1 files changed, 142 insertions, 33 deletions

diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index 63163fd651..c3789b7dd8 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -26,9 +26,9 @@
  */
 
 #include <string>
+#include <utility>
 
 #include "talk/app/webrtc/audiotrack.h"
-#include "talk/app/webrtc/fakeportallocatorfactory.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
 #include "talk/app/webrtc/mediastream.h"
 #include "talk/app/webrtc/mediastreaminterface.h"
@@ -37,6 +37,9 @@
 #include "talk/app/webrtc/rtpreceiverinterface.h"
 #include "talk/app/webrtc/rtpsenderinterface.h"
 #include "talk/app/webrtc/streamcollection.h"
+#ifdef WEBRTC_ANDROID
+#include "talk/app/webrtc/test/androidtestinitializer.h"
+#endif
 #include "talk/app/webrtc/test/fakeconstraints.h"
 #include "talk/app/webrtc/test/fakedtlsidentitystore.h"
 #include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
@@ -52,6 +55,7 @@
 #include "webrtc/base/sslstreamadapter.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/thread.h"
+#include "webrtc/p2p/client/fakeportallocator.h"
 
 static const char kStreamLabel1[] = "local_stream_1";
 static const char kStreamLabel2[] = "local_stream_2";
@@ -258,7 +262,6 @@ using webrtc::AudioTrackInterface;
 using webrtc::DataBuffer;
 using webrtc::DataChannelInterface;
 using webrtc::FakeConstraints;
-using webrtc::FakePortAllocatorFactory;
 using webrtc::IceCandidateInterface;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStream;
@@ -270,7 +273,6 @@ using webrtc::MockSetSessionDescriptionObserver;
 using webrtc::MockStatsObserver;
 using webrtc::PeerConnectionInterface;
 using webrtc::PeerConnectionObserver;
-using webrtc::PortAllocatorFactoryInterface;
 using webrtc::RtpReceiverInterface;
 using webrtc::RtpSenderInterface;
 using webrtc::SdpParseError;
@@ -515,6 +517,12 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
 
 class PeerConnectionInterfaceTest : public testing::Test {
  protected:
+  PeerConnectionInterfaceTest() {
+#ifdef WEBRTC_ANDROID
+    webrtc::InitializeAndroidObjects();
+#endif
+  }
+
   virtual void SetUp() {
     pc_factory_ = webrtc::CreatePeerConnectionFactory(
         rtc::Thread::Current(), rtc::Thread::Current(), NULL, NULL,
@@ -533,15 +541,17 @@ class PeerConnectionInterfaceTest : public testing::Test {
   void CreatePeerConnection(const std::string& uri,
                             const std::string& password,
                             webrtc::MediaConstraintsInterface* constraints) {
+    PeerConnectionInterface::RTCConfiguration config;
     PeerConnectionInterface::IceServer server;
-    PeerConnectionInterface::IceServers servers;
     if (!uri.empty()) {
       server.uri = uri;
       server.password = password;
-      servers.push_back(server);
+      config.servers.push_back(server);
     }
 
-    port_allocator_factory_ = FakePortAllocatorFactory::Create();
+    rtc::scoped_ptr<cricket::FakePortAllocator> port_allocator(
+        new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
+    port_allocator_ = port_allocator.get();
 
     // DTLS does not work in a loopback call, so is disabled for most of the
     // tests in this file. We only create a FakeIdentityService if the test
@@ -562,52 +572,47 @@ class PeerConnectionInterfaceTest : public testing::Test {
                        nullptr) && dtls) {
       dtls_identity_store.reset(new FakeDtlsIdentityStore());
     }
-    pc_ = pc_factory_->CreatePeerConnection(servers, constraints,
-                                            port_allocator_factory_.get(),
-                                            dtls_identity_store.Pass(),
-                                            &observer_);
+    pc_ = pc_factory_->CreatePeerConnection(
+        config, constraints, std::move(port_allocator),
+        std::move(dtls_identity_store), &observer_);
     ASSERT_TRUE(pc_.get() != NULL);
     observer_.SetPeerConnectionInterface(pc_.get());
     EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
   }
 
   void CreatePeerConnectionExpectFail(const std::string& uri) {
+    PeerConnectionInterface::RTCConfiguration config;
     PeerConnectionInterface::IceServer server;
-    PeerConnectionInterface::IceServers servers;
     server.uri = uri;
-    servers.push_back(server);
+    config.servers.push_back(server);
 
