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Convert PeerConnection integration tests to the track-based API

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Bug: webrtc:8742
Change-Id: I4e120f4c7a635201028155486530bb4fbdae2a8b
Reviewed-on: https://webrtc-review.googlesource.com/39386
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21666}
This commit is contained in:
Steve Anton
2018-01-16 10:26:49 -08:00
committed by Commit Bot
parent 772eb2115a
commit 1532477bbd
1 changed files with 165 additions and 224 deletions
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389
pc/peerconnection_integrationtest.cc
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@ -69,6 +69,7 @@ using webrtc::DtmfSender;
using webrtc::DtmfSenderInterface;
using webrtc::DtmfSenderObserverInterface;
using webrtc::FakeConstraints;
using webrtc::FakeVideoTrackRenderer;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
@ -81,10 +82,12 @@ using webrtc::PeerConnection;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionFactory;
using webrtc::PeerConnectionProxy;
using webrtc::RTCErrorType;
using webrtc::RtpReceiverInterface;
using webrtc::SdpType;
using webrtc::SessionDescriptionInterface;
using webrtc::StreamCollectionInterface;
using webrtc::VideoTrackInterface;
namespace {
@ -175,8 +178,7 @@ class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
// TODO(steveanton): See how this could become a subclass of
// PeerConnectionWrapper defined in peerconnectionwrapper.h .
class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
public SignalingMessageReceiver,
public ObserverInterface {
public SignalingMessageReceiver {
public:
// Different factory methods for convenience.
// TODO(deadbeef): Could use the pattern of:
@ -301,19 +303,14 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
return ice_gathering_state_history_;
}
// TODO(deadbeef): Switch the majority of these tests to use AddTrack instead
// of AddStream since AddStream is deprecated.
void AddAudioVideoMediaStream() {
AddMediaStreamFromTracks(CreateLocalAudioTrack(), CreateLocalVideoTrack());
void AddAudioVideoTracks() {
AddAudioTrack();
AddVideoTrack();
}
void AddAudioOnlyMediaStream() {
AddMediaStreamFromTracks(CreateLocalAudioTrack(), nullptr);
}
void AddAudioTrack() { AddTrack(CreateLocalAudioTrack()); }
void AddVideoOnlyMediaStream() {
AddMediaStreamFromTracks(nullptr, CreateLocalVideoTrack());
}
void AddVideoTrack() { AddTrack(CreateLocalVideoTrack()); }
rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() {
FakeConstraints constraints;
@ -342,19 +339,10 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
return CreateLocalVideoTrackInternal(FakeConstraints(), rotation);
}
void AddMediaStreamFromTracks(
const rtc::scoped_refptr<webrtc::AudioTrackInterface>& audio,
const rtc::scoped_refptr<webrtc::VideoTrackInterface>& video) {
rtc::scoped_refptr<MediaStreamInterface> stream =
peer_connection_factory_->CreateLocalMediaStream(
rtc::CreateRandomUuid());
if (audio) {
stream->AddTrack(audio);
}
if (video) {
stream->AddTrack(video);
}
EXPECT_TRUE(pc()->AddStream(stream));
void AddTrack(rtc::scoped_refptr<MediaStreamTrackInterface> track,
const std::vector<std::string>& stream_labels = {}) {
auto result = pc()->AddTrack(track, stream_labels);
EXPECT_EQ(RTCErrorType::NONE, result.error().type());
}
bool SignalingStateStable() {
@ -834,19 +822,26 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
webrtc::PeerConnectionInterface::SignalingState new_state) override {
EXPECT_EQ(pc()->signaling_state(), new_state);
}
void OnAddStream(
rtc::scoped_refptr<MediaStreamInterface> media_stream) override {
media_stream->RegisterObserver(this);
for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) {
const std::string id = media_stream->GetVideoTracks()[i]->id();
ASSERT_TRUE(fake_video_renderers_.find(id) ==
void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
streams) override {
if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
rtc::scoped_refptr<VideoTrackInterface> video_track(
static_cast<VideoTrackInterface*>(receiver->track().get()));
ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) ==
fake_video_renderers_.end());
fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
media_stream->GetVideoTracks()[i]));
fake_video_renderers_[video_track->id()] =
rtc::MakeUnique<FakeVideoTrackRenderer>(video_track);
}
}
void OnRemoveTrack(
rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
auto it = fake_video_renderers_.find(receiver->track()->id());
RTC_DCHECK(it != fake_video_renderers_.end());
fake_video_renderers_.erase(it);
}
}
void OnRemoveStream(
rtc::scoped_refptr<MediaStreamInterface> media_stream) override {}
void OnRenegotiationNeeded() override {}
void OnIceConnectionChange(
webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
@ -876,40 +871,6 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
data_observer_.reset(new MockDataChannelObserver(data_channel));
}
// MediaStreamInterface callback
void OnChanged() override {
// Track added or removed from MediaStream, so update our renderers.
