Cleanup and prepare for bundling.

- Add a GetOptions function. Needed for eventual bundle testing to confirm that channel options are preserved. - Simplify unit tests and cleanup unused code. This is a re-roll of 8237 (https://webrtc-codereview.appspot.com/39699004) with a default GetOption implementation. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/38909004 Cr-Commit-Position: refs/heads/master@{#8245} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8245 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-04 22:03:09 +00:00 · 2015-02-04 22:03:09 +00:00 · 877ac765ad
commit 877ac765ad
parent cf7efeba37
16 changed files with 191 additions and 133 deletions
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@ -323,28 +323,16 @@ bool PeerConnection::Initialize(
    PortAllocatorFactoryInterface* allocator_factory,
    DTLSIdentityServiceInterface* dtls_identity_service,
    PeerConnectionObserver* observer) {
+  ASSERT(observer != NULL);
+  if (!observer)
+    return false;
+  observer_ = observer;
+
  std::vector<PortAllocatorFactoryInterface::StunConfiguration> stun_config;
  std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turn_config;
  if (!ParseIceServers(configuration.servers, &stun_config, &turn_config)) {
    return false;
  }
-
-  return DoInitialize(configuration.type, stun_config, turn_config, constraints,
-                      allocator_factory, dtls_identity_service, observer);
-}
-
-bool PeerConnection::DoInitialize(
-    IceTransportsType type,
-    const StunConfigurations& stun_config,
-    const TurnConfigurations& turn_config,
-    const MediaConstraintsInterface* constraints,
-    webrtc::PortAllocatorFactoryInterface* allocator_factory,
-    DTLSIdentityServiceInterface* dtls_identity_service,
-    PeerConnectionObserver* observer) {
-  ASSERT(observer != NULL);
-  if (!observer)
-    return false;
-  observer_ = observer;
  port_allocator_.reset(
      allocator_factory->CreatePortAllocator(stun_config, turn_config));

@ -384,7 +372,9 @@ bool PeerConnection::DoInitialize(

  // Initialize the WebRtcSession. It creates transport channels etc.
  if (!session_->Initialize(factory_->options(), constraints,
-                            dtls_identity_service, type))
+                            dtls_identity_service,
+                            configuration.type,
+                            configuration.bundle_policy))
    return false;

  // Register PeerConnection as receiver of local ice candidates.
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@ -506,13 +506,13 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
  // http://dev.w3.org/2011/webrtc/editor/webrtc.html
  inline rtc::scoped_refptr<PeerConnectionInterface>
      CreatePeerConnection(
-          const PeerConnectionInterface::IceServers& configuration,
+          const PeerConnectionInterface::IceServers& servers,
          const MediaConstraintsInterface* constraints,
          PortAllocatorFactoryInterface* allocator_factory,
          DTLSIdentityServiceInterface* dtls_identity_service,
          PeerConnectionObserver* observer) {
      PeerConnectionInterface::RTCConfiguration rtc_config;
-      rtc_config.servers = configuration;
+      rtc_config.servers = servers;
      return CreatePeerConnection(rtc_config, constraints, allocator_factory,
                                  dtls_identity_service, observer);
  }
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@ -466,11 +466,12 @@ class IceRestartAnswerLatch {
  bool ice_restart_;
 };

-WebRtcSession::WebRtcSession(cricket::ChannelManager* channel_manager,
-                             rtc::Thread* signaling_thread,
-                             rtc::Thread* worker_thread,
-                             cricket::PortAllocator* port_allocator,
-                             MediaStreamSignaling* mediastream_signaling)
+WebRtcSession::WebRtcSession(
+    cricket::ChannelManager* channel_manager,
+    rtc::Thread* signaling_thread,
+    rtc::Thread* worker_thread,
+    cricket::PortAllocator* port_allocator,
+    MediaStreamSignaling* mediastream_signaling)
    : cricket::BaseSession(signaling_thread,
                           worker_thread,
                           port_allocator,
@ -516,7 +517,10 @@ bool WebRtcSession::Initialize(
    const PeerConnectionFactoryInterface::Options& options,
    const MediaConstraintsInterface*  constraints,
    DTLSIdentityServiceInterface* dtls_identity_service,
-    PeerConnectionInterface::IceTransportsType ice_transport) {
+    PeerConnectionInterface::IceTransportsType ice_transport_type,
+    PeerConnectionInterface::BundlePolicy bundle_policy) {
+  bundle_policy_ = bundle_policy;
+
  // TODO(perkj): Take |constraints| into consideration. Return false if not all
  // mandatory constraints can be fulfilled. Note that |constraints|
  // can be null.
@ -659,7 +663,7 @@ bool WebRtcSession::Initialize(
    webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
  }
  port_allocator()->set_candidate_filter(
-      ConvertIceTransportTypeToCandidateFilter(ice_transport));
+      ConvertIceTransportTypeToCandidateFilter(ice_transport_type));
  return true;
 }

