Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 06:15:48 +02:00
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions
--- a/pc/peerconnectionendtoend_unittest.cc
+++ b/pc/peerconnectionendtoend_unittest.cc
@ -0,0 +1,549 @@
+/*
+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include <memory>
+
+#include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h"
+#include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "webrtc/rtc_base/gunit.h"
+#include "webrtc/rtc_base/logging.h"
+#include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/ssladapter.h"
+#include "webrtc/rtc_base/sslstreamadapter.h"
+#include "webrtc/rtc_base/stringencode.h"
+#include "webrtc/rtc_base/stringutils.h"
+#include "webrtc/rtc_base/thread.h"
+#ifdef WEBRTC_ANDROID
+#include "webrtc/pc/test/androidtestinitializer.h"
+#endif
+#include "webrtc/pc/test/peerconnectiontestwrapper.h"
+// Notice that mockpeerconnectionobservers.h must be included after the above!
+#include "webrtc/pc/test/mockpeerconnectionobservers.h"
+#include "webrtc/test/mock_audio_decoder.h"
+#include "webrtc/test/mock_audio_decoder_factory.h"
+
+using testing::AtLeast;
+using testing::Invoke;
+using testing::StrictMock;
+using testing::_;
+
+using webrtc::DataChannelInterface;
+using webrtc::FakeConstraints;
+using webrtc::MediaConstraintsInterface;
+using webrtc::MediaStreamInterface;
+using webrtc::PeerConnectionInterface;
+
+namespace {
+
+const int kMaxWait = 10000;
+
+}  // namespace
+
+class PeerConnectionEndToEndTest
+    : public sigslot::has_slots<>,
+      public testing::Test {
+ public:
+  typedef std::vector<rtc::scoped_refptr<DataChannelInterface> >
+      DataChannelList;
+
+  PeerConnectionEndToEndTest() {
+    network_thread_ = rtc::Thread::CreateWithSocketServer();
+    worker_thread_ = rtc::Thread::Create();
+    RTC_CHECK(network_thread_->Start());
+    RTC_CHECK(worker_thread_->Start());
+    caller_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
+        "caller", network_thread_.get(), worker_thread_.get());
+    callee_ = new rtc::RefCountedObject<PeerConnectionTestWrapper>(
+        "callee", network_thread_.get(), worker_thread_.get());
+    webrtc::PeerConnectionInterface::IceServer ice_server;
+    ice_server.uri = "stun:stun.l.google.com:19302";
+    config_.servers.push_back(ice_server);
+
+#ifdef WEBRTC_ANDROID
+    webrtc::InitializeAndroidObjects();
+#endif
+  }
+
+  void CreatePcs(
+      const MediaConstraintsInterface* pc_constraints,
+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) {
+    EXPECT_TRUE(caller_->CreatePc(
+        pc_constraints, config_, audio_encoder_factory, audio_decoder_factory));
+    EXPECT_TRUE(callee_->CreatePc(
+        pc_constraints, config_, audio_encoder_factory, audio_decoder_factory));
+    PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
+
+    caller_->SignalOnDataChannel.connect(
+        this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel);
+    callee_->SignalOnDataChannel.connect(
+        this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel);
+  }
+
+  void GetAndAddUserMedia() {
+    FakeConstraints audio_constraints;
+    FakeConstraints video_constraints;
+    GetAndAddUserMedia(true, audio_constraints, true, video_constraints);
+  }
+
+  void GetAndAddUserMedia(bool audio, FakeConstraints audio_constraints,
+                          bool video, FakeConstraints video_constraints) {
+    caller_->GetAndAddUserMedia(audio, audio_constraints,
+                                video, video_constraints);
+    callee_->GetAndAddUserMedia(audio, audio_constraints,
+                                video, video_constraints);
+  }
+
+  void Negotiate() {
+    caller_->CreateOffer(NULL);
+  }
+
+  void WaitForCallEstablished() {
+    caller_->WaitForCallEstablished();
+    callee_->WaitForCallEstablished();
+  }
+
+  void WaitForConnection() {
+    caller_->WaitForConnection();
+    callee_->WaitForConnection();
+  }
+
+  void OnCallerAddedDataChanel(DataChannelInterface* dc) {
+    caller_signaled_data_channels_.push_back(dc);
+  }
+
+  void OnCalleeAddedDataChannel(DataChannelInterface* dc) {
+    callee_signaled_data_channels_.push_back(dc);
+  }
+
+  // Tests that |dc1| and |dc2| can send to and receive from each other.