rtc::scoped_refptr<StreamCollectionInterface> remote_streams =
pc()->remote_streams();
// Remove renderers for tracks that were removed.
for (auto it = fake_video_renderers_.begin();
it != fake_video_renderers_.end();) {
if (remote_streams->FindVideoTrack(it->first) == nullptr) {
auto to_remove = it++;
removed_fake_video_renderers_.push_back(std::move(to_remove->second));
fake_video_renderers_.erase(to_remove);
} else {
++it;
}
}
// Create renderers for new video tracks.
for (size_t stream_index = 0; stream_index < remote_streams->count();
++stream_index) {
MediaStreamInterface* remote_stream = remote_streams->at(stream_index);
for (size_t track_index = 0;
track_index < remote_stream->GetVideoTracks().size();
++track_index) {
const std::string id =
remote_stream->GetVideoTracks()[track_index]->id();
if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) {
continue;
}
fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
remote_stream->GetVideoTracks()[track_index]));
}
}
}
std::string debug_name_;
std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
@ -1184,8 +1145,8 @@ class PeerConnectionIntegrationTest : public testing::Test {
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
@ -1216,8 +1177,8 @@ TEST_F(PeerConnectionIntegrationTest,
RtpReceiverObserverOnFirstPacketReceived) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Start offer/answer exchange and wait for it to complete.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -1283,14 +1244,11 @@ class DummyDtmfObserver : public DtmfSenderObserverInterface {
// exchange is done.
void TestDtmfFromSenderToReceiver(PeerConnectionWrapper* sender,
PeerConnectionWrapper* receiver) {
// We should be able to get a DTMF sender from the local sender.
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender =
sender->pc()->GetSenders().at(0)->GetDtmfSender();
ASSERT_TRUE(dtmf_sender);
DummyDtmfObserver observer;
rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender;
// We should be able to create a DTMF sender from a local track.
webrtc::AudioTrackInterface* localtrack =
sender->local_streams()->at(0)->GetAudioTracks()[0];
dtmf_sender = sender->pc()->CreateDtmfSender(localtrack);
ASSERT_NE(nullptr, dtmf_sender.get());
dtmf_sender->RegisterObserver(&observer);
// Test the DtmfSender object just created.
@ -1310,8 +1268,8 @@ TEST_F(PeerConnectionIntegrationTest, DtmfSenderObserver) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Only need audio for DTMF.
caller()->AddAudioOnlyMediaStream();
callee()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// DTLS must finish before the DTMF sender can be used reliably.
@ -1327,8 +1285,8 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
ConnectFakeSignaling();
// Do normal offer/answer and wait for some frames to be received in each
// direction.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@ -1346,8 +1304,8 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
// Do normal offer/answer and wait for some frames to be received in each
// direction.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@ -1385,8 +1343,8 @@ TEST_F(PeerConnectionIntegrationTest,
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(caller()));
EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(callee()));
caller()->AddAudioOnlyMediaStream();
callee()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
@ -1415,10 +1373,10 @@ TEST_F(PeerConnectionIntegrationTest, SendAndReceive16To9AspectRatio) {
FakeConstraints constraints;
double requested_ratio = 16.0 / 9;
constraints.SetMandatoryMinAspectRatio(requested_ratio);
caller()->AddMediaStreamFromTracks(
nullptr, caller()->CreateLocalVideoTrackWithConstraints(constraints));
callee()->AddMediaStreamFromTracks(
nullptr, callee()->CreateLocalVideoTrackWithConstraints(constraints));
caller()->AddTrack(
caller()->CreateLocalVideoTrackWithConstraints(constraints));
callee()->AddTrack(
callee()->CreateLocalVideoTrackWithConstraints(constraints));
// Do normal offer/answer and wait for at least one frame to be received in
// each direction.
