/*
 *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include <algorithm>
#include <cstddef>
#include <cstdint>
#include <memory>
#include <optional>
#include <string>
#include <utility>
#include <vector>

#include "absl/strings/string_view.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/create_peerconnection_factory.h"
#include "api/jsep.h"
#include "api/make_ref_counted.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/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/set_remote_description_observer_interface.h"
#include "api/units/data_rate.h"
#include "api/video/render_resolution.h"
#include "api/video/video_codec_constants.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_decoder_factory_template.h"
#include "api/video_codecs/video_decoder_factory_template_dav1d_adapter.h"
#include "api/video_codecs/video_decoder_factory_template_libvpx_vp8_adapter.h"
#include "api/video_codecs/video_decoder_factory_template_libvpx_vp9_adapter.h"
#include "api/video_codecs/video_decoder_factory_template_open_h264_adapter.h"
#include "api/video_codecs/video_encoder_factory_template.h"
#include "api/video_codecs/video_encoder_factory_template_libaom_av1_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp8_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_libvpx_vp9_adapter.h"
#include "api/video_codecs/video_encoder_factory_template_open_h264_adapter.h"
#include "media/base/codec.h"
#include "media/base/media_channel.h"
#include "media/base/stream_params.h"
#include "media/engine/webrtc_video_engine.h"
#include "pc/media_session.h"
#include "pc/peer_connection_wrapper.h"
#include "pc/rtp_transceiver.h"
#include "pc/session_description.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/integration_test_helpers.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "rtc_base/checks.h"
#include "rtc_base/system/plan_b_only.h"
#include "rtc_base/thread.h"
#include "system_wrappers/include/metrics.h"
#include "test/create_test_field_trials.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/run_loop.h"
#include "test/wait_until.h"

// This file contains tests for RTP Media API-related behavior of
// `webrtc::PeerConnection`, see https://w3c.github.io/webrtc-pc/#rtp-media-api.

namespace webrtc {

using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
using ::testing::ElementsAre;
using ::testing::Eq;
using ::testing::UnorderedElementsAre;
using ::testing::Values;

template <typename MethodFunctor>
class OnSuccessObserver : public SetRemoteDescriptionObserverInterface {
 public:
  explicit OnSuccessObserver(MethodFunctor on_success)
      : on_success_(std::move(on_success)) {}

  // SetRemoteDescriptionObserverInterface implementation.
  void OnSetRemoteDescriptionComplete(RTCError error) override {
    RTC_CHECK(error.ok());
    on_success_();
  }

 private:
  MethodFunctor on_success_;
};

class PeerConnectionRtpBaseTest : public ::testing::Test {
 public:
  explicit PeerConnectionRtpBaseTest(SdpSemantics sdp_semantics)
      : sdp_semantics_(sdp_semantics),
        pc_factory_(CreatePeerConnectionFactory(
            Thread::Current(),
            Thread::Current(),
            Thread::Current(),
            FakeAudioCaptureModule::Create(),
            CreateBuiltinAudioEncoderFactory(),
            CreateBuiltinAudioDecoderFactory(),
            std::make_unique<
                VideoEncoderFactoryTemplate<LibvpxVp8EncoderTemplateAdapter,
                                            LibvpxVp9EncoderTemplateAdapter,
                                            OpenH264EncoderTemplateAdapter,
                                            LibaomAv1EncoderTemplateAdapter>>(),
            std::make_unique<
                VideoDecoderFactoryTemplate<LibvpxVp8DecoderTemplateAdapter,
                                            LibvpxVp9DecoderTemplateAdapter,
                                            OpenH264DecoderTemplateAdapter,
                                            Dav1dDecoderTemplateAdapter>>(),
            nullptr /* audio_mixer */,
            nullptr /* audio_processing */,
            nullptr /* audio_frame_processor */,
            CreateTestFieldTrialsPtr())) {
    metrics::Reset();
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
    return CreatePeerConnection(RTCConfiguration());
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithPlanB() {
    RTCConfiguration config;
    config.sdp_semantics = SdpSemantics::kPlanB_DEPRECATED;
    return CreatePeerConnectionInternal(config);
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithUnifiedPlan() {
    return CreatePeerConnectionWithUnifiedPlan("");
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithUnifiedPlan(
      absl::string_view field_trials) {
    RTCConfiguration config;
    config.sdp_semantics = SdpSemantics::kUnifiedPlan;
    return CreatePeerConnectionInternal(config, field_trials);
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
      const RTCConfiguration& config) {
    return CreatePeerConnection(config, "");
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
      absl::string_view field_trials) {
    return CreatePeerConnection(RTCConfiguration(), field_trials);
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
      const RTCConfiguration& config,
      absl::string_view field_trials) {
    RTCConfiguration modified_config = config;
    modified_config.sdp_semantics = sdp_semantics_;
    return CreatePeerConnectionInternal(modified_config, field_trials);
  }

 protected:
  const SdpSemantics sdp_semantics_;
  test::RunLoop run_loop_;
  scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;

 private:
  // Private so that tests don't accidentally bypass the SdpSemantics
  // adjustment.
  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionInternal(
      const RTCConfiguration& config) {
    return CreatePeerConnectionInternal(config, "");
  }

  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionInternal(
      const RTCConfiguration& config,
      absl::string_view field_trials) {
    auto observer = std::make_unique<MockPeerConnectionObserver>();
    PeerConnectionDependencies pc_dependencies(observer.get());
    if (!field_trials.empty()) {
      pc_dependencies.trials = CreateTestFieldTrialsPtr(field_trials);
    }
    auto result = pc_factory_->CreatePeerConnectionOrError(
        config, std::move(pc_dependencies));
    EXPECT_TRUE(result.ok());
    observer->SetPeerConnectionInterface(result.value().get());
    return std::make_unique<PeerConnectionWrapper>(
        pc_factory_, result.MoveValue(), std::move(observer));
  }
};

class PeerConnectionRtpTest
    : public PeerConnectionRtpBaseTest,
      public ::testing::WithParamInterface<SdpSemantics> {
 protected:
  PeerConnectionRtpTest() : PeerConnectionRtpBaseTest(GetParam()) {}
};

RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
class PeerConnectionRtpTestPlanB : public PeerConnectionRtpBaseTest {
 protected:
  PeerConnectionRtpTestPlanB()
      : PeerConnectionRtpBaseTest(SdpSemantics::kPlanB_DEPRECATED) {}
};
RTC_ALLOW_PLAN_B_DEPRECATION_END()

class PeerConnectionRtpTestUnifiedPlan : public PeerConnectionRtpBaseTest {
 protected:
  PeerConnectionRtpTestUnifiedPlan()
      : PeerConnectionRtpBaseTest(SdpSemantics::kUnifiedPlan) {}

  // Helper to emulate an SFU that rejects an offered media section
  // in answer.
  bool ExchangeOfferAnswerWhereRemoteStopsTransceiver(
      PeerConnectionWrapper* caller,
      PeerConnectionWrapper* callee,
      size_t mid_to_stop) {
    std::unique_ptr<SessionDescriptionInterface> offer = caller->CreateOffer();
    caller->SetLocalDescription(offer->Clone());
    callee->SetRemoteDescription(std::move(offer));
    EXPECT_LT(mid_to_stop, callee->pc()->GetTransceivers().size());
    // Must use StopInternal in order to do instant reject.
    callee->pc()->GetTransceivers()[mid_to_stop]->StopInternal();
    std::unique_ptr<SessionDescriptionInterface> answer =
        callee->CreateAnswer();
    EXPECT_TRUE(answer);
    bool set_local_answer = callee->SetLocalDescription(answer->Clone());
    EXPECT_TRUE(set_local_answer);
    bool set_remote_answer = caller->SetRemoteDescription(std::move(answer));
    EXPECT_TRUE(set_remote_answer);
    return set_remote_answer;
  }
};

// These tests cover `PeerConnectionObserver` callbacks firing upon
// setting the remote description.

TEST_P(PeerConnectionRtpTest, AddTrackWithoutStreamFiresOnAddTrack) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  ASSERT_TRUE(caller->AddAudioTrack("audio_track"));
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));

  ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
  const auto& add_track_event = callee->observer()->add_track_events_[0];
  EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());

  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    // Since we are not supporting the no stream case with Plan B, there should
    // be a generated stream, even though we didn't set one with AddTrack.
    ASSERT_EQ(1u, add_track_event.streams.size());
    EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
  } else {
    EXPECT_EQ(0u, add_track_event.streams.size());
  }
}

TEST_P(PeerConnectionRtpTest, AddTrackWithStreamFiresOnAddTrack) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  ASSERT_TRUE(caller->AddAudioTrack("audio_track", {"audio_stream"}));
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));

  ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
  auto& add_track_event = callee->observer()->add_track_events_[0];
  ASSERT_EQ(add_track_event.streams.size(), 1u);
  EXPECT_EQ("audio_stream", add_track_event.streams[0]->id());
  EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
  EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());
}

TEST_P(PeerConnectionRtpTest, RemoveTrackWithoutStreamFiresOnRemoveTrack) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("audio_track", {});
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
  ASSERT_TRUE(
      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));

  EXPECT_TRUE(caller->pc()->RemoveTrackOrError(sender).ok());
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));

  ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
  EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
            callee->observer()->remove_track_events_);
}

TEST_P(PeerConnectionRtpTest, RemoveTrackWithStreamFiresOnRemoveTrack) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("audio_track", {"audio_stream"});
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
  ASSERT_TRUE(
      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));

  EXPECT_TRUE(caller->pc()->RemoveTrackOrError(sender).ok());
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));

  ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
  EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
            callee->observer()->remove_track_events_);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  EXPECT_EQ(0u, callee->observer()->remote_streams()->count());
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
}

TEST_P(PeerConnectionRtpTest, RemoveTrackWithSharedStreamFiresOnRemoveTrack) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  const char kSharedStreamId[] = "shared_audio_stream";
  auto sender1 = caller->AddAudioTrack("audio_track1", {kSharedStreamId});
  auto sender2 = caller->AddAudioTrack("audio_track2", {kSharedStreamId});
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
  ASSERT_TRUE(
      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));

