Implemented the GetSources() in native code.

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

 /*
  *  Copyright (c) 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.
  */
 
 #include <algorithm>
 #include <memory>
 #include <vector>
 
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h"
 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace {
 
 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;
   int clockrate_hz;
 };
 
 const CngCodecSpec kCngCodecs[] = {{13, 8000},
                                    {103, 16000},
                                    {104, 32000},
                                    {105, 48000}};
@@ skipping 237 matching lines @@
                                           in_timestamp, -1, kTestPayload, 1,
                                           nullptr, nullptr, nullptr));
 
     EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
     EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
     EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
     in_timestamp += 10;
   }
 }
 
+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