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Added pureSpectrum filtering option.

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pull/13/head
boblark 3 years ago
parent 16aaab43d3
commit 59e11fd79f
3 changed files with 101 additions and 31 deletions
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  1. 38
      examples/TestFFT4096iq/TestFFT4096iq.ino
  2. 31
      synth_sin_cos_f32.cpp
  3. 63
      synth_sin_cos_f32.h

38
examples/TestFFT4096iq/TestFFT4096iq.ino
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@ -6,11 +6,15 @@
// Serial Print out powers of all 4096 bins in
// dB relative to Sine Wave Full Scale
// Rev- added i to print; added filters on oscillators. 23 Feb 2022
// Public Domain
#include "OpenAudio_ArduinoLibrary.h"
#include "AudioStream_F32.h"
int jj = 0;
// GUItool: begin automatically generated code
AudioSynthSineCosine_F32 sine_cos1; //xy=76,532
AudioAnalyzeFFT4096_IQ_F32 FFT4096iq1; //xy=243,532
@ -37,10 +41,13 @@ void setup(void) {
// or pick any old frequency
sine_cos1.frequency(1000.0f);
// elect the output format
// Engage the identical BP Filters on sine/cosine outputs (true).
sine_cos1.pureSpectrum(true);
// Select the output format FFT_POWER, FFT_RMS or FFT_DBFS
FFT4096iq1.setOutputType(FFT_DBFS);
// Select the wndow function
// Select the wndow function from these four:
//FFT4096iq1.windowFunction(AudioWindowNone);
//FFT4096iq1.windowFunction(AudioWindowHanning4096);
//FFT4096iq1.windowFunction(AudioWindowKaiser4096, 55.0f);
@ -50,24 +57,37 @@ void setup(void) {
// float* pw = FFT4096iq1.getWindow(); // Print window
// for (int i=0; i<4096; i++) Serial.println(pw[i], 7);
// xAxis, bit 0 left/right; bit 1 low to high; default 0X03
// x-Axis direction and offset for sine to I and cosine to Q.
// If xAxis=0 f=fs/2 in middle, f=0 on right edge
// If xAxis=1 f=fs/2 in middle, f=0 on left edge
// If xAxis=2 f=fs/2 on left edge, f=0 in middle
// If xAxis=3 f=fs/2 on right edge, f=0 in middle
// xAxis=3 is mathematically proper -fs/2 start to +fs/2 stop.
FFT4096iq1.setXAxis(0X03);
FFT4096iq1.setNAverage(1);
delay(100);
FFT4096iq1.setNAverage(5);
}
void loop(void) {
static bool doPrint=true;
float *pPwr;
// Print output, once
if( FFT4096iq1.available() && doPrint ) {
if( FFT4096iq1.available() && jj++>2 && doPrint )
{
pPwr = FFT4096iq1.getData();
for(int i=0; i<4096; i++)
{
// Serial.print((int)((float32_t)i * 44100.0/4096.0)); // Print freq
Serial.print(i); // FFT Output index (0, 4095)
Serial.print(" ");
Serial.println(*(pPwr + i), 8 );
}
doPrint = false;
}
Serial.print(" Audio MEM Float32 Peak: ");
Serial.println(AudioMemoryUsageMax_F32());
delay(500);
if(doPrint)
{
Serial.print(" Audio MEM Float32 Peak: ");
Serial.println(AudioMemoryUsageMax_F32());
}
delay(100);
}

