All modulators: use buffer for input audio that is always in use while generation is running. This fixes lockup problem reported in issue #11

plugins/channeltx/modam/ammod.cpp

@@ -57,8 +57,8 @@ AMMod::AMMod() :
 
 	apply();
 
-	//m_audioBuffer.resize(1<<14);
+	m_audioBuffer.resize(1<<14);
-	//m_audioBufferFill = 0;
+	m_audioBufferFill = 0;
 
 	m_movingAverage.resize(16, 0);
 	m_volumeAGC.resize(4096, 0.003, 0);
@@ -121,6 +121,7 @@ void AMMod::pull(Sample& sample)
         }
     }
 
+    m_audioBufferFill++;
     m_interpolatorDistanceRemain += m_interpolatorDistance;
 
     ci *= m_carrierNco.nextIQ(); // shift to carrier frequency
@@ -136,6 +137,19 @@ void AMMod::pull(Sample& sample)
 	sample.m_imag = (FixReal) ci.imag();
 }
 
+void AMMod::pullAudio(int nbSamples)
+{
+//    qDebug("AMMod::pullAudio: %d", nbSamples);
+
+    if (nbSamples > m_audioBuffer.size())
+    {
+        m_audioBuffer.resize(nbSamples);
+    }
+
+    m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioBuffer[0]), nbSamples*sizeof(AudioSample), 10);
+    m_audioBufferFill = 0;
+}
+
 void AMMod::modulateSample()
 {
 	Real t;
@@ -149,8 +163,6 @@ void AMMod::modulateSample()
 
 void AMMod::pullAF(Real& sample)
 {
-    int16_t audioSample[2];
-
     switch (m_afInput)
     {
     case AMModInputTone:
@@ -186,8 +198,7 @@ void AMMod::pullAF(Real& sample)
         }
         break;
     case AMModInputAudio:
-        m_audioFifo.read(reinterpret_cast<quint8*>(audioSample), 1, 10);
+        sample = ((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f) * m_running.m_volumeFactor;
-        sample = ((audioSample[0] + audioSample[1])  / 65536.0f) * m_running.m_volumeFactor;
         break;
     case AMModInputCWTone:
         Real fadeFactor;

plugins/channeltx/modam/ammod.h

@@ -185,6 +185,7 @@ public:
             bool playLoop);
 
     virtual void pull(Sample& sample);
+    virtual void pullAudio(int nbSamples);
     virtual void start();
     virtual void stop();
     virtual bool handleMessage(const Message& cmd);
@@ -294,8 +295,8 @@ private:
     MovingAverage<Real> m_movingAverage;
     SimpleAGC m_volumeAGC;
 
-    //AudioVector m_audioBuffer;
+    AudioVector m_audioBuffer;
-    //uint m_audioBufferFill;
+    uint m_audioBufferFill;
 
     AudioFifo m_audioFifo;
     SampleVector m_sampleBuffer;

plugins/channeltx/modnfm/nfmmod.cpp

@@ -59,8 +59,8 @@ NFMMod::NFMMod() :
 
 	apply();
 
-	//m_audioBuffer.resize(1<<14);
+	m_audioBuffer.resize(1<<14);
-	//m_audioBufferFill = 0;
+	m_audioBufferFill = 0;
 
 	m_movingAverage.resize(16, 0);
 	m_volumeAGC.resize(4096, 0.003, 0);
@@ -136,6 +136,7 @@ void NFMMod::pull(Sample& sample)
         }
     }
 
+    m_audioBufferFill++;
     m_interpolatorDistanceRemain += m_interpolatorDistance;
 
     ci *= m_carrierNco.nextIQ(); // shift to carrier frequency
@@ -151,6 +152,17 @@ void NFMMod::pull(Sample& sample)
 	sample.m_imag = (FixReal) ci.imag();
 }
 
+void NFMMod::pullAudio(int nbSamples)
+{
+    if (nbSamples > m_audioBuffer.size())
+    {
+        m_audioBuffer.resize(nbSamples);
+    }
+
+    m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioBuffer[0]), nbSamples*sizeof(AudioSample), 10);
+    m_audioBufferFill = 0;
+}
+
 void NFMMod::modulateSample()
 {
 	Real t;
@@ -174,8 +186,6 @@ void NFMMod::modulateSample()
 
