pc/peer_connection_integrationtest.cc

/*
 *  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.
 */

// Integration tests for PeerConnection.
// These tests exercise a full stack over a simulated network.
//
// NOTE: If your test takes a while (guideline: more than 5 seconds),
// do NOT add it here, but instead add it to the file
// slow_peer_connection_integrationtest.cc

#include <stdint.h>

#include <algorithm>
#include <memory>
#include <string>
#include <tuple>
#include <utility>
#include <vector>

#include "absl/algorithm/container.h"
#include "absl/memory/memory.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/async_resolver_factory.h"
#include "api/candidate.h"
#include "api/crypto/crypto_options.h"
#include "api/dtmf_sender_interface.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtc_event_log_output.h"
#include "api/rtp_parameters.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_direction.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/stats/rtc_stats.h"
#include "api/stats/rtc_stats_report.h"
#include "api/stats/rtcstats_objects.h"
#include "api/test/mock_encoder_selector.h"
#include "api/transport/rtp/rtp_source.h"
#include "api/uma_metrics.h"
#include "api/units/time_delta.h"
#include "api/video/video_rotation.h"
#include "logging/rtc_event_log/fake_rtc_event_log.h"
#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
#include "media/base/codec.h"
#include "media/base/media_constants.h"
#include "media/base/stream_params.h"
#include "p2p/base/mock_async_resolver.h"
#include "p2p/base/port.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/port_interface.h"
#include "p2p/base/test_stun_server.h"
#include "p2p/base/test_turn_customizer.h"
#include "p2p/base/test_turn_server.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
#include "pc/media_session.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_factory.h"
#include "pc/session_description.h"
#include "pc/test/fake_periodic_video_source.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/fake_mdns_responder.h"
#include "rtc_base/fake_network.h"
#include "rtc_base/firewall_socket_server.h"
#include "rtc_base/gunit.h"
#include "rtc_base/helpers.h"
#include "rtc_base/logging.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_fingerprint.h"
#include "rtc_base/ssl_identity.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/test_certificate_verifier.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/virtual_socket_server.h"
#include "system_wrappers/include/metrics.h"
#include "test/gmock.h"
#include "test/gtest.h"

namespace webrtc {

namespace {

class PeerConnectionIntegrationTest
    : public PeerConnectionIntegrationBaseTest,
      public ::testing::WithParamInterface<
          std::tuple<SdpSemantics, std::string>> {
 protected:
  PeerConnectionIntegrationTest()
      : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam()),
                                          std::get<1>(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 PeerConnectionIntegrationTestWithFakeClock
    : public FakeClockForTest,
      public PeerConnectionIntegrationTest {};

class PeerConnectionIntegrationTestPlanB
    : public PeerConnectionIntegrationBaseTest {
 protected:
  PeerConnectionIntegrationTestPlanB()
      : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB_DEPRECATED) {}
};

class PeerConnectionIntegrationTestUnifiedPlan
    : public PeerConnectionIntegrationBaseTest {
 protected:
  PeerConnectionIntegrationTestUnifiedPlan()
      : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
};

// Test the OnFirstPacketReceived callback from audio/video RtpReceivers.  This
// includes testing that the callback is invoked if an observer is connected
// after the first packet has already been received.
TEST_P(PeerConnectionIntegrationTest,
       RtpReceiverObserverOnFirstPacketReceived) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  // Start offer/answer exchange and wait for it to complete.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Should be one receiver each for audio/video.
  EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
  EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
  // Wait for all "first packet received" callbacks to be fired.
  EXPECT_TRUE_WAIT(
      absl::c_all_of(caller()->rtp_receiver_observers(),
                     [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                       return o->first_packet_received();
                     }),
      kMaxWaitForFramesMs);
  EXPECT_TRUE_WAIT(
      absl::c_all_of(callee()->rtp_receiver_observers(),
                     [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                       return o->first_packet_received();
                     }),
      kMaxWaitForFramesMs);
  // If new observers are set after the first packet was already received, the
  // callback should still be invoked.
  caller()->ResetRtpReceiverObservers();
  callee()->ResetRtpReceiverObservers();
  EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
  EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
  EXPECT_TRUE(
      absl::c_all_of(caller()->rtp_receiver_observers(),
                     [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                       return o->first_packet_received();
                     }));
  EXPECT_TRUE(
      absl::c_all_of(callee()->rtp_receiver_observers(),
                     [](const std::unique_ptr<MockRtpReceiverObserver>& o) {
                       return o->first_packet_received();
                     }));
}

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_;
};

// Assumes `sender` already has an audio track added and the offer/answer
// exchange is done.
void TestDtmfFromSenderToReceiver(PeerConnectionIntegrationWrapper* sender,
                                  PeerConnectionIntegrationWrapper* receiver) {
  // We should be able to get a DTMF sender from the local sender.
  rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender =
      sender->pc()->GetSenders().at(0)->GetDtmfSender();
  ASSERT_TRUE(dtmf_sender);
  DummyDtmfObserver observer;
  dtmf_sender->RegisterObserver(&observer);

  // Test the DtmfSender object just created.
  EXPECT_TRUE(dtmf_sender->CanInsertDtmf());
  EXPECT_TRUE(dtmf_sender->InsertDtmf("1a", 100, 50));

  EXPECT_TRUE_WAIT(observer.completed(), kDefaultTimeout);
  std::vector<std::string> tones = {"1", "a", ""};
  EXPECT_EQ(tones, observer.tones());
  dtmf_sender->UnregisterObserver();
  // TODO(deadbeef): Verify the tones were actually received end-to-end.
}

// Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each
// direction).
TEST_P(PeerConnectionIntegrationTest, DtmfSenderObserver) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Only need audio for DTMF.
  caller()->AddAudioTrack();
  callee()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // DTLS must finish before the DTMF sender can be used reliably.
  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
  TestDtmfFromSenderToReceiver(caller(), callee());
  TestDtmfFromSenderToReceiver(callee(), caller());
}

// Basic end-to-end test, verifying media can be encoded/transmitted/decoded
// between two connections, using DTLS-SRTP.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  // Do normal offer/answer and wait for some frames to be received in each
  // direction.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

#if defined(WEBRTC_FUCHSIA)
// Uses SDES instead of DTLS for key agreement.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
  PeerConnectionInterface::RTCConfiguration sdes_config;
  sdes_config.enable_dtls_srtp.emplace(false);
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config));
  ConnectFakeSignaling();

  // Do normal offer/answer and wait for some frames to be received in each
  // direction.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
#endif

// Basic end-to-end test specifying the `enable_encrypted_rtp_header_extensions`
// option to offer encrypted versions of all header extensions alongside the
// unencrypted versions.
TEST_P(PeerConnectionIntegrationTest,
       EndToEndCallWithEncryptedRtpHeaderExtensions) {
  CryptoOptions crypto_options;
  crypto_options.srtp.enable_encrypted_rtp_header_extensions = true;
  PeerConnectionInterface::RTCConfiguration config;
  config.crypto_options = crypto_options;
  // Note: This allows offering >14 RTP header extensions.
  config.offer_extmap_allow_mixed = true;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();

  // Do normal offer/answer and wait for some frames to be received in each
  // direction.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// This test sets up a call between two parties with a source resolution of
// 1280x720 and verifies that a 16:9 aspect ratio is received.
TEST_P(PeerConnectionIntegrationTest,
       Send1280By720ResolutionAndReceive16To9AspectRatio) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  // Add video tracks with 16:9 aspect ratio, size 1280 x 720.
  webrtc::FakePeriodicVideoSource::Config config;
  config.width = 1280;
  config.height = 720;
  config.timestamp_offset_ms = rtc::TimeMillis();
  caller()->AddTrack(caller()->CreateLocalVideoTrackWithConfig(config));
  callee()->AddTrack(callee()->CreateLocalVideoTrackWithConfig(config));

  // Do normal offer/answer and wait for at least one frame to be received in
  // each direction.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
                       callee()->min_video_frames_received_per_track() > 0,
                   kMaxWaitForFramesMs);

  // Check rendered aspect ratio.
  EXPECT_EQ(16.0 / 9, caller()->local_rendered_aspect_ratio());
  EXPECT_EQ(16.0 / 9, caller()->rendered_aspect_ratio());
  EXPECT_EQ(16.0 / 9, callee()->local_rendered_aspect_ratio());
  EXPECT_EQ(16.0 / 9, callee()->rendered_aspect_ratio());
}

// This test sets up an one-way call, with media only from caller to
// callee.
TEST_P(PeerConnectionIntegrationTest, OneWayMediaCall) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudioAndVideo();
  media_expectations.CallerExpectsNoAudio();
  media_expectations.CallerExpectsNoVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// Tests that send only works without the caller having a decoder factory and
// the callee having an encoder factory.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSendOnlyVideo) {
  ASSERT_TRUE(
      CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/true));
  ConnectFakeSignaling();
  // Add one-directional video, from caller to callee.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
      caller()->CreateLocalVideoTrack();
  caller()->AddTrack(caller_track);
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_video = 0;
  caller()->SetOfferAnswerOptions(options);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);

  // Expect video to be received in one direction.
  MediaExpectations media_expectations;
  media_expectations.CallerExpectsNoVideo();
  media_expectations.CalleeExpectsSomeVideo();

  EXPECT_TRUE(ExpectNewFrames(media_expectations));
}

// Tests that receive only works without the caller having an encoder factory
// and the callee having a decoder factory.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithReceiveOnlyVideo) {
  ASSERT_TRUE(
      CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/false));
  ConnectFakeSignaling();
  // Add one-directional video, from callee to caller.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
      callee()->CreateLocalVideoTrack();
  callee()->AddTrack(callee_track);
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_video = 1;
  caller()->SetOfferAnswerOptions(options);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_EQ(caller()->pc()->GetReceivers().size(), 1u);

  // Expect video to be received in one direction.
  MediaExpectations media_expectations;
  media_expectations.CallerExpectsSomeVideo();
  media_expectations.CalleeExpectsNoVideo();

  EXPECT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationTest,
       EndToEndCallAddReceiveVideoToSendOnlyCall) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add one-directional video, from caller to callee.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
      caller()->CreateLocalVideoTrack();
  caller()->AddTrack(caller_track);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Add receive video.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
      callee()->CreateLocalVideoTrack();
  callee()->AddTrack(callee_track);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Ensure that video frames are received end-to-end.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationTest,
       EndToEndCallAddSendVideoToReceiveOnlyCall) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add one-directional video, from callee to caller.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
      callee()->CreateLocalVideoTrack();
  callee()->AddTrack(callee_track);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Add send video.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
      caller()->CreateLocalVideoTrack();
  caller()->AddTrack(caller_track);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Expect video to be received in one direction.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationTest,
       EndToEndCallRemoveReceiveVideoFromSendReceiveCall) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add send video, from caller to callee.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
      caller()->CreateLocalVideoTrack();
  rtc::scoped_refptr<webrtc::RtpSenderInterface> caller_sender =
      caller()->AddTrack(caller_track);
  // Add receive video, from callee to caller.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
      callee()->CreateLocalVideoTrack();

  rtc::scoped_refptr<webrtc::RtpSenderInterface> callee_sender =
      callee()->AddTrack(callee_track);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Remove receive video (i.e., callee sender track).
  callee()->pc()->RemoveTrackOrError(callee_sender);

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Expect one-directional video.
  MediaExpectations media_expectations;
  media_expectations.CallerExpectsNoVideo();
  media_expectations.CalleeExpectsSomeVideo();

  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationTest,
       EndToEndCallRemoveSendVideoFromSendReceiveCall) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add send video, from caller to callee.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
      caller()->CreateLocalVideoTrack();
  rtc::scoped_refptr<webrtc::RtpSenderInterface> caller_sender =
      caller()->AddTrack(caller_track);
  // Add receive video, from callee to caller.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
      callee()->CreateLocalVideoTrack();

  rtc::scoped_refptr<webrtc::RtpSenderInterface> callee_sender =
      callee()->AddTrack(callee_track);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Remove send video (i.e., caller sender track).
  caller()->pc()->RemoveTrackOrError(caller_sender);

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Expect one-directional video.
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsNoVideo();
  media_expectations.CallerExpectsSomeVideo();

  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// This test sets up a audio call initially, with the callee rejecting video
// initially. Then later the callee decides to upgrade to audio/video, and
// initiates a new offer/answer exchange.
TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Initially, offer an audio/video stream from the caller, but refuse to
  // send/receive video on the callee side.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioTrack();
  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_video = 0;
    callee()->SetOfferAnswerOptions(options);
  } else {
    callee()->SetRemoteOfferHandler([this] {
      callee()
          ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
          ->StopInternal();
    });
  }
  // Do offer/answer and make sure audio is still received end-to-end.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  {
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudio();
    media_expectations.ExpectNoVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  // Sanity check that the callee's description has a rejected video section.
  ASSERT_NE(nullptr, callee()->pc()->local_description());
  const ContentInfo* callee_video_content =
      GetFirstVideoContent(callee()->pc()->local_description()->description());
  ASSERT_NE(nullptr, callee_video_content);
  EXPECT_TRUE(callee_video_content->rejected);

  // Now negotiate with video and ensure negotiation succeeds, with video
  // frames and additional audio frames being received.
  callee()->AddVideoTrack();
  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_video = 1;
    callee()->SetOfferAnswerOptions(options);
  } else {
    callee()->SetRemoteOfferHandler(nullptr);
    caller()->SetRemoteOfferHandler([this] {
      // The caller creates a new transceiver to receive video on when receiving
      // the offer, but by default it is send only.
      auto transceivers = caller()->pc()->GetTransceivers();
      ASSERT_EQ(2U, transceivers.size());
      ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO,
                transceivers[1]->receiver()->media_type());
      transceivers[1]->sender()->SetTrack(
          caller()->CreateLocalVideoTrack().get());
      transceivers[1]->SetDirectionWithError(
          RtpTransceiverDirection::kSendRecv);
    });
  }
  callee()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  {
    // Expect additional audio frames to be received after the upgrade.
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
}

