Implemented the GetSources() in native code.

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

Index: webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
diff --git a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
index 5e5d8eea49c04b89f9a13fd96bd62df7597d27f7..18aad7e1e62dbde8566c95217e9ae75e867a55e5 100644
--- a/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ b/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
@@ -27,6 +27,7 @@ const uint32_t kTestRate = 64000u;
 const uint8_t kTestPayload[] = { 't', 'e', 's', 't' };
 const uint8_t kPcmuPayloadType = 96;
 const uint8_t kDtmfPayloadType = 97;
+const int64_t kContributingSourcesTimeout = 10000;  // ms

[hbos, 2017/03/30 09:51:54]

+ kContributingSourcesTimeoutMs
- // ms

[Zhi Huang, 2017/03/31 06:44:04]

Done.

 
 struct CngCodecSpec {
   int payload_type;
@@ -284,4 +285,61 @@ TEST_F(RtpRtcpAudioTest, ComfortNoise) {
   }
 }
 
+TEST_F(RtpRtcpAudioTest, GetContributingSources) {
+  int64_t timestamp = fake_clock.TimeInMilliseconds();
+  RTPHeader header;
+  header.payloadType = kPcmuPayloadType;
+  header.ssrc = 1;
+  header.timestamp = timestamp;
+  header.numCSRCs = 2;
+  header.arrOfCSRCs[0] = 111;
+  header.arrOfCSRCs[1] = 222;
+
+  CodecInst voice_codec = {};
+  voice_codec.pltype = kPcmuPayloadType;
+  voice_codec.plfreq = 8000;
+  voice_codec.rate = kTestRate;
+  memcpy(voice_codec.plname, "PCMU", 5);
+  RegisterPayload(voice_codec);
+
+  PayloadUnion payload_specific;
+  bool in_order = false;
+
+  EXPECT_TRUE(rtp_receiver1_->IncomingRtpPacket(header, kTestPayload, 4,
+                                                payload_specific, in_order));
+  auto sources = rtp_receiver1_->GetContributingSources();
+  // Two sources use the CSRCs and one uses the SSRC.
+  ASSERT_EQ(3u, sources.size());
+  EXPECT_EQ(222u, sources[0]->source());
+  EXPECT_EQ(timestamp, sources[0]->timestamp());
+  EXPECT_EQ(111u, sources[1]->source());
+  EXPECT_EQ(timestamp, sources[1]->timestamp());
+  EXPECT_EQ(1u, sources[2]->source());
+  // The timestamp of the source using the SSRC is always the time when the
+  // method is called.
+  EXPECT_EQ(fake_clock.TimeInMilliseconds(), sources[2]->timestamp());
+
+  // Advance the fake clock and the method is expected to return the
+  // contributing source object with same |source| and updated |timestamp|.
+  fake_clock.AdvanceTimeMilliseconds(1);
+  EXPECT_TRUE(rtp_receiver1_->IncomingRtpPacket(header, kTestPayload, 4,
+                                                payload_specific, in_order));
+  sources = rtp_receiver1_->GetContributingSources();
+  ASSERT_EQ(3u, sources.size());
+  EXPECT_EQ(222u, sources[0]->source());
+  EXPECT_EQ(timestamp + 1, sources[0]->timestamp());
+  EXPECT_EQ(111u, sources[1]->source());
+  EXPECT_EQ(timestamp + 1, sources[1]->timestamp());
+  EXPECT_EQ(1u, sources[2]->source());
+  EXPECT_EQ(fake_clock.TimeInMilliseconds(), sources[2]->timestamp());
+
+  // Simulate the time out.
+  fake_clock.AdvanceTimeMilliseconds(kContributingSourcesTimeout + 1);
+  sources = rtp_receiver1_->GetContributingSources();
+  // The sources using the CSRCs should be out of date.
+  ASSERT_EQ(1u, sources.size());
+  EXPECT_EQ(1u, sources[0]->source());
+  EXPECT_EQ(fake_clock.TimeInMilliseconds(), sources[0]->timestamp());
+}
+
 }  // namespace webrtc