Split peer_connection_integrationtest.cc into pieces

This creates two integration tests: One for datachannel, the other
for every test that is not datachannel.

It separates out the common framework to a new file in pc/test.
Also applies some fixes to IWYU.

Bug: None
Change-Id: I919def1c360ffce205c20bec2d864aad9b179c3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/207060
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33244}

pc/BUILD.gn

@@ -1044,6 +1044,7 @@ if (rtc_include_tests && !build_with_chromium) {
   rtc_test("peerconnection_unittests") {
     testonly = true
     sources = [
+      "data_channel_integrationtest.cc",
       "data_channel_unittest.cc",
       "dtmf_sender_unittest.cc",
       "ice_server_parsing_unittest.cc",
@@ -1081,6 +1082,8 @@ if (rtc_include_tests && !build_with_chromium) {
       "sdp_serializer_unittest.cc",
       "stats_collector_unittest.cc",
       "test/fake_audio_capture_module_unittest.cc",
+      "test/integration_test_helpers.cc",
+      "test/integration_test_helpers.h",
       "test/test_sdp_strings.h",
       "track_media_info_map_unittest.cc",
       "video_rtp_track_source_unittest.cc",
@@ -1121,7 +1124,9 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api:libjingle_peerconnection_api",
       "../api:media_stream_interface",
       "../api:mock_rtp",
+      "../api:packet_socket_factory",
       "../api:rtc_error",
+      "../api:rtp_transceiver_direction",
       "../api:scoped_refptr",
       "../api/audio:audio_mixer_api",
       "../api/crypto:frame_decryptor_interface",
@@ -1129,11 +1134,14 @@ if (rtc_include_tests && !build_with_chromium) {
       "../api/crypto:options",
       "../api/rtc_event_log",
       "../api/rtc_event_log:rtc_event_log_factory",
+      "../api/task_queue",
       "../api/task_queue:default_task_queue_factory",
       "../api/transport:field_trial_based_config",
+      "../api/transport:webrtc_key_value_config",
       "../api/transport/rtp:rtp_source",
       "../api/units:time_delta",
       "../api/video:builtin_video_bitrate_allocator_factory",
+      "../api/video:video_rtp_headers",
       "../call/adaptation:resource_adaptation_test_utilities",
       "../logging:fake_rtc_event_log",
       "../media:rtc_media_config",
@@ -1144,6 +1152,7 @@ if (rtc_include_tests && !build_with_chromium) {
       "../modules/rtp_rtcp:rtp_rtcp_format",
       "../p2p:fake_ice_transport",
       "../p2p:fake_port_allocator",
+      "../p2p:p2p_server_utils",
       "../rtc_base:checks",
       "../rtc_base:gunit_helpers",
       "../rtc_base:ip_address",
@@ -1164,6 +1173,8 @@ if (rtc_include_tests && !build_with_chromium) {
       "//third_party/abseil-cpp/absl/memory",
       "//third_party/abseil-cpp/absl/strings",
       "//third_party/abseil-cpp/absl/types:optional",
+      "//third_party/googletest:gmock",
+      "//third_party/googletest:gtest",
     ]
     if (is_android) {
       deps += [ ":android_black_magic" ]

