reflow comments in platform/audio

third_party/WebKit/Source/platform/audio/AudioDestination.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_callback(callback),
       m_numberOfOutputChannels(numberOfOutputChannels),
       m_inputBus(AudioBus::create(numberOfInputChannels, renderBufferSize)),
       m_renderBus(
           AudioBus::create(numberOfOutputChannels, renderBufferSize, false)),
       m_sampleRate(sampleRate),
       m_isPlaying(false) {
   // Histogram for audioHardwareBufferSize
   DEFINE_STATIC_LOCAL(SparseHistogram, hardwareBufferSizeHistogram,
                       ("WebAudio.AudioDestination.HardwareBufferSize"));
-  // Histogram for the actual callback size used.  Typically, this is the same as
-  // audioHardwareBufferSize, but can be adjusted depending on some heuristics below.
+  // Histogram for the actual callback size used.  Typically, this is the same
+  // as audioHardwareBufferSize, but can be adjusted depending on some
+  // heuristics below.
   DEFINE_STATIC_LOCAL(SparseHistogram, callbackBufferSizeHistogram,
                       ("WebAudio.AudioDestination.CallbackBufferSize"));
 
   // Use the optimal buffer size recommended by the audio backend.
   size_t recommendedHardwareBufferSize =
       Platform::current()->audioHardwareBufferSize();
   m_callbackBufferSize = recommendedHardwareBufferSize;
 
 #if OS(ANDROID)
-  // The optimum low-latency hardware buffer size is usually too small on Android for WebAudio to
-  // render without glitching. So, if it is small, use a larger size. If it was already large, use
-  // the requested size.
+  // The optimum low-latency hardware buffer size is usually too small on
+  // Android for WebAudio to render without glitching. So, if it is small, use
+  // a larger size. If it was already large, use the requested size.
   //
-  // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144 for a Galaxy Nexus),
-  // cause significant processing jitter. Sometimes multiple blocks will processed, but other
-  // times will not be since the FIFO can satisfy the request. By using a larger
-  // callbackBufferSize, we smooth out the jitter.
+  // Since WebAudio renders in 128-frame blocks, the small buffer sizes (144
+  // for a Galaxy Nexus), cause significant processing jitter. Sometimes
+  // multiple blocks will processed, but other times will not be since the FIFO
+  // can satisfy the request. By using a larger callbackBufferSize, we smooth
+  // out the jitter.
   const size_t kSmallBufferSize = 1024;
   const size_t kDefaultCallbackBufferSize = 2048;
 
   if (m_callbackBufferSize <= kSmallBufferSize)
     m_callbackBufferSize = kDefaultCallbackBufferSize;
 #endif
 
   // Quick exit if the requested size is too large.
   ASSERT(m_callbackBufferSize + renderBufferSize <= fifoSize);
   if (m_callbackBufferSize + renderBufferSize > fifoSize)
@@ skipping 12 matching lines @@
   // not being a multiple of the render size. If the FIFO already
   // contains enough data, the data will be provided directly.
   // Otherwise, the FIFO will call the provider enough times to
   // satisfy the request for data.
   m_fifo = wrapUnique(new AudioPullFIFO(*this, numberOfOutputChannels, fifoSize,
                                         renderBufferSize));
 
   // Input buffering.
   m_inputFifo = wrapUnique(new AudioFIFO(numberOfInputChannels, fifoSize));
 
-  // If the callback size does not match the render size, then we need to buffer some
-  // extra silence for the input. Otherwise, we can over-consume the input FIFO.
+  // If the callback size does not match the render size, then we need to
+  // buffer some extra silence for the input. Otherwise, we can over-consume
+  // the input FIFO.
   if (m_callbackBufferSize != renderBufferSize) {
     // FIXME: handle multi-channel input and don't hard-code to stereo.
     RefPtr<AudioBus> silence = AudioBus::create(2, renderBufferSize);
     m_inputFifo->push(silence.get());
   }
 }
 
 AudioDestination::~AudioDestination() {
   stop();
 }
@@ skipping 54 matching lines @@
   AudioBus* sourceBus = nullptr;
   if (m_inputFifo->framesInFifo() >= framesToProcess) {
     m_inputFifo->consume(m_inputBus.get(), framesToProcess);
     sourceBus = m_inputBus.get();
   }
 
   m_callback.render(sourceBus, bus, framesToProcess);
 }
 
 }  // namespace blink