Add UMA for hardware buffer size and callback buffer size

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

Index: third_party/WebKit/Source/platform/audio/AudioDestination.cpp
diff --git a/third_party/WebKit/Source/platform/audio/AudioDestination.cpp b/third_party/WebKit/Source/platform/audio/AudioDestination.cpp
index 28649d669065421f369ffef2790765bb29decc5a..3c105a56172bb52ac15c73412e59a1531ee22501 100644
--- a/third_party/WebKit/Source/platform/audio/AudioDestination.cpp
+++ b/third_party/WebKit/Source/platform/audio/AudioDestination.cpp
@@ -27,6 +27,8 @@
  */
 
 #include "platform/audio/AudioDestination.h"
+
+#include "platform/Histogram.h"
 #include "platform/audio/AudioFIFO.h"
 #include "platform/audio/AudioPullFIFO.h"
 #include "public/platform/Platform.h"
@@ -54,8 +56,17 @@ AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inpu
     , m_sampleRate(sampleRate)
     , m_isPlaying(false)
 {
+    // Histogram for audioHardwareBufferSize
+    DEFINE_STATIC_LOCAL(SparseHistogram, hardwareBufferSizeHistogram,
+        ("WebAudio.AudioDestination.HardwareBuffersize"));
+    // Histogram for the actual callback size used.  Typically, this is the same as
+    // audioHardwareBufferSize, but can be adjusted depending on some heuristics below.
+    DEFINE_STATIC_LOCAL(SparseHistogram, callbackBufferSizeHistogram,

[Mark P, 2016/05/03 21:35:23]

Sparse histograms are slow.  see here for details:
https://code.google.com/p/chromium/codesearch#chromium/src/base/metrics/spars...
Are you sure this is the right choice of your needs?

(I think the answer is probably yes, but you should be aware you're putting a
call that acquires a lock on every emit.)

[Raymond Toy, 2016/05/03 21:43:20]

This is updated only when the AudioContext is created, and we don't expect more
than a handful per site so speed isn't a major factor here.  

The values are also fairly sparse, ranging from a low of 128 up to at least
~3000 (or more?) and typically a power of two.

+        ("WebAudio.AudioDestination.CallbackBufferSize"));
+
     // Use the optimal buffer size recommended by the audio backend.
-    m_callbackBufferSize = Platform::current()->audioHardwareBufferSize();
+    size_t recommendedHardwareBufferSize = Platform::current()->audioHardwareBufferSize();
+    m_callbackBufferSize = recommendedHardwareBufferSize;
 
 #if OS(ANDROID)
     // The optimum low-latency hardware buffer size is usually too small on Android for WebAudio to
@@ -81,6 +92,10 @@ AudioDestination::AudioDestination(AudioIOCallback& callback, const String& inpu
     m_audioDevice = adoptPtr(Platform::current()->createAudioDevice(m_callbackBufferSize, numberOfInputChannels, numberOfOutputChannels, sampleRate, this, inputDeviceId, securityOrigin));
     ASSERT(m_audioDevice);
 
+    // Record the sizes if we successfully created an output device.
+    hardwareBufferSizeHistogram.sample(recommendedHardwareBufferSize);
+    callbackBufferSizeHistogram.sample(m_callbackBufferSize);
+
     // Create a FIFO to handle the possibility of the callback size
     // not being a multiple of the render size. If the FIFO already
     // contains enough data, the data will be provided directly.