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
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 2dce562..bc16e5e 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -323,16 +323,28 @@
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));
@@ -372,9 +384,7 @@
// 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 1b52a56..35ba705 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& 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 83e613e..0145243 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -466,12 +466,11 @@
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 @@
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 @@
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 4f37340..8a77923 100644
--- a/talk/app/webrtc/webrtcsession.h
+++ b/talk/app/webrtc/webrtcsession.h
@@ -118,8 +118,7 @@
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 @@
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 a8ca497..d0fb805 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -342,7 +342,8 @@
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 @@
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 @@
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 @@
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 @@
webrtc::MediaConstraintsInterface::kNumUnsignalledRecvStreams,
value_set);
session_.reset();
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
@@ -908,7 +891,7 @@
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 @@
if (can) {
InitWithDtmfCodec();
} else {
- Init();
+ Init(NULL);
}
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
@@ -1067,7 +1050,7 @@
void TestLoopbackCall(const LoopbackNetworkConfiguration& config) {
LoopbackNetworkManager loopback_network_manager(this, config);
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
@@ -1257,6 +1240,7 @@
MockIceObserver observer_;
cricket::FakeVideoMediaChannel* video_channel_;
cricket::FakeVoiceMediaChannel* voice_channel_;
+ PeerConnectionInterface::IceTransportsType ice_type_;
FakeMetricsObserver metrics_observer_;
};
@@ -1267,7 +1251,7 @@
}
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, 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 @@
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 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, 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 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, 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 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 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, SetLocalSdpFailedOnCreateChannel) {
- Init();
+ Init(NULL);
media_engine_->set_fail_create_channel(true);
SessionDescriptionInterface* offer = CreateOffer();
@@ -1525,7 +1512,7 @@
// 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();
+ Init(NULL);
cricket::MediaSessionOptions options;
options.recv_video = true;
JsepSessionDescription* offer = CreateRemoteOffer(
@@ -1543,7 +1530,7 @@
// 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();
+ Init(NULL);
SessionDescriptionInterface* offer = NULL;
SessionDescriptionInterface* answer = NULL;
CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
@@ -1556,7 +1543,7 @@
// 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();
+ Init(NULL);
SessionDescriptionInterface* offer = NULL;
SessionDescriptionInterface* answer = NULL;
CreateCryptoOfferAndNonCryptoAnswer(&offer, &answer);
@@ -1741,7 +1728,7 @@
}
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, TestSetRemoteOfferTwice) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendNothing();
// SetLocalDescription take ownership of offer.
SessionDescriptionInterface* offer = CreateOffer();
@@ -1764,7 +1751,7 @@
}
TEST_F(WebRtcSessionTest, TestSetLocalAndRemoteOffer) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionWithoutError(offer);
@@ -1774,7 +1761,7 @@
}
TEST_F(WebRtcSessionTest, TestSetRemoteAndLocalOffer) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
@@ -1784,7 +1771,7 @@
}
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, TestSetRemotePrAnswer) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendNothing();
SessionDescriptionInterface* offer = CreateOffer();
SetLocalDescriptionExpectState(offer, BaseSession::STATE_SENTINITIATE);
@@ -1834,7 +1821,7 @@
}
TEST_F(WebRtcSessionTest, TestSetLocalAnswerWithoutOffer) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendNothing();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
@@ -1845,7 +1832,7 @@
}
TEST_F(WebRtcSessionTest, TestSetRemoteAnswerWithoutOffer) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendNothing();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
@@ -1856,7 +1843,7 @@
}
TEST_F(WebRtcSessionTest, TestAddRemoteCandidate) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
cricket::Candidate candidate;
@@ -1905,7 +1892,7 @@
// 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();
+ Init(NULL);
cricket::Candidate candidate1;
candidate1.set_component(1);
JsepIceCandidate ice_candidate1(kMediaContentName0, kMediaContentIndex0,
@@ -1958,7 +1945,7 @@
// 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 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 @@
// 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 @@
// 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<SessionDescriptionInterface> offer(CreateOffer());
@@ -2099,7 +2086,7 @@
// 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();
+ Init(NULL);
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
ASSERT_TRUE(offer != NULL);
@@ -2113,7 +2100,7 @@
// 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();
+ Init(NULL);
// Test Audio only offer.
