Quelle  video_send_stream_tests.cc   Sprache: C

/*
 *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */
#include <algorithm>  // max
#include <cstddef>
#include <cstdint>
#include <cstring>
#include <map>
#include <memory>
#include <numeric>
#include <optional>
#include <set>
#include <string>
#include <tuple>
#include <utility>
#include <vector>

#include "absl/strings/match.h"
#include "absl/types/variant.h"
#include "api/array_view.h"
#include "api/environment/environment.h"
#include "api/environment/environment_factory.h"
#include "api/fec_controller_override.h"
#include "api/make_ref_counted.h"
#include "api/rtp_headers.h"
#include "api/rtp_parameters.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/metrics/global_metrics_logger_and_exporter.h"
#include "api/test/metrics/metric.h"
#include "api/test/simulated_network.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "api/transport/bitrate_settings.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video/encoded_image.h"
#include "api/video/video_bitrate_allocation.h"
#include "api/video/video_bitrate_allocator.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "api/video/video_codec_type.h"
#include "api/video/video_content_type.h"
#include "api/video/video_frame_type.h"
#include "api/video/video_rotation.h"
#include "api/video/video_sink_interface.h"
#include "api/video/video_source_interface.h"
#include "api/video_codecs/scalability_mode.h"
#include "api/video_codecs/sdp_video_format.h"
#include "api/video_codecs/video_codec.h"
#include "api/video_codecs/video_encoder.h"
#include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "call/call.h"
#include "call/fake_network_pipe.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.h"
#include "media/engine/internal_encoder_factory.h"
#include "media/engine/simulcast_encoder_adapter.h"
#include "media/engine/webrtc_video_engine.h"
#include "modules/include/module_common_types_public.h"
#include "modules/rtp_rtcp/include/receive_statistics.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "modules/rtp_rtcp/source/rtcp_sender.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
#include "modules/video_coding/codecs/interface/common_constants.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
#include "modules/video_coding/codecs/vp9/include/vp9.h"
#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
#include "modules/video_coding/svc/create_scalability_structure.h"
#include "modules/video_coding/svc/scalability_mode_util.h"
#include "modules/video_coding/svc/scalable_video_controller.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/experiments/alr_experiment.h"
#include "rtc_base/logging.h"
#include "rtc_base/network_route.h"
#include "rtc_base/rate_limiter.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread.h"
#include "rtc_base/thread_annotations.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/unique_id_generator.h"
#include "system_wrappers/include/sleep.h"
#include "test/call_test.h"
#include "test/configurable_frame_size_encoder.h"
#include "test/encoder_settings.h"
#include "test/fake_encoder.h"
#include "test/field_trial.h"
#include "test/frame_forwarder.h"
#include "test/frame_generator_capturer.h"
#include "test/frame_utils.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/null_transport.h"
#include "test/rtcp_packet_parser.h"
#include "test/rtp_rtcp_observer.h"
#include "test/scoped_key_value_config.h"
#include "test/video_encoder_proxy_factory.h"
#include "test/video_test_constants.h"
#include "video/config/video_encoder_config.h"
#include "video/transport_adapter.h"
#include "video/video_send_stream_impl.h"

namespace webrtc {
namespace test {
class VideoSendStreamPeer {
 public:
  explicit VideoSendStreamPeer(webrtc::VideoSendStream* base_class_stream)
      : internal_stream_(
            static_cast<internal::VideoSendStreamImpl*>(base_class_stream)) {}
  std::optional<float> GetPacingFactorOverride() const {
    return internal_stream_->GetPacingFactorOverride();
  }

 private:
  internal::VideoSendStreamImpl const* const internal_stream_;
};
}  // namespace test

namespace {
enum : int {  // The first valid value is 1.
  kAbsSendTimeExtensionId = 1,
  kTimestampOffsetExtensionId,
  kTransportSequenceNumberExtensionId,
  kVideoContentTypeExtensionId,
  kVideoRotationExtensionId,
  kVideoTimingExtensionId,
};

