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Resampler.h
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Resampler.h
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/* Audio Library for Teensy 3.X
* Copyright (c) 2019, Paul Stoffregen, paul@pjrc.com
*
* Development of this audio library was funded by PJRC.COM, LLC by sales of
* Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
* open source software by purchasing Teensy or other PJRC products.
*
* 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.
*/
/*
by Alexander Walch
*/
#ifndef resampler_h_
#define resampler_h_
#include <Arduino.h> // github.com/PaulStoffregen/cores/blob/master/teensy4/Arduino.h
//#define DEBUG_RESAMPLER //activates debug output
#define MAX_FILTER_SAMPLES 40961 //=1024*20 +1
#define NO_EXACT_KAISER_SAMPLES 1025
#define MAX_HALF_FILTER_LENGTH 80
#define MAX_NO_CHANNELS 8
class Resampler {
public:
struct StepAdaptionParameters {
StepAdaptionParameters(){}
double alpha =0.2; //exponential smoothing parameter
double maxAdaption = 0.01; //maximum relative allowed adaption of resampler step 0.01 = 1%
double kp= 0.6;
double ki=0.00012;
double kd= 1.8;
};
Resampler(float attenuation=100, int32_t minHalfFilterLength=20, int32_t maxHalfFilterLength=80, StepAdaptionParameters settings=StepAdaptionParameters());
void reset();
///@param attenuation target attenuation [dB] of the anti-aliasing filter. Only used if newFs<fs. The attenuation can't be reached if the needed filter length exceeds 2*MAX_FILTER_SAMPLES+1
///@param minHalfFilterLength If newFs >= fs, the filter length of the resampling filter is 2*minHalfFilterLength+1. If fs y newFs the filter is maybe longer to reach the desired attenuation
void configure(float fs, float newFs);
///@param input0 first input array/ channel
///@param input1 second input array/ channel
///@param inputLength length of each input array
///@param processedLength number of samples of the input that were resampled to fill the output array
///@param output0 first output array/ channel
///@param output1 second output array/ channel
///@param outputLength length of each output array
///@param outputCount number of samples of each output array, that were filled with data
void resample(float* input0, float* input1, uint16_t inputLength, uint16_t& processedLength, float* output0, float* output1,uint16_t outputLength, uint16_t& outputCount);
bool addToSampleDiff(double diff);
double getXPos() const;
double getStep() const;
void addToPos(double val);
void fixStep();
bool initialized() const;
double getAttenuation() const;
int32_t getHalfFilterLength() const;
//resampling NOCHANNELS channels. Performance is increased a lot if the number of channels is known at compile time -> the number of channels is a template argument
template <uint8_t NOCHANNELS>
inline void resample(float** inputs, uint16_t inputLength, uint16_t& processedLength, float** outputs, uint16_t outputLength, uint16_t& outputCount){
outputCount=0;
int32_t successorIndex=(int32_t)(ceil(_cPos)); //negative number -> currently the _buffer0 of the last iteration is used
float* ip[NOCHANNELS];
float* fPtr;
float si0[NOCHANNELS];
float* si0Ptr;
float si1[NOCHANNELS];
float* si1Ptr;
while (floor(_cPos + _halfFilterLength) < inputLength && outputCount < outputLength){
float dist=successorIndex-_cPos;
float distScaled=dist*_overSamplingFactor;
