Parameterize PeerConnection end to end tests for Unified Plan

Bug: webrtc:8765
Change-Id: If4b797be7876a7680e99c698631c29b412f7a455
Reviewed-on: https://webrtc-review.googlesource.com/41540
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21755}

pc/peerconnectionendtoend_unittest.cc

@@ -34,6 +34,7 @@
 using testing::AtLeast;
 using testing::Invoke;
 using testing::StrictMock;
+using testing::Values;
 using testing::_;
 
 using webrtc::DataChannelInterface;
@@ -41,6 +42,7 @@ using webrtc::FakeConstraints;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
 using webrtc::PeerConnectionInterface;
+using webrtc::SdpSemantics;
 
 namespace {
 
@@ -48,14 +50,13 @@ const int kMaxWait = 10000;
 
 }  // namespace
 
-class PeerConnectionEndToEndTest
+class PeerConnectionEndToEndBaseTest : public sigslot::has_slots<>,
-    : public sigslot::has_slots<>,
                                        public testing::Test {
  public:
   typedef std::vector<rtc::scoped_refptr<DataChannelInterface> >
       DataChannelList;
 
-  PeerConnectionEndToEndTest() {
+  explicit PeerConnectionEndToEndBaseTest(SdpSemantics sdp_semantics) {
     network_thread_ = rtc::Thread::CreateWithSocketServer();
     worker_thread_ = rtc::Thread::Create();
     RTC_CHECK(network_thread_->Start());
@@ -67,6 +68,7 @@ class PeerConnectionEndToEndTest
     webrtc::PeerConnectionInterface::IceServer ice_server;
     ice_server.uri = "stun:stun.l.google.com:19302";
     config_.servers.push_back(ice_server);
+    config_.sdp_semantics = sdp_semantics;
 
 #ifdef WEBRTC_ANDROID
     webrtc::InitializeAndroidObjects();
@@ -85,9 +87,9 @@ class PeerConnectionEndToEndTest
     PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
 
     caller_->SignalOnDataChannel.connect(
-        this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel);
+        this, &PeerConnectionEndToEndBaseTest::OnCallerAddedDataChanel);
     callee_->SignalOnDataChannel.connect(
-        this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel);
+        this, &PeerConnectionEndToEndBaseTest::OnCalleeAddedDataChannel);
   }
 
   void GetAndAddUserMedia() {
@@ -181,6 +183,13 @@ class PeerConnectionEndToEndTest
   webrtc::PeerConnectionInterface::RTCConfiguration config_;
 };
 
+class PeerConnectionEndToEndTest
+    : public PeerConnectionEndToEndBaseTest,
+      public ::testing::WithParamInterface<SdpSemantics> {
+ protected:
+  PeerConnectionEndToEndTest() : PeerConnectionEndToEndBaseTest(GetParam()) {}
+};
+
 namespace {
 
 std::unique_ptr<webrtc::AudioDecoder> CreateForwardingMockDecoder(
@@ -343,7 +352,7 @@ struct AudioDecoderUnicornSparklesRainbow {
 #else
 #define MAYBE_Call Call
 #endif
-TEST_F(PeerConnectionEndToEndTest, MAYBE_Call) {
+TEST_P(PeerConnectionEndToEndTest, MAYBE_Call) {
   rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =
       webrtc::CreateBuiltinAudioDecoderFactory();
   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
@@ -354,7 +363,7 @@ TEST_F(PeerConnectionEndToEndTest, MAYBE_Call) {
 }
 
 #if !defined(ADDRESS_SANITIZER)
-TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
+TEST_P(PeerConnectionEndToEndTest, CallWithLegacySdp) {
   FakeConstraints pc_constraints;
   pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                               false);
@@ -366,7 +375,7 @@ TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
 }
 #endif  // !defined(ADDRESS_SANITIZER)
 
-TEST_F(PeerConnectionEndToEndTest, CallWithCustomCodec) {
+TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
   CreatePcs(
       nullptr,
       webrtc::CreateAudioEncoderFactory<AudioEncoderUnicornSparklesRainbow>(),
@@ -379,7 +388,7 @@ TEST_F(PeerConnectionEndToEndTest, CallWithCustomCodec) {
 #ifdef HAVE_SCTP
 // Verifies that a DataChannel created before the negotiation can transition to
 // "OPEN" and transfer data.
-TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
+TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
 
@@ -404,7 +413,7 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
 
 // Verifies that a DataChannel created after the negotiation can transition to
 // "OPEN" and transfer data.
-TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
+TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
 
@@ -436,7 +445,7 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
 }
 
 // Verifies that DataChannel IDs are even/odd based on the DTLS roles.
-TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
+TEST_P(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
 
