reflow comments in platform/audio

third_party/WebKit/Source/platform/audio/AudioBus.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 56 matching lines @@
   }
 
   m_layout = LayoutCanonical;  // for now this is the only layout we define
 }
 
 void AudioBus::setChannelMemory(unsigned channelIndex,
                                 float* storage,
                                 size_t length) {
   if (channelIndex < m_channels.size()) {
     channel(channelIndex)->set(storage, length);
-    m_length =
-        length;  // FIXME: verify that this length matches all the other channel lengths
+    // FIXME: verify that this length matches all the other channel lengths
+    m_length = length;
   }
 }
 
 void AudioBus::resizeSmaller(size_t newLength) {
   ASSERT(newLength <= m_length);
   if (newLength <= m_length)
     m_length = newLength;
 
   for (unsigned i = 0; i < m_channels.size(); ++i)
     m_channels[i]->resizeSmaller(newLength);
@@ skipping 399 matching lines @@
 
   AudioBus& sourceBusSafe = const_cast<AudioBus&>(sourceBus);
   const float* sources[MaxBusChannels];
   float* destinations[MaxBusChannels];
 
   for (unsigned i = 0; i < numberOfChannels; ++i) {
     sources[i] = sourceBusSafe.channel(i)->data();
     destinations[i] = channel(i)->mutableData();
   }
 
-  // We don't want to suddenly change the gain from mixing one time slice to the next,
-  // so we "de-zipper" by slowly changing the gain each sample-frame until we've achieved the target gain.
+  // We don't want to suddenly change the gain from mixing one time slice to
+  // the next, so we "de-zipper" by slowly changing the gain each sample-frame
+  // until we've achieved the target gain.
 
   // Take master bus gain into account as well as the targetGain.
   float totalDesiredGain = static_cast<float>(m_busGain * targetGain);
 
   // First time, snap directly to totalDesiredGain.
   float gain =
       static_cast<float>(m_isFirstTime ? totalDesiredGain : *lastMixGain);
   m_isFirstTime = false;
 
   const float DezipperRate = 0.005f;
   unsigned framesToProcess = length();
 
   // If the gain is within epsilon of totalDesiredGain, we can skip dezippering.
   // FIXME: this value may need tweaking.
   const float epsilon = 0.001f;
   float gainDiff = fabs(totalDesiredGain - gain);
 
-  // Number of frames to de-zipper before we are close enough to the target gain.
-  // FIXME: framesToDezipper could be smaller when target gain is close enough within this process loop.
+  // Number of frames to de-zipper before we are close enough to the target
+  // gain.
+  // FIXME: framesToDezipper could be smaller when target gain is close enough
+  // within this process loop.
   unsigned framesToDezipper = (gainDiff < epsilon) ? 0 : framesToProcess;
 
   if (framesToDezipper) {
     if (!m_dezipperGainValues.get() ||
         m_dezipperGainValues->size() < framesToDezipper)
       m_dezipperGainValues = wrapUnique(new AudioFloatArray(framesToDezipper));
 
     float* gainValues = m_dezipperGainValues->data();
     for (unsigned i = 0; i < framesToDezipper; ++i) {
       gain += (totalDesiredGain - gain) * DezipperRate;
 
-      // FIXME: If we are clever enough in calculating the framesToDezipper value, we can probably get
-      // rid of this DenormalDisabler::flushDenormalFloatToZero() call.
+      // FIXME: If we are clever enough in calculating the framesToDezipper
+      // value, we can probably get rid of this
+      // DenormalDisabler::flushDenormalFloatToZero() call.
       gain = DenormalDisabler::flushDenormalFloatToZero(gain);
       *gainValues++ = gain;
     }
 
     for (unsigned channelIndex = 0; channelIndex < numberOfChannels;
          ++channelIndex) {
       vmul(sources[channelIndex], 1, m_dezipperGainValues->data(), 1,
            destinations[channelIndex], 1, framesToDezipper);
       sources[channelIndex] += framesToDezipper;
       destinations[channelIndex] += framesToDezipper;
@@ skipping 212 matching lines @@
   // If the bus needs no conversion then return as is.
   if ((!mixToMono || audioBus->numberOfChannels() == 1) &&
       audioBus->sampleRate() == sampleRate)
     return audioBus;
 
   return AudioBus::createBySampleRateConverting(audioBus.get(), mixToMono,
                                                 sampleRate);
 }
 
 }  // namespace blink