  // Remove "audio_track1".
  EXPECT_TRUE(caller->pc()->RemoveTrackOrError(sender1).ok());
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
  EXPECT_EQ(
      std::vector<scoped_refptr<RtpReceiverInterface>>{
          callee->observer()->add_track_events_[0].receiver},
      callee->observer()->remove_track_events_);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  ASSERT_EQ(1u, callee->observer()->remote_streams()->count());
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  ASSERT_TRUE(
      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));

  // Remove "audio_track2".
  EXPECT_TRUE(caller->pc()->RemoveTrackOrError(sender2).ok());
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
  EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
            callee->observer()->remove_track_events_);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  EXPECT_EQ(0u, callee->observer()->remote_streams()->count());
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
}

// Tests the edge case that if a stream ID changes for a given track that both
// OnRemoveTrack and OnAddTrack is fired.
RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
TEST_F(PeerConnectionRtpTestPlanB,
       RemoteStreamIdChangesFiresOnRemoveAndOnAddTrack) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  const char kStreamId1[] = "stream1";
  const char kStreamId2[] = "stream2";
  caller->AddAudioTrack("audio_track1", {kStreamId1});
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  EXPECT_EQ(callee->observer()->add_track_events_.size(), 1u);

  // Change the stream ID of the sender in the session description.
  std::unique_ptr<SessionDescriptionInterface> offer =
      caller->CreateOfferAndSetAsLocal();
  auto* audio_desc = GetFirstAudioContentDescription(offer->description());
  ASSERT_EQ(audio_desc->mutable_streams().size(), 1u);
  audio_desc->mutable_streams()[0].set_stream_ids({kStreamId2});
  ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));

  ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
  EXPECT_EQ(callee->observer()->add_track_events_[1].streams[0]->id(),
            kStreamId2);
  ASSERT_EQ(callee->observer()->remove_track_events_.size(), 1u);
  EXPECT_EQ(callee->observer()->remove_track_events_[0]->streams()[0]->id(),
            kStreamId1);
}
RTC_ALLOW_PLAN_B_DEPRECATION_END()

// Tests that setting a remote description with sending transceivers will fire
// the OnTrack callback for each transceiver and setting a remote description
// with receive only transceivers will not call OnTrack. One transceiver is
// created without any stream_ids, while the other is created with multiple
// stream_ids.
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTransceiverCallsOnTrack) {
  const std::string kStreamId1 = "video_stream1";
  const std::string kStreamId2 = "video_stream2";
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto audio_transceiver = caller->AddTransceiver(MediaType::AUDIO);
  RtpTransceiverInit video_transceiver_init;
  video_transceiver_init.stream_ids = {kStreamId1, kStreamId2};
  auto video_transceiver =
      caller->AddTransceiver(MediaType::VIDEO, video_transceiver_init);

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  ASSERT_EQ(0u, caller->observer()->on_track_transceivers_.size());
  ASSERT_EQ(2u, callee->observer()->on_track_transceivers_.size());
  EXPECT_EQ(audio_transceiver->mid(),
            callee->pc()->GetTransceivers()[0]->mid());
  EXPECT_EQ(video_transceiver->mid(),
            callee->pc()->GetTransceivers()[1]->mid());
  std::vector<scoped_refptr<MediaStreamInterface>> audio_streams =
      callee->pc()->GetTransceivers()[0]->receiver()->streams();
  std::vector<scoped_refptr<MediaStreamInterface>> video_streams =
      callee->pc()->GetTransceivers()[1]->receiver()->streams();
  ASSERT_EQ(0u, audio_streams.size());
  ASSERT_EQ(2u, video_streams.size());
  EXPECT_EQ(kStreamId1, video_streams[0]->id());
  EXPECT_EQ(kStreamId2, video_streams[1]->id());
}

// Test that doing additional offer/answer exchanges with no changes to tracks
// will cause no additional OnTrack calls after the tracks have been negotiated.
TEST_F(PeerConnectionRtpTestUnifiedPlan, ReofferDoesNotCallOnTrack) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  caller->AddAudioTrack("audio");
  callee->AddAudioTrack("audio");

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());

  // If caller reoffers with no changes expect no additional OnTrack calls.
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());

  // Also if callee reoffers with no changes expect no additional OnTrack calls.
  ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
  EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
}

// Test that OnTrack is called when the transceiver direction changes to send
// the track.
TEST_F(PeerConnectionRtpTestUnifiedPlan, SetDirectionCallsOnTrack) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  EXPECT_TRUE(
      transceiver->SetDirectionWithError(RtpTransceiverDirection::kInactive)
          .ok());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
  EXPECT_EQ(0u, callee->observer()->on_track_transceivers_.size());

  EXPECT_TRUE(
      transceiver->SetDirectionWithError(RtpTransceiverDirection::kSendOnly)
          .ok());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());

  // If the direction changes but it is still receiving on the remote side, then
  // OnTrack should not be fired again.
  EXPECT_TRUE(
      transceiver->SetDirectionWithError(RtpTransceiverDirection::kSendRecv)
          .ok());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
}

// Test that OnTrack is called twice when a sendrecv call is started, the callee
// changes the direction to inactive, then changes it back to sendrecv.
TEST_F(PeerConnectionRtpTestUnifiedPlan, SetDirectionHoldCallsOnTrackTwice) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());

  // Put the call on hold by no longer receiving the track.
  EXPECT_TRUE(callee->pc()
                  ->GetTransceivers()[0]
                  ->SetDirectionWithError(RtpTransceiverDirection::kInactive)
                  .ok());

  ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());

  // Resume the call by changing the direction to recvonly. This should call
  // OnTrack again on the callee side.
  EXPECT_TRUE(callee->pc()
                  ->GetTransceivers()[0]
                  ->SetDirectionWithError(RtpTransceiverDirection::kRecvOnly)
                  .ok());

  ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
  EXPECT_EQ(2u, callee->observer()->on_track_transceivers_.size());
}

// Test that setting a remote offer twice with no answer in the middle results
// in OnAddTrack being fired only once.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       ApplyTwoRemoteOffersWithNoAnswerResultsInOneAddTrackEvent) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  caller->AddAudioTrack("audio_track", {});

  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  ASSERT_EQ(1u, callee->observer()->add_track_events_.size());

  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  EXPECT_EQ(1u, callee->observer()->add_track_events_.size());
}

// Test that setting a remote offer twice with no answer in the middle and the
// track being removed between the two offers results in OnAddTrack being called
// once the first time and OnRemoveTrack being called once the second time.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       ApplyRemoteOfferAddThenRemoteOfferRemoveResultsInOneRemoveTrackEvent) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("audio_track", {});

  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  ASSERT_EQ(1u, callee->observer()->add_track_events_.size());
  EXPECT_EQ(0u, callee->observer()->remove_track_events_.size());

  caller->pc()->RemoveTrackOrError(sender);

  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  EXPECT_EQ(1u, callee->observer()->add_track_events_.size());
  EXPECT_EQ(1u, callee->observer()->remove_track_events_.size());
}

// Test that changing the direction from receiving to not receiving between
// setting the remote offer and creating / setting the local answer results in
// a remove track event when SetLocalDescription is called.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       ChangeDirectionInAnswerResultsInRemoveTrackEvent) {
  auto caller = CreatePeerConnection();
  caller->AddTransceiver(MediaType::AUDIO);
  auto callee = CreatePeerConnection();
  callee->AddAudioTrack("audio_track", {});

  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
  EXPECT_EQ(1u, callee->observer()->add_track_events_.size());
  EXPECT_EQ(0u, callee->observer()->remove_track_events_.size());

  auto callee_transceiver = callee->pc()->GetTransceivers()[0];
  EXPECT_TRUE(callee_transceiver
                  ->SetDirectionWithError(RtpTransceiverDirection::kSendOnly)
                  .ok());

  ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
  EXPECT_EQ(1u, callee->observer()->add_track_events_.size());
  EXPECT_EQ(1u, callee->observer()->remove_track_events_.size());
}

TEST_F(PeerConnectionRtpTestUnifiedPlan, ChangeMsidWhileReceiving) {
  auto caller = CreatePeerConnection();
  caller->AddAudioTrack("audio_track", {"stream1"});
  auto callee = CreatePeerConnection();
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));

  ASSERT_EQ(1u, callee->observer()->on_track_transceivers_.size());
  auto transceiver = callee->observer()->on_track_transceivers_[0];
  ASSERT_EQ(1u, transceiver->receiver()->streams().size());
  EXPECT_EQ("stream1", transceiver->receiver()->streams()[0]->id());

  ASSERT_TRUE(callee->CreateAnswerAndSetAsLocal());

  // Change the stream ID in the offer.
  caller->pc()->GetSenders()[0]->SetStreams({"stream2"});
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  ASSERT_EQ(1u, transceiver->receiver()->streams().size());
  EXPECT_EQ("stream2", transceiver->receiver()->streams()[0]->id());
}

// These tests examine the state of the peer connection as a result of
// performing SetRemoteDescription().

