1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. 2 // Use of this source code is governed by a BSD-style license that can be 3 // found in the LICENSE file. 4 5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_UMA_HISTOGRAMS_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_UMA_HISTOGRAMS_H_ 7 8 #include "base/memory/singleton.h" 9 #include "base/threading/non_thread_safe.h" 10 #include "content/common/content_export.h" 11 #include "content/public/common/media_stream_request.h" 12 13 namespace content { 14 15 // Used to investigate where UserMediaRequests end up. 16 // Only UserMediaRequests that do not log with LogUserMediaRequestResult 17 // should call LogUserMediaRequestWithNoResult. 18 // 19 // Elements in this enum should not be deleted or rearranged; the only 20 // permitted operation is to add new elements before 21 // NUM_MEDIA_STREAM_REQUEST_WITH_NO_RESULT. 22 enum MediaStreamRequestState { 23 MEDIA_STREAM_REQUEST_EXPLICITLY_CANCELLED = 0, 24 MEDIA_STREAM_REQUEST_NOT_GENERATED = 1, 25 MEDIA_STREAM_REQUEST_PENDING_MEDIA_TRACKS = 2, 26 NUM_MEDIA_STREAM_REQUEST_WITH_NO_RESULT 27 }; 28 29 void LogUserMediaRequestWithNoResult(MediaStreamRequestState state); 30 void LogUserMediaRequestResult(MediaStreamRequestResult result); 31 32 // Helper enum used for histogramming calls to WebRTC APIs from JavaScript. 33 enum JavaScriptAPIName { 34 WEBKIT_GET_USER_MEDIA, 35 WEBKIT_PEER_CONNECTION, 36 WEBKIT_DEPRECATED_PEER_CONNECTION, 37 WEBKIT_RTC_PEER_CONNECTION, 38 WEBKIT_GET_MEDIA_DEVICES, 39 INVALID_NAME 40 }; 41 42 // Helper method used to collect information about the number of times 43 // different WebRTC APIs are called from JavaScript. 44 // 45 // This contributes to two histograms; the former is a raw count of 46 // the number of times the APIs are called, and be viewed at 47 // chrome://histograms/WebRTC.webkitApiCount. 48 // 49 // The latter is a count of the number of times the APIs are called 50 // that gets incremented only once per "session" as established by the 51 // PerSessionWebRTCAPIMetrics singleton below. It can be viewed at 52 // chrome://histograms/WebRTC.webkitApiCountPerSession. 53 void UpdateWebRTCMethodCount(JavaScriptAPIName api_name); 54 55 // A singleton that keeps track of the number of MediaStreams being 56 // sent over PeerConnections. It uses the transition to zero such 57 // streams to demarcate the start of a new "session". Note that this 58 // is a rough approximation of sessions, as you could conceivably have 59 // multiple tabs using this renderer process, and each of them using 60 // PeerConnections. 61 // 62 // The UpdateWebRTCMethodCount function above uses this class to log a 63 // metric at most once per session. 64 class CONTENT_EXPORT PerSessionWebRTCAPIMetrics : public base::NonThreadSafe { 65 public: 66 virtual ~PerSessionWebRTCAPIMetrics(); 67 68 static PerSessionWebRTCAPIMetrics* GetInstance(); 69 70 // Increment/decrement the number of streams being sent or received 71 // over any current PeerConnection. 72 void IncrementStreamCounter(); 73 void DecrementStreamCounter(); 74 75 protected: 76 friend struct DefaultSingletonTraits<PerSessionWebRTCAPIMetrics>; 77 friend void UpdateWebRTCMethodCount(JavaScriptAPIName); 78 79 // Protected so that unit tests can test without this being a 80 // singleton. 81 PerSessionWebRTCAPIMetrics(); 82 83 // Overridable by unit tests. 84 virtual void LogUsage(JavaScriptAPIName api_name); 85 86 // Called by UpdateWebRTCMethodCount above. Protected rather than 87 // private so that unit tests can call it. 88 void LogUsageOnlyOnce(JavaScriptAPIName api_name); 89 90 private: 91 void ResetUsage(); 92 93 int num_streams_; 94 bool has_used_api_[INVALID_NAME]; 95 96 DISALLOW_COPY_AND_ASSIGN(PerSessionWebRTCAPIMetrics); 97 }; 98 99 } // namespace content 100 101 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_UMA_HISTOGRAMS_H_ 102
