webrtc/video/end_to_end_tests.cc - Issue 1419193002: Call OnSentPacket for all packets sent in the test framework.

Unified Diff: webrtc/video/end_to_end_tests.cc

Issue 1419193002: Call OnSentPacket for all packets sent in the test framework. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: . Created 5 years, 2 months ago

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Index: webrtc/video/end_to_end_tests.cc

diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc

index 31ec7d5a764f9821c53da8673e74069eb459b4fc..71949c9933fd74369d95f2ea22d0933c5af0220a 100644

--- a/webrtc/video/end_to_end_tests.cc

+++ b/webrtc/video/end_to_end_tests.cc

@@ -81,7 +81,6 @@ class EndToEndTest : public test::CallTest {

void TestXrReceiverReferenceTimeReport(bool enable_rrtr);

void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);

void TestRtpStatePreservation(bool use_rtx);

- void TestReceivedFecPacketsNotNacked(const FakeNetworkPipe::Config& config);

void VerifyHistogramStats(bool use_rtx, bool use_red);

};

@@ -156,7 +155,8 @@ TEST_F(EndToEndTest, RendersSingleDelayedFrame) {

CreateCalls(Call::Config(), Call::Config());

- test::DirectTransport sender_transport, receiver_transport;

+ test::DirectTransport sender_transport(sender_call_.get());

+ test::DirectTransport receiver_transport(receiver_call_.get());

sender_transport.SetReceiver(receiver_call_->Receiver());

receiver_transport.SetReceiver(sender_call_->Receiver());

@@ -206,7 +206,8 @@ TEST_F(EndToEndTest, TransmitsFirstFrame) {

CreateCalls(Call::Config(), Call::Config());

- test::DirectTransport sender_transport, receiver_transport;

+ test::DirectTransport sender_transport(sender_call_.get());

+ test::DirectTransport receiver_transport(receiver_call_.get());

sender_transport.SetReceiver(receiver_call_->Receiver());

receiver_transport.SetReceiver(sender_call_->Receiver());

@@ -281,7 +282,7 @@ TEST_F(EndToEndTest, SendsAndReceivesVP9) {

int frame_counter_;

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, SendsAndReceivesH264) {

@@ -333,7 +334,7 @@ TEST_F(EndToEndTest, SendsAndReceivesH264) {

int frame_counter_;

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {

@@ -361,7 +362,7 @@ TEST_F(EndToEndTest, ReceiverUsesLocalSsrc) {

}

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {

@@ -379,6 +380,7 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {

private:

Action OnSendRtp(const uint8_t* packet, size_t length) override {

+ rtc::CritScope lock(&crit_);

RTPHeader header;

EXPECT_TRUE(rtp_parser_->Parse(packet, length, &header));

@@ -414,6 +416,7 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {

}

Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {

+ rtc::CritScope lock(&crit_);

RTCPUtility::RTCPParserV2 parser(packet, length, true);

EXPECT_TRUE(parser.IsValid());

@@ -441,6 +444,7 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {

"rendered.";

}

+ rtc::CriticalSection crit_;

pbos-webrtc 2015/10/27 14:06:30 annotate

stefan-webrtc 2015/10/27 14:33:16 Done.

rtc::scoped_ptr<RtpHeaderParser> rtp_parser_;

std::set<uint16_t> dropped_packets_;

std::set<uint16_t> retransmitted_packets_;

@@ -449,7 +453,7 @@ TEST_F(EndToEndTest, ReceivesAndRetransmitsNack) {

int nacks_left_;

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, CanReceiveFec) {

@@ -459,8 +463,8 @@ TEST_F(EndToEndTest, CanReceiveFec) {

: EndToEndTest(kDefaultTimeoutMs), state_(kFirstPacket) {}

private:

- Action OnSendRtp(const uint8_t* packet, size_t length) override

- EXCLUSIVE_LOCKS_REQUIRED(crit_) {

+ Action OnSendRtp(const uint8_t* packet, size_t length) override {

+ rtc::CritScope lock(&crit_);

RTPHeader header;

EXPECT_TRUE(parser_->Parse(packet, length, &header));

@@ -544,29 +548,20 @@ TEST_F(EndToEndTest, CanReceiveFec) {

<< "Timed out waiting for dropped frames frames to be rendered.";

}

+ rtc::CriticalSection crit_;

std::set<uint32_t> protected_sequence_numbers_ GUARDED_BY(crit_);

std::set<uint32_t> protected_timestamps_ GUARDED_BY(crit_);

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

// Flacky on all platforms. See webrtc:4328.

TEST_F(EndToEndTest, DISABLED_ReceivedFecPacketsNotNacked) {

- // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.

