Implemented the GetSources() in native code.

webrtc/pc/peerconnection_integrationtest.cc

Index: webrtc/pc/peerconnection_integrationtest.cc
diff --git a/webrtc/pc/peerconnection_integrationtest.cc b/webrtc/pc/peerconnection_integrationtest.cc
index b4acdaa423f7851b08ee4b2d6e7ec2288e7ef29f..8af5cee74485d06a655e29414f02e9b322a2c164 100644
--- a/webrtc/pc/peerconnection_integrationtest.cc
+++ b/webrtc/pc/peerconnection_integrationtest.cc
@@ -2755,6 +2755,27 @@ TEST_F(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
       kMaxWaitForFramesMs);
 }
 
+TEST_F(PeerConnectionIntegrationTest, GetSources) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddAudioVideoMediaStream();
+  callee()->AddAudioVideoMediaStream();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  ExpectNewFramesReceivedWithWait(
+      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
+      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,

[Taylor Brandstetter, 2017/04/05 04:27:55]

nit: Only really need one audio track, from caller to callee, and only need to
wait for one audio frame to be received.

[Zhi Huang, 2017/04/06 03:09:50]

I thought if it works in more complicated cases, it will also works in the one
way audio only case. But I guess the simple case would be enough since we are
only testing the pipeline.

+      kMaxWaitForFramesMs);
+  ASSERT_GT(callee()->pc()->GetReceivers().size(), 0u);
+  auto receiver = callee()->pc()->GetReceivers()[0];
+  ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
+
+  auto contributing_sources = receiver->GetSources();
+  ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
+  EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
+            contributing_sources[0].source_id());
+}
+
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)