Split "Online" and "Offline" AudioContext processing

Source/platform/audio/Reverb.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
@@ skipping 71 matching lines @@
     if (response->sampleRate())
         scale *= GainCalibrationSampleRate / response->sampleRate();
 
     // True-stereo compensation
     if (response->numberOfChannels() == 4)
         scale *= 0.5f;
 
     return scale;
 }
 
-Reverb::Reverb(AudioBus* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads, bool normalize)
+Reverb::Reverb(AudioBus* impulseResponse, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool hasRealtimeConstraint, bool normalize)
 {
     float scale = 1;
 
     if (normalize) {
         scale = calculateNormalizationScale(impulseResponse);
 
         if (scale)
             impulseResponse->scale(scale);
     }
 
-    initialize(impulseResponse, renderSliceSize, maxFFTSize, numberOfChannels, useBackgroundThreads);
+    initialize(impulseResponse, renderSliceSize, maxFFTSize, numberOfChannels, hasRealtimeConstraint);
 
     // Undo scaling since this shouldn't be a destructive operation on impulseResponse.
     // FIXME: What about roundoff? Perhaps consider making a temporary scaled copy
     // instead of scaling and unscaling in place.
     if (normalize && scale)
         impulseResponse->scale(1 / scale);
 }
 
-void Reverb::initialize(AudioBus* impulseResponseBuffer, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool useBackgroundThreads)
+void Reverb::initialize(AudioBus* impulseResponseBuffer, size_t renderSliceSize, size_t maxFFTSize, size_t numberOfChannels, bool hasRealtimeConstraint)
 {
     m_impulseResponseLength = impulseResponseBuffer->length();
 
     // The reverb can handle a mono impulse response and still do stereo processing
     size_t numResponseChannels = impulseResponseBuffer->numberOfChannels();
     m_convolvers.reserveCapacity(numberOfChannels);
 
     int convolverRenderPhase = 0;
     for (size_t i = 0; i < numResponseChannels; ++i) {
         AudioChannel* channel = impulseResponseBuffer->channel(i);
 
-        OwnPtr<ReverbConvolver> convolver = adoptPtr(new ReverbConvolver(channel, renderSliceSize, maxFFTSize, convolverRenderPhase, useBackgroundThreads));
+        OwnPtr<ReverbConvolver> convolver = adoptPtr(new ReverbConvolver(channel, renderSliceSize, maxFFTSize, convolverRenderPhase, hasRealtimeConstraint));
         m_convolvers.append(convolver.release());
 
         convolverRenderPhase += renderSliceSize;
     }
 
     // For "True" stereo processing we allocate a temporary buffer to avoid repeatedly allocating it in the process() method.
     // It can be bad to allocate memory in a real-time thread.
     if (numResponseChannels == 4)
         m_tempBuffer = AudioBus::create(2, MaxFrameSize);
 }
@@ skipping 97 matching lines @@
 }
 
 size_t Reverb::latencyFrames() const
 {
     return !m_convolvers.isEmpty() ? m_convolvers.first()->latencyFrames() : 0;
 }
 
 } // namespace blink
 
 #endif // ENABLE(WEB_AUDIO)