--- a/talk/app/webrtc/webrtcsession.h
+++ b/talk/app/webrtc/webrtcsession.h
@ -118,7 +118,8 @@ class WebRtcSession : public cricket::BaseSession,
  bool Initialize(const PeerConnectionFactoryInterface::Options& options,
                  const MediaConstraintsInterface* constraints,
                  DTLSIdentityServiceInterface* dtls_identity_service,
-                  PeerConnectionInterface::IceTransportsType ice_transport);
+                  PeerConnectionInterface::IceTransportsType ice_transport_type,
+                  PeerConnectionInterface::BundlePolicy bundle_policy);
  // Deletes the voice, video and data channel and changes the session state
  // to STATE_RECEIVEDTERMINATE.
  void Terminate();
@ -372,6 +373,9 @@ class WebRtcSession : public cricket::BaseSession,
  cricket::VideoOptions video_options_;
  MetricsObserverInterface* metrics_observer_;

+  // Declares the bundle policy for the WebRTCSession.
+  PeerConnectionInterface::BundlePolicy bundle_policy_;
+
  DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
 };
 }  // namespace webrtc
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@ -342,8 +342,7 @@ class WebRtcSessionTest : public testing::Test {
      stun_server_(cricket::TestStunServer::Create(Thread::Current(),
                                                   stun_socket_addr_)),
      turn_server_(Thread::Current(), kTurnUdpIntAddr, kTurnUdpExtAddr),
-      mediastream_signaling_(channel_manager_.get()),
-      ice_type_(PeerConnectionInterface::kAll) {
+      mediastream_signaling_(channel_manager_.get()) {
    tdesc_factory_->set_protocol(cricket::ICEPROTO_HYBRID);

    cricket::ServerAddresses stun_servers;
@ -365,11 +364,10 @@ class WebRtcSessionTest : public testing::Test {
    network_manager_.AddInterface(addr);
  }

-  void SetIceTransportType(PeerConnectionInterface::IceTransportsType type) {
-    ice_type_ = type;
-  }
-
-  void Init(DTLSIdentityServiceInterface* identity_service) {
+  void Init(
+      DTLSIdentityServiceInterface* identity_service,
+      PeerConnectionInterface::IceTransportsType ice_transport_type,
+      PeerConnectionInterface::BundlePolicy bundle_policy) {
    ASSERT_TRUE(session_.get() == NULL);
    session_.reset(new WebRtcSessionForTest(
        channel_manager_.get(), rtc::Thread::Current(),
@ -383,10 +381,35 @@ class WebRtcSessionTest : public testing::Test {
        observer_.ice_gathering_state_);

    EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
-                                     identity_service, ice_type_));
+                                     identity_service, ice_transport_type,
+                                     bundle_policy));
    session_->set_metrics_observer(&metrics_observer_);
  }

+  void Init() {
+    Init(NULL, PeerConnectionInterface::kAll,
+         PeerConnectionInterface::kBundlePolicyBalanced);
+  }
+
+  void InitWithIceTransport(
+      PeerConnectionInterface::IceTransportsType ice_transport_type) {
+    Init(NULL, ice_transport_type,
+         PeerConnectionInterface::kBundlePolicyBalanced);
+  }
+
+  void InitWithBundlePolicy(
+      PeerConnectionInterface::BundlePolicy bundle_policy) {
+    Init(NULL, PeerConnectionInterface::kAll, bundle_policy);
+  }
+
+  void InitWithDtls(bool identity_request_should_fail = false) {
+    FakeIdentityService* identity_service = new FakeIdentityService();
+    identity_service->set_should_fail(identity_request_should_fail);
+    Init(identity_service,
+         PeerConnectionInterface::kAll,
+         PeerConnectionInterface::kBundlePolicyBalanced);
+  }
+
  void InitWithDtmfCodec() {
    // Add kTelephoneEventCodec for dtmf test.
    const cricket::AudioCodec kTelephoneEventCodec(
@ -395,13 +418,7 @@ class WebRtcSessionTest : public testing::Test {
    codecs.push_back(kTelephoneEventCodec);
    media_engine_->SetAudioCodecs(codecs);
    desc_factory_->set_audio_codecs(codecs);
-    Init(NULL);
-  }
-
-  void InitWithDtls(bool identity_request_should_fail = false) {
-    FakeIdentityService* identity_service = new FakeIdentityService();
-    identity_service->set_should_fail(identity_request_should_fail);
-    Init(identity_service);
+    Init();
  }

  // Creates a local offer and applies it. Starts ice.
@ -572,7 +589,7 @@ class WebRtcSessionTest : public testing::Test {
        webrtc::MediaConstraintsInterface::kNumUnsignalledRecvStreams,
        value_set);
    session_.reset();
-    Init(NULL);
+    Init();
    mediastream_signaling_.SendAudioVideoStream1();
    SessionDescriptionInterface* offer = CreateOffer();

@ -891,7 +908,7 @@ class WebRtcSessionTest : public testing::Test {

  void TestSessionCandidatesWithBundleRtcpMux(bool bundle, bool rtcp_mux) {
    AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
-    Init(NULL);
+    Init();
    mediastream_signaling_.SendAudioVideoStream1();

    PeerConnectionInterface::RTCOfferAnswerOptions options;
@ -943,7 +960,7 @@ class WebRtcSessionTest : public testing::Test {
    if (can) {
      InitWithDtmfCodec();
    } else {
-      Init(NULL);
+      Init();
    }
    mediastream_signaling_.SendAudioVideoStream1();
    CreateAndSetRemoteOfferAndLocalAnswer();
@ -1050,7 +1067,7 @@ class WebRtcSessionTest : public testing::Test {

  void TestLoopbackCall(const LoopbackNetworkConfiguration& config) {
    LoopbackNetworkManager loopback_network_manager(this, config);
-    Init(NULL);
+    Init();
    mediastream_signaling_.SendAudioVideoStream1();
    SessionDescriptionInterface* offer = CreateOffer();

