From c5f697135e626044b15eacdc82fd840fbe74b351 Mon Sep 17 00:00:00 2001 From: "bjornv@webrtc.org" Date: Wed, 4 Feb 2015 10:22:14 +0000 Subject: [PATCH] Revert 8237 "Cleanup and prepare for bundling." libjingle_peerconnection_objc_test consistently failing on Mac64 Debug. > 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. > > BUG=1574 > R=pthatcher@webrtc.org, tommi@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/39699004 TBR=pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34959004 Cr-Commit-Position: refs/heads/master@{#8241} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8241 4adac7df-926f-26a2-2b94-8c16560cd09d --- talk/app/webrtc/peerconnection.cc | 26 ++- talk/app/webrtc/peerconnectioninterface.h | 4 +- talk/app/webrtc/webrtcsession.cc | 18 +- talk/app/webrtc/webrtcsession.h | 6 +- talk/app/webrtc/webrtcsession_unittest.cc | 219 ++++++++++------------ webrtc/p2p/base/dtlstransportchannel.h | 3 - webrtc/p2p/base/fakesession.h | 3 - webrtc/p2p/base/p2ptransportchannel.cc | 12 -- webrtc/p2p/base/p2ptransportchannel.h | 1 - webrtc/p2p/base/rawtransportchannel.cc | 4 - webrtc/p2p/base/rawtransportchannel.h | 1 - webrtc/p2p/base/session.cc | 5 +- webrtc/p2p/base/session.h | 3 +- webrtc/p2p/base/transportchannel.h | 1 - webrtc/p2p/base/transportchannelproxy.cc | 15 -- webrtc/p2p/base/transportchannelproxy.h | 1 - 16 files changed, 133 insertions(+), 189 deletions(-) diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc index 2dce562ae8..bc16e5e50c 100644 --- a/talk/app/webrtc/peerconnection.cc +++ b/talk/app/webrtc/peerconnection.cc @@ -323,16 +323,28 @@ bool PeerConnection::Initialize( PortAllocatorFactoryInterface* allocator_factory, DTLSIdentityServiceInterface* dtls_identity_service, PeerConnectionObserver* observer) { - ASSERT(observer != NULL); - if (!observer) - return false; - observer_ = observer; - std::vector stun_config; std::vector 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)); @@ -372,9 +384,7 @@ bool PeerConnection::Initialize( // Initialize the WebRtcSession. It creates transport channels etc. if (!session_->Initialize(factory_->options(), constraints, - dtls_identity_service, - configuration.type, - configuration.bundle_policy)) + dtls_identity_service, type)) return false; // Register PeerConnection as receiver of local ice candidates. diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h index 1b52a5615e..35ba705b88 100644 --- 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 CreatePeerConnection( - const PeerConnectionInterface::IceServers& servers, + const PeerConnectionInterface::IceServers& configuration, const MediaConstraintsInterface* constraints, PortAllocatorFactoryInterface* allocator_factory, DTLSIdentityServiceInterface* dtls_identity_service, PeerConnectionObserver* observer) { PeerConnectionInterface::RTCConfiguration rtc_config; - rtc_config.servers = servers; + rtc_config.servers = configuration; return CreatePeerConnection(rtc_config, constraints, allocator_factory, dtls_identity_service, observer); } diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc index 83e613ee5e..0145243fe1 100644 --- a/talk/app/webrtc/webrtcsession.cc +++ b/talk/app/webrtc/webrtcsession.cc @@ -466,12 +466,11 @@ 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, @@ -517,10 +516,7 @@ bool WebRtcSession::Initialize( const PeerConnectionFactoryInterface::Options& options, const MediaConstraintsInterface* constraints, DTLSIdentityServiceInterface* dtls_identity_service, - PeerConnectionInterface::IceTransportsType ice_transport_type, - PeerConnectionInterface::BundlePolicy bundle_policy) { - bundle_policy_ = bundle_policy; - + PeerConnectionInterface::IceTransportsType ice_transport) { // TODO(perkj): Take |constraints| into consideration. Return false if not all // mandatory constraints can be fulfilled. Note that |constraints| // can be null. @@ -663,7 +659,7 @@ bool