From c2dd5ee2c05b466949fedae3fcfac63838104392 Mon Sep 17 00:00:00 2001 From: "perkj@webrtc.org" Date: Tue, 4 Nov 2014 11:31:29 +0000 Subject: [PATCH] Prepare for removal of PeerConnectionObserver::OnError. Prepare for removal of constraints to PeerConnection::AddStream. OnError has never been implemented and has been removed from the spec. Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d --- talk/app/webrtc/java/jni/peerconnection_jni.cc | 14 ++------------ .../java/src/org/webrtc/PeerConnection.java | 11 +++-------- .../src/org/webrtc/PeerConnectionTest.java | 7 +------ talk/app/webrtc/objc/RTCPeerConnection.mm | 6 ++---- .../app/webrtc/objc/RTCPeerConnectionObserver.h | 2 -- .../webrtc/objc/RTCPeerConnectionObserver.mm | 4 ---- talk/app/webrtc/objc/public/RTCPeerConnection.h | 3 +-- .../objc/public/RTCPeerConnectionDelegate.h | 3 --- .../objctests/RTCPeerConnectionSyncObserver.m | 5 ----- .../webrtc/objctests/RTCPeerConnectionTest.mm | 3 +-- talk/app/webrtc/peerconnection.cc | 4 +--- talk/app/webrtc/peerconnection.h | 3 +-- talk/app/webrtc/peerconnection_unittest.cc | 3 +-- .../webrtc/peerconnectionfactory_unittest.cc | 3 ++- talk/app/webrtc/peerconnectioninterface.h | 17 ++++++++++++----- .../webrtc/peerconnectioninterface_unittest.cc | 11 +++++------ talk/app/webrtc/peerconnectionproxy.h | 3 +-- .../webrtc/test/peerconnectiontestwrapper.cc | 2 +- .../app/webrtc/test/peerconnectiontestwrapper.h | 1 - .../appspot/apprtc/PeerConnectionClient.java | 11 +++-------- .../objc/AppRTCDemo/APPRTCConnectionManager.m | 12 +----------- .../examples/peerconnection/client/conductor.cc | 11 +---------- talk/examples/peerconnection/client/conductor.h | 3 +-- 23 files changed, 40 insertions(+), 102 deletions(-) diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/talk/app/webrtc/java/jni/peerconnection_jni.cc index d82a54f4e..fb36cf82e 100644 --- a/talk/app/webrtc/java/jni/peerconnection_jni.cc +++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc @@ -586,13 +586,6 @@ class PCOJava : public PeerConnectionObserver { CHECK_EXCEPTION(jni()) << "error during CallVoidMethod"; } - virtual void OnError() OVERRIDE { - ScopedLocalRefFrame local_ref_frame(jni()); - jmethodID m = GetMethodID(jni(), *j_observer_class_, "onError", "()V"); - jni()->CallVoidMethod(*j_observer_global_, m); - CHECK_EXCEPTION(jni()) << "error during CallVoidMethod"; - } - virtual void OnSignalingChange( PeerConnectionInterface::SignalingState new_state) OVERRIDE { ScopedLocalRefFrame local_ref_frame(jni()); @@ -3144,12 +3137,9 @@ JOW(jboolean, PeerConnection_nativeAddIceCandidate)( } JOW(jboolean, PeerConnection_nativeAddLocalStream)( - JNIEnv* jni, jobject j_pc, jlong native_stream, jobject j_constraints) { - scoped_ptr constraints( - new ConstraintsWrapper(jni, j_constraints)); + JNIEnv* jni, jobject j_pc, jlong native_stream) { return ExtractNativePC(jni, j_pc)->AddStream( - reinterpret_cast(native_stream), - constraints.get()); + reinterpret_cast(native_stream)); } JOW(void, PeerConnection_nativeRemoveLocalStream)( diff --git a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java b/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java index c2617def1..3aef6ff8e 100644 --- a/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java +++ b/talk/app/webrtc/java/src/org/webrtc/PeerConnection.java @@ -71,9 +71,6 @@ public class PeerConnection { /** Triggered when a new ICE