diff options
Diffstat (limited to 'third_party/libwebrtc/call/rampup_tests.cc')
| -rw-r--r-- | third_party/libwebrtc/call/rampup_tests.cc | 704 |
1 files changed, 704 insertions, 0 deletions
diff --git a/third_party/libwebrtc/call/rampup_tests.cc b/third_party/libwebrtc/call/rampup_tests.cc new file mode 100644 index 0000000000..3c89670a9f --- /dev/null +++ b/third_party/libwebrtc/call/rampup_tests.cc @@ -0,0 +1,704 @@ +/* + * 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 <memory> + +#include "absl/flags/flag.h" +#include "absl/strings/string_view.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 "api/test/metrics/global_metrics_logger_and_exporter.h" +#include "api/test/metrics/metric.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/gtest.h" + +ABSL_FLAG(std::string, + ramp_dump_name, + "", + "Filename for dumped received RTP stream."); + +namespace webrtc { +namespace { + +using ::webrtc::test::GetGlobalMetricsLogger; +using ::webrtc::test::ImprovementDirection; +using ::webrtc::test::Unit; + +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<uint32_t> GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) { + std::vector<uint32_t> ssrcs; + for (size_t i = 0; i != num_streams; ++i) + ssrcs.push_back(static_cast<uint32_t>(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, + bool rtx, + bool red, + bool report_perf_stats, + TaskQueueBase* task_queue) + : EndToEndTest(test::CallTest::kLongTimeout), + 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), + 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<VideoReceiveStreamInterface*>& receive_streams) { + send_stream_ = send_stream; +} + +BuiltInNetworkBehaviorConfig RampUpTester::GetSendTransportConfig() const { + return forward_transport_config_; +} + +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<VideoStream> CreateEncoderStreams( + int frame_width, + int frame_height, + const VideoEncoderConfig& encoder_config) override { + std::vector<VideoStream> streams = + test::CreateVideoStreams(frame_width, frame_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<VideoReceiveStreamInterface::Config>* 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<RampUpTester::VideoStreamFactory>(); + 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<VideoStream> 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.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 (VideoReceiveStreamInterface::Config& recv_config : *receive_configs) { + 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<AudioReceiveStreamInterface::Config>* receive_configs) { + if (num_audio_streams_ == 0) + return; + + send_config->rtp.ssrc = audio_ssrcs_[0]; + send_config->min_bitrate_bps = 6000; + send_config->max_bitrate_bps = 60000; + + for (AudioReceiveStreamInterface::Config& recv_config : *receive_configs) { + recv_config.rtp.remote_ssrc = send_config->rtp.ssrc; + } +} + +void RampUpTester::ModifyFlexfecConfigs( + std::vector<FlexfecReceiveStream::Config>* 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]; +} + +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::OnTransportCreated( + test::PacketTransport* to_receiver, + SimulatedNetworkInterface* sender_network, + test::PacketTransport* to_sender, + SimulatedNetworkInterface* receiver_network) { + RTC_DCHECK_RUN_ON(task_queue_); + + send_transport_ = to_receiver; + send_simulated_network_ = sender_network; +} + +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( + absl::string_view measurement, + size_t value, + Unit unit, + ImprovementDirection improvement_direction) const { + GetGlobalMetricsLogger()->LogSingleValueMetric( + measurement, + ::testing::UnitTest::GetInstance()->current_test_info()->name(), value, + unit, improvement_direction); +} + +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(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(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, Unit::kBytes, + ImprovementDirection::kBiggerIsBetter); + ReportResult("ramp-up-padding-sent", padding_sent, Unit::kBytes, + ImprovementDirection::kSmallerIsBetter); + ReportResult("ramp-up-rtx-media-sent", rtx_media_sent, Unit::kBytes, + ImprovementDirection::kBiggerIsBetter); + ReportResult("ramp-up-rtx-padding-sent", rtx_padding_sent, Unit::kBytes, + ImprovementDirection::kSmallerIsBetter); + if (ramp_up_finished_ms_ >= 0) { + ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_, + Unit::kMilliseconds, ImprovementDirection::kSmallerIsBetter); + } + ReportResult("ramp-up-average-network-latency", + send_transport_->GetAverageDelayMs(), Unit::kMilliseconds, + ImprovementDirection::kSmallerIsBetter); + } +} + +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, + bool rtx, + bool red, + const std::vector<int>& loss_rates, + bool