Change WebRTC SslCipher to be exposed as number only.

talk/app/webrtc/peerconnectioninterface.h

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * 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,
@@ skipping 119 matching lines @@
 class MetricsObserverInterface : public rtc::RefCountInterface {
  public:
 
   // |type| is the type of the enum counter to be incremented. |counter|
   // is the particular counter in that type. |counter_max| is the next sequence
   // number after the highest counter.
   virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
                                     int counter,
                                     int counter_max) {}
 
+  // This is used to handle sparse counters like SSL cipher suites.
+  // TODO(guoweis): Remove the implementation once the dependency's interface
+  // definition is updated.
+  virtual void IncrementSparseEnumCounter(PeerConnectionEnumCounterType type,
+                                          int counter) {
+    IncrementEnumCounter(type, counter, 0 /* Ignored */);
+  }
+
   virtual void AddHistogramSample(PeerConnectionMetricsName type,
                                   int value) = 0;
-  // TODO(jbauch): Make method abstract when it is implemented by Chromium.
-  virtual void AddHistogramSample(PeerConnectionMetricsName type,
-                                  const std::string& value) {}
 
  protected:
   virtual ~MetricsObserverInterface() {}
 };
 
 typedef MetricsObserverInterface UMAObserver;
 
 class PeerConnectionInterface : public rtc::RefCountInterface {
  public:
   // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
@@ skipping 477 matching lines @@
 CreatePeerConnectionFactory(
     rtc::Thread* worker_thread,
     rtc::Thread* signaling_thread,
     AudioDeviceModule* default_adm,
     cricket::WebRtcVideoEncoderFactory* encoder_factory,
     cricket::WebRtcVideoDecoderFactory* decoder_factory);
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_