candidate has been found. */ public void onIceCandidate(IceCandidate candidate); - /** Triggered on any error. */ - public void onError(); - /** Triggered when media is received on a new stream from remote peer. */ public void onAddStream(MediaStream stream); @@ -147,9 +144,8 @@ public class PeerConnection { candidate.sdpMid, candidate.sdpMLineIndex, candidate.sdp); } - public boolean addStream( - MediaStream stream, MediaConstraints constraints) { - boolean ret = nativeAddLocalStream(stream.nativeStream, constraints); + public boolean addStream(MediaStream stream) { + boolean ret = nativeAddLocalStream(stream.nativeStream); if (!ret) { return false; } @@ -194,8 +190,7 @@ public class PeerConnection { private native boolean nativeAddIceCandidate( String sdpMid, int sdpMLineIndex, String iceCandidateSdp); - private native boolean nativeAddLocalStream( - long nativeStream, MediaConstraints constraints); + private native boolean nativeAddLocalStream(long nativeStream); private native void nativeRemoveLocalStream(long nativeStream); diff --git a/talk/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java b/talk/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java index 240e996bd..048d92b26 100644 --- a/talk/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java +++ b/talk/app/webrtc/javatests/src/org/webrtc/PeerConnectionTest.java @@ -117,11 +117,6 @@ public class PeerConnectionTest extends TestCase { ++expectedErrors; } - @Override - public synchronized void onError() { - assertTrue(--expectedErrors >= 0); - } - public synchronized void expectSetSize() { if (RENDER_TO_GUI) { // When new frames are delivered to the GUI renderer we don't get @@ -489,7 +484,7 @@ public class PeerConnectionTest extends TestCase { lMS.addTrack(videoTrack); lMS.addTrack(factory.createAudioTrack( audioTrackId, factory.createAudioSource(new MediaConstraints()))); - pc.addStream(lMS, new MediaConstraints()); + pc.addStream(lMS); return new WeakReference(lMS); } diff --git a/talk/app/webrtc/objc/RTCPeerConnection.mm b/talk/app/webrtc/objc/RTCPeerConnection.mm index 72ba37380..925de7339 100644 --- a/talk/app/webrtc/objc/RTCPeerConnection.mm +++ b/talk/app/webrtc/objc/RTCPeerConnection.mm @@ -151,10 +151,8 @@ class RTCStatsObserver : public StatsObserver { return self.peerConnection->AddIceCandidate(iceCandidate.get()); } -- (BOOL)addStream:(RTCMediaStream*)stream - constraints:(RTCMediaConstraints*)constraints { - BOOL ret = self.peerConnection->AddStream(stream.mediaStream, - constraints.constraints); +- (BOOL)addStream:(RTCMediaStream*)stream { + BOOL ret = self.peerConnection->AddStream(stream.mediaStream); if (!ret) { return NO; } diff --git a/talk/app/webrtc/objc/RTCPeerConnectionObserver.h b/talk/app/webrtc/objc/RTCPeerConnectionObserver.h index f66b5672e..8378ff8bd 100644 --- a/talk/app/webrtc/objc/RTCPeerConnectionObserver.h +++ b/talk/app/webrtc/objc/RTCPeerConnectionObserver.h @@ -41,8 +41,6 @@ class RTCPeerConnectionObserver : public PeerConnectionObserver { RTCPeerConnectionObserver(RTCPeerConnection* peerConnection); virtual ~RTCPeerConnectionObserver(); - virtual void OnError() OVERRIDE; - // Triggered when the SignalingState changed. virtual void OnSignalingChange( PeerConnectionInterface::SignalingState new_state) OVERRIDE; diff --git a/talk/app/webrtc/objc/RTCPeerConnectionObserver.mm b/talk/app/webrtc/objc/RTCPeerConnectionObserver.mm index