pc/data_channel_integrationtest.cc

@@ -0,0 +1,704 @@
+/*
+ *  Copyright 2012 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 <stdint.h>
+
+#include <algorithm>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/data_channel_interface.h"
+#include "api/dtmf_sender_interface.h"
+#include "api/peer_connection_interface.h"
+#include "api/scoped_refptr.h"
+#include "api/units/time_delta.h"
+#include "pc/test/integration_test_helpers.h"
+#include "pc/test/mock_peer_connection_observers.h"
+#include "rtc_base/fake_clock.h"
+#include "rtc_base/gunit.h"
+#include "rtc_base/ref_counted_object.h"
+#include "rtc_base/virtual_socket_server.h"
+
+namespace webrtc {
+
+namespace {
+
+class DataChannelIntegrationTest
+    : public PeerConnectionIntegrationBaseTest,
+      public ::testing::WithParamInterface<SdpSemantics> {
+ protected:
+  DataChannelIntegrationTest()
+      : PeerConnectionIntegrationBaseTest(GetParam()) {}
+};
+
+// Fake clock must be set before threads are started to prevent race on
+// Set/GetClockForTesting().
+// To achieve that, multiple inheritance is used as a mixin pattern
+// where order of construction is finely controlled.
+// This also ensures peerconnection is closed before switching back to non-fake
+// clock, avoiding other races and DCHECK failures such as in rtp_sender.cc.
+class FakeClockForTest : public rtc::ScopedFakeClock {
+ protected:
+  FakeClockForTest() {
+    // Some things use a time of "0" as a special value, so we need to start out
+    // the fake clock at a nonzero time.
+    // TODO(deadbeef): Fix this.
+    AdvanceTime(webrtc::TimeDelta::Seconds(1));
+  }
+
+  // Explicit handle.
+  ScopedFakeClock& FakeClock() { return *this; }
+};
+
+// Ensure FakeClockForTest is constructed first (see class for rationale).
+class DataChannelIntegrationTestWithFakeClock
+    : public FakeClockForTest,
+      public DataChannelIntegrationTest {};
+
+class DataChannelIntegrationTestPlanB
+    : public PeerConnectionIntegrationBaseTest {
+ protected:
+  DataChannelIntegrationTestPlanB()
+      : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB) {}
+};
+
+class DataChannelIntegrationTestUnifiedPlan
+    : public PeerConnectionIntegrationBaseTest {
+ protected:
+  DataChannelIntegrationTestUnifiedPlan()
+      : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
+};
+
+class DummyDtmfObserver : public DtmfSenderObserverInterface {
+ public:
+  DummyDtmfObserver() : completed_(false) {}
+
+  // Implements DtmfSenderObserverInterface.
+  void OnToneChange(const std::string& tone) override {
+    tones_.push_back(tone);
+    if (tone.empty()) {
+      completed_ = true;
+    }
+  }
+
+  const std::vector<std::string>& tones() const { return tones_; }
+  bool completed() const { return completed_; }
+
+ private:
+  bool completed_;
+  std::vector<std::string> tones_;
+};
+
+#ifdef WEBRTC_HAVE_SCTP
+
+// This test causes a PeerConnection to enter Disconnected state, and
+// sends data on a DataChannel while disconnected.
+// The data should be surfaced when the connection reestablishes.
+TEST_P(DataChannelIntegrationTest, DataChannelWhileDisconnected) {
+  CreatePeerConnectionWrappers();
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  std::string data1 = "hello first";
+  caller()->data_channel()->Send(DataBuffer(data1));
+  EXPECT_EQ_WAIT(data1, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  // Cause a network outage
+  virtual_socket_server()->set_drop_probability(1.0);
+  EXPECT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
+                 caller()->standardized_ice_connection_state(),
+                 kDefaultTimeout);
+  std::string data2 = "hello second";
+  caller()->data_channel()->Send(DataBuffer(data2));
+  // Remove the network outage. The connection should reestablish.
+  virtual_socket_server()->set_drop_probability(0.0);
+  EXPECT_EQ_WAIT(data2, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+// This test causes a PeerConnection to enter Disconnected state,
+// sends data on a DataChannel while disconnected, and then triggers
+// an ICE restart.
+// The data should be surfaced when the connection reestablishes.
+TEST_P(DataChannelIntegrationTest, DataChannelWhileDisconnectedIceRestart) {
+  CreatePeerConnectionWrappers();
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  std::string data1 = "hello first";
+  caller()->data_channel()->Send(DataBuffer(data1));
+  EXPECT_EQ_WAIT(data1, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  // Cause a network outage
+  virtual_socket_server()->set_drop_probability(1.0);
+  ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
+                 caller()->standardized_ice_connection_state(),
+                 kDefaultTimeout);