TEST_P(PeerConnectionRtpTest, AddTrackWithoutStreamAddsReceiver) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  ASSERT_TRUE(caller->AddAudioTrack("audio_track", {}));
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));

  EXPECT_EQ(callee->pc()->GetReceivers().size(), 1u);
  auto receiver_added = callee->pc()->GetReceivers()[0];
  EXPECT_EQ("audio_track", receiver_added->track()->id());

  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    // Since we are not supporting the no stream case with Plan B, there should
    // be a generated stream, even though we didn't set one with AddTrack.
    ASSERT_EQ(1u, receiver_added->streams().size());
    EXPECT_TRUE(receiver_added->streams()[0]->FindAudioTrack("audio_track"));
  } else {
    EXPECT_EQ(0u, receiver_added->streams().size());
  }
}

TEST_P(PeerConnectionRtpTest, AddTrackWithStreamAddsReceiver) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  ASSERT_TRUE(caller->AddAudioTrack("audio_track", {"audio_stream"}));
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));

  EXPECT_EQ(callee->pc()->GetReceivers().size(), 1u);
  auto receiver_added = callee->pc()->GetReceivers()[0];
  EXPECT_EQ("audio_track", receiver_added->track()->id());
  EXPECT_EQ(receiver_added->streams().size(), 1u);
  EXPECT_EQ("audio_stream", receiver_added->streams()[0]->id());
  EXPECT_TRUE(receiver_added->streams()[0]->FindAudioTrack("audio_track"));
}

TEST_P(PeerConnectionRtpTest, RemoveTrackWithoutStreamRemovesReceiver) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("audio_track", {});
  ASSERT_TRUE(sender);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  ASSERT_EQ(callee->pc()->GetReceivers().size(), 1u);
  auto receiver = callee->pc()->GetReceivers()[0];
  ASSERT_TRUE(caller->pc()->RemoveTrackOrError(sender).ok());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
    // With Unified Plan the receiver stays but the transceiver transitions to
    // inactive.
    ASSERT_EQ(1u, callee->pc()->GetReceivers().size());
    EXPECT_EQ(RtpTransceiverDirection::kInactive,
              callee->pc()->GetTransceivers()[0]->current_direction());
  } else {
    // With Plan B the receiver is removed.
    ASSERT_EQ(0u, callee->pc()->GetReceivers().size());
  }
}

TEST_P(PeerConnectionRtpTest, RemoveTrackWithStreamRemovesReceiver) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("audio_track", {"audio_stream"});
  ASSERT_TRUE(sender);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  ASSERT_EQ(callee->pc()->GetReceivers().size(), 1u);
  auto receiver = callee->pc()->GetReceivers()[0];
  ASSERT_TRUE(caller->pc()->RemoveTrackOrError(sender).ok());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
    // With Unified Plan the receiver stays but the transceiver transitions to
    // inactive.
    EXPECT_EQ(1u, callee->pc()->GetReceivers().size());
    EXPECT_EQ(RtpTransceiverDirection::kInactive,
              callee->pc()->GetTransceivers()[0]->current_direction());
  } else {
    // With Plan B the receiver is removed.
    EXPECT_EQ(0u, callee->pc()->GetReceivers().size());
  }
}

TEST_P(PeerConnectionRtpTest, RemoveTrackWithSharedStreamRemovesReceiver) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  const char kSharedStreamId[] = "shared_audio_stream";
  auto sender1 = caller->AddAudioTrack("audio_track1", {kSharedStreamId});
  auto sender2 = caller->AddAudioTrack("audio_track2", {kSharedStreamId});
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  ASSERT_EQ(2u, callee->pc()->GetReceivers().size());

  // Remove "audio_track1".
  EXPECT_TRUE(caller->pc()->RemoveTrackOrError(sender1).ok());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
    // With Unified Plan the receiver stays but the transceiver transitions to
    // inactive.
    ASSERT_EQ(2u, callee->pc()->GetReceivers().size());
    auto transceiver = callee->pc()->GetTransceivers()[0];
    EXPECT_EQ("audio_track1", transceiver->receiver()->track()->id());
    EXPECT_EQ(RtpTransceiverDirection::kInactive,
              transceiver->current_direction());
  } else {
    // With Plan B the receiver is removed.
    ASSERT_EQ(1u, callee->pc()->GetReceivers().size());
    EXPECT_EQ("audio_track2", callee->pc()->GetReceivers()[0]->track()->id());
  }

  // Remove "audio_track2".
  EXPECT_TRUE(caller->pc()->RemoveTrackOrError(sender2).ok());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
    // With Unified Plan the receiver stays but the transceiver transitions to
    // inactive.
    ASSERT_EQ(2u, callee->pc()->GetReceivers().size());
    auto transceiver = callee->pc()->GetTransceivers()[1];
    EXPECT_EQ("audio_track2", transceiver->receiver()->track()->id());
    EXPECT_EQ(RtpTransceiverDirection::kInactive,
              transceiver->current_direction());
  } else {
    // With Plan B the receiver is removed.
    ASSERT_EQ(0u, callee->pc()->GetReceivers().size());
  }
}

TEST_P(PeerConnectionRtpTest, AudioGetParametersHasHeaderExtensions) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();
  auto sender = caller->AddAudioTrack("audio_track");
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  ASSERT_GT(caller->pc()->GetSenders().size(), 0u);
  EXPECT_GT(sender->GetParameters().header_extensions.size(), 0u);

  ASSERT_GT(callee->pc()->GetReceivers().size(), 0u);
  auto receiver = callee->pc()->GetReceivers()[0];
  EXPECT_GT(receiver->GetParameters().header_extensions.size(), 0u);
}

TEST_P(PeerConnectionRtpTest, VideoGetParametersHasHeaderExtensions) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();
  auto sender = caller->AddVideoTrack("video_track");
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  ASSERT_GT(caller->pc()->GetSenders().size(), 0u);
  EXPECT_GT(sender->GetParameters().header_extensions.size(), 0u);

  ASSERT_GT(callee->pc()->GetReceivers().size(), 0u);
  auto receiver = callee->pc()->GetReceivers()[0];
  EXPECT_GT(receiver->GetParameters().header_extensions.size(), 0u);
}

// Invokes SetRemoteDescription() twice in a row without synchronizing the two
// calls and examine the state of the peer connection inside the callbacks to
// ensure that the second call does not occur prematurely, contaminating the
// state of the peer connection of the first callback.
RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
TEST_F(PeerConnectionRtpTestPlanB,
       StatesCorrelateWithSetRemoteDescriptionCall) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  // Create SDP for adding a track and for removing it. This will be used in the
  // first and second SetRemoteDescription() calls.
  auto sender = caller->AddAudioTrack("audio_track", {});
  auto srd1_sdp = caller->CreateOfferAndSetAsLocal();
  EXPECT_TRUE(caller->pc()->RemoveTrackOrError(sender).ok());
  auto srd2_sdp = caller->CreateOfferAndSetAsLocal();

  // In the first SetRemoteDescription() callback, check that we have a
  // receiver for the track.
  auto pc = callee->pc();
  bool srd1_callback_called = false;
  auto srd1_callback = [&srd1_callback_called, &pc]() {
    EXPECT_EQ(pc->GetReceivers().size(), 1u);
    srd1_callback_called = true;
  };

  // In the second SetRemoteDescription() callback, check that the receiver has
  // been removed.
  // TODO(hbos): When we implement Unified Plan, receivers will not be removed.
  // Instead, the transceiver owning the receiver will become inactive.
  // https://crbug.com/webrtc/7600
  bool srd2_callback_called = false;
  auto srd2_callback = [&srd2_callback_called, &pc]() {
    EXPECT_TRUE(pc->GetReceivers().empty());
    srd2_callback_called = true;
  };

  // Invoke SetRemoteDescription() twice in a row without synchronizing the two
  // calls. The callbacks verify that the two calls are synchronized, as in, the
  // effects of the second SetRemoteDescription() call must not have happened by
  // the time the first callback is invoked. If it has then the receiver that is
  // added as a result of the first SetRemoteDescription() call will already
  // have been removed as a result of the second SetRemoteDescription() call
  // when the first callback is invoked.
  callee->pc()->SetRemoteDescription(
      std::move(srd1_sdp),
      make_ref_counted<OnSuccessObserver<decltype(srd1_callback)>>(
          srd1_callback));
  callee->pc()->SetRemoteDescription(
      std::move(srd2_sdp),
      make_ref_counted<OnSuccessObserver<decltype(srd2_callback)>>(
          srd2_callback));
  EXPECT_TRUE(WaitUntil([&] { return srd1_callback_called; }));
  EXPECT_TRUE(WaitUntil([&] { return srd2_callback_called; }));
}
RTC_ALLOW_PLAN_B_DEPRECATION_END()

// Tests that a remote track is created with the signaled MSIDs when they are
// communicated with a=msid and no SSRCs are signaled at all (i.e., no a=ssrc
// lines).
TEST_F(PeerConnectionRtpTestUnifiedPlan, UnsignaledSsrcCreatesReceiverStreams) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();
  const char kStreamId1[] = "stream1";
  const char kStreamId2[] = "stream2";
  caller->AddTrack(caller->CreateAudioTrack("audio_track1"),
                   {kStreamId1, kStreamId2});

  std::unique_ptr<SessionDescriptionInterface> offer =
      caller->CreateOfferAndSetAsLocal();
  // Munge the offer to take out everything but the stream_ids.
  auto contents = offer->description()->contents();
  ASSERT_TRUE(!contents.empty());
  ASSERT_TRUE(!contents[0].media_description()->streams().empty());
  std::vector<std::string> stream_ids =
      contents[0].media_description()->streams()[0].stream_ids();
  contents[0].media_description()->mutable_streams().clear();
  StreamParams new_stream;
  new_stream.set_stream_ids(stream_ids);
  contents[0].media_description()->AddStream(new_stream);

  // Set the remote description and verify that the streams were added to the
  // receiver correctly.
  ASSERT_TRUE(callee->SetRemoteDescription(offer->Clone()));
  auto receivers = callee->pc()->GetReceivers();
  ASSERT_EQ(receivers.size(), 1u);
  ASSERT_EQ(receivers[0]->streams().size(), 2u);
  EXPECT_EQ(receivers[0]->streams()[0]->id(), kStreamId1);
  EXPECT_EQ(receivers[0]->streams()[1]->id(), kStreamId2);
}
TEST_F(PeerConnectionRtpTestUnifiedPlan, TracksDoNotEndWhenSsrcChanges) {
  constexpr uint32_t kFirstMungedSsrc = 1337u;

  // Munging allowed: kSsrcs (27)
  auto caller = CreatePeerConnection();
  auto callee =
      CreatePeerConnection("WebRTC-NoSdpMangleAllowForTesting/Enabled,27/");

  // Caller offers to receive audio and video.
  RtpTransceiverInit init;
  init.direction = RtpTransceiverDirection::kRecvOnly;
  caller->AddTransceiver(MediaType::AUDIO, init);
  caller->AddTransceiver(MediaType::VIDEO, init);

  // Callee wants to send audio and video tracks.
  callee->AddTrack(callee->CreateAudioTrack("audio_track"), {});
  callee->AddTrack(callee->CreateVideoTrack("video_track"), {});

  // Do inittial offer/answer exchange.
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  ASSERT_TRUE(
      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
  ASSERT_EQ(caller->observer()->add_track_events_.size(), 2u);
  ASSERT_EQ(caller->pc()->GetReceivers().size(), 2u);

  // Do a follow-up offer/answer exchange where the SSRCs are modified.
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  std::unique_ptr<SessionDescriptionInterface> answer = callee->CreateAnswer();
  auto& contents = answer->description()->contents();
  ASSERT_TRUE(!contents.empty());
  for (size_t i = 0; i < contents.size(); ++i) {
    auto& mutable_streams = contents[i].media_description()->mutable_streams();
    ASSERT_EQ(mutable_streams.size(), 1u);
    ReplaceFirstSsrc(mutable_streams[0],
                     kFirstMungedSsrc + static_cast<uint32_t>(i));
  }
  ASSERT_TRUE(callee->SetLocalDescription(answer->Clone()));
  ASSERT_TRUE(caller->SetRemoteDescription(answer->Clone()));

  // No furher track events should fire because we never changed direction, only
  // SSRCs.
  ASSERT_EQ(caller->observer()->add_track_events_.size(), 2u);
  // We should have the same number of receivers as before.
  auto receivers = caller->pc()->GetReceivers();
  ASSERT_EQ(receivers.size(), 2u);
  // The tracks are still alive.
  EXPECT_EQ(receivers[0]->track()->state(),
            MediaStreamTrackInterface::TrackState::kLive);
  EXPECT_EQ(receivers[1]->track()->state(),
            MediaStreamTrackInterface::TrackState::kLive);
}

// Tests that with Unified Plan if the the stream id changes for a track when
// when setting a new remote description, that the media stream is updated
// appropriately for the receiver.
// TODO(https://github.com/w3c/webrtc-pc/issues/1937): Resolve spec issue or fix
// test.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       DISABLED_RemoteStreamIdChangesUpdatesReceiver) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  const char kStreamId1[] = "stream1";
  const char kStreamId2[] = "stream2";
  caller->AddAudioTrack("audio_track1", {kStreamId1});
  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  EXPECT_EQ(callee->observer()->add_track_events_.size(), 1u);

  // Change the stream id of the sender in the session description.
  std::unique_ptr<SessionDescriptionInterface> offer =
      caller->CreateOfferAndSetAsLocal();
  auto contents = offer->description()->contents();
  ASSERT_EQ(contents.size(), 1u);
  ASSERT_EQ(contents[0].media_description()->mutable_streams().size(), 1u);
  contents[0].media_description()->mutable_streams()[0].set_stream_ids(
      {kStreamId2});

  // Set the remote description and verify that the stream was updated
  // properly.
  ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
  auto receivers = callee->pc()->GetReceivers();
  ASSERT_EQ(receivers.size(), 1u);
  ASSERT_EQ(receivers[0]->streams().size(), 1u);
  EXPECT_EQ(receivers[0]->streams()[0]->id(), kStreamId2);
}

// This tests a regression caught by a downstream client, that occured when
// applying a remote description with a SessionDescription object that
// contained StreamParams that didn't have ids. Although there were multiple
// remote audio senders, FindSenderInfo didn't find them as unique, because
// it looked up by StreamParam.id, which none had. This meant only one
// AudioRtpReceiver was created, as opposed to one for each remote sender.
RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
TEST_F(PeerConnectionRtpTestPlanB,
       MultipleRemoteSendersWithoutStreamParamIdAddsMultipleReceivers) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  const char kStreamId1[] = "stream1";
  const char kStreamId2[] = "stream2";
  caller->AddAudioTrack("audio_track1", {kStreamId1});
  caller->AddAudioTrack("audio_track2", {kStreamId2});

  std::unique_ptr<SessionDescriptionInterface> offer =
      caller->CreateOfferAndSetAsLocal();
  auto mutable_streams =
      GetFirstAudioContentDescription(offer->description())->mutable_streams();
  ASSERT_EQ(mutable_streams.size(), 2u);
  // Clear the IDs in the StreamParams.
  mutable_streams[0].id.clear();
  mutable_streams[1].id.clear();
  ASSERT_TRUE(callee->SetRemoteDescription(offer->Clone()));

  auto receivers = callee->pc()->GetReceivers();
  ASSERT_EQ(receivers.size(), 2u);
  ASSERT_EQ(receivers[0]->streams().size(), 1u);
  EXPECT_EQ(kStreamId1, receivers[0]->streams()[0]->id());
  ASSERT_EQ(receivers[1]->streams().size(), 1u);
  EXPECT_EQ(kStreamId2, receivers[1]->streams()[0]->id());
}
RTC_ALLOW_PLAN_B_DEPRECATION_END()

// Tests for the legacy SetRemoteDescription() function signature.

// Sanity test making sure the callback is invoked.
TEST_P(PeerConnectionRtpTest, LegacyObserverOnSuccess) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  std::string error;
  ASSERT_TRUE(
      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(), &error));
}

// Verifies legacy behavior: The observer is not called if if the peer
// connection is destroyed because the asynchronous callback is executed in the
// peer connection's message handler.
TEST_P(PeerConnectionRtpTest,
       LegacyObserverNotCalledIfPeerConnectionDereferenced) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  scoped_refptr<MockSetSessionDescriptionObserver> observer =
      make_ref_counted<MockSetSessionDescriptionObserver>();

  std::unique_ptr<SessionDescriptionInterface> offer =
      caller->CreateOfferAndSetAsLocal();
  callee->pc()->SetRemoteDescription(observer.get(), offer.release());
  callee = nullptr;
  run_loop_.Flush();
  EXPECT_FALSE(observer->called());
}

// RtpTransceiver Tests.

// Test that by default there are no transceivers with Unified Plan.
TEST_F(PeerConnectionRtpTestUnifiedPlan, PeerConnectionHasNoTransceivers) {
  auto caller = CreatePeerConnection();
  EXPECT_THAT(caller->pc()->GetTransceivers(), ElementsAre());
}

// Test that a transceiver created with the audio kind has the correct initial
// properties.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTransceiverHasCorrectInitProperties) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  EXPECT_EQ(std::nullopt, transceiver->mid());
  EXPECT_FALSE(transceiver->stopped());
  EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
  EXPECT_EQ(std::nullopt, transceiver->current_direction());
}

// Test that adding a transceiver with the audio kind creates an audio sender
// and audio receiver with the receiver having a live audio track.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddAudioTransceiverCreatesAudioSenderAndReceiver) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  EXPECT_EQ(MediaType::AUDIO, transceiver->media_type());

  ASSERT_TRUE(transceiver->sender());
  EXPECT_EQ(MediaType::AUDIO, transceiver->sender()->media_type());

  ASSERT_TRUE(transceiver->receiver());
  EXPECT_EQ(MediaType::AUDIO, transceiver->receiver()->media_type());

  auto track = transceiver->receiver()->track();
  ASSERT_TRUE(track);
  EXPECT_EQ(MediaStreamTrackInterface::kAudioKind, track->kind());
  EXPECT_EQ(MediaStreamTrackInterface::TrackState::kLive, track->state());
}

// Test that adding a transceiver with the video kind creates an video sender
// and video receiver with the receiver having a live video track.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddAudioTransceiverCreatesVideoSenderAndReceiver) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::VIDEO);
  EXPECT_EQ(MediaType::VIDEO, transceiver->media_type());

  ASSERT_TRUE(transceiver->sender());
  EXPECT_EQ(MediaType::VIDEO, transceiver->sender()->media_type());

  ASSERT_TRUE(transceiver->receiver());
  EXPECT_EQ(MediaType::VIDEO, transceiver->receiver()->media_type());

  auto track = transceiver->receiver()->track();
  ASSERT_TRUE(track);
  EXPECT_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
  EXPECT_EQ(MediaStreamTrackInterface::TrackState::kLive, track->state());
}

// Test that after a call to AddTransceiver, the transceiver shows in
// GetTransceivers(), the transceiver's sender shows in GetSenders(), and the
// transceiver's receiver shows in GetReceivers().
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTransceiverShowsInLists) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  EXPECT_EQ(std::vector<scoped_refptr<RtpTransceiverInterface>>{transceiver},
            caller->pc()->GetTransceivers());
  EXPECT_EQ(
      std::vector<scoped_refptr<RtpSenderInterface>>{transceiver->sender()},
      caller->pc()->GetSenders());
  EXPECT_EQ(
      std::vector<scoped_refptr<RtpReceiverInterface>>{transceiver->receiver()},
      caller->pc()->GetReceivers());
}

// Test that the direction passed in through the AddTransceiver init parameter
// is set in the returned transceiver.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTransceiverWithDirectionIsReflected) {
  auto caller = CreatePeerConnection();

  RtpTransceiverInit init;
  init.direction = RtpTransceiverDirection::kSendOnly;
  auto transceiver = caller->AddTransceiver(MediaType::AUDIO, init);
  EXPECT_EQ(RtpTransceiverDirection::kSendOnly, transceiver->direction());
}

// Test that calling AddTransceiver with a track creates a transceiver which has
// its sender's track set to the passed-in track.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTransceiverWithTrackCreatesSenderWithTrack) {
  auto caller = CreatePeerConnection();

  auto audio_track = caller->CreateAudioTrack("audio track");
  auto transceiver = caller->AddTransceiver(audio_track);

  auto sender = transceiver->sender();
  ASSERT_TRUE(sender->track());
  EXPECT_EQ(audio_track, sender->track());

  auto receiver = transceiver->receiver();
  ASSERT_TRUE(receiver->track());
  EXPECT_EQ(MediaStreamTrackInterface::kAudioKind, receiver->track()->kind());
  EXPECT_EQ(MediaStreamTrackInterface::TrackState::kLive,
            receiver->track()->state());
}

// Test that calling AddTransceiver twice with the same track creates distinct
// transceivers, senders with the same track.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTransceiverTwiceWithSameTrackCreatesMultipleTransceivers) {
  auto caller = CreatePeerConnection();

  auto audio_track = caller->CreateAudioTrack("audio track");

  auto transceiver1 = caller->AddTransceiver(audio_track);
  auto transceiver2 = caller->AddTransceiver(audio_track);

  EXPECT_NE(transceiver1, transceiver2);

  auto sender1 = transceiver1->sender();
  auto sender2 = transceiver2->sender();
  EXPECT_NE(sender1, sender2);
  EXPECT_EQ(audio_track, sender1->track());
  EXPECT_EQ(audio_track, sender2->track());

  EXPECT_THAT(caller->pc()->GetTransceivers(),
              UnorderedElementsAre(transceiver1, transceiver2));
  EXPECT_THAT(caller->pc()->GetSenders(),
              UnorderedElementsAre(sender1, sender2));
}

// RtpTransceiver error handling tests.