// Simpler than the above test; just add an audio track to an established
// video-only connection.
TEST_P(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Do initial offer/answer with just a video track.
  caller()->AddVideoTrack();
  callee()->AddVideoTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Now add an audio track and do another offer/answer.
  caller()->AddAudioTrack();
  callee()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Ensure both audio and video frames are received end-to-end.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// This test sets up a non-bundled call and negotiates bundling at the same
// time as starting an ICE restart. When bundling is in effect in the restart,
// the DTLS-SRTP context should be successfully reset.
TEST_P(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  // Remove the bundle group from the SDP received by the callee.
  callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
    desc->RemoveGroupByName("BUNDLE");
  });
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  {
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  // Now stop removing the BUNDLE group, and trigger an ICE restart.
  callee()->SetReceivedSdpMunger(nullptr);
  caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Expect additional frames to be received after the ICE restart.
  {
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
}

// Test CVO (Coordination of Video Orientation). If a video source is rotated
// and both peers support the CVO RTP header extension, the actual video frames
// don't need to be encoded in different resolutions, since the rotation is
// communicated through the RTP header extension.
TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add rotated video tracks.
  caller()->AddTrack(
      caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
  callee()->AddTrack(
      callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));

  // Wait for video frames to be received by both sides.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
                       callee()->min_video_frames_received_per_track() > 0,
                   kMaxWaitForFramesMs);

  // Ensure that the aspect ratio is unmodified.
  // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
  // not just assumed.
  EXPECT_EQ(4.0 / 3, caller()->local_rendered_aspect_ratio());
  EXPECT_EQ(4.0 / 3, caller()->rendered_aspect_ratio());
  EXPECT_EQ(4.0 / 3, callee()->local_rendered_aspect_ratio());
  EXPECT_EQ(4.0 / 3, callee()->rendered_aspect_ratio());
  // Ensure that the CVO bits were surfaced to the renderer.
  EXPECT_EQ(webrtc::kVideoRotation_270, caller()->rendered_rotation());
  EXPECT_EQ(webrtc::kVideoRotation_90, callee()->rendered_rotation());
}

// Test that when the CVO extension isn't supported, video is rotated the
// old-fashioned way, by encoding rotated frames.
TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add rotated video tracks.
  caller()->AddTrack(
      caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
  callee()->AddTrack(
      callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));

  // Remove the CVO extension from the offered SDP.
  callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
    cricket::VideoContentDescription* video =
        GetFirstVideoContentDescription(desc);
    video->ClearRtpHeaderExtensions();
  });
  // Wait for video frames to be received by both sides.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
                       callee()->min_video_frames_received_per_track() > 0,
                   kMaxWaitForFramesMs);

  // Expect that the aspect ratio is inversed to account for the 90/270 degree
  // rotation.
  // TODO(deadbeef): Where does 4:3 come from? Should be explicit in the test,
  // not just assumed.
  EXPECT_EQ(3.0 / 4, caller()->local_rendered_aspect_ratio());
  EXPECT_EQ(3.0 / 4, caller()->rendered_aspect_ratio());
  EXPECT_EQ(3.0 / 4, callee()->local_rendered_aspect_ratio());
  EXPECT_EQ(3.0 / 4, callee()->rendered_aspect_ratio());
  // Expect that each endpoint is unaware of the rotation of the other endpoint.
  EXPECT_EQ(webrtc::kVideoRotation_0, caller()->rendered_rotation());
  EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation());
}

// Test that if the answerer rejects the audio m= section, no audio is sent or
// received, but video still can be.
TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    // Only add video track for callee, and set offer_to_receive_audio to 0, so
    // it will reject the audio m= section completely.
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 0;
    callee()->SetOfferAnswerOptions(options);
  } else {
    // Stopping the audio RtpTransceiver will cause the media section to be
    // rejected in the answer.
    callee()->SetRemoteOfferHandler([this] {
      callee()
          ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
          ->StopInternal();
    });
  }
  callee()->AddTrack(callee()->CreateLocalVideoTrack());
  // Do offer/answer and wait for successful end-to-end video frames.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalVideo();
  media_expectations.ExpectNoAudio();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));

  // Sanity check that the callee's description has a rejected audio section.
  ASSERT_NE(nullptr, callee()->pc()->local_description());
  const ContentInfo* callee_audio_content =
      GetFirstAudioContent(callee()->pc()->local_description()->description());
  ASSERT_NE(nullptr, callee_audio_content);
  EXPECT_TRUE(callee_audio_content->rejected);
  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
    // The caller's transceiver should have stopped after receiving the answer,
    // and thus no longer listed in transceivers.
    EXPECT_EQ(nullptr,
              caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO));
  }
}

// Test that if the answerer rejects the video m= section, no video is sent or
// received, but audio still can be.
TEST_P(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    // Only add audio track for callee, and set offer_to_receive_video to 0, so
    // it will reject the video m= section completely.
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_video = 0;
    callee()->SetOfferAnswerOptions(options);
  } else {
    // Stopping the video RtpTransceiver will cause the media section to be
    // rejected in the answer.
    callee()->SetRemoteOfferHandler([this] {
      callee()
          ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
          ->StopInternal();
    });
  }
  callee()->AddTrack(callee()->CreateLocalAudioTrack());
  // Do offer/answer and wait for successful end-to-end audio frames.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudio();
  media_expectations.ExpectNoVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));

  // Sanity check that the callee's description has a rejected video section.
  ASSERT_NE(nullptr, callee()->pc()->local_description());
  const ContentInfo* callee_video_content =
      GetFirstVideoContent(callee()->pc()->local_description()->description());
  ASSERT_NE(nullptr, callee_video_content);
  EXPECT_TRUE(callee_video_content->rejected);
  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
    // The caller's transceiver should have stopped after receiving the answer,
    // and thus is no longer present.
    EXPECT_EQ(nullptr,
              caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO));
  }
}

// Test that if the answerer rejects both audio and video m= sections, nothing
// bad happens.
// TODO(deadbeef): Test that a data channel still works. Currently this doesn't
// test anything but the fact that negotiation succeeds, which doesn't mean
// much.
TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    // Don't give the callee any tracks, and set offer_to_receive_X to 0, so it
    // will reject both audio and video m= sections.
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_audio = 0;
    options.offer_to_receive_video = 0;
    callee()->SetOfferAnswerOptions(options);
  } else {
    callee()->SetRemoteOfferHandler([this] {
      // Stopping all transceivers will cause all media sections to be rejected.
      for (const auto& transceiver : callee()->pc()->GetTransceivers()) {
        transceiver->StopInternal();
      }
    });
  }
  // Do offer/answer and wait for stable signaling state.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Sanity check that the callee's description has rejected m= sections.
  ASSERT_NE(nullptr, callee()->pc()->local_description());
  const ContentInfo* callee_audio_content =
      GetFirstAudioContent(callee()->pc()->local_description()->description());
  ASSERT_NE(nullptr, callee_audio_content);
  EXPECT_TRUE(callee_audio_content->rejected);
  const ContentInfo* callee_video_content =
      GetFirstVideoContent(callee()->pc()->local_description()->description());
  ASSERT_NE(nullptr, callee_video_content);
  EXPECT_TRUE(callee_video_content->rejected);
}

// This test sets up an audio and video call between two parties. After the
// call runs for a while, the caller sends an updated offer with video being
// rejected. Once the re-negotiation is done, the video flow should stop and
// the audio flow should continue.
TEST_P(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  {
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudioAndVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  // Renegotiate, rejecting the video m= section.
  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    caller()->SetGeneratedSdpMunger(
        [](cricket::SessionDescription* description) {
          for (cricket::ContentInfo& content : description->contents()) {
            if (cricket::IsVideoContent(&content)) {
              content.rejected = true;
            }
          }
        });
  } else {
    caller()
        ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
        ->StopInternal();
  }
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);

  // Sanity check that the caller's description has a rejected video section.
  ASSERT_NE(nullptr, caller()->pc()->local_description());
  const ContentInfo* caller_video_content =
      GetFirstVideoContent(caller()->pc()->local_description()->description());
  ASSERT_NE(nullptr, caller_video_content);
  EXPECT_TRUE(caller_video_content->rejected);
  // Wait for some additional audio frames to be received.
  {
    MediaExpectations media_expectations;
    media_expectations.ExpectBidirectionalAudio();
    media_expectations.ExpectNoVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
}

// Do one offer/answer with audio, another that disables it (rejecting the m=
// section), and another that re-enables it. Regression test for:
// bugs.webrtc.org/6023
TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  // Add audio track, do normal offer/answer.
  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
      caller()->CreateLocalAudioTrack();
  rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
      caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Remove audio track, and set offer_to_receive_audio to false to cause the
  // m= section to be completely disabled, not just "recvonly".
  caller()->pc()->RemoveTrackOrError(sender);
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_audio = 0;
  caller()->SetOfferAnswerOptions(options);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Add the audio track again, expecting negotiation to succeed and frames to
  // flow.
  sender = caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
  options.offer_to_receive_audio = 1;
  caller()->SetOfferAnswerOptions(options);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudio();
  EXPECT_TRUE(ExpectNewFrames(media_expectations));
}

// Basic end-to-end test, but without SSRC/MSID signaling. This functionality
// is needed to support legacy endpoints.
// TODO(deadbeef): When we support the MID extension and demuxing on MID, also
// add a test for an end-to-end test without MID signaling either (basically,
// the minimum acceptable SDP).
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add audio and video, testing that packets can be demuxed on payload type.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  // Remove SSRCs and MSIDs from the received offer SDP.
  callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// Basic end-to-end test, without SSRC signaling. This means that the track
// was created properly and frames are delivered when the MSIDs are communicated
// with a=msid lines and no a=ssrc lines.
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       EndToEndCallWithoutSsrcSignaling) {
  const char kStreamId[] = "streamId";
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add just audio tracks.
  caller()->AddTrack(caller()->CreateLocalAudioTrack(), {kStreamId});
  callee()->AddAudioTrack();

  // Remove SSRCs from the received offer SDP.
  callee()->SetReceivedSdpMunger(RemoveSsrcsAndKeepMsids);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudio();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       EndToEndCallAddReceiveVideoToSendOnlyCall) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add one-directional video, from caller to callee.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
      caller()->CreateLocalVideoTrack();

  RtpTransceiverInit video_transceiver_init;
  video_transceiver_init.stream_ids = {"video1"};
  video_transceiver_init.direction = RtpTransceiverDirection::kSendOnly;
  auto video_sender =
      caller()->pc()->AddTransceiver(track, video_transceiver_init).MoveValue();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Add receive direction.
  video_sender->SetDirectionWithError(RtpTransceiverDirection::kSendRecv);

  rtc::scoped_refptr<webrtc::VideoTrackInterface> callee_track =
      callee()->CreateLocalVideoTrack();

  callee()->AddTrack(callee_track);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Ensure that video frames are received end-to-end.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// Tests that video flows between multiple video tracks when SSRCs are not
// signaled. This exercises the MID RTP header extension which is needed to
// demux the incoming video tracks.
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       EndToEndCallWithTwoVideoTracksAndNoSignaledSsrc) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddVideoTrack();
  caller()->AddVideoTrack();
  callee()->AddVideoTrack();
  callee()->AddVideoTrack();

  caller()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
  callee()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_EQ(2u, caller()->pc()->GetReceivers().size());
  ASSERT_EQ(2u, callee()->pc()->GetReceivers().size());

  // Expect video to be received in both directions on both tracks.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalVideo();
  EXPECT_TRUE(ExpectNewFrames(media_expectations));
}

// Used for the test below.
void RemoveBundleGroupSsrcsAndMidExtension(cricket::SessionDescription* desc) {
  RemoveSsrcsAndKeepMsids(desc);
  desc->RemoveGroupByName("BUNDLE");
  for (ContentInfo& content : desc->contents()) {
    cricket::MediaContentDescription* media = content.media_description();
    cricket::RtpHeaderExtensions extensions = media->rtp_header_extensions();
    extensions.erase(std::remove_if(extensions.begin(), extensions.end(),
                                    [](const RtpExtension& extension) {
                                      return extension.uri ==
                                             RtpExtension::kMidUri;
                                    }),
                     extensions.end());
    media->set_rtp_header_extensions(extensions);
  }
}

// Tests that video flows between multiple video tracks when BUNDLE is not used,
// SSRCs are not signaled and the MID RTP header extension is not used. This
// relies on demuxing by payload type, which normally doesn't work if you have
// multiple media sections using the same payload type, but which should work as
// long as the media sections aren't bundled.
// Regression test for: http://crbug.com/webrtc/12023
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       EndToEndCallWithTwoVideoTracksNoBundleNoSignaledSsrcAndNoMid) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddVideoTrack();
  caller()->AddVideoTrack();
  callee()->AddVideoTrack();
  callee()->AddVideoTrack();
  caller()->SetReceivedSdpMunger(&RemoveBundleGroupSsrcsAndMidExtension);
  callee()->SetReceivedSdpMunger(&RemoveBundleGroupSsrcsAndMidExtension);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_EQ(2u, caller()->pc()->GetReceivers().size());
  ASSERT_EQ(2u, callee()->pc()->GetReceivers().size());
  // Make sure we are not bundled.
  ASSERT_NE(caller()->pc()->GetSenders()[0]->dtls_transport(),
            caller()->pc()->GetSenders()[1]->dtls_transport());

  // Expect video to be received in both directions on both tracks.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalVideo();
  EXPECT_TRUE(ExpectNewFrames(media_expectations));
}

// Used for the test below.
void ModifyPayloadTypesAndRemoveMidExtension(
    cricket::SessionDescription* desc) {
  int pt = 96;
  for (ContentInfo& content : desc->contents()) {
    cricket::MediaContentDescription* media = content.media_description();
    cricket::RtpHeaderExtensions extensions = media->rtp_header_extensions();
    extensions.erase(std::remove_if(extensions.begin(), extensions.end(),
                                    [](const RtpExtension& extension) {
                                      return extension.uri ==
                                             RtpExtension::kMidUri;
                                    }),
                     extensions.end());
    media->set_rtp_header_extensions(extensions);
    cricket::VideoContentDescription* video = media->as_video();
    ASSERT_TRUE(video != nullptr);
    std::vector<cricket::VideoCodec> codecs = {{pt++, "VP8"}};
    video->set_codecs(codecs);
  }
}

