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315 lines
10 KiB
315 lines
10 KiB
/* ------------------------------------------------------------------------------------
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AudioAlignLR_F32.h
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Function: Waits for CODEC startup and measures L-R time alignment errors.
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Automatically corrects delay errors. Requires extra control to disable
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analog audio path from DAC Q to both ADC I and ADC Q.
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See the following for more background on the problem being solved:
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https://forum.pjrc.com/threads/42336-Reset-audio-board-codec-SGTL5000-in-realtime-processing/page2
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https://forum.pjrc.com/threads/57362-AudioSDR-A-single-Audio-block-SDR-(software-defined-radio)-processor-demodulator/page3
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"Twin Peaks", or TP, comes from Frank, DD4WH who found, in hsi Convolution SDR, double spectral responses when
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the L and R channels were out of time alignment.
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Author: Bob Larkin W7PUA
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Date: 28 Feb 2022
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Input: data_I, data_Q
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Outputs: dataOut_I, dataOut_Q, dataTransmit_Q
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Update() time: About 12 microseconds for a T4.x.
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Copyright (c) 2022 Robert Larkin
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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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The above copyright notice and this permission notice shall be included in all
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copies or substantial portions of the Software.
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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 THE
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SOFTWARE.
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------------------------------------------------------------------------------- */
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//REF:
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// Manual I2S codec Phase Correction Utility by Ron Carr, Updated for F32, K7MDL
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// Reference: [URL]https://forum.pjrc.com/threads/57362-AudioSDR-A-single-Audio-block-SDR-(software-defined-radio)-processor-demodulator?p=263048&viewfull=1#post263048[/URL]
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//
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// The variable, currentTPinfo.TPsignalHardware, changes source of
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// the signal for going to L&R for correlation.
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// If TP_SIGNAL_CODEC there needs to be a switchable hardware link between
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// the DAC right channel and both of ADC right and ADC left.
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// If TP_SIGNAL_IO_PIN the INO must update a roughly fs/4
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// square wave using an i/o pin. That signal is generated by
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// the .INO as it is not possible in an audio object.
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// Default is TP_SIGNAL_CODEC.
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//
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// BETA NOTE That calls and operations may change. BETA
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// Rev 12 Mar 2022 Added normalized decisions to eliminate need for a threshold.
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/* This is best placed in the setup() part of the INO. A signal needs to be generated
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* and this would look like:
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uint32_t timeSquareWave = 0;
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#define PIN_FOR_TP 2
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pinMode (PIN_FOR_TP, OUTPUT); // Digital output pin
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pData = TwinPeak.read(); // Base data to check error
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while (pData->TPerror < 0 && millis()-tMillis < 2000) // with timeout
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{
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if(micros()-timeSquareWave >= 45 && pData->TPstate==TP_MEASURE)
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{
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timeSquareWave = micros();
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squareWave = squareWave^1;
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digitalWrite(PIN_FOR_TP, squareWave);
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}
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pData = TwinPeak.read();
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}
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digitalWrite(PIN_FOR_TP, 0); // Set pin to zero
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* where the 45 microseconds needs to be adjusted for the sample rate of the Codec
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*/
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#ifndef audio_align_lr_f32_h_
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#define audio_align_lr_f32_h_
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#include "AudioStream_F32.h"
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#include "Arduino.h"
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#define TP_IDLE 0
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#define TP_MEASURE 1
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#define TP_RUN 2
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#define TP_SIGNAL_CODEC 0
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#define TP_SIGNAL_IO_PIN 1
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#define ERROR_TP_EARLY -2
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#define ERROR_TP_BAD_DATA -1
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#define ERROR_TP_NONE 0
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// Needs to be available to INO and AudioAlignLR_F32 also.
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struct TPinfo{
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uint16_t TPstate;
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uint32_t nMeas;
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float32_t xcVal[4]; // I-Q cross-correlation sums
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float32_t xNorm;
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int16_t neededShift;
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int16_t TPerror;
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uint16_t TPsignalHardware;
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};
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class AudioAlignLR_F32 : public AudioStream_F32
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{
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//GUI: inputs:2, outputs:3 //this line used for automatic generation of GUI node
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//GUI: shortName: AlignLR
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public:
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AudioAlignLR_F32(uint16_t _hardware, uint16_t _controlPinNumber, bool _controlPinInvert):
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AudioStream_F32(2,inputQueueArray)
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{
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currentTPinfo.TPsignalHardware = _hardware;
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controlPinNumber = _controlPinNumber;
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controlPinInvert = _controlPinInvert;
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sample_rate_Hz = AUDIO_SAMPLE_RATE;
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// Changes would be needed for block data sizes other than 128
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block_size = AUDIO_BLOCK_SAMPLES;
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initTP();
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}
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AudioAlignLR_F32
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(uint16_t _hardware, uint16_t _controlPinNumber, bool _controlPinInvert, const AudioSettings_F32 &settings):
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AudioStream_F32(2,inputQueueArray)
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{
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currentTPinfo.TPsignalHardware = _hardware;
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controlPinNumber = _controlPinNumber;
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controlPinInvert = _controlPinInvert;
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sample_rate_Hz = settings.sample_rate_Hz;
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block_size = settings.audio_block_samples;
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initTP();
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}
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void initTP(void) { // Common calls for constructors, not INO's
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currentTPinfo.TPstate = TP_IDLE;
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currentTPinfo.neededShift = 0;
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currentTPinfo.TPerror = ERROR_TP_EARLY;
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needOneMore = false;
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// Initialize FIR instance (ARM DSP Math Library)
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arm_fir_init_f32(&fir_instL, 101, firBP, &StateF32L[0], block_size);
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arm_fir_init_f32(&fir_instR, 101, firBP, &StateF32R[0], block_size);
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}
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// Returns all the status info, available anytime
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TPinfo *read(void) {
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return ¤tTPinfo;
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}
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void stateAlignLR(int _TPstate) {
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currentTPinfo.TPstate = _TPstate;
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if(currentTPinfo.TPstate==TP_MEASURE)
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{
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currentTPinfo.nMeas = 0; // Timing for the DAC & ADC delays
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currentTPinfo.neededShift = 0;
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currentTPinfo.TPerror = ERROR_TP_EARLY;
