content/renderer/media/webrtc_uma_histograms.h

   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_