/*
* radioVoiceClipper_F32.h
*
* 12 March 2023 (c) copyright Bob Larkin
* But with With much credit to:
* Chip Audette (OpenAudio)
* and of course, to PJRC for the Teensy and Teensy Audio Library
*
* The development of this Voice Clipper was by Bob Larkin, W7PUA, based
* entirely on ideas and suggestions from Dave Hershberger, W9GR.
* Many thanks to Dave. Note that this clipper is is a "real variable"
* version of the Single Sideband CESSB clipper. See the companion
* radioCESSBtransmit_F32.h class which uses all the same principles.
*
* The input signal is a voice (or tones) that will, in general, have
* been compressed in amplitude, keeping the maximum amplitude close to
* 1.0 peak-to-center. For this class, clipping occurs for any input
* greater than 1/gainIn where gainIn comes from the public function
* setGains(). Normally gainIn has a value around 1.5 and so clipping occurs
* for inputs above peak levels of 2/3=0.667. For this level of gaiIn,
* there will be about 3 dB of increase in the average power of the voice
* but still minimal perception of "over-processing."
*
* Internally the audio is clipped at the higher levels and the resulting
* out-of-band distion is low pass filtered. Next, the overshoot that
* occurs with the filter is removed by measuring the overshoot, low-pass
* filtering the overshoot and subtracting it off. All this requires
* care with the timing as all of the filtering steps involve delays.
*
* The compressor2 class in this F32 library is intended to precede this
* class.
*
* NOTE: Do NOT follow this block with any non-linear phase filtering,
* such as IIR. Minimize any linear-phase filtering such as FIR.
* Such activities enhance the overshoots and defeat the purpose of clipping.
*
* An important note: This clipper is suitable for voice modes, such as
* AM or NBFM. Do not use this clipper ahead of a single sideband
* transmitter. That is what the CESSB class is for.
*
* The following reference has information on CESSB, in detail, as well
* as on the use of clippers, similar to this one, in broadcast work:
* 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
*
* Status: Experimental
*
* Inputs: 0 is voice audio input
* Outputs: 0 is clipped voice.
*
* Functions, available during operation:
* void setSampleRate_Hz(float32_t fs_Hz) Allows dynamic sample rate change.
*
* 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 microseconds
* T4.0 For an update of a 128 sample block, measured microseconds
* These times are for a 48 ksps rate.
*
* NOTE: Do NOT follow this block with any non-linear phase filtering,
* such as IIR. Minimize any linear-phase filtering such as FIR.
* Such activities enhance the overshoots and defeat the purpose of clipping.
*/
#ifndef _radioVoiceClipper_f32_h
#define _radioVoiceClipper_f32_h
#include "Arduino.h"
#include "AudioStream_F32.h"
#include "arm_math.h"
#include "mathDSP_F32.h"
#define VC_SAMPLE_RATE_0 0
#define VC_SAMPLE_RATE_11_12 1
#define VC_SAMPLE_RATE_44_50 2
#define VC_SAMPLE_RATE_88_100 3
#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 levelClipper {
float32_t pwr0;
float32_t peak0;
float32_t pwr1;
float32_t peak1;
uint32_t countP; // Number of averaged samples for pwr0.
};
class radioVoiceClipper_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:
radioVoiceClipper_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
}
radioVoiceClipper_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>10900.0f && sample_rate_Hz<12600.0f)
{
// Design point is 12 ksps. No initial decimation. Interpolate
// to 24 ksps for clipping and then decimate back to 12 at the end.
sampleRate = VC_SAMPLE_RATE_11_12;
nW = 128;
nC = 256;
countLevelMax = 10; // About 0.1 sec for 12 ksps
inverseMaxCount = 1.0f/(float32_t)countLevelMax;
arm_fir_init_f32(&firInstInterpolate1I, 23, (float32_t*)interpolateFilter1,
&pStateInterpolate1I[0], nC);
arm_fir_init_f32(&firInstClipperI, 123, (float32_t*)clipperOut,
&pStateClipperI[0], nC);
arm_fir_init_f32(&firInstOShootI, 123, (float32_t*)clipperOut,
&pStateOShootI[0], nC);
}
else if(sample_rate_Hz>43900.0f && sample_rate_Hz<50100.0f)
{
// Design point is 48 ksps
sampleRate = VC_SAMPLE_RATE_44_50;
nW = 32;
nC = 64;
countLevelMax = 37; // About 0.1 sec for 48 ksps
inverseMaxCount = 1.0f/(float32_t)countLevelMax;
arm_fir_decimate_init_f32(&decimateInst, 65, 4,
(float32_t*)decimateFilter48, &pStateDecimate[0], 128);
arm_fir_init_f32(&firInstInterpolate1I, 23, (float32_t*)interpolateFilter1,
&pStateInterpolate1I[0], nC);
arm_fir_init_f32(&firInstClipperI, 123, (float32_t*)clipperOut,
&pStateClipperI[0], nC);
arm_fir_init_f32(&firInstOShootI, 123, (float32_t*)clipperOut,
&pStateOShootI[0], nC);
arm_fir_init_f32(&firInstInterpolate2I, 23, (float32_t*)interpolateFilter1,
&pStateInterpolate2I[0], nC);
}
else if(sample_rate_Hz>88000.0f && sample_rate_Hz<100100.0f)
{
// GET THINGS WORKING AT VC_SAMPLE_RATE_44_50 FIRST AND THEN FIX UP 96 ksps
// Design point is 96 ksps
/* sampleRate = VC_SAMPLE_RATE_88_100; //<<<<<<<<<<<<<<<<<<<<<<FIXUP
nW = 16;
nC = 32;
countLevelMax = 75; // About 0.1 sec for 96 ksps
inverseMaxCount = 1.0f/(float32_t)countLevelMax;
arm_fir_decimate_init_f32 (&decimateInst, 55, 4,
(float32_t*)decimateFilter48, pStateDecimate, 128);
arm_fir_init_f32(&firInstClipper, 199, basebandFilter,
&StateFirClipperF32[0], 128);
*/
}
else
{
// Unsupported sample rate
sampleRate = VC_SAMPLE_RATE_0;
nW = 1;
nC = 1;
}
newLevelDataReady = false;
}
struct levelClipper* getLevels(int what) {
if(what != 0) // 0 leaves a way to get pointer before data is ready
{
levelData.pwr0 = powerSum0/((float32_t)countPower0);
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;
}
virtual void update(void);
private:
void sincos_Z_(float32_t ph);
struct levelClipper levelData;
audio_block_f32_t *inputQueueArray_f32[1];
uint32_t jjj = 0; // Used for diagnostic printing
// Input/Output is at 12, 48 or 96 ksps.
