reflow comments in platform/audio

third_party/WebKit/Source/platform/audio/AudioResamplerKernel.cpp

Index: third_party/WebKit/Source/platform/audio/AudioResamplerKernel.cpp
diff --git a/third_party/WebKit/Source/platform/audio/AudioResamplerKernel.cpp b/third_party/WebKit/Source/platform/audio/AudioResamplerKernel.cpp
index 7936309d0ed0c681fa78dcb8ee2398449fca10b6..b400b1cbb4fa5c534340653bd0a68b093019c6f4 100644
--- a/third_party/WebKit/Source/platform/audio/AudioResamplerKernel.cpp
+++ b/third_party/WebKit/Source/platform/audio/AudioResamplerKernel.cpp
@@ -10,16 +10,17 @@
  *    notice, this list of conditions and the following disclaimer in the
  *    documentation and/or other materials provided with the distribution.
  *
- * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND ANY
- * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
- * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
- * DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE FOR ANY
- * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
- * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
- * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON
- * ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
- * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
- * SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS ``AS IS'' AND
+ * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
+ * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
+ * ARE DISCLAIMED. IN NO EVENT SHALL APPLE INC. OR ITS CONTRIBUTORS BE LIABLE
+ * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
+ * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
+ * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH
+ * DAMAGE.
  */
 
 #include "platform/audio/AudioResamplerKernel.h"
@@ -31,9 +32,9 @@ namespace blink {
 const size_t AudioResamplerKernel::MaxFramesToProcess = 128;
 
 AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler)
-    : m_resampler(resampler)
-      // The buffer size must be large enough to hold up to two extra sample frames for the linear interpolation.
-      ,
+    : m_resampler(resampler),
+      // The buffer size must be large enough to hold up to two extra sample
+      // frames for the linear interpolation.
       m_sourceBuffer(
           2 + static_cast<int>(MaxFramesToProcess * AudioResampler::MaxRate)),
       m_virtualReadIndex(0.0),
@@ -47,15 +48,18 @@ float* AudioResamplerKernel::getSourcePointer(
     size_t* numberOfSourceFramesNeededP) {
   ASSERT(framesToProcess <= MaxFramesToProcess);
 
-  // Calculate the next "virtual" index.  After process() is called, m_virtualReadIndex will equal this value.
+  // Calculate the next "virtual" index.  After process() is called,
+  // m_virtualReadIndex will equal this value.
   double nextFractionalIndex = m_virtualReadIndex + framesToProcess * rate();
 
-  // Because we're linearly interpolating between the previous and next sample we need to round up so we include the next sample.
+  // Because we're linearly interpolating between the previous and next sample
+  // we need to round up so we include the next sample.
   int endIndex = static_cast<int>(nextFractionalIndex +
                                   1.0);  // round up to next integer index
 
   // Determine how many input frames we'll need.
-  // We need to fill the buffer up to and including endIndex (so add 1) but we've already buffered m_fillIndex frames from last time.
+  // We need to fill the buffer up to and including endIndex (so add 1) but
+  // we've already buffered m_fillIndex frames from last time.
   size_t framesNeeded = 1 + endIndex - m_fillIndex;
   if (numberOfSourceFramesNeededP)
     *numberOfSourceFramesNeededP = framesNeeded;
@@ -111,7 +115,8 @@ void AudioResamplerKernel::process(float* destination, size_t framesToProcess) {
     virtualReadIndex += rate;
   }
 
-  // Save the last two sample-frames which will later be used at the beginning of the source buffer the next time around.
+  // Save the last two sample-frames which will later be used at the beginning
+  // of the source buffer the next time around.
   int readIndex = static_cast<int>(virtualReadIndex);
   m_lastValues[0] = source[readIndex];
   m_lastValues[1] = source[readIndex + 1];