webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc - src - Git at Google

 /*
 *  Copyright (c) 2013 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 <iterator>
 #include <list>
 #include <memory>
 #include <set>

 #include "webrtc/api/call/transport.h"
 #include "webrtc/call/rtp_stream_receiver_controller.h"
 #include "webrtc/call/rtx_receive_stream.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.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_packet_received.h"
 #include "webrtc/rtc_base/rate_limiter.h"
 #include "webrtc/test/gtest.h"

 namespace webrtc {

 const int kVideoNackListSize = 30;
 const uint32_t kTestSsrc = 3456;
 const uint32_t kTestRtxSsrc = kTestSsrc + 1;
 const uint16_t kTestSequenceNumber = 2345;
 const uint32_t kTestNumberOfPackets = 1350;
 const int kTestNumberOfRtxPackets = 149;
 const int kNumFrames = 30;
 const int kPayloadType = 123;
 const int kRtxPayloadType = 98;
 const int64_t kMaxRttMs = 1000;

 class VerifyingMediaStream : public RtpPacketSinkInterface {
  public:
   VerifyingMediaStream() {}

   void OnRtpPacket(const RtpPacketReceived& packet) override {
     if (!sequence_numbers_.empty())
       EXPECT_EQ(kTestSsrc, packet.Ssrc());

     sequence_numbers_.push_back(packet.SequenceNumber());
   }
   std::list<uint16_t> sequence_numbers_;
 };

 class RtxLoopBackTransport : public webrtc::Transport {
  public:
   explicit RtxLoopBackTransport(uint32_t rtx_ssrc)
       : count_(0),
         packet_loss_(0),
         consecutive_drop_start_(0),
         consecutive_drop_end_(0),
         rtx_ssrc_(rtx_ssrc),
         count_rtx_ssrc_(0),
         module_(NULL) {}

   void SetSendModule(RtpRtcp* rtpRtcpModule) {
     module_ = rtpRtcpModule;
   }

   void DropEveryNthPacket(int n) { packet_loss_ = n; }

   void DropConsecutivePackets(int start, int total) {
     consecutive_drop_start_ = start;
     consecutive_drop_end_ = start + total;
     packet_loss_ = 0;
   }

   bool SendRtp(const uint8_t* data,
                size_t len,
                const PacketOptions& options) override {
     count_++;
     RtpPacketReceived packet;
     if (!packet.Parse(data, len))
       return false;
     if (packet.Ssrc() == rtx_ssrc_) {
       count_rtx_ssrc_++;
     } else {
       // For non-RTX packets only.
       expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
                                         packet.SequenceNumber());
     }
     if (packet_loss_ > 0) {
       if ((count_ % packet_loss_) == 0) {
         return true;
       }
     } else if (count_ >= consecutive_drop_start_ &&
                count_ < consecutive_drop_end_) {
       return true;
     }
     EXPECT_TRUE(stream_receiver_controller_.OnRtpPacket(packet));
     return true;
   }

   bool SendRtcp(const uint8_t* data, size_t len) override {
     module_->IncomingRtcpPacket((const uint8_t*)data, len);
     return true;
   }
   int count_;
   int packet_loss_;
   int consecutive_drop_start_;
   int consecutive_drop_end_;
   uint32_t rtx_ssrc_;
   int count_rtx_ssrc_;
   RtpRtcp* module_;
   RtpStreamReceiverController stream_receiver_controller_;
   std::set<uint16_t> expected_sequence_numbers_;
 };

 class RtpRtcpRtxNackTest : public ::testing::Test {
  protected:
   RtpRtcpRtxNackTest()
       : rtp_rtcp_module_(nullptr),
         transport_(kTestRtxSsrc),
         rtx_stream_(&media_stream_,
                     rtx_associated_payload_types_,
                     kTestSsrc),
         payload_data_length(sizeof(payload_data)),
         fake_clock(123456),
         retransmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
   ~RtpRtcpRtxNackTest() {}

   void SetUp() override {
     RtpRtcp::Configuration configuration;
     configuration.audio = false;
     configuration.clock = &fake_clock;
     receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
     configuration.receive_statistics = receive_statistics_.get();
     configuration.outgoing_transport = &transport_;
     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
     rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);

     rtp_rtcp_module_->SetSSRC(kTestSsrc);
     rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
     rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
     EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));
     rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber);
     rtp_rtcp_module_->SetStartTimestamp(111111);

