Bug 1577932 - Remove using namespace std from dom/media r=padenot

Differential Revision: https://phabricator.services.mozilla.com/D44289

--HG--
extra : moz-landing-system : lando

dom/media/webaudio/PannerNode.cpp

@@ -23,8 +23,6 @@ using WebCore::HRTFPanner;
 namespace mozilla {
 namespace dom {
 
-using namespace std;
-
 NS_IMPL_CYCLE_COLLECTION_CLASS(PannerNode)
 NS_IMPL_CYCLE_COLLECTION_UNLINK_BEGIN_INHERITED(PannerNode, AudioNode)
   NS_IMPL_CYCLE_COLLECTION_UNLINK(mPositionX, mPositionY, mPositionZ,
@@ -448,7 +446,7 @@ void PannerNodeEngine::EqualPowerPanningFunction(const AudioBlock& aInput,
 
     // The following algorithm is described in the spec.
     // Clamp azimuth in the [-90, 90] range.
-    azimuth = min(180.f, max(-180.f, azimuth));
+    azimuth = std::min(180.f, std::max(-180.f, azimuth));
 
     // Wrap around
     if (azimuth < -90.f) {
@@ -548,7 +546,7 @@ void PannerNodeEngine::EqualPowerPanningFunction(const AudioBlock& aInput,
 
       // The following algorithm is described in the spec.
       // Clamp azimuth in the [-90, 90] range.
-      azimuth = min(180.f, max(-180.f, azimuth));
+      azimuth = std::min(180.f, std::max(-180.f, azimuth));
 
       // Wrap around
       if (azimuth < -90.f) {

dom/media/webaudio/StereoPannerNode.cpp

@@ -18,8 +18,6 @@
 namespace mozilla {
 namespace dom {
 
-using namespace std;
-
 NS_IMPL_CYCLE_COLLECTION_INHERITED(StereoPannerNode, AudioNode, mPan)
 
 NS_INTERFACE_MAP_BEGIN_CYCLE_COLLECTION(StereoPannerNode)

dom/media/webaudio/WebAudioUtils.h

@@ -38,14 +38,8 @@ const size_t MaxChannelCount = 32;
 const uint32_t MinSampleRate = 8000;
 const uint32_t MaxSampleRate = 192000;
 
-inline bool FuzzyEqual(float v1, float v2) {
-  using namespace std;
-  return fabsf(v1 - v2) < 1e-7f;
-}
-inline bool FuzzyEqual(double v1, double v2) {
-  using namespace std;
-  return fabs(v1 - v2) < 1e-7;
-}
+inline bool FuzzyEqual(float v1, float v2) { return fabsf(v1 - v2) < 1e-7f; }
+inline bool FuzzyEqual(double v1, double v2) { return fabs(v1 - v2) < 1e-7; }
 
 /**
  * Converts an AudioTimelineEvent's floating point time values to tick values
@@ -160,8 +154,7 @@ inline bool IsTimeValid(double aTime) {
  */
 template <typename IntType, typename FloatType>
 IntType TruncateFloatToInt(FloatType f) {
-  using namespace std;
-
+  using std::numeric_limits;
   static_assert(mozilla::IsIntegral<IntType>::value == true,
                 "IntType must be an integral type");
   static_assert(mozilla::IsFloatingPoint<FloatType>::value == true,

dom/media/webaudio/blink/DynamicsCompressorKernel.cpp

@@ -35,8 +35,6 @@
 #include "mozilla/FloatingPoint.h"
 #include "WebAudioUtils.h"
 
-using namespace std;
-
 using namespace mozilla::dom;  // for WebAudioUtils
 using mozilla::IsInfinite;
 using mozilla::IsNaN;
@@ -242,7 +240,7 @@ void DynamicsCompressorKernel::process(
       WebAudioUtils::ConvertDecibelsToLinear(dbPostGain) * fullRangeMakeupGain;
 
   // Attack parameters.
-  attackTime = max(0.001f, attackTime);
+  attackTime = std::max(0.001f, attackTime);
   float attackFrames = attackTime * sampleRate;
 
   // Release parameters.
@@ -337,8 +335,8 @@ void DynamicsCompressorKernel::process(
 
       // Contain within range: -12 -> 0 then scale to go from 0 -> 3
       float x = compressionDiffDb;
-      x = max(-12.0f, x);
-      x = min(0.0f, x);
+      x = std::max(-12.0f, x);
+      x = std::min(0.0f, x);
       x = 0.25f * (x + 12);
 