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTransceiverWithInvalidKindReturnsError) {
  auto caller = CreatePeerConnection();

  auto result = caller->pc()->AddTransceiver(MediaType::DATA);
  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       CanClosePeerConnectionWithoutCrashing) {
  auto caller = CreatePeerConnection();

  caller->pc()->Close();
}

// Unified Plan AddTrack tests.

// Test that adding an audio track creates a new audio RtpSender with the given
// track.
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddAudioTrackCreatesAudioSender) {
  auto caller = CreatePeerConnection();

  auto audio_track = caller->CreateAudioTrack("a");
  auto sender = caller->AddTrack(audio_track);
  ASSERT_TRUE(sender);

  EXPECT_EQ(MediaType::AUDIO, sender->media_type());
  EXPECT_EQ(audio_track, sender->track());
}

// Test that adding a video track creates a new video RtpSender with the given
// track.
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddVideoTrackCreatesVideoSender) {
  auto caller = CreatePeerConnection();

  auto video_track = caller->CreateVideoTrack("a");
  auto sender = caller->AddTrack(video_track);
  ASSERT_TRUE(sender);

  EXPECT_EQ(MediaType::VIDEO, sender->media_type());
  EXPECT_EQ(video_track, sender->track());
}

// Test that adding a track to a new PeerConnection creates an RtpTransceiver
// with the sender that AddTrack returns and in the sendrecv direction.
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddFirstTrackCreatesTransceiver) {
  auto caller = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("a");
  ASSERT_TRUE(sender);

  auto transceivers = caller->pc()->GetTransceivers();
  ASSERT_EQ(1u, transceivers.size());
  EXPECT_EQ(sender, transceivers[0]->sender());
  EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceivers[0]->direction());
}

// Test that if a transceiver of the same type but no track had been added to
// the PeerConnection and later a call to AddTrack is made, the resulting sender
// is the transceiver's sender and the sender's track is the newly-added track.
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackReusesTransceiver) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  auto audio_track = caller->CreateAudioTrack("a");
  auto sender = caller->AddTrack(audio_track);
  ASSERT_TRUE(sender);

  auto transceivers = caller->pc()->GetTransceivers();
  ASSERT_EQ(1u, transceivers.size());
  EXPECT_EQ(transceiver, transceivers[0]);
  EXPECT_EQ(sender, transceiver->sender());
  EXPECT_EQ(audio_track, sender->track());
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTrackWithSendEncodingDoesNotReuseTransceiver) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  auto audio_track = caller->CreateAudioTrack("a");
  RtpEncodingParameters encoding;
  auto sender = caller->AddTrack(audio_track, {}, {encoding});
  ASSERT_TRUE(sender);

  auto transceivers = caller->pc()->GetTransceivers();
  ASSERT_EQ(2u, transceivers.size());
  EXPECT_EQ(transceiver, transceivers[0]);
  EXPECT_NE(sender, transceiver->sender());
  EXPECT_EQ(audio_track, sender->track());
}

// Test that adding two tracks to a new PeerConnection creates two
// RtpTransceivers in the same order.
TEST_F(PeerConnectionRtpTestUnifiedPlan, TwoAddTrackCreatesTwoTransceivers) {
  auto caller = CreatePeerConnection();

  auto sender1 = caller->AddAudioTrack("a");
  auto sender2 = caller->AddVideoTrack("v");
  ASSERT_TRUE(sender2);

  auto transceivers = caller->pc()->GetTransceivers();
  ASSERT_EQ(2u, transceivers.size());
  EXPECT_EQ(sender1, transceivers[0]->sender());
  EXPECT_EQ(sender2, transceivers[1]->sender());
}

// Test that if there are multiple transceivers with no sending track then a
// later call to AddTrack will use the one of the same type as the newly-added
// track.
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackReusesTransceiverOfType) {
  auto caller = CreatePeerConnection();

  auto audio_transceiver = caller->AddTransceiver(MediaType::AUDIO);
  auto video_transceiver = caller->AddTransceiver(MediaType::VIDEO);
  auto sender = caller->AddVideoTrack("v");

  ASSERT_EQ(2u, caller->pc()->GetTransceivers().size());
  EXPECT_NE(sender, audio_transceiver->sender());
  EXPECT_EQ(sender, video_transceiver->sender());
}

// Test that if the only transceivers that do not have a sending track have a
// different type from the added track, then AddTrack will create a new
// transceiver for the track.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTrackDoesNotReuseTransceiverOfWrongType) {
  auto caller = CreatePeerConnection();

  caller->AddTransceiver(MediaType::AUDIO);
  auto sender = caller->AddVideoTrack("v");

  auto transceivers = caller->pc()->GetTransceivers();
  ASSERT_EQ(2u, transceivers.size());
  EXPECT_NE(sender, transceivers[0]->sender());
  EXPECT_EQ(sender, transceivers[1]->sender());
}

// Test that the first available transceiver is reused by AddTrack when multiple
// are available.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTrackReusesFirstMatchingTransceiver) {
  auto caller = CreatePeerConnection();

  caller->AddTransceiver(MediaType::AUDIO);
  caller->AddTransceiver(MediaType::AUDIO);
  auto sender = caller->AddAudioTrack("a");

  auto transceivers = caller->pc()->GetTransceivers();
  ASSERT_EQ(2u, transceivers.size());
  EXPECT_EQ(sender, transceivers[0]->sender());
  EXPECT_NE(sender, transceivers[1]->sender());
}

// Test that a call to AddTrack that reuses a transceiver will change the
// direction from inactive to sendonly.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTrackChangesDirectionFromInactiveToSendOnly) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  RtpTransceiverInit init;
  init.direction = RtpTransceiverDirection::kInactive;
  auto transceiver = caller->AddTransceiver(MediaType::AUDIO, init);
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();
  ASSERT_TRUE(caller->AddAudioTrack("a"));
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  EXPECT_EQ(RtpTransceiverDirection::kSendOnly, transceiver->direction());
}

// Test that a call to AddTrack that reuses a transceiver will change the
// direction from recvonly to sendrecv.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddTrackChangesDirectionFromRecvOnlyToSendRecv) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  RtpTransceiverInit init;
  init.direction = RtpTransceiverDirection::kRecvOnly;
  auto transceiver = caller->AddTransceiver(MediaType::AUDIO, init);
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();
  ASSERT_TRUE(caller->AddAudioTrack("a"));
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
}

TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackCreatesSenderWithTrackId) {
  const std::string kTrackId = "audio_track";

  auto caller = CreatePeerConnection();

  auto audio_track = caller->CreateAudioTrack(kTrackId);
  auto sender = caller->AddTrack(audio_track);

  EXPECT_EQ(kTrackId, sender->id());
}

// Unified Plan AddTrack error handling.

TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackErrorIfClosed) {
  auto caller = CreatePeerConnection();

  auto audio_track = caller->CreateAudioTrack("a");
  caller->pc()->Close();

  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();
  auto result = caller->pc()->AddTrack(audio_track, std::vector<std::string>());
  EXPECT_EQ(RTCErrorType::INVALID_STATE, result.error().type());
  EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
}

TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackErrorIfTrackAlreadyHasSender) {
  auto caller = CreatePeerConnection();

  auto audio_track = caller->CreateAudioTrack("a");
  ASSERT_TRUE(caller->AddTrack(audio_track));

  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();
  auto result = caller->pc()->AddTrack(audio_track, std::vector<std::string>());
  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
  EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
}

// Unified Plan RemoveTrack tests.

// Test that calling RemoveTrack on a sender with a previously-added track
// clears the sender's track.
TEST_F(PeerConnectionRtpTestUnifiedPlan, RemoveTrackClearsSenderTrack) {
  auto caller = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("a");
  ASSERT_TRUE(caller->pc()->RemoveTrackOrError(sender).ok());

  EXPECT_FALSE(sender->track());
}

// Test that calling RemoveTrack on a sender where the transceiver is configured
// in the sendrecv direction changes the transceiver's direction to recvonly.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       RemoveTrackChangesDirectionFromSendRecvToRecvOnly) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  RtpTransceiverInit init;
  init.direction = RtpTransceiverDirection::kSendRecv;
  auto transceiver =
      caller->AddTransceiver(caller->CreateAudioTrack("a"), init);
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();

  ASSERT_TRUE(caller->pc()->RemoveTrackOrError(transceiver->sender()).ok());
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, transceiver->direction());
}

// Test that calling RemoveTrack on a sender where the transceiver is configured
// in the sendonly direction changes the transceiver's direction to inactive.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       RemoveTrackChangesDirectionFromSendOnlyToInactive) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  RtpTransceiverInit init;
  init.direction = RtpTransceiverDirection::kSendOnly;
  auto transceiver =
      caller->AddTransceiver(caller->CreateAudioTrack("a"), init);
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();

  ASSERT_TRUE(caller->pc()->RemoveTrackOrError(transceiver->sender()).ok());
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  EXPECT_EQ(RtpTransceiverDirection::kInactive, transceiver->direction());
}

// Test that calling RemoveTrack with a sender that has a null track results in
// no change in state.
TEST_F(PeerConnectionRtpTestUnifiedPlan, RemoveTrackWithNullSenderTrackIsNoOp) {
  auto caller = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("a");
  auto transceiver = caller->pc()->GetTransceivers()[0];
  ASSERT_TRUE(sender->SetTrack(nullptr));

  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();
  ASSERT_TRUE(caller->pc()->RemoveTrackOrError(sender).ok());
  EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());

  EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
}

// Unified Plan RemoveTrack error handling.

TEST_F(PeerConnectionRtpTestUnifiedPlan, RemoveTrackErrorIfClosed) {
  auto caller = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("a");
  caller->pc()->Close();

  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();
  EXPECT_FALSE(caller->pc()->RemoveTrackOrError(sender).ok());
  EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       RemoveTrackNoErrorIfTrackAlreadyRemoved) {
  auto caller = CreatePeerConnection();

  auto sender = caller->AddAudioTrack("a");
  ASSERT_TRUE(caller->pc()->RemoveTrackOrError(sender).ok());

  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();
  EXPECT_TRUE(caller->pc()->RemoveTrackOrError(sender).ok());
  EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
}

// Test that setting offers that add/remove/add a track repeatedly without
// setting the appropriate answer in between works.
// These are regression tests for bugs.webrtc.org/9401
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddRemoveAddTrackOffersWorksAudio) {
  auto caller = CreatePeerConnection();

  auto sender1 = caller->AddAudioTrack("audio1");
  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));

  caller->pc()->RemoveTrackOrError(sender1);
  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));

  // This will re-use the transceiver created by the first AddTrack.
  auto sender2 = caller->AddAudioTrack("audio2");
  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));

  EXPECT_EQ(1u, caller->pc()->GetTransceivers().size());
  EXPECT_EQ(sender1, sender2);
}
TEST_F(PeerConnectionRtpTestUnifiedPlan, AddRemoveAddTrackOffersWorksVideo) {
  auto caller = CreatePeerConnection();

  auto sender1 = caller->AddVideoTrack("video1");
  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));

  caller->pc()->RemoveTrackOrError(sender1);
  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));

  // This will re-use the transceiver created by the first AddTrack.
  auto sender2 = caller->AddVideoTrack("video2");
  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));

  EXPECT_EQ(1u, caller->pc()->GetTransceivers().size());
  EXPECT_EQ(sender1, sender2);
}

// Test that CreateOffer succeeds if two tracks with the same label are added.
TEST_F(PeerConnectionRtpTestUnifiedPlan, CreateOfferSameTrackLabel) {
  auto caller = CreatePeerConnection();

  auto audio_sender = caller->AddAudioTrack("track", {});
  auto video_sender = caller->AddVideoTrack("track", {});

  EXPECT_TRUE(caller->CreateOffer());

  EXPECT_EQ(audio_sender->track()->id(), video_sender->track()->id());
  EXPECT_NE(audio_sender->id(), video_sender->id());
}

// Test that CreateAnswer succeeds if two tracks with the same label are added.
TEST_F(PeerConnectionRtpTestUnifiedPlan, CreateAnswerSameTrackLabel) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  RtpTransceiverInit recvonly;
  recvonly.direction = RtpTransceiverDirection::kRecvOnly;
  caller->AddTransceiver(MediaType::AUDIO, recvonly);
  caller->AddTransceiver(MediaType::VIDEO, recvonly);

  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));

  auto audio_sender = callee->AddAudioTrack("track", {});
  auto video_sender = callee->AddVideoTrack("track", {});

  EXPECT_TRUE(callee->CreateAnswer());

  EXPECT_EQ(audio_sender->track()->id(), video_sender->track()->id());
  EXPECT_NE(audio_sender->id(), video_sender->id());
}

// Test that calling AddTrack, RemoveTrack and AddTrack again creates a second
// m= section with a random sender id (different from the first, now rejected,
// m= section).
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       AddRemoveAddTrackGeneratesNewSenderId) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto track = caller->CreateVideoTrack("video");
  auto sender1 = caller->AddTrack(track);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  caller->pc()->RemoveTrackOrError(sender1);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  auto sender2 = caller->AddTrack(track);

  EXPECT_NE(sender1, sender2);
  EXPECT_NE(sender1->id(), sender2->id());
  std::string sender2_id = sender2->id();

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  // The sender's ID should not change after negotiation.
  EXPECT_EQ(sender2_id, sender2->id());
}

// Test that OnRenegotiationNeeded is fired if SetDirection is called on an
// active RtpTransceiver with a new direction.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       RenegotiationNeededAfterTransceiverSetDirection) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();
  EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();

  transceiver->SetDirectionWithError(RtpTransceiverDirection::kInactive);
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());
}

// Test that OnRenegotiationNeeded is not fired if SetDirection is called on an
// active RtpTransceiver with current direction.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       NoRenegotiationNeededAfterTransceiverSetSameDirection) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);

  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();
  transceiver->SetDirectionWithError(transceiver->direction());
  EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
}

// Test that OnRenegotiationNeeded is not fired if SetDirection is called on a
// stopped RtpTransceiver.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       NoRenegotiationNeededAfterSetDirectionOnStoppedTransceiver) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  transceiver->StopInternal();

  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();
  transceiver->SetDirectionWithError(RtpTransceiverDirection::kInactive);
  EXPECT_FALSE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_FALSE(caller->observer()->has_negotiation_needed_event());
}

// Test that currentDirection returnes "stopped" if the transceiver was stopped.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       CheckStoppedCurrentDirectionOnStoppedTransceiver) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  transceiver->StopInternal();

  EXPECT_TRUE(transceiver->stopping());
  EXPECT_TRUE(transceiver->stopped());
  EXPECT_EQ(RtpTransceiverDirection::kStopped,
            transceiver->current_direction());
}

// Test that InvalidState is thrown on a stopping transceiver.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       CheckForInvalidStateOnStoppingTransceiver) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  transceiver->StopStandard();

  EXPECT_TRUE(transceiver->stopping());
  EXPECT_FALSE(transceiver->stopped());
  EXPECT_EQ(
      RTCErrorType::INVALID_STATE,
      transceiver->SetDirectionWithError(RtpTransceiverDirection::kInactive)
          .type());
}

// Test that InvalidState is thrown on a stopped transceiver.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       CheckForInvalidStateOnStoppedTransceiver) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  transceiver->StopInternal();

  EXPECT_TRUE(transceiver->stopping());
  EXPECT_TRUE(transceiver->stopped());
  EXPECT_EQ(
      RTCErrorType::INVALID_STATE,
      transceiver->SetDirectionWithError(RtpTransceiverDirection::kInactive)
          .type());
}

// Test that TypeError is thrown if the direction is set to "stopped".
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       CheckForTypeErrorForStoppedOnTransceiver) {
  auto caller = CreatePeerConnection();

  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  EXPECT_EQ(
      RTCErrorType::INVALID_PARAMETER,
      transceiver->SetDirectionWithError(RtpTransceiverDirection::kStopped)
          .type());
}

// Test that you can do createOffer/setLocalDescription with a stopped
// media section.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       SetLocalDescriptionWithStoppedMediaSection) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();
  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  callee->pc()->GetTransceivers()[0]->StopStandard();
  ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
  EXPECT_EQ(RtpTransceiverDirection::kStopped,
            transceiver->current_direction());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       StopAndNegotiateCausesTransceiverToDisappear) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();
  auto transceiver = caller->AddTransceiver(MediaType::AUDIO);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  callee->pc()->GetTransceivers()[0]->StopStandard();
  ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
  EXPECT_EQ(RtpTransceiverDirection::kStopped,
            transceiver->current_direction());
  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());
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       SetLocalDescriptionWorksAfterRepeatedAddRemove) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();
  auto video_track = caller->CreateVideoTrack("v");
  auto track = caller->CreateAudioTrack("a");
  caller->AddTransceiver(video_track);
  auto transceiver = caller->AddTransceiver(track);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->pc()->RemoveTrackOrError(transceiver->sender());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->AddTrack(track);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->pc()->RemoveTrackOrError(transceiver->sender());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
}

// This is a repro of Chromium bug https://crbug.com/1134686
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       SetLocalDescriptionWorksAfterRepeatedAddRemoveWithRemoteReject) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();
  auto video_track = caller->CreateVideoTrack("v");
  auto track = caller->CreateAudioTrack("a");
  caller->AddTransceiver(video_track);
  auto transceiver = caller->AddTransceiver(track);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->pc()->RemoveTrackOrError(transceiver->sender());
  ExchangeOfferAnswerWhereRemoteStopsTransceiver(caller.get(), callee.get(), 1);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->AddTrack(track);
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->pc()->RemoveTrackOrError(transceiver->sender());
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
}

// Test that AddTransceiver fails if trying to use unimplemented RTP encoding
// parameters with the send_encodings parameters.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       CheckForUnsupportedEncodingParameters) {
  auto caller = CreatePeerConnection();

  RtpTransceiverInit init;
  init.send_encodings.emplace_back();

  auto default_send_encodings = init.send_encodings;

  // Unimplemented RtpParameters: ssrc, codec_payload_type, fec, rtx, dtx,
  // ptime, scale_framerate_down_by, dependency_rids.
  init.send_encodings[0].ssrc = 1;
  EXPECT_EQ(
      RTCErrorType::UNSUPPORTED_PARAMETER,
      caller->pc()->AddTransceiver(MediaType::AUDIO, init).error().type());
  init.send_encodings = default_send_encodings;
}

// Test that AddTransceiver fails if trying to use invalid RTP encoding
// parameters with the send_encodings parameters.
TEST_F(PeerConnectionRtpTestUnifiedPlan, CheckForInvalidEncodingParameters) {
  auto caller = CreatePeerConnection();

  RtpTransceiverInit init;
  init.send_encodings.emplace_back();

  auto default_send_encodings = init.send_encodings;

  init.send_encodings[0].scale_resolution_down_by = 0.5;
  EXPECT_EQ(
      RTCErrorType::INVALID_RANGE,
      caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type());
  init.send_encodings = default_send_encodings;

  init.send_encodings[0].bitrate_priority = 0;
  EXPECT_EQ(
      RTCErrorType::INVALID_RANGE,
      caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type());
  init.send_encodings = default_send_encodings;

  init.send_encodings[0].min_bitrate_bps = 200000;
  init.send_encodings[0].max_bitrate_bps = 100000;
  EXPECT_EQ(
      RTCErrorType::INVALID_RANGE,
      caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type());
  init.send_encodings = default_send_encodings;

  init.send_encodings[0].num_temporal_layers = 0;
  EXPECT_EQ(
      RTCErrorType::INVALID_RANGE,
      caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type());
  init.send_encodings = default_send_encodings;

  init.send_encodings[0].num_temporal_layers = 5;
  EXPECT_EQ(
      RTCErrorType::INVALID_RANGE,
      caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type());
  init.send_encodings = default_send_encodings;

  init.send_encodings[0].scalability_mode = std::nullopt;
  init.send_encodings[0].codec =
      CreateVideoCodec(SdpVideoFormat("VP8", {})).ToCodecParameters();
  EXPECT_EQ(
      RTCErrorType::NONE,
      caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type());
  init.send_encodings = default_send_encodings;

  init.send_encodings[0].scalability_mode = "L1T2";
  init.send_encodings[0].codec =
      CreateVideoCodec(SdpVideoFormat("VP8", {})).ToCodecParameters();
  EXPECT_EQ(
      RTCErrorType::NONE,
      caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type());
  init.send_encodings = default_send_encodings;

  init.send_encodings[0].scalability_mode = "L2T2";
  init.send_encodings[0].codec =
      CreateVideoCodec(SdpVideoFormat("VP8", {})).ToCodecParameters();
  EXPECT_EQ(
      RTCErrorType::UNSUPPORTED_OPERATION,
      caller->pc()->AddTransceiver(MediaType::VIDEO, init).error().type());
  init.send_encodings = default_send_encodings;
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       DefaultsMissingScaleResolutionDownByToOne) {
  auto caller = CreatePeerConnection();

  RtpTransceiverInit init;
  init.send_encodings.resize(2);
  init.send_encodings[1].scale_resolution_down_by = 3.0;

  auto result = caller->pc()->AddTransceiver(MediaType::VIDEO, init);
  ASSERT_TRUE(result.ok());

  const auto parameters = result.value()->sender()->GetParameters();
  ASSERT_EQ(parameters.encodings.size(), 2u);
  EXPECT_THAT(parameters.encodings[0].scale_resolution_down_by, Eq(1.0));
  EXPECT_THAT(parameters.encodings[1].scale_resolution_down_by, Eq(3.0));
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       DefaultsScaleResolutionDownByToPowersOfTwo) {
  auto caller = CreatePeerConnection();

  RtpTransceiverInit init;
  init.send_encodings.resize(2);

  auto result = caller->pc()->AddTransceiver(MediaType::VIDEO, init);
  ASSERT_TRUE(result.ok());

  const auto parameters = result.value()->sender()->GetParameters();
  ASSERT_EQ(parameters.encodings.size(), 2u);
  EXPECT_THAT(parameters.encodings[0].scale_resolution_down_by, Eq(2.0));
  EXPECT_THAT(parameters.encodings[1].scale_resolution_down_by, Eq(1.0));
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       DefaultsScaleResolutionDownByBeforeTrimmingEncodings) {
  auto caller = CreatePeerConnection();

  RtpTransceiverInit init;
  init.send_encodings.resize(kMaxSimulcastStreams + 1);
  init.send_encodings.back().scale_resolution_down_by = 3.0;

  auto result = caller->pc()->AddTransceiver(MediaType::VIDEO, init);
  ASSERT_TRUE(result.ok());

  const auto parameters = result.value()->sender()->GetParameters();
  ASSERT_EQ(parameters.encodings.size(), kMaxSimulcastStreams);
  for (const RtpEncodingParameters& encoding : parameters.encodings) {
    EXPECT_THAT(encoding.scale_resolution_down_by, Eq(1.0));
  }
}

// Test that AddTransceiver transfers the send_encodings to the sender and they
// are retained after SetLocalDescription().
TEST_F(PeerConnectionRtpTestUnifiedPlan, SendEncodingsPassedToSender) {
  auto caller = CreatePeerConnection();

  RtpTransceiverInit init;
  init.send_encodings.emplace_back();
  init.send_encodings[0].active = false;
  init.send_encodings[0].max_bitrate_bps = 180000;

  auto result = caller->pc()->AddTransceiver(MediaType::AUDIO, init);
  ASSERT_TRUE(result.ok());

  auto init_send_encodings = result.value()->sender()->init_send_encodings();
  EXPECT_FALSE(init_send_encodings[0].active);
  EXPECT_EQ(init_send_encodings[0].max_bitrate_bps, 180000);

  auto parameters = result.value()->sender()->GetParameters();
  EXPECT_FALSE(parameters.encodings[0].active);
  EXPECT_EQ(parameters.encodings[0].max_bitrate_bps, 180000);

  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));

  parameters = result.value()->sender()->GetParameters();
  EXPECT_FALSE(parameters.encodings[0].active);
  EXPECT_EQ(parameters.encodings[0].max_bitrate_bps, 180000);
}

// Test MSID signaling between Unified Plan and Plan B endpoints. There are two
// options for this kind of signaling: media section based (a=msid) and ssrc
// based (a=ssrc MSID). While JSEP only specifies media section MSID signaling,
// we want to ensure compatibility with older Plan B endpoints that might expect
// ssrc based MSID signaling. Thus we test here that Unified Plan offers both
// types but answers with the same type as the offer.

class PeerConnectionMsidSignalingTest
    : public PeerConnectionRtpTestUnifiedPlan {};

TEST_F(PeerConnectionMsidSignalingTest, UnifiedPlanTalkingToOurself) {
  auto caller = CreatePeerConnectionWithUnifiedPlan();
  caller->AddAudioTrack("caller_audio");
  auto callee = CreatePeerConnectionWithUnifiedPlan();
  callee->AddAudioTrack("callee_audio");

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  // Offer should have had both a=msid and a=ssrc MSID lines.
  auto* offer = callee->pc()->remote_description();
  EXPECT_EQ((kMsidSignalingSemantic | kMsidSignalingMediaSection |
             kMsidSignalingSsrcAttribute),
            offer->description()->msid_signaling());

  // Answer should have had only a=msid lines.
  auto* answer = caller->pc()->remote_description();
  EXPECT_EQ(kMsidSignalingSemantic | kMsidSignalingMediaSection,
            answer->description()->msid_signaling());
}

TEST_F(PeerConnectionMsidSignalingTest, PlanBOfferToUnifiedPlanAnswer) {
  auto caller = CreatePeerConnectionWithPlanB();
  caller->AddAudioTrack("caller_audio");
  auto callee = CreatePeerConnectionWithUnifiedPlan();
  callee->AddAudioTrack("callee_audio");

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  // Offer should have only a=ssrc MSID lines.
  auto* offer = callee->pc()->remote_description();
  EXPECT_EQ(kMsidSignalingSemantic | kMsidSignalingSsrcAttribute,
            offer->description()->msid_signaling());

  // Answer should have only a=ssrc MSID lines to match the offer.
  auto* answer = caller->pc()->remote_description();
  EXPECT_EQ(kMsidSignalingSemantic | kMsidSignalingSsrcAttribute,
            answer->description()->msid_signaling());
}

// This tests that a Plan B endpoint appropriately sets the remote description
// from a Unified Plan offer. When the Unified Plan offer contains a=msid lines
// that signal no stream ids or multiple stream ids we expect that the Plan B
// endpoint always has exactly one media stream per track.
TEST_F(PeerConnectionMsidSignalingTest, UnifiedPlanToPlanBAnswer) {
  const std::string kStreamId1 = "audio_stream_1";
  const std::string kStreamId2 = "audio_stream_2";

  auto caller = CreatePeerConnectionWithUnifiedPlan();
  caller->AddAudioTrack("caller_audio", {kStreamId1, kStreamId2});
  caller->AddVideoTrack("caller_video", {});
  auto callee = CreatePeerConnectionWithPlanB();
  callee->AddAudioTrack("callee_audio");
  caller->AddVideoTrack("callee_video");

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  // Offer should have had both a=msid and a=ssrc MSID lines.
  auto* offer = callee->pc()->remote_description();
  EXPECT_EQ((kMsidSignalingSemantic | kMsidSignalingMediaSection |
             kMsidSignalingSsrcAttribute),
            offer->description()->msid_signaling());

  // Callee should always have 1 stream for all of it's receivers.
  const auto& track_events = callee->observer()->add_track_events_;
  ASSERT_EQ(2u, track_events.size());
  ASSERT_EQ(1u, track_events[0].streams.size());
  EXPECT_EQ(kStreamId1, track_events[0].streams[0]->id());
  ASSERT_EQ(1u, track_events[1].streams.size());
  // This autogenerated a stream id for the empty one signalled.
  EXPECT_FALSE(track_events[1].streams[0]->id().empty());
}

TEST_F(PeerConnectionMsidSignalingTest, PureUnifiedPlanToUs) {
  // Munging allowed: kUnknownModification (MSID signaling semantic
  // modification)
  auto caller = CreatePeerConnectionWithUnifiedPlan(
      "WebRTC-NoSdpMangleAllowForTesting/Enabled,1/");
  caller->AddAudioTrack("caller_audio");
  auto callee = CreatePeerConnectionWithUnifiedPlan();
  callee->AddAudioTrack("callee_audio");

  std::unique_ptr<SessionDescriptionInterface> offer = caller->CreateOffer();
  // Simulate a pure Unified Plan offerer by setting the MSID signaling to media
  // section only.
  offer->description()->set_msid_signaling(kMsidSignalingSemantic |
                                           kMsidSignalingMediaSection);

  ASSERT_TRUE(caller->SetLocalDescription(offer->Clone()));
  ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));

  // Answer should have only a=msid to match the offer.
  std::unique_ptr<SessionDescriptionInterface> answer = callee->CreateAnswer();
  EXPECT_EQ(kMsidSignalingSemantic | kMsidSignalingMediaSection,
            answer->description()->msid_signaling());
}

// Sender setups in a call.