+  std::string data2 = "hello second";
+  caller()->data_channel()->Send(DataBuffer(data2));
+
+  // Trigger an ICE restart. The signaling channel is not affected by
+  // the network outage.
+  caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Remove the network outage. The connection should reestablish.
+  virtual_socket_server()->set_drop_probability(0.0);
+  EXPECT_EQ_WAIT(data2, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+#endif  // WEBRTC_HAVE_SCTP
+
+// This test sets up a call between two parties with audio, video and an RTP
+// data channel.
+TEST_P(DataChannelIntegrationTest, EndToEndCallWithRtpDataChannel) {
+  PeerConnectionInterface::RTCConfiguration rtc_config;
+  rtc_config.enable_rtp_data_channel = true;
+  rtc_config.enable_dtls_srtp = false;
+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
+  ConnectFakeSignaling();
+  // Expect that data channel created on caller side will show up for callee as
+  // well.
+  caller()->CreateDataChannel();
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Ensure the existence of the RTP data channel didn't impede audio/video.
+  MediaExpectations media_expectations;
+  media_expectations.ExpectBidirectionalAudioAndVideo();
+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_NE(nullptr, callee()->data_channel());
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Ensure data can be sent in both directions.
+  std::string data = "hello world";
+  SendRtpDataWithRetries(caller()->data_channel(), data, 5);
+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  SendRtpDataWithRetries(callee()->data_channel(), data, 5);
+  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+TEST_P(DataChannelIntegrationTest, RtpDataChannelWorksAfterRollback) {
+  PeerConnectionInterface::RTCConfiguration rtc_config;
+  rtc_config.enable_rtp_data_channel = true;
+  rtc_config.enable_dtls_srtp = false;
+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
+  ConnectFakeSignaling();
+  auto data_channel = caller()->pc()->CreateDataChannel("label_1", nullptr);
+  ASSERT_TRUE(data_channel.get() != nullptr);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+  caller()->CreateDataChannel("label_2", nullptr);
+  rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
+      new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
+  caller()->pc()->SetLocalDescription(observer,
+                                      caller()->CreateOfferAndWait().release());
+  EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
+  caller()->Rollback();
+
+  std::string data = "hello world";
+  SendRtpDataWithRetries(data_channel, data, 5);
+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+// Ensure that an RTP data channel is signaled as closed for the caller when
+// the callee rejects it in a subsequent offer.
+TEST_P(DataChannelIntegrationTest, RtpDataChannelSignaledClosedInCalleeOffer) {
+  // Same procedure as above test.
+  PeerConnectionInterface::RTCConfiguration rtc_config;
+  rtc_config.enable_rtp_data_channel = true;
+  rtc_config.enable_dtls_srtp = false;
+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_NE(nullptr, callee()->data_channel());
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Close the data channel on the callee, and do an updated offer/answer.
+  callee()->data_channel()->Close();
+  callee()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  EXPECT_FALSE(caller()->data_observer()->IsOpen());
+  EXPECT_FALSE(callee()->data_observer()->IsOpen());
+}
+
+#if !defined(THREAD_SANITIZER)
+// This test provokes TSAN errors. See bugs.webrtc.org/11282
+
+// Tests that data is buffered in an RTP data channel until an observer is
+// registered for it.
+//
+// NOTE: RTP data channels can receive data before the underlying
+// transport has detected that a channel is writable and thus data can be
+// received before the data channel state changes to open. That is hard to test
+// but the same buffering is expected to be used in that case.
+//
+// Use fake clock and simulated network delay so that we predictably can wait
+// until an SCTP message has been delivered without "sleep()"ing.
+TEST_P(DataChannelIntegrationTestWithFakeClock,
+       DataBufferedUntilRtpDataChannelObserverRegistered) {
+  virtual_socket_server()->set_delay_mean(5);  // 5 ms per hop.
+  virtual_socket_server()->UpdateDelayDistribution();
+
+  PeerConnectionInterface::RTCConfiguration rtc_config;
+  rtc_config.enable_rtp_data_channel = true;
+  rtc_config.enable_dtls_srtp = false;
+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE(caller()->data_channel() != nullptr);
+  ASSERT_TRUE_SIMULATED_WAIT(callee()->data_channel() != nullptr,
+                             kDefaultTimeout, FakeClock());
