In PC integration tests, create tracks/streams with random IDs.
Previously the helper methods (like "CreateLocalAudioTrack") were using
hard-coded IDs. This means if you try to add two tracks to the same
PeerConnection you'll get an ID conflict. Tests were getting around
this by using APIs to explicitly add tracks with different IDs. But
this leaks an implementation detail of the helper methods, and is a
hassle.
So instead, just use random UUIDs from the helper methods. This is how
IDs are created in the real world, so it's pretty reasonable and allows
the tests to be made more readable.
BUG=None
Review-Url: https://codereview.webrtc.org/3011713002
Cr-Original-Commit-Position: refs/heads/master@{#19736}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: b1a15d7711ffd419acd7dea3d307bf22f33ecba3
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 6b3ddd5..641be6f 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -87,9 +87,6 @@
static const int kDefaultExpectedAudioFrameCount = 3;
static const int kDefaultExpectedVideoFrameCount = 3;
-static const char kDefaultStreamLabel[] = "stream_label";
-static const char kDefaultVideoTrackId[] = "video_track";
-static const char kDefaultAudioTrackId[] = "audio_track";
static const char kDataChannelLabel[] = "data_channel";
// SRTP cipher name negotiated by the tests. This must be updated if the
@@ -306,45 +303,31 @@
peer_connection_factory_->CreateAudioSource(&constraints);
// TODO(perkj): Test audio source when it is implemented. Currently audio
// always use the default input.
- return peer_connection_factory_->CreateAudioTrack(kDefaultAudioTrackId,
+ return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(),
source);
}
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
- return CreateLocalVideoTrackInternal(
- kDefaultVideoTrackId, FakeConstraints(), webrtc::kVideoRotation_0);
+ return CreateLocalVideoTrackInternal(FakeConstraints(),
+ webrtc::kVideoRotation_0);
}
rtc::scoped_refptr<webrtc::VideoTrackInterface>
CreateLocalVideoTrackWithConstraints(const FakeConstraints& constraints) {
- return CreateLocalVideoTrackInternal(kDefaultVideoTrackId, constraints,
- webrtc::kVideoRotation_0);
+ return CreateLocalVideoTrackInternal(constraints, webrtc::kVideoRotation_0);
}
rtc::scoped_refptr<webrtc::VideoTrackInterface>
CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) {
- return CreateLocalVideoTrackInternal(kDefaultVideoTrackId,
- FakeConstraints(), rotation);
- }
-
- rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackWithId(
- const std::string& id) {
- return CreateLocalVideoTrackInternal(id, FakeConstraints(),
- webrtc::kVideoRotation_0);
+ return CreateLocalVideoTrackInternal(FakeConstraints(), rotation);
}
void AddMediaStreamFromTracks(
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio,
rtc::scoped_refptr<webrtc::VideoTrackInterface> video) {
- AddMediaStreamFromTracksWithLabel(audio, video, kDefaultStreamLabel);
- }
-
- void AddMediaStreamFromTracksWithLabel(
- rtc::scoped_refptr<webrtc::AudioTrackInterface> audio,
- rtc::scoped_refptr<webrtc::VideoTrackInterface> video,
- const std::string& stream_label) {
rtc::scoped_refptr<MediaStreamInterface> stream =
- peer_connection_factory_->CreateLocalMediaStream(stream_label);
+ peer_connection_factory_->CreateLocalMediaStream(
+ rtc::CreateRandomUuid());
if (audio) {
stream->AddTrack(audio);
}
@@ -634,7 +617,6 @@
}
rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal(
- const std::string& track_id,
const FakeConstraints& constraints,
webrtc::VideoRotation rotation) {
// Set max frame rate to 10fps to reduce the risk of test flakiness.
@@ -650,7 +632,8 @@
peer_connection_factory_->CreateVideoSource(fake_capturer,
&source_constraints);
rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
- peer_connection_factory_->CreateVideoTrack(track_id, source));
+ peer_connection_factory_->CreateVideoTrack(rtc::CreateRandomUuid(),
+ source));
if (!local_video_renderer_) {
local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
}
@@ -1411,8 +1394,7 @@
EXPECT_TRUE(callee_video_content->rejected);
// Now negotiate with video and ensure negotiation succeeds, with video
// frames and additional audio frames being received.
- callee()->AddMediaStreamFromTracksWithLabel(
- nullptr, callee()->CreateLocalVideoTrack(), "video_only_stream");
+ callee()->AddVideoOnlyMediaStream();
options.offer_to_receive_video = 1;
callee()->SetOfferAnswerOptions(options);
callee()->CreateAndSetAndSignalOffer();
@@ -1435,10 +1417,8 @@
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Now add an audio track and do another offer/answer.
- caller()->AddMediaStreamFromTracksWithLabel(caller()->CreateLocalAudioTrack(),
- nullptr, "audio_stream");
- callee()->AddMediaStreamFromTracksWithLabel(callee()->CreateLocalAudioTrack(),
- nullptr, "audio_stream");
+ caller()->AddAudioOnlyMediaStream();
+ callee()->AddAudioOnlyMediaStream();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Ensure both audio and video frames are received end-to-end.
@@ -1787,9 +1767,7 @@
ConnectFakeSignaling();
// Add one audio/video stream, and one video-only stream.
caller()->AddAudioVideoMediaStream();
- caller()->AddMediaStreamFromTracksWithLabel(
- nullptr, caller()->CreateLocalVideoTrackWithId("extra_track"),
- "extra_stream");
+ caller()->AddVideoOnlyMediaStream();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
ASSERT_EQ(2u, callee()->number_of_remote_streams());