// Tests that two video tracks can be demultiplexed by payload type alone, by
// using different payload types for the same codec in different m= sections.
// This practice is discouraged but historically has been supported.
// Regression test for: http://crbug.com/webrtc/12029
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       EndToEndCallWithTwoVideoTracksDemultiplexedByPayloadType) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddVideoTrack();
  caller()->AddVideoTrack();
  callee()->AddVideoTrack();
  callee()->AddVideoTrack();
  caller()->SetGeneratedSdpMunger(&ModifyPayloadTypesAndRemoveMidExtension);
  callee()->SetGeneratedSdpMunger(&ModifyPayloadTypesAndRemoveMidExtension);
  // We can't remove SSRCs from the generated SDP because then no send streams
  // would be created.
  caller()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
  callee()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_EQ(2u, caller()->pc()->GetReceivers().size());
  ASSERT_EQ(2u, callee()->pc()->GetReceivers().size());
  // Make sure we are bundled.
  ASSERT_EQ(caller()->pc()->GetSenders()[0]->dtls_transport(),
            caller()->pc()->GetSenders()[1]->dtls_transport());

  // Expect video to be received in both directions on both tracks.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalVideo();
  EXPECT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLinePresent) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioTrack();
  caller()->AddVideoTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  auto callee_receivers = callee()->pc()->GetReceivers();
  ASSERT_EQ(2u, callee_receivers.size());
  EXPECT_TRUE(callee_receivers[0]->stream_ids().empty());
  EXPECT_TRUE(callee_receivers[1]->stream_ids().empty());
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan, NoStreamsMsidLineMissing) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioTrack();
  caller()->AddVideoTrack();
  callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  auto callee_receivers = callee()->pc()->GetReceivers();
  ASSERT_EQ(2u, callee_receivers.size());
  ASSERT_EQ(1u, callee_receivers[0]->stream_ids().size());
  ASSERT_EQ(1u, callee_receivers[1]->stream_ids().size());
  EXPECT_EQ(callee_receivers[0]->stream_ids()[0],
            callee_receivers[1]->stream_ids()[0]);
  EXPECT_EQ(callee_receivers[0]->streams()[0],
            callee_receivers[1]->streams()[0]);
}

// Test that if two video tracks are sent (from caller to callee, in this test),
// they're transmitted correctly end-to-end.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Add one audio/video stream, and one video-only stream.
  caller()->AddAudioVideoTracks();
  caller()->AddVideoTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_EQ(3u, callee()->pc()->GetReceivers().size());

  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

static void MakeSpecCompliantMaxBundleOffer(cricket::SessionDescription* desc) {
  bool first = true;
  for (cricket::ContentInfo& content : desc->contents()) {
    if (first) {
      first = false;
      continue;
    }
    content.bundle_only = true;
  }
  first = true;
  for (cricket::TransportInfo& transport : desc->transport_infos()) {
    if (first) {
      first = false;
      continue;
    }
    transport.description.ice_ufrag.clear();
    transport.description.ice_pwd.clear();
    transport.description.connection_role = cricket::CONNECTIONROLE_NONE;
    transport.description.identity_fingerprint.reset(nullptr);
  }
}

// Test that if applying a true "max bundle" offer, which uses ports of 0,
// "a=bundle-only", omitting "a=fingerprint", "a=setup", "a=ice-ufrag" and
// "a=ice-pwd" for all but the audio "m=" section, negotiation still completes
// successfully and media flows.
// TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works.
// TODO(deadbeef): Won't need this test once we start generating actual
// standards-compliant SDP.
TEST_P(PeerConnectionIntegrationTest,
       EndToEndCallWithSpecCompliantMaxBundleOffer) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  // Do the equivalent of setting the port to 0, adding a=bundle-only, and
  // removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all
  // but the first m= section.
  callee()->SetReceivedSdpMunger(MakeSpecCompliantMaxBundleOffer);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// Test that we can receive the audio output level from a remote audio track.
// TODO(deadbeef): Use a fake audio source and verify that the output level is
// exactly what the source on the other side was configured with.
TEST_P(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Just add an audio track.
  caller()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Get the audio output level stats. Note that the level is not available
  // until an RTCP packet has been received.
  EXPECT_TRUE_WAIT(callee()->OldGetStats()->AudioOutputLevel() > 0,
                   kMaxWaitForFramesMs);
}

// Test that an audio input level is reported.
// TODO(deadbeef): Use a fake audio source and verify that the input level is
// exactly what the source was configured with.
TEST_P(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Just add an audio track.
  caller()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Get the audio input level stats. The level should be available very
  // soon after the test starts.
  EXPECT_TRUE_WAIT(caller()->OldGetStats()->AudioInputLevel() > 0,
                   kMaxWaitForStatsMs);
}

// Test that we can get incoming byte counts from both audio and video tracks.
TEST_P(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  // Do offer/answer, wait for the callee to receive some frames.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));

  // Get a handle to the remote tracks created, so they can be used as GetStats
  // filters.
  for (const auto& receiver : callee()->pc()->GetReceivers()) {
    // We received frames, so we definitely should have nonzero "received bytes"
    // stats at this point.
    EXPECT_GT(
        callee()->OldGetStatsForTrack(receiver->track().get())->BytesReceived(),
        0);
  }
}

// Test that we can get outgoing byte counts from both audio and video tracks.
TEST_P(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  auto audio_track = caller()->CreateLocalAudioTrack();
  auto video_track = caller()->CreateLocalVideoTrack();
  caller()->AddTrack(audio_track);
  caller()->AddTrack(video_track);
  // Do offer/answer, wait for the callee to receive some frames.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));

  // The callee received frames, so we definitely should have nonzero "sent
  // bytes" stats at this point.
  EXPECT_GT(caller()->OldGetStatsForTrack(audio_track.get())->BytesSent(), 0);
  EXPECT_GT(caller()->OldGetStatsForTrack(video_track.get())->BytesSent(), 0);
}

// Test that the track ID is associated with all local and remote SSRC stats
// using the old GetStats() and more than 1 audio and more than 1 video track.
// This is a regression test for crbug.com/906988
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       OldGetStatsAssociatesTrackIdForManyMediaSections) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  auto audio_sender_1 = caller()->AddAudioTrack();
  auto video_sender_1 = caller()->AddVideoTrack();
  auto audio_sender_2 = caller()->AddAudioTrack();
  auto video_sender_2 = caller()->AddVideoTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudioAndVideo();
  ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);

  std::vector<std::string> track_ids = {
      audio_sender_1->track()->id(), video_sender_1->track()->id(),
      audio_sender_2->track()->id(), video_sender_2->track()->id()};

  auto caller_stats = caller()->OldGetStats();
  EXPECT_THAT(caller_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
  auto callee_stats = callee()->OldGetStats();
  EXPECT_THAT(callee_stats->TrackIds(), UnorderedElementsAreArray(track_ids));
}

// Test that the new GetStats() returns stats for all outgoing/incoming streams
// with the correct track identifiers if there are more than one audio and more
// than one video senders/receivers.
TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  auto audio_sender_1 = caller()->AddAudioTrack();
  auto video_sender_1 = caller()->AddVideoTrack();
  auto audio_sender_2 = caller()->AddAudioTrack();
  auto video_sender_2 = caller()->AddVideoTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudioAndVideo();
  ASSERT_TRUE_WAIT(ExpectNewFrames(media_expectations), kDefaultTimeout);

  std::vector<std::string> track_ids = {
      audio_sender_1->track()->id(), video_sender_1->track()->id(),
      audio_sender_2->track()->id(), video_sender_2->track()->id()};

  rtc::scoped_refptr<const webrtc::RTCStatsReport> caller_report =
      caller()->NewGetStats();
  ASSERT_TRUE(caller_report);
  auto outbound_stream_stats =
      caller_report->GetStatsOfType<webrtc::RTCOutboundRTPStreamStats>();
  ASSERT_EQ(outbound_stream_stats.size(), 4u);
  std::vector<std::string> outbound_track_ids;
  for (const auto& stat : outbound_stream_stats) {
    ASSERT_TRUE(stat->bytes_sent.is_defined());
    EXPECT_LT(0u, *stat->bytes_sent);
    if (*stat->kind == "video") {
      ASSERT_TRUE(stat->key_frames_encoded.is_defined());
      EXPECT_GT(*stat->key_frames_encoded, 0u);
      ASSERT_TRUE(stat->frames_encoded.is_defined());
      EXPECT_GE(*stat->frames_encoded, *stat->key_frames_encoded);
    }
    ASSERT_TRUE(stat->media_source_id.is_defined());
    const RTCMediaSourceStats* media_source =
        static_cast<const RTCMediaSourceStats*>(
            caller_report->Get(*stat->media_source_id));
    ASSERT_TRUE(media_source);
    outbound_track_ids.push_back(*media_source->track_identifier);
  }
  EXPECT_THAT(outbound_track_ids, UnorderedElementsAreArray(track_ids));

  rtc::scoped_refptr<const webrtc::RTCStatsReport> callee_report =
      callee()->NewGetStats();
  ASSERT_TRUE(callee_report);
  auto inbound_stream_stats =
      callee_report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
  ASSERT_EQ(4u, inbound_stream_stats.size());
  std::vector<std::string> inbound_track_ids;
  for (const auto& stat : inbound_stream_stats) {
    ASSERT_TRUE(stat->bytes_received.is_defined());
    EXPECT_LT(0u, *stat->bytes_received);
    if (*stat->kind == "video") {
      ASSERT_TRUE(stat->key_frames_decoded.is_defined());
      EXPECT_GT(*stat->key_frames_decoded, 0u);
      ASSERT_TRUE(stat->frames_decoded.is_defined());
      EXPECT_GE(*stat->frames_decoded, *stat->key_frames_decoded);
    }
    inbound_track_ids.push_back(*stat->track_identifier);
  }
  EXPECT_THAT(inbound_track_ids, UnorderedElementsAreArray(track_ids));
}

// Test that we can get stats (using the new stats implementation) for
// unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
// SDP.
TEST_P(PeerConnectionIntegrationTest,
       GetStatsForUnsignaledStreamWithNewStatsApi) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioTrack();
  // Remove SSRCs and MSIDs from the received offer SDP.
  callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudio(1);
  ASSERT_TRUE(ExpectNewFrames(media_expectations));

  // We received a frame, so we should have nonzero "bytes received" stats for
  // the unsignaled stream, if stats are working for it.
  rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
      callee()->NewGetStats();
  ASSERT_NE(nullptr, report);
  auto inbound_stream_stats =
      report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
  ASSERT_EQ(1U, inbound_stream_stats.size());
  ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
  ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
  ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined());
}

// Same as above but for the legacy stats implementation.
TEST_P(PeerConnectionIntegrationTest,
       GetStatsForUnsignaledStreamWithOldStatsApi) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioTrack();
  // Remove SSRCs and MSIDs from the received offer SDP.
  callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Note that, since the old stats implementation associates SSRCs with tracks
  // using SDP, when SSRCs aren't signaled in SDP these stats won't have an
  // associated track ID. So we can't use the track "selector" argument.
  //
  // Also, we use "EXPECT_TRUE_WAIT" because the stats collector may decide to
  // return cached stats if not enough time has passed since the last update.
  EXPECT_TRUE_WAIT(callee()->OldGetStats()->BytesReceived() > 0,
                   kDefaultTimeout);
}

// Test that we can successfully get the media related stats (audio level
// etc.) for the unsignaled stream.
TEST_P(PeerConnectionIntegrationTest,
       GetMediaStatsForUnsignaledStreamWithNewStatsApi) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  // Remove SSRCs and MSIDs from the received offer SDP.
  callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudio(1);
  media_expectations.CalleeExpectsSomeVideo(1);
  ASSERT_TRUE(ExpectNewFrames(media_expectations));

  rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
      callee()->NewGetStats();
  ASSERT_NE(nullptr, report);

  auto inbound_rtps =
      report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
  auto index = FindFirstMediaStatsIndexByKind("audio", inbound_rtps);
  ASSERT_GE(index, 0);
  EXPECT_TRUE(inbound_rtps[index]->audio_level.is_defined());
}

// Helper for test below.
void ModifySsrcs(cricket::SessionDescription* desc) {
  for (ContentInfo& content : desc->contents()) {
    for (StreamParams& stream :
         content.media_description()->mutable_streams()) {
      for (uint32_t& ssrc : stream.ssrcs) {
        ssrc = rtc::CreateRandomId();
      }
    }
  }
}

// Test that the "DEPRECATED_RTCMediaStreamTrackStats"  object is updated
// correctly when SSRCs are unsignaled, and the SSRC of the received (audio)
// stream changes. This should result in two "RTCInboundRTPStreamStats", but
// only one "DEPRECATED_RTCMediaStreamTrackStats", whose counters go up
// continuously rather than being reset to 0 once the SSRC change occurs.
//
// Regression test for this bug:
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8158
//
// The bug causes the track stats to only represent one of the two streams:
// whichever one has the higher SSRC. So with this bug, there was a 50% chance
// that the track stat counters would reset to 0 when the new stream is
// received, and a 50% chance that they'll stop updating (while
// "concealed_samples" continues increasing, due to silence being generated for
// the inactive stream).
TEST_P(PeerConnectionIntegrationTest,
       TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioTrack();
  // Remove SSRCs and MSIDs from the received offer SDP, simulating an endpoint
  // that doesn't signal SSRCs (from the callee's perspective).
  callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Wait for 50 audio frames (500ms of audio) to be received by the callee.
  {
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio(50);
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  // Some audio frames were received, so we should have nonzero "samples
  // received" for the track.
  rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
      callee()->NewGetStats();
  ASSERT_NE(nullptr, report);
  auto track_stats =
      report->GetStatsOfType<webrtc::DEPRECATED_RTCMediaStreamTrackStats>();
  ASSERT_EQ(1U, track_stats.size());
  ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
  ASSERT_GT(*track_stats[0]->total_samples_received, 0U);
  // uint64_t prev_samples_received = *track_stats[0]->total_samples_received;