       // Compute adaptive release curve using 4th order polynomial.
@@ -366,7 +364,7 @@ void DynamicsCompressorKernel::process(
           m_maxAttackCompressionDiffDb < compressionDiffDb)
         m_maxAttackCompressionDiffDb = compressionDiffDb;
 
-      float effAttenDiffDb = max(0.5f, m_maxAttackCompressionDiffDb);
+      float effAttenDiffDb = std::max(0.5f, m_maxAttackCompressionDiffDb);
 
       float x = 0.25f / effAttenDiffDb;
       envelopeRate = 1 - powf(x, 1 / attackFrames);
@@ -415,7 +413,7 @@ void DynamicsCompressorKernel::process(
 
         float attenuationDb =
             -WebAudioUtils::ConvertLinearToDecibels(attenuation, -1000.0f);
-        attenuationDb = max(2.0f, attenuationDb);
+        attenuationDb = std::max(2.0f, attenuationDb);
 
         float dbPerFrame = attenuationDb / satReleaseFrames;
 
@@ -426,7 +424,7 @@ void DynamicsCompressorKernel::process(
         float rate = isRelease ? satReleaseRate : 1;
 
         detectorAverage += (attenuation - detectorAverage) * rate;
-        detectorAverage = min(1.0f, detectorAverage);
+        detectorAverage = std::min(1.0f, detectorAverage);
 
         // Fix gremlins.
         if (IsNaN(detectorAverage)) detectorAverage = 1;
@@ -439,7 +437,7 @@ void DynamicsCompressorKernel::process(
         } else {
           // Release - exponentially increase gain to 1.0
           compressorGain *= envelopeRate;
-          compressorGain = min(1.0f, compressorGain);
+          compressorGain = std::min(1.0f, compressorGain);
         }
 
         // Warp pre-compression gain to smooth out sharp exponential transition

dom/media/webaudio/blink/HRTFDatabase.cpp

@@ -30,8 +30,6 @@
 
 #include "HRTFElevation.h"
 
-using namespace std;
-
 namespace WebCore {
 
 const int HRTFDatabase::MinElevation = -45;

dom/media/webaudio/blink/HRTFElevation.cpp

@@ -34,7 +34,6 @@
 
 #include "IRC_Composite_C_R0195-incl.cpp"
 
-using namespace std;
 using namespace mozilla;
 
 namespace WebCore {
@@ -76,7 +75,7 @@ size_t HRTFElevation::fftSizeForSampleRate(float sampleRate) {
   unsigned resampledLength =
       floorf(ResponseFrameSize * sampleRate / rawSampleRate);
   // Keep things semi-sane, with max FFT size of 1024.
-  unsigned size = min(resampledLength, 1023U);
+  unsigned size = std::min(resampledLength, 1023U);
   // Ensure a minimum of 2 * WEBAUDIO_BLOCK_SIZE (with the size++ below) for
   // FFTConvolver and set the 8 least significant bits for rounding up to
   // the next power of 2 below.
@@ -242,7 +241,7 @@ nsReturnRef<HRTFElevation> HRTFElevation::createBuiltin(int elevation,
   for (unsigned rawIndex = 0; rawIndex < NumberOfRawAzimuths; ++rawIndex) {
     // Don't let elevation exceed maximum for this azimuth.
     int maxElevation = maxElevations[rawIndex];
-    int actualElevation = min(elevation, maxElevation);
+    int actualElevation = std::min(elevation, maxElevation);
 
     kernelListL[interpolatedIndex] = calculateKernelForAzimuthElevation(
         rawIndex * AzimuthSpacing, actualElevation, resampler, sampleRate);

dom/media/webaudio/blink/HRTFPanner.cpp

@@ -29,7 +29,6 @@
 #include "HRTFDatabase.h"
 #include "AudioBlock.h"
 
-using namespace std;
 using namespace mozilla;
 using dom::ChannelInterpretation;
 
@@ -111,8 +110,8 @@ int HRTFPanner::calculateDesiredAzimuthIndexAndBlend(double azimuth,
   // We don't immediately start using this azimuth index, but instead approach
   // this index from the last index we rendered at. This minimizes the clicks
   // and graininess for moving sources which occur otherwise.
-  desiredAzimuthIndex = max(0, desiredAzimuthIndex);
-  desiredAzimuthIndex = min(numberOfAzimuths - 1, desiredAzimuthIndex);
+  desiredAzimuthIndex = std::max(0, desiredAzimuthIndex);
+  desiredAzimuthIndex = std::min(numberOfAzimuths - 1, desiredAzimuthIndex);
   return desiredAzimuthIndex;
 }