WebRtcSession::Initialize( webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED); } port_allocator()->set_candidate_filter( - ConvertIceTransportTypeToCandidateFilter(ice_transport_type)); + ConvertIceTransportTypeToCandidateFilter(ice_transport)); return true; } diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h index 4f37340b9c..8a77923aaf 100644 --- a/talk/app/webrtc/webrtcsession.h +++ b/talk/app/webrtc/webrtcsession.h @@ -118,8 +118,7 @@ class WebRtcSession : public cricket::BaseSession, bool Initialize(const PeerConnectionFactoryInterface::Options& options, const MediaConstraintsInterface* constraints, DTLSIdentityServiceInterface* dtls_identity_service, - PeerConnectionInterface::IceTransportsType ice_transport_type, - PeerConnectionInterface::BundlePolicy bundle_policy); + PeerConnectionInterface::IceTransportsType ice_transport); // Deletes the voice, video and data channel and changes the session state // to STATE_RECEIVEDTERMINATE. void Terminate(); @@ -373,9 +372,6 @@ 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 diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc index a8ca497ac6..d0fb805a50 100644 --- a/talk/app/webrtc/webrtcsession_unittest.cc +++ b/talk/app/webrtc/webrtcsession_unittest.cc @@ -342,7 +342,8 @@ 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()) { + mediastream_signaling_(channel_manager_.get()), + ice_type_(PeerConnectionInterface::kAll) { tdesc_factory_->set_protocol(cricket::ICEPROTO_HYBRID); cricket::ServerAddresses stun_servers; @@ -364,10 +365,11 @@ class WebRtcSessionTest : public testing::Test { network_manager_.AddInterface(addr); } - void Init( - DTLSIdentityServiceInterface* identity_service, - PeerConnectionInterface::IceTransportsType ice_transport_type, - PeerConnectionInterface::BundlePolicy bundle_policy) { + void SetIceTransportType(PeerConnectionInterface::IceTransportsType type) { + ice_type_ = type; + } + + void Init(DTLSIdentityServiceInterface* identity_service) { ASSERT_TRUE(session_.get() == NULL); session_.reset(new WebRtcSessionForTest( channel_manager_.get(), rtc::Thread::Current(), @@ -381,35 +383,10 @@ class WebRtcSessionTest : public testing::Test { observer_.ice_gathering_state_); EXPECT_TRUE(session_->Initialize(options_, constraints_.get(), - identity_service, ice_transport_type, - bundle_policy)); + identity_service, ice_type_)); 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( @@ -418,7 +395,13 @@ class WebRtcSessionTest : public testing::Test { codecs.push_back(kTelephoneEventCodec); media_engine_->SetAudioCodecs(codecs); desc_factory_->set_audio_codecs(codecs); - Init(); + 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); } // Creates a local offer and applies it. Starts ice. @@ -589,7 +572,7 @@ class WebRtcSessionTest : public testing::Test { webrtc::MediaConstraintsInterface::kNumUnsignalledRecvStreams, value_set); session_.reset(); - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); @@ -908,7 +891,7 @@ class WebRtcSessionTest : public testing::Test { void TestSessionCandidatesWithBundleRtcpMux(bool bundle, bool rtcp_mux) { AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort)); - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); PeerConnectionInterface::RTCOfferAnswerOptions options; @@ -960,7 +943,7 @@ class WebRtcSessionTest : public testing::Test { if (can) { InitWithDtmfCodec(); } else { - Init(); + Init(NULL); } mediastream_signaling_.SendAudioVideoStream1(); CreateAndSetRemoteOfferAndLocalAnswer(); @@ -1067,7 +1050,7 @@ class WebRtcSessionTest : public testing::Test { void TestLoopbackCall(const LoopbackNetworkConfiguration& config) { LoopbackNetworkManager loopback_network_manager(this, config); - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); @@ -1257,6 +1240,7 @@ class WebRtcSessionTest : public testing::Test { MockIceObserver observer_; cricket::FakeVideoMediaChannel* video_channel_; cricket::FakeVoiceMediaChannel* voice_channel_; + PeerConnectionInterface::IceTransportsType ice_type_; FakeMetricsObserver metrics_observer_; }; @@ -1267,7 +1251,7 @@ TEST_F(WebRtcSessionTest, TestInitializeWithDtls) { } TEST_F(WebRtcSessionTest, TestInitializeWithoutDtls) { - Init(); + Init(NULL); // SDES is required if DTLS is off. EXPECT_EQ(cricket::SEC_REQUIRED, session_->SdesPolicy()); } @@ -1289,7 +1273,7 @@ TEST_F(WebRtcSessionTest, TestSessionCandidatesWithBundleRtcpMux) { TEST_F(WebRtcSessionTest, TestMultihomeCandidates) { AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort)); AddInterface(rtc::SocketAddress(kClientAddrHost2, kClientAddrPort)); - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); InitiateCall(); EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout); @@ -1304,7 +1288,7 @@ TEST_F(WebRtcSessionTest, TestStunError) { rtc::FP_UDP, rtc::FD_ANY, rtc::SocketAddress(kClientAddrHost1, kClientAddrPort)); - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); InitiateCall(); // Since kClientAddrHost1 is blocked, not expecting stun candidates for it. @@ -1316,7 +1300,8 @@ TEST_F(WebRtcSessionTest, TestStunError) { // Test session delivers no candidates gathered when constraint set to "none". TEST_F(WebRtcSessionTest, TestIceTransportsNone) { AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort)); - InitWithIceTransport(PeerConnectionInterface::kNone); + SetIceTransportType(PeerConnectionInterface::kNone); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); InitiateCall(); EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout); @@ -1329,7 +1314,8 @@ TEST_F(WebRtcSessionTest, TestIceTransportsNone) { TEST_F(WebRtcSessionTest, TestIceTransportsRelay) { AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort)); ConfigureAllocatorWithTurn(); - InitWithIceTransport(PeerConnectionInterface::kRelay); + SetIceTransportType(PeerConnectionInterface::kRelay); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); InitiateCall(); EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout); @@ -1348,7 +1334,8 @@ TEST_F(WebRtcSessionTest, TestIceTransportsRelay) { // Test session delivers all candidates gathered when constaint set to "all". TEST_F(WebRtcSessionTest, TestIceTransportsAll) { AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort)); - InitWithIceTransport(PeerConnectionInterface::kAll); + SetIceTransportType(PeerConnectionInterface::kAll); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); InitiateCall(); EXPECT_TRUE_WAIT(observer_.oncandidatesready_, kIceCandidatesTimeout); @@ -1358,7 +1345,7 @@ TEST_F(WebRtcSessionTest, TestIceTransportsAll) { } TEST_F(WebRtcSessionTest, SetSdpFailedOnInvalidSdp) { - Init(); + Init(NULL); SessionDescriptionInterface* offer = NULL; // Since |offer| is NULL, there's no way to tell if it's an offer or answer. std::string unknown_action; @@ -1369,7 +1356,7 @@ 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); const std::string session_id_orig = offer->session_id(); @@ -1423,7 +1410,7 @@ 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream2(); SessionDescriptionInterface* offer = CreateOffer(); VerifyCryptoParams(offer->description()); @@ -1479,7 +1466,7 @@ TEST_F(WebRtcSessionTest, TestReceiveSdesOfferCreateSdesAnswer) { } TEST_F(WebRtcSessionTest, SetLocalSdpFailedOnCreateChannel) { - Init(); + Init(NULL); media_engine_->set_fail_create_channel(true); SessionDescriptionInterface* offer = CreateOffer(); @@ -1525,7 +1512,7 @@ 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(); + Init(NULL); cricket::MediaSessionOptions options; options.recv_video = true; JsepSessionDescription* offer = CreateRemoteOffer( @@ -1543,7 +1530,7 @@ 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(); + Init(NULL); SessionDescriptionInterface* offer = NULL; SessionDescriptionInterface* answer = NULL; CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer); @@ -1556,7 +1543,7 @@ 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(); + Init(NULL); SessionDescriptionInterface* offer = NULL; SessionDescriptionInterface* answer = NULL; CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer); @@ -1741,7 +1728,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerReceiveOfferWithoutEncryption) { } TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) { - Init(); + Init(NULL); mediastream_signaling_.SendNothing(); // SetLocalDescription take ownership of offer. SessionDescriptionInterface* offer = CreateOffer(); @@ -1753,7 +1740,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalOfferTwice) { } TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) { - Init(); + Init(NULL); mediastream_signaling_.SendNothing(); // SetLocalDescription take ownership of offer. SessionDescriptionInterface* offer = CreateOffer(); @@ -1764,7 +1751,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteOfferTwice) { } TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) { - Init(); + Init(NULL); mediastream_signaling_.SendNothing(); SessionDescriptionInterface* offer = CreateOffer(); SetLocalDescriptionWithoutError(offer); @@ -1774,7 +1761,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) { } TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) { - Init(); + Init(NULL); mediastream_signaling_.SendNothing(); SessionDescriptionInterface* offer = CreateOffer(); SetRemoteDescriptionWithoutError(offer); @@ -1784,7 +1771,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) { } TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) { - Init(); + Init(NULL); mediastream_signaling_.SendNothing(); SessionDescriptionInterface* offer = CreateRemoteOffer(); SetRemoteDescriptionExpectState(offer, BaseSession::STATE_RECEIVEDINITIATE); @@ -1807,7 +1794,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalPrAnswer) { } TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) { - Init(); + Init(NULL); mediastream_signaling_.SendNothing(); SessionDescriptionInterface* offer = CreateOffer(); SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE); @@ -1834,7 +1821,7 @@ TEST_F(WebRtcSessionTest, TestSetRemotePrAnswer) { } TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) { - Init(); + Init(NULL); mediastream_signaling_.SendNothing(); rtc::scoped_ptr offer(CreateOffer()); @@ -1845,7 +1832,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) { } TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) { - Init(); + Init(NULL); mediastream_signaling_.SendNothing(); rtc::scoped_ptr offer(CreateOffer()); @@ -1856,7 +1843,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) { } TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); cricket::Candidate candidate; @@ -1905,7 +1892,7 @@ 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(); + Init(NULL); cricket::Candidate candidate1; candidate1.set_component(1); JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0, @@ -1958,7 +1945,7 @@ 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); CreateAndSetRemoteOfferAndLocalAnswer(); @@ -1993,7 +1980,7 @@ TEST_F(WebRtcSessionTest, TestLocalCandidatesAddedToSessionDescription) { // Test that we can set a remote session description with remote candidates. TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) { - Init(); + Init(NULL); cricket::Candidate candidate1; candidate1.set_component(1); @@ -2022,7 +2009,7 @@ TEST_F(WebRtcSessionTest, TestSetRemoteSessionDescriptionWithCandidates) { // been gathered. TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) { AddInterface(rtc::SocketAddress(kClientAddrHost1, kClientAddrPort)); - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); // Ice is started but candidates are not provided until SetLocalDescription // is called. @@ -2055,7 +2042,7 @@ TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteDescriptionWithCandidates) { // Verifies TransportProxy and media channels are created with content names // present in the SessionDescription. TEST_F(WebRtcSessionTest, TestChannelCreationsWithContentNames) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); rtc::scoped_ptr offer(CreateOffer()); @@ -2099,7 +2086,7 @@ 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(); + Init(NULL); rtc::scoped_ptr