/* * 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 "call/rampup_tests.h" #include #include "absl/flags/flag.h" #include "api/rtc_event_log/rtc_event_log_factory.h" #include "api/rtc_event_log_output_file.h" #include "api/task_queue/default_task_queue_factory.h" #include "api/task_queue/task_queue_base.h" #include "api/task_queue/task_queue_factory.h" #include "call/fake_network_pipe.h" #include "rtc_base/checks.h" #include "rtc_base/logging.h" #include "rtc_base/platform_thread.h" #include "rtc_base/string_encode.h" #include "rtc_base/task_queue_for_test.h" #include "rtc_base/time_utils.h" #include "test/encoder_settings.h" #include "test/field_trial.h" #include "test/gtest.h" #include "test/testsupport/perf_test.h" ABSL_FLAG(std::string, ramp_dump_name, "", "Filename for dumped received RTP stream."); namespace webrtc { namespace { constexpr TimeDelta kPollInterval = TimeDelta::Millis(20); static const int kExpectedHighVideoBitrateBps = 80000; static const int kExpectedHighAudioBitrateBps = 30000; static const int kLowBandwidthLimitBps = 20000; // Set target detected bitrate to slightly larger than the target bitrate to // avoid flakiness. static const int kLowBitrateMarginBps = 2000; std::vector GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) { std::vector ssrcs; for (size_t i = 0; i != num_streams; ++i) ssrcs.push_back(static_cast(ssrc_offset + i)); return ssrcs; } } // namespace RampUpTester::RampUpTester(size_t num_video_streams, size_t num_audio_streams, size_t num_flexfec_streams, unsigned int start_bitrate_bps, int64_t min_run_time_ms, const std::string& extension_type, bool rtx, bool red, bool report_perf_stats, TaskQueueBase* task_queue) : EndToEndTest(test::CallTest::kLongTimeoutMs), clock_(Clock::GetRealTimeClock()), num_video_streams_(num_video_streams), num_audio_streams_(num_audio_streams), num_flexfec_streams_(num_flexfec_streams), rtx_(rtx), red_(red), report_perf_stats_(report_perf_stats), sender_call_(nullptr), send_stream_(nullptr), send_transport_(nullptr), send_simulated_network_(nullptr), start_bitrate_bps_(start_bitrate_bps), min_run_time_ms_(min_run_time_ms), expected_bitrate_bps_(0), test_start_ms_(-1), ramp_up_finished_ms_(-1), extension_type_(extension_type), video_ssrcs_(GenerateSsrcs(num_video_streams_, 100)), video_rtx_ssrcs_(GenerateSsrcs(num_video_streams_, 200)), audio_ssrcs_(GenerateSsrcs(num_audio_streams_, 300)), task_queue_(task_queue) { if (red_) EXPECT_EQ(0u, num_flexfec_streams_); EXPECT_LE(num_audio_streams_, 1u); } RampUpTester::~RampUpTester() = default; void RampUpTester::ModifySenderBitrateConfig( BitrateConstraints* bitrate_config) { if (start_bitrate_bps_ != 0) { bitrate_config->start_bitrate_bps = start_bitrate_bps_; } bitrate_config->min_bitrate_bps = 10000; } void RampUpTester::OnVideoStreamsCreated( VideoSendStream* send_stream, const std::vector& receive_streams) { send_stream_ = send_stream; } std::unique_ptr RampUpTester::CreateSendTransport( TaskQueueBase* task_queue, Call* sender_call) { auto network = std::make_unique(forward_transport_config_); send_simulated_network_ = network.get(); auto send_transport = std::make_unique( task_queue, sender_call, this, test::PacketTransport::kSender, test::CallTest::payload_type_map_, std::make_unique(Clock::GetRealTimeClock(), std::move(network))); send_transport_ = send_transport.get(); return send_transport; } size_t RampUpTester::GetNumVideoStreams() const { return num_video_streams_; } size_t RampUpTester::GetNumAudioStreams() const { return num_audio_streams_; } size_t RampUpTester::GetNumFlexfecStreams() const { return num_flexfec_streams_; } class RampUpTester::VideoStreamFactory : public VideoEncoderConfig::VideoStreamFactoryInterface { public: VideoStreamFactory() {} private: std::vector CreateEncoderStreams( int width, int height, const VideoEncoderConfig& encoder_config) override { std::vector streams = test::CreateVideoStreams(width, height, encoder_config); if (encoder_config.number_of_streams == 1) { streams[0].target_bitrate_bps = streams[0].max_bitrate_bps = 2000000; } return streams; } }; void RampUpTester::ModifyVideoConfigs( VideoSendStream::Config* send_config, std::vector* receive_configs, VideoEncoderConfig* encoder_config) { send_config->suspend_below_min_bitrate = true; encoder_config->number_of_streams = num_video_streams_; encoder_config->max_bitrate_bps = 2000000; encoder_config->video_stream_factory = rtc::make_ref_counted(); if (num_video_streams_ == 1) { // For single stream rampup until 1mbps expected_bitrate_bps_ = kSingleStreamTargetBps; } else { // To ensure simulcast rate allocation. send_config->rtp.payload_name = "VP8"; encoder_config->codec_type = kVideoCodecVP8; std::vector streams = test::CreateVideoStreams( test::CallTest::kDefaultWidth, test::CallTest::kDefaultHeight, *encoder_config); // For multi stream rampup until all streams are being sent. That means // enough bitrate to send all the target streams plus the min bitrate of // the last one. expected_bitrate_bps_ = streams.back().min_bitrate_bps; for (size_t i = 0; i < streams.size() - 1; ++i) { expected_bitrate_bps_ += streams[i].target_bitrate_bps; } } send_config->rtp.extensions.clear(); bool remb; bool transport_cc; if (extension_type_ == RtpExtension::kAbsSendTimeUri) { remb = true; transport_cc = false; send_config->rtp.extensions.push_back( RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId)); } else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { remb = false; transport_cc = true; send_config->rtp.extensions.push_back(RtpExtension( extension_type_.c_str(), kTransportSequenceNumberExtensionId)); } else { remb = true; transport_cc = false; send_config->rtp.extensions.push_back(RtpExtension( extension_type_.c_str(), kTransmissionTimeOffsetExtensionId)); } send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs; send_config->rtp.ssrcs = video_ssrcs_; if (rtx_) { send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType; send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_; } if (red_) { send_config->rtp.ulpfec.ulpfec_payload_type = test::CallTest::kUlpfecPayloadType; send_config->rtp.ulpfec.red_payload_type = test::CallTest::kRedPayloadType; if (rtx_) { send_config->rtp.ulpfec.red_rtx_payload_type = test::CallTest::kRtxRedPayloadType; } } size_t i = 0; for (VideoReceiveStream::Config& recv_config : *receive_configs) { recv_config.rtp.transport_cc = transport_cc; recv_config.rtp.extensions = send_config->rtp.extensions; recv_config.decoders.reserve(1); recv_config.decoders[0].payload_type = send_config->rtp.payload_type; recv_config.decoders[0].video_format = SdpVideoFormat(send_config->rtp.payload_name); recv_config.rtp.remote_ssrc = video_ssrcs_[i]; recv_config.rtp.nack.rtp_history_ms = send_config->rtp.nack.rtp_history_ms; if (red_) { recv_config.rtp.red_payload_type = send_config->rtp.ulpfec.red_payload_type; recv_config.rtp.ulpfec_payload_type = send_config->rtp.ulpfec.ulpfec_payload_type; if (rtx_) { recv_config.rtp.rtx_associated_payload_types [send_config->rtp.ulpfec.red_rtx_payload_type] = send_config->rtp.ulpfec.red_payload_type; } } if (rtx_) { recv_config.rtp.rtx_ssrc = video_rtx_ssrcs_[i]; recv_config.rtp .rtx_associated_payload_types[send_config->rtp.rtx.payload_type] = send_config->rtp.payload_type; } ++i; } RTC_DCHECK_LE(num_flexfec_streams_, 1); if (num_flexfec_streams_ == 1) { send_config->rtp.flexfec.payload_type = test::CallTest::kFlexfecPayloadType; send_config->rtp.flexfec.ssrc = test::CallTest::kFlexfecSendSsrc; send_config->rtp.flexfec.protected_media_ssrcs = {video_ssrcs_[0]}; } } void RampUpTester::ModifyAudioConfigs( AudioSendStream::Config* send_config, std::vector* receive_configs) { if (num_audio_streams_ == 0) return; EXPECT_NE(RtpExtension::kTimestampOffsetUri, extension_type_) << "Audio BWE not supported with toffset."; EXPECT_NE(RtpExtension::kAbsSendTimeUri, extension_type_) << "Audio BWE not supported with abs-send-time."; send_config->rtp.ssrc = audio_ssrcs_[0]; send_config->rtp.extensions.clear(); send_config->min_bitrate_bps = 6000; send_config->max_bitrate_bps = 60000; bool transport_cc = false; if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { transport_cc = true; send_config->rtp.extensions.push_back(RtpExtension( extension_type_.c_str(), kTransportSequenceNumberExtensionId)); } for (AudioReceiveStream::Config& recv_config : *receive_configs) { recv_config.rtp.transport_cc = transport_cc; recv_config.rtp.extensions = send_config->rtp.extensions; recv_config.rtp.remote_ssrc = send_config->rtp.ssrc; } } void RampUpTester::ModifyFlexfecConfigs( std::vector* receive_configs) { if (num_flexfec_streams_ == 0) return; RTC_DCHECK_EQ(1, num_flexfec_streams_); (*receive_configs)[0].payload_type = test::CallTest::kFlexfecPayloadType; (*receive_configs)[0].rtp.remote_ssrc = test::CallTest::kFlexfecSendSsrc; (*receive_configs)[0].protected_media_ssrcs = {video_ssrcs_[0]}; (*receive_configs)[0].rtp.local_ssrc = video_ssrcs_[0]; if (extension_type_ == RtpExtension::kAbsSendTimeUri) { (*receive_configs)[0].rtp.transport_cc = false; (*receive_configs)[0].rtp.extensions.push_back( RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId)); } else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { (*receive_configs)[0].rtp.transport_cc = true; (*receive_configs)[0].rtp.extensions.push_back(RtpExtension( extension_type_.c_str(), kTransportSequenceNumberExtensionId)); } } void RampUpTester::OnCallsCreated(Call* sender_call, Call* receiver_call) { RTC_DCHECK(sender_call); sender_call_ = sender_call; pending_task_ = RepeatingTaskHandle::Start(task_queue_, [this] { PollStats(); return kPollInterval; }); } void RampUpTester::PollStats() { RTC_DCHECK_RUN_ON(task_queue_); Call::Stats stats = sender_call_->GetStats(); EXPECT_GE(expected_bitrate_bps_, 0); if (stats.send_bandwidth_bps >= expected_bitrate_bps_ && (min_run_time_ms_ == -1 || clock_->TimeInMilliseconds() - test_start_ms_ >= min_run_time_ms_)) { ramp_up_finished_ms_ = clock_->TimeInMilliseconds(); observation_complete_.Set(); pending_task_.Stop(); } } void RampUpTester::ReportResult(const std::string& measurement, size_t value, const std::string& units) const { webrtc::test::PrintResult( measurement, "", ::testing::UnitTest::GetInstance()->current_test_info()->name(), value, units, false); } void RampUpTester::AccumulateStats(const VideoSendStream::StreamStats& stream, size_t* total_packets_sent, size_t* total_sent, size_t* padding_sent, size_t* media_sent) const { *total_packets_sent += stream.rtp_stats.transmitted.packets + stream.rtp_stats.retransmitted.packets + stream.rtp_stats.fec.packets; *total_sent += stream.rtp_stats.transmitted.TotalBytes() + stream.rtp_stats.retransmitted.TotalBytes() + stream.rtp_stats.fec.TotalBytes(); *padding_sent += stream.rtp_stats.transmitted.padding_bytes + stream.rtp_stats.retransmitted.padding_bytes + stream.rtp_stats.fec.padding_bytes; *media_sent += stream.rtp_stats.MediaPayloadBytes(); } void RampUpTester::TriggerTestDone() { RTC_DCHECK_GE(test_start_ms_, 0); // Stop polling stats. // Corner case for