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New class for W9GR CESSB

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boblark 2 years ago
parent 2aa2bc444c
commit 5b4bca69ad
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  1. 273
      radioCESSBtransmit_F32.cpp
  2. 486
      radioCESSBtransmit_F32.h

273
radioCESSBtransmit_F32.cpp
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/*
* radioCESSBtransmit_F32.cpp
*
* Bob Larkin, Dec 2022, in support of the library:
* Chip Audette, OpenAudio_ArduinoLibrary
*
* MIT License, Use at your own risk.
*
* See radioCESSBtransmit_F32.h for technical info.
*
*/
#include "radioCESSBtransmit_F32.h"
// 513 values of the sine wave in a float array:
#include "sinTable512_f32.h"
// sincos(ph) inputs phase on (0, 512) and outputs private sn, cs
// A simplified version of the F32 synthesizer class
// AudioSynthSineCosine_F32. Full F32 accuracy
void radioCESSBtransmit_F32::sincos(float32_t ph) {
uint16_t index;
float32_t a, b, deltaPhase;
index = (uint16_t)ph;
deltaPhase = ph -(float32_t)index;
/* Read two nearest values of input value from the sin table */
a = sinTable512_f32[index];
b = sinTable512_f32[index+1];
sn = 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 */
cs = a +(b-a)*deltaPhase; /* Linear interpolation process */
// if(ttt++ <100){Serial.print(ttt); Serial.print(","); Serial.println(sn, 8); } <<<<<<
}
void radioCESSBtransmit_F32::update(void) {
audio_block_f32_t *blockIn, *blockOutI, *blockOutQ;
// Temporary storage. At an audio sample rate of 96 ksps, the used
// space will be half of the declared space.
// Todo: Cut 1 or two arrays out by more sharing
float32_t weaverIn[32];
float32_t weaverMI[32];
float32_t weaverMQ[32];
float32_t workingDataI[128];
float32_t workingDataQ[128];
float32_t delayedDataI[64]; // Allows batching of 64 data points
float32_t delayedDataQ[64];
float32_t diffI[64];
float32_t diffQ[64];
if(sampleRate!=SAMPLE_RATE_44_50 && sampleRate!=SAMPLE_RATE_88_100)
return;
// Get all needed resources, or return if not available.
blockIn = AudioStream_F32::receiveReadOnly_f32();
if (!blockIn)
{ return; }
blockOutI = AudioStream_F32::allocate_f32(); // a block for I output
if (!blockOutI)
{
AudioStream_F32::release(blockIn);
return;
}
blockOutQ = AudioStream_F32::allocate_f32(); // and for Q
if (!blockOutQ)
{
AudioStream_F32::release(blockOutI);
AudioStream_F32::release(blockIn);
return;
}
/* A +/- pulse to test timing of various delays. PULSE TEST
* This replaces any input from the audio stream
for(int kk=0; kk<128; kk++)
{
uint16_t y=(ny & 1023);
// pulse max at 1.548 is just starting to clip
// 2.189 is 3 dB increase
if (y>=100 && y<115) blockIn->data[kk] = 2.189f;
else if(y>=115 && y<130) blockIn->data[kk] = -2.189f;
else blockIn->data[kk] = 0.0f;
ny++;
// Serial.println(blockIn->data[kk]);
} */
// Decimate 48 ksps to 12 ksps, 128 to 32 samples
// or 96 ksps to 12 ksps, 128 to 16 samples (not yet)
arm_fir_decimate_f32(&decimateInst, &(blockIn->data[0]),
&weaverIn[0], 128);
// We now have 32 or 16 samples to process and interpolate out
float32_t gainIn2 = 2.0f*gainIn; // 2 because the mixers are 0.5
for(int k=0; k<nW; k++)
{
weaverIn[k] *= gainIn2; // Input gain for CESSB
phaseW += phaseIncrementW;
if(phaseW >=512.0f)
phaseW -= 512.0f;
sincos(phaseW); // Generate cs, sn
if(sidebandReverse)
weaverMI[k] = -weaverIn[k]*cs; // Quadrature mixers
else
weaverMI[k] = weaverIn[k]*cs;
weaverMQ[k] = weaverIn[k]*sn;
}
// Filter Weaver I and Q using first half of Out array.
// Bandwidth at this point is 0 to 1350 Hz.
arm_fir_f32(&firInstWeaverI, weaverMI, workingDataI, nW);
arm_fir_f32(&firInstWeaverQ, weaverMQ, workingDataQ, nW);
// Note: Sine wave envelope gain from blockIn->data[kk] to here is gainIn
// Mesaure input power and peak envelope, SSB before any CESSB processing
for(int k=0; k<nW; k++)
{
float32_t pwrWorkingData = workingDataI[k]*workingDataI[k] + workingDataQ[k]*workingDataQ[k];
float32_t vWD = sqrtf(pwrWorkingData); // Envelope
powerSum0 += pwrWorkingData;
if(vWD > maxMag0)
maxMag0 = vWD; // Peak envelope
countPower0++;
}
// Interpolate by 2 up to 24 ksps rate
for(int k=0; k<nW; k++) // 48 ksps: 0 to 31
{
int k2 = 2*(nW - k) - 1; // 48 ksps 63 to 1
// Zero pack, working from the bottom to not overwrite
workingDataI[k2] = 0.0f; // 64 element array
workingDataI[k2-1] = workingDataI[nW-k-1];
workingDataQ[k2] = 0.0f;
workingDataQ[k2-1] = workingDataQ[nW-k-1];
}
// LPF with gain of 2 built into coefficients, correct for zeros.