+  ASSERT_TRUE_SIMULATED_WAIT(caller()->data_observer()->IsOpen(),
+                             kDefaultTimeout, FakeClock());
+  ASSERT_EQ_SIMULATED_WAIT(DataChannelInterface::kOpen,
+                           callee()->data_channel()->state(), kDefaultTimeout,
+                           FakeClock());
+
+  // Unregister the observer which is normally automatically registered.
+  callee()->data_channel()->UnregisterObserver();
+  // Send data and advance fake clock until it should have been received.
+  std::string data = "hello world";
+  caller()->data_channel()->Send(DataBuffer(data));
+  SIMULATED_WAIT(false, 50, FakeClock());
+
+  // Attach data channel and expect data to be received immediately. Note that
+  // EXPECT_EQ_WAIT is used, such that the simulated clock is not advanced any
+  // further, but data can be received even if the callback is asynchronous.
+  MockDataChannelObserver new_observer(callee()->data_channel());
+  EXPECT_EQ_SIMULATED_WAIT(data, new_observer.last_message(), kDefaultTimeout,
+                           FakeClock());
+}
+
+#endif  // !defined(THREAD_SANITIZER)
+
+// This test sets up a call between two parties with audio, video and but only
+// the caller client supports RTP data channels.
+TEST_P(DataChannelIntegrationTest, RtpDataChannelsRejectedByCallee) {
+  PeerConnectionInterface::RTCConfiguration rtc_config_1;
+  rtc_config_1.enable_rtp_data_channel = true;
+  // Must disable DTLS to make negotiation succeed.
+  rtc_config_1.enable_dtls_srtp = false;
+  PeerConnectionInterface::RTCConfiguration rtc_config_2;
+  rtc_config_2.enable_dtls_srtp = false;
+  rtc_config_2.enable_dtls_srtp = false;
+  ASSERT_TRUE(
+      CreatePeerConnectionWrappersWithConfig(rtc_config_1, rtc_config_2));
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  ASSERT_TRUE(caller()->data_channel() != nullptr);
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // The caller should still have a data channel, but it should be closed, and
+  // one should ever have been created for the callee.
+  EXPECT_TRUE(caller()->data_channel() != nullptr);
+  EXPECT_FALSE(caller()->data_observer()->IsOpen());
+  EXPECT_EQ(nullptr, callee()->data_channel());
+}
+
+// This test sets up a call between two parties with audio, and video. When
+// audio and video is setup and flowing, an RTP data channel is negotiated.
+TEST_P(DataChannelIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
+  PeerConnectionInterface::RTCConfiguration rtc_config;
+  rtc_config.enable_rtp_data_channel = true;
+  rtc_config.enable_dtls_srtp = false;
+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(rtc_config, rtc_config));
+  ConnectFakeSignaling();
+  // Do initial offer/answer with audio/video.
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Create data channel and do new offer and answer.
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_NE(nullptr, callee()->data_channel());
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  // Ensure data can be sent in both directions.
+  std::string data = "hello world";
+  SendRtpDataWithRetries(caller()->data_channel(), data, 5);
+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  SendRtpDataWithRetries(callee()->data_channel(), data, 5);
+  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+#ifdef WEBRTC_HAVE_SCTP
+
+// This test sets up a call between two parties with audio, video and an SCTP
+// data channel.
+TEST_P(DataChannelIntegrationTest, EndToEndCallWithSctpDataChannel) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Expect that data channel created on caller side will show up for callee as
+  // well.
+  caller()->CreateDataChannel();
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Ensure the existence of the SCTP data channel didn't impede audio/video.
+  MediaExpectations media_expectations;
+  media_expectations.ExpectBidirectionalAudioAndVideo();
+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
+  // Caller data channel should already exist (it created one). Callee data
+  // channel may not exist yet, since negotiation happens in-band, not in SDP.
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Ensure data can be sent in both directions.
+  std::string data = "hello world";
+  caller()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  callee()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+// Ensure that when the callee closes an SCTP data channel, the closing
+// procedure results in the data channel being closed for the caller as well.
+TEST_P(DataChannelIntegrationTest, CalleeClosesSctpDataChannel) {
+  // Same procedure as above test.
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Close the data channel on the callee side, and wait for it to reach the
+  // "closed" state on both sides.
+  callee()->data_channel()->Close();
+  EXPECT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
+}
+
+TEST_P(DataChannelIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  webrtc::DataChannelInit init;
+  init.id = 53;
+  init.maxRetransmits = 52;
+  caller()->CreateDataChannel("data-channel", &init);
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  // Since "negotiated" is false, the "id" parameter should be ignored.
+  EXPECT_NE(init.id, callee()->data_channel()->id());
+  EXPECT_EQ("data-channel", callee()->data_channel()->label());
+  EXPECT_EQ(init.maxRetransmits, callee()->data_channel()->maxRetransmits());
+  EXPECT_FALSE(callee()->data_channel()->negotiated());
+}
+
+// Test usrsctp's ability to process unordered data stream, where data actually
+// arrives out of order using simulated delays. Previously there have been some
+// bugs in this area.
+TEST_P(DataChannelIntegrationTest, StressTestUnorderedSctpDataChannel) {
+  // Introduce random network delays.
+  // Otherwise it's not a true "unordered" test.
+  virtual_socket_server()->set_delay_mean(20);
+  virtual_socket_server()->set_delay_stddev(5);
+  virtual_socket_server()->UpdateDelayDistribution();
+  // Normal procedure, but with unordered data channel config.
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  webrtc::DataChannelInit init;
+  init.ordered = false;
+  caller()->CreateDataChannel(&init);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  static constexpr int kNumMessages = 100;
+  // Deliberately chosen to be larger than the MTU so messages get fragmented.
+  static constexpr size_t kMaxMessageSize = 4096;
+  // Create and send random messages.
+  std::vector<std::string> sent_messages;
+  for (int i = 0; i < kNumMessages; ++i) {
+    size_t length =
+        (rand() % kMaxMessageSize) + 1;  // NOLINT (rand_r instead of rand)
+    std::string message;
+    ASSERT_TRUE(rtc::CreateRandomString(length, &message));
+    caller()->data_channel()->Send(DataBuffer(message));
+    callee()->data_channel()->Send(DataBuffer(message));
+    sent_messages.push_back(message);
+  }
+
+  // Wait for all messages to be received.
+  EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
+                 caller()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+  EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
+                 callee()->data_observer()->received_message_count(),
+                 kDefaultTimeout);
+
+  // Sort and compare to make sure none of the messages were corrupted.
+  std::vector<std::string> caller_received_messages =
+      caller()->data_observer()->messages();
+  std::vector<std::string> callee_received_messages =
+      callee()->data_observer()->messages();
+  absl::c_sort(sent_messages);
+  absl::c_sort(caller_received_messages);
+  absl::c_sort(callee_received_messages);
+  EXPECT_EQ(sent_messages, caller_received_messages);
+  EXPECT_EQ(sent_messages, callee_received_messages);
+}
+
+// This test sets up a call between two parties with audio, and video. When
+// audio and video are setup and flowing, an SCTP data channel is negotiated.
+TEST_P(DataChannelIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Do initial offer/answer with audio/video.
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Create data channel and do new offer and answer.
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Caller data channel should already exist (it created one). Callee data
+  // channel may not exist yet, since negotiation happens in-band, not in SDP.
+  ASSERT_NE(nullptr, caller()->data_channel());
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  // Ensure data can be sent in both directions.
+  std::string data = "hello world";
+  caller()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  callee()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+// Set up a connection initially just using SCTP data channels, later upgrading
+// to audio/video, ensuring frames are received end-to-end. Effectively the
+// inverse of the test above.
+// This was broken in M57; see https://crbug.com/711243
+TEST_P(DataChannelIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  // Do initial offer/answer with just data channel.
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait until data can be sent over the data channel.
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  ASSERT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Do subsequent offer/answer with two-way audio and video. Audio and video
+  // should end up bundled on the DTLS/ICE transport already used for data.
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  MediaExpectations media_expectations;
