Merge pull request #358 from alex6679/master

using the spdif configuration of AudioOutputSPDIF3 and more

Resampler.cpp

@@ -30,7 +30,10 @@
 #include "Resampler.h"
 #include <math.h>
 
-Resampler::Resampler(StepAdaptionParameters settings){
+Resampler::Resampler(float attenuation, int32_t minHalfFilterLength, int32_t maxHalfFilterLength, StepAdaptionParameters settings): _targetAttenuation(attenuation)
+{
+	_maxHalfFilterLength=max(1, min(MAX_HALF_FILTER_LENGTH, maxHalfFilterLength));
+	_minHalfFilterLength=max(1, min(maxHalfFilterLength, minHalfFilterLength));
 #ifdef DEBUG_RESAMPLER
 	while (!Serial);
 #endif
@@ -129,16 +132,25 @@ void Resampler::setFilter(int32_t halfFiltLength,int32_t overSampling, float cut
 double Resampler::getStep() const {
     return  _stepAdapted;
 }
+double Resampler::getAttenuation() const {
+    return  _attenuation;
+}
+int32_t Resampler::getHalfFilterLength() const{
+	return  _halfFilterLength;
+}
 void Resampler::reset(){
     _initialized=false;
 }
-void Resampler::configure(float fs, float newFs, float attenuation, int32_t minHalfFilterLength){
+void Resampler::configure(float fs, float newFs){
     // Serial.print("configure, fs: ");
     // Serial.println(fs);
     if (fs<=0. || newFs <=0.){
+		_attenuation=0;
+		_halfFilterLength=0;
         _initialized=false;
         return;
     }
+	_attenuation=_targetAttenuation;
     _step=(double)fs/newFs;
     _configuredStep=_step;
     _stepAdapted=_step;
@@ -146,15 +158,16 @@ void Resampler::configure(float fs, float newFs, float attenuation, int32_t minH
     _oldDiffs[0]=0.;
     _oldDiffs[1]=0.;
     for (uint8_t i =0; i< MAX_NO_CHANNELS; i++){
-        memset(_buffer[i], 0, sizeof(float)*MAX_HALF_FILTER_LENGTH*2);
+        memset(_buffer[i], 0, sizeof(float)*_maxHalfFilterLength*2);
     }
 
     float cutOffFrequ, kaiserBeta;
     _overSamplingFactor=1024;
     if (fs <= newFs){
+		_attenuation=0;
         cutOffFrequ=1.;
         kaiserBeta=10;
-        _halfFilterLength=minHalfFilterLength;
+        _halfFilterLength=min(_minHalfFilterLength,_maxHalfFilterLength);
     }
     else{
         cutOffFrequ=newFs/fs;
@@ -163,38 +176,23 @@ void Resampler::configure(float fs, float newFs, float attenuation, int32_t minH
         Serial.print("b: ");
         Serial.println(b);
 #endif
-        double hfl=(int32_t)((attenuation-8)/(2.*2.285*TWO_PI*b)+0.5);
-        if (hfl >= minHalfFilterLength && hfl <= MAX_HALF_FILTER_LENGTH){
+        double hfl=(int32_t)((_attenuation-8)/(2.*2.285*TWO_PI*b)+0.5);
+        if (hfl >= _minHalfFilterLength && hfl <= _maxHalfFilterLength){
             _halfFilterLength=hfl;
-#ifdef DEBUG_RESAMPLER
-            Serial.print("Attenuation: ");
-#endif
         }
-        else if (hfl < minHalfFilterLength){
-            _halfFilterLength=minHalfFilterLength;
-            attenuation=((2*_halfFilterLength+1)-1)*(2.285*TWO_PI*b)+8;            
-#ifdef DEBUG_RESAMPLER
-            Serial.println("Resmapler: sinc filter length increased");
-            Serial.print("Attenuation increased to ");
-#endif
+        else if (hfl < _minHalfFilterLength){
+            _halfFilterLength=_minHalfFilterLength;
+            _attenuation=((2*_halfFilterLength+1)-1)*(2.285*TWO_PI*b)+8;            
         }
         else{
-            _halfFilterLength=MAX_HALF_FILTER_LENGTH;
-            attenuation=((2*_halfFilterLength+1)-1)*(2.285*TWO_PI*b)+8;
-#ifdef DEBUG_RESAMPLER
-            Serial.println("Resmapler: needed sinc filter length too long");
-            Serial.print("Attenuation decreased to ");
-#endif
+            _halfFilterLength=_maxHalfFilterLength;
+            _attenuation=((2*_halfFilterLength+1)-1)*(2.285*TWO_PI*b)+8;
         }
-#ifdef DEBUG_RESAMPLER
-        Serial.print(attenuation);
-        Serial.println("dB");
-#endif
-        if (attenuation>50.){
-            kaiserBeta=0.1102*(attenuation-8.7);
+        if (_attenuation>50.){
+            kaiserBeta=0.1102*(_attenuation-8.7);
         }
-        else if (21<=attenuation && attenuation<=50){
-            kaiserBeta=0.5842*(float)pow(attenuation-21.,0.4)+0.07886*(attenuation-21.);
+        else if (21<=_attenuation && _attenuation<=50){
+            kaiserBeta=0.5842*(float)pow(_attenuation-21.,0.4)+0.07886*(_attenuation-21.);
         }
         else{
             kaiserBeta=0.;