TEST_P(PeerConnectionRtpTest, CreateTwoSendersWithSameTrack) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto track = caller->CreateAudioTrack("audio_track");
  auto sender1 = caller->AddTrack(track);
  ASSERT_TRUE(sender1);
  // We need to temporarily reset the track for the subsequent AddTrack() to
  // succeed.
  EXPECT_TRUE(sender1->SetTrack(nullptr));
  auto sender2 = caller->AddTrack(track);
  EXPECT_TRUE(sender2);
  EXPECT_TRUE(sender1->SetTrack(track.get()));

  if (sdp_semantics_ == SdpSemantics::kPlanB_DEPRECATED) {
    // TODO(hbos): When https://crbug.com/webrtc/8734 is resolved, this should
    // return true, and doing `callee->SetRemoteDescription()` should work.
    EXPECT_FALSE(caller->CreateOfferAndSetAsLocal());
  } else {
    EXPECT_TRUE(caller->CreateOfferAndSetAsLocal());
  }
}

// This test exercises the code path that fires a NegotiationNeeded
// notification when the stream IDs of the local description differ from
// the ones in the transceiver.
TEST_F(PeerConnectionRtpTestUnifiedPlan,
       ChangeAssociatedStreamsTriggersRenegotiation) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  RtpTransceiverInit init;
  init.direction = RtpTransceiverDirection::kSendRecv;
  auto transceiver =
      caller->AddTransceiver(caller->CreateAudioTrack("a"), init);
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
  caller->observer()->clear_legacy_renegotiation_needed();
  caller->observer()->clear_latest_negotiation_needed_event();

  transceiver->sender()->SetStreams({"stream3", "stream4", "stream5"});
  EXPECT_TRUE(caller->observer()->legacy_renegotiation_needed());
  EXPECT_TRUE(caller->observer()->has_negotiation_needed_event());

  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
  auto callee_streams = callee->pc()->GetReceivers()[0]->streams();
  ASSERT_EQ(3u, callee_streams.size());
  EXPECT_EQ("stream3", callee_streams[0]->id());
  EXPECT_EQ("stream4", callee_streams[1]->id());
  EXPECT_EQ("stream5", callee_streams[2]->id());
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       SendParamsCorrectWhenSendCodecChanges) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto sender = caller->AddVideoTrack("video_track");
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  // Call GetParameters() to populate the cache.
  auto parameters1 = sender->GetParameters();
  ASSERT_GT(parameters1.codecs.size(), 0u);

  // Force renegotiation to change the negotiated codec.
  auto transceiver = callee->pc()->GetTransceivers()[0];
  auto capabilities =
      caller->pc_factory()->GetRtpReceiverCapabilities(MediaType::VIDEO);
  std::vector<RtpCodecCapability> codecs = capabilities.codecs;
  // Remove the currently negotiated codec from the receiver preferences
  codecs.erase(std::remove_if(codecs.begin(), codecs.end(),
                              [&](const RtpCodecCapability& c) {
                                return c.name == parameters1.codecs[0].name;
                              }),
               codecs.end());

  auto error = transceiver->SetCodecPreferences(codecs);
  EXPECT_TRUE(error.ok());

  // Exchange offer/answer again. This forces WebRtcVideoSendChannel to
  // change the send codec without notifying RtpSender!
  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  // Call GetParameters(). The DCHECK evaluates cached_parameters_ against
  // what's actually on the worker thread. Since we bypassed SetParameters,
  // this should crash if the bug is present, or return the right values.
  auto parameters2 = sender->GetParameters();
  ASSERT_GT(parameters2.codecs.size(), 0u);
  EXPECT_NE(parameters1.codecs[0].name, parameters2.codecs[0].name);
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       SendParamsCorrectWhenContentHintChanges) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto track = caller->CreateVideoTrack("video");

  RtpTransceiverInit init;
  init.direction = RtpTransceiverDirection::kSendRecv;
  RtpEncodingParameters encoding;
  encoding.active = true;
  // Set an unsupported scalability mode so that it gets mutated
  // by FallbackToDefaultScalabilityModeIfNotSupported later.
  encoding.scalability_mode = "L3T3_KEY";
  init.send_encodings.push_back(encoding);

  auto transceiver = caller->AddTransceiver(track, init);
  ASSERT_TRUE(transceiver != nullptr);
  auto sender = transceiver->sender();

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  // Get initial parameters to populate the signaling thread cache.
  auto parameters = sender->GetParameters();

  // Change content hint, which triggers VideoRtpSender::OnChanged() ->
  // SetVideoSend() -> WebRtcVideoSendStream::SetOptions() ->
  // ReconfigureEncoder() -> FallbackToDefaultScalabilityModeIfNotSupported()
  // on the worker thread.
  track->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);

  // Calling GetParameters will fetch the new parameters from the worker thread
  // and compare them to the signaling thread cache, triggering the DCHECK if
  // the cache is stale.
  auto parameters_after = sender->GetParameters();

  // If the DCHECK is not fatal, we manually verify the mismatch happened.
  EXPECT_EQ(parameters_after.encodings.size(), 1u);
}

TEST_F(PeerConnectionRtpTestUnifiedPlan,
       CacheMismatchWhenEncoderSelectorChanges) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto track = caller->CreateVideoTrack("video");

  auto transceiver = caller->AddTransceiver(track);
  ASSERT_TRUE(transceiver != nullptr);
  auto sender = transceiver->sender();

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  // Get initial parameters to populate the signaling thread cache.
  auto parameters = sender->GetParameters();

  class MockEncoderSelector
      : public VideoEncoderFactory::EncoderSelectorInterface {
   public:
    void OnCurrentEncoder(const SdpVideoFormat& format) override {}
    std::optional<SdpVideoFormat> OnAvailableBitrate(
        const DataRate& rate) override {
      return SdpVideoFormat("VP8");
    }
    std::optional<SdpVideoFormat> OnEncoderBroken() override {
      return std::nullopt;
    }
    std::optional<SdpVideoFormat> OnResolutionChange(
        const RenderResolution& resolution) override {
      return std::nullopt;
    }
  };

  scoped_refptr<MockEncoderSelector> encoder_selector =
      make_ref_counted<MockEncoderSelector>();
  auto format_to_switch_to =
      *encoder_selector->OnAvailableBitrate(DataRate::Zero());

  // Change encoder selector
  sender->SetEncoderSelector(encoder_selector);

  auto transceivers =
      caller->GetInternalPeerConnection()->GetTransceiversInternal();
  ASSERT_FALSE(transceivers.empty());
  auto* transceiver_internal = transceivers[0]->internal();
  ASSERT_TRUE(transceiver_internal != nullptr);

  auto* media_channel = transceiver_internal->media_send_channel();
  ASSERT_TRUE(media_channel != nullptr);

  auto* worker_thread = caller->GetInternalPeerConnection()->worker_thread();
  ASSERT_TRUE(worker_thread != nullptr);

  worker_thread->BlockingCall([&] {
    // Simulate VideoStreamEncoder calling RequestEncoderSwitch after getting
    // the format from the EncoderSelector.
    auto* video_send_channel = static_cast<webrtc::WebRtcVideoSendChannel*>(
        media_channel->AsVideoSendChannel());
    video_send_channel->RequestEncoderSwitch(std::move(format_to_switch_to),
                                             /*allow_default_fallback=*/false);
  });

  // Allow the signaling thread to process the cache invalidation task posted
  // by RequestEncoderSwitch.
  run_loop_.Flush();

  // Calling GetParameters will fetch the new parameters from the worker thread
  // and compare them to the signaling thread cache, triggering the DCHECK if
  // the cache is stale.
  auto parameters_after = sender->GetParameters();

  EXPECT_EQ(parameters_after.encodings.size(), 1u);
}

TEST_F(PeerConnectionRtpTestUnifiedPlan, SendParamsCorrectWhenEncoderFallback) {
  auto caller = CreatePeerConnection();
  auto callee = CreatePeerConnection();

  auto track = caller->CreateVideoTrack("video");

  auto transceiver = caller->AddTransceiver(track);
  ASSERT_TRUE(transceiver != nullptr);
  auto sender = transceiver->sender();

  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));

  // Get initial parameters to populate the signaling thread cache.
  auto parameters = sender->GetParameters();

  auto transceivers =
      caller->GetInternalPeerConnection()->GetTransceiversInternal();
  ASSERT_FALSE(transceivers.empty());
  auto* transceiver_internal = transceivers[0]->internal();
  ASSERT_TRUE(transceiver_internal != nullptr);

  auto* media_channel = transceiver_internal->media_send_channel();
  ASSERT_TRUE(media_channel != nullptr);

  auto* worker_thread = caller->GetInternalPeerConnection()->worker_thread();
  ASSERT_TRUE(worker_thread != nullptr);

  worker_thread->BlockingCall([&] {
    // Simulate an encoder fallback on the worker thread.
    auto* video_send_channel = static_cast<webrtc::WebRtcVideoSendChannel*>(
        media_channel->AsVideoSendChannel());
    video_send_channel->RequestEncoderSwitch(std::nullopt, true);
  });

  // Allow the signaling thread to process the cache invalidation task posted
  // by RequestEncoderSwitch(std::nullopt, true).
  run_loop_.Flush();

  // Call GetParameters. This triggers an internal consistency check in the
  // `GetParameters()` implementation.
  auto parameters_after = sender->GetParameters();
  EXPECT_EQ(parameters_after.codecs.size(), parameters.codecs.size());
}

RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN()
INSTANTIATE_TEST_SUITE_P(PeerConnectionRtpTest,
                         PeerConnectionRtpTest,
                         Values(SdpSemantics::kPlanB_DEPRECATED,
                                SdpSemantics::kUnifiedPlan));
RTC_ALLOW_PLAN_B_DEPRECATION_END()

}  // namespace webrtc