  // Create a new offer and munge it to cause the caller to use a new SSRC.
  caller()->SetGeneratedSdpMunger(ModifySsrcs);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Wait for 25 more audio frames (250ms of audio) to be received, from the new
  // SSRC.
  {
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio(25);
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }

  report = callee()->NewGetStats();
  ASSERT_NE(nullptr, report);
  track_stats =
      report->GetStatsOfType<webrtc::DEPRECATED_RTCMediaStreamTrackStats>();
  ASSERT_EQ(1U, track_stats.size());
  ASSERT_TRUE(track_stats[0]->total_samples_received.is_defined());
  // The "total samples received" stat should only be greater than it was
  // before.
  // TODO(deadbeef): Uncomment this assertion once the bug is completely fixed.
  // Right now, the new SSRC will cause the counters to reset to 0.
  // EXPECT_GT(*track_stats[0]->total_samples_received, prev_samples_received);

  // Additionally, the percentage of concealed samples (samples generated to
  // conceal packet loss) should be less than 50%. If it's greater, that's a
  // good sign that we're seeing stats from the old stream that's no longer
  // receiving packets, and is generating concealed samples of silence.
  constexpr double kAcceptableConcealedSamplesPercentage = 0.50;
  ASSERT_TRUE(track_stats[0]->concealed_samples.is_defined());
  EXPECT_LT(*track_stats[0]->concealed_samples,
            *track_stats[0]->total_samples_received *
                kAcceptableConcealedSamplesPercentage);

  // Also ensure that we have two "RTCInboundRTPStreamStats" as expected, as a
  // sanity check that the SSRC really changed.
  // TODO(deadbeef): This isn't working right now, because we're not returning
  // *any* stats for the inactive stream. Uncomment when the bug is completely
  // fixed.
  // auto inbound_stream_stats =
  //     report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
  // ASSERT_EQ(2U, inbound_stream_stats.size());
}

// Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
  PeerConnectionFactory::Options dtls_10_options;
  dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
                                                      dtls_10_options));
  ConnectFakeSignaling();
  // Do normal offer/answer and wait for some frames to be received in each
  // direction.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// Test getting cipher stats and UMA metrics when DTLS 1.0 is negotiated.
TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
  PeerConnectionFactory::Options dtls_10_options;
  dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
                                                      dtls_10_options));
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
  EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
                       caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
                   kDefaultTimeout);
  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                 caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
  // TODO(bugs.webrtc.org/9456): Fix it.
  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
                          "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
                          kDefaultSrtpCryptoSuite));
}

// Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
  PeerConnectionFactory::Options dtls_12_options;
  dtls_12_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options,
                                                      dtls_12_options));
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
  EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
                       caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
                   kDefaultTimeout);
  EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
                 caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
  // TODO(bugs.webrtc.org/9456): Fix it.
  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
                          "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
                          kDefaultSrtpCryptoSuite));
}

// Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the
// callee only supports 1.0.
TEST_P(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) {
  PeerConnectionFactory::Options caller_options;
  caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
  PeerConnectionFactory::Options callee_options;
  callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
  ASSERT_TRUE(
      CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
  ConnectFakeSignaling();
  // Do normal offer/answer and wait for some frames to be received in each
  // direction.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// Test that DTLS 1.0 can be used if the caller only supports DTLS 1.0 and the
// callee supports 1.2.
TEST_P(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
  PeerConnectionFactory::Options caller_options;
  caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
  PeerConnectionFactory::Options callee_options;
  callee_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
  ASSERT_TRUE(
      CreatePeerConnectionWrappersWithOptions(caller_options, callee_options));
  ConnectFakeSignaling();
  // Do normal offer/answer and wait for some frames to be received in each
  // direction.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// The three tests below verify that "enable_aes128_sha1_32_crypto_cipher"
// works as expected; the cipher should only be used if enabled by both sides.
TEST_P(PeerConnectionIntegrationTest,
       Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) {
  PeerConnectionFactory::Options caller_options;
  caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
  PeerConnectionFactory::Options callee_options;
  callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
      false;
  int expected_cipher_suite = rtc::kSrtpAes128CmSha1_80;
  TestNegotiatedCipherSuite(caller_options, callee_options,
                            expected_cipher_suite);
}

TEST_P(PeerConnectionIntegrationTest,
       Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) {
  PeerConnectionFactory::Options caller_options;
  caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher =
      false;
  PeerConnectionFactory::Options callee_options;
  callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
  int expected_cipher_suite = rtc::kSrtpAes128CmSha1_80;
  TestNegotiatedCipherSuite(caller_options, callee_options,
                            expected_cipher_suite);
}

TEST_P(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) {
  PeerConnectionFactory::Options caller_options;
  caller_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
  PeerConnectionFactory::Options callee_options;
  callee_options.crypto_options.srtp.enable_aes128_sha1_32_crypto_cipher = true;
  int expected_cipher_suite = rtc::kSrtpAes128CmSha1_32;
  TestNegotiatedCipherSuite(caller_options, callee_options,
                            expected_cipher_suite);
}

// Test that a non-GCM cipher is used if both sides only support non-GCM.
TEST_P(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) {
  bool local_gcm_enabled = false;
  bool remote_gcm_enabled = false;
  bool aes_ctr_enabled = true;
  int expected_cipher_suite = kDefaultSrtpCryptoSuite;
  TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                    aes_ctr_enabled, expected_cipher_suite);
}

// Test that a GCM cipher is used if both ends support it and non-GCM is
// disabled.
TEST_P(PeerConnectionIntegrationTest, GcmCipherUsedWhenOnlyGcmSupported) {
  bool local_gcm_enabled = true;
  bool remote_gcm_enabled = true;
  bool aes_ctr_enabled = false;
  int expected_cipher_suite = kDefaultSrtpCryptoSuiteGcm;
  TestGcmNegotiationUsesCipherSuite(local_gcm_enabled, remote_gcm_enabled,
                                    aes_ctr_enabled, expected_cipher_suite);
}

// Verify that media can be transmitted end-to-end when GCM crypto suites are
// enabled. Note that the above tests, such as GcmCipherUsedWhenGcmSupported,
// only verify that a GCM cipher is negotiated, and not necessarily that SRTP
// works with it.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
  PeerConnectionFactory::Options gcm_options;
  gcm_options.crypto_options.srtp.enable_gcm_crypto_suites = true;
  gcm_options.crypto_options.srtp.enable_aes128_sha1_80_crypto_cipher = false;
  ASSERT_TRUE(
      CreatePeerConnectionWrappersWithOptions(gcm_options, gcm_options));
  ConnectFakeSignaling();
  // Do normal offer/answer and wait for some frames to be received in each
  // direction.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// Test that the ICE connection and gathering states eventually reach
// "complete".
TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Do normal offer/answer.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
                 caller()->ice_gathering_state(), kMaxWaitForFramesMs);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
                 callee()->ice_gathering_state(), kMaxWaitForFramesMs);
  // After the best candidate pair is selected and all candidates are signaled,
  // the ICE connection state should reach "complete".
  // TODO(deadbeef): Currently, the ICE "controlled" agent (the
  // answerer/"callee" by default) only reaches "connected". When this is
  // fixed, this test should be updated.
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                 caller()->ice_connection_state(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 callee()->ice_connection_state(), kDefaultTimeout);
}

constexpr int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN |
                                cricket::PORTALLOCATOR_DISABLE_RELAY |
                                cricket::PORTALLOCATOR_DISABLE_TCP;

// Use a mock resolver to resolve the hostname back to the original IP on both
// sides and check that the ICE connection connects.
TEST_P(PeerConnectionIntegrationTest,
       IceStatesReachCompletionWithRemoteHostname) {
  auto caller_resolver_factory =
      std::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
  auto callee_resolver_factory =
      std::make_unique<NiceMock<webrtc::MockAsyncResolverFactory>>();
  NiceMock<rtc::MockAsyncResolver> callee_async_resolver;
  NiceMock<rtc::MockAsyncResolver> caller_async_resolver;

  // This also verifies that the injected AsyncResolverFactory is used by
  // P2PTransportChannel.
  EXPECT_CALL(*caller_resolver_factory, Create())
      .WillOnce(Return(&caller_async_resolver));
  webrtc::PeerConnectionDependencies caller_deps(nullptr);
  caller_deps.async_resolver_factory = std::move(caller_resolver_factory);

  EXPECT_CALL(*callee_resolver_factory, Create())
      .WillOnce(Return(&callee_async_resolver));
  webrtc::PeerConnectionDependencies callee_deps(nullptr);
  callee_deps.async_resolver_factory = std::move(callee_resolver_factory);

  PeerConnectionInterface::RTCConfiguration config;
  config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;

  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
      config, std::move(caller_deps), config, std::move(callee_deps)));

  caller()->SetRemoteAsyncResolver(&callee_async_resolver);
  callee()->SetRemoteAsyncResolver(&caller_async_resolver);

  // Enable hostname candidates with mDNS names.
  caller()->SetMdnsResponder(
      std::make_unique<webrtc::FakeMdnsResponder>(network_thread()));
  callee()->SetMdnsResponder(
      std::make_unique<webrtc::FakeMdnsResponder>(network_thread()));

  SetPortAllocatorFlags(kOnlyLocalPorts, kOnlyLocalPorts);

  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                 caller()->ice_connection_state(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 callee()->ice_connection_state(), kDefaultTimeout);

  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
                          "WebRTC.PeerConnection.CandidatePairType_UDP",
                          webrtc::kIceCandidatePairHostNameHostName));
  DestroyPeerConnections();
}

// Test that firewalling the ICE connection causes the clients to identify the
// disconnected state and then removing the firewall causes them to reconnect.
class PeerConnectionIntegrationIceStatesTest
    : public PeerConnectionIntegrationBaseTest,
      public ::testing::WithParamInterface<
          std::tuple<SdpSemantics, std::tuple<std::string, uint32_t>>> {
 protected:
  PeerConnectionIntegrationIceStatesTest()
      : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam())) {
    port_allocator_flags_ = std::get<1>(std::get<1>(GetParam()));
  }

  void StartStunServer(const SocketAddress& server_address) {
    stun_server_.reset(
        cricket::TestStunServer::Create(firewall(), server_address));
  }

  bool TestIPv6() {
    return (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6);
  }

  void SetPortAllocatorFlags() {
    PeerConnectionIntegrationBaseTest::SetPortAllocatorFlags(
        port_allocator_flags_, port_allocator_flags_);
  }

  std::vector<SocketAddress> CallerAddresses() {
    std::vector<SocketAddress> addresses;
    addresses.push_back(SocketAddress("1.1.1.1", 0));
    if (TestIPv6()) {
      addresses.push_back(SocketAddress("1111:0:a:b:c:d:e:f", 0));
    }
    return addresses;
  }

  std::vector<SocketAddress> CalleeAddresses() {
    std::vector<SocketAddress> addresses;
    addresses.push_back(SocketAddress("2.2.2.2", 0));
    if (TestIPv6()) {
      addresses.push_back(SocketAddress("2222:0:a:b:c:d:e:f", 0));
    }
    return addresses;
  }

  void SetUpNetworkInterfaces() {
    // Remove the default interfaces added by the test infrastructure.
    caller()->network_manager()->RemoveInterface(kDefaultLocalAddress);
    callee()->network_manager()->RemoveInterface(kDefaultLocalAddress);

    // Add network addresses for test.
    for (const auto& caller_address : CallerAddresses()) {
      caller()->network_manager()->AddInterface(caller_address);
    }
    for (const auto& callee_address : CalleeAddresses()) {
      callee()->network_manager()->AddInterface(callee_address);
    }
  }

 private:
  uint32_t port_allocator_flags_;
  std::unique_ptr<cricket::TestStunServer> stun_server_;
};

// Ensure FakeClockForTest is constructed first (see class for rationale).
class PeerConnectionIntegrationIceStatesTestWithFakeClock
    : public FakeClockForTest,
      public PeerConnectionIntegrationIceStatesTest {};

#if !defined(THREAD_SANITIZER)
// This test provokes TSAN errors. bugs.webrtc.org/11282

// Tests that if the connection doesn't get set up properly we eventually reach
// the "failed" iceConnectionState.
TEST_P(PeerConnectionIntegrationIceStatesTestWithFakeClock,
       IceStateSetupFailure) {
  // Block connections to/from the caller and wait for ICE to become
  // disconnected.
  for (const auto& caller_address : CallerAddresses()) {
    firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
  }

  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  SetPortAllocatorFlags();
  SetUpNetworkInterfaces();
  caller()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();

  // According to RFC7675, if there is no response within 30 seconds then the
  // peer should consider the other side to have rejected the connection. This
  // is signaled by the state transitioning to "failed".
  constexpr int kConsentTimeout = 30000;
  ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed,
                           caller()->standardized_ice_connection_state(),
                           kConsentTimeout, FakeClock());
}

#endif  // !defined(THREAD_SANITIZER)

// Tests that the best connection is set to the appropriate IPv4/IPv6 connection
// and that the statistics in the metric observers are updated correctly.
// TODO(bugs.webrtc.org/12591): Flaky on Windows.
#if defined(WEBRTC_WIN)
#define MAYBE_VerifyBestConnection DISABLED_VerifyBestConnection
#else
#define MAYBE_VerifyBestConnection VerifyBestConnection
#endif
TEST_P(PeerConnectionIntegrationIceStatesTest, MAYBE_VerifyBestConnection) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  SetPortAllocatorFlags();
  SetUpNetworkInterfaces();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();

  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                 caller()->ice_connection_state(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 callee()->ice_connection_state(), kDefaultTimeout);

  // TODO(bugs.webrtc.org/9456): Fix it.
  const int num_best_ipv4 = webrtc::metrics::NumEvents(
      "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
  const int num_best_ipv6 = webrtc::metrics::NumEvents(
      "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
  if (TestIPv6()) {
    // When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
    // connection.
    EXPECT_METRIC_EQ(0, num_best_ipv4);
    EXPECT_METRIC_EQ(1, num_best_ipv6);
  } else {
    EXPECT_METRIC_EQ(1, num_best_ipv4);
    EXPECT_METRIC_EQ(0, num_best_ipv6);
  }