a0206e5b2..f4cab7fc0 100644 --- a/talk/app/webrtc/objc/RTCPeerConnectionObserver.mm +++ b/talk/app/webrtc/objc/RTCPeerConnectionObserver.mm @@ -46,10 +46,6 @@ RTCPeerConnectionObserver::RTCPeerConnectionObserver( RTCPeerConnectionObserver::~RTCPeerConnectionObserver() { } -void RTCPeerConnectionObserver::OnError() { - [_peerConnection.delegate peerConnectionOnError:_peerConnection]; -} - void RTCPeerConnectionObserver::OnSignalingChange( PeerConnectionInterface::SignalingState new_state) { RTCSignalingState state = diff --git a/talk/app/webrtc/objc/public/RTCPeerConnection.h b/talk/app/webrtc/objc/public/RTCPeerConnection.h index 32a98306e..6d47f77de 100644 --- a/talk/app/webrtc/objc/public/RTCPeerConnection.h +++ b/talk/app/webrtc/objc/public/RTCPeerConnection.h @@ -64,8 +64,7 @@ // Add a new MediaStream to be sent on this PeerConnection. // Note that a SessionDescription negotiation is needed before the // remote peer can receive the stream. -- (BOOL)addStream:(RTCMediaStream *)stream - constraints:(RTCMediaConstraints *)constraints; +- (BOOL)addStream:(RTCMediaStream *)stream; // Remove a MediaStream from this PeerConnection. // Note that a SessionDescription negotiation is need before the diff --git a/talk/app/webrtc/objc/public/RTCPeerConnectionDelegate.h b/talk/app/webrtc/objc/public/RTCPeerConnectionDelegate.h index 4b177d504..ee6ec7a64 100644 --- a/talk/app/webrtc/objc/public/RTCPeerConnectionDelegate.h +++ b/talk/app/webrtc/objc/public/RTCPeerConnectionDelegate.h @@ -38,9 +38,6 @@ // implemented to get messages from PeerConnection. @protocol RTCPeerConnectionDelegate -// Triggered when there is an error. -- (void)peerConnectionOnError:(RTCPeerConnection *)peerConnection; - // Triggered when the SignalingState changed. - (void)peerConnection:(RTCPeerConnection *)peerConnection signalingStateChanged:(RTCSignalingState)stateChanged; diff --git a/talk/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m b/talk/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m index c3f898a29..fbcf217c5 100644 --- a/talk/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m +++ b/talk/app/webrtc/objctests/RTCPeerConnectionSyncObserver.m @@ -151,11 +151,6 @@ #pragma mark - RTCPeerConnectionDelegate methods -- (void)peerConnectionOnError:(RTCPeerConnection*)peerConnection { - NSLog(@"RTCPeerConnectionDelegate::onError"); - NSAssert(--_expectedErrors >= 0, @"Unexpected error"); -} - - (void)peerConnection:(RTCPeerConnection*)peerConnection signalingStateChanged:(RTCSignalingState)stateChanged { int expectedState = [self popFirstElementAsInt:_expectedSignalingChanges]; diff --git a/talk/app/webrtc/objctests/RTCPeerConnectionTest.mm b/talk/app/webrtc/objctests/RTCPeerConnectionTest.mm index 909503ac7..6c5950b89 100644 --- a/talk/app/webrtc/objctests/RTCPeerConnectionTest.mm +++ b/talk/app/webrtc/objctests/RTCPeerConnectionTest.mm @@ -89,8 +89,7 @@ [localMediaStream addVideoTrack:videoTrack]; RTCAudioTrack* audioTrack = [factory audioTrackWithID:audioTrackID]; [localMediaStream addAudioTrack:audioTrack]; - RTCMediaConstraints* constraints = [[RTCMediaConstraints alloc] init]; - [pc addStream:localMediaStream constraints:constraints]; + [pc addStream:localMediaStream]; return localMediaStream; } diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc index 479948940..64ddcad1b 100644 --- a/talk/app/webrtc/peerconnection.cc +++ b/talk/app/webrtc/peerconnection.cc @@ -404,8 +404,7 @@ PeerConnection::remote_streams() { return mediastream_signaling_->remote_streams(); } -bool PeerConnection::AddStream(MediaStreamInterface* local_stream, - const MediaConstraintsInterface* constraints) { +bool PeerConnection::AddStream(MediaStreamInterface* local_stream) { if (IsClosed()) { return false; } @@ -413,7 +412,6 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream, local_stream)) return false; - // TODO(perkj): Implement support for MediaConstraints in AddStream. if (!mediastream_signaling_->AddLocalStream(local_stream)) { return false; } diff --git a/talk/app/webrtc/peerconnection.h b/talk/app/webrtc/peerconnection.h index fb038020e..68aa15465 100644 --- a/talk/app/webrtc/peerconnection.h +++ b/talk/app/webrtc/peerconnection.h @@ -65,8 +65,7 @@ class PeerConnection : public PeerConnectionInterface, PeerConnectionObserver* observer); virtual rtc::scoped_refptr local_streams(); virtual rtc::scoped_refptr remote_streams(); - virtual bool AddStream(MediaStreamInterface* local_stream, - const MediaConstraintsInterface* constraints); + virtual bool AddStream(MediaStreamInterface* local_stream); virtual void RemoveStream(MediaStreamInterface* local_stream); virtual rtc::scoped_refptr CreateDtmfSender( diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc index ce745de25..c250eeab5 100644 --- a/talk/app/webrtc/peerconnection_unittest.cc +++ b/talk/app/webrtc/peerconnection_unittest.cc @@ -179,7 +179,7 @@ class PeerConnectionTestClientBase stream->AddTrack(CreateLocalVideoTrack(stream_label)); } - EXPECT_TRUE(peer_connection_->AddStream(stream, NULL)); + EXPECT_TRUE(peer_connection_->AddStream(stream)); } size_t NumberOfLocalMediaStreams() { @@ -426,7 +426,6 @@ class PeerConnectionTestClientBase } // PeerConnectionObserver callbacks. - virtual void OnError() {} virtual void OnMessage(const std::string&) {} virtual void OnSignalingMessage(const std::string& /*msg*/) {} virtual void OnSignalingChange( diff --git a/talk/app/webrtc/peerconnectionfactory_unittest.cc b/talk/app/webrtc/peerconnectionfactory_unittest.cc index 5995c46dc..e687b8bc6 100644 --- a/talk/app/webrtc/peerconnectionfactory_unittest.cc +++ b/talk/app/webrtc/peerconnectionfactory_unittest.cc @@ -40,6 +40,7 @@ #include "webrtc/base/thread.h" using webrtc::FakeVideoTrackRenderer; +using webrtc::DataChannelInterface; using webrtc::MediaStreamInterface; using webrtc::PeerConnectionFactoryInterface; using webrtc::PeerConnectionInterface; @@ -83,13 +84,13 @@ static const char kTurnIceServerWithIPv6Address[] = class NullPeerConnectionObserver : public PeerConnectionObserver { public: - virtual void OnError() {} virtual void OnMessage(const std::string& msg) {} virtual void OnSignalingMessage(const std::string& msg) {} virtual void OnSignalingChange( PeerConnectionInterface::SignalingState new_state) {} virtual void OnAddStream(MediaStreamInterface* stream) {} virtual void OnRemoveStream(MediaStreamInterface* stream) {} + virtual void OnDataChannel(DataChannelInterface* data_channel) {} virtual void OnRenegotiationNeeded() {} virtual void OnIceConnectionChange( PeerConnectionInterface::IceConnectionState new_state) {} diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h index 6ef48475a..edbf6e31c 100644 --- a/talk/app/webrtc/peerconnectioninterface.h +++ b/talk/app/webrtc/peerconnectioninterface.h @@ -252,11 +252,17 @@ class PeerConnectionInterface : public rtc::RefCountInterface { virtual rtc::scoped_refptr remote_streams() = 0; + // Deprecated: + // TODO(perkj): Remove once its not used by Chrome. + virtual bool AddStream(MediaStreamInterface* stream, + const MediaConstraintsInterface* constraints) { + return AddStream(stream); + } + // Add a new MediaStream to be sent on this PeerConnection. // Note that a SessionDescription negotiation is needed before the // remote peer can receive the stream. - virtual bool AddStream(MediaStreamInterface* stream, - const MediaConstraintsInterface* constraints) = 0; + virtual bool AddStream(MediaStreamInterface* stream) = 0; // Remove a MediaStream from this PeerConnection. // Note that a SessionDescription negotiation is need before the @@ -344,7 +350,9 @@ class PeerConnectionObserver { kIceState, }; - virtual void OnError() = 0; + // Deprecated. + // TODO(perkj): Remove once its not used by Chrome. + virtual void OnError() {} // Triggered when the SignalingState changed. virtual void OnSignalingChange( @@ -361,8 +369,7 @@ class PeerConnectionObserver { virtual void OnRemoveStream(MediaStreamInterface* stream) = 0; // Triggered when a remote peer open a data channel. - // TODO(perkj): Make pure virtual. - virtual void OnDataChannel(DataChannelInterface* data_channel) {} + virtual void OnDataChannel(DataChannelInterface* data_channel) = 0; // Triggered when renegotiation is needed, for example the ICE has restarted. virtual void OnRenegotiationNeeded() = 0; diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc index bf6067340..3be628055 100644 --- a/talk/app/webrtc/peerconnectioninterface_unittest.cc +++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc @@ -132,7 +132,6 @@ class MockPeerConnectionObserver : public PeerConnectionObserver { state_ = pc_->signaling_state(); } } - virtual void OnError() {} virtual void OnSignalingChange( PeerConnectionInterface::SignalingState new_state) { EXPECT_EQ(pc_->signaling_state(), new_state); @@ -320,7 +319,7 @@ class PeerConnectionInterfaceTest : public testing::Test { scoped_refptr video_track( pc_factory_->CreateVideoTrack(label + "v0", video_source)); stream->AddTrack(video_track.get()); - EXPECT_TRUE(pc_->AddStream(stream, NULL)); + EXPECT_TRUE(pc_->AddStream(stream)); EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout); observer_.renegotiation_needed_ = false; } @@ -332,7 +331,7 @@ class PeerConnectionInterfaceTest : public testing::Test { scoped_refptr audio_track( pc_factory_->CreateAudioTrack(label + "a0", NULL)); stream->AddTrack(audio_track.get()); - EXPECT_TRUE(pc_->AddStream(stream, NULL)); + EXPECT_TRUE(pc_->AddStream(stream)); EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout); observer_.renegotiation_needed_ = false; } @@ -350,7 +349,7 @@ class PeerConnectionInterfaceTest : public testing::Test { scoped_refptr video_track( pc_factory_->CreateVideoTrack(video_track_label, NULL)); stream->AddTrack(video_track.get()); - EXPECT_TRUE(pc_->AddStream(stream, NULL)); + EXPECT_TRUE(pc_->AddStream(stream)); EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout); observer_.renegotiation_needed_ = false; } @@ -574,7 +573,7 @@ TEST_F(PeerConnectionInterfaceTest, AddStreams) { pc_factory_->CreateAudioTrack( kStreamLabel3, static_cast(NULL))); stream->AddTrack(audio_track.get()); - EXPECT_TRUE(pc_->AddStream(stream, NULL)); + EXPECT_TRUE(pc_->AddStream(stream)); EXPECT_EQ(3u, pc_->local_streams()->count()); // Remove