Fix xAxis cmd for dBFS & dBFS offset (+12dB)

analyze_fft256_iq_F32.cpp

@@ -132,13 +132,18 @@ void AudioAnalyzeFFT256_IQ_F32::update(void)  {
        output[i] = sqrtf(inAf*sumsq[ii]);
     else if(outputType==FFT_POWER)
        output[i] = inAf*sumsq[ii];
-    else if(outputType==FFT_DBFS)
+
-       output[i] = 10.0f*log10f(inAf*sumsq[ii])-42.1442f;  // Scaled to FS sine wave
+    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] = 0.0f;
+          output[i] = -193.0f;   // lsb for 23 bit mantissa
-    }
        }
-  outputflag = true;
+    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()

analyze_fft256_iq_F32.h

@@ -2,6 +2,7 @@
  *
  * Rev 6 Mar 2021 - Added setXAxis()
  * Rev 7 Mar 2021 - Corrected bug in applying windowing
+ * Rev 10 Mar 2021 - Corrrected: dBFS offset (up 12 dB) & xAxis for dBFS
  * 
  * Does Fast Fourier Transform of a 256 point complex (I-Q) input.
  * Output is one of three measures of the power in each of the 256
@@ -17,6 +18,7 @@
  *   * Windowing None, Hann, Kaiser and Blackman-Harris.
  *   * Multiple bin-sum output to simulate wider bins.
  *   * Power averaging of multiple FFT
+ *   * Programmable frequency scale arrangement.
  *   * Soon: F32 audio outputs for I & Q
  *
  * Conversion Copyright (c) 2021 Bob Larkin
@@ -76,11 +78,10 @@
  * If there is 180 degree phase shift to I or Q these all get reversed.
  *
  * Timing, max is longest update() time:
- *   T3.6 Windowed, RMS out,  - uSec max
+ *   T3.6 Windowed, Power Out, 285 uSec max
- *   T3.6 Windowed, Power Out, - uSec max
+ *   T3.6 Windowed, dBFS out, 590 uSec max
- *   T3.6 Windowed, dBFS out, - uSec max
+ *   T3.6 No Window saves 28 uSec for any output.
- *   No Window saves 60 uSec on T3.6 for any output.
+ *   T4.0 Windowed, dBFS Out, 120 uSec
- *   T4.0 Windowed, RMS Out,   - uSec
  *
  * Scaling:
  *   Full scale for floating point DSP is a nebulous concept.  Normally the

examples/TestFFT256iq/TestFFT256iq.ino

@@ -18,14 +18,13 @@ AudioConnection_F32       patchCord2(sine_cos1, 1, FFT256iq1, 1);
 // GUItool: end automatically generated code
 
 void setup(void) {
-  float* pPwr;
-
   Serial.begin(9600);
   delay(1000);
   AudioMemory_F32(20);
-  Serial.println("FFT256IQ Test   v2");
+  Serial.println("FFT256IQ Test   v3");
 
   sine_cos1.amplitude(1.0);         // 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);
@@ -39,7 +38,7 @@ void setup(void) {
   // or pick any old frequency
   sine_cos1.frequency(7100.0);
 
-  // elect the output format
+  // Select the output format:  FFT_RMS, FFFT_POWER, FFT_DBFS
   FFT256iq1.setOutputType(FFT_DBFS);
 
   // Select the wndow function
@@ -55,14 +54,12 @@ void setup(void) {
   // xAxis, bit 0 left/right;  bit 1 low to high;  default 0X03
   FFT256iq1.setXAxis(0X03);
 
-  FFT256iq1.windowFunction(AudioWindowBlackmanHarris256);
-  //float* pw = FFT256iq1.getWindow();   // Print window
-  //for (int i=0; i<256; i++) Serial.println(pw[i], 4);
-
   // Do power averaging (outputs appear less often, as well)
-  FFT256iq1.setNAverage(5);  // nAverage >= 1
+  FFT256iq1.setNAverage(1);  // i.e., 16 or 50, etc.  nAverage >= 1
+  }
 
-  delay(1000);
+void loop(void)  {
+  float* pPwr;
   if( FFT256iq1.available() )  {
      pPwr = FFT256iq1.getData();
      for(int i=0; i<256; i++) {
@@ -72,7 +69,5 @@ void setup(void) {
          }
       Serial.print("\n\n");
       }
-  }
+  delay(500);
-
-void loop(void)  {
   }