  EXPECT_METRIC_EQ(0, webrtc::metrics::NumEvents(
                          "WebRTC.PeerConnection.CandidatePairType_UDP",
                          webrtc::kIceCandidatePairHostHost));
  EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
                          "WebRTC.PeerConnection.CandidatePairType_UDP",
                          webrtc::kIceCandidatePairHostPublicHostPublic));
}

constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |
                                      cricket::PORTALLOCATOR_DISABLE_STUN |
                                      cricket::PORTALLOCATOR_DISABLE_RELAY;
constexpr uint32_t kFlagsIPv6NoStun =
    cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_STUN |
    cricket::PORTALLOCATOR_ENABLE_IPV6 | cricket::PORTALLOCATOR_DISABLE_RELAY;
constexpr uint32_t kFlagsIPv4Stun =
    cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_RELAY;

INSTANTIATE_TEST_SUITE_P(
    PeerConnectionIntegrationTest,
    PeerConnectionIntegrationIceStatesTest,
    Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan),
            Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun),
                   std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun),
                   std::make_pair("IPv4 with STUN", kFlagsIPv4Stun))));

INSTANTIATE_TEST_SUITE_P(
    PeerConnectionIntegrationTest,
    PeerConnectionIntegrationIceStatesTestWithFakeClock,
    Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan),
            Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun),
                   std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun),
                   std::make_pair("IPv4 with STUN", kFlagsIPv4Stun))));

// This test sets up a call between two parties with audio and video.
// During the call, the caller restarts ICE and the test verifies that
// new ICE candidates are generated and audio and video still can flow, and the
// ICE state reaches completed again.
TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Do normal offer/answer and wait for ICE to complete.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                 caller()->ice_connection_state(), kMaxWaitForFramesMs);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 callee()->ice_connection_state(), kMaxWaitForFramesMs);

  // To verify that the ICE restart actually occurs, get
  // ufrag/password/candidates before and after restart.
  // Create an SDP string of the first audio candidate for both clients.
  const webrtc::IceCandidateCollection* audio_candidates_caller =
      caller()->pc()->local_description()->candidates(0);
  const webrtc::IceCandidateCollection* audio_candidates_callee =
      callee()->pc()->local_description()->candidates(0);
  ASSERT_GT(audio_candidates_caller->count(), 0u);
  ASSERT_GT(audio_candidates_callee->count(), 0u);
  std::string caller_candidate_pre_restart;
  ASSERT_TRUE(
      audio_candidates_caller->at(0)->ToString(&caller_candidate_pre_restart));
  std::string callee_candidate_pre_restart;
  ASSERT_TRUE(
      audio_candidates_callee->at(0)->ToString(&callee_candidate_pre_restart));
  const cricket::SessionDescription* desc =
      caller()->pc()->local_description()->description();
  std::string caller_ufrag_pre_restart =
      desc->transport_infos()[0].description.ice_ufrag;
  desc = callee()->pc()->local_description()->description();
  std::string callee_ufrag_pre_restart =
      desc->transport_infos()[0].description.ice_ufrag;

  EXPECT_EQ(caller()->ice_candidate_pair_change_history().size(), 1u);
  // Have the caller initiate an ICE restart.
  caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                 caller()->ice_connection_state(), kMaxWaitForFramesMs);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 callee()->ice_connection_state(), kMaxWaitForFramesMs);

  // Grab the ufrags/candidates again.
  audio_candidates_caller = caller()->pc()->local_description()->candidates(0);
  audio_candidates_callee = callee()->pc()->local_description()->candidates(0);
  ASSERT_GT(audio_candidates_caller->count(), 0u);
  ASSERT_GT(audio_candidates_callee->count(), 0u);
  std::string caller_candidate_post_restart;
  ASSERT_TRUE(
      audio_candidates_caller->at(0)->ToString(&caller_candidate_post_restart));
  std::string callee_candidate_post_restart;
  ASSERT_TRUE(
      audio_candidates_callee->at(0)->ToString(&callee_candidate_post_restart));
  desc = caller()->pc()->local_description()->description();
  std::string caller_ufrag_post_restart =
      desc->transport_infos()[0].description.ice_ufrag;
  desc = callee()->pc()->local_description()->description();
  std::string callee_ufrag_post_restart =
      desc->transport_infos()[0].description.ice_ufrag;
  // Sanity check that an ICE restart was actually negotiated in SDP.
  ASSERT_NE(caller_candidate_pre_restart, caller_candidate_post_restart);
  ASSERT_NE(callee_candidate_pre_restart, callee_candidate_post_restart);
  ASSERT_NE(caller_ufrag_pre_restart, caller_ufrag_post_restart);
  ASSERT_NE(callee_ufrag_pre_restart, callee_ufrag_post_restart);
  EXPECT_GT(caller()->ice_candidate_pair_change_history().size(), 1u);

  // Ensure that additional frames are received after the ICE restart.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// Verify that audio/video can be received end-to-end when ICE renomination is
// enabled.
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
  PeerConnectionInterface::RTCConfiguration config;
  config.enable_ice_renomination = true;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  // Do normal offer/answer and wait for some frames to be received in each
  // direction.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Sanity check that ICE renomination was actually negotiated.
  const cricket::SessionDescription* desc =
      caller()->pc()->local_description()->description();
  for (const cricket::TransportInfo& info : desc->transport_infos()) {
    ASSERT_THAT(info.description.transport_options, Contains("renomination"));
  }
  desc = callee()->pc()->local_description()->description();
  for (const cricket::TransportInfo& info : desc->transport_infos()) {
    ASSERT_THAT(info.description.transport_options, Contains("renomination"));
  }
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// With a max bundle policy and RTCP muxing, adding a new media description to
// the connection should not affect ICE at all because the new media will use
// the existing connection.
// TODO(bugs.webrtc.org/12538): Fails on tsan.
#if defined(THREAD_SANITIZER)
#define MAYBE_AddMediaToConnectedBundleDoesNotRestartIce \
  DISABLED_AddMediaToConnectedBundleDoesNotRestartIce
#else
#define MAYBE_AddMediaToConnectedBundleDoesNotRestartIce \
  AddMediaToConnectedBundleDoesNotRestartIce
#endif
TEST_P(PeerConnectionIntegrationTest,
       MAYBE_AddMediaToConnectedBundleDoesNotRestartIce) {
  PeerConnectionInterface::RTCConfiguration config;
  config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(
      config, PeerConnectionInterface::RTCConfiguration()));
  ConnectFakeSignaling();

  caller()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
                 caller()->ice_connection_state(), kDefaultTimeout);

  caller()->clear_ice_connection_state_history();

  caller()->AddVideoTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  EXPECT_EQ(0u, caller()->ice_connection_state_history().size());
}

// This test sets up a call between two parties with audio and video. It then
// renegotiates setting the video m-line to "port 0", then later renegotiates
// again, enabling video.
TEST_P(PeerConnectionIntegrationTest,
       VideoFlowsAfterMediaSectionIsRejectedAndRecycled) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  // Do initial negotiation, only sending media from the caller. Will result in
  // video and audio recvonly "m=" sections.
  caller()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Negotiate again, disabling the video "m=" section (the callee will set the
  // port to 0 due to offer_to_receive_video = 0).
  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_video = 0;
    callee()->SetOfferAnswerOptions(options);
  } else {
    callee()->SetRemoteOfferHandler([this] {
      callee()
          ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
          ->StopInternal();
    });
  }
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Sanity check that video "m=" section was actually rejected.
  const ContentInfo* answer_video_content = cricket::GetFirstVideoContent(
      callee()->pc()->local_description()->description());
  ASSERT_NE(nullptr, answer_video_content);
  ASSERT_TRUE(answer_video_content->rejected);

  // Enable video and do negotiation again, making sure video is received
  // end-to-end, also adding media stream to callee.
  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    PeerConnectionInterface::RTCOfferAnswerOptions options;
    options.offer_to_receive_video = 1;
    callee()->SetOfferAnswerOptions(options);
  } else {
    // The caller's transceiver is stopped, so we need to add another track.
    auto caller_transceiver =
        caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
    EXPECT_EQ(nullptr, caller_transceiver.get());
    caller()->AddVideoTrack();
  }
  callee()->AddVideoTrack();
  callee()->SetRemoteOfferHandler(nullptr);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Verify the caller receives frames from the newly added stream, and the
  // callee receives additional frames from the re-enabled video m= section.
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudio();
  media_expectations.ExpectBidirectionalVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// This tests that if we negotiate after calling CreateSender but before we
// have a track, then set a track later, frames from the newly-set track are
// received end-to-end.
TEST_F(PeerConnectionIntegrationTestPlanB,
       MediaFlowsAfterEarlyWarmupWithCreateSender) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  auto caller_audio_sender =
      caller()->pc()->CreateSender("audio", "caller_stream");
  auto caller_video_sender =
      caller()->pc()->CreateSender("video", "caller_stream");
  auto callee_audio_sender =
      callee()->pc()->CreateSender("audio", "callee_stream");
  auto callee_video_sender =
      callee()->pc()->CreateSender("video", "callee_stream");
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
  // Wait for ICE to complete, without any tracks being set.
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                 caller()->ice_connection_state(), kMaxWaitForFramesMs);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 callee()->ice_connection_state(), kMaxWaitForFramesMs);
  // Now set the tracks, and expect frames to immediately start flowing.
  EXPECT_TRUE(
      caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack().get()));
  EXPECT_TRUE(
      caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack().get()));
  EXPECT_TRUE(
      callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack().get()));
  EXPECT_TRUE(
      callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack().get()));
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// This tests that if we negotiate after calling AddTransceiver but before we
// have a track, then set a track later, frames from the newly-set tracks are
// received end-to-end.
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       MediaFlowsAfterEarlyWarmupWithAddTransceiver) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  auto audio_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
  ASSERT_EQ(RTCErrorType::NONE, audio_result.error().type());
  auto caller_audio_sender = audio_result.MoveValue()->sender();
  auto video_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
  ASSERT_EQ(RTCErrorType::NONE, video_result.error().type());
  auto caller_video_sender = video_result.MoveValue()->sender();
  callee()->SetRemoteOfferHandler([this] {
    ASSERT_EQ(2u, callee()->pc()->GetTransceivers().size());
    callee()->pc()->GetTransceivers()[0]->SetDirectionWithError(
        RtpTransceiverDirection::kSendRecv);
    callee()->pc()->GetTransceivers()[1]->SetDirectionWithError(
        RtpTransceiverDirection::kSendRecv);
  });
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
  // Wait for ICE to complete, without any tracks being set.
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
                 caller()->ice_connection_state(), kMaxWaitForFramesMs);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 callee()->ice_connection_state(), kMaxWaitForFramesMs);
  // Now set the tracks, and expect frames to immediately start flowing.
  auto callee_audio_sender = callee()->pc()->GetSenders()[0];
  auto callee_video_sender = callee()->pc()->GetSenders()[1];
  ASSERT_TRUE(
      caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack().get()));
  ASSERT_TRUE(
      caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack().get()));
  ASSERT_TRUE(
      callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack().get()));
  ASSERT_TRUE(
      callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack().get()));
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// This test verifies that a remote video track can be added via AddStream,
// and sent end-to-end. For this particular test, it's simply echoed back
// from the caller to the callee, rather than being forwarded to a third
// PeerConnection.
TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  // Just send a video track from the caller.
  caller()->AddVideoTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
  ASSERT_EQ(1U, callee()->remote_streams()->count());

  // Echo the stream back, and do a new offer/anwer (initiated by callee this
  // time).
  callee()->pc()->AddStream(callee()->remote_streams()->at(0));
  callee()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);

  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

#if !defined(THREAD_SANITIZER)
// This test provokes TSAN errors. bugs.webrtc.org/11282

// Test that we achieve the expected end-to-end connection time, using a
// fake clock and simulated latency on the media and signaling paths.
// We use a TURN<->TURN connection because this is usually the quickest to
// set up initially, especially when we're confident the connection will work
// and can start sending media before we get a STUN response.
//
// With various optimizations enabled, here are the network delays we expect to
// be on the critical path:
// 1. 2 signaling trips: Signaling offer and offerer's TURN candidate, then
//                       signaling answer (with DTLS fingerprint).
// 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when
//                  using TURN<->TURN pair, and DTLS exchange is 4 packets,
//                  the first of which should have arrived before the answer.
TEST_P(PeerConnectionIntegrationTestWithFakeClock,
       EndToEndConnectionTimeWithTurnTurnPair) {
  static constexpr int media_hop_delay_ms = 50;
  static constexpr int signaling_trip_delay_ms = 500;
  // For explanation of these values, see comment above.
  static constexpr int required_media_hops = 9;
  static constexpr int required_signaling_trips = 2;
  // For internal delays (such as posting an event asychronously).
  static constexpr int allowed_internal_delay_ms = 20;
  static constexpr int total_connection_time_ms =
      media_hop_delay_ms * required_media_hops +
      signaling_trip_delay_ms * required_signaling_trips +
      allowed_internal_delay_ms;

  static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
                                                                 3478};
  static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
                                                                 0};
  static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
                                                                 3478};
  static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
                                                                 0};
  cricket::TestTurnServer* turn_server_1 = CreateTurnServer(
      turn_server_1_internal_address, turn_server_1_external_address);

  cricket::TestTurnServer* turn_server_2 = CreateTurnServer(
      turn_server_2_internal_address, turn_server_2_external_address);
  // Bypass permission check on received packets so media can be sent before
  // the candidate is signaled.
  SendTask(network_thread(), [turn_server_1] {
    turn_server_1->set_enable_permission_checks(false);
  });
  SendTask(network_thread(), [turn_server_2] {
    turn_server_2->set_enable_permission_checks(false);
  });

  PeerConnectionInterface::RTCConfiguration client_1_config;
  webrtc::PeerConnectionInterface::IceServer ice_server_1;
  ice_server_1.urls.push_back("turn:88.88.88.0:3478");
  ice_server_1.username = "test";
  ice_server_1.password = "test";
  client_1_config.servers.push_back(ice_server_1);
  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
  client_1_config.presume_writable_when_fully_relayed = true;

  PeerConnectionInterface::RTCConfiguration client_2_config;
  webrtc::PeerConnectionInterface::IceServer ice_server_2;
  ice_server_2.urls.push_back("turn:99.99.99.0:3478");
  ice_server_2.username = "test";
  ice_server_2.password = "test";
  client_2_config.servers.push_back(ice_server_2);
  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
  client_2_config.presume_writable_when_fully_relayed = true;