@ -1240,7 +1257,6 @@ class WebRtcSessionTest : public testing::Test {
  MockIceObserver observer_;
  cricket::FakeVideoMediaChannel* video_channel_;
  cricket::FakeVoiceMediaChannel* voice_channel_;
-  PeerConnectionInterface::IceTransportsType ice_type_;
  FakeMetricsObserver metrics_observer_;
 };

@ -1251,7 +1267,7 @@ TEST_F(WebRtcSessionTest, TestInitializeWithDtls) {
 }

 TEST_F(WebRtcSessionTest, TestInitializeWithoutDtls) {
-  Init(NULL);
+  Init();
  // SDES is required if DTLS is off.
  EXPECT_EQ(cricket::SEC_REQUIRED, session_->SdesPolicy());
 }
@ -1273,7 +1289,7 @@ TEST_F(WebRtcSessionTest, TestSessionCandidatesWithBundleRtcpMux) {
 TEST_F(WebRtcSessionTest, TestMultihomeCandidates) {
  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
  AddInterface(rtc::SocketAddress(kClientAddrHost2, kClientAddrPort));
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  InitiateCall();
  EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
@ -1288,7 +1304,7 @@ TEST_F(WebRtcSessionTest, TestStunError) {
                rtc::FP_UDP,
                rtc::FD_ANY,
                rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  InitiateCall();
  // Since kClientAddrHost1 is blocked, not expecting stun candidates for it.
@ -1300,8 +1316,7 @@ TEST_F(WebRtcSessionTest, TestStunError) {
 // Test session delivers no candidates gathered when constraint set to "none".
 TEST_F(WebRtcSessionTest, TestIceTransportsNone) {
  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
-  SetIceTransportType(PeerConnectionInterface::kNone);
-  Init(NULL);
+  InitWithIceTransport(PeerConnectionInterface::kNone);
  mediastream_signaling_.SendAudioVideoStream1();
  InitiateCall();
  EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
@ -1314,8 +1329,7 @@ TEST_F(WebRtcSessionTest, TestIceTransportsNone) {
 TEST_F(WebRtcSessionTest, TestIceTransportsRelay) {
  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
  ConfigureAllocatorWithTurn();
-  SetIceTransportType(PeerConnectionInterface::kRelay);
-  Init(NULL);
+  InitWithIceTransport(PeerConnectionInterface::kRelay);
  mediastream_signaling_.SendAudioVideoStream1();
  InitiateCall();
  EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
@ -1334,8 +1348,7 @@ TEST_F(WebRtcSessionTest, TestIceTransportsRelay) {
 // Test session delivers all candidates gathered when constaint set to "all".
 TEST_F(WebRtcSessionTest, TestIceTransportsAll) {
  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
-  SetIceTransportType(PeerConnectionInterface::kAll);
-  Init(NULL);
+  InitWithIceTransport(PeerConnectionInterface::kAll);
  mediastream_signaling_.SendAudioVideoStream1();
  InitiateCall();
  EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout);
@ -1345,7 +1358,7 @@ TEST_F(WebRtcSessionTest, TestIceTransportsAll) {
 }

 TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
-  Init(NULL);
+  Init();
  SessionDescriptionInterface* offer = NULL;
  // Since |offer| is NULL, there's no way to tell if it's an offer or answer.
  std::string unknown_action;
@ -1356,7 +1369,7 @@ TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) {
 // Test creating offers and receive answers and make sure the
 // media engine creates the expected send and receive streams.
 TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();
  const std::string session_id_orig = offer->session_id();
@ -1410,7 +1423,7 @@ TEST_F(WebRtcSessionTest, TestCreateSdesOfferReceiveSdesAnswer) {
 // Test receiving offers and creating answers and make sure the
 // media engine creates the expected send and receive streams.
 TEST_F(WebRtcSessionTest, TestReceiveSdesOfferCreateSdesAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream2();
  SessionDescriptionInterface* offer = CreateOffer();
  VerifyCryptoParams(offer->description());
@ -1466,7 +1479,7 @@ TEST_F(WebRtcSessionTest, TestReceiveSdesOfferCreateSdesAnswer) {
 }

 TEST_F(WebRtcSessionTest, SetLocalSdpFailedOnCreateChannel) {
-  Init(NULL);
+  Init();
  media_engine_->set_fail_create_channel(true);