field_trials=WebRTC-QuickPerfTest/Enabled/ SendTask(RTC_FROM_HERE, task_queue_, [this] { pending_task_.Stop(); }); // TODO(holmer): Add audio send stats here too when those APIs are available. if (!send_stream_) return; VideoSendStream::Stats send_stats; SendTask(RTC_FROM_HERE, task_queue_, [&] { send_stats = send_stream_->GetStats(); }); send_stream_ = nullptr; // To avoid dereferencing a bad pointer. size_t total_packets_sent = 0; size_t total_sent = 0; size_t padding_sent = 0; size_t media_sent = 0; for (uint32_t ssrc : video_ssrcs_) { AccumulateStats(send_stats.substreams[ssrc], &total_packets_sent, &total_sent, &padding_sent, &media_sent); } size_t rtx_total_packets_sent = 0; size_t rtx_total_sent = 0; size_t rtx_padding_sent = 0; size_t rtx_media_sent = 0; for (uint32_t rtx_ssrc : video_rtx_ssrcs_) { AccumulateStats(send_stats.substreams[rtx_ssrc], &rtx_total_packets_sent, &rtx_total_sent, &rtx_padding_sent, &rtx_media_sent); } if (report_perf_stats_) { ReportResult("ramp-up-media-sent", media_sent, "bytes"); ReportResult("ramp-up-padding-sent", padding_sent, "bytes"); ReportResult("ramp-up-rtx-media-sent", rtx_media_sent, "bytes"); ReportResult("ramp-up-rtx-padding-sent", rtx_padding_sent, "bytes"); if (ramp_up_finished_ms_ >= 0) { ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_, "milliseconds"); } ReportResult("ramp-up-average-network-latency", send_transport_->GetAverageDelayMs(), "milliseconds"); } } void RampUpTester::PerformTest() { test_start_ms_ = clock_->TimeInMilliseconds(); EXPECT_TRUE(Wait()) << "Timed out while waiting for ramp-up to complete."; TriggerTestDone(); } RampUpDownUpTester::RampUpDownUpTester(size_t num_video_streams, size_t num_audio_streams, size_t num_flexfec_streams, unsigned int start_bitrate_bps, const std::string& extension_type, bool rtx, bool red, const std::vector& loss_rates, bool report_perf_stats, TaskQueueBase* task_queue) : RampUpTester(num_video_streams, num_audio_streams, num_flexfec_streams, start_bitrate_bps, 0, extension_type, rtx, red, report_perf_stats, task_queue), link_rates_({4 * GetExpectedHighBitrate() / (3 * 1000), kLowBandwidthLimitBps / 1000, 4 * GetExpectedHighBitrate() / (3 * 1000), 0}), test_state_(kFirstRampup), next_state_(kTransitionToNextState), state_start_ms_(clock_->TimeInMilliseconds()), interval_start_ms_(clock_->TimeInMilliseconds()), sent_bytes_(0), loss_rates_(loss_rates) { forward_transport_config_.link_capacity_kbps = link_rates_[test_state_]; forward_transport_config_.queue_delay_ms = 100; forward_transport_config_.loss_percent = loss_rates_[test_state_]; } RampUpDownUpTester::~RampUpDownUpTester() {} void RampUpDownUpTester::PollStats() { if (test_state_ == kTestEnd) { pending_task_.Stop(); } int transmit_bitrate_bps = 0; bool suspended = false; if (num_video_streams_ > 0 && send_stream_) { webrtc::VideoSendStream::Stats stats = send_stream_->GetStats(); for (const auto& it : stats.substreams) { transmit_bitrate_bps += it.second.total_bitrate_bps; } suspended = stats.suspended; } if (num_audio_streams_ > 0 && sender_call_) { // An audio send stream doesn't have bitrate stats, so the call send BW is // currently used instead. transmit_bitrate_bps = sender_call_->GetStats().send_bandwidth_bps; } EvolveTestState(transmit_bitrate_bps, suspended); } void RampUpDownUpTester::ModifyReceiverBitrateConfig( BitrateConstraints* bitrate_config) { bitrate_config->min_bitrate_bps = 10000; } std::string RampUpDownUpTester::GetModifierString() const { std::string str("_"); if (num_video_streams_ > 0) { str += rtc::ToString(num_video_streams_); str += "stream"; str += (num_video_streams_ > 1 ? "s" : ""); str += "_"; } if (num_audio_streams_ > 0) { str += rtc::ToString(num_audio_streams_); str += "stream"; str += (num_audio_streams_ > 1 ? "s" : ""); str += "_"; } str += (rtx_ ? "" : "no"); str += "rtx_"; str += (red_ ? "" : "no"); str += "red"; return str; } int RampUpDownUpTester::GetExpectedHighBitrate() const { int expected_bitrate_bps = 0; if (num_audio_streams_ > 0) expected_bitrate_bps += kExpectedHighAudioBitrateBps; if (num_video_streams_ > 0) expected_bitrate_bps += kExpectedHighVideoBitrateBps; return expected_bitrate_bps; } size_t RampUpDownUpTester::GetFecBytes() const { size_t flex_fec_bytes = 0; if (num_flexfec_streams_ > 0) { webrtc::VideoSendStream::Stats stats = send_stream_->GetStats(); for (const auto& kv : stats.substreams) flex_fec_bytes += kv.second.rtp_stats.fec.TotalBytes(); } return flex_fec_bytes; } bool RampUpDownUpTester::ExpectingFec() const { return num_flexfec_streams_ > 0 && forward_transport_config_.loss_percent > 0; } void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) { int64_t now = clock_->TimeInMilliseconds(); switch (test_state_) { case kFirstRampup: EXPECT_FALSE(suspended); if (bitrate_bps >= GetExpectedHighBitrate()) { if (report_perf_stats_) { webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(), "first_rampup", now - state_start_ms_, "ms", false); } // Apply loss during the transition between states if FEC is enabled. forward_transport_config_.loss_percent = loss_rates_[test_state_]; test_state_ = kTransitionToNextState; next_state_ = kLowRate; } break; case kLowRate: { // Audio streams are never suspended. bool check_suspend_state = num_video_streams_ > 0; if (bitrate_bps < kLowBandwidthLimitBps + kLowBitrateMarginBps && suspended == check_suspend_state) { if (report_perf_stats_) { webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(), "rampdown", now - state_start_ms_, "ms", false); } // Apply loss during the transition between states if FEC is enabled. forward_transport_config_.loss_percent = loss_rates_[test_state_]; test_state_ = kTransitionToNextState; next_state_ = kSecondRampup; } break; } case kSecondRampup: if (bitrate_bps >= GetExpectedHighBitrate() && !suspended) { if (report_perf_stats_) { webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(), "second_rampup", now - state_start_ms_, "ms", false); ReportResult("ramp-up-down-up-average-network-latency", send_transport_->GetAverageDelayMs(), "milliseconds"); } // Apply loss during the transition between states if FEC is enabled. forward_transport_config_.loss_percent = loss_rates_[test_state_]; test_state_ = kTransitionToNextState; next_state_ = kTestEnd; } break; case kTestEnd: observation_complete_.Set(); break; case kTransitionToNextState: if (!ExpectingFec() || GetFecBytes() > 0) { test_state_ = next_state_; forward_transport_config_.link_capacity_kbps = link_rates_[test_state_]; // No loss while ramping up and down as it may affect the BWE // negatively, making the test flaky. forward_transport_config_.loss_percent = 0; state_start_ms_ = now; interval_start_ms_ = now; sent_bytes_ = 0; send_simulated_network_->SetConfig(forward_transport_config_); } break; } } class RampUpTest : public test::CallTest { public: RampUpTest() : task_queue_factory_(CreateDefaultTaskQueueFactory()), rtc_event_log_factory_(task_queue_factory_.get()) { std::string dump_name(absl::GetFlag(FLAGS_ramp_dump_name)); if (!dump_name.empty()) { send_event_log_ = rtc_event_log_factory_.CreateRtcEventLog( RtcEventLog::EncodingType::Legacy); recv_event_log_ = rtc_event_log_factory_.CreateRtcEventLog( RtcEventLog::EncodingType::Legacy); bool event_log_started = send_event_log_->StartLogging( std::make_unique( dump_name + ".send.rtc.dat", RtcEventLog::kUnlimitedOutput), RtcEventLog::kImmediateOutput) && recv_event_log_->StartLogging( std::make_unique( dump_name + ".recv.rtc.dat", RtcEventLog::kUnlimitedOutput), RtcEventLog::kImmediateOutput); RTC_DCHECK(event_log_started); } } private: const std::unique_ptr task_queue_factory_; RtcEventLogFactory rtc_event_log_factory_; }; static const uint32_t kStartBitrateBps = 60000; TEST_F(RampUpTest, UpDownUpAbsSendTimeSimulcastRedRtx) { std::vector loss_rates = {0, 0, 0, 0}; RampUpDownUpTester test(3, 0, 0, kStartBitrateBps, RtpExtension::kAbsSendTimeUri, true, true, loss_rates, true, task_queue()); RunBaseTest(&test); } // TODO(bugs.webrtc.org/8878) #if defined(WEBRTC_MAC) #define MAYBE_UpDownUpTransportSequenceNumberRtx \ DISABLED_UpDownUpTransportSequenceNumberRtx #else #define MAYBE_UpDownUpTransportSequenceNumberRtx \ UpDownUpTransportSequenceNumberRtx #endif TEST_F(RampUpTest, MAYBE_UpDownUpTransportSequenceNumberRtx) { std::vector loss_rates = {0, 0, 0, 0}; RampUpDownUpTester test(3, 0, 0, kStartBitrateBps, RtpExtension::kTransportSequenceNumberUri, true, false, loss_rates, true, task_queue()); RunBaseTest(&test); } // TODO(holmer): Tests which don't report perf stats should be moved to a // different executable since they per definition are not perf tests. // This test is disabled because it crashes on Linux, and is flaky on other // platforms. See: crbug.com/webrtc/7919 TEST_F(RampUpTest, DISABLED_UpDownUpTransportSequenceNumberPacketLoss) { std::vector loss_rates = {20, 0, 0, 0}; RampUpDownUpTester test(1, 0, 1, kStartBitrateBps, RtpExtension::kTransportSequenceNumberUri, true, false, loss_rates, false, task_queue()); RunBaseTest(&test); } // TODO(bugs.webrtc.org/8878) #if defined(WEBRTC_MAC) #define MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx \ DISABLED_UpDownUpAudioVideoTransportSequenceNumberRtx #else #define MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx \ UpDownUpAudioVideoTransportSequenceNumberRtx #endif TEST_F(RampUpTest, MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx) { std::vector loss_rates = {0, 0, 0, 0}; RampUpDownUpTester test(3, 1, 0, kStartBitrateBps, RtpExtension::kTransportSequenceNumberUri, true, false, loss_rates, false, task_queue()); RunBaseTest(&test); } TEST_F(RampUpTest, UpDownUpAudioTransportSequenceNumberRtx) { std::vector loss_rates = {0, 0, 0, 0}; RampUpDownUpTester test(0, 1, 0, kStartBitrateBps, RtpExtension::kTransportSequenceNumberUri, true, false, loss_rates, false, task_queue()); RunBaseTest(&test); } TEST_F(RampUpTest, TOffsetSimulcastRedRtx) { RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTimestampOffsetUri, true, true, true, task_queue()); RunBaseTest(&test); } TEST_F(RampUpTest, AbsSendTime) { RampUpTester test(1, 0, 0, 0, 0, RtpExtension::kAbsSendTimeUri, false, false, false, task_queue()); RunBaseTest(&test); } TEST_F(RampUpTest, AbsSendTimeSimulcastRedRtx) { RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kAbsSendTimeUri, true, true, true, task_queue()); RunBaseTest(&test); } TEST_F(RampUpTest, TransportSequenceNumber) { RampUpTester test(1, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri, false, false, false, task_queue()); RunBaseTest(&test); } TEST_F(RampUpTest, TransportSequenceNumberSimulcast) { RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri, false, false, false, task_queue()); RunBaseTest(&test); } TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) { RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri, true, true, true, task_queue()); RunBaseTest(&test); } TEST_F(RampUpTest, AudioTransportSequenceNumber) { RampUpTester test(0, 1, 0, 300000, 10000, RtpExtension::kTransportSequenceNumberUri, false, false, false, task_queue()); RunBaseTest(&test); } } // namespace webrtc