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
Cr-Commit-Position: refs/heads/master@{#8237}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8237 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index bc16e5e..2dce562 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -323,28 +323,16 @@
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 @@
// 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.
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index 35ba705..1b52a56 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -506,13 +506,13 @@
// 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);
}
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 27b936b..7144e76 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -466,11 +466,12 @@
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 @@
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 @@
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
}
port_allocator()->set_candidate_filter(
- ConvertIceTransportTypeToCandidateFilter(ice_transport));
+ ConvertIceTransportTypeToCandidateFilter(ice_transport_type));
return true;
}
diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
index 8a77923..4f37340 100644
--- a/talk/app/webrtc/webrtcsession.h
+++ b/talk/app/webrtc/webrtcsession.h
@@ -118,7 +118,8 @@
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 @@
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 d0fb805..a8ca497 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -342,8 +342,7 @@
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 @@
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 @@
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 @@
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 @@
webrtc::MediaConstraintsInterface::kNumUnsignalledRecvStreams,
value_set);
session_.reset();
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
@@ -891,7 +908,7 @@
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 @@
if (can) {
InitWithDtmfCodec();
} else {
- Init(NULL);
+ Init();
}
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
@@ -1050,7 +1067,7 @@
void TestLoopbackCall(const LoopbackNetworkConfiguration& config) {
LoopbackNetworkManager loopback_network_manager(this, config);
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
@@ -1240,7 +1257,6 @@
MockIceObserver observer_;
cricket::FakeVideoMediaChannel* video_channel_;
cricket::FakeVoiceMediaChannel* voice_channel_;
- PeerConnectionInterface::IceTransportsType ice_type_;
FakeMetricsObserver metrics_observer_;
};
@@ -1251,7 +1267,7 @@
}
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, 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 @@
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 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, 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 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, 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 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 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, SetLocalSdpFailedOnCreateChannel) {
- Init(NULL);
+ Init();
media_engine_->set_fail_create_channel(true);
SessionDescriptionInterface* offer = CreateOffer();
@@ -1512,7 +1525,7 @@
// 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 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 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, TestSetLocalOfferTwice) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendNothing();
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer = CreateOffer();
@@ -1740,7 +1753,7 @@
}
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, TestSetLocalAndRemoteOffer) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
@@ -1761,7 +1774,7 @@
}
TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
@@ -1771,7 +1784,7 @@
}
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, TestSetRemotePrAnswer) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE);
@@ -1821,7 +1834,7 @@
}
TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendNothing();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
@@ -1832,7 +1845,7 @@
}
TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendNothing();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
@@ -1843,7 +1856,7 @@
}
TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
cricket::Candidate candidate;
@@ -1892,7 +1905,7 @@
// 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 @@
// 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 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 @@
// 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 @@
// 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 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 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 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 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 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 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 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 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 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 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 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, CreateOfferWithoutCNCodecs) {
AddCNCodecs();
- Init(NULL);
+ Init();
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio =
RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
@@ -2336,7 +2349,7 @@
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 @@
// 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 @@
// 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, VerifyCryptoParamsInSDP) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
VerifyCryptoParams(offer->description());
@@ -2475,26 +2488,26 @@
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 @@
// 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 @@
// 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 @@
// 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 @@
// 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, TestDisabledBundleInAnswer) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE &
allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE);
@@ -2628,7 +2641,7 @@
// 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, SetAudioPlayout) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@@ -2683,7 +2696,7 @@
}
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, AudioRendererForLocalStream) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@@ -2736,7 +2749,7 @@
}
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, SetVideoSend) {
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
@@ -2778,7 +2791,7 @@
TEST_F(WebRtcSessionTest, InsertDtmf) {
// Setup
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@@ -2804,7 +2817,7 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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 @@
// 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, 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, SetSdpFailedOnSessionError) {
- Init(NULL);
+ Init();
cricket::MediaSessionOptions options;
options.recv_video = true;
@@ -3209,7 +3222,7 @@
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 @@
// 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 @@
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableDscp, true);
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
@@ -3487,7 +3500,7 @@
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
true);
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
@@ -3516,7 +3529,7 @@
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kCombinedAudioVideoBwe,
true);
- Init(NULL);
+ Init();
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
diff --git a/webrtc/p2p/base/dtlstransportchannel.h b/webrtc/p2p/base/dtlstransportchannel.h
index e629bb5..4c9c879 100644
--- a/webrtc/p2p/base/dtlstransportchannel.h
+++ b/webrtc/p2p/base/dtlstransportchannel.h
@@ -126,6 +126,9 @@
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 375d36d..13eceef 100644
--- a/webrtc/p2p/base/fakesession.h
+++ b/webrtc/p2p/base/fakesession.h
@@ -198,6 +198,9 @@
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 4bbe9cf..735dbd5 100644
--- a/webrtc/p2p/base/p2ptransportchannel.cc
+++ b/webrtc/p2p/base/p2ptransportchannel.cc
@@ -817,6 +817,7 @@
// 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 @@
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 10e19f0..f8756dc 100644
--- a/webrtc/p2p/base/p2ptransportchannel.h
+++ b/webrtc/p2p/base/p2ptransportchannel.h
@@ -79,6 +79,7 @@
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);
diff --git a/webrtc/p2p/base/rawtransportchannel.cc b/webrtc/p2p/base/rawtransportchannel.cc
index f0d7d5d..b032e63 100644
--- a/webrtc/p2p/base/rawtransportchannel.cc
+++ b/webrtc/p2p/base/rawtransportchannel.cc
@@ -72,6 +72,10 @@
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 bc84316..a4d9ce0 100644
--- a/webrtc/p2p/base/rawtransportchannel.h
+++ b/webrtc/p2p/base/rawtransportchannel.h
@@ -50,6 +50,7 @@
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 ba10a14..93a4a2b 100644
--- a/webrtc/p2p/base/session.cc
+++ b/webrtc/p2p/base/session.cc
@@ -787,8 +787,11 @@
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;
}
diff --git a/webrtc/p2p/base/session.h b/webrtc/p2p/base/session.h
index d91fb57..7f2d816 100644
--- a/webrtc/p2p/base/session.h
+++ b/webrtc/p2p/base/session.h
@@ -415,6 +415,8 @@
virtual void OnMessage(rtc::Message *pmsg);
protected:
+ bool IsCandidateAllocationDone() const;
+
State state_;
Error error_;
std::string error_desc_;
@@ -428,7 +430,6 @@
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 d50f025..fb592e5 100644
--- a/webrtc/p2p/base/transportchannel.h
+++ b/webrtc/p2p/base/transportchannel.h
@@ -81,6 +81,7 @@
// 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 e6fb557..a8535fa 100644
--- a/webrtc/p2p/base/transportchannelproxy.cc
+++ b/webrtc/p2p/base/transportchannelproxy.cc
@@ -104,6 +104,21 @@
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 188039e..46803f7 100644
--- a/webrtc/p2p/base/transportchannelproxy.h
+++ b/webrtc/p2p/base/transportchannelproxy.h
@@ -52,6 +52,7 @@
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);