  SessionDescriptionInterface* offer = CreateOffer();
@ -1512,7 +1525,7 @@ TEST_F(WebRtcSessionTest, SetLocalSdpFailedOnCreateChannel) {
 // Test that we return a failure when applying a remote/local offer that doesn't
 // have cryptos enabled when DTLS is off.
 TEST_F(WebRtcSessionTest, TestSetNonSdesOfferWhenSdesOn) {
-  Init(NULL);
+  Init();
  cricket::MediaSessionOptions options;
  options.recv_video = true;
  JsepSessionDescription* offer = CreateRemoteOffer(
@ -1530,7 +1543,7 @@ TEST_F(WebRtcSessionTest, TestSetNonSdesOfferWhenSdesOn) {
 // Test that we return a failure when applying a local answer that doesn't have
 // cryptos enabled when DTLS is off.
 TEST_F(WebRtcSessionTest, TestSetLocalNonSdesAnswerWhenSdesOn) {
-  Init(NULL);
+  Init();
  SessionDescriptionInterface* offer = NULL;
  SessionDescriptionInterface* answer = NULL;
  CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
@ -1543,7 +1556,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalNonSdesAnswerWhenSdesOn) {
 // Test we will return fail when apply an remote answer that doesn't have
 // crypto enabled when DTLS is off.
 TEST_F(WebRtcSessionTest, TestSetRemoteNonSdesAnswerWhenSdesOn) {
-  Init(NULL);
+  Init();
  SessionDescriptionInterface* offer = NULL;
  SessionDescriptionInterface* answer = NULL;
  CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
@ -1728,7 +1741,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerReceiveOfferWithoutEncryption) {
 }

 TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendNothing();
  // SetLocalDescription take ownership of offer.
  SessionDescriptionInterface* offer = CreateOffer();
@ -1740,7 +1753,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) {
 }

 TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendNothing();
  // SetLocalDescription take ownership of offer.
  SessionDescriptionInterface* offer = CreateOffer();
@ -1751,7 +1764,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) {
 }

 TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendNothing();
  SessionDescriptionInterface* offer = CreateOffer();
  SetLocalDescriptionWithoutError(offer);
@ -1761,7 +1774,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
 }

 TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendNothing();
  SessionDescriptionInterface* offer = CreateOffer();
  SetRemoteDescriptionWithoutError(offer);
@ -1771,7 +1784,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
 }

 TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendNothing();
  SessionDescriptionInterface* offer = CreateRemoteOffer();
  SetRemoteDescriptionExpectState(offer, BaseSession::STATE_RECEIVEDINITIATE);
@ -1794,7 +1807,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) {
 }

 TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendNothing();
  SessionDescriptionInterface* offer = CreateOffer();
  SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE);
@ -1821,7 +1834,7 @@ TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) {
 }

 TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendNothing();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());

@ -1832,7 +1845,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
 }

 TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendNothing();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());

@ -1843,7 +1856,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
 }

 TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();

  cricket::Candidate candidate;
@ -1892,7 +1905,7 @@ TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
 // Test that a remote candidate is added to the remote session description and
 // that it is retained if the remote session description is changed.
 TEST_F(WebRtcSessionTest, TestRemoteCandidatesAddedToSessionDescription) {
-  Init(NULL);
+  Init();
  cricket::Candidate candidate1;
  candidate1.set_component(1);
  JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0,
@ -1945,7 +1958,7 @@ TEST_F(WebRtcSessionTest, TestRemoteCandidatesAddedToSessionDescription) {
 // that they are retained if the local session description is changed.
 TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {
  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  CreateAndSetRemoteOfferAndLocalAnswer();

@ -1980,7 +1993,7 @@ TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) {

 // Test that we can set a remote session description with remote candidates.
 TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
-  Init(NULL);
+  Init();

  cricket::Candidate candidate1;
  candidate1.set_component(1);
@ -2009,7 +2022,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) {
 // been gathered.
 TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
  AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort));
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  // Ice is started but candidates are not provided until SetLocalDescription
  // is called.
@ -2042,7 +2055,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) {
 // Verifies TransportProxy and media channels are created with content names
 // present in the SessionDescription.
 TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());

@ -2086,7 +2099,7 @@ TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) {
 // Test that an offer contains the correct media content descriptions based on
 // the send streams when no constraints have been set.
 TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
-  Init(NULL);
+  Init();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());

  ASSERT_TRUE(offer != NULL);
@ -2100,7 +2113,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraintsOrStreams) {
 // Test that an offer contains the correct media content descriptions based on
 // the send streams when no constraints have been set.
 TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
-  Init(NULL);
+  Init();
  // Test Audio only offer.
  mediastream_signaling_.UseOptionsAudioOnly();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
@ -2123,7 +2136,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutConstraints) {
 // Test that an offer contains no media content descriptions if
 // kOfferToReceiveVideo and kOfferToReceiveAudio constraints are set to false.
 TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
-  Init(NULL);
+  Init();
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_audio = 0;
  options.offer_to_receive_video = 0;
@ -2142,7 +2155,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithConstraintsWithoutStreams) {
 // Test that an offer contains only audio media content descriptions if
 // kOfferToReceiveAudio constraints are set to true.
 TEST_F(WebRtcSessionTest, CreateAudioOnlyOfferWithConstraints) {
-  Init(NULL);
+  Init();
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_audio =
      RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
@ -2160,7 +2173,7 @@ TEST_F(WebRtcSessionTest, CreateAudioOnlyOfferWithConstraints) {
 // Test that an offer contains audio and video media content descriptions if
 // kOfferToReceiveAudio and kOfferToReceiveVideo constraints are set to true.
 TEST_F(WebRtcSessionTest, CreateOfferWithConstraints) {
-  Init(NULL);
+  Init();
  // Test Audio / Video offer.
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_audio =
@ -2193,7 +2206,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithConstraints) {
 // Test that an answer can not be created if the last remote description is not
 // an offer.
 TEST_F(WebRtcSessionTest, CreateAnswerWithoutAnOffer) {
-  Init(NULL);
+  Init();
  SessionDescriptionInterface* offer = CreateOffer();
  SetLocalDescriptionWithoutError(offer);
  SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
@ -2204,7 +2217,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithoutAnOffer) {
 // Test that an answer contains the correct media content descriptions when no
 // constraints have been set.
 TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraintsOrStreams) {
-  Init(NULL);
+  Init();
  // Create a remote offer with audio and video content.
  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
  SetRemoteDescriptionWithoutError(offer.release());
@ -2223,7 +2236,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraintsOrStreams) {
 // Test that an answer contains the correct media content descriptions when no
 // constraints have been set and the offer only contain audio.
 TEST_F(WebRtcSessionTest, CreateAudioAnswerWithoutConstraintsOrStreams) {
-  Init(NULL);
+  Init();
  // Create a remote offer with audio only.
  cricket::MediaSessionOptions options;

