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peer_connection_rtp_unittest.cc
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peer_connection_rtp_unittest.cc
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/*
* 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 <stdint.h>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.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_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 "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/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/ref_counted_object.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 rtc::RefCountedObject<
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;
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 pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr,
observer.get());
EXPECT_TRUE(pc.get());
observer->SetPeerConnectionInterface(pc.get());
return std::make_unique<PeerConnectionWrapper>(pc_factory_, pc,
std::move(observer));
}
};
class PeerConnectionRtpTest
: public PeerConnectionRtpBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionRtpTest() : PeerConnectionRtpBaseTest(GetParam()) {}
};
class PeerConnectionRtpTestPlanB : public PeerConnectionRtpBaseTest {
protected:
PeerConnectionRtpTestPlanB()
: PeerConnectionRtpBaseTest(SdpSemantics::kPlanB) {}
};
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) {
// 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()->RemoveTrack(sender));
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()->RemoveTrack(sender));
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()->RemoveTrack(sender1));
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()->RemoveTrack(sender2));
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()->RemoveTrack(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) {
// 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()->RemoveTrack(sender));
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()->RemoveTrack(sender));
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()->RemoveTrack(sender1));
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()->RemoveTrack(sender2));
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()->RemoveTrack(sender));
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),
new OnSuccessObserver<decltype(srd1_callback)>(srd1_callback));
callee->pc()->SetRemoteDescription(
std::move(srd2_sdp),
new 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, 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);