_D2M1 - n];
}
}
};

222
examples/TestFFT1024/TestFFT1024.ino
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@ -1,3 +1,193 @@
#if 0
#include "Arduino.h"
#include "AudioStream_F32.h"
#include "arm_math.h"
#include "mathDSP_F32.h"
#if defined(__IMXRT1062__)
#include "arm_const_structs.h"
#endif
#define NFFT 1024
#define NFFT_D2 NFFT/2
#define FFT_PI 3.14159265359f
#define PI2 2.0f*FFT_PI
void setup(void) {
float x[NFFT]; // Real DFT input
float Xre[NFFT], Xim[NFFT]; // DFT of x
float P[NFFT]; // power spectrum of x
float kf, nf;
float fft_buffer[2*NFFT]; // 2 is fo CMSIS FFT
float sinN[NFFT_D2];
float cosN[NFFT_D2];
uint32_t tt;
// Instantiate FFT, T4.x ONLY
arm_cfft_instance_f32 Sfft;
Sfft = arm_cfft_sR_f32_len1024;
// Instantiate FFT, T4.x ONLY
arm_cfft_instance_f32 Sfft_128;
Sfft_128 = arm_cfft_sR_f32_len512;
Serial.begin(300); // Any speed works
delay(1000);
// Factors for using half size complex FFT
for(int n=0; n<NFFT_D2; n++) {
sinN[n] = sinf(FFT_PI*((float)n)/((float)NFFT_D2));
cosN[n] = cosf(FFT_PI*((float)n)/((float)NFFT_D2));
}
// The signal, a sine wave that fits integer number (2 here) of times
Serial.println("The input waveform, NFFT points");
for (int n=0; n<NFFT; n++) {
x[n] = sinf(5.0f*FFT_PI*((float)n)/((float)NFFT_D2));
Serial.println(x[n], 8);
}
Serial.println();
Serial.println("The DFT by full NxN DFT, k, Real, Imag, Power");
// Calculate DFT of x[n] with NFFT point DFT
for (int k=0 ; k<NFFT ; ++k) {
kf = (float)k;
// Real part of X[k]
Xre[k] = 0.0f;
Xim[k] = 0.0f;
for (int n=0 ; n<NFFT ; ++n) {
nf = (float)n;
Xre[k] += x[n] * cos(nf * kf * PI2 / ((float)NFFT));
Xim[k] -= x[n] * sin(nf * kf * PI2 / ((float)NFFT));
}
// Power at kth frequency bin
P[k] = 10.0f*log10f(Xre[k]*Xre[k] + Xim[k]*Xim[k]);
Serial.print(k); Serial.print(",");
Serial.print(Xre[k],8); Serial.print(",");
Serial.print(Xim[k],8); Serial.print(",");
Serial.println(P[k],3);
}
Serial.println("And now the same thing with FFT size NFFT/2");
for (int k = 0; k < NFFT_D2; k++) { // For each output element
kf = (float)k;
float sumreal = 0;
float sumimag = 0;
for (int n = 0; n < NFFT_D2; n++) { // For each input element
nf = (float)n;
float angle = PI2 * nf * kf / ((float)NFFT_D2);
sumreal += x[2*n] * cos(angle) + x[2*n+1] * sin(angle);
sumimag += -x[2*n] * sin(angle) + x[2*n+1] * cos(angle);
}
Xre[k] = sumreal;
Xim[k] = sumimag;
}
// Rearrange the outputs to look like the FFT
for(int k=0; k<NFFT_D2; k++) {
fft_buffer[2*k] = Xre[k];
fft_buffer[2*k+1] = Xim[k];
}
// Now the post FT processing for using half-length transform
Xre[0] = 0.0f;
for(int n=0; n<NFFT; n++)
Xre[0] += x[n]/((float)NFFT); // DC real
Xim[0] = 0.0f; // DC Imag
P[0] = 10.0f*log10f(Xre[0]*Xre[0]);
// And the non-DC values
for(int i=1; i<NFFT_D2; i++) {
float rns = 0.5f*(fft_buffer[2*i] + fft_buffer[NFFT-2*i]);
float ins = 0.5f*(fft_buffer[2*i+1] + fft_buffer[NFFT-2*i+1]);
float rnd = 0.5f*(fft_buffer[2*i] - fft_buffer[NFFT-2*i]);
float ind = 0.5f*(fft_buffer[2*i+1] - fft_buffer[NFFT-2*i+1]);
Xre[i] = rns + cosN[i]*ins - sinN[i]*rnd;
Xim[i] = ind - sinN[i]*ins - cosN[i]*rnd;
P[i] = 10.0f*log10f(Xre[i]*Xre[i] + Xim[i]*Xim[i]);
}
for(int k=0; k<NFFT_D2; k++) {
Serial.print(k); Serial.print(",");
Serial.print(Xre[k],8); Serial.print(",");
Serial.print(Xim[k],8); Serial.print(",");
Serial.println(P[k],3);
}
Serial.println();
// And the same data with the CMSIS FFT
Serial.println("And now the same thing with CMSIS FFT size NFFT, 0.0 input for imag");
// 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)
for(int k=0; k<NFFT; k++) {
fft_buffer[2*k] = x[k];
fft_buffer[2*k+1] = 0.0f;
}
tt = micros();
arm_cfft_f32 (&Sfft, fft_buffer, 0, 1);
Serial.print("NFFT length FFT uSec = ");
Serial.println(micros()-tt);
for (int i=0; i < NFFT; i++) {
float magsq = fft_buffer[2*i]*fft_buffer[2*i] + fft_buffer[2*i+1]*fft_buffer[2*i+1];
Serial.print(i); Serial.print(",");
// Serial.print(Xre[k],8); Serial.print(",");
// Serial.print(Xim[k],8); Serial.print(",");
Serial.println(10.0f*log10f(magsq), 3);
}
// And the same data with the CMSIS FFT of size NFFT/2 and interleaved Trig sorting
// Time: for NFFT=256, this method is 18 uSec, down from 23 uSec using 256 length
// and 0's in the imag inputs. Memory use is about half and accuracy is the same.