- // Configure some network delay.

- const int kNetworkDelayMs = 50;

- FakeNetworkPipe::Config config;

- config.queue_delay_ms = kNetworkDelayMs;

- TestReceivedFecPacketsNotNacked(config);

-void EndToEndTest::TestReceivedFecPacketsNotNacked(

- const FakeNetworkPipe::Config& config) {

class FecNackObserver : public test::EndToEndTest {

public:

- explicit FecNackObserver(const FakeNetworkPipe::Config& config)

- : EndToEndTest(kDefaultTimeoutMs, config),

+ explicit FecNackObserver()

+ : EndToEndTest(kDefaultTimeoutMs),

state_(kFirstPacket),

fec_sequence_number_(0),

has_last_sequence_number_(false),

@@ -674,9 +669,14 @@ void EndToEndTest::TestReceivedFecPacketsNotNacked(

uint16_t fec_sequence_number_;

bool has_last_sequence_number_;

uint16_t last_sequence_number_;

- } test(config);

+ } test;

- RunBaseTest(&test);

+ // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.

+ // Configure some network delay.

+ const int kNetworkDelayMs = 50;

+ FakeNetworkPipe::Config config;

+ config.queue_delay_ms = kNetworkDelayMs;

+ RunBaseTest(&test, config);

}

// This test drops second RTP packet with a marker bit set, makes sure it's

@@ -700,6 +700,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {

private:

Action OnSendRtp(const uint8_t* packet, size_t length) override {

+ rtc::CritScope lock(&crit_);

RTPHeader header;

EXPECT_TRUE(parser_->Parse(packet, length, &header));

@@ -773,6 +774,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {

: (use_red ? kRedPayloadType : kFakeSendPayloadType);

}

+ rtc::CriticalSection crit_;

pbos-webrtc 2015/10/27 14:06:30 guarded_by

stefan-webrtc 2015/10/27 14:33:16 Done.

const int payload_type_;

const uint32_t retransmission_ssrc_;

const int retransmission_payload_type_;

@@ -782,7 +784,7 @@ void EndToEndTest::DecodesRetransmittedFrame(bool use_rtx, bool use_red) {

bool frame_retransmitted_;

} test(use_rtx, use_red);

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, DecodesRetransmittedFrame) {

@@ -861,7 +863,8 @@ TEST_F(EndToEndTest, UsesFrameCallbacks) {

CreateCalls(Call::Config(), Call::Config());

- test::DirectTransport sender_transport, receiver_transport;

+ test::DirectTransport sender_transport(sender_call_.get());

+ test::DirectTransport receiver_transport(receiver_call_.get());

sender_transport.SetReceiver(receiver_call_->Receiver());

receiver_transport.SetReceiver(sender_call_->Receiver());

@@ -918,6 +921,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {

private:

Action OnSendRtp(const uint8_t* packet, size_t length) override {

+ rtc::CritScope lock(&crit_);

RTPHeader header;

EXPECT_TRUE(parser_->Parse(packet, length, &header));

@@ -935,6 +939,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {

}

Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {

+ rtc::CritScope lock(&crit_);

RTCPUtility::RTCPParserV2 parser(packet, length, true);

EXPECT_TRUE(parser.IsValid());

@@ -979,6 +984,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {

"rendered afterwards.";

}

+ rtc::CriticalSection crit_;

pbos-webrtc 2015/10/27 14:06:30 guarded_by

stefan-webrtc 2015/10/27 14:33:16 Done.

int rtp_history_ms_;

bool nack_enabled_;

uint32_t highest_dropped_timestamp_;

@@ -986,7 +992,7 @@ void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {

bool received_pli_;

} test(rtp_history_ms);

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, ReceivesPliAndRecoversWithNack) {

@@ -1031,7 +1037,8 @@ TEST_F(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {

CreateCalls(Call::Config(), Call::Config());

- test::DirectTransport send_transport, receive_transport;

+ test::DirectTransport send_transport(sender_call_.get());

+ test::DirectTransport receive_transport(receiver_call_.get());

PacketInputObserver input_observer(receiver_call_->Receiver());

send_transport.SetReceiver(&input_observer);

receive_transport.SetReceiver(sender_call_->Receiver());

@@ -1135,7 +1142,7 @@ void EndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) {

int sent_rtcp_;

} test(rtcp_mode);

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, UsesRtcpCompoundMode) {

@@ -1167,13 +1174,12 @@ class MultiStreamTest {

virtual ~MultiStreamTest() {}

void RunTest() {

- rtc::scoped_ptr<test::DirectTransport> sender_transport(

- CreateSendTransport());

- rtc::scoped_ptr<test::DirectTransport> receiver_transport(

- CreateReceiveTransport());

rtc::scoped_ptr<Call> sender_call(Call::Create(Call::Config()));

rtc::scoped_ptr<Call> receiver_call(Call::Create(Call::Config()));

+ rtc::scoped_ptr<test::DirectTransport> sender_transport(

+ CreateSendTransport(sender_call.get()));

+ rtc::scoped_ptr<test::DirectTransport> receiver_transport(

+ CreateReceiveTransport(receiver_call.get()));

sender_transport->SetReceiver(receiver_call->Receiver());

receiver_transport->SetReceiver(sender_call->Receiver());

@@ -1257,11 +1263,11 @@ class MultiStreamTest {

virtual void UpdateReceiveConfig(size_t stream_index,

VideoReceiveStream::Config* receive_config) {

}

- virtual test::DirectTransport* CreateSendTransport() {

- return new test::DirectTransport();

+ virtual test::DirectTransport* CreateSendTransport(Call* sender_call) {

pbos-webrtc 2015/10/27 14:06:30 Inline if this isn't overriding anything. If so ta

stefan-webrtc 2015/10/27 14:33:16 Done.

+ return new test::DirectTransport(sender_call);

}

- virtual test::DirectTransport* CreateReceiveTransport() {

- return new test::DirectTransport();

+ virtual test::DirectTransport* CreateReceiveTransport(Call* receiver_call) {

+ return new test::DirectTransport(receiver_call);

}

};

@@ -1340,9 +1346,11 @@ TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) {

class RtpExtensionHeaderObserver : public test::DirectTransport {

public:

- RtpExtensionHeaderObserver(const uint32_t& first_media_ssrc,

+ RtpExtensionHeaderObserver(Call* sender_call,

+ const uint32_t& first_media_ssrc,

const std::map<uint32_t, uint32_t>& ssrc_map)

- : done_(EventWrapper::Create()),

+ : DirectTransport(sender_call),

+ done_(EventWrapper::Create()),

parser_(RtpHeaderParser::Create()),

first_media_ssrc_(first_media_ssrc),

rtx_to_media_ssrcs_(ssrc_map),

@@ -1507,8 +1515,8 @@ TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) {

RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));

}

- virtual test::DirectTransport* CreateSendTransport() {

- observer_ = new RtpExtensionHeaderObserver(first_media_ssrc_,

+ virtual test::DirectTransport* CreateSendTransport(Call* sender_call) {

pbos-webrtc 2015/10/27 14:06:30 Mark as override or inline if not overriding.

stefan-webrtc 2015/10/27 14:33:16 Done.

+ observer_ = new RtpExtensionHeaderObserver(sender_call, first_media_ssrc_,

rtx_to_media_ssrcs_);

return observer_;

}

@@ -1527,7 +1535,8 @@ TEST_F(EndToEndTest, ReceivesTransportFeedback) {

class TransportFeedbackObserver : public test::DirectTransport {

public:

- TransportFeedbackObserver(rtc::Event* done_event) : done_(done_event) {}

+ TransportFeedbackObserver(Call* receiver_call, rtc::Event* done_event)

+ : DirectTransport(receiver_call), done_(done_event) {}

virtual ~TransportFeedbackObserver() {}

bool SendRtcp(const uint8_t* data, size_t length) override {

@@ -1575,8 +1584,8 @@ TEST_F(EndToEndTest, ReceivesTransportFeedback) {

RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));

}

- virtual test::DirectTransport* CreateReceiveTransport() {

- return new TransportFeedbackObserver(&done_);

+ virtual test::DirectTransport* CreateReceiveTransport(Call* receiver_call) {

pbos-webrtc 2015/10/27 14:06:30 override?

stefan-webrtc 2015/10/27 14:33:16 Done.

+ return new TransportFeedbackObserver(receiver_call, &done_);

}

private:

@@ -1624,7 +1633,8 @@ TEST_F(EndToEndTest, ObserversEncodedFrames) {

CreateCalls(Call::Config(), Call::Config());

- test::DirectTransport sender_transport, receiver_transport;

+ test::DirectTransport sender_transport(sender_call_.get());

+ test::DirectTransport receiver_transport(receiver_call_.get());

sender_transport.SetReceiver(receiver_call_->Receiver());

receiver_transport.SetReceiver(sender_call_->Receiver());

@@ -1694,11 +1704,11 @@ TEST_F(EndToEndTest, ReceiveStreamSendsRemb) {

}

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, VerifyBandwidthStats) {

- class RtcpObserver : public test::EndToEndTest, public PacketReceiver {

+ class RtcpObserver : public test::EndToEndTest {

public:

RtcpObserver()

: EndToEndTest(kDefaultTimeoutMs),

@@ -1706,10 +1716,7 @@ TEST_F(EndToEndTest, VerifyBandwidthStats) {

receiver_call_(nullptr),

has_seen_pacer_delay_(false) {}

- DeliveryStatus DeliverPacket(MediaType media_type,

- const uint8_t* packet,

- size_t length,

- const PacketTime& packet_time) override {

+ Action OnSendRtp(const uint8_t* packet, size_t length) override {

Call::Stats sender_stats = sender_call_->GetStats();

Call::Stats receiver_stats = receiver_call_->GetStats();

if (!has_seen_pacer_delay_)

@@ -1718,8 +1725,7 @@ TEST_F(EndToEndTest, VerifyBandwidthStats) {

receiver_stats.recv_bandwidth_bps > 0 && has_seen_pacer_delay_) {

observation_complete_->Set();

}

- return receiver_call_->Receiver()->DeliverPacket(media_type, packet,

- length, packet_time);

+ return SEND_PACKET;

}

void OnCallsCreated(Call* sender_call, Call* receiver_call) override {

@@ -1732,18 +1738,13 @@ TEST_F(EndToEndTest, VerifyBandwidthStats) {

"non-zero bandwidth stats.";

}

- void SetReceivers(PacketReceiver* send_transport_receiver,

- PacketReceiver* receive_transport_receiver) override {

- test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);

- }

private:

Call* sender_call_;

Call* receiver_call_;

bool has_seen_pacer_delay_;

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, VerifyNackStats) {

@@ -1844,7 +1845,7 @@ TEST_F(EndToEndTest, VerifyNackStats) {

} test;

test::ClearHistograms();

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

EXPECT_EQ(1, test::NumHistogramSamples(

"WebRTC.Video.UniqueNackRequestsSentInPercent"));

@@ -1857,7 +1858,7 @@ TEST_F(EndToEndTest, VerifyNackStats) {

}

void EndToEndTest::VerifyHistogramStats(bool use_rtx, bool use_red) {

- class StatsObserver : public test::EndToEndTest, public PacketReceiver {

+ class StatsObserver : public test::EndToEndTest {

public:

StatsObserver(bool use_rtx, bool use_red)

: EndToEndTest(kLongTimeoutMs),

@@ -1872,20 +1873,13 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx, bool use_red) {

if (MinMetricRunTimePassed())

observation_complete_->Set();

- return SEND_PACKET;

- }

- DeliveryStatus DeliverPacket(MediaType media_type,

- const uint8_t* packet,

- size_t length,

- const PacketTime& packet_time) override {

// GetStats calls GetSendChannelRtcpStatistics

// (via VideoSendStream::GetRtt) which updates ReportBlockStats used by

// WebRTC.Video.SentPacketsLostInPercent.

// TODO(asapersson): Remove dependency on calling GetStats.

sender_call_->GetStats();

- return receiver_call_->Receiver()->DeliverPacket(media_type, packet,

- length, packet_time);

+ return SEND_PACKET;

}

bool MinMetricRunTimePassed() {

@@ -1927,11 +1921,6 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx, bool use_red) {

receiver_call_ = receiver_call;

}

- void SetReceivers(PacketReceiver* send_transport_receiver,

- PacketReceiver* receive_transport_receiver) override {

- test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);

- }

void PerformTest() override {

EXPECT_EQ(kEventSignaled, Wait())

<< "Timed out waiting for packet to be NACKed.";

@@ -1945,7 +1934,7 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx, bool use_red) {

} test(use_rtx, use_red);

test::ClearHistograms();

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

// Verify that stats have been updated once.

EXPECT_EQ(1, test::NumHistogramSamples(

@@ -2129,7 +2118,7 @@ void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {

int sent_rtcp_dlrr_;

} test(enable_rrtr);

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,

@@ -2227,7 +2216,7 @@ void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,

VideoEncoderConfig encoder_config_all_streams_;

} test(kSendSsrcs, num_ssrcs, send_single_ssrc_first);

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, ReportsSetEncoderRates) {

@@ -2282,11 +2271,12 @@ TEST_F(EndToEndTest, ReportsSetEncoderRates) {

}

private:

+ rtc::CriticalSection crit_;

VideoSendStream* send_stream_;

uint32_t bitrate_kbps_ GUARDED_BY(crit_);

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, GetStats) {

@@ -2294,8 +2284,8 @@ TEST_F(EndToEndTest, GetStats) {

static const int kExpectedRenderDelayMs = 20;

class StatsObserver : public test::EndToEndTest, public I420FrameCallback {

public:

- explicit StatsObserver(const FakeNetworkPipe::Config& config)

- : EndToEndTest(kLongTimeoutMs, config),

+ StatsObserver()

+ : EndToEndTest(kLongTimeoutMs),

send_stream_(nullptr),

expected_send_ssrcs_(),

check_stats_event_(EventWrapper::Create()) {}

@@ -2542,13 +2532,11 @@ TEST_F(EndToEndTest, GetStats) {

std::string expected_cname_;

rtc::scoped_ptr<EventWrapper> check_stats_event_;

- };

+ } test;

FakeNetworkPipe::Config network_config;

network_config.loss_percent = 5;

- StatsObserver test(network_config);

- RunBaseTest(&test);

+ RunBaseTest(&test, network_config);

}

TEST_F(EndToEndTest, ReceiverReferenceTimeReportEnabled) {

@@ -2596,7 +2584,7 @@ TEST_F(EndToEndTest, TestReceivedRtpPacketStats) {

uint32_t sent_rtp_;

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, SendsSetSsrc) { TestSendsSetSsrcs(1, false); }

@@ -2676,7 +2664,7 @@ TEST_F(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {

std::map<uint32_t, bool> registered_rtx_ssrc_;

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {

@@ -2771,9 +2759,18 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {

} observer(use_rtx);

CreateCalls(Call::Config(), Call::Config());

- observer.SetReceivers(sender_call_->Receiver(), nullptr);

- CreateSendConfig(kNumSsrcs, observer.SendTransport());

+ test::PacketTransport send_transport(sender_call_.get(), &observer,

+ test::PacketTransport::kSender,

+ FakeNetworkPipe::Config());

+ test::PacketTransport receive_transport(nullptr, &observer,

+ test::PacketTransport::kReceiver,

+ FakeNetworkPipe::Config());

+ send_transport.SetReceiver(receiver_call_->Receiver());

+ receive_transport.SetReceiver(sender_call_->Receiver());

+ observer.SetTransports(&send_transport, &receive_transport);

+ CreateSendConfig(kNumSsrcs, &send_transport);

if (use_rtx) {

for (size_t i = 0; i < kNumSsrcs; ++i) {

@@ -2802,7 +2799,7 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {

encoder_config_.streams[i].max_bitrate_bps;

}

- CreateMatchingReceiveConfigs(observer.ReceiveTransport());

+ CreateMatchingReceiveConfigs(&receive_transport);

CreateStreams();

CreateFrameGeneratorCapturer();

@@ -2847,7 +2844,8 @@ void EndToEndTest::TestRtpStatePreservation(bool use_rtx) {

<< "Timed out waiting for all SSRCs to send packets.";

}

- observer.StopSending();

+ send_transport.StopSending();

+ receive_transport.StopSending();

Stop();

DestroyStreams();

@@ -3033,7 +3031,7 @@ TEST_F(EndToEndTest, RespectsNetworkState) {

int down_frames_ GUARDED_BY(test_crit_);

} test;

- RunBaseTest(&test);

+ RunBaseTest(&test, FakeNetworkPipe::Config());

}

TEST_F(EndToEndTest, CallReportsRttForSender) {

@@ -3044,9 +3042,9 @@ TEST_F(EndToEndTest, CallReportsRttForSender) {

FakeNetworkPipe::Config config;

config.queue_delay_ms = kSendDelayMs;

- test::DirectTransport sender_transport(config);

+ test::DirectTransport sender_transport(config, sender_call_.get());

config.queue_delay_ms = kReceiveDelayMs;

- test::DirectTransport receiver_transport(config);

+ test::DirectTransport receiver_transport(config, receiver_call_.get());

sender_transport.SetReceiver(receiver_call_->Receiver());

receiver_transport.SetReceiver(sender_call_->Receiver());

@@ -3108,7 +3106,7 @@ TEST_F(EndToEndTest, NewReceiveStreamsRespectNetworkDown) {

CreateCalls(Call::Config(), Call::Config());

receiver_call_->SignalNetworkState(kNetworkDown);

- test::DirectTransport sender_transport;

+ test::DirectTransport sender_transport(sender_call_.get());

sender_transport.SetReceiver(receiver_call_->Receiver());

CreateSendConfig(1, &sender_transport);

UnusedTransport transport;

« webrtc/test/rtp_rtcp_observer.h ('K') | « webrtc/test/rtp_rtcp_observer.h ('k') | webrtc/video/rampup_tests.h » ('j') | webrtc/video/video_send_stream_tests.cc » ('J')