-    scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store;
-    port_allocator_factory_ = FakePortAllocatorFactory::Create();
     scoped_refptr<PeerConnectionInterface> pc;
-    pc = pc_factory_->CreatePeerConnection(
-        servers, nullptr, port_allocator_factory_.get(),
-        dtls_identity_store.Pass(), &observer_);
-    ASSERT_EQ(nullptr, pc);
+    pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr, nullptr,
+                                           &observer_);
+    EXPECT_EQ(nullptr, pc);
   }
 
   void CreatePeerConnectionWithDifferentConfigurations() {
     CreatePeerConnection(kStunAddressOnly, "", NULL);
-    EXPECT_EQ(1u, port_allocator_factory_->stun_configs().size());
-    EXPECT_EQ(0u, port_allocator_factory_->turn_configs().size());
-    EXPECT_EQ("address",
-        port_allocator_factory_->stun_configs()[0].server.hostname());
+    EXPECT_EQ(1u, port_allocator_->stun_servers().size());
+    EXPECT_EQ(0u, port_allocator_->turn_servers().size());
+    EXPECT_EQ("address", port_allocator_->stun_servers().begin()->hostname());
     EXPECT_EQ(kDefaultStunPort,
-        port_allocator_factory_->stun_configs()[0].server.port());
+              port_allocator_->stun_servers().begin()->port());
 
     CreatePeerConnectionExpectFail(kStunInvalidPort);
     CreatePeerConnectionExpectFail(kStunAddressPortAndMore1);
     CreatePeerConnectionExpectFail(kStunAddressPortAndMore2);
 
     CreatePeerConnection(kTurnIceServerUri, kTurnPassword, NULL);
-    EXPECT_EQ(0u, port_allocator_factory_->stun_configs().size());
-    EXPECT_EQ(1u, port_allocator_factory_->turn_configs().size());
+    EXPECT_EQ(0u, port_allocator_->stun_servers().size());
+    EXPECT_EQ(1u, port_allocator_->turn_servers().size());
     EXPECT_EQ(kTurnUsername,
-              port_allocator_factory_->turn_configs()[0].username);
+              port_allocator_->turn_servers()[0].credentials.username);
     EXPECT_EQ(kTurnPassword,
-              port_allocator_factory_->turn_configs()[0].password);
+              port_allocator_->turn_servers()[0].credentials.password);
     EXPECT_EQ(kTurnHostname,
-              port_allocator_factory_->turn_configs()[0].server.hostname());
+              port_allocator_->turn_servers()[0].ports[0].address.hostname());
   }
 
   void ReleasePeerConnection() {
@@ -926,7 +931,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
     ASSERT_TRUE(stream->AddTrack(video_track));
   }
 
-  scoped_refptr<FakePortAllocatorFactory> port_allocator_factory_;
+  cricket::FakePortAllocator* port_allocator_ = nullptr;
   scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
   scoped_refptr<PeerConnectionInterface> pc_;
   MockPeerConnectionObserver observer_;
@@ -1156,6 +1161,64 @@ TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
   EXPECT_NE(audio_ssrc, video_ssrc);
 }
 
+// Test that it's possible to call AddTrack on a MediaStream after adding
+// the stream to a PeerConnection.
+// TODO(deadbeef): Remove this test once this behavior is no longer supported.
+TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
+  CreatePeerConnection();
+  // Create audio stream and add to PeerConnection.
+  AddVoiceStream(kStreamLabel1);
+  MediaStreamInterface* stream = pc_->local_streams()->at(0);
+
+  // Add video track to the audio-only stream.
+  scoped_refptr<VideoTrackInterface> video_track(
+      pc_factory_->CreateVideoTrack("video_label", nullptr));
+  stream->AddTrack(video_track.get());
+
+  scoped_ptr<SessionDescriptionInterface> offer;
+  ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
+
+  const cricket::MediaContentDescription* video_desc =
+      cricket::GetFirstVideoContentDescription(offer->description());
+  EXPECT_TRUE(video_desc != nullptr);
+}
+
+// Test that it's possible to call RemoveTrack on a MediaStream after adding
+// the stream to a PeerConnection.
+// TODO(deadbeef): Remove this test once this behavior is no longer supported.
+TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
+  CreatePeerConnection();
+  // Create audio/video stream and add to PeerConnection.
+  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
+  MediaStreamInterface* stream = pc_->local_streams()->at(0);
+
+  // Remove the video track.
+  stream->RemoveTrack(stream->GetVideoTracks()[0]);
+
+  scoped_ptr<SessionDescriptionInterface> offer;
+  ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
+
+  const cricket::MediaContentDescription* video_desc =
+      cricket::GetFirstVideoContentDescription(offer->description());
+  EXPECT_TRUE(video_desc == nullptr);
+}
+
+// Test creating a sender with a stream ID, and ensure the ID is populated
+// in the offer.
+TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
+  CreatePeerConnection();
+  pc_->CreateSender("video", kStreamLabel1);
+
+  scoped_ptr<SessionDescriptionInterface> offer;
+  ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
+
+  const cricket::MediaContentDescription* video_desc =
+      cricket::GetFirstVideoContentDescription(offer->description());
+  ASSERT_TRUE(video_desc != nullptr);
+  ASSERT_EQ(1u, video_desc->streams().size());
+  EXPECT_EQ(kStreamLabel1, video_desc->streams()[0].sync_label);
+}
+
 // Test that we can specify a certain track that we want statistics about.
 TEST_F(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
   InitiateCall();
@@ -1660,6 +1723,22 @@ TEST_F(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
   ASSERT_EQ(cricket::MD_INACTIVE, audio_desc->direction());
 }
 