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needOneMore = true;
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if(currentTPinfo.TPsignalHardware==TP_SIGNAL_CODEC)
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{
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pinMode(controlPinNumber, OUTPUT); // Turn on special audio path
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digitalWrite(controlPinNumber, 1 & (uint16_t)controlPinInvert);
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}
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}
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}
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/*
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void setThreshold(float32_t _TPthreshold) {
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//TPthreshold = _TPthreshold;
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Serial.println("ERROR: Threshold is no longer used. 12 Mar 2022");
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}
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*/
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// Do we need this after experiments are done? <<<<<<<<<<<<<<<<<<<<<<
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void setLRfilter(bool _useLRfilter) {
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useLRfilter = _useLRfilter;
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}
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virtual void update(void);
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private:
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audio_block_f32_t *inputQueueArray[2];
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float32_t sample_rate_Hz = AUDIO_SAMPLE_RATE;
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uint16_t controlPinNumber = 0;
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bool controlPinInvert = true;
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uint16_t block_size = 128;
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bool needOneMore = false;
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// float32_t TPthreshold = 1.0f;
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float32_t TPextraL = 0.0f;
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float32_t TPextraR = 0.0f;
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TPinfo currentTPinfo;
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bool outputFlag = false;
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uint16_t count = 0;
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bool useLRfilter = true;
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// ARM DSP Math library filter instances
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arm_fir_instance_f32 fir_instL;
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float32_t StateF32L[229];
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arm_fir_instance_f32 fir_instR;
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float32_t StateF32R[229];
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/* Sampling frequency: 48000 Hz
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* 0 Hz - 10100 Hz attenuation = 39.7 dB
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* 10800 Hz - 13200 Hz ripple = 1.2 dB
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* 13900 Hz - 24000 Hz attenuation = 39.7 dB
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* NOTE: This will scale with sample rate and stay withpassband at fs/4.
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*/
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float32_t firBP[101] = {
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0.0010784874240328994,
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-1.8511940285394994e-15,
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0.007173020540888694,
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-2.446906631093228e-15,
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-0.007077486809913102,
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-2.3412063121164944e-15,
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0.009011618946946028,
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-2.5923206852409818e-15,
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-0.010712293622384057,
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-2.3964501699408722e-15,
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0.011482971456733491,
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-2.666527614775755e-15,
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-0.010885141080842342,
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-2.4082964205096483e-15,
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0.008647733991415402,
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-2.4633438337819814e-15,
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-0.004721228640000285,
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-2.479178552837757e-15,
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-0.0006641071730367476,
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-2.5286128045656088e-15,
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0.006973730987499941,
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-2.477532927164074e-15,
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-0.01339871502510467,
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-2.431267551488873e-15,
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0.01892093480750003,
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-2.4098508014460433e-15,
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-0.022430541924076213,
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-2.405917616535165e-15,
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0.02285350193041076,
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-2.509429940385098e-15,
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-0.01935677493053742,
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-2.483893222086823e-15,
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0.01141749520487472,
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-2.601900577555639e-15,
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0.000985366589667628,
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-2.5322636675246416e-15,
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-0.01735448964139692,
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-2.592937123951427e-15,
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0.03668097751462642,
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-2.617655860016599e-15,
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-0.057518688568419914,
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-2.3833597705424817e-15,
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0.07812082879163866,
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-2.532991478487954e-15,
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-0.09663455293743198,
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-2.412807130934187e-15,
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0.11132156511296272,
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-2.5145289109982724e-15,
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-0.12075435023719062,
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-2.4314403797559724e-15,
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0.12400702102359241,
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-2.4314403797559724e-15,
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-0.12075435023719062,
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-2.5145289109982724e-15,
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0.11132156511296272,
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-2.412807130934187e-15,
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-0.09663455293743198,
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-2.532991478487954e-15,
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0.07812082879163866,
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-2.3833597705424817e-15,
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-0.057518688568419914,
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-2.617655860016599e-15,
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0.03668097751462642,
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-2.592937123951427e-15,
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-0.01735448964139692,
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-2.5322636675246416e-15,
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0.000985366589667628,
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-2.601900577555639e-15,
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0.01141749520487472,
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-2.483893222086823e-15,
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-0.01935677493053742,
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-2.509429940385098e-15,
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0.02285350193041076,
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-2.405917616535165e-15,
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-0.022430541924076213,
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-2.4098508014460433e-15,
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0.01892093480750003,
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-2.431267551488873e-15,
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-0.01339871502510467,
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-2.477532927164074e-15,
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0.006973730987499941,
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-2.5286128045656088e-15,
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-0.0006641071730367476,
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-2.479178552837757e-15,
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-0.004721228640000285,
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-2.4633438337819814e-15,
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0.008647733991415402,
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-2.4082964205096483e-15,
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-0.010885141080842342,
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-2.666527614775755e-15,
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0.011482971456733491,
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-2.3964501699408722e-15,
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-0.010712293622384057,
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-2.5923206852409818e-15,
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0.009011618946946028,
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-2.3412063121164944e-15,
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-0.007077486809913102,
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-2.446906631093228e-15,
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0.007173020540888694,
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-1.8511940285394994e-15,
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0.0010784874240328994};
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};
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#endif
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