@ -1446,10 +1404,10 @@ TEST_F(PeerConnectionIntegrationTest,
FakeConstraints constraints;
constraints.SetMandatoryMinWidth(1280);
constraints.SetMandatoryMinHeight(720);
caller()->AddMediaStreamFromTracks(
nullptr, caller()->CreateLocalVideoTrackWithConstraints(constraints));
callee()->AddMediaStreamFromTracks(
nullptr, callee()->CreateLocalVideoTrackWithConstraints(constraints));
caller()->AddTrack(
caller()->CreateLocalVideoTrackWithConstraints(constraints));
callee()->AddTrack(
callee()->CreateLocalVideoTrackWithConstraints(constraints));
// Do normal offer/answer and wait for at least one frame to be received in
// each direction.
@ -1470,7 +1428,7 @@ TEST_F(PeerConnectionIntegrationTest,
TEST_F(PeerConnectionIntegrationTest, OneWayMediaCall) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
int caller_received_frames = 0;
ExpectNewFramesReceivedWithWait(
@ -1487,8 +1445,8 @@ TEST_F(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
ConnectFakeSignaling();
// Initially, offer an audio/video stream from the caller, but refuse to
// send/receive video on the callee side.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioOnlyMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioTrack();
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_video = 0;
callee()->SetOfferAnswerOptions(options);
@ -1506,7 +1464,7 @@ TEST_F(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
EXPECT_TRUE(callee_video_content->rejected);
// Now negotiate with video and ensure negotiation succeeds, with video
// frames and additional audio frames being received.
callee()->AddVideoOnlyMediaStream();
callee()->AddVideoTrack();
options.offer_to_receive_video = 1;
callee()->SetOfferAnswerOptions(options);
callee()->CreateAndSetAndSignalOffer();
@ -1524,13 +1482,13 @@ TEST_F(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do initial offer/answer with just a video track.
caller()->AddVideoOnlyMediaStream();
callee()->AddVideoOnlyMediaStream();
caller()->AddVideoTrack();
callee()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Now add an audio track and do another offer/answer.
caller()->AddAudioOnlyMediaStream();
callee()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure both audio and video frames are received end-to-end.
@ -1545,8 +1503,8 @@ TEST_F(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
TEST_F(PeerConnectionIntegrationTest, CallTransferredForCallee) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -1560,7 +1518,7 @@ TEST_F(PeerConnectionIntegrationTest, CallTransferredForCallee) {
original_peer->pc()->Close();
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for some additional frames to be transmitted end-to-end.
@ -1575,8 +1533,8 @@ TEST_F(PeerConnectionIntegrationTest, CallTransferredForCallee) {
TEST_F(PeerConnectionIntegrationTest, CallTransferredForCaller) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -1590,7 +1548,7 @@ TEST_F(PeerConnectionIntegrationTest, CallTransferredForCaller) {
original_peer->pc()->Close();
ConnectFakeSignaling();
callee()->AddAudioVideoMediaStream();
callee()->AddAudioVideoTracks();
caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -1608,8 +1566,8 @@ TEST_F(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Remove the bundle group from the SDP received by the callee.
callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
desc->RemoveGroupByName("BUNDLE");
@ -1642,11 +1600,9 @@ TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add rotated video tracks.
caller()->AddMediaStreamFromTracks(
nullptr,
caller()->AddTrack(
caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
callee()->AddMediaStreamFromTracks(
nullptr,
callee()->AddTrack(
callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
// Wait for video frames to be received by both sides.
@ -1674,11 +1630,9 @@ TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add rotated video tracks.
caller()->AddMediaStreamFromTracks(
nullptr,
caller()->AddTrack(
caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
callee()->AddMediaStreamFromTracks(
nullptr,
callee()->AddTrack(
callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
// Remove the CVO extension from the offered SDP.