Resampler.h

@@ -49,11 +49,11 @@ class Resampler {
             double ki=0.00012;
             double kd= 1.8;
         };
-        Resampler(StepAdaptionParameters settings=StepAdaptionParameters());
+        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, float attenuation=100, int32_t minHalfFilterLength=20);
+        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
@@ -69,6 +69,8 @@ class Resampler {
         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>
@@ -205,6 +207,8 @@ class Resampler {
         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;     
@@ -218,6 +222,9 @@ class Resampler {
         double _cPos;
         double _sum;
         double _oldDiffs[2]; 
+		
+		double _attenuation=0;
+		float _targetAttenuation=100;
 };
 
 #endif

async_input_spdif3.cpp

@@ -29,6 +29,8 @@
 #if defined(__IMXRT1052__) || defined(__IMXRT1062__)
 
 #include "async_input_spdif3.h"
+#include "output_spdif3.h"
+
 #include "biquad.h"
 #include <utility/imxrt_hw.h>
 //Parameters
@@ -36,8 +38,8 @@ namespace {
 	#define SPDIF_RX_BUFFER_LENGTH AUDIO_BLOCK_SAMPLES
 	const int32_t bufferLength=8*AUDIO_BLOCK_SAMPLES;
 	const uint16_t noSamplerPerIsr=SPDIF_RX_BUFFER_LENGTH/4;
+	const float toFloatAudio= 1.f/pow(2., 23.);
 }
-volatile bool AsyncAudioInputSPDIF3::resetResampler=true;
 
 #ifdef DEBUG_SPDIF_IN
 volatile bool AsyncAudioInputSPDIF3::bufferOverflow=false;
@@ -53,8 +55,6 @@ static float bufferL[bufferLength];
 volatile int32_t AsyncAudioInputSPDIF3::buffer_offset = 0;	// read by resample/ written in spdif input isr -> copied at the beginning of 'resmaple' protected by __disable_irq() in resample
 int32_t AsyncAudioInputSPDIF3::resample_offset = 0; // read/written by resample/ read in spdif input isr -> no protection needed?
 