+  media_expectations.ExpectBidirectionalAudioAndVideo();
+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
+}
+
+static void MakeSpecCompliantSctpOffer(cricket::SessionDescription* desc) {
+  cricket::SctpDataContentDescription* dcd_offer =
+      GetFirstSctpDataContentDescription(desc);
+  // See https://crbug.com/webrtc/11211 - this function is a no-op
+  ASSERT_TRUE(dcd_offer);
+  dcd_offer->set_use_sctpmap(false);
+  dcd_offer->set_protocol("UDP/DTLS/SCTP");
+}
+
+// Test that the data channel works when a spec-compliant SCTP m= section is
+// offered (using "a=sctp-port" instead of "a=sctpmap", and using
+// "UDP/DTLS/SCTP" as the protocol).
+TEST_P(DataChannelIntegrationTest,
+       DataChannelWorksWhenSpecCompliantSctpOfferReceived) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->SetGeneratedSdpMunger(MakeSpecCompliantSctpOffer);
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
+  EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
+  EXPECT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+
+  // Ensure data can be sent in both directions.
+  std::string data = "hello world";
+  caller()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, callee()->data_observer()->last_message(),
+                 kDefaultTimeout);
+  callee()->data_channel()->Send(DataBuffer(data));
+  EXPECT_EQ_WAIT(data, caller()->data_observer()->last_message(),
+                 kDefaultTimeout);
+}
+
+#endif  // WEBRTC_HAVE_SCTP
+
+// Test that after closing PeerConnections, they stop sending any packets (ICE,
+// DTLS, RTP...).
+TEST_P(DataChannelIntegrationTest, ClosingConnectionStopsPacketFlow) {
+  // Set up audio/video/data, wait for some frames to be received.
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddAudioVideoTracks();
+#ifdef WEBRTC_HAVE_SCTP
+  caller()->CreateDataChannel();
+#endif
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  MediaExpectations media_expectations;
+  media_expectations.CalleeExpectsSomeAudioAndVideo();
+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
+  // Close PeerConnections.
+  ClosePeerConnections();
+  // Pump messages for a second, and ensure no new packets end up sent.
+  uint32_t sent_packets_a = virtual_socket_server()->sent_packets();
+  WAIT(false, 1000);
+  uint32_t sent_packets_b = virtual_socket_server()->sent_packets();
+  EXPECT_EQ(sent_packets_a, sent_packets_b);
+}
+
+// Test that transport stats are generated by the RTCStatsCollector for a
+// connection that only involves data channels. This is a regression test for
+// crbug.com/826972.
+#ifdef WEBRTC_HAVE_SCTP
+TEST_P(DataChannelIntegrationTest,
+       TransportStatsReportedForDataChannelOnlyConnection) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+
+  auto caller_report = caller()->NewGetStats();
+  EXPECT_EQ(1u, caller_report->GetStatsOfType<RTCTransportStats>().size());
+  auto callee_report = callee()->NewGetStats();
+  EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size());
+}
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan,
+       EndToEndCallWithBundledSctpDataChannel) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->AddAudioVideoTracks();
+  callee()->AddAudioVideoTracks();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
+    ASSERT_EQ_WAIT(SctpTransportState::kConnected,
+                   caller()->pc()->GetSctpTransport()->Information().state(),
+                   kDefaultTimeout);
+  });
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+}
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan,
+       EndToEndCallWithDataChannelOnlyConnects) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  ASSERT_TRUE(caller()->data_observer()->IsOpen());
+}
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan, DataChannelClosesWhenClosed) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  caller()->data_channel()->Close();
+  ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
+}
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan,
+       DataChannelClosesWhenClosedReverse) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  callee()->data_channel()->Close();
+  ASSERT_TRUE_WAIT(!caller()->data_observer()->IsOpen(), kDefaultTimeout);
+}
+
+TEST_F(DataChannelIntegrationTestUnifiedPlan,
+       DataChannelClosesWhenPeerConnectionClosed) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->CreateDataChannel();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer(), kDefaultTimeout);
+  ASSERT_TRUE_WAIT(callee()->data_observer()->IsOpen(), kDefaultTimeout);
+  caller()->pc()->Close();
+  ASSERT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
+}
+
+#endif  // WEBRTC_HAVE_SCTP
+
+}  // namespace
+
+}  // namespace webrtc