+// Test that we can use SetConfiguration to change the ICE servers of the
+// PortAllocator.
+TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
+  CreatePeerConnection();
+
+  PeerConnectionInterface::RTCConfiguration config;
+  PeerConnectionInterface::IceServer server;
+  server.uri = "stun:test_hostname";
+  config.servers.push_back(server);
+  EXPECT_TRUE(pc_->SetConfiguration(config));
+
+  EXPECT_EQ(1u, port_allocator_->stun_servers().size());
+  EXPECT_EQ("test_hostname",
+            port_allocator_->stun_servers().begin()->hostname());
+}
+
 // Test that PeerConnection::Close changes the states to closed and all remote
 // tracks change state to ended.
 TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
@@ -1977,6 +2056,28 @@ TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
   EXPECT_EQ(0u, observer_.remote_streams()->count());
 }
 
+// This tests that when setting a new description, the old default tracks are
+// not destroyed and recreated.
+// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
+TEST_F(PeerConnectionInterfaceTest, DefaultTracksNotDestroyedAndRecreated) {
+  FakeConstraints constraints;
+  constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
+                           true);
+  CreatePeerConnection(&constraints);
+  CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
+
+  ASSERT_EQ(1u, observer_.remote_streams()->count());
+  MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
+  ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
+
+  // Set the track to "disabled", then set a new description and ensure the
+  // track is still disabled, which ensures it hasn't been recreated.
+  remote_stream->GetAudioTracks()[0]->set_enabled(false);
+  CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
+  ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
+  EXPECT_FALSE(remote_stream->GetAudioTracks()[0]->enabled());
+}
+
 // This tests that a default MediaStream is not created if a remote session
 // description is updated to not have any MediaStreams.
 TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
@@ -2020,8 +2121,10 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
   EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
 
   // Remove an audio and video track.
+  pc_->RemoveStream(reference_collection_->at(0));
   rtc::scoped_ptr<SessionDescriptionInterface> desc_2;
   CreateSessionDescriptionAndReference(1, 1, desc_2.accept());
+  pc_->AddStream(reference_collection_->at(0));
   EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
   senders = pc_->GetSenders();
   EXPECT_EQ(2u, senders.size());
@@ -2220,7 +2323,9 @@ TEST(CreateSessionOptionsTest, GetDefaultMediaSessionOptionsForOffer) {
   EXPECT_FALSE(options.has_video());
   EXPECT_TRUE(options.bundle_enabled);
   EXPECT_TRUE(options.vad_enabled);
-  EXPECT_FALSE(options.transport_options.ice_restart);
+  EXPECT_FALSE(options.audio_transport_options.ice_restart);
+  EXPECT_FALSE(options.video_transport_options.ice_restart);
+  EXPECT_FALSE(options.data_transport_options.ice_restart);
 }
 
 // Test that a correct MediaSessionOptions is created for an offer if
@@ -2255,18 +2360,22 @@ TEST(CreateSessionOptionsTest,
 
 // Test that a correct MediaSessionOptions is created to restart ice if
 // IceRestart is set. It also tests that subsequent MediaSessionOptions don't
-// have |transport_options.ice_restart| set.
+// have |audio_transport_options.ice_restart| etc. set.
 TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithIceRestart) {
   RTCOfferAnswerOptions rtc_options;
   rtc_options.ice_restart = true;
 
   cricket::MediaSessionOptions options;
   EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
-  EXPECT_TRUE(options.transport_options.ice_restart);
+  EXPECT_TRUE(options.audio_transport_options.ice_restart);
+  EXPECT_TRUE(options.video_transport_options.ice_restart);
+  EXPECT_TRUE(options.data_transport_options.ice_restart);
 
   rtc_options = RTCOfferAnswerOptions();
   EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
-  EXPECT_FALSE(options.transport_options.ice_restart);
+  EXPECT_FALSE(options.audio_transport_options.ice_restart);
+  EXPECT_FALSE(options.video_transport_options.ice_restart);
+  EXPECT_FALSE(options.data_transport_options.ice_restart);
 }
 
 // Test that the MediaConstraints in an answer don't affect if audio and video