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/*
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* AudioLMSDenoiseNotch_F32
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*
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* 22 January 2022 copyright (c)Robert Larkin 2022
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy
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* of this software and associated documentation files (the "Software"), to deal
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* in the Software without restriction, including without limitation the rights
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* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
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* copies of the Software, and to permit persons to whom the Software is
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* furnished to do so, subject to the following conditions:
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*
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* The above copyright notice, development funding notice, and this permission
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* notice shall be included in all copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
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* THE SOFTWARE.
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*/
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#include "AudioStream_F32.h"
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#include "AudioLMSDenoiseNotch_F32.h"
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void AudioLMSDenoiseNotch_F32::update(void)
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{
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uint16_t j, k;
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float32_t blockDataIn, error, firOut;
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audio_block_f32_t *block;
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block = AudioStream_F32::receiveWritable_f32();
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if (!block) return;
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if(!doLMS)
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{
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AudioStream_F32::transmit(block);
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AudioStream_F32::release(block);
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return;
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}
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audio_block_f32_t *blockOut;
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blockOut = AudioStream_F32::allocate_f32(); // Output block
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if (!blockOut)
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{
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AudioStream_F32::transmit(block);
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AudioStream_F32::release(block);
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return;
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}
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for(int i=0; i<128; i++)
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{
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blockDataIn = block->data[i];
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// Leakage on one coefficient
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coeff[numLeak] *= decay; // Decay one coefficient
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if(++numLeak >= lengthDataF) // Wrap around, if needed
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numLeak = 0;
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// Circular delay line to find correlated components
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dataD[kNextD] = blockDataIn; // Get a new data point from block
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#ifdef LMS_NORMALIZE
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powerNorm[i] = blockDataIn*blockDataIn;
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pNorm += powerNorm[i];
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if(i==127)
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pNorm -= powerNorm[0];
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else
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pNorm -= powerNorm[i+1];
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#endif
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if(++kNextD >= lengthDataD) // Next spot in delay line
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kNextD = 0;
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// Update the FIR.
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dataF[kOffsetF] = dataD[kNextD]; // Input FIR is output Delay
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firOut = 0.0f;
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for(j=0; j<lengthDataF; j++) // Over all coefficients
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{
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k = (j + kOffsetF) & kMask; // Data circular buffer
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firOut +=coeff[j]*dataF[k];
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}
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// Compute the error, the difference between the data point
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// just received and the FIR output.
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error = blockDataIn - firOut;
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// Update the coefficients
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#ifdef LMS_NORMALIZE
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float32_t kcf = error*beta/pNorm;
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#else
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float32_t kcf = error*beta;
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#endif
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for(j=0; j<lengthDataF; j++)
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{
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k = (j + kOffsetF) & kMask;
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coeff[j] = coeff[j] + kcf*dataF[k];
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}
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// Move to next positions in circular data buffer via kOffsetF
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if(++kOffsetF >= lengthDataF)
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kOffsetF = 0; // Wrap the FIR circular buffer
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// fir out to output block
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if(what == DENOISE)
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blockOut->data[i] = firOut;
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else
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blockOut->data[i] = error; // Auto-Notch
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}
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//transmit the block and be done
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AudioStream_F32::transmit(blockOut);
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AudioStream_F32::release(block);
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AudioStream_F32::release(blockOut);
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}
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