report_perf_stats, + TaskQueueBase* task_queue) + : RampUpTester(num_video_streams, + num_audio_streams, + num_flexfec_streams, + start_bitrate_bps, + 0, + 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_) { + GetGlobalMetricsLogger()->LogSingleValueMetric( + "ramp_up_down_up" + GetModifierString(), "first_rampup", + now - state_start_ms_, Unit::kMilliseconds, + ImprovementDirection::kSmallerIsBetter); + } + // 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_) { + GetGlobalMetricsLogger()->LogSingleValueMetric( + "ramp_up_down_up" + GetModifierString(), "rampdown", + now - state_start_ms_, Unit::kMilliseconds, + ImprovementDirection::kSmallerIsBetter); + } + // 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_) { + GetGlobalMetricsLogger()->LogSingleValueMetric( + "ramp_up_down_up" + GetModifierString(), "second_rampup", + now - state_start_ms_, Unit::kMilliseconds, + ImprovementDirection::kSmallerIsBetter); + ReportResult("ramp-up-down-up-average-network-latency", + send_transport_->GetAverageDelayMs(), + Unit::kMilliseconds, + ImprovementDirection::kSmallerIsBetter); + } + // 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<RtcEventLogOutputFile>( + dump_name + ".send.rtc.dat", RtcEventLog::kUnlimitedOutput), + RtcEventLog::kImmediateOutput) && + recv_event_log_->StartLogging( + std::make_unique<RtcEventLogOutputFile>( + dump_name + ".recv.rtc.dat", RtcEventLog::kUnlimitedOutput), + RtcEventLog::kImmediateOutput); + RTC_DCHECK(event_log_started); + } + } + + private: + const std::unique_ptr<TaskQueueFactory> task_queue_factory_; + RtcEventLogFactory rtc_event_log_factory_; +}; + +static const uint32_t kStartBitrateBps = 60000; + +TEST_F(RampUpTest, UpDownUpAbsSendTimeSimulcastRedRtx) { + std::vector<int> loss_rates = {0, 0, 0, 0}; + RegisterRtpExtension( + RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); + RampUpDownUpTester test(3, 0, 0, kStartBitrateBps, 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<int> loss_rates = {0, 0, 0, 0}; + RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, + kTransportSequenceNumberExtensionId)); + RampUpDownUpTester test(3, 0, 0, kStartBitrateBps, 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<int> loss_rates = {20, 0, 0, 0}; + RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, + kTransportSequenceNumberExtensionId)); + RampUpDownUpTester test(1, 0, 1, kStartBitrateBps, 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<int> loss_rates = {0, 0, 0, 0}; + RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, + kTransportSequenceNumberExtensionId)); + RampUpDownUpTester test(3, 1, 0, kStartBitrateBps, true, false, loss_rates, + false, task_queue()); + RunBaseTest(&test); +} + +TEST_F(RampUpTest, UpDownUpAudioTransportSequenceNumberRtx) { + std::vector<int> loss_rates = {0, 0, 0, 0}; + RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, + kTransportSequenceNumberExtensionId)); + RampUpDownUpTester test(0, 1, 0, kStartBitrateBps, true, false, loss_rates, + false, task_queue()); + RunBaseTest(&test); +} + +TEST_F(RampUpTest, TOffsetSimulcastRedRtx) { + RegisterRtpExtension(RtpExtension(RtpExtension::kTimestampOffsetUri, + kTransmissionTimeOffsetExtensionId)); + RampUpTester test(3, 0, 0, 0, 0, true, true, true, task_queue()); + RunBaseTest(&test); +} + +TEST_F(RampUpTest, AbsSendTime) { + RegisterRtpExtension( + RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); + RampUpTester test(1, 0, 0, 0, 0, false, false, false, task_queue()); + RunBaseTest(&test); +} + +TEST_F(RampUpTest, AbsSendTimeSimulcastRedRtx) { + RegisterRtpExtension( + RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId)); + RampUpTester test(3, 0, 0, 0, 0, true, true, true, task_queue()); + RunBaseTest(&test); +} + +TEST_F(RampUpTest, TransportSequenceNumber) { + RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, + kTransportSequenceNumberExtensionId)); + RampUpTester test(1, 0, 0, 0, 0, false, false, false, task_queue()); + RunBaseTest(&test); +} + +TEST_F(RampUpTest, TransportSequenceNumberSimulcast) { + RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, + kTransportSequenceNumberExtensionId)); + RampUpTester test(3, 0, 0, 0, 0, false, false, false, task_queue()); + RunBaseTest(&test); +} + +TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) { + RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, + kTransportSequenceNumberExtensionId)); + RampUpTester test(3, 0, 0, 0, 0, true, true, true, task_queue()); + RunBaseTest(&test); +} + +TEST_F(RampUpTest, AudioTransportSequenceNumber) { + RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, + kTransportSequenceNumberExtensionId)); + RampUpTester test(0, 1, 0, 300000, 10000, false, false, false, task_queue()); + RunBaseTest(&test); +} + +} // namespace webrtc |