mediastream_signaling_.UseOptionsAudioOnly();
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
@@ -2136,7 +2123,7 @@
// 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 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 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 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 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<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
@@ -2236,7 +2223,7 @@
// 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();
+ Init(NULL);
// Create a remote offer with audio only.
cricket::MediaSessionOptions options;
@@ -2259,7 +2246,7 @@
// 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<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
@@ -2280,7 +2267,7 @@
// 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();
+ Init(NULL);
// Create a remote offer with audio and video content.
rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
@@ -2304,7 +2291,7 @@
// Test that an answer contains the correct media content descriptions when
// constraints have been set and streams are sent.
TEST_F(WebRtcSessionTest, CreateAnswerWithConstraints) {
- Init();
+ Init(NULL);
// Create a remote offer with audio and video content.
rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
@@ -2332,7 +2319,7 @@
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, CreateAnswerWithoutCNCodecs) {
AddCNCodecs();
- Init();
+ Init(NULL);
// Create a remote offer with audio and video content.
rtc::scoped_ptr<JsepSessionDescription> offer(CreateRemoteOffer());
SetRemoteDescriptionWithoutError(offer.release());
@@ -2367,7 +2354,7 @@
// 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 @@
// 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, VerifyCryptoParamsInSDP) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
VerifyCryptoParams(offer->description());
@@ -2488,26 +2475,26 @@
TEST_F(WebRtcSessionTest, VerifyNoCryptoParamsInSDP) {
options_.disable_encryption = true;
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
scoped_ptr<SessionDescriptionInterface> 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<SessionDescriptionInterface> offer(CreateOffer());
@@ -2521,7 +2508,7 @@
// 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();
+ Init(NULL);
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateRemoteOffer());
std::string sdp;
RemoveIceUfragPwdLines(offer.get(), &sdp);
@@ -2533,7 +2520,7 @@
// 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<SessionDescriptionInterface> offer(CreateOffer());
@@ -2559,7 +2546,7 @@
// 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<SessionDescriptionInterface> offer(CreateRemoteOffer());
std::string sdp;
@@ -2583,7 +2570,7 @@
// 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();
+ Init(NULL);
EXPECT_TRUE((cricket::PORTALLOCATOR_ENABLE_BUNDLE &
allocator_->flags()) == cricket::PORTALLOCATOR_ENABLE_BUNDLE);
rtc::scoped_ptr<SessionDescriptionInterface> offer(CreateOffer());
@@ -2600,7 +2587,7 @@
}
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 @@
// 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, SetAudioPlayout) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@@ -2696,7 +2683,7 @@
}
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, AudioRendererForLocalStream) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@@ -2749,7 +2736,7 @@
}
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, SetVideoSend) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
cricket::FakeVideoMediaChannel* channel = media_engine_->GetVideoChannel(0);
@@ -2791,7 +2778,7 @@
TEST_F(WebRtcSessionTest, InsertDtmf) {
// Setup
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
CreateAndSetRemoteOfferAndLocalAnswer();
FakeVoiceMediaChannel* channel = media_engine_->GetVoiceChannel(0);
@@ -2817,7 +2804,7 @@
// 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 @@
// 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 @@
// 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<SessionDescriptionInterface> answer(
@@ -2865,7 +2852,7 @@
// This test verifies the ice protocol type at initiator of the call
// if ICE RFC5245 is supported in answer.