  ASSERT_TRUE(
      CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
  // Set up the simulated delays.
  SetSignalingDelayMs(signaling_trip_delay_ms);
  ConnectFakeSignaling();
  virtual_socket_server()->set_delay_mean(media_hop_delay_ms);
  virtual_socket_server()->UpdateDelayDistribution();

  // Set "offer to receive audio/video" without adding any tracks, so we just
  // set up ICE/DTLS with no media.
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_audio = 1;
  options.offer_to_receive_video = 1;
  caller()->SetOfferAnswerOptions(options);
  caller()->CreateAndSetAndSignalOffer();
  EXPECT_TRUE_SIMULATED_WAIT(DtlsConnected(), total_connection_time_ms,
                             FakeClock());
  // Closing the PeerConnections destroys the ports before the ScopedFakeClock.
  // If this is not done a DCHECK can be hit in ports.cc, because a large
  // negative number is calculated for the rtt due to the global clock changing.
  ClosePeerConnections();
}

TEST_P(PeerConnectionIntegrationTestWithFakeClock,
       OnIceCandidateFlushesGetStatsCache) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioTrack();

  // Call getStats, assert there are no candidates.
  rtc::scoped_refptr<const webrtc::RTCStatsReport> first_report =
      caller()->NewGetStats();
  ASSERT_TRUE(first_report);
  auto first_candidate_stats =
      first_report->GetStatsOfType<webrtc::RTCLocalIceCandidateStats>();
  ASSERT_EQ(first_candidate_stats.size(), 0u);

  // Create an offer at the caller and set it as remote description on the
  // callee.
  caller()->CreateAndSetAndSignalOffer();
  // Call getStats again, assert there are candidates now.
  rtc::scoped_refptr<const webrtc::RTCStatsReport> second_report =
      caller()->NewGetStats();
  ASSERT_TRUE(second_report);
  auto second_candidate_stats =
      second_report->GetStatsOfType<webrtc::RTCLocalIceCandidateStats>();
  ASSERT_NE(second_candidate_stats.size(), 0u);

  // The fake clock ensures that no time has passed so the cache must have been
  // explicitly invalidated.
  EXPECT_EQ(first_report->timestamp_us(), second_report->timestamp_us());
}

TEST_P(PeerConnectionIntegrationTestWithFakeClock,
       AddIceCandidateFlushesGetStatsCache) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignalingForSdpOnly();
  caller()->AddAudioTrack();

  // Start candidate gathering and wait for it to complete. Candidates are not
  // signalled.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_SIMULATED_WAIT(caller()->IceGatheringStateComplete(),
                             kDefaultTimeout, FakeClock());

  // Call getStats, assert there are no candidates.
  rtc::scoped_refptr<const webrtc::RTCStatsReport> first_report =
      caller()->NewGetStats();
  ASSERT_TRUE(first_report);
  auto first_candidate_stats =
      first_report->GetStatsOfType<webrtc::RTCRemoteIceCandidateStats>();
  ASSERT_EQ(first_candidate_stats.size(), 0u);

  // Add a "fake" candidate.
  absl::optional<RTCError> result;
  caller()->pc()->AddIceCandidate(
      absl::WrapUnique(webrtc::CreateIceCandidate(
          "", 0,
          "candidate:2214029314 1 udp 2122260223 127.0.0.1 49152 typ host",
          nullptr)),
      [&result](RTCError r) { result = r; });
  ASSERT_TRUE_WAIT(result.has_value(), kDefaultTimeout);
  ASSERT_TRUE(result.value().ok());

  // Call getStats again, assert there is a remote candidate now.
  rtc::scoped_refptr<const webrtc::RTCStatsReport> second_report =
      caller()->NewGetStats();
  ASSERT_TRUE(second_report);
  auto second_candidate_stats =
      second_report->GetStatsOfType<webrtc::RTCRemoteIceCandidateStats>();
  ASSERT_EQ(second_candidate_stats.size(), 1u);

  // The fake clock ensures that no time has passed so the cache must have been
  // explicitly invalidated.
  EXPECT_EQ(first_report->timestamp_us(), second_report->timestamp_us());
}

#endif  // !defined(THREAD_SANITIZER)

// Verify that a TurnCustomizer passed in through RTCConfiguration
// is actually used by the underlying TURN candidate pair.
// Note that turnport_unittest.cc contains more detailed, lower-level tests.
TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) {
  static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
                                                                 3478};
  static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
                                                                 0};
  static const rtc::SocketAddress turn_server_2_internal_address{"99.99.99.0",
                                                                 3478};
  static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
                                                                 0};
  CreateTurnServer(turn_server_1_internal_address,
                   turn_server_1_external_address);
  CreateTurnServer(turn_server_2_internal_address,
                   turn_server_2_external_address);

  PeerConnectionInterface::RTCConfiguration client_1_config;
  webrtc::PeerConnectionInterface::IceServer ice_server_1;
  ice_server_1.urls.push_back("turn:88.88.88.0:3478");
  ice_server_1.username = "test";
  ice_server_1.password = "test";
  client_1_config.servers.push_back(ice_server_1);
  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
  auto* customizer1 = CreateTurnCustomizer();
  client_1_config.turn_customizer = customizer1;

  PeerConnectionInterface::RTCConfiguration client_2_config;
  webrtc::PeerConnectionInterface::IceServer ice_server_2;
  ice_server_2.urls.push_back("turn:99.99.99.0:3478");
  ice_server_2.username = "test";
  ice_server_2.password = "test";
  client_2_config.servers.push_back(ice_server_2);
  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
  auto* customizer2 = CreateTurnCustomizer();
  client_2_config.turn_customizer = customizer2;

  ASSERT_TRUE(
      CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
  ConnectFakeSignaling();

  // Set "offer to receive audio/video" without adding any tracks, so we just
  // set up ICE/DTLS with no media.
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_audio = 1;
  options.offer_to_receive_video = 1;
  caller()->SetOfferAnswerOptions(options);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);

  ExpectTurnCustomizerCountersIncremented(customizer1);
  ExpectTurnCustomizerCountersIncremented(customizer2);
}

// Verifies that you can use TCP instead of UDP to connect to a TURN server and
// send media between the caller and the callee.
TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
  static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                               3478};
  static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};

  // Enable TCP for the fake turn server.
  CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                   cricket::PROTO_TCP);

  webrtc::PeerConnectionInterface::IceServer ice_server;
  ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
  ice_server.username = "test";
  ice_server.password = "test";

  PeerConnectionInterface::RTCConfiguration client_1_config;
  client_1_config.servers.push_back(ice_server);
  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;

  PeerConnectionInterface::RTCConfiguration client_2_config;
  client_2_config.servers.push_back(ice_server);
  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;

  ASSERT_TRUE(
      CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));

  // Do normal offer/answer and wait for ICE to complete.
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 callee()->ice_connection_state(), kMaxWaitForFramesMs);

  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  EXPECT_TRUE(ExpectNewFrames(media_expectations));
}

// Verify that a SSLCertificateVerifier passed in through
// PeerConnectionDependencies is actually used by the underlying SSL
// implementation to determine whether a certificate presented by the TURN
// server is accepted by the client. Note that openssladapter_unittest.cc
// contains more detailed, lower-level tests.
TEST_P(PeerConnectionIntegrationTest,
       SSLCertificateVerifierUsedForTurnConnections) {
  static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                               3478};
  static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};

  // Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
  // that host name verification passes on the fake certificate.
  CreateTurnServer(turn_server_internal_address, turn_server_external_address,
                   cricket::PROTO_TLS, "88.88.88.0");

  webrtc::PeerConnectionInterface::IceServer ice_server;
  ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
  ice_server.username = "test";
  ice_server.password = "test";

  PeerConnectionInterface::RTCConfiguration client_1_config;
  client_1_config.servers.push_back(ice_server);
  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;

  PeerConnectionInterface::RTCConfiguration client_2_config;
  client_2_config.servers.push_back(ice_server);
  // Setting the type to kRelay forces the connection to go through a TURN
  // server.
  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;

  // Get a copy to the pointer so we can verify calls later.
  rtc::TestCertificateVerifier* client_1_cert_verifier =
      new rtc::TestCertificateVerifier();
  client_1_cert_verifier->verify_certificate_ = true;
  rtc::TestCertificateVerifier* client_2_cert_verifier =
      new rtc::TestCertificateVerifier();
  client_2_cert_verifier->verify_certificate_ = true;

  // Create the dependencies with the test certificate verifier.
  webrtc::PeerConnectionDependencies client_1_deps(nullptr);
  client_1_deps.tls_cert_verifier =
      std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
  webrtc::PeerConnectionDependencies client_2_deps(nullptr);
  client_2_deps.tls_cert_verifier =
      std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);

  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
      client_1_config, std::move(client_1_deps), client_2_config,
      std::move(client_2_deps)));
  ConnectFakeSignaling();

  // Set "offer to receive audio/video" without adding any tracks, so we just
  // set up ICE/DTLS with no media.
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_audio = 1;
  options.offer_to_receive_video = 1;
  caller()->SetOfferAnswerOptions(options);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);

  EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
  EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
}

// Test that the injected ICE transport factory is used to create ICE transports
// for WebRTC connections.
TEST_P(PeerConnectionIntegrationTest, IceTransportFactoryUsedForConnections) {
  PeerConnectionInterface::RTCConfiguration default_config;
  PeerConnectionDependencies dependencies(nullptr);
  auto ice_transport_factory = std::make_unique<MockIceTransportFactory>();
  EXPECT_CALL(*ice_transport_factory, RecordIceTransportCreated()).Times(1);
  dependencies.ice_transport_factory = std::move(ice_transport_factory);
  auto wrapper = CreatePeerConnectionWrapper("Caller", nullptr, &default_config,
                                             std::move(dependencies), nullptr,
                                             /*reset_encoder_factory=*/false,
                                             /*reset_decoder_factory=*/false);
  ASSERT_TRUE(wrapper);
  wrapper->CreateDataChannel();
  auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
  wrapper->pc()->SetLocalDescription(observer.get(),
                                     wrapper->CreateOfferAndWait().release());
}

// Test that audio and video flow end-to-end when codec names don't use the
// expected casing, given that they're supposed to be case insensitive. To test
// this, all but one codec is removed from each media description, and its
// casing is changed.
//
// In the past, this has regressed and caused crashes/black video, due to the
// fact that code at some layers was doing case-insensitive comparisons and
// code at other layers was not.
TEST_P(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();

  // Remove all but one audio/video codec (opus and VP8), and change the
  // casing of the caller's generated offer.
  caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* description) {
    cricket::AudioContentDescription* audio =
        GetFirstAudioContentDescription(description);
    ASSERT_NE(nullptr, audio);
    auto audio_codecs = audio->codecs();
    audio_codecs.erase(std::remove_if(audio_codecs.begin(), audio_codecs.end(),
                                      [](const cricket::AudioCodec& codec) {
                                        return codec.name != "opus";
                                      }),
                       audio_codecs.end());
    ASSERT_EQ(1u, audio_codecs.size());
    audio_codecs[0].name = "OpUs";
    audio->set_codecs(audio_codecs);

    cricket::VideoContentDescription* video =
        GetFirstVideoContentDescription(description);
    ASSERT_NE(nullptr, video);
    auto video_codecs = video->codecs();
    video_codecs.erase(std::remove_if(video_codecs.begin(), video_codecs.end(),
                                      [](const cricket::VideoCodec& codec) {
                                        return codec.name != "VP8";
                                      }),
                       video_codecs.end());
    ASSERT_EQ(1u, video_codecs.size());
    video_codecs[0].name = "vP8";
    video->set_codecs(video_codecs);
  });

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Verify frames are still received end-to-end.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationTest, GetSourcesAudio) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Wait for one audio frame to be received by the callee.
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudio(1);
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
  ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
  auto receiver = callee()->pc()->GetReceivers()[0];
  ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
  auto sources = receiver->GetSources();
  ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
  EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
            sources[0].source_id());
  EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
}

TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddVideoTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Wait for one video frame to be received by the callee.
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeVideo(1);
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
  ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
  auto receiver = callee()->pc()->GetReceivers()[0];
  ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_VIDEO);
  auto sources = receiver->GetSources();
  ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
  ASSERT_GT(sources.size(), 0u);
  EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
            sources[0].source_id());
  EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
}

// Test that if a track is removed and added again with a different stream ID,
// the new stream ID is successfully communicated in SDP and media continues to
// flow end-to-end.
// TODO(webrtc.bugs.org/8734): This test does not work for Unified Plan because
// it will not reuse a transceiver that has already been sending. After creating
// a new transceiver it tries to create an offer with two senders of the same
// track ids and it fails.
TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  // Add track using stream 1, do offer/answer.
  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
      caller()->CreateLocalAudioTrack();
  rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
      caller()->AddTrack(track, {"stream_1"});
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  {
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio(1);
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
  // Remove the sender, and create a new one with the new stream.
  caller()->pc()->RemoveTrackOrError(sender);
  sender = caller()->AddTrack(track, {"stream_2"});
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Wait for additional audio frames to be received by the callee.
  {
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
}

TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  auto output = std::make_unique<testing::NiceMock<MockRtcEventLogOutput>>();
  ON_CALL(*output, IsActive()).WillByDefault(::testing::Return(true));
  ON_CALL(*output, Write(::testing::A<absl::string_view>()))
      .WillByDefault(::testing::Return(true));
  EXPECT_CALL(*output, Write(::testing::A<absl::string_view>()))
      .Times(::testing::AtLeast(1));
  EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
      std::move(output), webrtc::RtcEventLog::kImmediateOutput));

  caller()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
}

// Test that if candidates are only signaled by applying full session
// descriptions (instead of using AddIceCandidate), the peers can connect to
// each other and exchange media.
TEST_P(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  // Each side will signal the session descriptions but not candidates.
  ConnectFakeSignalingForSdpOnly();

  // Add audio video track and exchange the initial offer/answer with media
  // information only. This will start ICE gathering on each side.
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();

  // Wait for all candidates to be gathered on both the caller and callee.
  ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
                 caller()->ice_gathering_state(), kDefaultTimeout);
  ASSERT_EQ_WAIT(PeerConnectionInterface::kIceGatheringComplete,
                 callee()->ice_gathering_state(), kDefaultTimeout);

  // The candidates will now be included in the session description, so
  // signaling them will start the ICE connection.
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Ensure that media flows in both directions.
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

#if !defined(THREAD_SANITIZER)
// These tests provokes TSAN errors. See bugs.webrtc.org/11305.

// Test that SetAudioPlayout can be used to disable audio playout from the
// start, then later enable it. This may be useful, for example, if the caller
// needs to play a local ringtone until some event occurs, after which it
// switches to playing the received audio.
TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  // Set up audio-only call where audio playout is disabled on caller's side.
  caller()->pc()->SetAudioPlayout(false);
  caller()->AddAudioTrack();
  callee()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Pump messages for a second.
  WAIT(false, 1000);
  // Since audio playout is disabled, the caller shouldn't have received
  // anything (at the playout level, at least).
  EXPECT_EQ(0, caller()->audio_frames_received());
  // As a sanity check, make sure the callee (for which playout isn't disabled)
  // did still see frames on its audio level.
  ASSERT_GT(callee()->audio_frames_received(), 0);

  // Enable playout again, and ensure audio starts flowing.
  caller()->pc()->SetAudioPlayout(true);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudio();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

double GetAudioEnergyStat(PeerConnectionIntegrationWrapper* pc) {
  auto report = pc->NewGetStats();
  auto inbound_rtps =
      report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
  RTC_CHECK(!inbound_rtps.empty());
  auto* inbound_rtp = inbound_rtps[0];
  if (!inbound_rtp->total_audio_energy.is_defined()) {
    return 0.0;
  }
  return *inbound_rtp->total_audio_energy;
}

// Test that if audio playout is disabled via the SetAudioPlayout() method, then
// incoming audio is still processed and statistics are generated.
TEST_P(PeerConnectionIntegrationTest,
       DisableAudioPlayoutStillGeneratesAudioStats) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  // Set up audio-only call where playout is disabled but audio-processing is
  // still active.
  caller()->AddAudioTrack();
  callee()->AddAudioTrack();
  caller()->pc()->SetAudioPlayout(false);

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Wait for the callee to receive audio stats.
  EXPECT_TRUE_WAIT(GetAudioEnergyStat(caller()) > 0, kMaxWaitForFramesMs);
}

#endif  // !defined(THREAD_SANITIZER)

// Test that SetAudioRecording can be used to disable audio recording from the
// start, then later enable it. This may be useful, for example, if the caller
// wants to ensure that no audio resources are active before a certain state
// is reached.
TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();

  // Set up audio-only call where audio recording is disabled on caller's side.
  caller()->pc()->SetAudioRecording(false);
  caller()->AddAudioTrack();
  callee()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Pump messages for a second.
  WAIT(false, 1000);
  // Since caller has disabled audio recording, the callee shouldn't have
  // received anything.
  EXPECT_EQ(0, callee()->audio_frames_received());
  // As a sanity check, make sure the caller did still see frames on its
  // audio level since audio recording is enabled on the calle side.
  ASSERT_GT(caller()->audio_frames_received(), 0);

  // Enable audio recording again, and ensure audio starts flowing.
  caller()->pc()->SetAudioRecording(true);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudio();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationTest,
       IceEventsGeneratedAndLoggedInRtcEventLog) {
  ASSERT_TRUE(CreatePeerConnectionWrappersWithFakeRtcEventLog());
  ConnectFakeSignaling();
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_audio = 1;
  caller()->SetOfferAnswerOptions(options);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
  ASSERT_NE(nullptr, caller()->event_log_factory());
  ASSERT_NE(nullptr, callee()->event_log_factory());
  webrtc::FakeRtcEventLog* caller_event_log =
      caller()->event_log_factory()->last_log_created();
  webrtc::FakeRtcEventLog* callee_event_log =
      callee()->event_log_factory()->last_log_created();
  ASSERT_NE(nullptr, caller_event_log);
  ASSERT_NE(nullptr, callee_event_log);
  int caller_ice_config_count = caller_event_log->GetEventCount(
      webrtc::RtcEvent::Type::IceCandidatePairConfig);
  int caller_ice_event_count = caller_event_log->GetEventCount(
      webrtc::RtcEvent::Type::IceCandidatePairEvent);
  int callee_ice_config_count = callee_event_log->GetEventCount(
      webrtc::RtcEvent::Type::IceCandidatePairConfig);
  int callee_ice_event_count = callee_event_log->GetEventCount(
      webrtc::RtcEvent::Type::IceCandidatePairEvent);
  EXPECT_LT(0, caller_ice_config_count);
  EXPECT_LT(0, caller_ice_event_count);
  EXPECT_LT(0, callee_ice_config_count);
  EXPECT_LT(0, callee_ice_event_count);
}

TEST_P(PeerConnectionIntegrationTest, RegatherAfterChangingIceTransportType) {
  static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                               3478};
  static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};

  CreateTurnServer(turn_server_internal_address, turn_server_external_address);

  webrtc::PeerConnectionInterface::IceServer ice_server;
  ice_server.urls.push_back("turn:88.88.88.0:3478");
  ice_server.username = "test";
  ice_server.password = "test";

  PeerConnectionInterface::RTCConfiguration caller_config;
  caller_config.servers.push_back(ice_server);
  caller_config.type = webrtc::PeerConnectionInterface::kRelay;
  caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
  caller_config.surface_ice_candidates_on_ice_transport_type_changed = true;

  PeerConnectionInterface::RTCConfiguration callee_config;
  callee_config.servers.push_back(ice_server);
  callee_config.type = webrtc::PeerConnectionInterface::kRelay;
  callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;
  callee_config.surface_ice_candidates_on_ice_transport_type_changed = true;

  ASSERT_TRUE(
      CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));

  // Do normal offer/answer and wait for ICE to complete.
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Since we are doing continual gathering, the ICE transport does not reach
  // kIceGatheringComplete (see
  // P2PTransportChannel::OnCandidatesAllocationDone), and consequently not
  // kIceConnectionComplete.
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 caller()->ice_connection_state(), kDefaultTimeout);
  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                 callee()->ice_connection_state(), kDefaultTimeout);
  // Note that we cannot use the metric
  // `WebRTC.PeerConnection.CandidatePairType_UDP` in this test since this
  // metric is only populated when we reach kIceConnectionComplete in the
  // current implementation.
  EXPECT_EQ(cricket::RELAY_PORT_TYPE,
            caller()->last_candidate_gathered().type());
  EXPECT_EQ(cricket::RELAY_PORT_TYPE,
            callee()->last_candidate_gathered().type());

  // Loosen the caller's candidate filter.
  caller_config = caller()->pc()->GetConfiguration();
  caller_config.type = webrtc::PeerConnectionInterface::kAll;
  caller()->pc()->SetConfiguration(caller_config);
  // We should have gathered a new host candidate.
  EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
                 caller()->last_candidate_gathered().type(), kDefaultTimeout);

  // Loosen the callee's candidate filter.
  callee_config = callee()->pc()->GetConfiguration();
  callee_config.type = webrtc::PeerConnectionInterface::kAll;
  callee()->pc()->SetConfiguration(callee_config);
  EXPECT_EQ_WAIT(cricket::LOCAL_PORT_TYPE,
                 callee()->last_candidate_gathered().type(), kDefaultTimeout);

  // Create an offer and verify that it does not contain an ICE restart (i.e new
  // ice credentials).
  std::string caller_ufrag_pre_offer = caller()
                                           ->pc()
                                           ->local_description()
                                           ->description()
                                           ->transport_infos()[0]
                                           .description.ice_ufrag;
  caller()->CreateAndSetAndSignalOffer();
  std::string caller_ufrag_post_offer = caller()
                                            ->pc()
                                            ->local_description()
                                            ->description()
                                            ->transport_infos()[0]
                                            .description.ice_ufrag;
  EXPECT_EQ(caller_ufrag_pre_offer, caller_ufrag_post_offer);
}

TEST_P(PeerConnectionIntegrationTest, OnIceCandidateError) {
  static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
                                                               3478};
  static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};

  CreateTurnServer(turn_server_internal_address, turn_server_external_address);

  webrtc::PeerConnectionInterface::IceServer ice_server;
  ice_server.urls.push_back("turn:88.88.88.0:3478");
  ice_server.username = "test";
  ice_server.password = "123";

  PeerConnectionInterface::RTCConfiguration caller_config;
  caller_config.servers.push_back(ice_server);
  caller_config.type = webrtc::PeerConnectionInterface::kRelay;
  caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;

  PeerConnectionInterface::RTCConfiguration callee_config;
  callee_config.servers.push_back(ice_server);
  callee_config.type = webrtc::PeerConnectionInterface::kRelay;
  callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;

  ASSERT_TRUE(
      CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));

  // Do normal offer/answer and wait for ICE to complete.
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ_WAIT(401, caller()->error_event().error_code, kDefaultTimeout);
  EXPECT_EQ("Unauthorized", caller()->error_event().error_text);
  EXPECT_EQ("turn:88.88.88.0:3478?transport=udp", caller()->error_event().url);
  EXPECT_NE(caller()->error_event().address, "");
}

TEST_P(PeerConnectionIntegrationTest, OnIceCandidateErrorWithEmptyAddress) {
  webrtc::PeerConnectionInterface::IceServer ice_server;
  ice_server.urls.push_back("turn:127.0.0.1:3478?transport=tcp");
  ice_server.username = "test";
  ice_server.password = "test";

  PeerConnectionInterface::RTCConfiguration caller_config;
  caller_config.servers.push_back(ice_server);
  caller_config.type = webrtc::PeerConnectionInterface::kRelay;
  caller_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;

  PeerConnectionInterface::RTCConfiguration callee_config;
  callee_config.servers.push_back(ice_server);
  callee_config.type = webrtc::PeerConnectionInterface::kRelay;
  callee_config.continual_gathering_policy = PeerConnection::GATHER_CONTINUALLY;

  ASSERT_TRUE(
      CreatePeerConnectionWrappersWithConfig(caller_config, callee_config));

  // Do normal offer/answer and wait for ICE to complete.
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ_WAIT(701, caller()->error_event().error_code, kDefaultTimeout);
  EXPECT_EQ(caller()->error_event().address, "");
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       AudioKeepsFlowingAfterImplicitRollback) {
  PeerConnectionInterface::RTCConfiguration config;
  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
  config.enable_implicit_rollback = true;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  caller()->AddAudioTrack();
  callee()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudio();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
  SetSignalIceCandidates(false);  // Workaround candidate outrace sdp.
  caller()->AddVideoTrack();
  callee()->AddVideoTrack();
  auto observer = rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
  callee()->pc()->SetLocalDescription(observer.get(),
                                      callee()->CreateOfferAndWait().release());
  EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
  caller()->CreateAndSetAndSignalOffer();  // Implicit rollback.
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       ImplicitRollbackVisitsStableState) {
  RTCConfiguration config;
  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
  config.enable_implicit_rollback = true;

  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));

  auto sld_observer =
      rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
  callee()->pc()->SetLocalDescription(sld_observer.get(),
                                      callee()->CreateOfferAndWait().release());
  EXPECT_TRUE_WAIT(sld_observer->called(), kDefaultTimeout);
  EXPECT_EQ(sld_observer->error(), "");

  auto srd_observer =
      rtc::make_ref_counted<MockSetSessionDescriptionObserver>();
  callee()->pc()->SetRemoteDescription(
      srd_observer.get(), caller()->CreateOfferAndWait().release());
  EXPECT_TRUE_WAIT(srd_observer->called(), kDefaultTimeout);
  EXPECT_EQ(srd_observer->error(), "");

  EXPECT_THAT(callee()->peer_connection_signaling_state_history(),
              ElementsAre(PeerConnectionInterface::kHaveLocalOffer,
                          PeerConnectionInterface::kStable,
                          PeerConnectionInterface::kHaveRemoteOffer));
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       H264FmtpSpsPpsIdrInKeyframeParameterUsage) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddVideoTrack();
  callee()->AddVideoTrack();
  auto munger = [](cricket::SessionDescription* desc) {
    cricket::VideoContentDescription* video =
        GetFirstVideoContentDescription(desc);
    auto codecs = video->codecs();
    for (auto&& codec : codecs) {
      if (codec.name == "H264") {
        std::string value;
        // The parameter is not supposed to be present in SDP by default.
        EXPECT_FALSE(
            codec.GetParam(cricket::kH264FmtpSpsPpsIdrInKeyframe, &value));
        codec.SetParam(std::string(cricket::kH264FmtpSpsPpsIdrInKeyframe),
                       std::string(""));
      }
    }
    video->set_codecs(codecs);
  };
  // Munge local offer for SLD.
  caller()->SetGeneratedSdpMunger(munger);
  // Munge remote answer for SRD.
  caller()->SetReceivedSdpMunger(munger);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Observe that after munging the parameter is present in generated SDP.
  caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* desc) {
    cricket::VideoContentDescription* video =
        GetFirstVideoContentDescription(desc);
    for (auto&& codec : video->codecs()) {
      if (codec.name == "H264") {
        std::string value;
        EXPECT_TRUE(
            codec.GetParam(cricket::kH264FmtpSpsPpsIdrInKeyframe, &value));
      }
    }
  });
  caller()->CreateOfferAndWait();
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       RenegotiateManyAudioTransceivers) {
  PeerConnectionInterface::RTCConfiguration config;
  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  int current_size = caller()->pc()->GetTransceivers().size();
  // Add more tracks until we get close to having issues.
  // Issues have been seen at:
  // - 32 tracks on android_arm64_rel and android_arm_dbg bots
  // - 16 tracks on android_arm_dbg (flaky)
  while (current_size < 8) {
    // Double the number of tracks
    for (int i = 0; i < current_size; i++) {
      caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
    }
    current_size = caller()->pc()->GetTransceivers().size();
    RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
    auto start_time_ms = rtc::TimeMillis();
    caller()->CreateAndSetAndSignalOffer();
    // We want to stop when the time exceeds one second.
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    auto elapsed_time_ms = rtc::TimeMillis() - start_time_ms;
    RTC_LOG(LS_INFO) << "Renegotiating took " << elapsed_time_ms << " ms";
    ASSERT_GT(1000, elapsed_time_ms)
        << "Audio transceivers: Negotiation took too long after "
        << current_size << " tracks added";
  }
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       RenegotiateManyVideoTransceivers) {
  PeerConnectionInterface::RTCConfiguration config;
  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  int current_size = caller()->pc()->GetTransceivers().size();
  // Add more tracks until we get close to having issues.
  // Issues have been seen at:
  // - 96 on a Linux workstation
  // - 64 at win_x86_more_configs and win_x64_msvc_dbg
  // - 32 on android_arm64_rel and linux_dbg bots
  // - 16 on Android 64 (Nexus 5x)
  while (current_size < 8) {
    // Double the number of tracks
    for (int i = 0; i < current_size; i++) {
      caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
    }
    current_size = caller()->pc()->GetTransceivers().size();
    RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
    auto start_time_ms = rtc::TimeMillis();
    caller()->CreateAndSetAndSignalOffer();
    // We want to stop when the time exceeds one second.
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    auto elapsed_time_ms = rtc::TimeMillis() - start_time_ms;
    RTC_LOG(LS_INFO) << "Renegotiating took " << elapsed_time_ms << " ms";
    ASSERT_GT(1000, elapsed_time_ms)
        << "Video transceivers: Negotiation took too long after "
        << current_size << " tracks added";
  }
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       RenegotiateManyVideoTransceiversAndWatchAudioDelay) {
  PeerConnectionInterface::RTCConfiguration config;
  config.sdp_semantics = SdpSemantics::kUnifiedPlan;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  caller()->AddAudioTrack();
  callee()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  // Wait until we can see the audio flowing.
  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudio();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));

