Use correct FFT scaling for WebAudio AnalyserNode

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 160 matching lines @@
     // 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).
-    const double magnitudeScale = 1.0 / DefaultFFTSize;
+    const double magnitudeScale = 1.0 / fftSize;
 
     // A value of 0 does no averaging with the previous result.  Larger values produce slower, but smoother changes.
     double k = m_smoothingTimeConstant;
     k = max(0.0, k);
     k = min(1.0, k);
 
     // 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) {
@@ skipping 99 matching lines @@
                 scaledValue = UCHAR_MAX;
 
             destination[i] = static_cast<unsigned char>(scaledValue);
         }
     }
 }
 
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)