-volatile bool AsyncAudioInputSPDIF3::lockChanged=false;
-volatile bool AsyncAudioInputSPDIF3::locked=false;
 DMAChannel AsyncAudioInputSPDIF3::dma(false);
 
 AsyncAudioInputSPDIF3::~AsyncAudioInputSPDIF3(){
@@ -64,10 +64,11 @@ AsyncAudioInputSPDIF3::~AsyncAudioInputSPDIF3(){
 	delete quantizer[1];
 }
 
-PROGMEM
-AsyncAudioInputSPDIF3::AsyncAudioInputSPDIF3(bool dither, bool noiseshaping,float attenuation, int32_t minHalfFilterLength) : AudioStream(0, NULL) {
-	_attenuation=attenuation;
-	_minHalfFilterLength=minHalfFilterLength;
+FLASHMEM
+AsyncAudioInputSPDIF3::AsyncAudioInputSPDIF3(bool dither, bool noiseshaping,float attenuation, int32_t minHalfFilterLength, int32_t maxHalfFilterLength):
+	AudioStream(0, NULL),
+	_resampler(attenuation, minHalfFilterLength, maxHalfFilterLength)
+	{
 	const float factor = powf(2, 15)-1.f; // to 16 bit audio
 	quantizer[0]=new Quantizer(AUDIO_SAMPLE_RATE_EXACT);
 	quantizer[0]->configure(noiseshaping, dither, factor);
@@ -75,9 +76,12 @@ AsyncAudioInputSPDIF3::AsyncAudioInputSPDIF3(bool dither, bool noiseshaping,floa
 	quantizer[1]->configure(noiseshaping, dither, factor);
 	begin();
 	}
-PROGMEM
+FLASHMEM
 void AsyncAudioInputSPDIF3::begin()
 {
+	
+	AudioOutputSPDIF3::config_spdif3();
+		
 	dma.begin(true); // Allocate the DMA channel first
 	const uint32_t noByteMinorLoop=2*4;
 	dma.TCD->SOFF = 4;
@@ -94,10 +98,9 @@ void AsyncAudioInputSPDIF3::begin()
 	dma.TCD->DADDR = spdif_rx_buffer;
 	dma.triggerAtHardwareEvent(DMAMUX_SOURCE_SPDIF_RX);
 
-	SPDIF_SCR |=SPDIF_SCR_DMA_RX_EN;		//DMA Receive Request Enable
+	//SPDIF_SCR |=SPDIF_SCR_DMA_RX_EN;		//DMA Receive Request Enable
 	dma.enable();
 	dma.attachInterrupt(isr);
-	config_spdifIn();
 #ifdef DEBUG_SPDIF_IN
 	while (!Serial);
 #endif
@@ -105,44 +108,24 @@ void AsyncAudioInputSPDIF3::begin()
 	_bufferLPFilter.numStages=1;
 	_bufferLPFilter.pState=new float[2];
 	getCoefficients(_bufferLPFilter.pCoeffs, BiquadType::LOW_PASS, 0., 5., AUDIO_SAMPLE_RATE_EXACT/AUDIO_BLOCK_SAMPLES, 0.5);
+	SPDIF_SCR &=(~SPDIF_SCR_RXFIFO_OFF_ON);	//receive fifo is turned on again
+	
+	SPDIF_SRCD = 0;
+	SPDIF_SCR |= SPDIF_SCR_DMA_RX_EN;
+	CORE_PIN15_CONFIG = 3;
+	IOMUXC_SPDIF_IN_SELECT_INPUT = 0; // GPIO_AD_B1_03_ALT3
 }
-bool AsyncAudioInputSPDIF3::isLocked() const {
-	__disable_irq();
-	bool l=locked;
-	__enable_irq();
-	return l;
-}
-void AsyncAudioInputSPDIF3::spdif_interrupt(){
-	if(SPDIF_SIS & SPDIF_SIS_LOCK){
-		if (!locked){
-			locked=true;
-			lockChanged=true;
-		}
-	}
-	else if(SPDIF_SIS & SPDIF_SIS_LOCKLOSS){
-		if (locked){
-			locked=false;
-			lockChanged=true;
-			resetResampler=true;
-		}
-	}
-	SPDIF_SIC |= SPDIF_SIC_LOCKLOSS;//clear SPDIF_SIC_LOCKLOSS interrupt
-	SPDIF_SIC |= SPDIF_SIC_LOCK;	//clear SPDIF_SIC_LOCK interrupt
+bool AsyncAudioInputSPDIF3::isLocked() {
+	return (SPDIF_SRPC & SPDIF_SRPC_LOCK) == SPDIF_SRPC_LOCK;
 }
 
