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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 17:32:43 +0000 |
commit | 6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch) | |
tree | a68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc | |
parent | Initial commit. (diff) | |
download | thunderbird-upstream.tar.xz thunderbird-upstream.zip |
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc | 1967 |
1 files changed, 1967 insertions, 0 deletions
diff --git a/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc b/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc new file mode 100644 index 0000000000..e5861ea815 --- /dev/null +++ b/third_party/libwebrtc/pc/peer_connection_rtp_unittest.cc @@ -0,0 +1,1967 @@ +/* + * 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 <stddef.h> + +#include <cstdint> +#include <memory> +#include <string> +#include <utility> +#include <vector> + +#include "absl/types/optional.h" +#include "api/audio/audio_mixer.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/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/uma_metrics.h" +#include "api/video_codecs/builtin_video_decoder_factory.h" +#include "api/video_codecs/builtin_video_encoder_factory.h" +#include "media/base/stream_params.h" +#include "modules/audio_device/include/audio_device.h" +#include "modules/audio_processing/include/audio_processing.h" +#include "p2p/base/port_allocator.h" +#include "pc/media_session.h" +#include "pc/peer_connection_wrapper.h" +#include "pc/sdp_utils.h" +#include "pc/session_description.h" +#include "pc/test/fake_audio_capture_module.h" +#include "pc/test/mock_peer_connection_observers.h" +#include "rtc_base/checks.h" +#include "rtc_base/gunit.h" +#include "rtc_base/rtc_certificate_generator.h" +#include "rtc_base/thread.h" +#include "system_wrappers/include/metrics.h" +#include "test/gmock.h" +#include "test/gtest.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::Pair; +using ::testing::UnorderedElementsAre; +using ::testing::Values; + +const uint32_t kDefaultTimeout = 10000u; + +template <typename MethodFunctor> +class OnSuccessObserver : public webrtc::SetRemoteDescriptionObserverInterface { + public: + explicit OnSuccessObserver(MethodFunctor on_success) + : on_success_(std::move(on_success)) {} + + // webrtc::SetRemoteDescriptionObserverInterface implementation. + void OnSetRemoteDescriptionComplete(webrtc::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(rtc::Thread::Current(), + rtc::Thread::Current(), + rtc::Thread::Current(), + FakeAudioCaptureModule::Create(), + CreateBuiltinAudioEncoderFactory(), + CreateBuiltinAudioDecoderFactory(), + CreateBuiltinVideoEncoderFactory(), + CreateBuiltinVideoDecoderFactory(), + nullptr /* audio_mixer */, + nullptr /* audio_processing */)) { + webrtc::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() { + RTCConfiguration config; + config.sdp_semantics = SdpSemantics::kUnifiedPlan; + return CreatePeerConnectionInternal(config); + } + + std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection( + const RTCConfiguration& config) { + RTCConfiguration modified_config = config; + modified_config.sdp_semantics = sdp_semantics_; + return CreatePeerConnectionInternal(modified_config); + } + + protected: + const SdpSemantics sdp_semantics_; + rtc::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) { + auto observer = std::make_unique<MockPeerConnectionObserver>(); + auto result = pc_factory_->CreatePeerConnectionOrError( + config, PeerConnectionDependencies(observer.get())); + EXPECT_TRUE(result.ok()); + observer->SetPeerConnectionInterface(result.value().get()); + return std::make_unique<PeerConnectionWrapper>( + pc_factory_, result.MoveValue(), std::move(observer)); + } + + rtc::AutoThread main_thread_; +}; + +class PeerConnectionRtpTest + : public PeerConnectionRtpBaseTest, + public ::testing::WithParamInterface<SdpSemantics> { + protected: + PeerConnectionRtpTest() : PeerConnectionRtpBaseTest(GetParam()) {} +}; + +class PeerConnectionRtpTestPlanB : public PeerConnectionRtpBaseTest { + protected: + PeerConnectionRtpTestPlanB() + : PeerConnectionRtpBaseTest(SdpSemantics::kPlanB_DEPRECATED) {} +}; + +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) { + auto offer = caller->CreateOffer(); + caller->SetLocalDescription(CloneSessionDescription(offer.get())); + 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(); + auto answer = callee->CreateAnswer(); + EXPECT_TRUE(answer); + bool set_local_answer = + callee->SetLocalDescription(CloneSessionDescription(answer.get())); + 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 `webrtc::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_); + EXPECT_EQ(0u, callee->observer()->remote_streams()->count()); +} + +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<rtc::scoped_refptr<RtpReceiverInterface>>{ + callee->observer()->add_track_events_[0].receiver}, + callee->observer()->remove_track_events_); + ASSERT_EQ(1u, callee->observer()->remote_streams()->count()); + 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_); + EXPECT_EQ(0u, callee->observer()->remote_streams()->count()); +} + +// Tests the edge case that if a stream ID changes for a given track that both +// OnRemoveTrack and OnAddTrack is fired. +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. + auto offer = caller->CreateOfferAndSetAsLocal(); + auto* audio_desc = + cricket::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); +} + +// 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(cricket::MEDIA_TYPE_AUDIO); + RtpTransceiverInit video_transceiver_init; + video_transceiver_init.stream_ids = {kStreamId1, kStreamId2}; + auto video_transceiver = + caller->AddTransceiver(cricket::MEDIA_TYPE_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<rtc::scoped_refptr<MediaStreamInterface>> audio_streams = + callee->pc()->GetTransceivers()[0]->receiver()->streams(); + std::vector<rtc::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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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. +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), + rtc::make_ref_counted<OnSuccessObserver<decltype(srd1_callback)>>( + srd1_callback)); + callee->pc()->SetRemoteDescription( + std::move(srd2_sdp), + rtc::make_ref_counted<OnSuccessObserver<decltype(srd2_callback)>>( + srd2_callback)); + EXPECT_TRUE_WAIT(srd1_callback_called, kDefaultTimeout); + EXPECT_TRUE_WAIT(srd2_callback_called, kDefaultTimeout); +} + +// 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}); + + auto 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(); + cricket::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(CloneSessionDescription(offer.get()))); + 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; + + auto caller = CreatePeerConnection(); + auto callee = CreatePeerConnection(); + + // Caller offers to receive audio and video. + RtpTransceiverInit init; + init.direction = RtpTransceiverDirection::kRecvOnly; + caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init); + caller->AddTransceiver(cricket::MEDIA_TYPE_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())); + auto 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); + mutable_streams[0].ssrcs = {kFirstMungedSsrc + static_cast<uint32_t>(i)}; + } + ASSERT_TRUE( + callee->SetLocalDescription(CloneSessionDescription(answer.get()))); + ASSERT_TRUE( + caller->SetRemoteDescription(CloneSessionDescription(answer.get()))); + + // 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. + auto 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. +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}); + + auto offer = caller->CreateOfferAndSetAsLocal(); + auto mutable_streams = + cricket::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(CloneSessionDescription(offer.get()))); + + 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()); +} + +// 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(); + + rtc::scoped_refptr<webrtc::MockSetSessionDescriptionObserver> observer = + rtc::make_ref_counted<webrtc::MockSetSessionDescriptionObserver>(); + + auto offer = caller->CreateOfferAndSetAsLocal(); + callee->pc()->SetRemoteDescription(observer.get(), offer.release()); + callee = nullptr; + rtc::Thread::Current()->ProcessMessages(0); + 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(cricket::MEDIA_TYPE_AUDIO); + EXPECT_EQ(absl::nullopt, transceiver->mid()); + EXPECT_FALSE(transceiver->stopped()); + EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction()); + EXPECT_EQ(absl::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(cricket::MEDIA_TYPE_AUDIO); + EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->media_type()); + + ASSERT_TRUE(transceiver->sender()); + EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->sender()->media_type()); + + ASSERT_TRUE(transceiver->receiver()); + EXPECT_EQ(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_VIDEO); + EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type()); + + ASSERT_TRUE(transceiver->sender()); + EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->sender()->media_type()); + + ASSERT_TRUE(transceiver->receiver()); + EXPECT_EQ(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_AUDIO); + EXPECT_EQ( + std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>{transceiver}, + caller->pc()->GetTransceivers()); + EXPECT_EQ( + std::vector<rtc::scoped_refptr<RtpSenderInterface>>{ + transceiver->sender()}, + caller->pc()->GetSenders()); + EXPECT_EQ( + std::vector<rtc::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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_AUDIO); + auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_AUDIO); + caller->AddTransceiver(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_AUDIO, recvonly); + caller->AddTransceiver(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_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(cricket::MEDIA_TYPE_VIDEO, init) + .error() + .type()); + init.send_encodings = default_send_encodings; +} + +// 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(cricket::MEDIA_TYPE_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((cricket::kMsidSignalingMediaSection | + cricket::kMsidSignalingSsrcAttribute), + offer->description()->msid_signaling()); + + // Answer should have had only a=msid lines. + auto* answer = caller->pc()->remote_description(); + EXPECT_EQ(cricket::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(cricket::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(cricket::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((cricket::kMsidSignalingMediaSection | + cricket::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) { + auto caller = CreatePeerConnectionWithUnifiedPlan(); + caller->AddAudioTrack("caller_audio"); + auto callee = CreatePeerConnectionWithUnifiedPlan(); + callee->AddAudioTrack("callee_audio"); + + auto offer = caller->CreateOffer(); + // Simulate a pure Unified Plan offerer by setting the MSID signaling to media + // section only. + offer->description()->set_msid_signaling(cricket::kMsidSignalingMediaSection); + + ASSERT_TRUE( + caller->SetLocalDescription(CloneSessionDescription(offer.get()))); + ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer))); + + // Answer should have only a=msid to match the offer. + auto answer = callee->CreateAnswer(); + EXPECT_EQ(cricket::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()); +} + +INSTANTIATE_TEST_SUITE_P(PeerConnectionRtpTest, + PeerConnectionRtpTest, + Values(SdpSemantics::kPlanB_DEPRECATED, + SdpSemantics::kUnifiedPlan)); + +} // namespace webrtc |