Add the WebRTC.webkitApiCountPerSession metric.

BUG=348616

Review URL: https://codereview.chromium.org/214983003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@260211 0039d316-1c4b-4281-b951-d872f2087c98

content/content_renderer.gypi

@@ -735,6 +735,8 @@
         'renderer/media/webrtc_local_audio_track.h',
         'renderer/media/webrtc_logging.cc',
         'renderer/media/webrtc_logging.h',
+        'renderer/media/webrtc_uma_histograms.cc',
+        'renderer/media/webrtc_uma_histograms.h',
         'renderer/p2p/host_address_request.cc',
         'renderer/p2p/host_address_request.h',
         'renderer/p2p/ipc_network_manager.cc',

content/content_tests.gypi

@@ -731,6 +731,7 @@
             'renderer/media/webrtc_identity_service_unittest.cc',
             'renderer/media/webrtc_local_audio_source_provider_unittest.cc',
             'renderer/media/webrtc_local_audio_track_unittest.cc',
+            'renderer/media/webrtc_uma_histograms_unittest.cc',
           ],
           'dependencies': [
             '../third_party/libjingle/libjingle.gyp:libjingle_webrtc',

content/renderer/media/rtc_peer_connection_handler.cc

@@ -27,6 +27,7 @@
 #include "content/renderer/media/rtc_media_constraints.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/render_thread_impl.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h"
@@ -553,6 +554,8 @@ bool RTCPeerConnectionHandler::addStream(
     peer_connection_tracker_->TrackAddStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
 
+  PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
+
   track_metrics_.AddStream(MediaStreamTrackMetrics::SENT_STREAM,
                            MediaStream::GetAdapter(stream));
 
@@ -591,6 +594,7 @@ void RTCPeerConnectionHandler::removeStream(
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackRemoveStream(
         this, stream, PeerConnectionTracker::SOURCE_LOCAL);
+  PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
   track_metrics_.RemoveStream(MediaStreamTrackMetrics::SENT_STREAM,
                               MediaStream::GetAdapter(stream));
 }
@@ -767,6 +771,8 @@ void RTCPeerConnectionHandler::OnAddStream(
         this, remote_stream->webkit_stream(),
         PeerConnectionTracker::SOURCE_REMOTE);
 
+  PerSessionWebRTCAPIMetrics::GetInstance()->IncrementStreamCounter();
+
   track_metrics_.AddStream(MediaStreamTrackMetrics::RECEIVED_STREAM,
                            stream_interface);
 
@@ -784,6 +790,7 @@ void RTCPeerConnectionHandler::OnRemoveStream(
 
   track_metrics_.RemoveStream(MediaStreamTrackMetrics::RECEIVED_STREAM,
                               stream_interface);
+  PerSessionWebRTCAPIMetrics::GetInstance()->DecrementStreamCounter();
 
   scoped_ptr<RemoteMediaStreamImpl> remote_stream(it->second);
   const blink::WebMediaStream& webkit_stream = remote_stream->webkit_stream();

content/renderer/media/webrtc_uma_histograms.cc

+// Copyright 2014 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "content/renderer/media/webrtc_uma_histograms.h"
+
+#include "base/metrics/histogram.h"
+
+namespace content {
+
+void UpdateWebRTCMethodCount(JavaScriptAPIName api_name) {
+  DVLOG(3) << "Incrementing WebRTC.webkitApiCount for " << api_name;
+  UMA_HISTOGRAM_ENUMERATION("WebRTC.webkitApiCount", api_name, INVALID_NAME);
+  PerSessionWebRTCAPIMetrics::GetInstance()->LogUsageOnlyOnce(api_name);
+}
+
+PerSessionWebRTCAPIMetrics::~PerSessionWebRTCAPIMetrics() {
+}
+
+// static
+PerSessionWebRTCAPIMetrics* PerSessionWebRTCAPIMetrics::GetInstance() {
+  return Singleton<PerSessionWebRTCAPIMetrics>::get();
+}
+
+void PerSessionWebRTCAPIMetrics::IncrementStreamCounter() {
+  DCHECK(CalledOnValidThread());
+  ++num_streams_;
+}
+
+void PerSessionWebRTCAPIMetrics::DecrementStreamCounter() {
+  DCHECK(CalledOnValidThread());
+  if (--num_streams_ == 0) {
+    ResetUsage();
+  }
+}
+
+PerSessionWebRTCAPIMetrics::PerSessionWebRTCAPIMetrics() : num_streams_(0) {
+  ResetUsage();
+}
+
+void PerSessionWebRTCAPIMetrics::LogUsage(JavaScriptAPIName api_name) {
+  DVLOG(3) << "Incrementing WebRTC.webkitApiCountPerSession for " << api_name;
+  UMA_HISTOGRAM_ENUMERATION("WebRTC.webkitApiCountPerSession",
+                            api_name, INVALID_NAME);
+}
+
+void PerSessionWebRTCAPIMetrics::LogUsageOnlyOnce(JavaScriptAPIName api_name) {
+  DCHECK(CalledOnValidThread());
+  if (!has_used_api_[api_name]) {
+    has_used_api_[api_name] = true;
+    LogUsage(api_name);
+  }
+}
+
+void PerSessionWebRTCAPIMetrics::ResetUsage() {
+  for (size_t i = 0; i < arraysize(has_used_api_); ++i)
+    has_used_api_[i] = false;
+}
+
+}  // namespace content