 void AsyncAudioInputSPDIF3::resample(int16_t* data_left, int16_t* data_right, int32_t& block_offset){
 	block_offset=0;
-	if(!_resampler.initialized()){
-		return;
-	}
-	__disable_irq();
-	if(!locked){
-		__enable_irq();
+	if(!_resampler.initialized() || !isLocked()){
 		return;
 	}
 	int32_t bOffset=buffer_offset;
 	int32_t resOffset=resample_offset;
-	__enable_irq();
 		
 	uint16_t inputBufferStop = bOffset >= resOffset ? bOffset-resOffset : bufferLength-resOffset;
 	if (inputBufferStop==0){
@@ -199,14 +182,13 @@ void AsyncAudioInputSPDIF3::isr(void)
 		#endif
 		float *destR = &(bufferR[buffer_offset]);
 		float *destL = &(bufferL[buffer_offset]);
-		const float factor= pow(2., 23.)+1;
 		do {			
 			int32_t n=(*src) & 0x800000 ? (*src)|0xFF800000  : (*src) & 0xFFFFFF;
-			*destL++ = (float)(n)/factor;
+			*destL++ = (float)(n)*toFloatAudio;
 			++src;
 
 			n=(*src) & 0x800000 ? (*src)|0xFF800000  : (*src) & 0xFFFFFF;
-			*destR++ = (float)(n)/factor;
+			*destR++ = (float)(n)*toFloatAudio;
 			++src;
 		} while (src < end);
 		buffer_offset=(buffer_offset+SPDIF_RX_BUFFER_LENGTH/4)%bufferLength;
@@ -219,8 +201,8 @@ void AsyncAudioInputSPDIF3::isr(void)
 }
 double AsyncAudioInputSPDIF3::getNewValidInputFrequ(){
 	//page 2129: FrequMeas[23:0]=FreqMeas_CLK / BUS_CLK * 2^10 * GAIN
-	if (SPDIF_SRPC & SPDIF_SRPC_LOCK){
-		const double f=(float)F_BUS_ACTUAL/(1024.*1024.*24.*128.);// bit clock = 128 * sampling frequency
+	if (isLocked()){
+		const double f=(float)F_BUS_ACTUAL/(1024.*1024.*AudioOutputSPDIF3::dpll_Gain()*128.);// bit clock = 128 * sampling frequency
 		const double freqMeas=(SPDIF_SRFM & 0xFFFFFF)*f;
 		if (_lastValidInputFrequ != freqMeas){//frequency not stable yet;
 			_lastValidInputFrequ=freqMeas;
@@ -239,26 +221,14 @@ double AsyncAudioInputSPDIF3::getBufferedTime() const{
 }
 