the third stream. @@ -1180,7 +1179,7 @@ TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) { pc_->Close(); pc_->RemoveStream(local_stream); - EXPECT_FALSE(pc_->AddStream(local_stream, NULL)); + EXPECT_FALSE(pc_->AddStream(local_stream)); ASSERT_FALSE(local_stream->GetAudioTracks().empty()); rtc::scoped_refptr dtmf_sender( diff --git a/talk/app/webrtc/peerconnectionproxy.h b/talk/app/webrtc/peerconnectionproxy.h index ed26eb870..852d8520e 100644 --- a/talk/app/webrtc/peerconnectionproxy.h +++ b/talk/app/webrtc/peerconnectionproxy.h @@ -39,8 +39,7 @@ BEGIN_PROXY_MAP(PeerConnection) local_streams) PROXY_METHOD0(rtc::scoped_refptr, remote_streams) - PROXY_METHOD2(bool, AddStream, MediaStreamInterface*, - const MediaConstraintsInterface*) + PROXY_METHOD1(bool, AddStream, MediaStreamInterface*) PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*) PROXY_METHOD1(rtc::scoped_refptr, CreateDtmfSender, AudioTrackInterface*) diff --git a/talk/app/webrtc/test/peerconnectiontestwrapper.cc b/talk/app/webrtc/test/peerconnectiontestwrapper.cc index 24932b89f..e3b801574 100644 --- a/talk/app/webrtc/test/peerconnectiontestwrapper.cc +++ b/talk/app/webrtc/test/peerconnectiontestwrapper.cc @@ -253,7 +253,7 @@ void PeerConnectionTestWrapper::GetAndAddUserMedia( bool video, const webrtc::FakeConstraints& video_constraints) { rtc::scoped_refptr stream = GetUserMedia(audio, audio_constraints, video, video_constraints); - EXPECT_TRUE(peer_connection_->AddStream(stream, NULL)); + EXPECT_TRUE(peer_connection_->AddStream(stream)); } rtc::scoped_refptr diff --git a/talk/app/webrtc/test/peerconnectiontestwrapper.h b/talk/app/webrtc/test/peerconnectiontestwrapper.h index d4a0e4ecb..d8299ec85 100644 --- a/talk/app/webrtc/test/peerconnectiontestwrapper.h +++ b/talk/app/webrtc/test/peerconnectiontestwrapper.h @@ -57,7 +57,6 @@ class PeerConnectionTestWrapper const webrtc::DataChannelInit& init); // Implements PeerConnectionObserver. - virtual void OnError() {} virtual void OnSignalingChange( webrtc::PeerConnectionInterface::SignalingState new_state) {} virtual void OnStateChange( diff --git a/talk/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java b/talk/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java index daea5fb06..9c917bbbb 100644 --- a/talk/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java +++ b/talk/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java @@ -110,7 +110,7 @@ public class PeerConnectionClient { if (videoConstraints != null) { videoMediaStream = factory.createLocalMediaStream("ARDAMSVideo"); videoMediaStream.addTrack(createVideoTrack(useFrontFacingCamera)); - pc.addStream(videoMediaStream, new MediaConstraints()); + pc.addStream(videoMediaStream); } if (appRtcParameters.audioConstraints != null) { @@ -118,7 +118,7 @@ public class PeerConnectionClient { lMS.addTrack(factory.createAudioTrack( "ARDAMSa0", factory.createAudioSource(appRtcParameters.audioConstraints))); - pc.addStream(lMS, new MediaConstraints()); + pc.addStream(lMS); } } @@ -409,7 +409,7 @@ public class PeerConnectionClient { useFrontFacingCamera = !useFrontFacingCamera; VideoTrack newTrack = createVideoTrack(useFrontFacingCamera); videoMediaStream.addTrack(newTrack); - pc.addStream(videoMediaStream, new MediaConstraints()); + pc.addStream(videoMediaStream); SessionDescription remoteDesc = pc.getRemoteDescription(); if (localSdp == null || remoteDesc == null) { @@ -440,11 +440,6 @@ public class PeerConnectionClient { }); } - @Override - public void onError() { - reportError("PeerConnection error!"); - } - @Override public void onSignalingChange( PeerConnection.SignalingState newState) { diff --git a/talk/examples/objc/AppRTCDemo/APPRTCConnectionManager.m b/talk/examples/objc/AppRTCDemo/APPRTCConnectionManager.m index b411a6215..9a39528df 100644 --- a/talk/examples/objc/AppRTCDemo/APPRTCConnectionManager.m +++ b/talk/examples/objc/AppRTCDemo/APPRTCConnectionManager.m @@ -170,7 +170,7 @@ #endif [lms addAudioTrack:[self.peerConnectionFactory audioTrackWithID:@"ARDAMSa0"]]; - [self.peerConnection addStream:lms constraints:constraints]; + [self.peerConnection addStream:lms]; [self.logger logMessage:@"onICEServers - added local stream."]; } @@ -243,16 +243,6 @@ #pragma mark - RTCPeerConnectionDelegate -- (void)peerConnectionOnError:(RTCPeerConnection*)peerConnection { - dispatch_async(dispatch_get_main_queue(), ^{ - NSString* message = @"PeerConnection error"; - NSLog(@"%@", message); - NSAssert(NO, @"PeerConnection failed."); - [self.delegate connectionManager:self - didErrorWithMessage:message]; - }); -} - - (void)peerConnection:(RTCPeerConnection*)peerConnection signalingStateChanged:(RTCSignalingState)stateChanged { dispatch_async(dispatch_get_main_queue(), ^{ diff --git a/talk/examples/peerconnection/client/conductor.cc b/talk/examples/peerconnection/client/conductor.cc index f49aee6c6..e81f7fc8c 100644 --- a/talk/examples/peerconnection/client/conductor.cc +++ b/talk/examples/peerconnection/client/conductor.cc @@ -137,11 +137,6 @@ void Conductor::EnsureStreamingUI() { // PeerConnectionObserver implementation. // -void Conductor::OnError() { - LOG(LS_ERROR) << __FUNCTION__; - main_wnd_->QueueUIThreadCallback(PEER_CONNECTION_ERROR, NULL); -} - // Called when a remote stream is added void Conductor::OnAddStream(webrtc::MediaStreamInterface* stream) { LOG(INFO) << __FUNCTION__ << " " << stream->label(); @@ -373,7 +368,7 @@ void Conductor::AddStreams() { stream->AddTrack(audio_track); stream->AddTrack(video_track); - if (!peer_connection_->AddStream(stream, NULL)) { + if (!peer_connection_->AddStream(stream)) { LOG(LS_ERROR) << "Adding stream to PeerConnection failed"; } typedef std::pairMessageBox("Error", "an unknown error occurred", true); - break; - case NEW_STREAM_ADDED: { webrtc::MediaStreamInterface* stream = reinterpret_cast( diff --git a/talk/examples/peerconnection/client/conductor.h b/talk/examples/peerconnection/client/conductor.h index 0aff53132..3ef525355 100644 --- a/talk/examples/peerconnection/client/conductor.h +++ b/talk/examples/peerconnection/client/conductor.h @@ -58,7 +58,6 @@ class Conductor MEDIA_CHANNELS_INITIALIZED = 1, PEER_CONNECTION_CLOSED, SEND_MESSAGE_TO_PEER, - PEER_CONNECTION_ERROR, NEW_STREAM_ADDED, STREAM_REMOVED, }; @@ -80,11 +79,11 @@ class Conductor // // PeerConnectionObserver implementation. // - virtual void OnError(); virtual void OnStateChange( webrtc::PeerConnectionObserver::StateType state_changed) {} virtual void OnAddStream(webrtc::MediaStreamInterface* stream); virtual void OnRemoveStream(webrtc::MediaStreamInterface* stream); + virtual void OnDataChannel(webrtc::DataChannelInterface* channel) {} virtual void OnRenegotiationNeeded() {} virtual void OnIceChange() {} virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate);