Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(43)

Unified Diff: webrtc/api/peerconnectioninterface_unittest.cc

Issue 1871993002: Only generate one CNAME per PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Renaming: cname -> rtcp_cname. Modified the peerconnectioninterface unit tests. Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/api/peerconnection.cc ('k') | webrtc/pc/mediasession.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/api/peerconnectioninterface_unittest.cc
diff --git a/webrtc/api/peerconnectioninterface_unittest.cc b/webrtc/api/peerconnectioninterface_unittest.cc
index 1a8dd57f08a6565f883c8ba5e73d69996fa1d623..2594b6c10637996c3a554757a188da633d5b12cb 100644
--- a/webrtc/api/peerconnectioninterface_unittest.cc
+++ b/webrtc/api/peerconnectioninterface_unittest.cc
@@ -934,6 +934,34 @@ class PeerConnectionInterfaceTest : public testing::Test {
ASSERT_TRUE(stream->AddTrack(video_track));
}
+ rtc::scoped_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
+ CreatePeerConnection();
+ AddVoiceStream(kStreamLabel1);
+ rtc::scoped_ptr<SessionDescriptionInterface> offer;
+ EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
+ return offer;
+ }
+
+ rtc::scoped_ptr<SessionDescriptionInterface>
+ CreateAnswerWithOneAudioStream() {
+ rtc::scoped_ptr<SessionDescriptionInterface> offer =
+ CreateOfferWithOneAudioStream();
+ EXPECT_TRUE(DoSetRemoteDescription(offer.release()));
+ rtc::scoped_ptr<SessionDescriptionInterface> answer;
+ EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
+ return answer;
+ }
+
+ const std::string& GetFirstAudioStreamCname(
+ const SessionDescriptionInterface* desc) {
+ const cricket::ContentInfo* audio_content =
+ cricket::GetFirstAudioContent(desc->description());
+ const cricket::AudioContentDescription* audio_desc =
+ static_cast<const cricket::AudioContentDescription*>(
+ audio_content->description);
+ return audio_desc->streams()[0].cname;
+ }
+
cricket::FakePortAllocator* port_allocator_ = nullptr;
scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
scoped_refptr<PeerConnectionInterface> pc_;
@@ -941,6 +969,27 @@ class PeerConnectionInterfaceTest : public testing::Test {
rtc::scoped_refptr<StreamCollection> reference_collection_;
};
+// Generate different CNAMEs when PeerConnections are created.
+// The CNAMEs are expected to be generated randomly. It is possible
+// that the test fails, though the possibility is very low.
+TEST_F(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
+ rtc::scoped_ptr<SessionDescriptionInterface> offer1 =
+ CreateOfferWithOneAudioStream();
+ rtc::scoped_ptr<SessionDescriptionInterface> offer2 =
+ CreateOfferWithOneAudioStream();
+ EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
+ GetFirstAudioStreamCname(offer2.get()));
+}
+
+TEST_F(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
+ rtc::scoped_ptr<SessionDescriptionInterface> answer1 =
+ CreateAnswerWithOneAudioStream();
+ rtc::scoped_ptr<SessionDescriptionInterface> answer2 =
+ CreateAnswerWithOneAudioStream();
+ EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
+ GetFirstAudioStreamCname(answer2.get()));
+}
+
TEST_F(PeerConnectionInterfaceTest,
CreatePeerConnectionWithDifferentConfigurations) {
CreatePeerConnectionWithDifferentConfigurations();
« no previous file with comments | « webrtc/api/peerconnection.cc ('k') | webrtc/pc/mediasession.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698