Require use of AudioBus::create() to avoid ref-counting issues

Source/modules/webaudio/ScriptProcessorNode.cpp

 /*
  * Copyright (C) 2010, Google Inc. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
  * are met:
  * 1.  Redistributions of source code must retain the above copyright
  *    notice, this list of conditions and the following disclaimer.
  * 2.  Redistributions in binary form must reproduce the above copyright
  *    notice, this list of conditions and the following disclaimer in the
@@ skipping 61 matching lines @@
 
 ScriptProcessorNode::ScriptProcessorNode(AudioContext* context, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels)
     : AudioNode(context, sampleRate)
     , m_doubleBufferIndex(0)
     , m_doubleBufferIndexForEvent(0)
     , m_bufferSize(bufferSize)
     , m_bufferReadWriteIndex(0)
     , m_isRequestOutstanding(false)
     , m_numberOfInputChannels(numberOfInputChannels)
     , m_numberOfOutputChannels(numberOfOutputChannels)
-    , m_internalInputBus(numberOfInputChannels, AudioNode::ProcessingSizeInFrames, false)
+    , m_internalInputBus(AudioBus::create(numberOfInputChannels, AudioNode::ProcessingSizeInFrames, false))
 {
     // Regardless of the allowed buffer sizes, we still need to process at the granularity of the AudioNode.
     if (m_bufferSize < AudioNode::ProcessingSizeInFrames)
         m_bufferSize = AudioNode::ProcessingSizeInFrames;
 
     ASSERT(numberOfInputChannels <= AudioContext::maxNumberOfChannels());
 
     addInput(adoptPtr(new AudioNodeInput(this)));
     addOutput(adoptPtr(new AudioNodeOutput(this, numberOfOutputChannels)));
 
@@ skipping 54 matching lines @@
     unsigned doubleBufferIndex = this->doubleBufferIndex();
     bool isDoubleBufferIndexGood = doubleBufferIndex < 2 && doubleBufferIndex < m_inputBuffers.size() && doubleBufferIndex < m_outputBuffers.size();
     ASSERT(isDoubleBufferIndexGood);
     if (!isDoubleBufferIndexGood)
         return;
     
     AudioBuffer* inputBuffer = m_inputBuffers[doubleBufferIndex].get();
     AudioBuffer* outputBuffer = m_outputBuffers[doubleBufferIndex].get();
 
     // Check the consistency of input and output buffers.
-    unsigned numberOfInputChannels = m_internalInputBus.numberOfChannels();
+    unsigned numberOfInputChannels = m_internalInputBus->numberOfChannels();
     bool buffersAreGood = outputBuffer && bufferSize() == outputBuffer->length() && m_bufferReadWriteIndex + framesToProcess <= bufferSize();
 
     // If the number of input channels is zero, it's ok to have inputBuffer = 0.
-    if (m_internalInputBus.numberOfChannels())
+    if (m_internalInputBus->numberOfChannels())
         buffersAreGood = buffersAreGood && inputBuffer && bufferSize() == inputBuffer->length();
 
     ASSERT(buffersAreGood);
     if (!buffersAreGood)
         return;
 
     // We assume that bufferSize() is evenly divisible by framesToProcess - should always be true, but we should still check.
     bool isFramesToProcessGood = framesToProcess && bufferSize() >= framesToProcess && !(bufferSize() % framesToProcess);
     ASSERT(isFramesToProcessGood);
     if (!isFramesToProcessGood)
         return;
 
     unsigned numberOfOutputChannels = outputBus->numberOfChannels();
 
     bool channelsAreGood = (numberOfInputChannels == m_numberOfInputChannels) && (numberOfOutputChannels == m_numberOfOutputChannels);
     ASSERT(channelsAreGood);
     if (!channelsAreGood)
         return;
 
     for (unsigned i = 0; i < numberOfInputChannels; i++)
-        m_internalInputBus.setChannelMemory(i, inputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex, framesToProcess);
+        m_internalInputBus->setChannelMemory(i, inputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex, framesToProcess);
 
     if (numberOfInputChannels)
-        m_internalInputBus.copyFrom(*inputBus);
+        m_internalInputBus->copyFrom(*inputBus);
 
     // Copy from the output buffer to the output. 
     for (unsigned i = 0; i < numberOfOutputChannels; ++i)
         memcpy(outputBus->channel(i)->mutableData(), outputBuffer->getChannelData(i)->data() + m_bufferReadWriteIndex, sizeof(float) * framesToProcess);
 
     // Update the buffering index.
     m_bufferReadWriteIndex = (m_bufferReadWriteIndex + framesToProcess) % bufferSize();
 
     // m_bufferReadWriteIndex will wrap back around to 0 when the current input and output buffers are full.
     // When this happens, fire an event and swap buffers.
@@ skipping 83 matching lines @@
 }
 
 double ScriptProcessorNode::latencyTime() const
 {
     return std::numeric_limits<double>::infinity();
 }
 
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)