 void AsyncAudioInputSPDIF3::configure(){
-	__disable_irq();
-	if(resetResampler){
+	if(!isLocked()){
 		_resampler.reset();
-		resetResampler=false;
-	}
-	if(!locked){
-		__enable_irq();
-#ifdef DEBUG_SPDIF_IN
-		Serial.println("lock lost");
-#endif
 		return;
-	}	
+	}
+		
 #ifdef DEBUG_SPDIF_IN
 	const bool bOverf=bufferOverflow;
 	bufferOverflow=false;
-#endif
-	const bool lc=lockChanged;
-	__enable_irq();
-
-#ifdef DEBUG_SPDIF_IN
 	if (bOverf){
 		Serial.print("buffer overflow, buffer offset: ");
 		Serial.print(buffer_offset);
@@ -269,34 +239,30 @@ void AsyncAudioInputSPDIF3::configure(){
 		}
 	}
 #endif
-	if (lc || !_resampler.initialized()){
-		const double inputF=getNewValidInputFrequ();	//returns: -1 ... invalid frequency
-		if (inputF > 0.){
+	const double inputF=getNewValidInputFrequ();	//returns: -1 ... invalid frequency
+	if (inputF > 0.){
+		//we got a valid sample frequency
+		const double frequDiff=inputF/_inputFrequency-1.;
+		if (abs(frequDiff) > 0.01 || !_resampler.initialized()){
+			//the new sample frequency differs from the last one -> configure the _resampler again
+			_inputFrequency=inputF;		
+			_targetLatencyS=max(0.001,(noSamplerPerIsr*3./2./_inputFrequency));
+			_maxLatency=max(2.*_blockDuration, 2*noSamplerPerIsr/_inputFrequency);
+			const int32_t targetLatency=round(_targetLatencyS*inputF);
 			__disable_irq();
-			lockChanged=false;	//only reset lockChanged if a valid frequency was received (inputFrequ > 0.)
+			resample_offset =  targetLatency <= buffer_offset ? buffer_offset - targetLatency : bufferLength -(targetLatency-buffer_offset);
 			__enable_irq();
-			//we got a valid sample frequency
-			const double frequDiff=inputF/_inputFrequency-1.;
-			if (abs(frequDiff) > 0.01 || !_resampler.initialized()){
-				//the new sample frequency differs from the last one -> configure the _resampler again
-				_inputFrequency=inputF;		
-				const int32_t targetLatency=round(_targetLatencyS*inputF);
-				_targetLatencyS=max(0.001,(noSamplerPerIsr*3./2./_inputFrequency));
-				__disable_irq();
-				resample_offset =  targetLatency <= buffer_offset ? buffer_offset - targetLatency : bufferLength -(targetLatency-buffer_offset);
-				__enable_irq();
-				_resampler.configure(inputF, AUDIO_SAMPLE_RATE_EXACT, _attenuation, _minHalfFilterLength);
-		#ifdef DEBUG_SPDIF_IN
-				Serial.print("_maxLatency: ");
-				Serial.println(_maxLatency);
-				Serial.print("targetLatency: ");
-				Serial.println(targetLatency);
-				Serial.print("relative frequ diff: ");
-				Serial.println(frequDiff, 8);
-				Serial.print("configure _resampler with frequency ");
-				Serial.println(inputF,8);			
-		#endif			
-			}
+			_resampler.configure(inputF, AUDIO_SAMPLE_RATE_EXACT);
+	#ifdef DEBUG_SPDIF_IN
+			Serial.print("_maxLatency: ");
+			Serial.println(_maxLatency);
+			Serial.print("targetLatency: ");
+			Serial.println(targetLatency);
+			Serial.print("relative frequ diff: ");
+			Serial.println(frequDiff, 8);
+			Serial.print("configure _resampler with frequency ");
+			Serial.println(inputF,8);			
+	#endif			
 		}
 	}
 }
@@ -332,24 +298,23 @@ void AsyncAudioInputSPDIF3::monitorResampleBuffer(){
 		while (resample_offset<0){
 			resample_offset+=bufferLength;
 		}	
-		//int32_t b_offset=buffer_offset;
 #ifdef DEBUG_SPDIF_IN
 		double bTimeFixed = resample_offset <= buffer_offset ? (buffer_offset-resample_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset : (bufferLength-resample_offset +buffer_offset-_resampler.getXPos())/_lastValidInputFrequ+dmaOffset;	//[seconds]
 #endif
 		__enable_irq();
 #ifdef DEBUG_SPDIF_IN
-		// Serial.print("settled: ");
-		// Serial.println(settled);
+		Serial.print("settled: ");
+		Serial.println(settled);
 		Serial.print("bTime: ");
-		Serial.print(bTime*1e6,3);
+		Serial.println(bTime*1e6,3);
 		Serial.print("_maxLatency: ");
 		Serial.println(_maxLatency*1e6,3);
 		Serial.print("bTime-dmaOffset: ");
-		Serial.print((bTime-dmaOffset)*1e6,3);
+		Serial.println((bTime-dmaOffset)*1e6,3);
 		Serial.print(", _blockDuration: ");
-		Serial.print(_blockDuration*1e6,3);
+		Serial.println(_blockDuration*1e6,3);
 		Serial.print("bTimeFixed: ");
-		Serial.print(bTimeFixed*1e6,3);
+		Serial.println(bTimeFixed*1e6,3);
 
