Add UMA histograms for WebAudio

third_party/WebKit/Source/modules/webaudio/AbstractAudioContext.cpp

Index: third_party/WebKit/Source/modules/webaudio/AbstractAudioContext.cpp
diff --git a/third_party/WebKit/Source/modules/webaudio/AbstractAudioContext.cpp b/third_party/WebKit/Source/modules/webaudio/AbstractAudioContext.cpp
index fcfa28d9c3f55cbcff6d24dc9f2ce5767d40f202..089da9423c97380e6a2eb57c004c80378adf23b3 100644
--- a/third_party/WebKit/Source/modules/webaudio/AbstractAudioContext.cpp
+++ b/third_party/WebKit/Source/modules/webaudio/AbstractAudioContext.cpp
@@ -64,6 +64,7 @@
 #include "modules/webaudio/ScriptProcessorNode.h"
 #include "modules/webaudio/StereoPannerNode.h"
 #include "modules/webaudio/WaveShaperNode.h"
+#include "platform/Histogram.h"
 #include "platform/ThreadSafeFunctional.h"
 #include "platform/audio/IIRFilter.h"
 #include "public/platform/Platform.h"
@@ -206,7 +207,35 @@ AudioBuffer* AbstractAudioContext::createBuffer(unsigned numberOfChannels, size_
     // It's ok to call createBuffer, even if the context is closed because the AudioBuffer doesn't
     // really "belong" to any particular context.
 
-    return AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate, exceptionState);
+    AudioBuffer* buffer = AudioBuffer::create(numberOfChannels, numberOfFrames, sampleRate, exceptionState);
+
+    if (buffer) {
+        // Only record the data if the creation succeeded.
+        DEFINE_STATIC_LOCAL(SparseHistogram, audioBufferChannelsHistogram,
+            ("WebAudio.AudioBuffer.NumberOfChannels"));
+        DEFINE_STATIC_LOCAL(SparseHistogram, audioBufferLengthHistogram,
+            ("WebAudio.AudioBuffer.Length"));
+        DEFINE_STATIC_LOCAL(SparseHistogram, audioBufferSampleRateHistogram,
+            ("WebAudio.AudioBuffer.SampleRate"));
+
+        audioBufferChannelsHistogram.sample(numberOfChannels);
+        audioBufferLengthHistogram.sample(clampTo(numberOfFrames, 0, std::numeric_limits<int>::max()));
+        audioBufferSampleRateHistogram.sample(static_cast<int>(sampleRate));
+
+        // Compute the ratio of the buffer rate and the context rate so we know
+        // how often the buffer needs to be resampled to match the context.  For
+        // the histogram, we multiply the ratio by 100 and round to the nearest
+        // integer.  If the context is closed, don't record this because we
+        // don't have a sample rate for closed context.
+        if (!isContextClosed()) {
+            DEFINE_STATIC_LOCAL(SparseHistogram, audioBufferSampleRateRatioHistogram,
+                ("WebAudio.AudioBuffer.SampleRateRatio"));
+            float ratio = 100 * sampleRate / this->sampleRate();
+            audioBufferSampleRateRatioHistogram.sample(static_cast<int>(ratio + 0.5));
+        }
+    }
+
+    return buffer;
 }
 
 ScriptPromise AbstractAudioContext::decodeAudioData(ScriptState* scriptState, DOMArrayBuffer* audioData, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback, ExceptionState& exceptionState)