| Index: webrtc/pc/peerconnection_unittest.cc
|
| diff --git a/webrtc/pc/peerconnection_unittest.cc b/webrtc/pc/peerconnection_unittest.cc
|
| index e5e310ea0fd17be4fdf8bddfbc3a1bbd942bb994..03503fbde1bfb8ee16d0c7e4d36fbe82287d9c7b 100644
|
| --- a/webrtc/pc/peerconnection_unittest.cc
|
| +++ b/webrtc/pc/peerconnection_unittest.cc
|
| @@ -2638,6 +2638,22 @@ TEST_F(P2PTestConductor, EndToEndConnectionTimeWithTurnTurnPair) {
|
| delete set_receiving_client(nullptr);
|
| }
|
|
|
| +// The end to end test of the method GetContributingSources(). Tests the
|
| +// function calling chain from API layer down to the rtp_rtcp module.
|
| +TEST_F(P2PTestConductor, TestGetContributingSources) {
|
| + ASSERT_TRUE(CreateTestClients());
|
| + LocalP2PTest();
|
| +
|
| + ASSERT_GT(receiving_client()->pc()->GetReceivers().size(), 0);
|
| + auto receiver = receiving_client()->pc()->GetReceivers()[0];
|
| + ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
|
| +
|
| + auto contributing_sources = receiver->GetContributingSources();
|
| + ASSERT_GT(receiver->GetParameters().encodings.size(), 0);
|
| + EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
|
| + contributing_sources[0].source);
|
| +}
|
| +
|
| class IceServerParsingTest : public testing::Test {
|
| public:
|
| // Convenience for parsing a single URL.
|
|
|