offer(CreateOffer()); ASSERT_TRUE(offer != NULL); @@ -2113,7 +2100,7 @@ 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(); + Init(NULL); // Test Audio only offer. mediastream_signaling_.UseOptionsAudioOnly(); rtc::scoped_ptr offer(CreateOffer()); @@ -2136,7 +2123,7 @@ 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(); + Init(NULL); PeerConnectionInterface::RTCOfferAnswerOptions options; options.offer_to_receive_audio = 0; options.offer_to_receive_video = 0; @@ -2155,7 +2142,7 @@ 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(); + Init(NULL); PeerConnectionInterface::RTCOfferAnswerOptions options; options.offer_to_receive_audio = RTCOfferAnswerOptions::kOfferToReceiveMediaTrue; @@ -2173,7 +2160,7 @@ 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(); + Init(NULL); // Test Audio / Video offer. PeerConnectionInterface::RTCOfferAnswerOptions options; options.offer_to_receive_audio = @@ -2206,7 +2193,7 @@ 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(); + Init(NULL); SessionDescriptionInterface* offer = CreateOffer(); SetLocalDescriptionWithoutError(offer); SessionDescriptionInterface* answer = CreateRemoteAnswer(offer); @@ -2217,7 +2204,7 @@ 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(); + Init(NULL); // Create a remote offer with audio and video content. rtc::scoped_ptr offer(CreateRemoteOffer()); SetRemoteDescriptionWithoutError(offer.release()); @@ -2236,7 +2223,7 @@ 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(); + Init(NULL); // Create a remote offer with audio only. cricket::MediaSessionOptions options; @@ -2259,7 +2246,7 @@ 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(); + Init(NULL); // Create a remote offer with audio and video content. rtc::scoped_ptr offer(CreateRemoteOffer()); SetRemoteDescriptionWithoutError(offer.release()); @@ -2280,7 +2267,7 @@ 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(); + Init(NULL); // Create a remote offer with audio and video content. rtc::scoped_ptr offer(CreateRemoteOffer()); SetRemoteDescriptionWithoutError(offer.release()); @@ -2304,7 +2291,7 @@ 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(); + Init(NULL); // Create a remote offer with audio and video content. rtc::scoped_ptr offer(CreateRemoteOffer()); SetRemoteDescriptionWithoutError(offer.release()); @@ -2332,7 +2319,7 @@ TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) { TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) { AddCNCodecs(); - Init(); + Init(NULL); PeerConnectionInterface::RTCOfferAnswerOptions options; options.offer_to_receive_audio = RTCOfferAnswerOptions::kOfferToReceiveMediaTrue; @@ -2349,7 +2336,7 @@ TEST_F(WebRtcSessionTest, CreateOfferWithoutCNCodecs) { TEST_F(WebRtcSessionTest, CreateAnswerWithoutCNCodecs) { AddCNCodecs(); - Init(); + Init(NULL); // Create a remote offer with audio and video content. rtc::scoped_ptr offer(CreateRemoteOffer()); SetRemoteDescriptionWithoutError(offer.release()); @@ -2367,7 +2354,7 @@ 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(); + Init(NULL); EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL); EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL); @@ -2424,7 +2411,7 @@ 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(); + Init(NULL); EXPECT_TRUE(media_engine_->GetVideoChannel(0) == NULL); EXPECT_TRUE(media_engine_->GetVoiceChannel(0) == NULL); mediastream_signaling_.SendAudioVideoStream1(); @@ -2477,7 +2464,7 @@ TEST_F(WebRtcSessionTest, TestAVOfferWithVideoOnlyAnswer) { } TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); scoped_ptr offer(CreateOffer()); VerifyCryptoParams(offer->description()); @@ -2488,26 +2475,26 @@ TEST_F(WebRtcSessionTest, VerifyCryptoParamsInSDP) { TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) { options_.disable_encryption = true; - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); scoped_ptr