Roll WebRTC 8573:8602, Libjingle 8571:8600

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>
 
@@ skipping 390 matching lines @@
     request_->requestSucceeded(response);
     request_ = nullptr;  // must be freed on the main thread.
   }
 
   void AddReport(LocalRTCStatsResponse* response, const Report& report) {
     int idx = response->addReport(blink::WebString::fromUTF8(report.id),
                                   blink::WebString::fromUTF8(report.type),
                                   report.timestamp);
     for (const auto& value : report.values) {
       response->addStatistic(idx,
-          blink::WebString::fromUTF8(value->display_name()),
-          blink::WebString::fromUTF8(value->ToString()));
+          blink::WebString::fromUTF8(value.second->display_name()),
+          blink::WebString::fromUTF8(value.second->ToString()));
     }
   }
 
   rtc::scoped_refptr<LocalRTCStatsRequest> request_;
   const scoped_refptr<base::SingleThreadTaskRunner> main_thread_;
   base::ThreadChecker signaling_thread_checker_;
 };
 
 void GetStatsOnSignalingThread(
     const scoped_refptr<webrtc::PeerConnectionInterface>& pc,
@@ skipping 1036 matching lines @@
     base::WaitableEvent event(false, false);
     thread->PostTask(FROM_HERE,
         base::Bind(&RunSynchronousClosure, closure,
                    base::Unretained(trace_event_name),
                    base::Unretained(&event)));
     event.Wait();
   }
 }
 
 }  // namespace content