Add UMA metrics and finch experiment for DTLS1.2 in WebRTC.

content/renderer/media/rtc_peer_connection_handler.cc

 // Copyright (c) 2012 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/rtc_peer_connection_handler.h"
 
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "base/command_line.h"
 #include "base/lazy_instance.h"
 #include "base/location.h"
 #include "base/logging.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/metrics/histogram.h"
+#include "base/metrics/sparse_histogram.h"
 #include "base/stl_util.h"
 #include "base/strings/utf_string_conversions.h"
 #include "base/thread_task_runner_handle.h"
 #include "base/trace_event/trace_event.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_track.h"
 #include "content/renderer/media/peer_connection_tracker.h"
 #include "content/renderer/media/remote_media_stream_impl.h"
 #include "content/renderer/media/rtc_data_channel_handler.h"
 #include "content/renderer/media/rtc_dtmf_sender_handler.h"
@@ skipping 518 matching lines @@
         UMA_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
                                   counter, counter_max);
         break;
       default:
         // The default clause is expected to reach when new enum types are
         // added.
         break;
     }
   }
 
+  void IncrementSparseEnumCounter(
+      webrtc::PeerConnectionEnumCounterType counter_type,
+      int counter) override {
+    switch (counter_type) {
+      case webrtc::kEnumCounterAudioSrtpCipher:
+        UMA_HISTOGRAM_SPARSE_SLOWLY(
+            "WebRTC.PeerConnection.SrtpCryptoSuite.Audio", counter);
+        break;
+      case webrtc::kEnumCounterAudioSslCipher:
+        UMA_HISTOGRAM_SPARSE_SLOWLY(
+            "WebRTC.PeerConnection.SslCipherSuite.Audio", counter);
+        break;
+      case webrtc::kEnumCounterVideoSrtpCipher:
+        UMA_HISTOGRAM_SPARSE_SLOWLY(
+            "WebRTC.PeerConnection.SrtpCryptoSuite.Video", counter);
+        break;
+      case webrtc::kEnumCounterVideoSslCipher:
+        UMA_HISTOGRAM_SPARSE_SLOWLY(
+            "WebRTC.PeerConnection.SslCipherSuite.Video", counter);
+        break;
+      case webrtc::kEnumCounterDataSrtpCipher:
+        UMA_HISTOGRAM_SPARSE_SLOWLY(
+            "WebRTC.PeerConnection.SrtpCryptoSuite.Data", counter);
+        break;
+      case webrtc::kEnumCounterDataSslCipher:
+        UMA_HISTOGRAM_SPARSE_SLOWLY("WebRTC.PeerConnection.SslCipherSuite.Data",
+                                    counter);
+        break;
+      default:
+        // The default clause is expected to reach when new enum types are
+        // added.
+        break;
+    }
+  }
+
   void AddHistogramSample(webrtc::PeerConnectionUMAMetricsName type,
                           int value) override {
     // Runs on libjingle's signaling thread.
     switch (type) {
       case webrtc::kTimeToConnect:
         UMA_HISTOGRAM_MEDIUM_TIMES(
             "WebRTC.PeerConnection.TimeToConnect",
             base::TimeDelta::FromMilliseconds(value));
         break;
       case webrtc::kNetworkInterfaces_IPv4:
@@ skipping 1063 matching lines @@
 }
 
 void RTCPeerConnectionHandler::ResetUMAStats() {
   DCHECK(thread_checker_.CalledOnValidThread());
   num_local_candidates_ipv6_ = 0;
   num_local_candidates_ipv4_ = 0;
   ice_connection_checking_start_ = base::TimeTicks();
   memset(ice_state_seen_, 0, sizeof(ice_state_seen_));
 }
 }  // namespace content