offer(CreateOffer()); VerifyNoCryptoParams(offer->description(), false); } TEST_F(WebRtcSessionTest, VerifyAnswerFromNonCryptoOffer) { - Init(); + Init(NULL); VerifyAnswerFromNonCryptoOffer(); } TEST_F(WebRtcSessionTest, VerifyAnswerFromCryptoOffer) { - Init(); + Init(NULL); 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); rtc::scoped_ptr offer(CreateOffer()); @@ -2521,7 +2508,7 @@ 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(); + Init(NULL); rtc::scoped_ptr offer(CreateRemoteOffer()); std::string sdp; RemoveIceUfragPwdLines(offer.get(), &sdp); @@ -2533,7 +2520,7 @@ 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(); + Init(NULL); tdesc_factory_->set_protocol(cricket::ICEPROTO_RFC5245); mediastream_signaling_.SendAudioVideoStream1(); rtc::scoped_ptr offer(CreateOffer()); @@ -2559,7 +2546,7 @@ 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(); + Init(NULL); tdesc_factory_->set_protocol(cricket::ICEPROTO_RFC5245); rtc::scoped_ptr offer(CreateRemoteOffer()); std::string sdp; @@ -2583,7 +2570,7 @@ 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(); + Init(NULL); EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE); rtc::scoped_ptr offer(CreateOffer()); @@ -2600,7 +2587,7 @@ TEST_F(WebRtcSessionTest, VerifyBundleFlagInPA) { } TEST_F(WebRtcSessionTest, TestDisabledBundleInAnswer) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE); @@ -2641,7 +2628,7 @@ 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) { - Init(); + WebRtcSessionTest::Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE & allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE); @@ -2671,7 +2658,7 @@ TEST_F(WebRtcSessionTest, TestDisabledRtcpMuxWithBundleEnabled) { } TEST_F(WebRtcSessionTest, SetAudioPlayout) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); CreateAndSetRemoteOfferAndLocalAnswer(); cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0); @@ -2696,7 +2683,7 @@ TEST_F(WebRtcSessionTest, SetAudioPlayout) { } TEST_F(WebRtcSessionTest, SetAudioSend) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); CreateAndSetRemoteOfferAndLocalAnswer(); cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0); @@ -2726,7 +2713,7 @@ TEST_F(WebRtcSessionTest, SetAudioSend) { } TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); CreateAndSetRemoteOfferAndLocalAnswer(); cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0); @@ -2749,7 +2736,7 @@ TEST_F(WebRtcSessionTest, AudioRendererForLocalStream) { } TEST_F(WebRtcSessionTest, SetVideoPlayout) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); CreateAndSetRemoteOfferAndLocalAnswer(); cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0); @@ -2766,7 +2753,7 @@ TEST_F(WebRtcSessionTest, SetVideoPlayout) { } TEST_F(WebRtcSessionTest, SetVideoSend) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); CreateAndSetRemoteOfferAndLocalAnswer(); cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0); @@ -2791,7 +2778,7 @@ TEST_F(WebRtcSessionTest, CanInsertDtmf) { TEST_F(WebRtcSessionTest, InsertDtmf) { // Setup - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); CreateAndSetRemoteOfferAndLocalAnswer(); FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0); @@ -2817,7 +2804,7 @@ TEST_F(WebRtcSessionTest, InsertDtmf) { // This test verifies the |initiator| flag when session initiates the call. TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) { - Init(); + Init(NULL); EXPECT_FALSE(session_->initiator()); SessionDescriptionInterface* offer = CreateOffer(); SessionDescriptionInterface* answer = CreateRemoteAnswer(offer); @@ -2829,7 +2816,7 @@ TEST_F(WebRtcSessionTest, TestInitiatorFlagAsOriginator) { // This test verifies the |initiator| flag when session receives the call. TEST_F(WebRtcSessionTest, TestInitiatorFlagAsReceiver) { - Init(); + Init(NULL); EXPECT_FALSE(session_->initiator()); SessionDescriptionInterface* offer = CreateRemoteOffer(); SetRemoteDescriptionWithoutError(offer); @@ -2843,7 +2830,7 @@ 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); rtc::scoped_ptr answer( @@ -2865,7 +2852,7 @@ 