  // Get the baseline numbers for audio_packets and audio_delay
  // in both directions.
  caller()->StartWatchingDelayStats();
  callee()->StartWatchingDelayStats();

  int current_size = caller()->pc()->GetTransceivers().size();
  // Add more tracks until we get close to having issues.
  // Making this number very large makes the test very slow.
  while (current_size < 16) {
    // Double the number of tracks
    for (int i = 0; i < current_size; i++) {
      caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
    }
    current_size = caller()->pc()->GetTransceivers().size();
    RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
    auto start_time_ms = rtc::TimeMillis();
    caller()->CreateAndSetAndSignalOffer();
    // We want to stop when the time exceeds one second.
    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
    auto elapsed_time_ms = rtc::TimeMillis() - start_time_ms;
    RTC_LOG(LS_INFO) << "Renegotiating took " << elapsed_time_ms << " ms";
    // This is a guard against the test using excessive amounts of time.
    ASSERT_GT(5000, elapsed_time_ms)
        << "Video transceivers: Negotiation took too long after "
        << current_size << " tracks added";
    caller()->UpdateDelayStats("caller reception", current_size);
    callee()->UpdateDelayStats("callee reception", current_size);
  }
}

INSTANTIATE_TEST_SUITE_P(
    PeerConnectionIntegrationTest,
    PeerConnectionIntegrationTest,
    Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan),
            Values("WebRTC-FrameBuffer3/arm:FrameBuffer2/",
                   "WebRTC-FrameBuffer3/arm:FrameBuffer3/",
                   "WebRTC-FrameBuffer3/arm:SyncDecoding/")));

INSTANTIATE_TEST_SUITE_P(
    PeerConnectionIntegrationTest,
    PeerConnectionIntegrationTestWithFakeClock,
    Combine(Values(SdpSemantics::kPlanB_DEPRECATED, SdpSemantics::kUnifiedPlan),
            Values("WebRTC-FrameBuffer3/arm:FrameBuffer2/",
                   "WebRTC-FrameBuffer3/arm:FrameBuffer3/",
                   "WebRTC-FrameBuffer3/arm:SyncDecoding/")));

// Tests that verify interoperability between Plan B and Unified Plan
// PeerConnections.
class PeerConnectionIntegrationInteropTest
    : public PeerConnectionIntegrationBaseTest,
      public ::testing::WithParamInterface<
          std::tuple<SdpSemantics, SdpSemantics>> {
 protected:
  // Setting the SdpSemantics for the base test to kDefault does not matter
  // because we specify not to use the test semantics when creating
  // PeerConnectionIntegrationWrappers.
  PeerConnectionIntegrationInteropTest()
      : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB_DEPRECATED),
        caller_semantics_(std::get<0>(GetParam())),
        callee_semantics_(std::get<1>(GetParam())) {}

  bool CreatePeerConnectionWrappersWithSemantics() {
    return CreatePeerConnectionWrappersWithSdpSemantics(caller_semantics_,
                                                        callee_semantics_);
  }

  const SdpSemantics caller_semantics_;
  const SdpSemantics callee_semantics_;
};

TEST_P(PeerConnectionIntegrationInteropTest, NoMediaLocalToNoMediaRemote) {
  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
  ConnectFakeSignaling();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
}

TEST_P(PeerConnectionIntegrationInteropTest, OneAudioLocalToNoMediaRemote) {
  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
  ConnectFakeSignaling();
  auto audio_sender = caller()->AddAudioTrack();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Verify that one audio receiver has been created on the remote and that it
  // has the same track ID as the sending track.
  auto receivers = callee()->pc()->GetReceivers();
  ASSERT_EQ(1u, receivers.size());
  EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, receivers[0]->media_type());
  EXPECT_EQ(receivers[0]->track()->id(), audio_sender->track()->id());

  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudio();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationInteropTest, OneAudioOneVideoToNoMediaRemote) {
  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
  ConnectFakeSignaling();
  auto video_sender = caller()->AddVideoTrack();
  auto audio_sender = caller()->AddAudioTrack();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Verify that one audio and one video receiver have been created on the
  // remote and that they have the same track IDs as the sending tracks.
  auto audio_receivers =
      callee()->GetReceiversOfType(cricket::MEDIA_TYPE_AUDIO);
  ASSERT_EQ(1u, audio_receivers.size());
  EXPECT_EQ(audio_receivers[0]->track()->id(), audio_sender->track()->id());
  auto video_receivers =
      callee()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO);
  ASSERT_EQ(1u, video_receivers.size());
  EXPECT_EQ(video_receivers[0]->track()->id(), video_sender->track()->id());

  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationInteropTest,
       OneAudioOneVideoLocalToOneAudioOneVideoRemote) {
  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  callee()->AddAudioVideoTracks();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  MediaExpectations media_expectations;
  media_expectations.ExpectBidirectionalAudioAndVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationInteropTest,
       ReverseRolesOneAudioLocalToOneVideoRemote) {
  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
  ConnectFakeSignaling();
  caller()->AddAudioTrack();
  callee()->AddVideoTrack();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Verify that only the audio track has been negotiated.
  EXPECT_EQ(0u, caller()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO).size());
  // Might also check that the callee's NegotiationNeeded flag is set.

  // Reverse roles.
  callee()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  MediaExpectations media_expectations;
  media_expectations.CallerExpectsSomeVideo();
  media_expectations.CalleeExpectsSomeAudio();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

TEST_P(PeerConnectionIntegrationTest, NewTracksDoNotCauseNewCandidates) {
  ASSERT_TRUE(CreatePeerConnectionWrappers());
  ConnectFakeSignaling();
  caller()->AddAudioVideoTracks();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
  caller()->ExpectCandidates(0);
  callee()->ExpectCandidates(0);
  caller()->AddAudioTrack();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
}

TEST_P(PeerConnectionIntegrationTest, MediaCallWithoutMediaEngineFails) {
  ASSERT_TRUE(CreatePeerConnectionWrappersWithoutMediaEngine());
  // AddTrack should fail.
  EXPECT_FALSE(
      caller()->pc()->AddTrack(caller()->CreateLocalAudioTrack(), {}).ok());
}

INSTANTIATE_TEST_SUITE_P(
    PeerConnectionIntegrationTest,
    PeerConnectionIntegrationInteropTest,
    Values(std::make_tuple(SdpSemantics::kPlanB_DEPRECATED,
                           SdpSemantics::kUnifiedPlan),
           std::make_tuple(SdpSemantics::kUnifiedPlan,
                           SdpSemantics::kPlanB_DEPRECATED)));

// Test that if the Unified Plan side offers two video tracks then the Plan B
// side will only see the first one and ignore the second.
TEST_F(PeerConnectionIntegrationTestPlanB, TwoVideoUnifiedPlanToNoMediaPlanB) {
  ASSERT_TRUE(CreatePeerConnectionWrappersWithSdpSemantics(
      SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB_DEPRECATED));
  ConnectFakeSignaling();
  auto first_sender = caller()->AddVideoTrack();
  caller()->AddVideoTrack();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  // Verify that there is only one receiver and it corresponds to the first
  // added track.
  auto receivers = callee()->pc()->GetReceivers();
  ASSERT_EQ(1u, receivers.size());
  EXPECT_TRUE(receivers[0]->track()->enabled());
  EXPECT_EQ(first_sender->track()->id(), receivers[0]->track()->id());

  MediaExpectations media_expectations;
  media_expectations.CalleeExpectsSomeVideo();
  ASSERT_TRUE(ExpectNewFrames(media_expectations));
}

// Test that if the initial offer tagged BUNDLE section is rejected due to its
// associated RtpTransceiver being stopped and another transceiver is added,
// then renegotiation causes the callee to receive the new video track without
// error.
// This is a regression test for bugs.webrtc.org/9954
TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       ReOfferWithStoppedBundleTaggedTransceiver) {
  RTCConfiguration config;
  config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  auto audio_transceiver_or_error =
      caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
  ASSERT_TRUE(audio_transceiver_or_error.ok());
  auto audio_transceiver = audio_transceiver_or_error.MoveValue();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  {
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeAudio();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }

  audio_transceiver->StopInternal();
  caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack());

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  {
    MediaExpectations media_expectations;
    media_expectations.CalleeExpectsSomeVideo();
    ASSERT_TRUE(ExpectNewFrames(media_expectations));
  }
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       StopTransceiverRemovesDtlsTransports) {
  RTCConfiguration config;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  auto audio_transceiver_or_error =
      caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
  ASSERT_TRUE(audio_transceiver_or_error.ok());
  auto audio_transceiver = audio_transceiver_or_error.MoveValue();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);

  audio_transceiver->StopStandard();
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_EQ(0U, caller()->pc()->GetTransceivers().size());
  EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
            caller()->pc()->ice_gathering_state());
  EXPECT_THAT(caller()->ice_gathering_state_history(),
              ElementsAre(PeerConnectionInterface::kIceGatheringGathering,
                          PeerConnectionInterface::kIceGatheringComplete,
                          PeerConnectionInterface::kIceGatheringNew));
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       StopTransceiverStopsAndRemovesTransceivers) {
  RTCConfiguration config;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  auto audio_transceiver_or_error =
      caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
  ASSERT_TRUE(audio_transceiver_or_error.ok());
  auto caller_transceiver = audio_transceiver_or_error.MoveValue();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  caller_transceiver->StopStandard();

  auto callee_transceiver = callee()->pc()->GetTransceivers()[0];
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ(0U, caller()->pc()->GetTransceivers().size());
  EXPECT_EQ(0U, callee()->pc()->GetTransceivers().size());
  EXPECT_EQ(0U, caller()->pc()->GetSenders().size());
  EXPECT_EQ(0U, callee()->pc()->GetSenders().size());
  EXPECT_EQ(0U, caller()->pc()->GetReceivers().size());
  EXPECT_EQ(0U, callee()->pc()->GetReceivers().size());
  EXPECT_TRUE(caller_transceiver->stopped());
  EXPECT_TRUE(callee_transceiver->stopped());
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       StopTransceiverEndsIncomingAudioTrack) {
  RTCConfiguration config;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  auto audio_transceiver_or_error =
      caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
  ASSERT_TRUE(audio_transceiver_or_error.ok());
  auto audio_transceiver = audio_transceiver_or_error.MoveValue();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  auto caller_track = audio_transceiver->receiver()->track();
  auto callee_track = callee()->pc()->GetReceivers()[0]->track();
  audio_transceiver->StopStandard();
  EXPECT_EQ(MediaStreamTrackInterface::TrackState::kEnded,
            caller_track->state());
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ(MediaStreamTrackInterface::TrackState::kEnded,
            callee_track->state());
}

TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
       StopTransceiverEndsIncomingVideoTrack) {
  RTCConfiguration config;
  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
  ConnectFakeSignaling();
  auto audio_transceiver_or_error =
      caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack());
  ASSERT_TRUE(audio_transceiver_or_error.ok());
  auto audio_transceiver = audio_transceiver_or_error.MoveValue();

  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  auto caller_track = audio_transceiver->receiver()->track();
  auto callee_track = callee()->pc()->GetReceivers()[0]->track();
  audio_transceiver->StopStandard();
  EXPECT_EQ(MediaStreamTrackInterface::TrackState::kEnded,
            caller_track->state());
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  EXPECT_EQ(MediaStreamTrackInterface::TrackState::kEnded,
            callee_track->state());
}

TEST_P(PeerConnectionIntegrationTest, EndToEndRtpSenderVideoEncoderSelector) {
  ASSERT_TRUE(
      CreateOneDirectionalPeerConnectionWrappers(/*caller_to_callee=*/true));
  ConnectFakeSignaling();
  // Add one-directional video, from caller to callee.
  rtc::scoped_refptr<webrtc::VideoTrackInterface> caller_track =
      caller()->CreateLocalVideoTrack();
  auto sender = caller()->AddTrack(caller_track);
  PeerConnectionInterface::RTCOfferAnswerOptions options;
  options.offer_to_receive_video = 0;
  caller()->SetOfferAnswerOptions(options);
  caller()->CreateAndSetAndSignalOffer();
  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
  ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);

  std::unique_ptr<MockEncoderSelector> encoder_selector =
      std::make_unique<MockEncoderSelector>();
  EXPECT_CALL(*encoder_selector, OnCurrentEncoder);

  sender->SetEncoderSelector(std::move(encoder_selector));

  // Expect video to be received in one direction.
  MediaExpectations media_expectations;
  media_expectations.CallerExpectsNoVideo();
  media_expectations.CalleeExpectsSomeVideo();

  EXPECT_TRUE(ExpectNewFrames(media_expectations));
}

}  // namespace

}  // namespace webrtc