pc/test/integration_test_helpers.cc

@@ -0,0 +1,59 @@
+/*
+ *  Copyright 2012 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 "pc/test/integration_test_helpers.h"
+
+namespace webrtc {
+
+PeerConnectionInterface::RTCOfferAnswerOptions IceRestartOfferAnswerOptions() {
+  PeerConnectionInterface::RTCOfferAnswerOptions options;
+  options.ice_restart = true;
+  return options;
+}
+
+void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) {
+  for (ContentInfo& content : desc->contents()) {
+    content.media_description()->mutable_streams().clear();
+  }
+  desc->set_msid_supported(false);
+  desc->set_msid_signaling(0);
+}
+
+void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) {
+  for (ContentInfo& content : desc->contents()) {
+    std::string track_id;
+    std::vector<std::string> stream_ids;
+    if (!content.media_description()->streams().empty()) {
+      const StreamParams& first_stream =
+          content.media_description()->streams()[0];
+      track_id = first_stream.id;
+      stream_ids = first_stream.stream_ids();
+    }
+    content.media_description()->mutable_streams().clear();
+    StreamParams new_stream;
+    new_stream.id = track_id;
+    new_stream.set_stream_ids(stream_ids);
+    content.media_description()->AddStream(new_stream);
+  }
+}
+
+int FindFirstMediaStatsIndexByKind(
+    const std::string& kind,
+    const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
+        media_stats_vec) {
+  for (size_t i = 0; i < media_stats_vec.size(); i++) {
+    if (media_stats_vec[i]->kind.ValueToString() == kind) {
+      return i;
+    }
+  }
+  return -1;
+}
+
+}  // namespace webrtc

tools_webrtc/iwyu/apply-iwyu

@@ -15,28 +15,48 @@ FILE=$1
 # the following variable to "yes". This is a style guide violation.
 REMOVE_CC_INCLUDES=no
 
-if [ ! -f $FILE.h ]; then
+if [ ! -f $FILE ]; then
+  # See if we have the root name of a .cc/.h pair
+  if [ ! -f $FILE.h ]; then
     echo "$FILE.h not found"
     exit 1
-fi
-
-if [ ! -f $FILE.cc ]; then
+  fi
+  FILE_H=$FILE.h
+  if [ ! -f $FILE.cc ]; then
     echo "$FILE.cc not found"
     exit 1
+  fi
+  FILE_CC=$FILE.cc
+else
+  # Exact file, no .h file
+  FILE_CC=$FILE
+  FILE_H=""
 fi
 
-iwyu -Xiwyu --no_fwd_decls -D__X86_64__  -DWEBRTC_POSIX -I . -I third_party/abseil-cpp $FILE.cc |& fix_include || echo "Some files modified"
+iwyu -Xiwyu --no_fwd_decls -D__X86_64__  -DWEBRTC_POSIX -I . \
+  -I third_party/abseil-cpp \
+  -I third_party/googletest/src/googlemock/include \
+  -I third_party/googletest/src/googletest/include \
+  $FILE_CC |& fix_include || echo "Some files modified"
 
 if [ $REMOVE_CC_INCLUDES == "yes" ]; then
-  grep ^#include $FILE.h | grep -v -f - $FILE.cc > $FILE.ccnew
-  grep -v -f tools_webrtc/iwyu/iwyu-filter-list $FILE.ccnew > $FILE.cc
+  if [ -n "$FILE_H" ]; then
+    # Don't include in .cc what's already included in .h
+    grep ^#include $FILE_H | grep -v -f - $FILE_CC > $FILE_CC.new
+  else
+    cp $FILE_CC $FILE_CC.new
+  fi
+  # Don't include stuff on the banlist
+  grep -v -f tools_webrtc/iwyu/iwyu-filter-list $FILE_CC.new > $FILE_CC
   rm $FILE.ccnew
 else
-  grep -v -f tools_webrtc/iwyu/iwyu-filter-list $FILE.cc > $FILE.ccnew
-  mv $FILE.ccnew $FILE.cc
+  grep -v -f tools_webrtc/iwyu/iwyu-filter-list $FILE_CC > $FILE_CC.new
+  mv $FILE_CC.new $FILE_CC
+fi
+if [ -n "$FILE_H" ]; then
+  grep -v -f tools_webrtc/iwyu/iwyu-filter-list $FILE_H > $FILE_H.new
+  mv $FILE_H.new $FILE_H
 fi
-grep -v -f tools_webrtc/iwyu/iwyu-filter-list $FILE.h > $FILE.hnew
-mv $FILE.hnew $FILE.h
 
 echo "Finished. Check diff, compile and git cl format before uploading."