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); SessionDescriptionInterface* answer = CreateRemoteAnswer(offer); @@ -2879,7 +2866,7 @@ 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); SetRemoteDescriptionWithoutError(offer); @@ -2901,7 +2888,7 @@ 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); SetRemoteDescriptionWithoutError(offer); @@ -2914,7 +2901,7 @@ 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); rtc::scoped_ptr offer(CreateOffer()); @@ -2946,7 +2933,7 @@ TEST_F(WebRtcSessionTest, TestIceOfferGIceOnlyAnswer) { // Verifing local offer and remote answer have matching m-lines as per RFC 3264. TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); SetLocalDescriptionWithoutError(offer); @@ -2994,7 +2981,7 @@ TEST_F(WebRtcSessionTest, TestIncorrectMLinesInRemoteAnswer) { // Verifying remote offer and local answer have matching m-lines as per // RFC 3264. TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateRemoteOffer(); SetRemoteDescriptionWithoutError(offer); @@ -3015,7 +3002,7 @@ TEST_F(WebRtcSessionTest, TestIncorrectMLinesInLocalAnswer) { // This test verifies that WebRtcSession does not start candidate allocation // before SetLocalDescription is called. TEST_F(WebRtcSessionTest, TestIceStartAfterSetLocalDescriptionOnly) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateRemoteOffer(); cricket::Candidate candidate; @@ -3047,7 +3034,7 @@ 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(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); rtc::scoped_ptr offer(CreateOffer()); @@ -3066,7 +3053,7 @@ TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescription) { // This test verifies the crypto parameter when security is disabled. TEST_F(WebRtcSessionTest, TestCryptoAfterSetLocalDescriptionWithDisabled) { options_.disable_encryption = true; - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); rtc::scoped_ptr offer(CreateOffer()); @@ -3085,7 +3072,7 @@ 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(); + Init(NULL); cricket::MediaSessionOptions options; options.recv_video = true; rtc::scoped_ptr offer( @@ -3116,7 +3103,7 @@ 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(); + Init(NULL); cricket::MediaSessionOptions options; options.recv_video = true; rtc::scoped_ptr offer( @@ -3145,7 +3132,7 @@ TEST_F(WebRtcSessionTest, TestCreateAnswerWithOldUfragAndPassword) { } TEST_F(WebRtcSessionTest, TestSessionContentError) { - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); const std::string session_id_orig = offer->session_id(); @@ -3197,7 +3184,7 @@ TEST_F(WebRtcSessionTest, TestIceStatesBundle) { } TEST_F(WebRtcSessionTest, SetSdpFailedOnSessionError) { - Init(); + Init(NULL); cricket::MediaSessionOptions options; options.recv_video = true; @@ -3222,7 +3209,7 @@ TEST_F(WebRtcSessionTest, TestRtpDataChannel) { constraints_.reset(new FakeConstraints()); constraints_->AddOptional( webrtc::MediaConstraintsInterface::kEnableRtpDataChannels, true); - Init(); + Init(NULL); SetLocalDescriptionWithDataChannel(); EXPECT_EQ(cricket::DCT_RTP, data_engine_->last_channel_type()); @@ -3452,7 +3439,7 @@ TEST_F(WebRtcSessionTest, // offer has no SDES crypto but only DTLS fingerprint. TEST_F(WebRtcSessionTest, TestSetRemoteOfferFailIfDtlsDisabledAndNoCrypto) { // Init without DTLS. - Init(); + Init(NULL); // Create a remote offer with secured transport disabled. cricket::MediaSessionOptions options; JsepSessionDescription* offer(CreateRemoteOffer( @@ -3474,7 +3461,7 @@ TEST_F(WebRtcSessionTest, TestDscpConstraint) { constraints_.reset(new FakeConstraints()); constraints_->AddOptional( webrtc::MediaConstraintsInterface::kEnableDscp, true); - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); @@ -3500,7 +3487,7 @@ TEST_F(WebRtcSessionTest, TestSuspendBelowMinBitrateConstraint) { constraints_->AddOptional( webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate, true); - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); @@ -3529,7 +3516,7 @@ TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) { constraints_->AddOptional( webrtc::MediaConstraintsInterface::kCombinedAudioVideoBwe, true); - Init(); + Init(NULL); mediastream_signaling_.SendAudioVideoStream1(); SessionDescriptionInterface* offer = CreateOffer(); diff --git a/webrtc/p2p/base/dtlstransportchannel.h b/webrtc/p2p/base/dtlstransportchannel.h index 4c9c879b78..e629bb53c0 100644 --- a/webrtc/p2p/base/dtlstransportchannel.h +++ b/webrtc/p2p/base/dtlstransportchannel.h @@ -126,9 +126,6 @@ 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(); } diff --git a/webrtc/p2p/base/fakesession.h b/webrtc/p2p/base/fakesession.h index 13eceefe19..375d36d473 100644 --- a/webrtc/p2p/base/fakesession.h +++ b/webrtc/p2p/base/fakesession.h @@ -198,9 +198,6 @@ 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; } diff --git a/webrtc/p2p/base/p2ptransportchannel.cc b/webrtc/p2p/base/p2ptransportchannel.cc index 735dbd598f..4bbe9cf239 100644 --- a/webrtc/p2p/base/p2ptransportchannel.cc +++ b/webrtc/p2p/base/p2ptransportchannel.cc @@ -817,7 +817,6 @@ 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)); @@ -839,17 +838,6 @@ 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, diff --git a/webrtc/p2p/base/p2ptransportchannel.h b/webrtc/p2p/base/p2ptransportchannel.h index f8756dc02d..10e19f03f9 100644 --- a/webrtc/p2p/base/p2ptransportchannel.h +++ b/webrtc/p2p/base/p2ptransportchannel.h @@ -79,7 +79,6 @@ 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* stats); diff --git a/webrtc/p2p/base/rawtransportchannel.cc b/webrtc/p2p/base/rawtransportchannel.cc index b032e63cda..f0d7d5d9f6 100644 --- a/webrtc/p2p/base/rawtransportchannel.cc +++ b/webrtc/p2p/base/rawtransportchannel.cc @@ -72,10 +72,6 @@ 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; } diff --git a/webrtc/p2p/base/rawtransportchannel.h b/webrtc/p2p/base/rawtransportchannel.h index a4d9ce01d0..bc8431616b 100644 --- a/webrtc/p2p/base/rawtransportchannel.h +++ b/webrtc/p2p/base/rawtransportchannel.h @@ -50,7 +50,6 @@ 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. diff --git a/webrtc/p2p/base/session.cc b/webrtc/p2p/base/session.cc index 69b6a9cab5..1a126a6983 100644 --- a/webrtc/p2p/base/session.cc +++ b/webrtc/p2p/base/session.cc @@ -823,11 +823,8 @@ 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()) { - LOG(LS_INFO) << "Candidate allocation not done for " - << iter->second->content_name(); + if (!iter->second->candidates_allocated()) return false; - } } return true; } diff --git a/webrtc/p2p/base/session.h b/webrtc/p2p/base/session.h index 7e89123ef4..f809b3095d 100644 --- a/webrtc/p2p/base/session.h +++ b/webrtc/p2p/base/session.h @@ -427,8 +427,6 @@ class BaseSession : public sigslot::has_slots<>, virtual void OnMessage(rtc::Message *pmsg); protected: - bool IsCandidateAllocationDone() const; - State state_; Error error_; std::string error_desc_; @@ -442,6 +440,7 @@ 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 diff --git a/webrtc/p2p/base/transportchannel.h b/webrtc/p2p/base/transportchannel.h index fb592e5580..d50f025ea0 100644 --- a/webrtc/p2p/base/transportchannel.h +++ b/webrtc/p2p/base/transportchannel.h @@ -81,7 +81,6 @@ 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; - virtual bool GetOption(rtc::Socket::Option opt, int* value) = 0; // Returns the most recent error that occurred on this channel. virtual int GetError() = 0; diff --git a/webrtc/p2p/base/transportchannelproxy.cc b/webrtc/p2p/base/transportchannelproxy.cc index a8535fa461..e6fb557f77 100644 --- a/webrtc/p2p/base/transportchannelproxy.cc +++ b/webrtc/p2p/base/transportchannelproxy.cc @@ -104,21 +104,6 @@ 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_) { diff --git a/webrtc/p2p/base/transportchannelproxy.h b/webrtc/p2p/base/transportchannelproxy.h index 46803f7369..188039ef5a 100644 --- a/webrtc/p2p/base/transportchannelproxy.h +++ b/webrtc/p2p/base/transportchannelproxy.h @@ -52,7 +52,6 @@ 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);