Rename RTCPeerConnection.updateIce to setConfiguration and make it work.

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.h>
 
 #include <string>
 #include <utility>
@@ skipping 22 matching lines @@
 #include "content/renderer/media/webrtc/rtc_stats.h"
 #include "content/renderer/media/webrtc/webrtc_media_stream_adapter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/media/webrtc_uma_histograms.h"
 #include "content/renderer/render_thread_impl.h"
 #include "media/base/media_switches.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 #include "third_party/WebKit/public/platform/WebRTCAnswerOptions.h"
 #include "third_party/WebKit/public/platform/WebRTCConfiguration.h"
 #include "third_party/WebKit/public/platform/WebRTCDataChannelInit.h"
+#include "third_party/WebKit/public/platform/WebRTCError.h"
 #include "third_party/WebKit/public/platform/WebRTCICECandidate.h"
 #include "third_party/WebKit/public/platform/WebRTCLegacyStats.h"
 #include "third_party/WebKit/public/platform/WebRTCOfferOptions.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescription.h"
 #include "third_party/WebKit/public/platform/WebRTCSessionDescriptionRequest.h"
 #include "third_party/WebKit/public/platform/WebRTCVoidRequest.h"
 #include "third_party/WebKit/public/platform/WebURL.h"
 #include "third_party/webrtc/pc/mediasession.h"
 
 using webrtc::DataChannelInterface;
@@ skipping 100 matching lines @@
 
   std::string sdp;
   if (!native_desc->ToString(&sdp)) {
     LOG(ERROR) << "Failed to get SDP string of native session description.";
     return blink::WebRTCSessionDescription();
   }
 
   return CreateWebKitSessionDescription(sdp, native_desc->type());
 }
 
+void ConvertToWebKitRTCError(const webrtc::RTCError& webrtc_error,
+                             blink::WebRTCError* blink_error) {
+  switch (webrtc_error.type()) {
+    case webrtc::RTCErrorType::NONE:
+      blink_error->setType(blink::WebRTCErrorType::kNone);
+      break;
+    case webrtc::RTCErrorType::UNSUPPORTED_PARAMETER:
+      blink_error->setType(blink::WebRTCErrorType::kUnsupportedParameter);
+      break;
+    case webrtc::RTCErrorType::INVALID_PARAMETER:
+      blink_error->setType(blink::WebRTCErrorType::kInvalidParameter);
+      break;
+    case webrtc::RTCErrorType::INVALID_RANGE:
+      blink_error->setType(blink::WebRTCErrorType::kInvalidRange);
+      break;
+    case webrtc::RTCErrorType::SYNTAX_ERROR:
+      blink_error->setType(blink::WebRTCErrorType::kSyntaxError);
+      break;
+    case webrtc::RTCErrorType::INVALID_STATE:
+      blink_error->setType(blink::WebRTCErrorType::kInvalidState);
+      break;
+    case webrtc::RTCErrorType::INVALID_MODIFICATION:
+      blink_error->setType(blink::WebRTCErrorType::kInvalidModification);
+      break;
+    case webrtc::RTCErrorType::NETWORK_ERROR:
+      blink_error->setType(blink::WebRTCErrorType::kNetworkError);
+      break;
+    case webrtc::RTCErrorType::INTERNAL_ERROR:
+      blink_error->setType(blink::WebRTCErrorType::kInternalError);
+      break;
+    default:

[perkj_chrome, 2017/01/12 08:56:19]

nit: remove default to make sure this has to be updated if a new type is added.

[Taylor_Brandstetter, 2017/01/13 19:14:19]

Done.

+      blink_error->setType(blink::WebRTCErrorType::kInternalError);
+      break;
+  }
+}
+
 void RunClosureWithTrace(const base::Closure& closure,
                          const char* trace_event_name) {
   TRACE_EVENT0("webrtc", trace_event_name);
   closure.Run();
 }
 
 void RunSynchronousClosure(const base::Closure& closure,
                            const char* trace_event_name,
                            base::WaitableEvent* event) {
   {
@@ skipping 1214 matching lines @@
                  native_peer_connection_);
   RunSynchronousClosureOnSignalingThread(
       base::Bind(&GetSdpAndTypeFromSessionDescription, description_cb,
           base::Unretained(&sdp), base::Unretained(&type)),
       "remoteDescription");
 
   return CreateWebKitSessionDescription(sdp, type);
 }
 
 bool RTCPeerConnectionHandler::setConfiguration(
-    const blink::WebRTCConfiguration& blink_config) {
+    const blink::WebRTCConfiguration& blink_config,
+    blink::WebRTCError& error) {
   DCHECK(thread_checker_.CalledOnValidThread());
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::setConfiguration");
   GetNativeRtcConfiguration(blink_config, &configuration_);
 
   if (peer_connection_tracker_)
     peer_connection_tracker_->TrackSetConfiguration(this, configuration_);
 
-  return native_peer_connection_->SetConfiguration(configuration_);
+  webrtc::RTCError webrtc_error;
+  bool ret =
+      native_peer_connection_->SetConfiguration(configuration_, &webrtc_error);
+  ConvertToWebKitRTCError(webrtc_error, &error);
+  return ret;
 }
 
 bool RTCPeerConnectionHandler::addICECandidate(
     const blink::WebRTCVoidRequest& request,
     const blink::WebRTCICECandidate& candidate) {
   DCHECK(thread_checker_.CalledOnValidThread());
   TRACE_EVENT0("webrtc", "RTCPeerConnectionHandler::addICECandidate");
   // Libjingle currently does not accept callbacks for addICECandidate.
   // For that reason we are going to call callbacks from here.
 
@@ skipping 539 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