Update webrtc/libjingle 5523:5548.

This is a clone of https://codereview.chromium.org/166753002/ + changes to handle OPT_RTP_SENDTIME_EXTN_ID in ipc_socket_factory.cc.

R=mallinath@chromium.org, wjia@chromium.org, xians@chromium.org
TBR=sergeyu

Review URL: https://codereview.chromium.org/167313002

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@251415 0039d316-1c4b-4281-b951-d872f2087c98

DEPS

@@ -54,8 +54,7 @@ vars = {
   # Three lines of non-changing comments so that
   # the commit queue can handle CLs rolling WebRTC
   # and V8 without interference from each other.
-  "webrtc_revision": "5523",
-  #"webrtc_revision": "5521",
+  "webrtc_revision": "5548",
   "jsoncpp_revision": "248",
   "nss_revision": "246067",
   # Three lines of non-changing comments so that

content/browser/media/webrtc_browsertest.cc

@@ -99,7 +99,7 @@ IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest,
 #endif
 IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream) {
   MakeTypicalPeerConnectionCall(
-      "callAndForwardRemoteStream({video: true, audio: true});");
+      "callAndForwardRemoteStream({video: true, audio: false});");
 }
 
 IN_PROC_BROWSER_TEST_F(WebRtcBrowserTest, MAYBE_CanForwardRemoteStream720p) {

content/renderer/media/mock_peer_connection_impl.h

@@ -40,6 +40,11 @@ class MockPeerConnectionImpl : public webrtc::PeerConnectionInterface {
 
   virtual bool GetStats(webrtc::StatsObserver* observer,
                         webrtc::MediaStreamTrackInterface* track) OVERRIDE;
+  virtual bool GetStats(webrtc::StatsObserver* observer,
+                        webrtc::MediaStreamTrackInterface* track,
+                        StatsOutputLevel level) OVERRIDE {
+    return false;
+  }
   // Set Call this function to make sure next call to GetStats fail.
   void SetGetStatsResult(bool result) { getstats_result_ = result; }
 

content/renderer/p2p/ipc_socket_factory.cc

@@ -48,6 +48,7 @@ bool JingleSocketOptionToP2PSocketOption(talk_base::Socket::Option option,
     case talk_base::Socket::OPT_DONTFRAGMENT:
     case talk_base::Socket::OPT_NODELAY:
     case talk_base::Socket::OPT_IPV6_V6ONLY:
+    case talk_base::Socket::OPT_RTP_SENDTIME_EXTN_ID:
       return false;  // Not supported by the chrome sockets.
     default:
       NOTREACHED();

remoting/jingle_glue/chromium_socket_factory.cc

@@ -246,6 +246,10 @@ int UdpPacketSocket::SetOption(talk_base::Socket::Option option, int value) {
     case talk_base::Socket::OPT_DSCP:
       NOTIMPLEMENTED();
       return -1;
+
+    case talk_base::Socket::OPT_RTP_SENDTIME_EXTN_ID:
+      NOTIMPLEMENTED();
+      return -1;
   }
 
   NOTREACHED();

third_party/libjingle/README.chromium

 Name: libjingle
 URL: http://code.google.com/p/webrtc/
 Version: unknown
-Revision: 5420
+Revision: 5548
 License: BSD
 License File: source/talk/COPYING
 Security Critical: yes

third_party/libjingle/libjingle.gyp

@@ -582,6 +582,7 @@
         }],
         ['use_openssl==1', {
           'sources': [
+            '<(libjingle_source)/talk/base/openssl.h',
             '<(libjingle_source)/talk/base/openssladapter.cc',
             '<(libjingle_source)/talk/base/openssldigest.cc',
             '<(libjingle_source)/talk/base/opensslidentity.cc',
@@ -679,6 +680,8 @@
             '<(libjingle_source)/talk/app/webrtc/peerconnectioninterface.h',
             '<(libjingle_source)/talk/app/webrtc/portallocatorfactory.cc',
             '<(libjingle_source)/talk/app/webrtc/portallocatorfactory.h',
+            '<(libjingle_source)/talk/app/webrtc/remoteaudiosource.cc',
+            '<(libjingle_source)/talk/app/webrtc/remoteaudiosource.h',
             '<(libjingle_source)/talk/app/webrtc/remotevideocapturer.cc',
             '<(libjingle_source)/talk/app/webrtc/remotevideocapturer.h',
             '<(libjingle_source)/talk/app/webrtc/sctputils.cc',