// Clipping and overshoot processing is at 24 ksps.
// Next line is to indicate that setSampleRateHz() has not executed
int sampleRate = VC_SAMPLE_RATE_0;
float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE; // 44.1 ksps
int16_t nW = 32; // 128, 32 or 16
int16_t nC = 64; // 256, 64 or 32
uint16_t block_length = 128;
float32_t pStateDecimate[128 + 65 - 1]; // Goes with CMSIS decimate function
arm_fir_decimate_instance_f32 decimateInst;
// For 12 ksps case, 24 kHz clipper uses 256 points
float32_t pStateInterpolate1I[256 + 23 - 1]; // For interpolate 12 to 24 ksps
arm_fir_instance_f32 firInstInterpolate1I;
float32_t pStateClipperI[256 + 123 - 1]; // Goes with Clipper filter
arm_fir_instance_f32 firInstClipperI; // at 24 ksps
float32_t pStateOShootI[256+123-1];
arm_fir_instance_f32 firInstOShootI;
float32_t pStateInterpolate2I[256 + 23 - 1]; // For interpolate 12 to 24 ksps
arm_fir_instance_f32 firInstInterpolate2I;
float32_t gainIn = 1.0f;
float32_t gainCompensate = 1.4f;
float32_t gainOut = 1.0f; // Does not change Clipping, here for convenience to set out level
// A tiny delay to allow negative time for the previous path
float32_t osEnv[4];
uint16_t indexOsEnv = 4; // 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. Some where around 128 taps works
// but if we make the delay exactly 2^6=64 the delay line is simple
// resulting in a FIR size of 2*64+1=129 taps.
float32_t osDelayI[64];
uint16_t indexOsDelay = 64;
// 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;
/* 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};
/* Filter for outputs of clipper
* Use also overshoot corrector, but might be able to use less terms.
* FIR filter designed with http://t-filter.appspot.com
* Sample frequency: 24000 Hz
* 0 Hz - 2800 Hz ripple = 0.14 dB
* 3200 Hz - 12000 Hz atten = 40.51 dB */
const float32_t clipperOut[123] = {
-0.003947255f, 0.001759588f, 0.002221444f, 0.002407244f, 0.001833343f, 0.000524622f,
-0.000946260f,-0.001768428f,-0.001395297f, 0.000055916f, 0.001779024f, 0.002694998f,
0.002099736f, 0.000157764f,-0.002092190f,-0.003282801f,-0.002542927f,-0.000116969f,
0.002694319f, 0.004153363f, 0.003197589f, 0.000143560f,-0.003346600f,-0.005148200f,
-0.003947437f,-0.000152425f, 0.004166345f, 0.006378882f, 0.004871469f, 0.000164557f,
-0.005173898f,-0.007896395f,-0.006014470f,-0.000173552f, 0.006447615f, 0.009828080f,
0.007480359f, 0.000184482f,-0.008116957f,-0.012379161f,-0.009436712f,-0.000194737f,
0.010412610f, 0.015941971f, 0.012213107f, 0.000200845f,-0.013823966f,-0.021360759f,
-0.016552097f,-0.000205707f, 0.019544260f, 0.030836344f, 0.024523278f, 0.000211298f,
-0.031509151f,-0.052450055f,-0.044811840f,-0.000214078f, 0.074661107f, 0.158953216f,
0.225159581f, 0.250214862f, 0.225159581f, 0.158953216f, 0.074661107f,-0.000214078f,
-0.044811840f,-0.052450055f,-0.031509151f, 0.000211298f, 0.024523278f, 0.030836344f,
0.019544260f,-0.000205707f,-0.016552097f,-0.021360759f,-0.013823966f, 0.000200845f,
0.012213107f, 0.015941971f, 0.010412610f,-0.000194737f,-0.009436712f,-0.012379161f,
-0.008116957f, 0.000184482f, 0.007480359f, 0.009828080f, 0.006447615f,-0.000173552f,
-0.006014470f,-0.007896395f,-0.005173898f, 0.000164557f, 0.004871469f, 0.006378882f,
0.004166345f,-0.000152425f,-0.003947437f,-0.005148200f,-0.003346600f, 0.000143560f,
0.003197589f, 0.004153363f, 0.002694319f,-0.000116969f,-0.002542927f,-0.003282801f,
-0.002092190f, 0.000157764f, 0.002099736f, 0.002694998f, 0.001779024f, 0.000055916f,
-0.001395297f,-0.001768428f,-0.000946260f, 0.000524622f, 0.001833343f, 0.002407244f,
0.002221444f, 0.001759588f,-0.003947255f};
/* 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};
}; // end Class
#endif