| Index: Source/modules/webaudio/AudioBufferSourceNode.cpp
|
| diff --git a/Source/modules/webaudio/AudioBufferSourceNode.cpp b/Source/modules/webaudio/AudioBufferSourceNode.cpp
|
| index dc55f76d4d63b8a2b3f37d39d48124071619c55b..09f4fb975d24228e062236b44d0f22ffddb78e21 100644
|
| --- a/Source/modules/webaudio/AudioBufferSourceNode.cpp
|
| +++ b/Source/modules/webaudio/AudioBufferSourceNode.cpp
|
| @@ -38,8 +38,6 @@
|
| #include "wtf/MathExtras.h"
|
| #include <algorithm>
|
|
|
| -using namespace std;
|
| -
|
| namespace WebCore {
|
|
|
| const double DefaultGrainDuration = 0.020; // 20ms
|
| @@ -226,7 +224,7 @@ bool AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destination
|
| double loopStartFrame = m_loopStart * buffer()->sampleRate();
|
| double loopEndFrame = m_loopEnd * buffer()->sampleRate();
|
|
|
| - virtualEndFrame = min(loopEndFrame, virtualEndFrame);
|
| + virtualEndFrame = std::min(loopEndFrame, virtualEndFrame);
|
| virtualDeltaFrames = virtualEndFrame - loopStartFrame;
|
| }
|
|
|
| @@ -256,8 +254,8 @@ bool AudioBufferSourceNode::renderFromBuffer(AudioBus* bus, unsigned destination
|
| endFrame = static_cast<unsigned>(virtualEndFrame);
|
| while (framesToProcess > 0) {
|
| int framesToEnd = endFrame - readIndex;
|
| - int framesThisTime = min(framesToProcess, framesToEnd);
|
| - framesThisTime = max(0, framesThisTime);
|
| + int framesThisTime = std::min(framesToProcess, framesToEnd);
|
| + framesThisTime = std::max(0, framesThisTime);
|
|
|
| for (unsigned i = 0; i < numberOfChannels; ++i)
|
| memcpy(destinationChannels[i] + writeIndex, sourceChannels[i] + readIndex, sizeof(float) * framesThisTime);
|
| @@ -392,14 +390,14 @@ void AudioBufferSourceNode::start(double when, double grainOffset, double grainD
|
| // Do sanity checking of grain parameters versus buffer size.
|
| double bufferDuration = buffer()->duration();
|
|
|
| - grainOffset = max(0.0, grainOffset);
|
| - grainOffset = min(bufferDuration, grainOffset);
|
| + grainOffset = std::max(0.0, grainOffset);
|
| + grainOffset = std::min(bufferDuration, grainOffset);
|
| m_grainOffset = grainOffset;
|
|
|
| double maxDuration = bufferDuration - grainOffset;
|
|
|
| - grainDuration = max(0.0, grainDuration);
|
| - grainDuration = min(maxDuration, grainDuration);
|
| + grainDuration = std::max(0.0, grainDuration);
|
| + grainDuration = std::min(maxDuration, grainDuration);
|
| m_grainDuration = grainDuration;
|
|
|
| m_isGrain = true;
|
| @@ -431,10 +429,10 @@ double AudioBufferSourceNode::totalPitchRate()
|
| double totalRate = dopplerRate * sampleRateFactor * basePitchRate;
|
|
|
| // Sanity check the total rate. It's very important that the resampler not get any bad rate values.
|
| - totalRate = max(0.0, totalRate);
|
| + totalRate = std::max(0.0, totalRate);
|
| if (!totalRate)
|
| totalRate = 1; // zero rate is considered illegal
|
| - totalRate = min(MaxRate, totalRate);
|
| + totalRate = std::min(MaxRate, totalRate);
|
|
|
| bool isTotalRateValid = !std::isnan(totalRate) && !std::isinf(totalRate);
|
| ASSERT(isTotalRateValid);
|
|
|