     // Used for NACK processing.
     // TODO(nisse): Unclear on which side? It's confusing to use a
     // single rtp_rtcp module for both send and receive side.
     rtp_rtcp_module_->SetRemoteSSRC(kTestSsrc);

     rtp_rtcp_module_->RegisterVideoSendPayload(kPayloadType, "video");
     rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
     transport_.SetSendModule(rtp_rtcp_module_);
     media_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
         kTestSsrc, &media_stream_);

     for (size_t n = 0; n < payload_data_length; n++) {
       payload_data[n] = n % 10;
     }
   }

   int BuildNackList(uint16_t* nack_list) {
     media_stream_.sequence_numbers_.sort();
     std::list<uint16_t> missing_sequence_numbers;
     std::list<uint16_t>::iterator it = media_stream_.sequence_numbers_.begin();

     while (it != media_stream_.sequence_numbers_.end()) {
       uint16_t sequence_number_1 = *it;
       ++it;
       if (it != media_stream_.sequence_numbers_.end()) {
         uint16_t sequence_number_2 = *it;
         // Add all missing sequence numbers to list
         for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
           missing_sequence_numbers.push_back(i);
         }
       }
     }
     int n = 0;
     for (it = missing_sequence_numbers.begin();
          it != missing_sequence_numbers.end(); ++it) {
       nack_list[n++] = (*it);
     }
     return n;
   }

   bool ExpectedPacketsReceived() {
     std::list<uint16_t> received_sorted;
     std::copy(media_stream_.sequence_numbers_.begin(),
               media_stream_.sequence_numbers_.end(),
               std::back_inserter(received_sorted));
     received_sorted.sort();
     return received_sorted.size() ==
                transport_.expected_sequence_numbers_.size() &&
            std::equal(received_sorted.begin(), received_sorted.end(),
                       transport_.expected_sequence_numbers_.begin());
   }

   void RunRtxTest(RtxMode rtx_method, int loss) {
     rtx_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
         kTestRtxSsrc, &rtx_stream_);
     rtp_rtcp_module_->SetRtxSendStatus(rtx_method);
     rtp_rtcp_module_->SetRtxSsrc(kTestRtxSsrc);
     transport_.DropEveryNthPacket(loss);
     uint32_t timestamp = 3000;
     uint16_t nack_list[kVideoNackListSize];
     for (int frame = 0; frame < kNumFrames; ++frame) {
       EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
           webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
           payload_data, payload_data_length, nullptr, nullptr, nullptr));
       // Min required delay until retransmit = 5 + RTT ms (RTT = 0).
       fake_clock.AdvanceTimeMilliseconds(5);
       int length = BuildNackList(nack_list);
       if (length > 0)
         rtp_rtcp_module_->SendNACK(nack_list, length);
       fake_clock.AdvanceTimeMilliseconds(28);  //  33ms - 5ms delay.
       rtp_rtcp_module_->Process();
       // Prepare next frame.
       timestamp += 3000;
     }
     media_stream_.sequence_numbers_.sort();
   }

   void TearDown() override { delete rtp_rtcp_module_; }

   std::unique_ptr<ReceiveStatistics> receive_statistics_;
   RtpRtcp* rtp_rtcp_module_;
   RtxLoopBackTransport transport_;
   const std::map<int, int> rtx_associated_payload_types_ =
       {{kRtxPayloadType, kPayloadType}};
   VerifyingMediaStream media_stream_;
   RtxReceiveStream rtx_stream_;
   uint8_t payload_data[65000];
   size_t payload_data_length;
   SimulatedClock fake_clock;
   RateLimiter retransmission_rate_limiter_;
   std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
   std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
 };

 TEST_F(RtpRtcpRtxNackTest, LongNackList) {
   const int kNumPacketsToDrop = 900;
   const int kNumRequiredRtcp = 4;
   uint32_t timestamp = 3000;
   uint16_t nack_list[kNumPacketsToDrop];
   // Disable StorePackets to be able to set a larger packet history.
   rtp_rtcp_module_->SetStorePacketsStatus(false, 0);
   // Enable StorePackets with a packet history of 2000 packets.
   rtp_rtcp_module_->SetStorePacketsStatus(true, 2000);
   // Drop 900 packets from the second one so that we get a NACK list which is
   // big enough to require 4 RTCP packets to be fully transmitted to the sender.
   transport_.DropConsecutivePackets(2, kNumPacketsToDrop);
   // Send 30 frames which at the default size is roughly what we need to get
   // enough packets.
   for (int frame = 0; frame < kNumFrames; ++frame) {
     EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
         webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
         payload_data, payload_data_length, nullptr, nullptr, nullptr));
     // Prepare next frame.
     timestamp += 3000;
     fake_clock.AdvanceTimeMilliseconds(33);
     rtp_rtcp_module_->Process();
   }
   EXPECT_FALSE(transport_.expected_sequence_numbers_.empty());
   EXPECT_FALSE(media_stream_.sequence_numbers_.empty());
   size_t last_receive_count = media_stream_.sequence_numbers_.size();
   int length = BuildNackList(nack_list);
   for (int i = 0; i < kNumRequiredRtcp - 1; ++i) {
     rtp_rtcp_module_->SendNACK(nack_list, length);
     EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
     last_receive_count = media_stream_.sequence_numbers_.size();
     EXPECT_FALSE(ExpectedPacketsReceived());
   }
   rtp_rtcp_module_->SendNACK(nack_list, length);
   EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }

 TEST_F(RtpRtcpRtxNackTest, RtxNack) {
   RunRtxTest(kRtxRetransmitted, 10);
   EXPECT_EQ(kTestSequenceNumber, *(media_stream_.sequence_numbers_.begin()));
   EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
             *(media_stream_.sequence_numbers_.rbegin()));
   EXPECT_EQ(kTestNumberOfPackets, media_stream_.sequence_numbers_.size());
   EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
   EXPECT_TRUE(ExpectedPacketsReceived());
 }

 }  // namespace webrtc
	/*
	* Copyright (c) 2013 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 <iterator>
	#include <list>
	#include <memory>
	#include <set>

	#include "webrtc/api/call/transport.h"
	#include "webrtc/call/rtp_stream_receiver_controller.h"
	#include "webrtc/call/rtx_receive_stream.h"
	#include "webrtc/common_types.h"
	#include "webrtc/modules/rtp_rtcp/include/receive_statistics.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_packet_received.h"
	#include "webrtc/rtc_base/rate_limiter.h"
	#include "webrtc/test/gtest.h"

	namespace webrtc {

	const int kVideoNackListSize = 30;
	const uint32_t kTestSsrc = 3456;
	const uint32_t kTestRtxSsrc = kTestSsrc + 1;
	const uint16_t kTestSequenceNumber = 2345;
	const uint32_t kTestNumberOfPackets = 1350;
	const int kTestNumberOfRtxPackets = 149;
	const int kNumFrames = 30;
	const int kPayloadType = 123;
	const int kRtxPayloadType = 98;
	const int64_t kMaxRttMs = 1000;

	class VerifyingMediaStream : public RtpPacketSinkInterface {
	public:
	VerifyingMediaStream() {}

	void OnRtpPacket(const RtpPacketReceived& packet) override {
	if (!sequence_numbers_.empty())
	EXPECT_EQ(kTestSsrc, packet.Ssrc());

	sequence_numbers_.push_back(packet.SequenceNumber());
	}
	std::list<uint16_t> sequence_numbers_;
	};

	class RtxLoopBackTransport : public webrtc::Transport {
	public:
	explicit RtxLoopBackTransport(uint32_t rtx_ssrc)
	: count_(0),
	packet_loss_(0),
	consecutive_drop_start_(0),
	consecutive_drop_end_(0),
	rtx_ssrc_(rtx_ssrc),
	count_rtx_ssrc_(0),
	module_(NULL) {}

	void SetSendModule(RtpRtcp* rtpRtcpModule) {
	module_ = rtpRtcpModule;
	}

	void DropEveryNthPacket(int n) { packet_loss_ = n; }

	void DropConsecutivePackets(int start, int total) {
	consecutive_drop_start_ = start;
	consecutive_drop_end_ = start + total;
	packet_loss_ = 0;
	}

	bool SendRtp(const uint8_t* data,
	size_t len,
	const PacketOptions& options) override {
	count_++;
	RtpPacketReceived packet;
	if (!packet.Parse(data, len))
	return false;
	if (packet.Ssrc() == rtx_ssrc_) {
	count_rtx_ssrc_++;
	} else {
	// For non-RTX packets only.
	expected_sequence_numbers_.insert(expected_sequence_numbers_.end(),
	packet.SequenceNumber());
	}
	if (packet_loss_ > 0) {
	if ((count_ % packet_loss_) == 0) {
	return true;
	}
	} else if (count_ >= consecutive_drop_start_ &&
	count_ < consecutive_drop_end_) {
	return true;
	}
	EXPECT_TRUE(stream_receiver_controller_.OnRtpPacket(packet));
	return true;
	}

	bool SendRtcp(const uint8_t* data, size_t len) override {
	module_->IncomingRtcpPacket((const uint8_t*)data, len);
	return true;
	}
	int count_;
	int packet_loss_;
	int consecutive_drop_start_;
	int consecutive_drop_end_;
	uint32_t rtx_ssrc_;
	int count_rtx_ssrc_;
	RtpRtcp* module_;
	RtpStreamReceiverController stream_receiver_controller_;
	std::set<uint16_t> expected_sequence_numbers_;
	};

	class RtpRtcpRtxNackTest : public ::testing::Test {
	protected:
	RtpRtcpRtxNackTest()
	: rtp_rtcp_module_(nullptr),
	transport_(kTestRtxSsrc),
	rtx_stream_(&media_stream_,
	rtx_associated_payload_types_,
	kTestSsrc),
	payload_data_length(sizeof(payload_data)),
	fake_clock(123456),
	retransmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
	~RtpRtcpRtxNackTest() {}

	void SetUp() override {
	RtpRtcp::Configuration configuration;
	configuration.audio = false;
	configuration.clock = &fake_clock;
	receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
	configuration.receive_statistics = receive_statistics_.get();
	configuration.outgoing_transport = &transport_;
	configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
	rtp_rtcp_module_ = RtpRtcp::CreateRtpRtcp(configuration);

	rtp_rtcp_module_->SetSSRC(kTestSsrc);
	rtp_rtcp_module_->SetRTCPStatus(RtcpMode::kCompound);
	rtp_rtcp_module_->SetStorePacketsStatus(true, 600);
	EXPECT_EQ(0, rtp_rtcp_module_->SetSendingStatus(true));
	rtp_rtcp_module_->SetSequenceNumber(kTestSequenceNumber);
	rtp_rtcp_module_->SetStartTimestamp(111111);

	// Used for NACK processing.
	// TODO(nisse): Unclear on which side? It's confusing to use a
	// single rtp_rtcp module for both send and receive side.
	rtp_rtcp_module_->SetRemoteSSRC(kTestSsrc);

	rtp_rtcp_module_->RegisterVideoSendPayload(kPayloadType, "video");
	rtp_rtcp_module_->SetRtxSendPayloadType(kRtxPayloadType, kPayloadType);
	transport_.SetSendModule(rtp_rtcp_module_);
	media_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
	kTestSsrc, &media_stream_);

	for (size_t n = 0; n < payload_data_length; n++) {
	payload_data[n] = n % 10;
	}
	}

	int BuildNackList(uint16_t* nack_list) {
	media_stream_.sequence_numbers_.sort();
	std::list<uint16_t> missing_sequence_numbers;
	std::list<uint16_t>::iterator it = media_stream_.sequence_numbers_.begin();

	while (it != media_stream_.sequence_numbers_.end()) {
	uint16_t sequence_number_1 = *it;
	++it;
	if (it != media_stream_.sequence_numbers_.end()) {
	uint16_t sequence_number_2 = *it;
	// Add all missing sequence numbers to list
	for (uint16_t i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
	missing_sequence_numbers.push_back(i);
	}
	}
	}
	int n = 0;
	for (it = missing_sequence_numbers.begin();
	it != missing_sequence_numbers.end(); ++it) {
	nack_list[n++] = (*it);
	}
	return n;
	}

	bool ExpectedPacketsReceived() {
	std::list<uint16_t> received_sorted;
	std::copy(media_stream_.sequence_numbers_.begin(),
	media_stream_.sequence_numbers_.end(),
	std::back_inserter(received_sorted));
	received_sorted.sort();
	return received_sorted.size() ==
	transport_.expected_sequence_numbers_.size() &&
	std::equal(received_sorted.begin(), received_sorted.end(),
	transport_.expected_sequence_numbers_.begin());
	}

	void RunRtxTest(RtxMode rtx_method, int loss) {
	rtx_receiver_ = transport_.stream_receiver_controller_.CreateReceiver(
	kTestRtxSsrc, &rtx_stream_);
	rtp_rtcp_module_->SetRtxSendStatus(rtx_method);
	rtp_rtcp_module_->SetRtxSsrc(kTestRtxSsrc);
	transport_.DropEveryNthPacket(loss);
	uint32_t timestamp = 3000;
	uint16_t nack_list[kVideoNackListSize];
	for (int frame = 0; frame < kNumFrames; ++frame) {
	EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
	webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
	payload_data, payload_data_length, nullptr, nullptr, nullptr));
	// Min required delay until retransmit = 5 + RTT ms (RTT = 0).
	fake_clock.AdvanceTimeMilliseconds(5);
	int length = BuildNackList(nack_list);
	if (length > 0)
	rtp_rtcp_module_->SendNACK(nack_list, length);
	fake_clock.AdvanceTimeMilliseconds(28); // 33ms - 5ms delay.
	rtp_rtcp_module_->Process();
	// Prepare next frame.
	timestamp += 3000;
	}
	media_stream_.sequence_numbers_.sort();
	}

	void TearDown() override { delete rtp_rtcp_module_; }

	std::unique_ptr<ReceiveStatistics> receive_statistics_;
	RtpRtcp* rtp_rtcp_module_;
	RtxLoopBackTransport transport_;
	const std::map<int, int> rtx_associated_payload_types_ =
	{{kRtxPayloadType, kPayloadType}};
	VerifyingMediaStream media_stream_;
	RtxReceiveStream rtx_stream_;
	uint8_t payload_data[65000];
	size_t payload_data_length;
	SimulatedClock fake_clock;
	RateLimiter retransmission_rate_limiter_;
	std::unique_ptr<RtpStreamReceiverInterface> media_receiver_;
	std::unique_ptr<RtpStreamReceiverInterface> rtx_receiver_;
	};

	TEST_F(RtpRtcpRtxNackTest, LongNackList) {
	const int kNumPacketsToDrop = 900;
	const int kNumRequiredRtcp = 4;
	uint32_t timestamp = 3000;
	uint16_t nack_list[kNumPacketsToDrop];
	// Disable StorePackets to be able to set a larger packet history.
	rtp_rtcp_module_->SetStorePacketsStatus(false, 0);
	// Enable StorePackets with a packet history of 2000 packets.
	rtp_rtcp_module_->SetStorePacketsStatus(true, 2000);
	// Drop 900 packets from the second one so that we get a NACK list which is
	// big enough to require 4 RTCP packets to be fully transmitted to the sender.
	transport_.DropConsecutivePackets(2, kNumPacketsToDrop);
	// Send 30 frames which at the default size is roughly what we need to get
	// enough packets.
	for (int frame = 0; frame < kNumFrames; ++frame) {
	EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
	webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
	payload_data, payload_data_length, nullptr, nullptr, nullptr));
	// Prepare next frame.
	timestamp += 3000;
	fake_clock.AdvanceTimeMilliseconds(33);
	rtp_rtcp_module_->Process();
	}
	EXPECT_FALSE(transport_.expected_sequence_numbers_.empty());
	EXPECT_FALSE(media_stream_.sequence_numbers_.empty());
	size_t last_receive_count = media_stream_.sequence_numbers_.size();
	int length = BuildNackList(nack_list);
	for (int i = 0; i < kNumRequiredRtcp - 1; ++i) {
	rtp_rtcp_module_->SendNACK(nack_list, length);
	EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
	last_receive_count = media_stream_.sequence_numbers_.size();
	EXPECT_FALSE(ExpectedPacketsReceived());
	}
	rtp_rtcp_module_->SendNACK(nack_list, length);
	EXPECT_GT(media_stream_.sequence_numbers_.size(), last_receive_count);
	EXPECT_TRUE(ExpectedPacketsReceived());
	}

	TEST_F(RtpRtcpRtxNackTest, RtxNack) {
	RunRtxTest(kRtxRetransmitted, 10);
	EXPECT_EQ(kTestSequenceNumber, *(media_stream_.sequence_numbers_.begin()));
	EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
	*(media_stream_.sequence_numbers_.rbegin()));
	EXPECT_EQ(kTestNumberOfPackets, media_stream_.sequence_numbers_.size());
	EXPECT_EQ(kTestNumberOfRtxPackets, transport_.count_rtx_ssrc_);
	EXPECT_TRUE(ExpectedPacketsReceived());
	}

	} // namespace webrtc