@ -1716,14 +1670,13 @@ TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
// Only add video track for callee, and set offer_to_receive_audio to 0, so
// it will reject the audio m= section completely.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 0;
callee()->SetOfferAnswerOptions(options);
callee()->AddMediaStreamFromTracks(nullptr,
callee()->CreateLocalVideoTrack());
callee()->AddTrack(callee()->CreateLocalVideoTrack());
// Do offer/answer and wait for successful end-to-end video frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -1746,14 +1699,13 @@ TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) {
TEST_F(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
// Only add audio track for callee, and set offer_to_receive_video to 0, so
// it will reject the video m= section completely.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_video = 0;
callee()->SetOfferAnswerOptions(options);
callee()->AddMediaStreamFromTracks(callee()->CreateLocalAudioTrack(),
nullptr);
callee()->AddTrack(callee()->CreateLocalAudioTrack());
// Do offer/answer and wait for successful end-to-end audio frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -1779,7 +1731,7 @@ TEST_F(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
// Don't give the callee any tracks, and set offer_to_receive_X to 0, so it
// will reject both audio and video m= sections.
PeerConnectionInterface::RTCOfferAnswerOptions options;
@ -1808,8 +1760,8 @@ TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
TEST_F(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@ -1860,8 +1812,8 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add audio and video, testing that packets can be demuxed on payload type.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Remove SSRCs and MSIDs from the received offer SDP.
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
@ -1878,11 +1830,11 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Add one audio/video stream, and one video-only stream.
caller()->AddAudioVideoMediaStream();
caller()->AddVideoOnlyMediaStream();
caller()->AddAudioVideoTracks();
caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_EQ(2u, callee()->number_of_remote_streams());
ASSERT_EQ(3u, callee()->pc()->GetReceivers().size());
int expected_callee_received_frames = kDefaultExpectedVideoFrameCount;
ExpectNewFramesReceivedWithWait(0, 0, 0, expected_callee_received_frames,
kMaxWaitForFramesMs);
@ -1921,8 +1873,8 @@ TEST_F(PeerConnectionIntegrationTest,
EndToEndCallWithSpecCompliantMaxBundleOffer) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Do the equivalent of setting the port to 0, adding a=bundle-only, and
// removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all
// but the first m= section.
@ -1942,8 +1894,7 @@ TEST_F(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Just add an audio track.
caller()->AddMediaStreamFromTracks(caller()->CreateLocalAudioTrack(),
nullptr);
caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -1960,8 +1911,7 @@ TEST_F(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Just add an audio track.
caller()->AddMediaStreamFromTracks(caller()->CreateLocalAudioTrack(),
nullptr);
caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -1975,7 +1925,7 @@ TEST_F(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
TEST_F(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
// Do offer/answer, wait for the callee to receive some frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -1987,30 +1937,23 @@ TEST_F(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
// Get a handle to the remote tracks created, so they can be used as GetStats
// filters.
StreamCollectionInterface* remote_streams = callee()->remote_streams();
ASSERT_EQ(1u, remote_streams->count());
ASSERT_EQ(1u, remote_streams->at(0)->GetAudioTracks().size());
ASSERT_EQ(1u, remote_streams->at(0)->GetVideoTracks().size());
MediaStreamTrackInterface* remote_audio_track =
remote_streams->at(0)->GetAudioTracks()[0];
MediaStreamTrackInterface* remote_video_track =
remote_streams->at(0)->GetVideoTracks()[0];
// We received frames, so we definitely should have nonzero "received bytes"
// stats at this point.
EXPECT_GT(callee()->OldGetStatsForTrack(remote_audio_track)->BytesReceived(),
0);
EXPECT_GT(callee()->OldGetStatsForTrack(remote_video_track)->BytesReceived(),
0);
for (auto receiver : callee()->pc()->GetReceivers()) {
// We received frames, so we definitely should have nonzero "received bytes"
// stats at this point.
EXPECT_GT(callee()->OldGetStatsForTrack(receiver->track())->BytesReceived(),
0);
}
}
// Test that we can get outgoing byte counts from both audio and video tracks.