// Readability convenience enum for `WaitBitrateChanged()`.
enum class WaitUntil : bool { kZero = false, kNonZero = true };

constexpr int64_t kRtcpIntervalMs = 1000;

// Some of the test cases are expected to time out.
// Use a shorter timeout window than the default one for those.
constexpr TimeDelta kReducedTimeout = TimeDelta::Seconds(10);

enum VideoFormat {
  kGeneric,
  kVP8,
};

struct Vp9TestParams {
  std::string scalability_mode;
  uint8_t num_spatial_layers;
  uint8_t num_temporal_layers;
  InterLayerPredMode inter_layer_pred;
};

using ParameterizationType = std::tuple<Vp9TestParams, bool>;

std::string ParamInfoToStr(
    const testing::TestParamInfo<ParameterizationType>& info) {
  rtc::StringBuilder sb;
  sb << std::get<0>(info.param).scalability_mode << "_"
     << (std::get<1>(info.param) ? "WithIdentifier" : "WithoutIdentifier");
  return sb.str();
}

}  // namespace

class VideoSendStreamTest : public test::CallTest {
 public:
  VideoSendStreamTest() {
    RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri,
                                      kTransportSequenceNumberExtensionId));
  }

 protected:
  void TestNackRetransmission(uint32_t retransmit_ssrc,
                              uint8_t retransmit_payload_type);
  void TestPacketFragmentationSize(VideoFormat format, bool with_fec);

  void TestVp9NonFlexMode(const Vp9TestParams& params,
                          bool use_scalability_mode_identifier);

  void TestRequestSourceRotateVideo(bool support_orientation_ext);

  void TestTemporalLayers(VideoEncoderFactory* encoder_factory,
                          const std::string& payload_name,
                          const std::vector<int>& num_temporal_layers,
                          const std::vector<ScalabilityMode>& scalability_mode);
};

TEST_F(VideoSendStreamTest, CanStartStartedStream) {
  SendTask(task_queue(), [this]() {
    CreateSenderCall();

    test::NullTransport transport;
    CreateSendConfig(1, 0, 0, &transport);
    CreateVideoStreams();
    GetVideoSendStream()->Start();
    GetVideoSendStream()->Start();
    DestroyStreams();
    DestroyCalls();
  });
}

TEST_F(VideoSendStreamTest, CanStopStoppedStream) {
  SendTask(task_queue(), [this]() {
    CreateSenderCall();

    test::NullTransport transport;
    CreateSendConfig(1, 0, 0, &transport);
    CreateVideoStreams();
    GetVideoSendStream()->Stop();
    GetVideoSendStream()->Stop();
    DestroyStreams();
    DestroyCalls();
  });
}

TEST_F(VideoSendStreamTest, SupportsCName) {
  static std::string kCName = "PjQatC14dGfbVwGPUOA9IH7RlsFDbWl4AhXEiDsBizo=";
  class CNameObserver : public test::SendTest {
   public:
    CNameObserver() : SendTest(test::VideoTestConstants::kDefaultTimeout) {}

   private:
    Action OnSendRtcp(rtc::ArrayView<const uint8_t> packet) override {
      test::RtcpPacketParser parser;
      EXPECT_TRUE(parser.Parse(packet));
      if (parser.sdes()->num_packets() > 0) {
        EXPECT_EQ(1u, parser.sdes()->chunks().size());
        EXPECT_EQ(kCName, parser.sdes()->chunks()[0].cname);

        observation_complete_.Set();
      }

      return SEND_PACKET;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      send_config->rtp.c_name = kCName;
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait()) << "Timed out while waiting for RTCP with CNAME.";
    }
  } test;

  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsAbsoluteSendTime) {
  class AbsoluteSendTimeObserver : public test::SendTest {
   public:
    AbsoluteSendTimeObserver()
        : SendTest(test::VideoTestConstants::kDefaultTimeout) {
      extensions_.Register<AbsoluteSendTime>(kAbsSendTimeExtensionId);
    }

    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      RtpPacket rtp_packet(&extensions_);
      EXPECT_TRUE(rtp_packet.Parse(packet));

      uint32_t abs_send_time = 0;
      EXPECT_FALSE(rtp_packet.HasExtension<TransmissionOffset>());
      EXPECT_TRUE(rtp_packet.GetExtension<AbsoluteSendTime>(&abs_send_time));
      if (abs_send_time != 0) {
        // Wait for at least one packet with a non-zero send time. The send time
        // is a 16-bit value derived from the system clock, and it is valid
        // for a packet to have a zero send time. To tell that from an
        // unpopulated value we'll wait for a packet with non-zero send time.
        observation_complete_.Set();
      } else {
        RTC_LOG(LS_WARNING)
            << "Got a packet with zero absoluteSendTime, waiting"
               " for another packet...";
      }

      return SEND_PACKET;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      send_config->rtp.extensions.clear();
      send_config->rtp.extensions.push_back(
          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
    }

   private:
    RtpHeaderExtensionMap extensions_;
  } test;

  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
  static const int kEncodeDelayMs = 5;
  class TransmissionTimeOffsetObserver : public test::SendTest {
   public:
    TransmissionTimeOffsetObserver()
        : SendTest(test::VideoTestConstants::kDefaultTimeout),
          encoder_factory_([](const Environment& env,
                              const SdpVideoFormat& format) {
            return std::make_unique<test::DelayedEncoder>(env, kEncodeDelayMs);
          }) {
      extensions_.Register<TransmissionOffset>(kTimestampOffsetExtensionId);
    }

   private:
    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      RtpPacket rtp_packet(&extensions_);
      EXPECT_TRUE(rtp_packet.Parse(packet));

      int32_t toffset = 0;
      EXPECT_TRUE(rtp_packet.GetExtension<TransmissionOffset>(&toffset));
      EXPECT_FALSE(rtp_packet.HasExtension<AbsoluteSendTime>());
      EXPECT_GT(toffset, 0);
      observation_complete_.Set();

      return SEND_PACKET;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      send_config->encoder_settings.encoder_factory = &encoder_factory_;
      send_config->rtp.extensions.clear();
      send_config->rtp.extensions.push_back(RtpExtension(
          RtpExtension::kTimestampOffsetUri, kTimestampOffsetExtensionId));
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
    }

    test::FunctionVideoEncoderFactory encoder_factory_;
    RtpHeaderExtensionMap extensions_;
  } test;

  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsTransportWideSequenceNumbers) {
  static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
  class TransportWideSequenceNumberObserver : public test::SendTest {
   public:
    TransportWideSequenceNumberObserver()
        : SendTest(test::VideoTestConstants::kDefaultTimeout),
          encoder_factory_(
              [](const Environment& env, const SdpVideoFormat& format) {
                return std::make_unique<test::FakeEncoder>(env);
              }) {
      extensions_.Register<TransportSequenceNumber>(kExtensionId);
    }

   private:
    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      RtpPacket rtp_packet(&extensions_);
      EXPECT_TRUE(rtp_packet.Parse(packet));

      EXPECT_TRUE(rtp_packet.HasExtension<TransportSequenceNumber>());
      EXPECT_FALSE(rtp_packet.HasExtension<TransmissionOffset>());
      EXPECT_FALSE(rtp_packet.HasExtension<AbsoluteSendTime>());

      observation_complete_.Set();

      return SEND_PACKET;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      send_config->encoder_settings.encoder_factory = &encoder_factory_;
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
    }

    test::FunctionVideoEncoderFactory encoder_factory_;
    RtpHeaderExtensionMap extensions_;
  } test;

  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsVideoRotation) {
  class VideoRotationObserver : public test::SendTest {
   public:
    VideoRotationObserver()
        : SendTest(test::VideoTestConstants::kDefaultTimeout) {
      extensions_.Register<VideoOrientation>(kVideoRotationExtensionId);
    }

    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      RtpPacket rtp_packet(&extensions_);
      EXPECT_TRUE(rtp_packet.Parse(packet));
      // Only the last packet of the frame is required to have the extension.
      if (!rtp_packet.Marker())
        return SEND_PACKET;
      EXPECT_EQ(rtp_packet.GetExtension<VideoOrientation>(), kVideoRotation_90);
      observation_complete_.Set();
      return SEND_PACKET;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      send_config->rtp.extensions.clear();
      send_config->rtp.extensions.push_back(RtpExtension(
          RtpExtension::kVideoRotationUri, kVideoRotationExtensionId));
    }

    void OnFrameGeneratorCapturerCreated(
        test::FrameGeneratorCapturer* frame_generator_capturer) override {
      frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
    }

   private:
    RtpHeaderExtensionMap extensions_;
  } test;

  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsVideoContentType) {
  class VideoContentTypeObserver : public test::SendTest {
   public:
    VideoContentTypeObserver()
        : SendTest(test::VideoTestConstants::kDefaultTimeout),
          first_frame_sent_(false) {
      extensions_.Register<VideoContentTypeExtension>(
          kVideoContentTypeExtensionId);
    }

    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      RtpPacket rtp_packet(&extensions_);
      EXPECT_TRUE(rtp_packet.Parse(packet));
      // Only the last packet of the key-frame must have extension.
      if (!rtp_packet.Marker() || first_frame_sent_)
        return SEND_PACKET;
      // First marker bit seen means that the first frame is sent.
      first_frame_sent_ = true;
      VideoContentType type;
      EXPECT_TRUE(rtp_packet.GetExtension<VideoContentTypeExtension>(&type));
      EXPECT_TRUE(videocontenttypehelpers::IsScreenshare(type));
      observation_complete_.Set();
      return SEND_PACKET;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      send_config->rtp.extensions.clear();
      send_config->rtp.extensions.push_back(RtpExtension(
          RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
    }

   private:
    bool first_frame_sent_;
    RtpHeaderExtensionMap extensions_;
  } test;

  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsVideoTimingFrames) {
  class VideoTimingObserver : public test::SendTest {
   public:
    VideoTimingObserver()
        : SendTest(test::VideoTestConstants::kDefaultTimeout),
          first_frame_sent_(false) {
      extensions_.Register<VideoTimingExtension>(kVideoTimingExtensionId);
    }

    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      RtpPacket rtp_packet(&extensions_);
      EXPECT_TRUE(rtp_packet.Parse(packet));
      // Only the last packet of the frame must have extension.
      // Also don't check packets of the second frame if they happen to get
      // through before the test terminates.
      if (!rtp_packet.Marker() || first_frame_sent_)
        return SEND_PACKET;
      EXPECT_TRUE(rtp_packet.HasExtension<VideoTimingExtension>());
      observation_complete_.Set();
      first_frame_sent_ = true;
      return SEND_PACKET;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      send_config->rtp.extensions.clear();
      send_config->rtp.extensions.push_back(
          RtpExtension(RtpExtension::kVideoTimingUri, kVideoTimingExtensionId));
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait()) << "Timed out while waiting for timing frames.";
    }

   private:
    RtpHeaderExtensionMap extensions_;
    bool first_frame_sent_;
  } test;

  RunBaseTest(&test);
}

class FakeReceiveStatistics : public ReceiveStatisticsProvider {
 public:
  FakeReceiveStatistics(uint32_t send_ssrc,
                        uint32_t last_sequence_number,
                        uint32_t cumulative_lost,
                        uint8_t fraction_lost) {
    stat_.SetMediaSsrc(send_ssrc);
    stat_.SetExtHighestSeqNum(last_sequence_number);
    stat_.SetCumulativeLost(cumulative_lost);
    stat_.SetFractionLost(fraction_lost);
  }

  std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override {
    EXPECT_GE(max_blocks, 1u);
    return {stat_};
  }

 private:
  rtcp::ReportBlock stat_;
};