@ -2246,7 +2259,7 @@ TEST_F(WebRtcSessionTest, CreateAudioAnswerWithoutConstraintsOrStreams) {
 // Test that an answer contains the correct media content descriptions when no
 // constraints have been set.
 TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
-  Init(NULL);
+  Init();
  // Create a remote offer with audio and video content.
  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
  SetRemoteDescriptionWithoutError(offer.release());
@ -2267,7 +2280,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithoutConstraints) {
 // Test that an answer contains the correct media content descriptions when
 // constraints have been set but no stream is sent.
 TEST_F(WebRtcSessionTest, CreateAnswerWithConstraintsWithoutStreams) {
-  Init(NULL);
+  Init();
  // Create a remote offer with audio and video content.
  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
  SetRemoteDescriptionWithoutError(offer.release());
@ -2291,7 +2304,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithConstraintsWithoutStreams) {
 // Test that an answer contains the correct media content descriptions when
 // constraints have been set and streams are sent.
 TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
-  Init(NULL);
+  Init();
  // Create a remote offer with audio and video content.
  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
  SetRemoteDescriptionWithoutError(offer.release());
@ -2319,7 +2332,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {

 TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) {
  AddCNCodecs();
-  Init(NULL);
+  Init();
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_audio =
      RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
@ -2336,7 +2349,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) {

 TEST_F(WebRtcSessionTest, CreateAnswerWithoutCNCodecs) {
  AddCNCodecs();
-  Init(NULL);
+  Init();
  // Create a remote offer with audio and video content.
  rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
  SetRemoteDescriptionWithoutError(offer.release());
@ -2354,7 +2367,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithoutCNCodecs) {
 // This test verifies the call setup when remote answer with audio only and
 // later updates with video.
 TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
-  Init(NULL);
+  Init();
  EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
  EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);

@ -2411,7 +2424,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithAudioOnlyAnswer) {
 // This test verifies the call setup when remote answer with video only and
 // later updates with audio.
 TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
-  Init(NULL);
+  Init();
  EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL);
  EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL);
  mediastream_signaling_.SendAudioVideoStream1();
@ -2464,7 +2477,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) {
 }

 TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
  VerifyCryptoParams(offer->description());
@ -2475,26 +2488,26 @@ TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) {

 TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
  options_.disable_encryption = true;
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
  VerifyNoCryptoParams(offer->description(), false);
 }

 TEST_F(WebRtcSessionTest, VerifyAnswerFromNonCryptoOffer) {
-  Init(NULL);
+  Init();
  VerifyAnswerFromNonCryptoOffer();
 }

 TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) {
-  Init(NULL);
+  Init();
  VerifyAnswerFromCryptoOffer();
 }

 // This test verifies that setLocalDescription fails if
 // no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
 TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());

@ -2508,7 +2521,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalDescriptionWithoutIce) {
 // This test verifies that setRemoteDescription fails if
 // no a=ice-ufrag and a=ice-pwd lines are present in the SDP.
 TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) {
-  Init(NULL);
+  Init();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
  std::string sdp;
  RemoveIceUfragPwdLines(offer.get(), &sdp);
@ -2520,7 +2533,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionWithoutIce) {
 // This test verifies that setLocalDescription fails if local offer has
 // too short ice ufrag and pwd strings.
 TEST_F(WebRtcSessionTest, TestSetLocalDescriptionInvalidIceCredentials) {
-  Init(NULL);
+  Init();
  tdesc_factory_->set_protocol(cricket::ICEPROTO_RFC5245);
  mediastream_signaling_.SendAudioVideoStream1();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
@ -2546,7 +2559,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalDescriptionInvalidIceCredentials) {
 // This test verifies that setRemoteDescription fails if remote offer has
 // too short ice ufrag and pwd strings.
 TEST_F(WebRtcSessionTest, TestSetRemoteDescriptionInvalidIceCredentials) {
-  Init(NULL);
+  Init();
  tdesc_factory_->set_protocol(cricket::ICEPROTO_RFC5245);
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
  std::string sdp;
@ -2570,7 +2583,7 @@ TEST_F(WebRtcSessionTest, VerifyBundleFlagInPA) {
  // This test verifies BUNDLE flag in PortAllocator, if BUNDLE information in
  // local description is removed by the application, BUNDLE flag should be
  // disabled in PortAllocator. By default BUNDLE is enabled in the WebRtc.
-  Init(NULL);
+  Init();
  EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE &
      allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE);
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
@ -2587,7 +2600,7 @@ TEST_F(WebRtcSessionTest, VerifyBundleFlagInPA) {
 }