TEST_F(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
auto audio_track = caller()->CreateLocalAudioTrack();
auto video_track = caller()->CreateLocalVideoTrack();
caller()->AddMediaStreamFromTracks(audio_track, video_track);
caller()->AddTrack(audio_track);
caller()->AddTrack(video_track);
// Do offer/answer, wait for the callee to receive some frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -2030,10 +1973,9 @@ TEST_F(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
TEST_F(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
auto audio_track = callee()->CreateLocalAudioTrack();
callee()->AddMediaStreamFromTracks(audio_track, nullptr);
callee()->AddAudioTrack();
// Do offer/answer, wait for the callee to receive some frames.
caller()->CreateAndSetAndSignalOffer();
@ -2060,7 +2002,7 @@ TEST_F(PeerConnectionIntegrationTest,
GetStatsForUnsignaledStreamWithNewStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
// Remove SSRCs and MSIDs from the received offer SDP.
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
@ -2087,7 +2029,7 @@ TEST_F(PeerConnectionIntegrationTest,
GetMediaStatsForUnsignaledStreamWithNewStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
// Remove SSRCs and MSIDs from the received offer SDP.
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
caller()->CreateAndSetAndSignalOffer();
@ -2136,7 +2078,7 @@ TEST_F(PeerConnectionIntegrationTest,
TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
// Remove SSRCs and MSIDs from the received offer SDP, simulating an endpoint
// that doesn't signal SSRCs (from the callee's perspective).
callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
@ -2204,8 +2146,8 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
ConnectFakeSignaling();
// Do normal offer/answer and wait for some frames to be received in each
// direction.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@ -2225,8 +2167,8 @@ TEST_F(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
@ -2250,8 +2192,8 @@ TEST_F(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
@ -2276,8 +2218,8 @@ TEST_F(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) {
ConnectFakeSignaling();
// Do normal offer/answer and wait for some frames to be received in each
// direction.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@ -2298,8 +2240,8 @@ TEST_F(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
ConnectFakeSignaling();
// Do normal offer/answer and wait for some frames to be received in each
// direction.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@ -2358,8 +2300,8 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
ConnectFakeSignaling();
// Do normal offer/answer and wait for some frames to be received in each
// direction.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@ -2379,8 +2321,8 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
// Expect that data channel created on caller side will show up for callee as
// well.
caller()->CreateDataChannel();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure the existence of the RTP data channel didn't impede audio/video.
@ -2414,8 +2356,8 @@ TEST_F(PeerConnectionIntegrationTest,
&setup_constraints));
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_NE(nullptr, caller()->data_channel());
@ -2496,8 +2438,8 @@ TEST_F(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
&setup_constraints_1, &setup_constraints_2));
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// The caller should still have a data channel, but it should be closed, and
@ -2516,8 +2458,8 @@ TEST_F(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
&setup_constraints));
ConnectFakeSignaling();
// Do initial offer/answer with audio/video.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Create data channel and do new offer and answer.
@ -2548,8 +2490,8 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) {
// Expect that data channel created on caller side will show up for callee as
// well.
caller()->CreateDataChannel();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure the existence of the SCTP data channel didn't impede audio/video.
@ -2581,8 +2523,8 @@ TEST_F(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->CreateDataChannel();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_NE(nullptr, caller()->data_channel());
@ -2604,8 +2546,8 @@ TEST_F(PeerConnectionIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
init.id = 53;
init.maxRetransmits = 52;
caller()->CreateDataChannel("data-channel", &init);
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
@ -2679,8 +2621,8 @@ TEST_F(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do initial offer/answer with audio/video.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Create data channel and do new offer and answer.
@ -2721,8 +2663,8 @@ TEST_F(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
// Do subsequent offer/answer with two-way audio and video. Audio and video
// should end up bundled on the DTLS/ICE transport already used for data.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@ -2772,8 +2714,8 @@ TEST_F(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do normal offer/answer.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
@ -2876,8 +2818,8 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyIceStates) {
ConnectFakeSignaling();
SetPortAllocatorFlags();
SetUpNetworkInterfaces();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Initial state before anything happens.