class UlpfecObserver : public test::EndToEndTest {
 public:
  UlpfecObserver(
      bool header_extensions_enabled,
      bool use_nack,
      bool expect_red,
      bool expect_ulpfec,
      const std::string& codec,
      VideoEncoderFactory* encoder_factory,
      const TimeDelta& timeout = test::VideoTestConstants::kDefaultTimeout)
      : EndToEndTest(timeout),
        encoder_factory_(encoder_factory),
        payload_name_(codec),
        use_nack_(use_nack),
        expect_red_(expect_red),
        expect_ulpfec_(expect_ulpfec),
        sent_media_(false),
        sent_ulpfec_(false),
        header_extensions_enabled_(header_extensions_enabled) {
    extensions_.Register<AbsoluteSendTime>(kAbsSendTimeExtensionId);
    extensions_.Register<TransportSequenceNumber>(
        kTransportSequenceNumberExtensionId);
  }

 private:
  Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
    RtpPacket rtp_packet(&extensions_);
    EXPECT_TRUE(rtp_packet.Parse(packet));

    int encapsulated_payload_type = -1;
    if (rtp_packet.PayloadType() == test::VideoTestConstants::kRedPayloadType) {
      EXPECT_TRUE(expect_red_);
      encapsulated_payload_type = rtp_packet.payload()[0];
      if (encapsulated_payload_type !=
          test::VideoTestConstants::kFakeVideoSendPayloadType) {
        EXPECT_EQ(test::VideoTestConstants::kUlpfecPayloadType,
                  encapsulated_payload_type);
      }
    } else {
      EXPECT_EQ(test::VideoTestConstants::kFakeVideoSendPayloadType,
                rtp_packet.PayloadType());
      if (rtp_packet.payload_size() > 0) {
        // Not padding-only, media received outside of RED.
        EXPECT_FALSE(expect_red_);
        sent_media_ = true;
      }
    }

    if (header_extensions_enabled_) {
      uint32_t abs_send_time;
      EXPECT_TRUE(rtp_packet.GetExtension<AbsoluteSendTime>(&abs_send_time));
      uint16_t transport_seq_num;
      EXPECT_TRUE(
          rtp_packet.GetExtension<TransportSequenceNumber>(&transport_seq_num));
      if (!first_packet_) {
        uint32_t kHalf24BitsSpace = 0xFFFFFF / 2;
        if (abs_send_time <= kHalf24BitsSpace &&
            prev_abs_send_time_ > kHalf24BitsSpace) {
          // 24 bits wrap.
          EXPECT_GT(prev_abs_send_time_, abs_send_time);
        } else {
          EXPECT_GE(abs_send_time, prev_abs_send_time_);
        }

        uint16_t seq_num_diff = transport_seq_num - prev_transport_seq_num_;
        EXPECT_EQ(1, seq_num_diff);
      }
      first_packet_ = false;
      prev_abs_send_time_ = abs_send_time;
      prev_transport_seq_num_ = transport_seq_num;
    }

    if (encapsulated_payload_type != -1) {
      if (encapsulated_payload_type ==
          test::VideoTestConstants::kUlpfecPayloadType) {
        EXPECT_TRUE(expect_ulpfec_);
        sent_ulpfec_ = true;
      } else {
        sent_media_ = true;
      }
    }

    if (sent_media_ && sent_ulpfec_) {
      observation_complete_.Set();
    }

    return SEND_PACKET;
  }

  BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
    // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
    // Configure some network delay.
    const int kNetworkDelayMs = 100;
    BuiltInNetworkBehaviorConfig config;
    config.loss_percent = 5;
    config.queue_delay_ms = kNetworkDelayMs;
    return config;
  }

  void ModifyVideoConfigs(
      VideoSendStream::Config* send_config,
      std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
      VideoEncoderConfig* encoder_config) override {
    if (use_nack_) {
      send_config->rtp.nack.rtp_history_ms =
          (*receive_configs)[0].rtp.nack.rtp_history_ms =
              test::VideoTestConstants::kNackRtpHistoryMs;
    }
    send_config->encoder_settings.encoder_factory = encoder_factory_;
    send_config->rtp.payload_name = payload_name_;
    send_config->rtp.ulpfec.red_payload_type =
        test::VideoTestConstants::kRedPayloadType;
    send_config->rtp.ulpfec.ulpfec_payload_type =
        test::VideoTestConstants::kUlpfecPayloadType;
    if (!header_extensions_enabled_) {
      send_config->rtp.extensions.clear();
    } else {
      send_config->rtp.extensions.push_back(
          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
    }
    encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
    (*receive_configs)[0].rtp.red_payload_type =
        send_config->rtp.ulpfec.red_payload_type;
    (*receive_configs)[0].rtp.ulpfec_payload_type =
        send_config->rtp.ulpfec.ulpfec_payload_type;
  }

  void PerformTest() override {
    EXPECT_EQ(expect_ulpfec_, Wait())
        << "Timed out waiting for ULPFEC and/or media packets.";
  }