 TEST_F(WebRtcSessionTest, TestDisabledBundleInAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE &
      allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE);
@ -2628,7 +2641,7 @@ TEST_F(WebRtcSessionTest, TestDisabledBundleInAnswer) {
 // This test verifies that SetLocalDescription and SetRemoteDescription fails
 // if BUNDLE is enabled but rtcp-mux is disabled in m-lines.
 TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
-  WebRtcSessionTest::Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE &
      allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE);
@ -2658,7 +2671,7 @@ TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) {
 }

 TEST_F(WebRtcSessionTest, SetAudioPlayout) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  CreateAndSetRemoteOfferAndLocalAnswer();
  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@ -2683,7 +2696,7 @@ TEST_F(WebRtcSessionTest, SetAudioPlayout) {
 }

 TEST_F(WebRtcSessionTest, SetAudioSend) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  CreateAndSetRemoteOfferAndLocalAnswer();
  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@ -2713,7 +2726,7 @@ TEST_F(WebRtcSessionTest, SetAudioSend) {
 }

 TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  CreateAndSetRemoteOfferAndLocalAnswer();
  cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@ -2736,7 +2749,7 @@ TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) {
 }

 TEST_F(WebRtcSessionTest, SetVideoPlayout) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  CreateAndSetRemoteOfferAndLocalAnswer();
  cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
@ -2753,7 +2766,7 @@ TEST_F(WebRtcSessionTest, SetVideoPlayout) {
 }

 TEST_F(WebRtcSessionTest, SetVideoSend) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  CreateAndSetRemoteOfferAndLocalAnswer();
  cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
@ -2778,7 +2791,7 @@ TEST_F(WebRtcSessionTest, CanInsertDtmf) {

 TEST_F(WebRtcSessionTest, InsertDtmf) {
  // Setup
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  CreateAndSetRemoteOfferAndLocalAnswer();
  FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@ -2804,7 +2817,7 @@ TEST_F(WebRtcSessionTest, InsertDtmf) {

 // This test verifies the |initiator| flag when session initiates the call.
 TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) {
-  Init(NULL);
+  Init();
  EXPECT_FALSE(session_->initiator());
  SessionDescriptionInterface* offer = CreateOffer();
  SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
@ -2816,7 +2829,7 @@ TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) {

 // This test verifies the |initiator| flag when session receives the call.
 TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) {
-  Init(NULL);
+  Init();
  EXPECT_FALSE(session_->initiator());
  SessionDescriptionInterface* offer = CreateRemoteOffer();
  SetRemoteDescriptionWithoutError(offer);
@ -2830,7 +2843,7 @@ TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) {
 // This test verifies the ice protocol type at initiator of the call
 // if |a=ice-options:google-ice| is present in answer.
 TEST_F(WebRtcSessionTest, TestInitiatorGIceInAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();
  rtc::scoped_ptr<SessionDescriptionInterface> answer(
@ -2852,7 +2865,7 @@ TEST_F(WebRtcSessionTest, TestInitiatorGIceInAnswer) {
 // This test verifies the ice protocol type at initiator of the call
 // if ICE RFC5245 is supported in answer.
 TEST_F(WebRtcSessionTest, TestInitiatorIceInAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();
  SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
@ -2866,7 +2879,7 @@ TEST_F(WebRtcSessionTest, TestInitiatorIceInAnswer) {
 // This test verifies the ice protocol type at receiver side of the call if
 // receiver decides to use google-ice.
 TEST_F(WebRtcSessionTest, TestReceiverGIceInOffer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();
  SetRemoteDescriptionWithoutError(offer);
@ -2888,7 +2901,7 @@ TEST_F(WebRtcSessionTest, TestReceiverGIceInOffer) {
 // This test verifies the ice protocol type at receiver side of the call if
 // receiver decides to use ice RFC 5245.
 TEST_F(WebRtcSessionTest, TestReceiverIceInOffer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();
  SetRemoteDescriptionWithoutError(offer);
@ -2901,7 +2914,7 @@ TEST_F(WebRtcSessionTest, TestReceiverIceInOffer) {
 // This test verifies the session state when ICE RFC5245 in offer and
 // ICE google-ice in answer.
 TEST_F(WebRtcSessionTest, TestIceOfferGIceOnlyAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());

@ -2933,7 +2946,7 @@ TEST_F(WebRtcSessionTest, TestIceOfferGIceOnlyAnswer) {

 // Verifing local offer and remote answer have matching m-lines as per RFC 3264.
 TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();
  SetLocalDescriptionWithoutError(offer);
@ -2981,7 +2994,7 @@ TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) {
 // Verifying remote offer and local answer have matching m-lines as per
 // RFC 3264.
 TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateRemoteOffer();
  SetRemoteDescriptionWithoutError(offer);
@ -3002,7 +3015,7 @@ TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) {
 // This test verifies that WebRtcSession does not start candidate allocation
 // before SetLocalDescription is called.
 TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateRemoteOffer();
  cricket::Candidate candidate;
@ -3034,7 +3047,7 @@ TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) {
 // This test verifies that crypto parameter is updated in local session
 // description as per security policy set in MediaSessionDescriptionFactory.
 TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());

