reflow comments in modules/{webaudio,vr}

third_party/WebKit/Source/modules/webaudio/RealtimeAnalyser.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 22 matching lines @@
 #include <complex>
 #include <limits.h>
 
 namespace blink {
 
 const double RealtimeAnalyser::DefaultSmoothingTimeConstant = 0.8;
 const double RealtimeAnalyser::DefaultMinDecibels = -100;
 const double RealtimeAnalyser::DefaultMaxDecibels = -30;
 
 const unsigned RealtimeAnalyser::DefaultFFTSize = 2048;
-// All FFT implementations are expected to handle power-of-two sizes MinFFTSize <= size <= MaxFFTSize.
+// All FFT implementations are expected to handle power-of-two sizes
+// MinFFTSize <= size <= MaxFFTSize.
 const unsigned RealtimeAnalyser::MinFFTSize = 32;
 const unsigned RealtimeAnalyser::MaxFFTSize = 32768;
 const unsigned RealtimeAnalyser::InputBufferSize =
     RealtimeAnalyser::MaxFFTSize * 2;
 
 RealtimeAnalyser::RealtimeAnalyser()
     : m_inputBuffer(InputBufferSize),
       m_writeIndex(0),
       m_downMixBus(AudioBus::create(1, AudioUtilities::kRenderQuantumFrames)),
       m_fftSize(DefaultFFTSize),
@@ skipping 10 matching lines @@
 
   // Only allow powers of two.
   unsigned log2size = static_cast<unsigned>(log2(size));
   bool isPOT(1UL << log2size == size);
 
   if (!isPOT || size > MaxFFTSize || size < MinFFTSize)
     return false;
 
   if (m_fftSize != size) {
     m_analysisFrame = wrapUnique(new FFTFrame(size));
-    // m_magnitudeBuffer has size = fftSize / 2 because it contains floats reduced from complex values in m_analysisFrame.
+    // m_magnitudeBuffer has size = fftSize / 2 because it contains floats
+    // reduced from complex values in m_analysisFrame.
     m_magnitudeBuffer.allocate(size / 2);
     m_fftSize = size;
   }
 
   return true;
 }
 
 void RealtimeAnalyser::writeInput(AudioBus* bus, size_t framesToProcess) {
   bool isBusGood = bus && bus->numberOfChannels() > 0 &&
                    bus->channel(0)->length() >= framesToProcess;
   DCHECK(isBusGood);
   if (!isBusGood)
     return;
 
   // FIXME : allow to work with non-FFTSize divisible chunking
   bool isDestinationGood =
       m_writeIndex < m_inputBuffer.size() &&
       m_writeIndex + framesToProcess <= m_inputBuffer.size();
   DCHECK(isDestinationGood);
   if (!isDestinationGood)
     return;
 
   // Perform real-time analysis
   float* dest = m_inputBuffer.data() + m_writeIndex;
 
-  // Clear the bus and downmix the input according to the down mixing rules.  Then save the result
-  // in the m_inputBuffer at the appropriate place.
+  // Clear the bus and downmix the input according to the down mixing rules.
+  // Then save the result in the m_inputBuffer at the appropriate place.
   m_downMixBus->zero();
   m_downMixBus->sumFrom(*bus);
   memcpy(dest, m_downMixBus->channel(0)->data(),
          framesToProcess * sizeof(*dest));
 
   m_writeIndex += framesToProcess;
   if (m_writeIndex >= InputBufferSize)
     m_writeIndex = 0;
 }
 
@@ skipping 14 matching lines @@
         a0 - a1 * cos(twoPiDouble * x) + a2 * cos(twoPiDouble * 2.0 * x);
     p[i] *= float(window);
   }
 }
 
 }  // namespace
 
 void RealtimeAnalyser::doFFTAnalysis() {
   DCHECK(isMainThread());
 
-  // Unroll the input buffer into a temporary buffer, where we'll apply an analysis window followed by an FFT.
+  // Unroll the input buffer into a temporary buffer, where we'll apply an
+  // analysis window followed by an FFT.
   size_t fftSize = this->fftSize();
 
   AudioFloatArray temporaryBuffer(fftSize);
   float* inputBuffer = m_inputBuffer.data();
   float* tempP = temporaryBuffer.data();
 
-  // Take the previous fftSize values from the input buffer and copy into the temporary buffer.
+  // Take the previous fftSize values from the input buffer and copy into the
+  // temporary buffer.
   unsigned writeIndex = m_writeIndex;
   if (writeIndex < fftSize) {
     memcpy(tempP, inputBuffer + writeIndex - fftSize + InputBufferSize,
            sizeof(*tempP) * (fftSize - writeIndex));
     memcpy(tempP + fftSize - writeIndex, inputBuffer,
            sizeof(*tempP) * writeIndex);
   } else {
     memcpy(tempP, inputBuffer + writeIndex - fftSize, sizeof(*tempP) * fftSize);
   }
 
   // Window the input samples.
   applyWindow(tempP, fftSize);
 
   // Do the analysis.
   m_analysisFrame->doFFT(tempP);
 
   float* realP = m_analysisFrame->realData();
   float* imagP = m_analysisFrame->imagData();
 
   // Blow away the packed nyquist component.
   imagP[0] = 0;
 
-  // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
+  // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT
+  // scaling factor).
   const double magnitudeScale = 1.0 / fftSize;
 
-  // A value of 0 does no averaging with the previous result.  Larger values produce slower, but smoother changes.
+  // A value of 0 does no averaging with the previous result.  Larger values
+  // produce slower, but smoother changes.
   double k = m_smoothingTimeConstant;
   k = std::max(0.0, k);
   k = std::min(1.0, k);
 
-  // Convert the analysis data from complex to magnitude and average with the previous result.
+  // Convert the analysis data from complex to magnitude and average with the
+  // previous result.
   float* destination = magnitudeBuffer().data();
   size_t n = magnitudeBuffer().size();
   for (size_t i = 0; i < n; ++i) {
     std::complex<double> c(realP[i], imagP[i]);
     double scalarMagnitude = abs(c) * magnitudeScale;
     destination[i] = float(k * destination[i] + (1 - k) * scalarMagnitude);
   }
 }
 
 void RealtimeAnalyser::convertFloatToDb(DOMFloat32Array* destinationArray) {
@@ skipping 39 matching lines @@
                                         : 1 / (m_maxDecibels - m_minDecibels);
     const double minDecibels = m_minDecibels;
 
     const float* source = magnitudeBuffer().data();
     unsigned char* destination = destinationArray->data();
 
     for (unsigned i = 0; i < len; ++i) {
       float linearValue = source[i];
       double dbMag = AudioUtilities::linearToDecibels(linearValue);
 
-      // The range m_minDecibels to m_maxDecibels will be scaled to byte values from 0 to UCHAR_MAX.
+      // The range m_minDecibels to m_maxDecibels will be scaled to byte values
+      // from 0 to UCHAR_MAX.
       double scaledValue = UCHAR_MAX * (dbMag - minDecibels) * rangeScaleFactor;
 
       // Clip to valid range.
       if (scaledValue < 0)
         scaledValue = 0;
       if (scaledValue > UCHAR_MAX)
         scaledValue = UCHAR_MAX;
 
       destination[i] = static_cast<unsigned char>(scaledValue);
     }
   }
 }
 
 void RealtimeAnalyser::getByteFrequencyData(DOMUint8Array* destinationArray,
                                             double currentTime) {
   DCHECK(isMainThread());
   DCHECK(destinationArray);
 
   if (currentTime <= m_lastAnalysisTime) {
-    // FIXME: Is it worth caching the data so we don't have to do the conversion every time?
-    // Perhaps not, since we expect many calls in the same rendering quantum.
+    // FIXME: Is it worth caching the data so we don't have to do the conversion
+    // every time?  Perhaps not, since we expect many calls in the same
+    // rendering quantum.
     convertToByteData(destinationArray);
     return;
   }
 
   // Time has advanced since the last call; update the FFT data.
   m_lastAnalysisTime = currentTime;
   doFFTAnalysis();
 
   convertToByteData(destinationArray);
 }
@@ skipping 58 matching lines @@
         scaledValue = 0;
       if (scaledValue > UCHAR_MAX)
         scaledValue = UCHAR_MAX;
 
       destination[i] = static_cast<unsigned char>(scaledValue);
     }
   }
 }
 
 }  // namespace blink