| 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();
|
|
|