content/renderer/media/webrtc_uma_histograms.h

@@ -5,7 +5,9 @@
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_UMA_HISTOGRAMS_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_UMA_HISTOGRAMS_H_
 
-#include "base/metrics/histogram.h"
+#include "base/memory/singleton.h"
+#include "base/threading/non_thread_safe.h"
+#include "content/common/content_export.h"
 
 namespace content {
 
@@ -19,11 +21,61 @@ enum JavaScriptAPIName {
 };
 
 // Helper method used to collect information about the number of times
-// different WebRTC API:s are called from JavaScript.
-// The histogram can be viewed at chrome://histograms/WebRTC.webkitApiCount.
-inline void UpdateWebRTCMethodCount(JavaScriptAPIName api_name) {
-  UMA_HISTOGRAM_ENUMERATION("WebRTC.webkitApiCount", api_name, INVALID_NAME);
-}
+// different WebRTC APIs are called from JavaScript.
+//
+// This contributes to two histograms; the former is a raw count of
+// the number of times the APIs are called, and be viewed at
+// chrome://histograms/WebRTC.webkitApiCount.
+//
+// The latter is a count of the number of times the APIs are called
+// that gets incremented only once per "session" as established by the
+// PerSessionWebRTCAPIMetrics singleton below. It can be viewed at
+// chrome://histograms/WebRTC.webkitApiCountPerSession.
+void UpdateWebRTCMethodCount(JavaScriptAPIName api_name);
+
+// A singleton that keeps track of the number of MediaStreams being
+// sent over PeerConnections. It uses the transition to zero such
+// streams to demarcate the start of a new "session". Note that this
+// is a rough approximation of sessions, as you could conceivably have
+// multiple tabs using this renderer process, and each of them using
+// PeerConnections.
+//
+// The UpdateWebRTCMethodCount function above uses this class to log a
+// metric at most once per session.
+class CONTENT_EXPORT PerSessionWebRTCAPIMetrics : public base::NonThreadSafe {
+ public:
+  virtual ~PerSessionWebRTCAPIMetrics();
+
+  static PerSessionWebRTCAPIMetrics* GetInstance();
+
+  // Increment/decrement the number of streams being sent or received
+  // over any current PeerConnection.
+  void IncrementStreamCounter();
+  void DecrementStreamCounter();
+
+ protected:
+  friend struct DefaultSingletonTraits<PerSessionWebRTCAPIMetrics>;
+  friend void UpdateWebRTCMethodCount(JavaScriptAPIName);
+
+  // Protected so that unit tests can test without this being a
+  // singleton.
+  PerSessionWebRTCAPIMetrics();
+
+  // Overridable by unit tests.
+  virtual void LogUsage(JavaScriptAPIName api_name);
+
+  // Called by UpdateWebRTCMethodCount above. Protected rather than
+  // private so that unit tests can call it.
+  void LogUsageOnlyOnce(JavaScriptAPIName api_name);
+
+ private:
+  void ResetUsage();
+
+  int num_streams_;
+  bool has_used_api_[INVALID_NAME];
+
+  DISALLOW_COPY_AND_ASSIGN(PerSessionWebRTCAPIMetrics);
+};
 
