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OpenAudio_ArduinoLibrary/AudioLMSDenoiseNotch_F32.h

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/*
* AudioLMSDenoiseNotch_F32.h
*
* Created: Bob Larkin, January 2022
* Purpose; LMS DeNoise and Auto-notch for audio.
* Assumes floating-point data.
*
* 22 January 2022 copyright (c)Robert Larkin 2022
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice, development notice, and this permission
* notice shall be included in all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
/* *** Notes ***
* The LMS DeNoise is effective for improving the signal-to-noise ratio (S/N)
* when the input S/N is reasonably high. When the signal is "buried" in the noise
* it is much less effective. Thus it is effective as a radio "squelch" for SSB.
*
* The auto-notch is very effective at removing annoying tones when they are
* reasonably strong. Again for radio systems, this can be quite useful.
* The initialization selects whether DeNoise or AutoNotch is used. It makes
* no sense to use both at once as, in a perfect world, that would remove everything.
*
* The LMS algorithm for optimization was first proposed by
* Widrow and Hoff in 1960.
* It has been applied extensively due to its simplicity. The form here
* optimizes the coefficients of a FIR filter to recognize any coherency
* to the input signal. This can be use to reduce non-coherent noise by
* using the FIR filter output. Alternatively, the input signal can be
* subtracted from the FIR filter output to remove coherent signals,
* producing the so called "auto-notch."
*
* This particular write of the denoise and auto-notch traces back to
* Johan Forrer, KC7WW, per September 1994 QEX. From there it was used
* in the DSP-10 project, http://www.janbob.com/electron/dsp10/dsp10.htm
*
* The normalized version of coefficient update is generally best. If it
* is not desired, it can be removed at compile time by commenting out
* "#define LMS_NORMALIZE" below.
*
* Initialization also sets the size of the FIR buffer used to filter signal
* and noise. Small buffers respond to change quickly. Large buffers can work
* on lower audio frequencies. Experiment with this. The FIR buffer is set in
* powers of 2, such as 32, 64 or 128. The maximum value is set at compile
* time by the #define MAX_FIR (default 128).
*
* Initialization sets the decorrelation delay size. If the LMS is preceded by
* a narrow band filter, this delay must be greater. Wide band systems can
* work with less delay. Experiment with this, also. The DELAY buffer size
* can be any value from 2 to MAX DELAY. The maximum value is set at compile
* time by the #define MAX_DELAY (default 16).
*
* This block behaves as a pass-through filter with one input and one output.
*
* There are two parameters that are set in the .ino via the function
* setParameters(float32_t beta, float32_t decay)
* The first, beta determines the rate of convergence of the coefficients.
* This changes the "sound" of the audio and normally is one of a radio's
* front panel adjustments. The second parameter, decay, slowly turns
* the algorithm off when signals are absent. It is normally very
* slightly less than 1.0. This can also change the "sound."
*
* The Teensy 3.6 needs 690 microseconds per 128 block update using a FIR
* buffer size of 32. It needs 1335 microseconds using 64 FIR Buffer.
* Note that the ARM library LMS routines might improve these
* numbers. Those routines use double buffer sizes to remove the
* need for the circular buffering used here. It also uses x4 loop un-wrapping.
* The price is a signifigantly more complex setup involving moving of data
* and the added memory.
*
* Teensy 4.x needs 140 microseconds for 32 FIR word buffer size,
* 270 for 64, and 529 microseconds for 128.
*
* All timing was done with a delay buffer of 4, but this size has
* very little effect, anyway. Normalization was off, also, but
* again, this has a minor effect.
*/
#ifndef _AudioLMSDenoiseNotch_F32_h
#define _AudioLMSDenoiseNotch_F32_h
#include <AudioStream_F32.h>
#include "arm_math.h"
// Default is to use the normalized form of coefficient update
#define LMS_NORMALIZE
#define MAX_FIR 256
#define MAX_DELAY 16
#define DENOISE 1
#define NOTCH 2
class AudioLMSDenoiseNotch_F32 : public AudioStream_F32
{
//GUI: inputs:1, outputs:1 //this line used for automatic generation of GUI node
public:
//constructor
AudioLMSDenoiseNotch_F32(void) : AudioStream_F32(1, inputQueueArray_f32) {};
AudioLMSDenoiseNotch_F32(const AudioSettings_F32 &settings) :
AudioStream_F32(1, inputQueueArray_f32) {};
uint16_t initializeLMS(uint16_t _what, uint16_t _lengthDataF, uint16_t _lengthDataD)
{
what = _what;
if(what != DENOISE && what != NOTCH) what = DENOISE;
lengthDataF = powf(2.0f, log2f(_lengthDataF)+0.000001f); //Make sure a power of 2
lengthDataF = (lengthDataF>MAX_FIR ? MAX_FIR : lengthDataF); // Limit length
kMask = lengthDataF - 1;
lengthDataD = _lengthDataD;
lengthDataD = (lengthDataD>MAX_DELAY ? MAX_DELAY : lengthDataD); // Limit length
#ifdef LMS_NORMALIZE
for(int i=0; i<128; i++) powerNorm[i] = 0.01f;
pNorm = 0.01f * 128.0f;
#endif
return lengthDataF;
}
// If setEnable is false the LMS object update() becomes pass-though.
void enable(bool setEnable) {
if(setEnable) doLMS=true;
else doLMS=false;
}
void setParameters(float32_t _beta, float32_t _decay)
{
beta = _beta;
if(beta>=1.0f) beta = 0.999999f;
if(beta<0.000001) beta = 0.000001f;
decay = _decay;
if(decay>=1.0f) decay = 0.999999f;
if(decay<0.000001) decay = 0.000001f;
}
virtual void update(void);
private:
audio_block_f32_t *inputQueueArray_f32[1]; //memory pointer for the input to this module
uint16_t what = DENOISE; // DENOISE or NOTCH
bool doLMS = false;
float32_t dataD[16]; // Can be made less than 16 by lengthDataD
uint16_t kNextD = 0;
uint16_t kOffsetD = 0;
uint16_t lengthDataD = 4; // Any value, 2 to MAX_DELAY
float32_t coeff[128];
#ifdef LMS_NORMALIZE
float32_t powerNorm[128];
float32_t pNorm = 0.0f;
#endif
// dataF[] is arranged, by added variables kOffset and
// lengthDataF, to be circular. A power-of-2 mask makes it circular.
float32_t dataF[128]; // Can be made less than 128 by lengthDataF
float32_t dataOutF = 0.0f;
uint16_t kOffsetF = 0;
uint16_t lengthDataF = 64;
uint16_t kMask = 63;
float32_t beta = 0.03f;
float32_t decay = 0.995f;
uint16_t numLeak = 0;
};
#endif