Upstream version 10.39.225.0

[platform/framework/web/crosswalk.git] / src / third_party / WebKit / Source / platform / audio / HRTFKernel.cpp

diff --git a/src/third_party/WebKit/Source/platform/audio/HRTFKernel.cpp b/src/third_party/WebKit/Source/platform/audio/HRTFKernel.cpp
index 1bc8bd1..8b828f9 100644 (file)
--- a/src/third_party/WebKit/Source/platform/audio/HRTFKernel.cpp
+++ b/src/third_party/WebKit/Source/platform/audio/HRTFKernel.cpp
@@ -36,8 +36,6 @@
 #include "platform/FloatConversion.h"
 #include "wtf/MathExtras.h"
 
-using namespace std;
-
 namespace blink {
 
 // Takes the input AudioChannel as an input impulse response and calculates the average group delay.
@@ -80,7 +78,7 @@ HRTFKernel::HRTFKernel(AudioChannel* channel, size_t fftSize, float sampleRate)
     size_t responseLength = channel->length();
 
     // We need to truncate to fit into 1/2 the FFT size (with zero padding) in order to do proper convolution.
-    size_t truncatedResponseLength = min(responseLength, fftSize / 2); // truncate if necessary to max impulse response length allowed by FFT
+    size_t truncatedResponseLength = std::min(responseLength, fftSize / 2); // truncate if necessary to max impulse response length allowed by FFT
 
     // Quick fade-out (apply window) at truncation point
     unsigned numberOfFadeOutFrames = static_cast<unsigned>(sampleRate / 4410); // 10 sample-frames @44.1KHz sample-rate
@@ -116,7 +114,7 @@ PassRefPtr<HRTFKernel> HRTFKernel::createInterpolatedKernel(HRTFKernel* kernel1,
         return nullptr;
 
     ASSERT(x >= 0.0 && x < 1.0);
-    x = min(1.0f, max(0.0f, x));
+    x = std::min(1.0f, std::max(0.0f, x));
 
     float sampleRate1 = kernel1->sampleRate();
     float sampleRate2 = kernel2->sampleRate();