arm_fir_f32(&firInstInterpolate1I, workingDataI, workingDataI, nC);
arm_fir_f32(&firInstInterpolate1Q, workingDataQ, workingDataQ, nC);
// WorkingDataI and Q are now at 24 ksps and ready for clipping
// For input 48 ksps this produces 64 numbers
// Voltage gain from blockIn->data to here for small sine wave is 1.0
for(int kk=0; kk<nC; kk++)
{
float32_t power = workingDataI[kk]*workingDataI[kk] + workingDataQ[kk]*workingDataQ[kk];
float32_t mag = sqrtf(power);
if(mag > 1.0f) // This the clipping, scaled to 1.0, desired max
{
workingDataI[kk] /= mag;
workingDataQ[kk] /= mag;
}
powerSum0 += power; // For measuring amount of clipping
if(mag > maxMag0)
maxMag0 = mag;
}
// clipperIn needs spectrum control, so LP filter it. Same filter coeffs as Weaver.
// Both BW of the signal and the sample rate have been doubled.
arm_fir_f32(&firInstClipperI, workingDataI, workingDataI, nC);
arm_fir_f32(&firInstClipperQ, workingDataQ, workingDataQ, nC);
// Ready to compensate for filter overshoots
for (int k=0; k<64; k++)
{
/* ======= Sidebar: Circular 2^n length delay arrays ========
*
* The length of the array, N,
* must be a power of 2. For example N=2^6 = 64. The minimum
* delay possible is the trivial case of 0 up to N-1.
* As in C, let i be the index of the N array elements which
* would range from 0 to N-1. If p is an integer, that is a power
* of 2 also, with p >= n, it can serve as an index to the
* delay array by "ANDing" it with (N-1). That is,
* i = p & (N-1). It can be convenient if the largest
* possible value of the integer p, plus 1, is an integer multiple
* of the arrray size N, as then the rollover of p will not cause
* a jump in i. For instance, if p is an uint8_t with a maximum
* value of pmax=255, (pmax+1)/N = (255+1)/64 = 4, which is an
* integer. This combination will have no problems from rollover
* of p.
*
* The new data point is entered at index p & (N - 1). To
* achieve a delay of d, the output of the delay array is taken
* at index ((p-d) & (N-1)). The index is then incremented by 1.
* ========================================================== */
// Circular delay line for signal to align data with FIR output
// Put I & Q data points into the delay arrays
osDelayI[indexOsDelay & 0X3F] = workingDataI[k];
osDelayQ[indexOsDelay & 0X3F] = workingDataQ[k];
// Remove 64 points delayed data from line and save for later
delayedDataI[k] = osDelayI[(indexOsDelay - 63) & 0X3F];
delayedDataQ[k] = osDelayQ[(indexOsDelay - 63) & 0X3F];
indexOsDelay++;
// Delay line to allow strongest envelope to be used for compensation
// We only need to look ahead 1 or behind 1, so delay line of 4 is OK.
// Enter latest envelope to delay array
osEnv[indexOsEnv & 0X03] = sqrtf(
workingDataI[k]*workingDataI[k] + workingDataQ[k]*workingDataQ[k]);
// look over the envelope curve to find the max
float32_t eMax = 0.0f;
if(osEnv[(indexOsEnv) & 0X03] > eMax) // One just entered
eMax = osEnv[(indexOsEnv) & 0X03];
if(osEnv[(indexOsEnv-1) & 0X03] > eMax) // Entered one before
eMax = osEnv[(indexOsEnv-1) & 0X03];
if(osEnv[(indexOsEnv-2) & 0X03] > eMax) // Entered one before that
eMax = osEnv[(indexOsEnv-2) & 0X03];
if(eMax < 1.0f)
eMax = 1.0f; // Below clipping region
indexOsEnv++;
// Clip the signal to 1.0. -2 allows 1 look ahead on signal.
float32_t eCorrectedI = osDelayI[(indexOsDelay - 2) & 0X3F] / eMax;
float32_t eCorrectedQ = osDelayQ[(indexOsDelay - 2) & 0X3F] / eMax;
// Filtering is linear, so we only need to filter the difference between
// the signal and the clipper output. This needs less filtering, as the
// difference is many dB below the signal to begin with. Hershberger 2014
diffI[k] = osDelayI[(indexOsDelay - 2) & 0X3F] - eCorrectedI;
diffQ[k] = osDelayQ[(indexOsDelay - 2) & 0X3F] - eCorrectedQ;
} // End, for k=0 to 63
// Filter the differences, osFilter has 129 taps and 64 delay
arm_fir_f32(&firInstOShootI, diffI, diffI, nC);
arm_fir_f32(&firInstOShootQ, diffQ, diffQ, nC);
// Do the overshoot compensation
for(int k=0; k<64; k++)
{
workingDataI[k] = delayedDataI[k] - gainCompensate*diffI[k];
workingDataQ[k] = delayedDataQ[k] - gainCompensate*diffQ[k];
}
// Finally interpolate to 48 or 96 ksps. Data is in workingDataI[k]
// and is 64 samples for audio 48 ksps.
