Merge pull request #364 from alex6679/master

AsyncAudioInputSPDIF3 minor bug fix/clipping prevention

Quantizer.cpp

@@ -82,6 +82,21 @@ void Quantizer::configure(bool noiseShaping, bool dither, float factor){
      _noiseShaping=noiseShaping;
      _dither=dither;
      _factor=factor;
+     if (_dither){
+         _factor-=1.f;
+     }
+     if (_noiseShaping){
+        // the maximum rounding error is 0.5
+        // Assuming the rounding errors of the last NOISE_SHAPE_F_LENGTH samples was 0.5 at the positive noise-shaping-coefficients and -0.5 at the negative coefficients,
+        // the maximum added value can be computed as follows:
+        float* f=_noiseSFilter;
+        float maxAddedVal=0.f;
+        for (uint16_t j =0; j< NOISE_SHAPE_F_LENGTH; j++){
+            maxAddedVal+=abs(*f++);
+        }
+        maxAddedVal/=2.f;  
+        _factor-=maxAddedVal;
+     }
      reset();
 }
 void Quantizer::reset(){
@@ -141,7 +156,7 @@ void Quantizer::quantize(float* input, int16_t* output, uint16_t length){
             *output=(int16_t)_factor;
         }
         else if (xnDR < -_factor){
-            *output=(int16_t)_factor;
+            *output=-(int16_t)_factor;
         }
         else {
             *output=(int16_t)xnDR;

async_input_spdif3.h

@@ -43,7 +43,7 @@ public:
 	///@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(bool dither=false, bool noiseshaping=false,float attenuation=100, int32_t minHalfFilterLength=20, int32_t maxHalfFilterLength=80);
 	~AsyncAudioInputSPDIF3();
 	void begin();
 	virtual void update(void);

examples/HardwareTesting/PassThroughAsyncSpdif/PassThroughAsyncSpdif.ino

@@ -1,7 +1,7 @@
 
 #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
+AsyncAudioInputSPDIF3     spdifIn(false, false, 100, 20, 80);	//dither = false, noiseshaping = false, anti-aliasing attenuation=100dB, minimum half resampling filter length=20, maximum half resampling filter length=80
 AudioOutputSPDIF3   spdifOut;
 
 AudioConnection          patchCord1(spdifIn, 0, spdifOut, 0);