Split "Online" and "Offline" AudioContext processing

Source/modules/webaudio/ScriptProcessorNode.h

 /*
  * 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 16 matching lines @@
 
 #include "modules/webaudio/AudioNode.h"
 #include "platform/audio/AudioBus.h"
 #include "wtf/Forward.h"
 #include "wtf/PassRefPtr.h"
 #include "wtf/RefPtr.h"
 #include "wtf/Vector.h"
 
 namespace blink {
 
+class AbstractAudioContext;
 class AudioBuffer;
-class AudioContext;
 
 // ScriptProcessorNode is an AudioNode which allows for arbitrary synthesis or processing directly using JavaScript.
 // The API allows for a variable number of inputs and outputs, although it must have at least one input or output.
 // This basic implementation supports no more than one input and output.
 // The "onaudioprocess" attribute is an event listener which will get called periodically with an AudioProcessingEvent which has
 // AudioBuffers for each input and output.
 
 class ScriptProcessorHandler final : public AudioHandler {
 public:
     static PassRefPtr<ScriptProcessorHandler> create(AudioNode&, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
@@ skipping 45 matching lines @@
     DEFINE_WRAPPERTYPEINFO();
 public:
     // bufferSize must be one of the following values: 256, 512, 1024, 2048,
     // 4096, 8192, 16384.
     // This value controls how frequently the onaudioprocess event handler is
     // called and how many sample-frames need to be processed each call.
     // Lower numbers for bufferSize will result in a lower (better)
     // latency. Higher numbers will be necessary to avoid audio breakup and
     // glitches.
     // The value chosen must carefully balance between latency and audio quality.
-    static ScriptProcessorNode* create(AudioContext&, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
+    static ScriptProcessorNode* create(AbstractAudioContext&, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
 
     DEFINE_ATTRIBUTE_EVENT_LISTENER(audioprocess);
     size_t bufferSize() const;
 
 private:
-    ScriptProcessorNode(AudioContext&, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
+    ScriptProcessorNode(AbstractAudioContext&, float sampleRate, size_t bufferSize, unsigned numberOfInputChannels, unsigned numberOfOutputChannels);
 };
 
 } // namespace blink
 
 #endif // ScriptProcessorNode_h