reflow comments in modules/{webaudio,vr}

third_party/WebKit/Source/modules/webaudio/GainNode.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 19 matching lines @@
 #include "platform/audio/AudioBus.h"
 
 namespace blink {
 
 GainHandler::GainHandler(AudioNode& node,
                          float sampleRate,
                          AudioParamHandler& gain)
     : AudioHandler(NodeTypeGain, node, sampleRate),
       m_lastGain(1.0),
       m_gain(gain),
-      m_sampleAccurateGainValues(
-          ProcessingSizeInFrames)  // FIXME: can probably share temp buffer in context
+      m_sampleAccurateGainValues(ProcessingSizeInFrames)  // FIXME: can probably
+                                                          // share temp buffer
+                                                          // in context
 {
   addInput();
   addOutput(1);
 
   initialize();
 }
 
 PassRefPtr<GainHandler> GainHandler::create(AudioNode& node,
                                             float sampleRate,
                                             AudioParamHandler& gain) {
   return adoptRef(new GainHandler(node, sampleRate, gain));
 }
 
 void GainHandler::process(size_t framesToProcess) {
-  // FIXME: for some cases there is a nice optimization to avoid processing here, and let the gain change
-  // happen in the summing junction input of the AudioNode we're connected to.
-  // Then we can avoid all of the following:
+  // FIXME: for some cases there is a nice optimization to avoid processing
+  // here, and let the gain change happen in the summing junction input of the
+  // AudioNode we're connected to.  Then we can avoid all of the following:
 
   AudioBus* outputBus = output(0).bus();
   DCHECK(outputBus);
 
   if (!isInitialized() || !input(0).isConnected()) {
     outputBus->zero();
   } else {
     AudioBus* inputBus = input(0).bus();
 
     if (m_gain->hasSampleAccurateValues()) {
-      // Apply sample-accurate gain scaling for precise envelopes, grain windows, etc.
+      // Apply sample-accurate gain scaling for precise envelopes, grain
+      // windows, etc.
       DCHECK_LE(framesToProcess, m_sampleAccurateGainValues.size());
       if (framesToProcess <= m_sampleAccurateGainValues.size()) {
         float* gainValues = m_sampleAccurateGainValues.data();
         m_gain->calculateSampleAccurateValues(gainValues, framesToProcess);
         outputBus->copyWithSampleAccurateGainValuesFrom(*inputBus, gainValues,
                                                         framesToProcess);
         // Update m_lastGain so if the timeline ever ends, we get
         // consistent data for the smoothing below.  (Without this,
         // m_lastGain was the last value before the timeline started
         // procesing.
         m_lastGain = gainValues[framesToProcess - 1];
       }
     } else {
       // Apply the gain with de-zippering into the output bus.
       if (!m_lastGain && m_lastGain == m_gain->value()) {
-        // If the gain is 0 (and we've converged on dezippering), just zero the bus and set
-        // the silence hint.
+        // If the gain is 0 (and we've converged on dezippering), just zero the
+        // bus and set the silence hint.
         outputBus->zero();
       } else {
         outputBus->copyWithGainFrom(*inputBus, &m_lastGain, m_gain->value());
       }
     }
   }
 }
 
-// FIXME: this can go away when we do mixing with gain directly in summing junction of AudioNodeInput
+// FIXME: this can go away when we do mixing with gain directly in summing
+// junction of AudioNodeInput
 //
 // As soon as we know the channel count of our input, we can lazily initialize.
-// Sometimes this may be called more than once with different channel counts, in which case we must safely
-// uninitialize and then re-initialize with the new channel count.
+// Sometimes this may be called more than once with different channel counts, in
+// which case we must safely uninitialize and then re-initialize with the new
+// channel count.
 void GainHandler::checkNumberOfChannelsForInput(AudioNodeInput* input) {
   DCHECK(context()->isAudioThread());
   ASSERT(context()->isGraphOwner());
 
   DCHECK(input);
   DCHECK_EQ(input, &this->input(0));
   if (input != &this->input(0))
     return;
 
   unsigned numberOfChannels = input->numberOfChannels();
 
   if (isInitialized() && numberOfChannels != output(0).numberOfChannels()) {
     // We're already initialized but the channel count has changed.
     uninitialize();
   }
 
   if (!isInitialized()) {
-    // This will propagate the channel count to any nodes connected further downstream in the graph.
+    // This will propagate the channel count to any nodes connected further
+    // downstream in the graph.
     output(0).setNumberOfChannels(numberOfChannels);
     initialize();
   }
 
   AudioHandler::checkNumberOfChannelsForInput(input);
 }
 
 // ----------------------------------------------------------------
 
 GainNode::GainNode(BaseAudioContext& context)
@@ skipping 34 matching lines @@
 AudioParam* GainNode::gain() const {
   return m_gain;
 }
 
 DEFINE_TRACE(GainNode) {
   visitor->trace(m_gain);
   AudioNode::trace(visitor);
 }
 
 }  // namespace blink