Index: Source/platform/audio/HRTFKernel.cpp |
diff --git a/Source/platform/audio/HRTFKernel.cpp b/Source/platform/audio/HRTFKernel.cpp |
index 1bc8bd1ca14cef6faca398f654bd290d4ff34dce..8b828f9c3655fbb05c8232bf51433f0406e684e3 100644 |
--- a/Source/platform/audio/HRTFKernel.cpp |
+++ b/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(); |