Convert RELEASE_ASSERT()/ASSERT(...) to CHECK()/DCHECK_op(...) in platform/audio

third_party/WebKit/Source/platform/audio/HRTFPanner.cpp

Index: third_party/WebKit/Source/platform/audio/HRTFPanner.cpp
diff --git a/third_party/WebKit/Source/platform/audio/HRTFPanner.cpp b/third_party/WebKit/Source/platform/audio/HRTFPanner.cpp
index 0861ebc23ae511f239daf24310dabf9bf9c14ebf..efc2f8b405b7e49bee2492085b53eed33794628a 100644
--- a/third_party/WebKit/Source/platform/audio/HRTFPanner.cpp
+++ b/third_party/WebKit/Source/platform/audio/HRTFPanner.cpp
@@ -210,7 +210,7 @@ void HRTFPanner::pan(double desiredAzimuth,
 
   // This algorithm currently requires that we process in power-of-two size
   // chunks at least AudioUtilities::kRenderQuantumFrames.
-  ASSERT(1UL << static_cast<int>(log2(framesToProcess)) == framesToProcess);
+  DCHECK_EQ(1UL << static_cast<int>(log2(framesToProcess)), framesToProcess);
   DCHECK_GE(framesToProcess, AudioUtilities::kRenderQuantumFrames);
 
   const unsigned framesPerSegment = AudioUtilities::kRenderQuantumFrames;
@@ -240,10 +240,10 @@ void HRTFPanner::pan(double desiredAzimuth,
       return;
     }
 
-    ASSERT(frameDelayL1 / sampleRate() < MaxDelayTimeSeconds &&
-           frameDelayR1 / sampleRate() < MaxDelayTimeSeconds);
-    ASSERT(frameDelayL2 / sampleRate() < MaxDelayTimeSeconds &&
-           frameDelayR2 / sampleRate() < MaxDelayTimeSeconds);
+    DCHECK_LT(frameDelayL1 / sampleRate(), MaxDelayTimeSeconds);
+    DCHECK_LT(frameDelayR1 / sampleRate(), MaxDelayTimeSeconds);
+    DCHECK_LT(frameDelayL2 / sampleRate(), MaxDelayTimeSeconds);
+    DCHECK_LT(frameDelayR2 / sampleRate(), MaxDelayTimeSeconds);
 
     // Crossfade inter-aural delays based on transitions.
     double frameDelayL =