int32_t rightIndex=abs((int32_t)(ceilf(distScaled))-_overSamplingFactor*_halfFilterLength);
const int32_t indexData=successorIndex-_halfFilterLength;
if (indexData>=0){
for (uint8_t i =0; i< NOCHANNELS; i++){
ip[i]=inputs[i]+indexData;
}
}
else {
for (uint8_t i =0; i< NOCHANNELS; i++){
ip[i]=_buffer[i]+indexData+_filterLength;
}
}
fPtr=filter+rightIndex;
memset(si0, 0, NOCHANNELS*sizeof(float));
if (rightIndex==_overSamplingFactor*_halfFilterLength){
si1Ptr=si1;
for (uint8_t i=0; i< NOCHANNELS; i++){
*(si1Ptr++)=*ip[i]++**fPtr;
}
fPtr-=_overSamplingFactor;
rightIndex=(int32_t)(ceilf(distScaled))+_overSamplingFactor; //needed below
}
else {
memset(si1, 0, NOCHANNELS*sizeof(float));
rightIndex=(int32_t)(ceilf(distScaled)); //needed below
}
for (uint16_t i =0 ; i<_halfFilterLength; i++){
if(ip[0]==_endOfBuffer[0]){
for (uint8_t i =0; i< NOCHANNELS; i++){
ip[i]=inputs[i];
}
}
const float fPtrSucc=*(fPtr+1);
si0Ptr=si0;
si1Ptr=si1;
for (uint8_t i =0; i< NOCHANNELS; i++){
*(si0Ptr++)+=*ip[i]*fPtrSucc;
*(si1Ptr++)+=*ip[i]**fPtr;
++ip[i];
}
fPtr-=_overSamplingFactor;
}
fPtr=filter+rightIndex-1;
for (uint16_t i =0 ; i<_halfFilterLength; i++){
if(ip[0]==_endOfBuffer[0]){
for (uint8_t i =0; i< NOCHANNELS; i++){
ip[i]=inputs[i];
}
}
const float fPtrSucc=*(fPtr+1);
si0Ptr=si0;
si1Ptr=si1;
for (uint8_t i =0; i< NOCHANNELS; i++){
*(si0Ptr++)+=*ip[i]**fPtr;
*(si1Ptr++)+=*ip[i]*fPtrSucc;
++ip[i];
}
fPtr+=_overSamplingFactor;
}
const float w0=ceilf(distScaled)-distScaled;
const float w1=1.0f-w0;
si0Ptr=si0;
si1Ptr=si1;
for (uint8_t i =0; i< NOCHANNELS; i++){
*outputs[i]++=*(si0Ptr++)*w0 + *(si1Ptr++)*w1;
}
outputCount++;
_cPos+=_stepAdapted;
while (_cPos >successorIndex){
successorIndex++;
}
}
if(outputCount < outputLength){
//ouput vector not full -> we ran out of input samples
processedLength=inputLength;
}
else{
processedLength=min(inputLength, (int16_t)floor(_cPos + _halfFilterLength));
}
//fill _buffer
const int32_t indexData=processedLength-_filterLength;
if (indexData>=0){
const unsigned long long bytesToCopy= _filterLength*sizeof(float);
float** inPtr=inputs;
for (uint8_t i =0; i< NOCHANNELS; i++){
memcpy((void *)_buffer[i], (void *)((*inPtr)+indexData), bytesToCopy);
++inPtr;
}
}
else {
float** inPtr=inputs;
for (uint8_t i =0; i< NOCHANNELS; i++){
float* b=_buffer[i];
float* ip=b+indexData+_filterLength;
for (uint16_t j =0; j< _filterLength; j++){
if(ip==_endOfBuffer[i]){
ip=*inPtr;
}
*b++ = *ip++;
}
++inPtr;
}
}
_cPos-=processedLength;
if (_cPos < -_halfFilterLength){
_cPos=-_halfFilterLength;
}
}
private:
void getKaiserExact(double beta);
void setKaiserWindow(double beta, int32_t noSamples);
void setFilter(int32_t halfFiltLength,int32_t overSampling, double cutOffFrequ, double kaiserBeta);
float filter[MAX_FILTER_SAMPLES];
double kaiserWindowSamples[NO_EXACT_KAISER_SAMPLES];
double tempRes[NO_EXACT_KAISER_SAMPLES-1];
double kaiserWindowXsq[NO_EXACT_KAISER_SAMPLES-1];
float _buffer[MAX_NO_CHANNELS][MAX_HALF_FILTER_LENGTH*2];
float* _endOfBuffer[MAX_NO_CHANNELS];
int32_t _minHalfFilterLength;
int32_t _maxHalfFilterLength;
int32_t _overSamplingFactor;
int32_t _halfFilterLength;
int32_t _filterLength;
bool _initialized=false;
const double _settledThrs = 1e-6;
StepAdaptionParameters _settings;
double _configuredStep;
double _step;
double _stepAdapted;
double _cPos;
double _sum;
double _oldDiffs[2];
double _attenuation=0;
float _targetAttenuation=100;
};
#endif