+  void TestDataChannelSendAndReceive(
+      DataChannelInterface* dc1, DataChannelInterface* dc2) {
+    std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer(
+        new webrtc::MockDataChannelObserver(dc1));
+
+    std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer(
+        new webrtc::MockDataChannelObserver(dc2));
+
+    static const std::string kDummyData = "abcdefg";
+    webrtc::DataBuffer buffer(kDummyData);
+    EXPECT_TRUE(dc1->Send(buffer));
+    EXPECT_EQ_WAIT(kDummyData, dc2_observer->last_message(), kMaxWait);
+
+    EXPECT_TRUE(dc2->Send(buffer));
+    EXPECT_EQ_WAIT(kDummyData, dc1_observer->last_message(), kMaxWait);
+
+    EXPECT_EQ(1U, dc1_observer->received_message_count());
+    EXPECT_EQ(1U, dc2_observer->received_message_count());
+  }
+
+  void WaitForDataChannelsToOpen(DataChannelInterface* local_dc,
+                                 const DataChannelList& remote_dc_list,
+                                 size_t remote_dc_index) {
+    EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait);
+
+    EXPECT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait);
+    EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
+                   remote_dc_list[remote_dc_index]->state(),
+                   kMaxWait);
+    EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id());
+  }
+
+  void CloseDataChannels(DataChannelInterface* local_dc,
+                         const DataChannelList& remote_dc_list,
+                         size_t remote_dc_index) {
+    local_dc->Close();
+    EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait);
+    EXPECT_EQ_WAIT(DataChannelInterface::kClosed,
+                   remote_dc_list[remote_dc_index]->state(),
+                   kMaxWait);
+  }
+
+ protected:
+  std::unique_ptr<rtc::Thread> network_thread_;
+  std::unique_ptr<rtc::Thread> worker_thread_;
+  rtc::scoped_refptr<PeerConnectionTestWrapper> caller_;
+  rtc::scoped_refptr<PeerConnectionTestWrapper> callee_;
+  DataChannelList caller_signaled_data_channels_;
+  DataChannelList callee_signaled_data_channels_;
+  webrtc::PeerConnectionInterface::RTCConfiguration config_;
+};
+
+std::unique_ptr<webrtc::AudioDecoder> CreateForwardingMockDecoder(
+    std::unique_ptr<webrtc::AudioDecoder> real_decoder) {
+  class ForwardingMockDecoder : public StrictMock<webrtc::MockAudioDecoder> {
+   public:
+    ForwardingMockDecoder(std::unique_ptr<AudioDecoder> decoder)
+        : decoder_(std::move(decoder)) {}
+
+   private:
+    std::unique_ptr<AudioDecoder> decoder_;
+  };
+
+  const auto dec = real_decoder.get();  // For lambda capturing.