 #endif
 		preload(&_bufferLPFilter, diff);
@@ -403,7 +368,7 @@ double AsyncAudioInputSPDIF3::getInputFrequency() const{
 	__disable_irq();
 	double f=_lastValidInputFrequ;
 	__enable_irq();
-	return f;
+	return isLocked() ? f : 0.;
 }
 double AsyncAudioInputSPDIF3::getTargetLantency() const {
 	__disable_irq();
@@ -411,57 +376,12 @@ double AsyncAudioInputSPDIF3::getTargetLantency() const {
 	__enable_irq();
 	return l ;
 }
-void AsyncAudioInputSPDIF3::config_spdifIn(){
-	//CCM Clock Gating Register 5, imxrt1060_rev1.pdf page 1145
-	CCM_CCGR5 |=CCM_CCGR5_SPDIF(CCM_CCGR_ON); //turn spdif clock on - necessary for receiver!
-	
-	SPDIF_SCR |=SPDIF_SCR_RXFIFO_OFF_ON;	//turn receive fifo off 1->off, 0->on
-
-	SPDIF_SCR&=~(SPDIF_SCR_RXFIFO_CTR);		//reset rx fifo control: normal opertation
-
-	SPDIF_SCR&=~(SPDIF_SCR_RXFIFOFULL_SEL(3));	//reset rx full select
-	SPDIF_SCR|=SPDIF_SCR_RXFIFOFULL_SEL(2);	//full interrupt if at least 8 sample in Rx left and right FIFOs
-
-	SPDIF_SCR|=SPDIF_SCR_RXAUTOSYNC; //Rx FIFO auto sync on
-
-	SPDIF_SCR&=(~SPDIF_SCR_USRC_SEL(3));	//No embedded U channel
-
-    CORE_PIN15_CONFIG  = 3;  //pin 15 set to alt3 -> spdif input
-
-	/// from eval board sample code
-	//   IOMUXC_SetPinConfig(
-	//       IOMUXC_GPIO_AD_B1_03_SPDIF_IN,        /* GPIO_AD_B1_03 PAD functional properties : */
-	//       0x10B0u);                               /* Slew Rate Field: Slow Slew Rate
-	//                                                  Drive Strength Field: R0/6
-	//                                                  Speed Field: medium(100MHz)
-	//                                                  Open Drain Enable Field: Open Drain Disabled
-	//                                                  Pull / Keep Enable Field: Pull/Keeper Enabled
-	//                                                  Pull / Keep Select Field: Keeper
-	//                                                  Pull Up / Down Config. Field: 100K Ohm Pull Down
-	//                                                  Hyst. Enable Field: Hysteresis Disabled */
-	CORE_PIN15_PADCONFIG=0x10B0;
-	SPDIF_SCR &=(~SPDIF_SCR_RXFIFO_OFF_ON);	//receive fifo is turned on again
-
-
-	SPDIF_SRPC &= ~SPDIF_SRPC_CLKSRC_SEL(15);	//reset clock selection page 2136
-	//SPDIF_SRPC |=SPDIF_SRPC_CLKSRC_SEL(6);		//if (DPLL Locked) SPDIF_RxClk else tx_clk (SPDIF0_CLK_ROOT)
-	//page 2129: FrequMeas[23:0]=FreqMeas_CLK / BUS_CLK * 2^10 * GAIN
-	SPDIF_SRPC &=~SPDIF_SRPC_GAINSEL(7);	//reset gain select 0 -> gain = 24*2^10
-	//SPDIF_SRPC |= SPDIF_SRPC_GAINSEL(3);	//gain select: 8*2^10
-	//==============================================
-
-	//interrupts
-	SPDIF_SIE |= SPDIF_SIE_LOCK;	//enable spdif receiver lock interrupt
-	SPDIF_SIE |=SPDIF_SIE_LOCKLOSS;
-
-	lockChanged=true;
-	attachInterruptVector(IRQ_SPDIF, spdif_interrupt);
-	NVIC_SET_PRIORITY(IRQ_SPDIF, 208); // 255 = lowest priority, 208 = priority of update
-	NVIC_ENABLE_IRQ(IRQ_SPDIF);
-
-	SPDIF_SIC |= SPDIF_SIC_LOCK;	//clear SPDIF_SIC_LOCK interrupt
-	SPDIF_SIC |= SPDIF_SIC_LOCKLOSS;//clear SPDIF_SIC_LOCKLOSS interrupt
-	locked=(SPDIF_SRPC & SPDIF_SRPC_LOCK);
+double AsyncAudioInputSPDIF3::getAttenuation() const{
+	return _resampler.getAttenuation();
 }
+int32_t AsyncAudioInputSPDIF3::getHalfFilterLength() const{
+	return _resampler.getHalfFilterLength();
+}
+
 #endif
 

