Add UMA histograms for WebAudio

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

 /*
  * Copyright (C) 2010, Google Inc. All rights reserved.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
  * are met:
  * 1.  Redistributions of source code must retain the above copyright
  *    notice, this list of conditions and the following disclaimer.
  * 2.  Redistributions in binary form must reproduce the above copyright
  *    notice, this list of conditions and the following disclaimer in the
@@ skipping 216 matching lines @@
 void AbstractAudioContext::throwExceptionForClosedState(ExceptionState& exceptionState)
 {
     exceptionState.throwDOMException(InvalidStateError, "AudioContext has been closed.");
 }
 
 AudioBuffer* AbstractAudioContext::createBuffer(unsigned numberOfChannels, size_t numberOfFrames, float sampleRate, ExceptionState& exceptionState)
 {
     // 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);
+
+        // To limit the size of the histogram, record 10*log10(numberOfFrames),
+        // clipping the value at 60 (buffer size of 1 million frames). This
+        // gives 60 entries in the histogram.
+        double histogramValue = 10*log10(numberOfFrames);
+        audioBufferLengthHistogram.sample(clampTo(static_cast<int>(0.5 + histogramValue), 0, 60));

[Mark P, 2016/06/09 20:18:23]

If you're doing a histogram with min and max and exponentially spaced buckets
between them, you might want to use a CustomCountHistogram.  It does all this
log math for you.
(Admittedly this is no longer sparse, but it'll yield cleaner code, be faster,
and handle edge cases correctly.)

ditto everywhere below, and elsewhere in this changelist

[Raymond Toy, 2016/06/09 22:34:28]

Done.

+
+        // To limit the size of the histogram, record
+        // 40+10*log2(sampleRate/48000), clipping the value to lie between 0
+        // and 60.  These limits correspond to a min rate of 3000 and a max of
+        // 192000.  This gives 60 entries in the histogram.
+        histogramValue = 40 + 10*log2(sampleRate / 48000);
+        audioBufferSampleRateHistogram.sample(clampTo(static_cast<int>(0.5 + histogramValue), 0, 60));
+
+        // 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 = sampleRate / this->sampleRate();
+            histogramValue = clampTo(40 + 10*log2(ratio), 0.0, 60.0);
+            audioBufferSampleRateRatioHistogram.sample(static_cast<int>(0.5 + histogramValue));
+        }
+    }
+
+    return buffer;
 }
 
 ScriptPromise AbstractAudioContext::decodeAudioData(ScriptState* scriptState, DOMArrayBuffer* audioData, AudioBufferCallback* successCallback, AudioBufferCallback* errorCallback, ExceptionState& exceptionState)
 {
     ASSERT(isMainThread());
     ASSERT(audioData);
 
     ScriptPromiseResolver* resolver = ScriptPromiseResolver::create(scriptState);
     ScriptPromise promise = resolver->promise();
 
@@ skipping 542 matching lines @@
 
 SecurityOrigin* AbstractAudioContext::getSecurityOrigin() const
 {
     if (getExecutionContext())
         return getExecutionContext()->getSecurityOrigin();
 
     return nullptr;
 }
 
 } // namespace blink