Merge 94002

Source/WebCore/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 31 matching lines @@
 
 using namespace std;
 
 namespace WebCore {
 
 const double RealtimeAnalyser::DefaultSmoothingTimeConstant  = 0.8;
 const double RealtimeAnalyser::DefaultMinDecibels = -100.0;
 const double RealtimeAnalyser::DefaultMaxDecibels = -30.0;
 
 const unsigned RealtimeAnalyser::DefaultFFTSize = 2048;
+// All FFT implementations are expected to handle power-of-two sizes MinFFTSize <= size <= MaxFFTSize.
+const unsigned RealtimeAnalyser::MinFFTSize = 128;
 const unsigned RealtimeAnalyser::MaxFFTSize = 2048;
 const unsigned RealtimeAnalyser::InputBufferSize = RealtimeAnalyser::MaxFFTSize * 2;
 
 RealtimeAnalyser::RealtimeAnalyser()
     : m_inputBuffer(InputBufferSize)
     , m_writeIndex(0)
     , m_fftSize(DefaultFFTSize)
     , m_magnitudeBuffer(DefaultFFTSize / 2)
     , m_smoothingTimeConstant(DefaultSmoothingTimeConstant)
     , m_minDecibels(DefaultMinDecibels)
@@ skipping 13 matching lines @@
     m_magnitudeBuffer.zero();
 }
 
 void RealtimeAnalyser::setFftSize(size_t size)
 {
     ASSERT(isMainThread());
 
     // Only allow powers of two.
     unsigned log2size = static_cast<unsigned>(log2(size));
     bool isPOT(1UL << log2size == size);
-    
-    if (!isPOT || size > MaxFFTSize) {
+
+    if (!isPOT || size > MaxFFTSize || size < MinFFTSize) {
         // FIXME: It would be good to also set an exception.
         return;
     }
 
     if (m_fftSize != size) {
-        m_analysisFrame = adoptPtr(new FFTFrame(m_fftSize));
-        m_magnitudeBuffer.allocate(size);
+        m_analysisFrame = adoptPtr(new FFTFrame(size));
+        // 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;
     }
 }
 
 void RealtimeAnalyser::writeInput(AudioBus* bus, size_t framesToProcess)
 {
     bool isBusGood = bus && bus->numberOfChannels() > 0 && bus->channel(0)->length() >= framesToProcess;
     ASSERT(isBusGood);
     if (!isBusGood)
         return;
@@ skipping 54 matching lines @@
     unsigned writeIndex = m_writeIndex;
     for (unsigned i = 0; i < fftSize; ++i)
         tempP[i] = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
     
     // Window the input samples.
     applyWindow(tempP, fftSize);
     
     // Do the analysis.
     m_analysisFrame->doFFT(tempP);
 
-    size_t n = DefaultFFTSize / 2;
-
     float* realP = m_analysisFrame->realData();
     float* imagP = m_analysisFrame->imagData();
 
     // Blow away the packed nyquist component.
     imagP[0] = 0.0f;
     
     // Normalize so than an input sine wave at 0dBfs registers as 0dBfs (undo FFT scaling factor).
     const double MagnitudeScale = 1.0 / DefaultFFTSize;
 
     // 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();
-    for (unsigned i = 0; i < n; ++i) {
+    size_t n = magnitudeBuffer().size();
+    for (size_t i = 0; i < n; ++i) {
         Complex c(realP[i], imagP[i]);
         double scalarMagnitude = abs(c) * MagnitudeScale;        
         destination[i] = float(k * destination[i] + (1.0 - k) * scalarMagnitude);
     }
 }
 
 void RealtimeAnalyser::getFloatFrequencyData(Float32Array* destinationArray)
 {
     ASSERT(isMainThread());
 
@@ skipping 88 matching lines @@
                 scaledValue = UCHAR_MAX;
             
             destination[i] = static_cast<unsigned char>(scaledValue);
         }
     }
 }
 
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)