async_input_spdif3.h

@@ -40,17 +40,19 @@
 class AsyncAudioInputSPDIF3 : public AudioStream
 {
 public:
-	///@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
-	AsyncAudioInputSPDIF3(bool dither=true, bool noiseshaping=true,float attenuation=100, int32_t minHalfFilterLength=20);
+	///@param attenuation target attenuation [dB] of the anti-aliasing filter. Only used if AUDIO_SAMPLE_RATE_EXACT < input sample rate (input fs). The attenuation can't be reached if the needed filter length exceeds 2*MAX_FILTER_SAMPLES+1
+	///@param minHalfFilterLength If AUDIO_SAMPLE_RATE_EXACT >= input fs), the filter length of the resampling filter is 2*minHalfFilterLength+1. If AUDIO_SAMPLE_RATE_EXACT < input fs the filter is maybe longer to reach the desired attenuation
+	///@param maxHalfFilterLength Can be used to restrict the maximum filter length at the cost of a lower attenuation
+	AsyncAudioInputSPDIF3(bool dither=true, bool noiseshaping=true,float attenuation=100, int32_t minHalfFilterLength=20, int32_t maxHalfFilterLength=80);
 	~AsyncAudioInputSPDIF3();
-	virtual void update(void);
 	void begin();
-	void stop();
+	virtual void update(void);
 	double getBufferedTime() const;
 	double getInputFrequency() const;
-	bool isLocked() const;
+	static bool isLocked();
 	double getTargetLantency() const;
+	double getAttenuation() const;
+	int32_t getHalfFilterLength() const;
 protected:	
 	static DMAChannel dma;
 	static void isr(void);
@@ -67,27 +69,16 @@ private:
     static volatile uint32_t microsLast;
 	//====================
 
-	// spdif lock-changed interrupt
-	static volatile bool locked;
-	static volatile bool lockChanged;
-	static volatile bool resetResampler;
-  	static void spdif_interrupt();
-	#ifdef MEASURE_FREQ
-	static FrequencyMeasurement frequMeasure;	
-	#endif
-	//=============================
-	float _attenuation;
-	int32_t _minHalfFilterLength;
 	Resampler _resampler;
 	Quantizer* quantizer[2];
 	arm_biquad_cascade_df2T_instance_f32 _bufferLPFilter;
 	
 	volatile double _bufferedTime;
 	volatile double _lastValidInputFrequ;
-	double _inputFrequency;
+	double _inputFrequency=0.;
 	double _targetLatencyS;	//target latency [seconds]
-	const double _blockDuration=AUDIO_BLOCK_SAMPLES/AUDIO_SAMPLE_RATE; //[seconds] 
-	const double _maxLatency=2.*_blockDuration;
+	const double _blockDuration=AUDIO_BLOCK_SAMPLES/AUDIO_SAMPLE_RATE_EXACT; //[seconds] 
+	double _maxLatency=2.*_blockDuration;
 