ASSERT_EQ(PeerConnectionInterface::kIceGatheringNew,
@ -2937,8 +2879,8 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
ConnectFakeSignaling();
SetPortAllocatorFlags();
SetUpNetworkInterfaces();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
rtc::scoped_refptr<webrtc::FakeMetricsObserver> metrics_observer(
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>());
@ -2994,8 +2936,8 @@ TEST_F(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Do normal offer/answer and wait for ICE to complete.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
@ -3074,8 +3016,8 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
ConnectFakeSignaling();
// Do normal offer/answer and wait for some frames to be received in each
// direction.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Sanity check that ICE renomination was actually negotiated.
@ -3112,7 +3054,7 @@ TEST_F(PeerConnectionIntegrationTest,
config, PeerConnectionInterface::RTCConfiguration()));
ConnectFakeSignaling();
caller()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
@ -3120,7 +3062,7 @@ TEST_F(PeerConnectionIntegrationTest,
caller()->clear_ice_connection_state_history();
caller()->AddVideoOnlyMediaStream();
caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -3137,7 +3079,7 @@ TEST_F(PeerConnectionIntegrationTest,
// Do initial negotiation, only sending media from the caller. Will result in
// video and audio recvonly "m=" sections.
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -3158,7 +3100,7 @@ TEST_F(PeerConnectionIntegrationTest,
// end-to-end, also adding media stream to callee.
options.offer_to_receive_video = 1;
callee()->SetOfferAnswerOptions(options);
callee()->AddAudioVideoMediaStream();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Verify the caller receives frames from the newly added stream, and the
@ -3178,8 +3120,8 @@ TEST_F(PeerConnectionIntegrationTest,
ASSERT_TRUE(CreatePeerConnectionWrappers());
EnableVideoDecoderFactory();
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ExpectNewFramesReceivedWithWait(
@ -3231,8 +3173,7 @@ TEST_F(PeerConnectionIntegrationTest, CanSendRemoteVideoTrack) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
// Just send a video track from the caller.
caller()->AddMediaStreamFromTracks(nullptr,
caller()->CreateLocalVideoTrack());
caller()->AddVideoTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
ASSERT_EQ(1, callee()->remote_streams()->count());
@ -3417,8 +3358,8 @@ TEST_F(PeerConnectionIntegrationTest, \
TEST_F(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Remove all but one audio/video codec (opus and VP8), and change the
// casing of the caller's generated offer.
@ -3463,7 +3404,7 @@ TEST_F(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
TEST_F(PeerConnectionIntegrationTest, GetSources) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for one audio frame to be received by the callee.
@ -3521,7 +3462,7 @@ TEST_F(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
std::move(output), webrtc::RtcEventLog::kImmediateOutput));
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
}
@ -3536,8 +3477,8 @@ TEST_F(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
// Add audio video track and exchange the initial offer/answer with media
// information only. This will start ICE gathering on each side.
caller()->AddAudioVideoMediaStream();
callee()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
// Wait for all candidates to be gathered on both the caller and callee.
@ -3568,8 +3509,8 @@ TEST_F(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
// Set up audio-only call where audio playout is disabled on caller's side.
caller()->pc()->SetAudioPlayout(false);
caller()->AddAudioOnlyMediaStream();
callee()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -3617,8 +3558,8 @@ TEST_F(PeerConnectionIntegrationTest,
// Set up audio-only call where playout is disabled but audio-processing is
// still active.
caller()->AddAudioOnlyMediaStream();
callee()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->pc()->SetAudioPlayout(false);
caller()->CreateAndSetAndSignalOffer();
@ -3638,8 +3579,8 @@ TEST_F(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) {
// Set up audio-only call where audio recording is disabled on caller's side.
caller()->pc()->SetAudioRecording(false);
caller()->AddAudioOnlyMediaStream();
callee()->AddAudioOnlyMediaStream();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
@ -3665,7 +3606,7 @@ TEST_F(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
// Set up audio/video/data, wait for some frames to be received.
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoMediaStream();
caller()->AddAudioVideoTracks();
#ifdef HAVE_SCTP
caller()->CreateDataChannel();
#endif