reflow comments in platform/audio

third_party/WebKit/Source/platform/audio/AudioBus.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
@@ skipping 20 matching lines @@
 
 #include "platform/audio/AudioChannel.h"
 #include "wtf/Noncopyable.h"
 #include "wtf/ThreadSafeRefCounted.h"
 #include "wtf/Vector.h"
 #include <memory>
 
 namespace blink {
 
 // An AudioBus represents a collection of one or more AudioChannels.
-// The data layout is "planar" as opposed to "interleaved".
-// An AudioBus with one channel is mono, an AudioBus with two channels is stereo, etc.
+// The data layout is "planar" as opposed to "interleaved".  An AudioBus with
+// one channel is mono, an AudioBus with two channels is stereo, etc.
 class PLATFORM_EXPORT AudioBus : public ThreadSafeRefCounted<AudioBus> {
   WTF_MAKE_NONCOPYABLE(AudioBus);
 
  public:
   enum {
     ChannelLeft = 0,
     ChannelRight = 1,
     ChannelCenter = 2,  // center and mono are the same
     ChannelMono = 2,
     ChannelLFE = 3,
     ChannelSurroundLeft = 4,
     ChannelSurroundRight = 5,
   };
 
   enum {
     LayoutCanonical = 0
     // Can define non-standard layouts here
   };
 
   enum ChannelInterpretation {
     Speakers,
     Discrete,
   };
 
-  // allocate indicates whether or not to initially have the AudioChannels created with managed storage.
-  // Normal usage is to pass true here, in which case the AudioChannels will memory-manage their own storage.
-  // If allocate is false then setChannelMemory() has to be called later on for each channel before the AudioBus is useable...
+  // allocate indicates whether or not to initially have the AudioChannels
+  // created with managed storage.  Normal usage is to pass true here, in which
+  // case the AudioChannels will memory-manage their own storage.  If allocate
+  // is false then setChannelMemory() has to be called later on for each
+  // channel before the AudioBus is useable...
   static PassRefPtr<AudioBus> create(unsigned numberOfChannels,
                                      size_t length,
                                      bool allocate = true);
 
   // Tells the given channel to use an externally allocated buffer.
   void setChannelMemory(unsigned channelIndex, float* storage, size_t length);
 
   // Channels
   unsigned numberOfChannels() const { return m_channels.size(); }
 
@@ skipping 27 matching lines @@
   // Returns true if the channel count and frame-size match.
   bool topologyMatches(const AudioBus& sourceBus) const;
 
   // Creates a new buffer from a range in the source buffer.
   // 0 may be returned if the range does not fit in the sourceBuffer
   static PassRefPtr<AudioBus> createBufferFromRange(
       const AudioBus* sourceBuffer,
       unsigned startFrame,
       unsigned endFrame);
 
-  // Creates a new AudioBus by sample-rate converting sourceBus to the newSampleRate.
+  // Creates a new AudioBus by sample-rate converting sourceBus to the
+  // newSampleRate.
   // setSampleRate() must have been previously called on sourceBus.
-  // Note: sample-rate conversion is already handled in the file-reading code for the mac port, so we don't need this.
+  // Note: sample-rate conversion is already handled in the file-reading code
+  // for the mac port, so we don't need this.
   static PassRefPtr<AudioBus> createBySampleRateConverting(
       const AudioBus* sourceBus,
       bool mixToMono,
       double newSampleRate);
 
   // Creates a new AudioBus by mixing all the channels down to mono.
-  // If sourceBus is already mono, then the returned AudioBus will simply be a copy.
+  // If sourceBus is already mono, then the returned AudioBus will simply be a
+  // copy.
   static PassRefPtr<AudioBus> createByMixingToMono(const AudioBus* sourceBus);
 
   // Scales all samples by the same amount.
   void scale(float scale);
 
   void reset() { m_isFirstTime = true; }  // for de-zippering
 
   // Copies the samples from the source bus to this one.
-  // This is just a simple per-channel copy if the number of channels match, otherwise an up-mix or down-mix is done.
+  // This is just a simple per-channel copy if the number of channels match,
+  // otherwise an up-mix or down-mix is done.
   void copyFrom(const AudioBus& sourceBus, ChannelInterpretation = Speakers);
 
   // Sums the samples from the source bus to this one.
-  // This is just a simple per-channel summing if the number of channels match, otherwise an up-mix or down-mix is done.
+  // This is just a simple per-channel summing if the number of channels match,
+  // otherwise an up-mix or down-mix is done.
   void sumFrom(const AudioBus& sourceBus, ChannelInterpretation = Speakers);
 
   // Copy each channel from sourceBus into our corresponding channel.
-  // We scale by targetGain (and our own internal gain m_busGain), performing "de-zippering" to smoothly change from *lastMixGain to (targetGain*m_busGain).
-  // The caller is responsible for setting up lastMixGain to point to storage which is unique for every "stream" which will be applied to this bus.
+  // We scale by targetGain (and our own internal gain m_busGain), performing
+  // "de-zippering" to smoothly change from *lastMixGain to
+  // (targetGain*m_busGain).  The caller is responsible for setting up
+  // lastMixGain to point to storage which is unique for every "stream" which
+  // will be applied to this bus.
   // This represents the dezippering memory.
   void copyWithGainFrom(const AudioBus& sourceBus,
                         float* lastMixGain,
                         float targetGain);
 
   // Copies the sourceBus by scaling with sample-accurate gain values.
   void copyWithSampleAccurateGainValuesFrom(const AudioBus& sourceBus,
                                             float* gainValues,
                                             unsigned numberOfGainValues);
 
-  // Returns maximum absolute value across all channels (useful for normalization).
+  // Returns maximum absolute value across all channels (useful for
+  // normalization).
   float maxAbsValue() const;
 
   // Makes maximum absolute value == 1.0 (if possible).
   void normalize();
 
   static PassRefPtr<AudioBus> loadPlatformResource(const char* name,
                                                    float sampleRate);
 
  protected:
   AudioBus() {}
@@ skipping 12 matching lines @@
   int m_layout;
   float m_busGain;
   std::unique_ptr<AudioFloatArray> m_dezipperGainValues;
   bool m_isFirstTime;
   float m_sampleRate;  // 0.0 if unknown or N/A
 };
 
 }  // namespace blink
 
 #endif  // AudioBus_h