 #ifdef DEBUG_SPDIF_IN
 	static volatile bool bufferOverflow;

examples/HardwareTesting/PassThroughAsyncSpdif/PassThroughAsyncSpdif.ino

@@ -1,41 +1,45 @@
 
 #include <Audio.h>
 
+AsyncAudioInputSPDIF3     spdifIn(false, false, 100, 20, 80);	//dither = true, noiseshaping = true, anti-aliasing attenuation=100dB, minimum half resampling filter length=20, maximum half resampling filter length=80
 AudioOutputSPDIF3   spdifOut;
-AsyncAudioInputSPDIF3     spdifIn(true, true, 100, 20);	//dither = true, noiseshaping = true, anti-aliasing attenuation=100dB, minimum resampling filter length=20
-//
 
 AudioConnection          patchCord1(spdifIn, 0, spdifOut, 0);
 AudioConnection          patchCord2(spdifIn, 1, spdifOut, 1);
 
 void setup() {
-
-  // put your setup code here, to run once:
   AudioMemory(12);
+  Serial.begin(57600);
   while (!Serial);
 
 }
 
 void loop() {
-	double bufferedTine=spdifIn.getBufferedTime();
-	//double targetLatency = spdifIn.getTargetLantency();
-	Serial.print("buffered time [micro seconds]: ");
-	Serial.println(bufferedTine*1e6,2);
-	// Serial.print(", target: ");
-	// Serial.println(targetLatency*1e6,2);
-	
-	double pUsageIn=spdifIn.processorUsage(); 
-	Serial.print("processor usage [%]: ");
-	Serial.println(pUsageIn);
-
-	// bool islocked=spdifIn.isLocked(); 
-	// Serial.print("isLocked: ");
-	// Serial.println(islocked);
-
-	// double f=spdifIn.getInputFrequency();
-	// Serial.print("frequency: ");
-	// Serial.println(f);
-	// Serial.print("Memory max: ");
-  	// Serial.println(AudioMemoryUsageMax());
-	delay(500);
-}
+  double bufferedTime=spdifIn.getBufferedTime();
+  Serial.print("buffered time [micro seconds]: ");
+  Serial.println(bufferedTime*1e6,2);
+  
+  
+  Serial.print("locked: ");
+  Serial.println(spdifIn.isLocked());
+  Serial.print("input frequency: ");
+  double inputFrequency=spdifIn.getInputFrequency();
+  Serial.println(inputFrequency);
+  Serial.print("anti-aliasing attenuation: ");
+  Serial.println(spdifIn.getAttenuation());
+  
+  Serial.print("resampling goup delay [milli seconds]: ");
+  Serial.println(spdifIn.getHalfFilterLength()/inputFrequency*1e3,2);
+  
+  Serial.print("half filter length: ");
+  Serial.println(spdifIn.getHalfFilterLength()); 
+  
+  double pUsageIn=spdifIn.processorUsage(); 
+  Serial.print("processor usage [%]: ");
+  Serial.println(pUsageIn);
+
+  Serial.print("max number of used blocks: ");
+  Serial.println(AudioMemoryUsageMax()); 
+
+  delay(500);
+}

output_spdif3.h

@@ -34,6 +34,7 @@ public:
 	virtual void update(void);
 	void begin(void);
 	friend class AudioInputSPDIF3;
+	friend class AsyncAudioInputSPDIF3;
 	static void mute_PCM(const bool mute);
 	static bool pll_locked(void);
 protected: