Implemented the GetSources() in native code.

webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc

+/*
+ *  Copyright (c) 2017 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 <memory>
+
+#include "webrtc/common_types.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/test/gtest.h"
+
+namespace webrtc {
+
+const int64_t kContributingSourcesTimeoutMs = 10000;
+const uint32_t kTestRate = 64000u;
+const uint8_t kTestPayload[] = {'t', 'e', 's', 't'};
+const uint8_t kPcmuPayloadType = 96;
+
+class RtpReceiverTest : public ::testing::Test {
+ protected:
+  RtpReceiverTest() : fake_clock(123456) {}
+  ~RtpReceiverTest() {}
+
+  void SetUp() override {
+    rtp_payload_registry_.reset(new RTPPayloadRegistry());
+
+    rtp_receiver_.reset(RtpReceiver::CreateAudioReceiver(
+        &fake_clock, nullptr, nullptr, rtp_payload_registry_.get()));
+
+    CodecInst voice_codec = {};
+    voice_codec.pltype = kPcmuPayloadType;
+    voice_codec.plfreq = 8000;
+    voice_codec.rate = kTestRate;
+    memcpy(voice_codec.plname, "PCMU", 5);
+    rtp_receiver_->RegisterReceivePayload(voice_codec);
+  }
+
+  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
+  std::unique_ptr<RtpReceiver> rtp_receiver_;
+  SimulatedClock fake_clock;
+};
+
+TEST_F(RtpReceiverTest, 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;
+  PayloadUnion payload_specific;
+  bool in_order = false;
+
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
+                                               payload_specific, in_order));
+  auto sources = rtp_receiver_->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].ssrc());
+  EXPECT_EQ(1u, sources[2].source());
+  EXPECT_EQ(timestamp, 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_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
+                                               payload_specific, in_order));
+  sources = rtp_receiver_->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].ssrc());
+  EXPECT_EQ(1u, sources[2].source());
+  EXPECT_EQ(timestamp + 1, sources[2].timestamp());
+
+  // Simulate the time out.
+  fake_clock.AdvanceTimeMilliseconds(kContributingSourcesTimeoutMs + 1);
+  sources = rtp_receiver_->GetContributingSources();
+  // All the sources should be out of date.
+  ASSERT_EQ(0u, sources.size());
+}
+
+// Test the cases that the SSRC is changed.
+TEST_F(RtpReceiverTest, GetContributingSourcesChangeSSRC) {
+  int64_t prev_time = -1;
+  int64_t cur_time = fake_clock.TimeInMilliseconds();
+  RTPHeader header;
+  header.payloadType = kPcmuPayloadType;
+  header.ssrc = 1;
+  header.timestamp = cur_time;
+  PayloadUnion payload_specific;
+  bool in_order = false;
+
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
+                                               payload_specific, in_order));
+  auto sources = rtp_receiver_->GetContributingSources();
+  ASSERT_EQ(1u, sources.size());
+  EXPECT_EQ(1u, sources[0].source());
+  EXPECT_EQ(cur_time, sources[0].timestamp());
+
+  // The SSRC is changed and the old SSRC is expected to be returned.
+  fake_clock.AdvanceTimeMilliseconds(100);
+  prev_time = cur_time;
+  cur_time = fake_clock.TimeInMilliseconds();
+  header.ssrc = 2;
+  header.timestamp = cur_time;
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
+                                               payload_specific, in_order));
+  sources = rtp_receiver_->GetContributingSources();
+  ASSERT_EQ(2u, sources.size());
+  EXPECT_EQ(2u, sources[0].source());
+  EXPECT_EQ(cur_time, sources[0].timestamp());
+  EXPECT_EQ(1u, sources[1].source());
+  EXPECT_EQ(prev_time, sources[1].timestamp());
+
+  // The SSRC is changed again and happen to be changed back to 1. No
+  // duplication is expected.
+  fake_clock.AdvanceTimeMilliseconds(100);
+  header.ssrc = 1;
+  header.timestamp = cur_time;
+  prev_time = cur_time;
+  cur_time = fake_clock.TimeInMilliseconds();
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
+                                               payload_specific, in_order));
+  sources = rtp_receiver_->GetContributingSources();
+  ASSERT_EQ(2u, sources.size());
+  EXPECT_EQ(1u, sources[0].source());
+  EXPECT_EQ(cur_time, sources[0].timestamp());
+  EXPECT_EQ(2u, sources[1].source());
+  EXPECT_EQ(prev_time, sources[1].timestamp());
+
+  // Old SSRC source timeout.
+  fake_clock.AdvanceTimeMilliseconds(kContributingSourcesTimeoutMs);
+  cur_time = fake_clock.TimeInMilliseconds();
+  EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(header, kTestPayload, 4,
+                                               payload_specific, in_order));
+  sources = rtp_receiver_->GetContributingSources();
+  ASSERT_EQ(1u, sources.size());
+  EXPECT_EQ(1u, sources[0].source());
+  EXPECT_EQ(cur_time, sources[0].timestamp());
+}
+
+}  // namespace webrtc