 void NFMMod::pullAF(Real& sample)
 {
-    int16_t audioSample[2];
-
     switch (m_afInput)
     {
     case NFMModInputTone:
@@ -211,8 +221,7 @@ void NFMMod::pullAF(Real& sample)
         }
         break;
     case NFMModInputAudio:
-        m_audioFifo.read(reinterpret_cast<quint8*>(audioSample), 1, 10);
+        sample = ((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f) * m_running.m_volumeFactor;
-        sample = ((audioSample[0] + audioSample[1])  / 65536.0f) * m_running.m_volumeFactor;
         break;
     case NFMModInputCWTone:
         Real fadeFactor;

plugins/channeltx/modnfm/nfmmod.h

@@ -190,6 +190,7 @@ public:
             float ctcssFrequency);
 
     virtual void pull(Sample& sample);
+    virtual void pullAudio(int nbSamples);
     virtual void start();
     virtual void stop();
     virtual bool handleMessage(const Message& cmd);
@@ -342,8 +343,8 @@ private:
     MovingAverage<Real> m_movingAverage;
     SimpleAGC m_volumeAGC;
 
-    //AudioVector m_audioBuffer;
+    AudioVector m_audioBuffer;
-    //uint m_audioBufferFill;
+    uint m_audioBufferFill;
 
     AudioFifo m_audioFifo;
     SampleVector m_sampleBuffer;

plugins/channeltx/modssb/ssbmod.cpp

@@ -69,8 +69,8 @@ SSBMod::SSBMod(BasebandSampleSink* sampleSink) :
 	m_config.m_toneFrequency = 1000.0f;
 	m_config.m_audioSampleRate = DSPEngine::instance()->getAudioSampleRate();
 
-	//m_audioBuffer.resize(1<<14);
+	m_audioBuffer.resize(1<<14);
-	//m_audioBufferFill = 0;
+	m_audioBufferFill = 0;
 
 //    m_magsqSpectrum = 0.0f;
 //    m_magsqSum = 0.0f;
@@ -166,6 +166,7 @@ void SSBMod::pull(Sample& sample)
         }
     }
 
+    m_audioBufferFill++;
     m_interpolatorDistanceRemain += m_interpolatorDistance;
 
     ci *= m_carrierNco.nextIQ(); // shift to carrier frequency
@@ -182,6 +183,17 @@ void SSBMod::pull(Sample& sample)
 	sample.m_imag = (FixReal) ci.imag();
 }
 
+void SSBMod::pullAudio(int nbSamples)
+{
+    if (nbSamples > m_audioBuffer.size())
+    {
+        m_audioBuffer.resize(nbSamples);
+    }
+
+    m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioBuffer[0]), nbSamples*sizeof(AudioSample), 10);
+    m_audioBufferFill = 0;
+}
+
 void SSBMod::modulateSample()
 {
     pullAF(m_modSample);
@@ -197,7 +209,6 @@ void SSBMod::pullAF(Complex& sample)
         return;
 	}
 
-    int16_t audioSample[2];
     Complex ci;
     fftfilt::cmplx *filtered;
     int n_out = 0;
@@ -279,24 +290,22 @@ void SSBMod::pullAF(Complex& sample)
         }
         break;
     case SSBModInputAudio:
-        m_audioFifo.read(reinterpret_cast<quint8*>(audioSample), 1, 10);
-
         if (m_running.m_audioBinaural)
     	{
         	if (m_running.m_audioFlipChannels)
         	{
-                ci.real((audioSample[1] / 32768.0f) * m_running.m_volumeFactor);
+                ci.real((m_audioBuffer[m_audioBufferFill].r / 32768.0f) * m_running.m_volumeFactor);
-                ci.imag((audioSample[0] / 32768.0f) * m_running.m_volumeFactor);
+                ci.imag((m_audioBuffer[m_audioBufferFill].l / 32768.0f) * m_running.m_volumeFactor);
         	}
         	else
         	{
-                ci.real((audioSample[0] / 32768.0f) * m_running.m_volumeFactor);
+                ci.real((m_audioBuffer[m_audioBufferFill].l / 32768.0f) * m_running.m_volumeFactor);
-                ci.imag((audioSample[1] / 32768.0f) * m_running.m_volumeFactor);
+                ci.imag((m_audioBuffer[m_audioBufferFill].r / 32768.0f) * m_running.m_volumeFactor);
         	}
     	}
         else
         {
-            ci.real(((audioSample[0] + audioSample[1])  / 65536.0f) * m_running.m_volumeFactor);
+            ci.real(((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r)  / 65536.0f) * m_running.m_volumeFactor);
             ci.imag(0.0f);
         }
 

plugins/channeltx/modssb/ssbmod.h

@@ -190,6 +190,7 @@ public:
             bool playLoop);
 
     virtual void pull(Sample& sample);
+    virtual void pullAudio(int nbSamples);
     virtual void start();
     virtual void stop();
     virtual bool handleMessage(const Message& cmd);
@@ -366,8 +367,8 @@ private:
     MovingAverage<Real> m_movingAverage;
     SimpleAGC m_volumeAGC;
 