 } //  namespace content
 

content/renderer/media/webrtc_uma_histograms_unittest.cc

+// Copyright 2014 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+#include "content/renderer/media/webrtc_uma_histograms.h"
+
+#include "testing/gmock/include/gmock/gmock.h"
+#include "testing/gtest/include/gtest/gtest.h"
+
+using ::testing::_;
+
+namespace content {
+
+class MockPerSessionWebRTCAPIMetrics : public PerSessionWebRTCAPIMetrics {
+ public:
+  MockPerSessionWebRTCAPIMetrics() {}
+
+  using PerSessionWebRTCAPIMetrics::LogUsageOnlyOnce;
+
+  MOCK_METHOD1(LogUsage, void(JavaScriptAPIName));
+};
+
+TEST(PerSessionWebRTCAPIMetrics, NoCallOngoingGetUserMedia) {
+  MockPerSessionWebRTCAPIMetrics metrics;
+  EXPECT_CALL(metrics, LogUsage(_)).Times(1);
+  metrics.LogUsageOnlyOnce(WEBKIT_GET_USER_MEDIA);
+}
+
+TEST(PerSessionWebRTCAPIMetrics, CallOngoingGetUserMedia) {
+  MockPerSessionWebRTCAPIMetrics metrics;
+  metrics.IncrementStreamCounter();
+  EXPECT_CALL(metrics, LogUsage(WEBKIT_GET_USER_MEDIA)).Times(1);
+  metrics.LogUsageOnlyOnce(WEBKIT_GET_USER_MEDIA);
+}
+
+TEST(PerSessionWebRTCAPIMetrics, NoCallOngoingRTCPeerConnection) {
+  MockPerSessionWebRTCAPIMetrics metrics;
+  EXPECT_CALL(metrics, LogUsage(WEBKIT_RTC_PEER_CONNECTION));
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+}
+
+TEST(PerSessionWebRTCAPIMetrics, NoCallOngoingMultiplePC) {
+  MockPerSessionWebRTCAPIMetrics metrics;
+  EXPECT_CALL(metrics, LogUsage(WEBKIT_RTC_PEER_CONNECTION)).Times(1);
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+}
+
+TEST(PerSessionWebRTCAPIMetrics, BeforeAfterCallMultiplePC) {
+  MockPerSessionWebRTCAPIMetrics metrics;
+  EXPECT_CALL(metrics, LogUsage(WEBKIT_RTC_PEER_CONNECTION)).Times(1);
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+  metrics.IncrementStreamCounter();
+  metrics.IncrementStreamCounter();
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+  metrics.DecrementStreamCounter();
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+  metrics.DecrementStreamCounter();
+  EXPECT_CALL(metrics, LogUsage(WEBKIT_RTC_PEER_CONNECTION)).Times(1);
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+  metrics.LogUsageOnlyOnce(WEBKIT_RTC_PEER_CONNECTION);
+}
+
+}  // namespace content

tools/metrics/histograms/histograms.xml

@@ -29148,6 +29148,27 @@ other types of suffix sets.
   <summary>Counts number of calls to WebRTC APIs from JavaScript.</summary>
 </histogram>
 
+<histogram name="WebRTC.webkitApiCountPerSession" enum="JavaScriptAPIName">
+  <owner>tommi@chromium.org</owner>
+  <summary>
+    Counts number of calls to WebRTC APIs from JavaScript, once   per
+    &quot;session&quot; which is demarcated by the total number of MediaStreams
+    within a renderer process that are being sent over a PeerConnection
+    transitioning from or to 0.
+  </summary>
+</histogram>
+
+<histogram name="WebRTC.webkitApiCountUniqueByOrigin" enum="JavaScriptAPIName">
+  <obsolete>
+    Deprecated as of r253828 (27 Feb 2014).
+  </obsolete>
+  <owner>tommi@chromium.org</owner>
+  <summary>
+    Counts number of calls to WebRTC APIs from JavaScript, once per origin per
+    renderer process.
+  </summary>
+</histogram>
+
 <histogram name="Webstore.ExtensionInstallResult" enum="BooleanSuccess">
   <owner>jackhou@chromium.org</owner>
   <summary>