for(int k=0; k<nC; k++) // Audio sampling at 48 ksps: 0 to 63
{
int k2 = 2*(nC - k) - 1; // 48 ksps 63 to 1
// Zero pack, working from the bottom to not overwrite
workingDataI[k2] = 0.0f;
workingDataI[k2-1] = workingDataI[nC-k-1];
workingDataQ[k2] = 0.0f;
workingDataQ[k2-1] = workingDataQ[nC-k-1];
}
// LPF with gain of 2 built into coefficients, correct for zeros.
arm_fir_f32(&firInstInterpolate2I, workingDataI, &blockOutI->data[0], 2*nC);
arm_fir_f32(&firInstInterpolate2Q, workingDataQ, &blockOutQ->data[0], 2*nC);
// Voltage gain from blockIn->data to here for small sine wave is 1.0
// Measure output power and peak envelope, after CESSB
for(int k=0; k<128; k++)
{
float32_t pwrOut = blockOutI->data[k]*blockOutI->data[k] + blockOutQ->data[k]*blockOutQ->data[k];
float32_t vWD = sqrtf(pwrOut); // Envelope
powerSum1 += pwrOut;
if(vWD > maxMag1)
maxMag1 = vWD; // Peak envelope
countPower1++;
}
AudioStream_F32::transmit(blockOutI, 0); // send the outputs
AudioStream_F32::transmit(blockOutQ, 1);
AudioStream_F32::release(blockIn); // Release the blocks
AudioStream_F32::release(blockOutI);
AudioStream_F32::release(blockOutQ);
} // end update()

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radioCESSBtransmit_F32.h
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/*
* radioCESSBtransmit_F32.h
*
* 2 Dec 2022 Bob Larkin
* With much credit to:
* Chip Audette (OpenAudio) Feb 2017
* and of course, to PJRC for the Teensy and Teensy Audio Library
*
* The development of the Controlled Envelope Single Side Band (CESSB)
* was done by Dave Hershberger, W9GR. Many thanks to Dave.
* The following description is mostly taken
* from Frank, DD4WH and is on line at the GNU Radio site, ref:
* https://github-wiki-see.page/m/df8oe/UHSDR/wiki/Controlled-Envelope-Single-Sideband-CESSB
*
* Controlled Envelope Single Sideband is an invention by Dave Hershberger
* W9GR with the aim to "allow your rig to output more average power while
* keeping peak envelope power PEP the same". The increase in perceived
* loudness can be up to 4dB without any audible increase in distortion
* and without making you sound "processed" (Hershberger 2014, 2016b).
*
* The principle to achieve this is relatively simple. The process
* involves only audio baseband processing which can be done digitally in
* software without the need for modifications in the hardware or messing
* with the RF output of your rig.
*
* Controlled Envelope Single Sideband can be produced using three
* processing blocks making up a complete CESSB system:
* 1. An SSB modulator. This is implemented as a Weaver system to allow
* minimum (12 kHz) decimated sample rate with the output of I & Q
* signals (a complex SSB signal).
* 2. A baseband envelope clipper. This takes the modulus of the I & Q
* signals (also called the magnitude), which is sqrt(I * I + Q * Q)
* and divides the I & Q signals by the modulus, IF the signal is
* larger than 1.0. If not, the signal remains untouched. After
* clipping, the signal is lowpass filtered with a linear phase FIR
* low pass filter with a stopband frequency of 3.0kHz
* 3. An overshoot controller . This does something similar as the
* envelope clipper: Again, the modulus is calculated (but now on
* the basis of the current and two preceding and two subsequent
* samples). If the signals modulus is larger than 1 (clipping),
* the signals I and Q are divided by the maximum of 1 or of
* (1.9 * signal). That means the clipping is overcompensated by 1.9
* which leads to a much better suppression of the overshoots from
* the first two stages. Finally, the resulting signal is again
* lowpass-filtered with a linear phase FIR filter with stopband
* frequency of 3.0khz
*
* It is important that the sample rate is high enough so that the higher
* frequency components of the output of the modulator, clipper and
* overshoot controller do not alias back into the desired signal. Also
* all the filters should be linear phase filters (FIR, not IIR).
*
* This CESSB system can reduce the overshoot of the SSB modulator from
* 61% to 1.3%, meaning about 2.5 times higher perceived SSB output power
* (Hershberger 2014).
*
* References:
* 1-Hershberger, D.L. (2014): Controlled Envelope Single Sideband. QEX
* November/December 2014 pp3-13.
* http://www.arrl.org/files/file/QEX_Next_Issue/2014/Nov-Dec_2014/Hershberger_QEX_11_14.pdf
* 2-Hershberger, D.L. (2016a): External Processing for Controlled
* Envelope Single Sideband. - QEX January/February 2016 pp9-12.