-    //AudioVector m_audioBuffer;
+    AudioVector m_audioBuffer;
-    //uint m_audioBufferFill;
+    uint m_audioBufferFill;
 
     AudioFifo m_audioFifo;
     QMutex m_settingsMutex;

plugins/channeltx/modwfm/wfmmod.cpp

@@ -62,8 +62,8 @@ WFMMod::WFMMod() :
 
 	apply();
 
-	//m_audioBuffer.resize(1<<14);
+	m_audioBuffer.resize(1<<14);
-	//m_audioBufferFill = 0;
+	m_audioBufferFill = 0;
 
 	m_movingAverage.resize(16, 0);
 	m_volumeAGC.resize(4096, 0.003, 0);
@@ -116,8 +116,10 @@ void WFMMod::pull(Sample& sample)
     {
         pullAF(m_modSample);
         calculateLevel(m_modSample.real());
+        m_audioBufferFill++;
     }
 
+    m_audioBufferFill++;
     m_interpolatorDistanceRemain += m_interpolatorDistance;
 
     m_modPhasor += (m_running.m_fmDeviation / (float) m_running.m_outputSampleRate) * ri.real() * M_PI;
@@ -139,10 +141,19 @@ void WFMMod::pull(Sample& sample)
 	sample.m_imag = (FixReal) ci.imag();
 }
 
+void WFMMod::pullAudio(int nbSamples)
+{
+    if (nbSamples > m_audioBuffer.size())
+    {
+        m_audioBuffer.resize(nbSamples);
+    }
+
+    m_audioFifo.read(reinterpret_cast<quint8*>(&m_audioBuffer[0]), nbSamples*sizeof(AudioSample), 10);
+    m_audioBufferFill = 0;
+}
+
 void WFMMod::pullAF(Complex& sample)
 {
-    int16_t audioSample[2];
-
     switch (m_afInput)
     {
     case WFMModInputTone:
@@ -184,9 +195,7 @@ void WFMMod::pullAF(Complex& sample)
         break;
     case WFMModInputAudio:
         {
-            Real s = (audioSample[0] + audioSample[1])  / 65536.0f;
+            sample.real(((m_audioBuffer[m_audioBufferFill].l + m_audioBuffer[m_audioBufferFill].r) / 65536.0f) * m_running.m_volumeFactor);
-            m_audioFifo.read(reinterpret_cast<quint8*>(audioSample), 1, 10);
-            sample.real(s * m_running.m_volumeFactor);
             sample.imag(0.0f);
         }
         break;

plugins/channeltx/modwfm/wfmmod.h

@@ -187,6 +187,7 @@ public:
             bool playLoop);
 
     virtual void pull(Sample& sample);
+    virtual void pullAudio(int nbSamples);
     virtual void start();
     virtual void stop();
     virtual bool handleMessage(const Message& cmd);
@@ -322,8 +323,8 @@ private:
     MovingAverage<Real> m_movingAverage;
     SimpleAGC m_volumeAGC;
 
-    //AudioVector m_audioBuffer;
+    AudioVector m_audioBuffer;
-    //uint m_audioBufferFill;
+    uint m_audioBufferFill;
 
     AudioFifo m_audioFifo;
     SampleVector m_sampleBuffer;

sdrbase/dsp/basebandsamplesource.h

@@ -35,12 +35,14 @@ public:
 	virtual void start() = 0;
 	virtual void stop() = 0;
 	virtual void pull(Sample& sample) = 0;
+	virtual void pullAudio(int nbSamples) {}
 
     /** direct feeding of sample source FIFO */
 	void feed(SampleSourceFifo* sampleFifo, int nbSamples)
 	{
 	    SampleVector::iterator writeAt;
 	    sampleFifo->getWriteIterator(writeAt);
+	    pullAudio(nbSamples); // Pre-fetch input audio samples this is mandatory to keep things running smoothly
 
 	    for (int i = 0; i < nbSamples; i++)
 	    {

sdrbase/dsp/dspdevicesinkengine.cpp

@@ -176,7 +176,9 @@ void DSPDeviceSinkEngine::work(int nbWriteSamples)
 	// single channel source handling
 	if ((m_threadedBasebandSampleSources.size() + m_basebandSampleSources.size()) == 1)
 	{
-		for (ThreadedBasebandSampleSources::iterator it = m_threadedBasebandSampleSources.begin(); it != m_threadedBasebandSampleSources.end(); ++it)
+        qDebug("DSPDeviceSinkEngine::work: single channel source handling");
+
+        for (ThreadedBasebandSampleSources::iterator it = m_threadedBasebandSampleSources.begin(); it != m_threadedBasebandSampleSources.end(); ++it)
 		{
 			(*it)->feed(sampleFifo, nbWriteSamples);
 		}
@@ -189,47 +191,52 @@ void DSPDeviceSinkEngine::work(int nbWriteSamples)
 	else if ((m_threadedBasebandSampleSources.size() + m_basebandSampleSources.size()) > 1)
 	{
 	    qDebug("DSPDeviceSinkEngine::work: multiple channel sources handling");
-	    SampleVector::iterator writeBegin;
+	    // TODO properly
-	    sampleFifo->getWriteIterator(writeBegin);
+	    if (m_threadedBasebandSampleSources.size() > 0)
-	    SampleVector::iterator writeAt = writeBegin;
+	    {
-	    Sample s;
+	        (*m_threadedBasebandSampleSources.begin())->feed(sampleFifo, nbWriteSamples);
-	    int sourceOccurence = 0;
+	    }
-
+//	    SampleVector::iterator writeBegin;
-	    for (int is = 0; is < nbWriteSamples; is++)
+//	    sampleFifo->getWriteIterator(writeBegin);
-		{
+//	    SampleVector::iterator writeAt = writeBegin;
-			// pull data from threaded sources and merge them in the device sample FIFO
+//	    Sample s;
-			for (ThreadedBasebandSampleSources::iterator it = m_threadedBasebandSampleSources.begin(); it != m_threadedBasebandSampleSources.end(); ++it)
+//	    int sourceOccurence = 0;
-			{
+//
-				(*it)->pull(s);
+//	    for (int is = 0; is < nbWriteSamples; is++)
-				s /= (m_threadedBasebandSampleSources.size() + m_basebandSampleSources.size());
+//		{
-
+//			// pull data from threaded sources and merge them in the device sample FIFO
-				if (sourceOccurence == 0) {
+//			for (ThreadedBasebandSampleSources::iterator it = m_threadedBasebandSampleSources.begin(); it != m_threadedBasebandSampleSources.end(); ++it)
-				    (*writeAt) = s;
+//			{
-				} else {
+//				(*it)->pull(s);
-	                (*writeAt) += s;
+//				s /= (m_threadedBasebandSampleSources.size() + m_basebandSampleSources.size());
-				}
+//
-
+//				if (sourceOccurence == 0) {
-				sourceOccurence++;
+//				    (*writeAt) = s;
-			}
+//				} else {
-
+//	                (*writeAt) += s;
-			// pull data from direct sources and merge them in the device sample FIFO
+//				}
-			for (BasebandSampleSources::iterator it = m_basebandSampleSources.begin(); it != m_basebandSampleSources.end(); ++it)
+//
-			{
+//				sourceOccurence++;
-				(*it)->pull(s);
+//			}
-				s /= (m_threadedBasebandSampleSources.size() + m_basebandSampleSources.size());
+//
-
+//			// pull data from direct sources and merge them in the device sample FIFO
-                if (sourceOccurence == 0) {
+//			for (BasebandSampleSources::iterator it = m_basebandSampleSources.begin(); it != m_basebandSampleSources.end(); ++it)
-                    (*writeAt) = s;
+//			{
-                } else {
+//				(*it)->pull(s);
-                    (*writeAt) += s;
+//				s /= (m_threadedBasebandSampleSources.size() + m_basebandSampleSources.size());
-                }
+//
-
+//                if (sourceOccurence == 0) {
-                sourceOccurence++;
+//                    (*writeAt) = s;
-			}
+//                } else {
-
+//                    (*writeAt) += s;
-			sampleFifo->bumpIndex(writeAt);
+//                }
-            sourceOccurence = 0;
+//
-		}
+//                sourceOccurence++;
+//			}
+//
+//			sampleFifo->bumpIndex(writeAt);
+//            sourceOccurence = 0;
+//		}
 
 		// feed the mix to the main spectrum sink
 //		if (m_spectrumSink)

sdrbase/dsp/upchannelizer.h

@@ -63,6 +63,7 @@ public:
     virtual void start();
     virtual void stop();
     virtual void pull(Sample& sample);
+    virtual void pullAudio(int nbSamples) { if (m_sampleSource) m_sampleSource->pullAudio(nbSamples); }
 
     virtual bool handleMessage(const Message& cmd);