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OpenAudio_ArduinoLibrary/AudioLMSDenoiseNotch_F32.cpp

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3.8 KiB

/*
* AudioLMSDenoiseNotch_F32
*
* 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 funding 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.
*/
#include "AudioStream_F32.h"
#include "AudioLMSDenoiseNotch_F32.h"
void AudioLMSDenoiseNotch_F32::update(void)
{
uint16_t j, k;
float32_t blockDataIn, error, firOut;
audio_block_f32_t *block;
block = AudioStream_F32::receiveWritable_f32();
if (!block) return;
if(!doLMS)
{
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
return;
}
audio_block_f32_t *blockOut;
blockOut = AudioStream_F32::allocate_f32(); // Output block
if (!blockOut)
{
AudioStream_F32::transmit(block);
AudioStream_F32::release(block);
return;
}
for(int i=0; i<128; i++)
{
blockDataIn = block->data[i];
// Leakage on one coefficient
coeff[numLeak] *= decay; // Decay one coefficient
if(++numLeak >= lengthDataF) // Wrap around, if needed
numLeak = 0;
// Circular delay line to find correlated components
dataD[kNextD] = blockDataIn; // Get a new data point from block
#ifdef LMS_NORMALIZE
powerNorm[i] = blockDataIn*blockDataIn;
pNorm += powerNorm[i];
if(i==127)
pNorm -= powerNorm[0];
else
pNorm -= powerNorm[i+1];
#endif
if(++kNextD >= lengthDataD) // Next spot in delay line
kNextD = 0;
// Update the FIR.
dataF[kOffsetF] = dataD[kNextD]; // Input FIR is output Delay
firOut = 0.0f;
for(j=0; j<lengthDataF; j++) // Over all coefficients
{
k = (j + kOffsetF) & kMask; // Data circular buffer
firOut +=coeff[j]*dataF[k];
}
// Compute the error, the difference between the data point
// just received and the FIR output.
error = blockDataIn - firOut;
// Update the coefficients
#ifdef LMS_NORMALIZE
float32_t kcf = error*beta/pNorm;
#else
float32_t kcf = error*beta;
#endif
for(j=0; j<lengthDataF; j++)
{
k = (j + kOffsetF) & kMask;
coeff[j] = coeff[j] + kcf*dataF[k];
}
// Move to next positions in circular data buffer via kOffsetF
if(++kOffsetF >= lengthDataF)
kOffsetF = 0; // Wrap the FIR circular buffer
// fir out to output block
if(what == DENOISE)
blockOut->data[i] = firOut;
else
blockOut->data[i] = error; // Auto-Notch
}
//transmit the block and be done
AudioStream_F32::transmit(blockOut);
AudioStream_F32::release(block);
AudioStream_F32::release(blockOut);
}