* http://www.arrl.org/files/file/QEX_Next_Issue/2016/January_February_2016/Hershberger_QEX_1_16.pdf
* 3-Hershberger, D.L. (2016b): Understanding Controlled Envelope Single
* Sideband. - QST February 2016 pp30-36.
* 4-Forum discussion on CESSB on the Flex-Radio forum,
* https://community.flexradio.com/discussion/6432965/cessb-questions
*
* Weaver Method of SSB: Note that this class includes a good umplementation
* of the Weaver method. To use this without invoking the CESSB corrections,
* just keep the input peak level below 1.0. One could disable CESSB by
* setting gainCompensate=0.0, but that serves no purpose if the input level
* is below the clipping point.
*
* Status: 44 to 50 ksps sample rate working per ref 1 above.
* 96 ksps is not yet implemented. Anyone need this?
*
* Inputs: 0 is voice audio input
* Outputs: 0 is I 1 is Q
*
* Functions, available during operation:
* void frequency(float32_t fr) Sets LO frequency Hz
*
* void setSampleRate_Hz(float32_t fs_Hz) Allows dynamic sample rate
* change for this function
*
* struct levels* getLevels(int what) {
* what = 0 returns a pointer to struct levels before data is ready
* what = 1 returns a pointer to struct levels
*
* uint32_t levelDataCount() return countPower0
*
* void setGains(float32_t gIn, float32_t gCompensate, float32_t gOut)
*
* Time: T3.6 For an update of a 128 sample block, estimated 750 microseconds
* T4.0 For an update of a 128 sample block, measured 252 microseconds
* These times are for a 48 ksps rate, for which about 2667 microseconds
* are available.
*/
#ifndef _radioCESSBtransmit_f32_h
#define _radioCESSBtransmit_f32_h
#include "Arduino.h"
#include "AudioStream_F32.h"
#include "arm_math.h"
#include "mathDSP_F32.h"
#define SAMPLE_RATE_0 0
#define SAMPLE_RATE_44_50 1
#define SAMPLE_RATE_88_100 2
#ifndef M_PI
#define M_PI 3.141592653589793f
#endif
#ifndef M_PI_2
#define M_PI_2 1.570796326794897f
#endif
#ifndef M_TWOPI
#define M_TWOPI (M_PI * 2.0f)
#endif
// For the average power and peak voltage readings, global
struct levels {
float32_t pwr0;
float32_t peak0;
float32_t pwr1;
float32_t peak1;
uint32_t countP; // Number of averaged samples for pwr0.
};
class radioCESSBtransmit_F32 : public AudioStream_F32 {
//GUI: inputs:1, outputs:2 //this line used for automatic generation of GUI node
//GUI: shortName:CESSBTransmit //this line used for automatic generation of GUI node
public:
radioCESSBtransmit_F32(void) :
AudioStream_F32(1, inputQueueArray_f32)
{
setSampleRate_Hz(AUDIO_SAMPLE_RATE);
//uses default AUDIO_SAMPLE_RATE from AudioStream.h
//setBlockLength(128); Always default 128
}
radioCESSBtransmit_F32(const AudioSettings_F32 &settings) :
AudioStream_F32(1, inputQueueArray_f32)
{
setSampleRate_Hz(settings.sample_rate_Hz);
//setBlockLength(128); Always default 128
}
// Sample rate starts at default 44.1 ksps. That will work. Filters
// are designed for 48 and 96 ksps, however. This is a *required*
// function at setup().
void setSampleRate_Hz(const float fs_Hz) {
sample_rate_Hz = fs_Hz;
if(sample_rate_Hz>44000.0f && sample_rate_Hz<50100.0f)
{
// Design point is 48 ksps
sampleRate = SAMPLE_RATE_44_50;
nW = 32;
nC = 64;
countLevelMax = 37; // About 0.1 sec for 48 ksps
inverseMaxCount = 1.0f/(float32_t)countLevelMax;
Serial.print("Status, decimate init = "); Serial.println(
arm_fir_decimate_init_f32(&decimateInst, 65, 4,
(float32_t*)decimateFilter48, &pStateDecimate[0], 128) );
// Putting this init stuff here is in anticipation of
// adding 96 ksps support later.
arm_fir_init_f32(&firInstWeaverI, 213, (float32_t*)weaverFilter,
&pStateWeaverI[0], nW);
arm_fir_init_f32(&firInstWeaverQ, 213, (float32_t*)weaverFilter,
&pStateWeaverQ[0], nW);
arm_fir_init_f32(&firInstInterpolate1I, 23, (float32_t*)interpolateFilter1,
&pStateInterpolate1I[0], nC);
arm_fir_init_f32(&firInstInterpolate1Q, 23, (float32_t*)interpolateFilter1,
&pStateInterpolate1Q[0], nC);
arm_fir_init_f32(&firInstClipperI, 213, (float32_t*)weaverFilter,
&pStateClipperI[0], nC);
arm_fir_init_f32(&firInstClipperQ, 213, (float32_t*)weaverFilter,
&pStateClipperQ[0], nC);
arm_fir_init_f32(&firInstOShootI, 125, (float32_t*)osFilter,
&pStateOShootI[0], nC);
arm_fir_init_f32(&firInstOShootQ, 125, (float32_t*)osFilter,
&pStateOShootQ[0], nC);
arm_fir_init_f32(&firInstInterpolate2I, 23, (float32_t*)interpolateFilter1,
&pStateInterpolate2I[0], nC);
arm_fir_init_f32(&firInstInterpolate2Q, 23, (float32_t*)interpolateFilter1,
&pStateInterpolate2Q[0], nC);
}
/* else if(sample_rate_Hz>88000.0f && sample_rate_Hz<100100.0f)
{
// GET THINGS WORKING AT SAMPLE_RATE_44_50 FIRST AND THEN FIX UP 96 ksps
// Design point is 96 ksps
}
*/
else
{
// Unsupported sample rate
sampleRate = SAMPLE_RATE_0;
nW = 1;
nC = 1;
}
phaseIncrementW = 512.0f * freqW / 12000.0f; // 57.6 for 12ksps
newLevelDataReady = false;
}
struct levels* getLevels(int what) {
if(what != 0) // 0 leaves a way to get pointer before data is ready
{
levelData.pwr0 = powerSum0/(2.975f*(float32_t)countPower0); // WHY????
levelData.peak0 = maxMag0;
levelData.pwr1 = powerSum1/(float32_t)countPower1;
levelData.peak1 = maxMag1;
levelData.countP = countPower0;
// Automatic reset for next set of readings
powerSum0 = 0.0f;
maxMag0 = -1.0f;
powerSum1 = 0.0f;
maxMag1 = -1.0f;
countPower0 = 0;
countPower1 = 0;
}
return &levelData;
}
uint32_t levelDataCount(void) {
return countPower0; // Input count, out may be different
}
void setGains(float32_t gIn, float32_t gCompensate, float32_t gOut)
{
gainIn = gIn;
gainCompensate = gCompensate;
gainOut = gOut;
}
// The LSB/USB selection depends on the processing of the
// IQ outputs of this class. But, what we can do here is to reverse the
// selectio by reversing the phase of one of the Weaver LO's.
void setSideband(bool _sbReverse)
{
sidebandReverse = _sbReverse;
}
virtual void update(void);
private:
void sincos(float32_t ph);
struct levels levelData;
audio_block_f32_t *inputQueueArray_f32[1];
float32_t freqW = 1350.0f; // Set here and not changed
// Input/Output is at 48 (or later 96 ksps). Weaver generation is at 12 ksps.
// Clipping and overshoot processing is at 24 ksps.
// Next line is to indicate that setSampleRateHz() has not executed
int sampleRate = SAMPLE_RATE_0;
float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE; // 44.1 ksps
int16_t nW = 32; // 32 or 16
int16_t nC = 64; // 64 or 32
float32_t phaseIncrementW = 512.0f * freqW / 24000.0f;
float32_t phaseW = 0.0f; // Weaver signal 0.0 to 512.0
uint16_t block_length = 128;
bool sidebandReverse = false;
float32_t pStateDecimate[128 + 65 - 1]; // Goes with CMSIS decimate function
arm_fir_decimate_instance_f32 decimateInst;
float32_t pStateWeaverI[32 + 213 - 1]; // Goes with Weaver filter out
arm_fir_instance_f32 firInstWeaverI; // at 12 ksps
float32_t pStateWeaverQ[32 + 213 - 1];
arm_fir_instance_f32 firInstWeaverQ;
float32_t pStateInterpolate1I[64 + 23 - 1]; // For interpolate 12 to 24 ksps
arm_fir_instance_f32 firInstInterpolate1I;
float32_t pStateInterpolate1Q[64 + 23 - 1];
arm_fir_instance_f32 firInstInterpolate1Q;
float32_t pStateClipperI[64 + 213 - 1]; // Goes with Clipper filter
arm_fir_instance_f32 firInstClipperI; // at 24 ksps
float32_t pStateClipperQ[64 + 213 - 1];
arm_fir_instance_f32 firInstClipperQ;
float32_t pStateOShootI[64+125-1]; // 129-1];
arm_fir_instance_f32 firInstOShootI;
float32_t pStateOShootQ[64+125-1];
arm_fir_instance_f32 firInstOShootQ;
float32_t pStateInterpolate2I[128 + 23 - 1]; // For interpolate 12 to 24 ksps
arm_fir_instance_f32 firInstInterpolate2I;
float32_t pStateInterpolate2Q[128 + 23 - 1];
arm_fir_instance_f32 firInstInterpolate2Q;
float32_t sn, cs;
float32_t gainIn = 1.0f;
float32_t gainCompensate = 2.0f;
float32_t gainOut = 1.0f;
// In the overshoot compensator, we need to search for the highest
// filter output over several samples.
// And a tiny delay to allow negative time for the previous path
float32_t osEnv[4];
uint16_t indexOsEnv = 0; // 0 to 3 by using a 2-bit mask
// We need a delay for overshoot remove to account for the FIR
// filter in the correction path. Making the delay array
// exactly 2^6=64 allows a simple circular structure.
float32_t osDelayI[64];
float32_t osDelayQ[64];
uint16_t indexOsDelay = 0;
// RMS and Peak variable for monitoring levels and changes to the
// Peak to RMS ratio. These are temporary storage. Data is
// transferred by global levelData struct at the top of this file.
float32_t powerSum0 = 0.0f;
float32_t maxMag0 = -1.0f;
float32_t powerSum1 = 0.0f;
float32_t maxMag1 = -1.0f;
uint32_t countPower0 = 0;
uint32_t countPower1 = 0;
bool newLevelDataReady = false;
int countLevel = 0;
int countLevelMax = 37; // About 0.1 sec for 48 ksps
float32_t inverseMaxCount = 1.0f/(float32_t)countLevelMax;
// uint16_t ny = 0; // For test pulse generation
/* Input filter for decimate by 4:
* FIR filter designed with http://t-filter.appspot.com
* Sampling frequency: 48000 Hz
* 0 Hz - 3000 Hz ripple = 0.075 dB
* 6000 Hz - 24000 Hz atten = -95.93 dB */
const float32_t decimateFilter48[65] = {
0.00004685f, 0.00016629f, 0.00038974f, 0.00073279f, 0.00113663f, 0.00148721f,
0.00159057f, 0.00125129f, 0.00032821f,-0.00114283f,-0.00289782f,-0.00441933f,
-0.00505118f,-0.00418143f,-0.00151748f, 0.00268876f, 0.00751487f, 0.01147689f,
0.01286243f, 0.01027735f, 0.00323528f,-0.00737003f,-0.01913035f,-0.02842381f,
-0.03117447f,-0.02390063f,-0.00480378f, 0.02544011f, 0.06344286f, 0.10357132f,
0.13904464f, 0.16342506f, 0.17210799f, 0.16342506f, 0.13904464f, 0.10357132f,
0.06344286f, 0.02544011f,-0.00480378f,-0.02390063f,-0.03117447f,-0.02842381f,
-0.01913035f,-0.00737003f, 0.00323528f, 0.01027735f, 0.01286243f, 0.01147689f,
0.00751487f, 0.00268876f,-0.00151748f,-0.00418143f,-0.00505118f,-0.00441933f,
-0.00289782f,-0.00114283f, 0.00032821f, 0.00125129f, 0.00159057f, 0.00148721f,
0.00113663f, 0.00073279f, 0.00038974f, 0.00016629f, 0.00004685};
/* FIR filter for Weaver I & Q
* Filter designed with http://t-filter.appspot.com
* Sampling frequency: 12000 ksps
* 0 Hz - 1350 Hz ripple = 0.14 dB
* 1500 Hz - 6000 Hz atten = -60.2 dB
* ALSO: 0 to 2700 Hz at 24 ksps */
const float32_t weaverFilter[213] = {
0.00069446f, 0.00037170f, 0.00016640f,-0.00025667f,-0.00077930f,-0.00120663f,
-0.00134867f,-0.00111550f,-0.00057687f, 0.00005147f, 0.00049736f, 0.00056149f,
0.00022366f,-0.00033377f,-0.00080586f,-0.00091552f,-0.00056344f, 0.00010449f,
0.00075723f, 0.00104136f, 0.00077294f, 0.00005168f,-0.00076730f,-0.00124489f,
-0.00108978f,-0.00033029f, 0.00067306f, 0.00139546f, 0.00142002f, 0.00067429f,
-0.00050084f,-0.00150186f,-0.00176980f,-0.00109852f, 0.00022372f, 0.00153080f,
0.00211108f, 0.00159111f, 0.00016633f,-0.00146039f,-0.00242101f,-0.00214184f,
-0.00067864f, 0.00126494f, 0.00267008f, 0.00273272f, 0.00131711f,-0.00091957f,
-0.00282456f,-0.00333871f,-0.00207907f, 0.00040237f, 0.00284896f, 0.00392959f,
0.00295636f, 0.00030577f,-0.00270677f,-0.00447189f,-0.00393839f,-0.00122551f,
0.00235504f, 0.00492259f, 0.00500607f, 0.00237350f,-0.00174927f,-0.00523381f,
-0.00613636f,-0.00376725f, 0.00083831f, 0.00534869f, 0.00730076f, 0.00542689f,
0.00043859f,-0.00520046f,-0.00846933f,-0.00738444f,-0.00216395f, 0.00470259f,
0.00960921f, 0.00969387f, 0.00446038f,-0.00373274f,-0.01068416f,-0.01245333f,
-0.00752832f, 0.00210318f, 0.01166261f, 0.01586953f, 0.01175214f, 0.00053376f,
-0.01251222f,-0.02039576f,-0.01795974f,-0.00492844f, 0.01320402f, 0.02719248f,
0.02832779f, 0.01314687f,-0.01371714f,-0.04016441f,-0.05091338f,-0.03387251f,
0.01403178f, 0.08421962f, 0.15843610f, 0.21483324f, 0.23586349f, 0.21483324f,
0.15843610f, 0.08421962f, 0.01403178f,-0.03387251f,-0.05091338f,-0.04016441f,
-0.01371714f, 0.01314687f, 0.02832779f, 0.02719248f, 0.01320402f,-0.00492844f,
-0.01795974f,-0.02039576f,-0.01251222f, 0.00053376f, 0.01175214f, 0.01586953f,
0.01166261f, 0.00210318f,-0.00752832f,-0.01245333f,-0.01068416f,-0.00373274f,
0.00446038f, 0.00969387f, 0.00960921f, 0.00470259f,-0.00216395f,-0.00738444f,
-0.00846933f,-0.00520046f, 0.00043859f, 0.00542689f, 0.00730076f, 0.00534869f,
0.00083831f,-0.00376725f,-0.00613636f,-0.00523381f,-0.00174927f, 0.00237350f,
0.00500607f, 0.00492259f, 0.00235504f,-0.00122551f,-0.00393839f,-0.00447189f,
-0.00270677f, 0.00030577f, 0.00295636f, 0.00392959f, 0.00284896f, 0.00040237f,
-0.00207907f,-0.00333871f,-0.00282456f,-0.00091957f, 0.00131711f, 0.00273272f,
0.00267008f, 0.00126494f,-0.00067864f,-0.00214184f,-0.00242101f,-0.00146039f,
0.00016633f, 0.00159111f, 0.00211108f, 0.00153080f, 0.00022372f,-0.00109852f,
-0.00176980f,-0.00150186f,-0.00050084f, 0.00067429f, 0.00142002f, 0.00139546f,
0.00067306f,-0.00033029f,-0.00108978f,-0.00124489f,-0.00076730f, 0.00005168f,
0.00077294f, 0.00104136f, 0.00075723f, 0.00010449f,-0.00056344f,-0.00091552f,
-0.00080586f,-0.00033377f, 0.00022366f, 0.00056149f, 0.00049736f, 0.00005147f,
-0.00057687f,-0.00111550f,-0.00134867f,-0.00120663f,-0.00077930f,-0.00025667f,
0.00016640f, 0.00037170f, 0.00069446f};
/* FIR for filtering limiter and overshoot correction
* FIR filter designed with http://t-filter.appspot.com
* Sampling frequency: 24000 Hz
* 0 Hz-1400 Hz gain=1 ripple=0.07 dB
* 1820 Hz-12000 Hz attenuation=40.4 dB
*/
float32_t osFilter[125] = {
//-0.00207432f, 0.00402547f,
0.00200766f, 0.00106812f, 0.00044566f,-0.00014761f,
-0.00074036f,-0.00129580f,-0.00169464f,-0.00183414f,-0.00164520f,-0.00111129f,
-0.00029199f, 0.00069623f, 0.00168197f, 0.00246922f, 0.00287793f, 0.00277706f,
0.00212434f, 0.00097933f,-0.00049561f,-0.00205243f,-0.00339945f,-0.00424955f,
-0.00438005f,-0.00368304f,-0.00219719f,-0.00011885f, 0.00222062f, 0.00440171f,
0.00598772f, 0.00660803f, 0.00603436f, 0.00424134f, 0.00143235f,-0.00197384f,
-0.00539709f,-0.00818867f,-0.00974422f,-0.00962242f,-0.00764568f,-0.00396213f,
0.00094275f, 0.00629665f, 0.01114674f, 0.01451066f, 0.01555071f, 0.01374059f,
0.00899944f, 0.00176454f,-0.00701380f,-0.01596042f,-0.02344211f,-0.02778959f,
-0.02754621f,-0.02170618f,-0.00990373f, 0.00747576f, 0.02928698f, 0.05372275f,
0.07850988f, 0.10117969f, 0.11937421f, 0.13114808f, 0.13522153f, 0.13114808f,
0.11937421f, 0.10117969f, 0.07850988f, 0.05372275f, 0.02928698f, 0.00747576f,
-0.00990373f,-0.02170618f,-0.02754621f,-0.02778959f,-0.02344211f,-0.01596042f,
-0.00701380f, 0.00176454f, 0.00899944f, 0.01374059f, 0.01555071f, 0.01451066f,
0.01114674f, 0.00629665f, 0.00094275f,-0.00396213f,-0.00764568f,-0.00962242f,
-0.00974422f,-0.00818867f,-0.00539709f,-0.00197384f, 0.00143235f, 0.00424134f,
0.00603436f, 0.00660803f, 0.00598772f, 0.00440171f, 0.00222062f,-0.00011885f,
-0.00219719f,-0.00368304f,-0.00438005f,-0.00424955f,-0.00339945f,-0.00205243f,
-0.00049561f, 0.00097933f, 0.00212434f, 0.00277706f, 0.00287793f, 0.00246922f,
0.00168197f, 0.00069623f,-0.00029199f,-0.00111129f,-0.00164520f,-0.00183414f,
-0.00169464f,-0.00129580f,-0.00074036f,-0.00014761f, 0.00044566f, 0.00106812f,
0.00200766f};