@ -3053,7 +3066,7 @@ TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) {
 // This test verifies the crypto parameter when security is disabled.
 TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
  options_.disable_encryption = true;
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());

@ -3072,7 +3085,7 @@ TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) {
 // This test verifies that an answer contains new ufrag and password if an offer
 // with new ufrag and password is received.
 TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
-  Init(NULL);
+  Init();
  cricket::MediaSessionOptions options;
  options.recv_video = true;
  rtc::scoped_ptr<JsepSessionDescription> offer(
@ -3103,7 +3116,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithNewUfragAndPassword) {
 // This test verifies that an answer contains old ufrag and password if an offer
 // with old ufrag and password is received.
 TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
-  Init(NULL);
+  Init();
  cricket::MediaSessionOptions options;
  options.recv_video = true;
  rtc::scoped_ptr<JsepSessionDescription> offer(
@ -3132,7 +3145,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) {
 }

 TEST_F(WebRtcSessionTest, TestSessionContentError) {
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();
  const std::string session_id_orig = offer->session_id();
@ -3184,7 +3197,7 @@ TEST_F(WebRtcSessionTest, TestIceStatesBundle) {
 }

 TEST_F(WebRtcSessionTest, SetSdpFailedOnSessionError) {
-  Init(NULL);
+  Init();
  cricket::MediaSessionOptions options;
  options.recv_video = true;

@ -3209,7 +3222,7 @@ TEST_F(WebRtcSessionTest, TestRtpDataChannel) {
  constraints_.reset(new FakeConstraints());
  constraints_->AddOptional(
      webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true);
-  Init(NULL);
+  Init();

  SetLocalDescriptionWithDataChannel();
  EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type());
@ -3439,7 +3452,7 @@ TEST_F(WebRtcSessionTest,
 // offer has no SDES crypto but only DTLS fingerprint.
 TEST_F(WebRtcSessionTest, TestSetRemoteOfferFailIfDtlsDisabledAndNoCrypto) {
  // Init without DTLS.
-  Init(NULL);
+  Init();
  // Create a remote offer with secured transport disabled.
  cricket::MediaSessionOptions options;
  JsepSessionDescription* offer(CreateRemoteOffer(
@ -3461,7 +3474,7 @@ TEST_F(WebRtcSessionTest, TestDscpConstraint) {
  constraints_.reset(new FakeConstraints());
  constraints_->AddOptional(
      webrtc::MediaConstraintsInterface::kEnableDscp, true);
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();

@ -3487,7 +3500,7 @@ TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) {
  constraints_->AddOptional(
      webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
      true);
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();

@ -3516,7 +3529,7 @@ TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) {
  constraints_->AddOptional(
      webrtc::MediaConstraintsInterface::kCombinedAudioVideoBwe,
      true);
-  Init(NULL);
+  Init();
  mediastream_signaling_.SendAudioVideoStream1();
  SessionDescriptionInterface* offer = CreateOffer();

--- a/webrtc/p2p/base/dtlstransportchannel.h
+++ b/webrtc/p2p/base/dtlstransportchannel.h
@ -126,6 +126,9 @@ class DtlsTransportChannelWrapper : public TransportChannelImpl {
  virtual int SetOption(rtc::Socket::Option opt, int value) {
    return channel_->SetOption(opt, value);
  }
+  virtual bool GetOption(rtc::Socket::Option opt, int* value) {
+    return channel_->GetOption(opt, value);
+  }
  virtual int GetError() {
    return channel_->GetError();
  }
--- a/webrtc/p2p/base/fakesession.h
+++ b/webrtc/p2p/base/fakesession.h
@ -198,6 +198,9 @@ class FakeTransportChannel : public TransportChannelImpl,
  virtual int SetOption(rtc::Socket::Option opt, int value) {
    return true;
  }
+  virtual bool GetOption(rtc::Socket::Option opt, int* value) {
+    return true;
+  }
  virtual int GetError() {
    return 0;
  }
--- a/webrtc/p2p/base/p2ptransportchannel.cc
+++ b/webrtc/p2p/base/p2ptransportchannel.cc
@ -817,6 +817,7 @@ void P2PTransportChannel::RememberRemoteCandidate(
 // Set options on ourselves is simply setting options on all of our available
 // port objects.
 int P2PTransportChannel::SetOption(rtc::Socket::Option opt, int value) {
+  ASSERT(worker_thread_ == rtc::Thread::Current());
  OptionMap::iterator it = options_.find(opt);
  if (it == options_.end()) {
    options_.insert(std::make_pair(opt, value));
@ -838,6 +839,17 @@ int P2PTransportChannel::SetOption(rtc::Socket::Option opt, int value) {
  return 0;
 }

