Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 711 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..232fe0b3fe
--- /dev/null
+++ b/third_party/libwebrtc/call/rampup_tests.cc
@@ -0,0 +1,711 @@
+/*
+ *  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"
+#include "test/video_test_constants.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::VideoTestConstants::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::VideoTestConstants::kDefaultWidth,
+        test::VideoTestConstants::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::VideoTestConstants::kNackRtpHistoryMs;
+  send_config->rtp.ssrcs = video_ssrcs_;
+  if (rtx_) {
+    send_config->rtp.rtx.payload_type =
+        test::VideoTestConstants::kSendRtxPayloadType;
+    send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_;
+  }
+  if (red_) {
+    send_config->rtp.ulpfec.ulpfec_payload_type =
+        test::VideoTestConstants::kUlpfecPayloadType;
+    send_config->rtp.ulpfec.red_payload_type =
+        test::VideoTestConstants::kRedPayloadType;
+    if (rtx_) {
+      send_config->rtp.ulpfec.red_rtx_payload_type =
+          test::VideoTestConstants::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::VideoTestConstants::kFlexfecPayloadType;
+    send_config->rtp.flexfec.ssrc = test::VideoTestConstants::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::VideoTestConstants::kFlexfecPayloadType;
+  (*receive_configs)[0].rtp.remote_ssrc =
+      test::VideoTestConstants::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 webrtc_quick_perf_test
+  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