Implemented the GetSources() in native code.

webrtc/pc/peerconnection_unittest.cc

Index: webrtc/pc/peerconnection_unittest.cc
diff --git a/webrtc/pc/peerconnection_unittest.cc b/webrtc/pc/peerconnection_unittest.cc
index e5e310ea0fd17be4fdf8bddfbc3a1bbd942bb994..60046f57cee557bfa1e0cffe933f7dc63e029f32 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) {

[Taylor Brandstetter, 2017/03/30 22:55:38]

I recently refactored the tests in this file, FYI. There's no more
"LocalP2PTest", so you'll have to explicitly add an audio track to the caller's
PeerConnection.

[Zhi Huang, 2017/03/31 06:44:05]

Acknowledged.

+  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.