+bool P2PTransportChannel::GetOption(rtc::Socket::Option opt, int* value) {
+  ASSERT(worker_thread_ == rtc::Thread::Current());
+
+  const auto& found = options_.find(opt);
+  if (found == options_.end()) {
+    return false;
+  }
+  *value = found->second;
+  return true;
+}
+
 // Send data to the other side, using our best connection.
 int P2PTransportChannel::SendPacket(const char *data, size_t len,
                                    const rtc::PacketOptions& options,
--- a/webrtc/p2p/base/p2ptransportchannel.h
+++ b/webrtc/p2p/base/p2ptransportchannel.h
@ -79,6 +79,7 @@ class P2PTransportChannel : public TransportChannelImpl,
  virtual int SendPacket(const char *data, size_t len,
                         const rtc::PacketOptions& options, int flags);
  virtual int SetOption(rtc::Socket::Option opt, int value);
+  virtual bool GetOption(rtc::Socket::Option opt, int* value);
  virtual int GetError() { return error_; }
  virtual bool GetStats(std::vector<ConnectionInfo>* stats);

--- a/webrtc/p2p/base/rawtransportchannel.cc
+++ b/webrtc/p2p/base/rawtransportchannel.cc
@ -72,6 +72,10 @@ int RawTransportChannel::SetOption(rtc::Socket::Option opt, int value) {
  return port_->SetOption(opt, value);
 }

+bool RawTransportChannel::GetOption(rtc::Socket::Option opt, int* value) {
+  return false;
+}
+
 int RawTransportChannel::GetError() {
  return (port_ != NULL) ? port_->GetError() : 0;
 }
--- a/webrtc/p2p/base/rawtransportchannel.h
+++ b/webrtc/p2p/base/rawtransportchannel.h
@ -50,6 +50,7 @@ class RawTransportChannel : public TransportChannelImpl,
  virtual int SendPacket(const char *data, size_t len,
                         const rtc::PacketOptions& options, int flags);
  virtual int SetOption(rtc::Socket::Option opt, int value);
+  virtual bool GetOption(rtc::Socket::Option opt, int* value);
  virtual int GetError();

  // Implements TransportChannelImpl.
--- a/webrtc/p2p/base/session.cc
+++ b/webrtc/p2p/base/session.cc
@ -787,8 +787,11 @@ void BaseSession::OnTransportCandidatesAllocationDone(Transport* transport) {
 bool BaseSession::IsCandidateAllocationDone() const {
  for (TransportMap::const_iterator iter = transports_.begin();
       iter != transports_.end(); ++iter) {
-    if (!iter->second->candidates_allocated())
+    if (!iter->second->candidates_allocated()) {
+      LOG(LS_INFO) << "Candidate allocation not done for "
+                   << iter->second->content_name();
      return false;
+    }
  }
  return true;
 }
--- a/webrtc/p2p/base/session.h
+++ b/webrtc/p2p/base/session.h
@ -415,6 +415,8 @@ class BaseSession : public sigslot::has_slots<>,
  virtual void OnMessage(rtc::Message *pmsg);

 protected:
+  bool IsCandidateAllocationDone() const;
+
  State state_;
  Error error_;
  std::string error_desc_;
@ -428,7 +430,6 @@ class BaseSession : public sigslot::has_slots<>,
      const SessionDescription* sdesc, ContentAction action,
      std::string* error_desc);

-  bool IsCandidateAllocationDone() const;
  void MaybeCandidateAllocationDone();

  // This method will delete the Transport and TransportChannelImpls and
--- a/webrtc/p2p/base/transportchannel.h
+++ b/webrtc/p2p/base/transportchannel.h
@ -81,6 +81,9 @@ class TransportChannel : public sigslot::has_slots<> {
  // Sets a socket option on this channel.  Note that not all options are
  // supported by all transport types.
  virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
+  // TODO(pthatcher): Once Chrome's MockTransportChannel implments
+  // this, remove the default implementation.
+  virtual bool GetOption(rtc::Socket::Option opt, int* value) { return false; }

  // Returns the most recent error that occurred on this channel.
  virtual int GetError() = 0;
--- a/webrtc/p2p/base/transportchannelproxy.cc
+++ b/webrtc/p2p/base/transportchannelproxy.cc
@ -104,6 +104,21 @@ int TransportChannelProxy::SetOption(rtc::Socket::Option opt, int value) {
  return impl_->SetOption(opt, value);
 }

+bool TransportChannelProxy::GetOption(rtc::Socket::Option opt, int* value) {
+  ASSERT(rtc::Thread::Current() == worker_thread_);
+  if (impl_) {
+    return impl_->GetOption(opt, value);
+  }
+
+  for (const auto& pending : pending_options_) {
+    if (pending.first == opt) {
+      *value = pending.second;
+      return true;
+    }
+  }
+  return false;
+}
+
 int TransportChannelProxy::GetError() {
  ASSERT(rtc::Thread::Current() == worker_thread_);
  if (!impl_) {
--- a/webrtc/p2p/base/transportchannelproxy.h
+++ b/webrtc/p2p/base/transportchannelproxy.h
@ -52,6 +52,7 @@ class TransportChannelProxy : public TransportChannel,
                         const rtc::PacketOptions& options,
                         int flags);
  virtual int SetOption(rtc::Socket::Option opt, int value);
+  virtual bool GetOption(rtc::Socket::Option opt, int* value);
  virtual int GetError();
  virtual IceRole GetIceRole() const;
  virtual bool GetStats(ConnectionInfos* infos);