@@ -463,7 +472,7 @@ TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
 
 // Verifies that the message is received by the right remote DataChannel when
 // there are multiple DataChannels.
-TEST_F(PeerConnectionEndToEndTest,
+TEST_P(PeerConnectionEndToEndTest,
        MessageTransferBetweenTwoPairsOfDataChannels) {
   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
@@ -507,7 +516,7 @@ TEST_F(PeerConnectionEndToEndTest,
 // caused by the fact that a data channel signals that it's closed before it
 // really is. Re-enable this test once that's fixed.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
-TEST_F(PeerConnectionEndToEndTest,
+TEST_P(PeerConnectionEndToEndTest,
        DISABLED_DataChannelFromOpenWorksAfterClose) {
   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
@@ -535,7 +544,7 @@ TEST_F(PeerConnectionEndToEndTest,
 // by the application (meaning only the PeerConnection contributes to its
 // reference count), no memory access violation will occur.
 // See: https://code.google.com/p/chromium/issues/detail?id=565048
-TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
+TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
 
@@ -557,3 +566,8 @@ TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
   rtc::Thread::Current()->ProcessMessages(100);
 }
 #endif  // HAVE_SCTP
+
+INSTANTIATE_TEST_CASE_P(PeerConnectionEndToEndTest,
+                        PeerConnectionEndToEndTest,
+                        Values(SdpSemantics::kPlanB,
+                               SdpSemantics::kUnifiedPlan));

pc/test/peerconnectiontestwrapper.cc

@@ -10,6 +10,7 @@
 
 #include <string>
 #include <utility>
+#include <vector>
 
 #include "p2p/base/fakeportallocator.h"
 #include "pc/sdputils.h"
@@ -24,8 +25,10 @@ using webrtc::FakeVideoTrackRenderer;
 using webrtc::IceCandidateInterface;
 using webrtc::MediaConstraintsInterface;
 using webrtc::MediaStreamInterface;
+using webrtc::MediaStreamTrackInterface;
 using webrtc::MockSetSessionDescriptionObserver;
 using webrtc::PeerConnectionInterface;
+using webrtc::RtpReceiverInterface;
 using webrtc::SdpType;
 using webrtc::SessionDescriptionInterface;
 using webrtc::VideoTrackInterface;
@@ -99,12 +102,14 @@ PeerConnectionTestWrapper::CreateDataChannel(
   return peer_connection_->CreateDataChannel(label, &init);
 }
 
-void PeerConnectionTestWrapper::OnAddStream(
+void PeerConnectionTestWrapper::OnAddTrack(
-    rtc::scoped_refptr<MediaStreamInterface> stream) {
+    rtc::scoped_refptr<RtpReceiverInterface> receiver,
-  RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": OnAddStream";
+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
-  // TODO(ronghuawu): support multiple streams.
+  RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": OnAddTrack";
-  if (stream->GetVideoTracks().size() > 0) {
+  if (receiver->track()->kind() == MediaStreamTrackInterface::kVideoKind) {
-    renderer_.reset(new FakeVideoTrackRenderer(stream->GetVideoTracks()[0]));
+    auto* video_track =
+        static_cast<VideoTrackInterface*>(receiver->track().get());
+    renderer_ = rtc::MakeUnique<FakeVideoTrackRenderer>(video_track);
   }
 }
 
@@ -244,7 +249,14 @@ void PeerConnectionTestWrapper::GetAndAddUserMedia(
     bool video, const webrtc::FakeConstraints& video_constraints) {
   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
       GetUserMedia(audio, audio_constraints, video, video_constraints);
-  EXPECT_TRUE(peer_connection_->AddStream(stream));
+  for (auto audio_track : stream->GetAudioTracks()) {
+    EXPECT_TRUE(
+        peer_connection_->AddTrack(audio_track, {stream->label()}).ok());
+  }
+  for (auto video_track : stream->GetVideoTracks()) {
+    EXPECT_TRUE(
+        peer_connection_->AddTrack(video_track, {stream->label()}).ok());
+  }
 }
 
 rtc::scoped_refptr<webrtc::MediaStreamInterface>

pc/test/peerconnectiontestwrapper.h

@@ -13,6 +13,7 @@
 
 #include <memory>
 #include <string>
+#include <vector>
 
 #include "api/peerconnectioninterface.h"
 #include "api/test/fakeconstraints.h"
@@ -48,10 +49,10 @@ class PeerConnectionTestWrapper
   // Implements PeerConnectionObserver.
   void OnSignalingChange(
      webrtc::PeerConnectionInterface::SignalingState new_state) override {}
-  void OnAddStream(
+  void OnAddTrack(
-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
+      rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
-  void OnRemoveStream(
+      const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override {}
+          streams) override;
   void OnDataChannel(
       rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
   void OnRenegotiationNeeded() override {}

tools_webrtc/valgrind/gtest_exclude/peerconnection_unittests.gtest-memcheck.txt

@@ -1,7 +1,7 @@
 # Tests that are failing when run under memcheck.
 # https://code.google.com/p/webrtc/issues/detail?id=4387
 DtmfSenderTest.*
-PeerConnectionEndToEndTest.*
+PeerConnectionEndToEndTest*
 PeerConnectionIntegrationTest.*
 PeerConnectionInterfaceTest.*
 RTCStatsIntegrationTest.*