Audio is dead silent, howver, it is much closer now than before.

audiohandler.cpp

@@ -303,7 +303,7 @@ void audioHandler::setupLADSP()
     // setup for the plugin:
 
     int lPluginIndex;
-    pre_control = 0.01f;
+    pre_control = 0.99f;
 
     void *pvPluginHandle = loadLADSPAPluginLibrary(pcPluginFilename);
     dlerror();
@@ -323,11 +323,12 @@ void audioHandler::setupLADSP()
     }
 
 
-    BUF_SIZE = (this->chunkSize);
 
-    BUF_SIZE = 1112; // hard coding what seems to be the correct value here.
 
-    qDebug(logAudio()) << __PRETTY_FUNCTION__ << ": Setting BUF_SIZE to " << BUF_SIZE; // 960
+    BUF_SIZE = 2728; // starting with the largest likely size
+
+    qDebug(logAudio()) << __PRETTY_FUNCTION__ << ": Setting BUF_SIZE to " << BUF_SIZE;
+    qDebug(logAudio()) << __PRETTY_FUNCTION__ << ": Setting sample rate to: " << SAMPLE_RATE;
 
     pInBuffer[0]= (float*)malloc(sizeof(LADSPA_Data) * BUF_SIZE/2);
     pInBuffer[1]= (float*)malloc(sizeof(LADSPA_Data) * BUF_SIZE/2);
@@ -403,16 +404,16 @@ void audioHandler::setupLADSP()
 
 void audioHandler::runPlugin()
 {
-    int numread=0;
+    // numread actually sets the
+    int numSamplesToProcess= BUF_SIZE/2;
     if (psDescriptor->activate != NULL)
         psDescriptor->activate(handle);
 
     // grab audio (already done, no sound file to read from)
     // sfload does this: copy from input file to pInBuffer, at size BUF_SIZE
 
-    // Crash here:
-    psDescriptor->run(handle, numread); // run the audio through the plugin. Number of pieces processed is numread. Compare to size of data needing to be read and run again if needed.
-    qDebug(logAudio()) << "Number of samples processed: " << numread;
+    psDescriptor->run(handle, numSamplesToProcess); // run the audio through the plugin. Number of pieces processed is numread. Compare to size of data needing to be read and run again if needed.
+    //qDebug(logAudio()) << "Number of samples processed: " << numSamplesToProcess;
 }
 
 void audioHandler::setVolume(unsigned char volume)
@@ -669,20 +670,45 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 
     pluginMutex.lock();
 
+    // out has size inPacket.data = BUF_SIZE
+    // pOut/inBuffer[1] has size BUF_SIZE/2
+    // inPacket.data has size BUF_SIZE
+    // pluginOutput16bitInterlaced has size BUF_SIZE
+
+    // Flow:
+    // inPacket --> out --> pInBuffer --> runPlugin() --> pOutBuffer --> pluginOutput16bitInterlaced --> inPacket
+
+
     //
     // Note, size of qint16 array is data.size() / 2 .
     //
     qint16* out = (qint16*)inPacket.data.constData();
 
-    qDebug() << "about to make byte array:";
-    // Why does the next line down crash!?!?!
-    qDebug() << "made byte array.";
+    BUF_SIZE = inPacket.data.size() / 2; // divide by two because the inPacket is 8 bits per member
+                                         // and all our usage is at 16 bits per.
 
-    qDebug(logAudio()) << "size of inPacket data: " << inPacket.data.size();
-    qDebug(logAudio()) << "size of BUF_SIZE: " << BUF_SIZE;
+    // --- Input and Output buffer strategy:
+    // The plugin does not look at the size of the allocation. And the pointer
+    // needs to be to the same original location as the plugin runs
+    // Therefore, we allocate a large amount in the setupLADSPA() function,
+    // and simply adjust the BUF_SIZE so that the runPlugin() call only goes
+    // as far as is needed into the memory.
+
+    // I think these lines caused the pointer location at the base of the array to change, no good.
+    //    pInBuffer[0]= (float*)realloc(pInBuffer[0], sizeof(LADSPA_Data) * BUF_SIZE / 2);
+    //    pInBuffer[1]= (float*)realloc(pInBuffer[1], sizeof(LADSPA_Data) * BUF_SIZE / 2);
+    //    pOutBuffer[0]= (float*)realloc(pOutBuffer[0], sizeof(LADSPA_Data) * BUF_SIZE / 2);
+    //    pOutBuffer[1]= (float*)realloc(pOutBuffer[1], sizeof(LADSPA_Data) * BUF_SIZE / 2);
 
 
-    pluginOutput16bitInterlaced = (uint16_t *)malloc(sizeof(uint16_t) * inPacket.data.size() );
+
+    //qDebug(logAudio()) << "size of inPacket data: " << inPacket.data.size();
+
+    //qDebug(logAudio()) << "size of BUF_SIZE: " << BUF_SIZE;
+
+
+    // Allocated for 16 bit samples, interlaced stereo
+    pluginOutput16bitInterlaced = (uint16_t *)malloc(sizeof(uint16_t) * BUF_SIZE );
 
     // Copy data from "out" to pInBuffer:
     // copyDataIn();
@@ -698,12 +724,14 @@ void audioHandler::incomingAudio(audioPacket inPacket)
         pInBuffer[1][n/2] = (out[n+1] - mid) * scalingFactor;
     }
 
-    control = pre_control;
+    // control is a number that the plugin has a pointer to.
+    // In this case, control is a float from 0 to 1.
+    control = pre_control = 0.9;
     runPlugin();
 
     // copy data back out from pOutBuffer to inPacket.data
     // pOut/inBuffer[0] has size BUF_SIZE/2
-    // pOut/inBuffer[1] has size BUF_SIZE/2
+
 
     // Maybe the issue is here?
 
@@ -715,9 +743,12 @@ void audioHandler::incomingAudio(audioPacket inPacket)
 
     // copyDataOut();
 
-    for(int i=0; i < BUF_SIZE; i++)
+    // CAREFUL, get the 16 bits into the 8 bit format carefully:
+    for(int i=0; i < BUF_SIZE*2; i+=2)
     {
-        inPacket.data[i] = pluginOutput16bitInterlaced[i];
+        inPacket.data[i] =   pluginOutput16bitInterlaced[i/2] & 0x0f;
+        inPacket.data[i+1] = (pluginOutput16bitInterlaced[i/2] & 0xf0 >> 8);
+
     }