// 0.00402547f,-0.00207432f};
/* FIR filter designed with http://t-filter.appspot.com
* Sampling frequency: 24000 sps
* 0 Hz - 3000 Hz gain = 2 ripple = 0.11 dB
* 6000 Hz - 12000 Hz atten = -62.4 dB
* (At Sampling Frequency=48ksps, double all frequency values) */
const float32_t interpolateFilter1[23] = {
-0.00413402f,-0.01306124f,-0.01106321f, 0.01383359f, 0.04386756f, 0.02731837f,
-0.05470066f,-0.12407408f,-0.04389386f, 0.23355907f, 0.56707488f, 0.71763165f,
0.56707488f, 0.23355907f,-0.04389386f,-0.12407408f,-0.05470066f, 0.02731837f,
0.04386756f, 0.01383359f,-0.01106321f,-0.01306124f,-0.00413402};
/* Linear phase baseband filter
* FIR filter designed with http://t-filter.appspot.com
* Sampling frequency: 24000 Hz
* 0 Hz - 1420 Hz ripple = 0.146 dB
* 1700 Hz - 12000 Hz attenuation = -50.1 dB */
float32_t basebandFilter[199] = {
0.00196058f, 0.00082632f, 0.00085733f, 0.00078043f, 0.00059145f, 0.00030448f,
-0.00004829f,-0.00042015f,-0.00075631f,-0.00100164f,-0.00110987f,-0.00105351f,
-0.00083052f,-0.00046510f,-0.00000746f, 0.00047037f, 0.00089019f, 0.00117401f,
0.00126254f, 0.00112385f, 0.00076287f, 0.00022299f,-0.00041828f,-0.00105968f,
-0.00159130f,-0.00191324f,-0.00195342f,-0.00168166f,-0.00111897f,-0.00033785f,
0.00054658f, 0.00139192f, 0.00205194f, 0.00240019f, 0.00235381f, 0.00189072f,
0.00105796f,-0.00003104f,-0.00121055f,-0.00228720f,-0.00307062f,-0.00340596f,
-0.00320312f,-0.00245657f,-0.00125253f, 0.00023880f, 0.00178631f, 0.00313236f,
0.00403460f, 0.00430822f, 0.00386101f, 0.00271591f, 0.00101544f,-0.00099378f,
-0.00299483f,-0.00464878f,-0.00565026f,-0.00578103f,-0.00495344f,-0.00323470f,
-0.00084708f, 0.00185766f, 0.00444725f, 0.00647565f, 0.00755579f, 0.00742946f,
0.00601997f, 0.00345944f, 0.00008392f,-0.00360622f,-0.00701534f,-0.00954279f,
-0.01068201f,-0.01011191f,-0.00776604f,-0.00386666f, 0.00108417f, 0.00636072f,
0.01110737f, 0.01446572f, 0.01570891f, 0.01437252f, 0.01035602f, 0.00397827f,
-0.00402157f,-0.01254475f,-0.02025120f,-0.02572083f,-0.02763900f,-0.02498240f,
-0.01717994f,-0.00422067f, 0.01329264f, 0.03419240f, 0.05682312f, 0.07923505f,
0.09938512f, 0.11536507f, 0.12562657f, 0.12916328f, 0.12562657f, 0.11536507f,
0.09938512f, 0.07923505f, 0.05682312f, 0.03419240f, 0.01329264f,-0.00422067f,
-0.01717994f,-0.02498240f,-0.02763900f,-0.02572083f,-0.02025120f,-0.01254475f,
-0.00402157f, 0.00397827f, 0.01035602f, 0.01437252f, 0.01570891f, 0.01446572f,
0.01110737f, 0.00636072f, 0.00108417f,-0.00386666f,-0.00776604f,-0.01011191f,
-0.01068201f,-0.00954279f,-0.00701534f,-0.00360622f, 0.00008392f, 0.00345944f,
0.00601997f, 0.00742946f, 0.00755579f, 0.00647565f, 0.00444725f, 0.00185766f,
-0.00084708f,-0.00323470f,-0.00495344f,-0.00578103f,-0.00565026f,-0.00464878f,
-0.00299483f,-0.00099378f, 0.00101544f, 0.00271591f, 0.00386101f, 0.00430822f,
0.00403460f, 0.00313236f, 0.00178631f, 0.00023880f,-0.00125253f,-0.00245657f,
-0.00320312f,-0.00340596f,-0.00307062f,-0.00228720f,-0.00121055f,-0.00003104f,
0.00105796f, 0.00189072f, 0.00235381f, 0.00240019f, 0.00205194f, 0.00139192f,
0.00054658f,-0.00033785f,-0.00111897f,-0.00168166f,-0.00195342f,-0.00191324f,
-0.00159130f,-0.00105968f,-0.00041828f, 0.00022299f, 0.00076287f, 0.00112385f,
0.00126254f, 0.00117401f, 0.00089019f, 0.00047037f,-0.00000746f,-0.00046510f,
-0.00083052f,-0.00105351f,-0.00110987f,-0.00100164f,-0.00075631f,-0.00042015f,
-0.00004829f, 0.00030448f, 0.00059145f, 0.00078043f, 0.00085733f, 0.00082632f,
0.00196058};
}; // end Class
#endif
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