Serial.println("And now with CMSIS FFT size NFFT/2, 0.0 input for imag");
// 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)
for(int k=0; k<NFFT; k++) {
fft_buffer[k] = x[k];
}
arm_cfft_f32 (&Sfft_128, fft_buffer, 0, 1);
// Now the post FT processing for using half-length transform
Xre[0] = 0.0f;
for(int n=0; n<NFFT; n++)
Xre[0] += x[n]/((float)NFFT); // DC real
Xim[0] = 0.0f; // DC Imag
P[0] = 10.0f*log10f(Xre[0]*Xre[0]);
// And the non-DC values
for(int i=1; i<NFFT_D2; i++) {
float rns = 0.5f*(fft_buffer[2*i] + fft_buffer[NFFT-2*i]);
float ins = 0.5f*(fft_buffer[2*i+1] + fft_buffer[NFFT-2*i+1]);
float rnd = 0.5f*(fft_buffer[2*i] - fft_buffer[NFFT-2*i]);
float ind = 0.5f*(fft_buffer[2*i+1] - fft_buffer[NFFT-2*i+1]);
Xre[i] = rns + cosN[i]*ins - sinN[i]*rnd;
Xim[i] = ind - sinN[i]*ins - cosN[i]*rnd;
P[i] = 10.0f*log10f(Xre[i]*Xre[i] + Xim[i]*Xim[i]);
}
for(int k=0; k<NFFT_D2; k++) {
Serial.print(k); Serial.print(",");
Serial.print(Xre[k],8); Serial.print(",");
Serial.print(Xim[k],8); Serial.print(",");
Serial.println(P[k],3);
}
Serial.println();
}
void loop(void) {
}
#endif
// *****************************************************************
// *****************************************************************
// TestFFT1024.ino Bob Larkin W7PUA
// Started from PJRC Teensy Examples/Audio/Analysis/FFT
//
@ -16,9 +206,6 @@
#include <Audio.h>
#include "OpenAudio_ArduinoLibrary.h"
const int myInput = AUDIO_INPUT_LINEIN;
//const int myInput = AUDIO_INPUT_MIC;
// Create the Audio components. These should be created in the
// order data flows, inputs/sources -> processing -> outputs
//
@ -30,7 +217,7 @@ AudioOutputI2S_F32 audioOutput; // audio shield: headphones & line-out N
// AudioConnection_F32 patchCord1(audioInput, 0, myFFT, 0);
AudioConnection_F32 patchCord1(sinewave, 0, myFFT, 0);
AudioControlSGTL5000 audioShield;
float xxx[1024];
uint32_t ct = 0;
uint32_t count = 0;
@ -42,15 +229,15 @@ void setup() {
// Enable the audio shield and set the output volume.
audioShield.enable();
audioShield.inputSelect(myInput);
audioShield.volume(0.5);
audioShield.inputSelect(AUDIO_INPUT_LINEIN);
// Create a synthetic sine wave, for testing
// To use this, edit the connections above
// sinewave.frequency(1033.99f); // Bin 24 T3.x
// sinewave.frequency(1033.59375f); // Bin 24 T4.x at 44100
// sinewave.frequency(1055.0f); // Bin 24.5, demonstrates windowing
sinewave.frequency(1076.0f);
// Or some random frequency:
sinewave.frequency(1234.5f);
sinewave.amplitude(1.0f);
@ -73,16 +260,19 @@ void setup() {
void loop() {
if (myFFT.available() /*&& ++ct == 4*/ ) {
// each time new FFT data is available
// print it all to the Arduino Serial Monitor
Serial.println("FFT Output: ");
for (int i=0; i<512; i++) {
// each time new FFT data is available
// print it all to the Arduino Serial Monitor
float* pin = myFFT.getData();
for (int gg=0; gg<512; gg++)
xxx[gg]= *(pin + gg);
for (int i=0; i<512; i++) {
Serial.print(i);
Serial.print(",");
Serial.println(myFFT.read(i), 3);
Serial.print(", ");
Serial.println(xxx[i], 8);
}
Serial.println();
}
Serial.println();
}
/*
if(count++<200) {
@ -92,5 +282,5 @@ void loop() {
Serial.println(AudioMemoryUsageMax_F32());
}
*/
delay(500);
}

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