31
synth_sin_cos_f32.cpp
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@ -1,5 +1,5 @@
/* synth_sin_cos_f32.cpp
*
*
* SynthSinCos_F32 Bob Larkin April 17, 2020
*
* Based on Chip Audette's OpenAudio sine(), that was
@ -44,13 +44,13 @@ void AudioSynthSineCosine_F32::update(void) {
float32_t a, b, deltaPhase, phaseC;
blockS = AudioStream_F32::allocate_f32(); // Output blocks
if (!blockS) return;
blockC = AudioStream_F32::allocate_f32();
if (!blockC) {
AudioStream_F32::release(blockS);
return;
}
// doSimple has amplitude (-1, 1) and sin/cos differ by 90.00 degrees.
if (doSimple) {
for (i=0; i < block_length; i++) {
@ -62,15 +62,16 @@ void AudioSynthSineCosine_F32::update(void) {
/* Read two nearest values of input value from the sin table */
a = sinTable512_f32[index];
b = sinTable512_f32[index+1];
blockS->data[i] = a + 0.001953125*(b-a)*deltaPhase; /* Linear interpolation process */
// Corrected
// blockS->data[i] = a + 0.001953125*(b-a)*deltaPhase; /* Linear interpolation process */
blockS->data[i] = a+(b-a)*deltaPhase; /* Linear interpolation process */
/* Repeat for cosine by adding 90 degrees phase */
index = (index + 128) & 0x01ff;
/* Read two nearest values of input value from the sin table */
a = sinTable512_f32[index];
b = sinTable512_f32[index+1];
/* deltaPhase will be the same as used for sin */
blockC->data[i] = a + 0.001953125*(b-a)*deltaPhase; /* Linear interpolation process */
blockC->data[i] = a +(b-a)*deltaPhase; /* Linear interpolation process */
}
}
else { // Do a more flexible update, i.e., not doSimple
@ -82,8 +83,8 @@ void AudioSynthSineCosine_F32::update(void) {
/* Read two nearest values of input value from the sin table */
a = sinTable512_f32[index];
b = sinTable512_f32[index+1];
blockS->data[i] = amplitude_pk*(a + 0.001953125*(b-a)*deltaPhase); /* Linear interpolation process */
blockS->data[i] = amplitude_pk*(a +(b-a)*deltaPhase); /* Linear interpolation process */
/* Shift forward phaseS_C and get cos. First, the calculation of index of the table */
phaseC = phaseS + phaseS_C;
if (phaseC > 512.0f) phaseC -= 512.0f;
@ -92,13 +93,21 @@ void AudioSynthSineCosine_F32::update(void) {
/* Read two nearest values of input value from the sin table */
a = sinTable512_f32[index];
b = sinTable512_f32[index+1];
blockC->data[i] = amplitude_pk*(a + 0.001953125*(b-a)*deltaPhase); /* Linear interpolation process */
blockC->data[i] = amplitude_pk*(a +(b-a)*deltaPhase); /* Linear interpolation process */
}
}
// For higher frequencies, an optional bandpass filter the output
// This does a pass through for lower frequencies
if(doPureSpectrum)
{
arm_biquad_cascade_df1_f32(&bq_instS, blockS->data, blockS->data, 128);
arm_biquad_cascade_df1_f32(&bq_instC, blockC->data, blockC->data, 128);
}
AudioStream_F32::transmit(blockS, 0);
AudioStream_F32::release (blockS);
AudioStream_F32::release (blockS);
AudioStream_F32::transmit(blockC, 1);
AudioStream_F32::release (blockC);
return;
}

