Removing using declarations that import names in the C++ Standard library.

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 "platform/audio/VectorMath.h"
 
 #include <algorithm>
 #include <limits.h>
 #include "wtf/Complex.h"
 #include "wtf/Float32Array.h"
 #include "wtf/MainThread.h"
 #include "wtf/MathExtras.h"
 #include "wtf/Uint8Array.h"
 
-using namespace std;
-
 namespace WebCore {
 
 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.
 const unsigned RealtimeAnalyser::MinFFTSize = 32;
 const unsigned RealtimeAnalyser::MaxFFTSize = 2048;
@@ skipping 119 matching lines @@
     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 / 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);
+    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.
     float* destination = magnitudeBuffer().data();
     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 - k) * scalarMagnitude);
     }
 }
 
 void RealtimeAnalyser::getFloatFrequencyData(Float32Array* destinationArray)
 {
     ASSERT(isMainThread());
 
     if (!destinationArray)
         return;
 
     doFFTAnalysis();
 
     // Convert from linear magnitude to floating-point decibels.
     const double minDecibels = m_minDecibels;
     unsigned sourceLength = magnitudeBuffer().size();
-    size_t len = min(sourceLength, destinationArray->length());
+    size_t len = std::min(sourceLength, destinationArray->length());
     if (len > 0) {
         const float* source = magnitudeBuffer().data();
         float* destination = destinationArray->data();
 
         for (unsigned i = 0; i < len; ++i) {
             float linearValue = source[i];
             double dbMag = !linearValue ? minDecibels : AudioUtilities::linearToDecibels(linearValue);
             destination[i] = float(dbMag);
         }
     }
 }
 
 void RealtimeAnalyser::getByteFrequencyData(Uint8Array* destinationArray)
 {
     ASSERT(isMainThread());
 
     if (!destinationArray)
         return;
 
     doFFTAnalysis();
 
     // Convert from linear magnitude to unsigned-byte decibels.
     unsigned sourceLength = magnitudeBuffer().size();
-    size_t len = min(sourceLength, destinationArray->length());
+    size_t len = std::min(sourceLength, destinationArray->length());
     if (len > 0) {
         const double rangeScaleFactor = m_maxDecibels == m_minDecibels ? 1 : 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 = !linearValue ? minDecibels : AudioUtilities::linearToDecibels(linearValue);
@@ skipping 13 matching lines @@
 }
 
 void RealtimeAnalyser::getFloatTimeDomainData(Float32Array* destinationArray)
 {
     ASSERT(isMainThread());
 
     if (!destinationArray)
         return;
 
     unsigned fftSize = this->fftSize();
-    size_t len = min(fftSize, destinationArray->length());
+    size_t len = std::min(fftSize, destinationArray->length());
     if (len > 0) {
         bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize && m_inputBuffer.size() > fftSize;
         ASSERT(isInputBufferGood);
         if (!isInputBufferGood)
             return;
 
         float* inputBuffer = m_inputBuffer.data();
         float* destination = destinationArray->data();
 
         unsigned writeIndex = m_writeIndex;
 
         for (unsigned i = 0; i < len; ++i) {
             // Buffer access is protected due to modulo operation.
             float value = inputBuffer[(i + writeIndex - fftSize + InputBufferSize) % InputBufferSize];
 
             destination[i] = value;
         }
     }
 }
 
 void RealtimeAnalyser::getByteTimeDomainData(Uint8Array* destinationArray)
 {
     ASSERT(isMainThread());
 
     if (!destinationArray)
         return;
 
     unsigned fftSize = this->fftSize();
-    size_t len = min(fftSize, destinationArray->length());
+    size_t len = std::min(fftSize, destinationArray->length());
     if (len > 0) {
         bool isInputBufferGood = m_inputBuffer.size() == InputBufferSize && m_inputBuffer.size() > fftSize;
         ASSERT(isInputBufferGood);
         if (!isInputBufferGood)
             return;
 
         float* inputBuffer = m_inputBuffer.data();
         unsigned char* destination = destinationArray->data();
 
         unsigned writeIndex = m_writeIndex;
@@ skipping 12 matching lines @@
                 scaledValue = UCHAR_MAX;
 
             destination[i] = static_cast<unsigned char>(scaledValue);
         }
     }
 }
 
 } // namespace WebCore
 
 #endif // ENABLE(WEB_AUDIO)