Implemented the GetSources() in native code.

webrtc/pc/peerconnection_integrationtest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 2737 matching lines @@
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
 
   // Verify frames are still received end-to-end.
   ExpectNewFramesReceivedWithWait(
       kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
       kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
       kMaxWaitForFramesMs);
 }
 
+TEST_F(PeerConnectionIntegrationTest, GetSources) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddAudioOnlyMediaStream();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait for one audio frame received by callee.
+  ExpectNewFramesReceivedWithWait(0, 0, 1, 0, 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)