63
synth_sin_cos_f32.h
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@ -1,8 +1,10 @@
/* synth_sin_cos_f32.h
* AudioSynthSineCosine_F32
*
*
* Status: Checked for function and accuracy. 19 April 2020
* 10 March 2021 Corrected Interpolation equations. Bob L
* 23 Feb 2022 Added "pure spectrum" filtering. RSL
* This adds the function pureSpectrum(bool _setPure); // Default: false
*
* Created: Bob Larkin 15 April 2020
*
@ -19,19 +21,19 @@
* (-amplitude, amplitude) range. If these are at the default values,
* called doSimple, the caluclation is faster.
* For doSimple either true or false, the frequency can be changed
* at will using the frequency() method.
*
* at will using the frequency() method.
*
* Defaults:
* Frequency: 1000.0 Hz
* Phase of Sine: 0.0 radians (0.0 deg)
* Phase of Cosine: pi/2 radians (90.0 deg) ahead of Sine
* Amplitude: -1.0 to 1.0
*
*
* Time: T3.6 update() block of 128 with doSimple is 36 microseconds
* Same using flexible doSimple=false is 49 microseconds
* T4.0 update() block of 128 with doSimple is 16 microseconds
* Same using flexible doSimple=false is 24 microseconds
*
*
* Copyright (c) 2020 Bob Larkin
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
@ -79,16 +81,47 @@ class AudioSynthSineCosine_F32 : public AudioStream_F32 {
public:
AudioSynthSineCosine_F32(void) : AudioStream_F32(0, NULL) { } //uses default AUDIO_SAMPLE_RATE from AudioStream.h
AudioSynthSineCosine_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
AudioSynthSineCosine_F32(const AudioSettings_F32 &settings) : AudioStream_F32(0, NULL) {
setSampleRate_Hz(settings.sample_rate_Hz);
setBlockLength(settings.audio_block_samples);
}
void frequency(float32_t fr) { // Frequency in Hz
freq = fr;
if (freq < 0.0f) freq = 0.0f;
else if (freq > sample_rate_Hz/2.0f) freq = sample_rate_Hz/2.0f;
phaseIncrement = 512.0f * freq / sample_rate_Hz;
// Find coeff for 2 stages of BPF to remove harmoncs
// Always compute these in case pureSpectrum is enabled later.
if(freq > 0.003f*sample_rate_Hz)
{
float32_t q = 20.0f;
float32_t w0 = freq * (2.0f * 3.141592654f / sample_rate_Hz);
float32_t alpha = sin(w0) / (q * 2.0);
float32_t scale = 1.0f / (1.0f + alpha);
/* b0 */ coeff32[0] = alpha * scale;
/* b1 */ coeff32[1] = 0;
/* b2 */ coeff32[2] = (-alpha) * scale;
/* a1 */ coeff32[3] = -(-2.0 * cos(w0)) * scale;
/* a2 */ coeff32[4] = -(1.0 - alpha) * scale;
/* b0 */ coeff32[5] = coeff32[0];
/* b1 */ coeff32[6] = coeff32[1];
/* b2 */ coeff32[7] = coeff32[2];
/* a1 */ coeff32[8] = coeff32[3];
/* a2 */ coeff32[9] = coeff32[4];
arm_biquad_cascade_df1_init_f32( &bq_instS, 2, coeff32, state32S );
arm_biquad_cascade_df1_init_f32( &bq_instC, 2, coeff32, state32C );
}
else
{
for(int ii=0; ii<10; ii++) // Coeff for BiQuad BPF
coeff32[ii] = 0.0;
coeff32[0] = 1.0; // b0 = 1 for pass through
coeff32[5] = 1.0;
arm_biquad_cascade_df1_init_f32( &bq_instS, 2, coeff32, state32S );
arm_biquad_cascade_df1_init_f32( &bq_instC, 2, coeff32, state32C );
}
}
/* Externally, phase comes in the range (0,2*M_PI) keeping with C math functions
@ -102,7 +135,7 @@ public:
doSimple = false;
return;
}
// phaseS_C_r is the number of radians that the cosine output leads the
// sine output. The default is M_PI_2 = pi/2 = 1.57079633 radians,
// corresponding to 90.00 degrees cosine leading sine.
@ -123,7 +156,7 @@ public:
return;
}
// Speed up calculations by setting phaseS_C=90deg, amplitude=1
// Speed up calculations by setting phaseS_C=90deg, amplitude=1
// Note, s=true will override any setting of phaseS_C_r or amplitude.
void simple(bool s) {
doSimple = s;
@ -132,7 +165,7 @@ public:
amplitude_pk = 1.0f;
}
return;
}
}
void setSampleRate_Hz(float32_t fs_Hz) {
// Check freq range
@ -145,7 +178,9 @@ public:
if(bl > 128) bl = 128;
block_length = bl;
}
void pureSpectrum(bool _setPure) { doPureSpectrum = _setPure; }
virtual void update(void);
private:
@ -159,5 +194,11 @@ private:
// if only freq() is used, the complexities of phase, phaseS_C,
// and amplitude are not used, speeding up the sin and cos:
bool doSimple = true;
bool doPureSpectrum = false; // Adds bandpass filter (not normally needed)
float32_t coeff32[10]; // 2 biquad stages for filtering output shared S&C
float32_t state32S[8];
arm_biquad_casd_df1_inst_f32 bq_instS; // ARM DSP Math library filter instance.
float32_t state32C[8];
arm_biquad_casd_df1_inst_f32 bq_instC; // ARM DSP Math library filter instance.
};
#endif

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