TEST_F(WebRtcSessionTest, TestInitiatorIceInAnswer) {
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
SessionDescriptionInterface* answer = CreateRemoteAnswer(offer);
@@ -2879,7 +2866,7 @@
// 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();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
@@ -2901,7 +2888,7 @@
// 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();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
SetRemoteDescriptionWithoutError(offer);
@@ -2914,7 +2901,7 @@
// 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<SessionDescriptionInterface> offer(CreateOffer());
@@ -2946,7 +2933,7 @@
// 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 @@
// 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 @@
// 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 @@
// 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<SessionDescriptionInterface> offer(CreateOffer());
@@ -3066,7 +3053,7 @@
// 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<SessionDescriptionInterface> offer(CreateOffer());
@@ -3085,7 +3072,7 @@
// 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();
+ Init(NULL);
cricket::MediaSessionOptions options;
options.recv_video = true;
rtc::scoped_ptr<JsepSessionDescription> offer(
@@ -3116,7 +3103,7 @@
// 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();
+ Init(NULL);
cricket::MediaSessionOptions options;
options.recv_video = true;
rtc::scoped_ptr<JsepSessionDescription> offer(
@@ -3145,7 +3132,7 @@
}
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, SetSdpFailedOnSessionError) {
- Init();
+ Init(NULL);
cricket::MediaSessionOptions options;
options.recv_video = true;
@@ -3222,7 +3209,7 @@
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 @@
// 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 @@
constraints_.reset(new FakeConstraints());
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableDscp, true);
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
@@ -3500,7 +3487,7 @@
constraints_->AddOptional(
webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
true);
- Init();
+ Init(NULL);
mediastream_signaling_.SendAudioVideoStream1();
SessionDescriptionInterface* offer = CreateOffer();
@@ -3529,7 +3516,7 @@
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 4c9c879..e629bb5 100644
--- a/webrtc/p2p/base/dtlstransportchannel.h
+++ b/webrtc/p2p/base/dtlstransportchannel.h
@@ -126,9 +126,6 @@
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 13eceef..375d36d 100644
--- a/webrtc/p2p/base/fakesession.h
+++ b/webrtc/p2p/base/fakesession.h
@@ -198,9 +198,6 @@
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 735dbd5..4bbe9cf 100644
--- a/webrtc/p2p/base/p2ptransportchannel.cc
+++ b/webrtc/p2p/base/p2ptransportchannel.cc
@@ -817,7 +817,6 @@
// 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 @@
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 f8756dc..10e19f0 100644
--- a/webrtc/p2p/base/p2ptransportchannel.h
+++ b/webrtc/p2p/base/p2ptransportchannel.h
@@ -79,7 +79,6 @@
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 b032e63..f0d7d5d 100644
--- a/webrtc/p2p/base/rawtransportchannel.cc
+++ b/webrtc/p2p/base/rawtransportchannel.cc
@@ -72,10 +72,6 @@
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 a4d9ce0..bc84316 100644
--- a/webrtc/p2p/base/rawtransportchannel.h
+++ b/webrtc/p2p/base/rawtransportchannel.h
@@ -50,7 +50,6 @@
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 69b6a9c..1a126a6 100644
--- a/webrtc/p2p/base/session.cc
+++ b/webrtc/p2p/base/session.cc
@@ -823,11 +823,8 @@
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 7e89123..f809b30 100644
--- a/webrtc/p2p/base/session.h
+++ b/webrtc/p2p/base/session.h
@@ -427,8 +427,6 @@
virtual void OnMessage(rtc::Message *pmsg);
protected:
- bool IsCandidateAllocationDone() const;
-
State state_;
Error error_;
std::string error_desc_;
@@ -442,6 +440,7 @@
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 fb592e5..d50f025 100644
--- a/webrtc/p2p/base/transportchannel.h
+++ b/webrtc/p2p/base/transportchannel.h
@@ -81,7 +81,6 @@
// 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 a8535fa..e6fb557 100644
--- a/webrtc/p2p/base/transportchannelproxy.cc
+++ b/webrtc/p2p/base/transportchannelproxy.cc
@@ -104,21 +104,6 @@
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 46803f7..188039e 100644
--- a/webrtc/p2p/base/transportchannelproxy.h
+++ b/webrtc/p2p/base/transportchannelproxy.h
@@ -52,7 +52,6 @@
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);