+  auto mock_decoder =
+      rtc::MakeUnique<ForwardingMockDecoder>(std::move(real_decoder));
+  EXPECT_CALL(*mock_decoder, Channels())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke([dec] { return dec->Channels(); }));
+  EXPECT_CALL(*mock_decoder, DecodeInternal(_, _, _, _, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(
+          Invoke([dec](const uint8_t* encoded, size_t encoded_len,
+                       int sample_rate_hz, int16_t* decoded,
+                       webrtc::AudioDecoder::SpeechType* speech_type) {
+            return dec->Decode(encoded, encoded_len, sample_rate_hz,
+                               std::numeric_limits<size_t>::max(), decoded,
+                               speech_type);
+          }));
+  EXPECT_CALL(*mock_decoder, Die());
+  EXPECT_CALL(*mock_decoder, HasDecodePlc()).WillRepeatedly(Invoke([dec] {
+    return dec->HasDecodePlc();
+  }));
+  EXPECT_CALL(*mock_decoder, IncomingPacket(_, _, _, _, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke([dec](const uint8_t* payload, size_t payload_len,
+                                   uint16_t rtp_sequence_number,
+                                   uint32_t rtp_timestamp,
+                                   uint32_t arrival_timestamp) {
+        return dec->IncomingPacket(payload, payload_len, rtp_sequence_number,
+                                   rtp_timestamp, arrival_timestamp);
+      }));
+  EXPECT_CALL(*mock_decoder, PacketDuration(_, _))
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke([dec](const uint8_t* encoded, size_t encoded_len) {
+        return dec->PacketDuration(encoded, encoded_len);
+      }));
+  EXPECT_CALL(*mock_decoder, SampleRateHz())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke([dec] { return dec->SampleRateHz(); }));
+
+  return std::move(mock_decoder);
+}
+
+rtc::scoped_refptr<webrtc::AudioDecoderFactory>
+CreateForwardingMockDecoderFactory(
+    webrtc::AudioDecoderFactory* real_decoder_factory) {
+  rtc::scoped_refptr<webrtc::MockAudioDecoderFactory> mock_decoder_factory =
+      new rtc::RefCountedObject<StrictMock<webrtc::MockAudioDecoderFactory>>;
+  EXPECT_CALL(*mock_decoder_factory, GetSupportedDecoders())
+      .Times(AtLeast(1))
+      .WillRepeatedly(Invoke([real_decoder_factory] {
+        return real_decoder_factory->GetSupportedDecoders();
+      }));
+  EXPECT_CALL(*mock_decoder_factory, IsSupportedDecoder(_))
+      .Times(AtLeast(1))
+      .WillRepeatedly(
+          Invoke([real_decoder_factory](const webrtc::SdpAudioFormat& format) {
+            return real_decoder_factory->IsSupportedDecoder(format);
+          }));
+  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _))
+      .Times(AtLeast(2))
+      .WillRepeatedly(
+          Invoke([real_decoder_factory](
+                     const webrtc::SdpAudioFormat& format,
+                     std::unique_ptr<webrtc::AudioDecoder>* return_value) {
+            auto real_decoder = real_decoder_factory->MakeAudioDecoder(format);
+            *return_value =
+                real_decoder
+                    ? CreateForwardingMockDecoder(std::move(real_decoder))
+                    : nullptr;
+          }));
+  return mock_decoder_factory;
+}
+
+// Disabled for TSan v2, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=4719 for details.
+// Disabled for Mac, see
+// https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details.
+#if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
+TEST_F(PeerConnectionEndToEndTest, Call) {
+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =
+      webrtc::CreateBuiltinAudioDecoderFactory();
+  CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+            CreateForwardingMockDecoderFactory(real_decoder_factory.get()));
+  GetAndAddUserMedia();
+  Negotiate();
+  WaitForCallEstablished();
+}
+#endif // if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
+
+#if !defined(ADDRESS_SANITIZER)
+TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
+  FakeConstraints pc_constraints;
+  pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
+                              false);
+  CreatePcs(&pc_constraints, webrtc::CreateBuiltinAudioEncoderFactory(),
+            webrtc::CreateBuiltinAudioDecoderFactory());
+  GetAndAddUserMedia();
+  Negotiate();
+  WaitForCallEstablished();
+}
+#endif  // !defined(ADDRESS_SANITIZER)
+
+#ifdef HAVE_SCTP
+// Verifies that a DataChannel created before the negotiation can transition to
+// "OPEN" and transfer data.
+TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
+  CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
+      caller_->CreateDataChannel("data", init));
+  rtc::scoped_refptr<DataChannelInterface> callee_dc(
+      callee_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+
+  WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
+  WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
+
+  TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[0]);
+  TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
+
+  CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
+  CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
+}
+
+// Verifies that a DataChannel created after the negotiation can transition to
+// "OPEN" and transfer data.
+TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
+  CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+
+  // This DataChannel is for creating the data content in the negotiation.
+  rtc::scoped_refptr<DataChannelInterface> dummy(
+      caller_->CreateDataChannel("data", init));
+  Negotiate();
+  WaitForConnection();
+
+  // Wait for the data channel created pre-negotiation to be opened.
+  WaitForDataChannelsToOpen(dummy, callee_signaled_data_channels_, 0);
+
+  // Create new DataChannels after the negotiation and verify their states.
+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
+      caller_->CreateDataChannel("hello", init));
+  rtc::scoped_refptr<DataChannelInterface> callee_dc(
+      callee_->CreateDataChannel("hello", init));
+
+  WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
+  WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
+
+  TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
+  TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0]);
+
+  CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
+  CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
+}
+
+// Verifies that DataChannel IDs are even/odd based on the DTLS roles.
+TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
+  CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
+      caller_->CreateDataChannel("data", init));
+  rtc::scoped_refptr<DataChannelInterface> callee_dc_1(
+      callee_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+
+  EXPECT_EQ(1U, caller_dc_1->id() % 2);
+  EXPECT_EQ(0U, callee_dc_1->id() % 2);
+
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
+      caller_->CreateDataChannel("data", init));
+  rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
+      callee_->CreateDataChannel("data", init));
+
+  EXPECT_EQ(1U, caller_dc_2->id() % 2);
+  EXPECT_EQ(0U, callee_dc_2->id() % 2);
+}
+
+// Verifies that the message is received by the right remote DataChannel when
+// there are multiple DataChannels.
+TEST_F(PeerConnectionEndToEndTest,
+       MessageTransferBetweenTwoPairsOfDataChannels) {
+  CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
+      caller_->CreateDataChannel("data", init));
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
+      caller_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+  WaitForDataChannelsToOpen(caller_dc_1, callee_signaled_data_channels_, 0);
+  WaitForDataChannelsToOpen(caller_dc_2, callee_signaled_data_channels_, 1);
+
+  std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
+      new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[0]));
+
+  std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
+      new webrtc::MockDataChannelObserver(callee_signaled_data_channels_[1]));
+
+  const std::string message_1 = "hello 1";
+  const std::string message_2 = "hello 2";
+
+  caller_dc_1->Send(webrtc::DataBuffer(message_1));
+  EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
+
+  caller_dc_2->Send(webrtc::DataBuffer(message_2));
+  EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
+
+  EXPECT_EQ(1U, dc_1_observer->received_message_count());
+  EXPECT_EQ(1U, dc_2_observer->received_message_count());
+}
+#endif  // HAVE_SCTP
+
+#ifdef HAVE_QUIC
+// Test that QUIC data channels can be used and that messages go to the correct
+// remote data channel when both peers want to use QUIC. It is assumed that the
+// application has externally negotiated the data channel parameters.
+TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) {
+  config_.enable_quic = true;
+  CreatePcs();
+
+  webrtc::DataChannelInit init_1;
+  init_1.id = 0;
+  init_1.ordered = false;
+  init_1.reliable = true;
+
+  webrtc::DataChannelInit init_2;
+  init_2.id = 1;
+  init_2.ordered = false;
+  init_2.reliable = true;
+
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_1(
+      caller_->CreateDataChannel("data", init_1));
+  ASSERT_NE(nullptr, caller_dc_1);
+  rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
+      caller_->CreateDataChannel("data", init_2));
+  ASSERT_NE(nullptr, caller_dc_2);
+  rtc::scoped_refptr<DataChannelInterface> callee_dc_1(
+      callee_->CreateDataChannel("data", init_1));
+  ASSERT_NE(nullptr, callee_dc_1);
+  rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
+      callee_->CreateDataChannel("data", init_2));
+  ASSERT_NE(nullptr, callee_dc_2);
+
+  Negotiate();
+  WaitForConnection();
+  EXPECT_TRUE_WAIT(caller_dc_1->state() == webrtc::DataChannelInterface::kOpen,
+                   kMaxWait);
+  EXPECT_TRUE_WAIT(callee_dc_1->state() == webrtc::DataChannelInterface::kOpen,
+                   kMaxWait);
+  EXPECT_TRUE_WAIT(caller_dc_2->state() == webrtc::DataChannelInterface::kOpen,
+                   kMaxWait);
+  EXPECT_TRUE_WAIT(callee_dc_2->state() == webrtc::DataChannelInterface::kOpen,
+                   kMaxWait);
+
+  std::unique_ptr<webrtc::MockDataChannelObserver> dc_1_observer(
+      new webrtc::MockDataChannelObserver(callee_dc_1.get()));
+
+  std::unique_ptr<webrtc::MockDataChannelObserver> dc_2_observer(
+      new webrtc::MockDataChannelObserver(callee_dc_2.get()));
+
+  const std::string message_1 = "hello 1";
+  const std::string message_2 = "hello 2";
+
+  // Send data from caller to callee.
+  caller_dc_1->Send(webrtc::DataBuffer(message_1));
+  EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
+
+  caller_dc_2->Send(webrtc::DataBuffer(message_2));
+  EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
+
+  EXPECT_EQ(1U, dc_1_observer->received_message_count());
+  EXPECT_EQ(1U, dc_2_observer->received_message_count());
+
+  // Send data from callee to caller.
+  dc_1_observer.reset(new webrtc::MockDataChannelObserver(caller_dc_1.get()));
+  dc_2_observer.reset(new webrtc::MockDataChannelObserver(caller_dc_2.get()));
+
+  callee_dc_1->Send(webrtc::DataBuffer(message_1));
+  EXPECT_EQ_WAIT(message_1, dc_1_observer->last_message(), kMaxWait);
+
+  callee_dc_2->Send(webrtc::DataBuffer(message_2));
+  EXPECT_EQ_WAIT(message_2, dc_2_observer->last_message(), kMaxWait);
+
+  EXPECT_EQ(1U, dc_1_observer->received_message_count());
+  EXPECT_EQ(1U, dc_2_observer->received_message_count());
+}
+#endif  // HAVE_QUIC
+
+#ifdef HAVE_SCTP
+// Verifies that a DataChannel added from an OPEN message functions after
+// a channel has been previously closed (webrtc issue 3778).
+// This previously failed because the new channel re-uses the ID of the closed
+// channel, and the closed channel was incorrectly still assigned to the id.
+// TODO(deadbeef): This is disabled because there's currently a race condition
+// 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,
+       DISABLED_DataChannelFromOpenWorksAfterClose) {
+  CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
+      caller_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+
+  WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
+  CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
+
+  // Create a new channel and ensure it works after closing the previous one.
+  caller_dc = caller_->CreateDataChannel("data2", init);
+
+  WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
+  TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
+
+  CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
+}
+
+// This tests that if a data channel is closed remotely while not referenced
+// 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) {
+  CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
+
+  webrtc::DataChannelInit init;
+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
+      caller_->CreateDataChannel("data", init));
+
+  Negotiate();
+  WaitForConnection();
+
+  WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
+  // This removes the reference to the remote data channel that we hold.
+  callee_signaled_data_channels_.clear();
+  caller_dc->Close();
+  EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
+
+  // Wait for a bit longer so the remote data channel will receive the
+  // close message and be destroyed.
+  rtc::Thread::Current()->ProcessMessages(100);
+}
+#endif  // HAVE_SCTP