  VideoEncoderFactory* encoder_factory_;
  RtpHeaderExtensionMap extensions_;
  const std::string payload_name_;
  const bool use_nack_;
  const bool expect_red_;
  const bool expect_ulpfec_;
  bool sent_media_;
  bool sent_ulpfec_;
  const bool header_extensions_enabled_;
  bool first_packet_ = true;
  uint32_t prev_abs_send_time_ = 0;
  uint16_t prev_transport_seq_num_ = 0;
};

TEST_F(VideoSendStreamTest, SupportsUlpfecWithExtensions) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp8Encoder(env);
      });
  UlpfecObserver test(true, false, true, true, "VP8", &encoder_factory);
  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsUlpfecWithoutExtensions) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp8Encoder(env);
      });
  UlpfecObserver test(false, false, true, true, "VP8", &encoder_factory);
  RunBaseTest(&test);
}

class VideoSendStreamWithoutUlpfecTest : public test::CallTest {
 protected:
  VideoSendStreamWithoutUlpfecTest()
      : field_trial_(field_trials_, "WebRTC-DisableUlpFecExperiment/Enabled/") {
  }

  test::ScopedKeyValueConfig field_trial_;
};

TEST_F(VideoSendStreamWithoutUlpfecTest, NoUlpfecIfDisabledThroughFieldTrial) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp8Encoder(env);
      });
  UlpfecObserver test(false, false, false, false, "VP8", &encoder_factory,
                      kReducedTimeout);
  RunBaseTest(&test);
}

// The FEC scheme used is not efficient for H264, so we should not use RED/FEC
// since we'll still have to re-request FEC packets, effectively wasting
// bandwidth since the receiver has to wait for FEC retransmissions to determine
// that the received state is actually decodable.
TEST_F(VideoSendStreamTest, DoesNotUtilizeUlpfecForH264WithNackEnabled) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return std::make_unique<test::FakeH264Encoder>(env);
      });
  UlpfecObserver test(false, true, false, false, "H264", &encoder_factory,
                      kReducedTimeout);
  RunBaseTest(&test);
}

// Without retransmissions FEC for H264 is fine.
TEST_F(VideoSendStreamTest, DoesUtilizeUlpfecForH264WithoutNackEnabled) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return std::make_unique<test::FakeH264Encoder>(env);
      });
  UlpfecObserver test(false, false, true, true, "H264", &encoder_factory);
  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, DoesUtilizeUlpfecForVp8WithNackEnabled) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp8Encoder(env);
      });
  UlpfecObserver test(false, true, true, true, "VP8", &encoder_factory);
  RunBaseTest(&test);
}

#if defined(RTC_ENABLE_VP9)
TEST_F(VideoSendStreamTest, DoesUtilizeUlpfecForVp9WithNackEnabled) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp9Encoder(env);
      });
  // Use kLongTimeout timeout because the test is flaky with kDefaultTimeout.
  UlpfecObserver test(false, true, true, true, "VP9", &encoder_factory,
                      test::VideoTestConstants::kLongTimeout);
  RunBaseTest(&test);
}
#endif  // defined(RTC_ENABLE_VP9)

TEST_F(VideoSendStreamTest, SupportsUlpfecWithMultithreadedH264) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [&](const Environment& env, const SdpVideoFormat& format) {
        return std::make_unique<test::MultithreadedFakeH264Encoder>(env);
      });
  UlpfecObserver test(false, false, true, true, "H264", &encoder_factory);
  RunBaseTest(&test);
}

// TODO(brandtr): Move these FlexFEC tests when we have created
// FlexfecSendStream.
class FlexfecObserver : public test::EndToEndTest {
 public:
  FlexfecObserver(bool header_extensions_enabled,
                  bool use_nack,
                  const std::string& codec,
                  VideoEncoderFactory* encoder_factory,
                  size_t num_video_streams)
      : EndToEndTest(test::VideoTestConstants::kDefaultTimeout),
        encoder_factory_(encoder_factory),
        payload_name_(codec),
        use_nack_(use_nack),
        sent_media_(false),
        sent_flexfec_(false),
        header_extensions_enabled_(header_extensions_enabled),
        num_video_streams_(num_video_streams) {
    extensions_.Register<AbsoluteSendTime>(kAbsSendTimeExtensionId);
    extensions_.Register<TransmissionOffset>(kTimestampOffsetExtensionId);
    extensions_.Register<TransportSequenceNumber>(
        kTransportSequenceNumberExtensionId);
  }

  size_t GetNumFlexfecStreams() const override { return 1; }
  size_t GetNumVideoStreams() const override { return num_video_streams_; }

 private:
  Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
    RtpPacket rtp_packet(&extensions_);
    EXPECT_TRUE(rtp_packet.Parse(packet));

    if (rtp_packet.PayloadType() ==
        test::VideoTestConstants::kFlexfecPayloadType) {
      EXPECT_EQ(test::VideoTestConstants::kFlexfecSendSsrc, rtp_packet.Ssrc());
      sent_flexfec_ = true;
    } else {
      EXPECT_EQ(test::VideoTestConstants::kFakeVideoSendPayloadType,
                rtp_packet.PayloadType());
      EXPECT_THAT(
          ::testing::make_tuple(test::VideoTestConstants::kVideoSendSsrcs,
                                num_video_streams_),
          ::testing::Contains(rtp_packet.Ssrc()));
      sent_media_ = true;
    }

    if (header_extensions_enabled_) {
      EXPECT_TRUE(rtp_packet.HasExtension<AbsoluteSendTime>());
      EXPECT_TRUE(rtp_packet.HasExtension<TransmissionOffset>());
      EXPECT_TRUE(rtp_packet.HasExtension<TransportSequenceNumber>());
    }

    if (sent_media_ && sent_flexfec_) {
      observation_complete_.Set();
    }

    return SEND_PACKET;
  }

  BuiltInNetworkBehaviorConfig GetSendTransportConfig() const {
    // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
    // Therefore we need some network delay.
    const int kNetworkDelayMs = 100;
    BuiltInNetworkBehaviorConfig config;
    config.loss_percent = 5;
    config.queue_delay_ms = kNetworkDelayMs;
    return config;
  }

  BuiltInNetworkBehaviorConfig GetReceiveTransportConfig() const {
    // We need the RTT to be >200 ms to send FEC and the network delay for the
    // send transport is 100 ms, so add 100 ms (but no loss) on the return link.
    BuiltInNetworkBehaviorConfig config;
    config.loss_percent = 0;
    config.queue_delay_ms = 100;
    return config;
  }

  void ModifyVideoConfigs(
      VideoSendStream::Config* send_config,
      std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
      VideoEncoderConfig* encoder_config) override {
    if (use_nack_) {
      send_config->rtp.nack.rtp_history_ms =
          (*receive_configs)[0].rtp.nack.rtp_history_ms =
              test::VideoTestConstants::kNackRtpHistoryMs;
    }
    send_config->encoder_settings.encoder_factory = encoder_factory_;
    send_config->rtp.payload_name = payload_name_;
    if (header_extensions_enabled_) {
      send_config->rtp.extensions.push_back(
          RtpExtension(RtpExtension::kAbsSendTimeUri, kAbsSendTimeExtensionId));
      send_config->rtp.extensions.push_back(RtpExtension(
          RtpExtension::kTimestampOffsetUri, kTimestampOffsetExtensionId));
    } else {
      send_config->rtp.extensions.clear();
    }
    encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
  }

  void PerformTest() override {
    EXPECT_TRUE(Wait())
        << "Timed out waiting for FlexFEC and/or media packets.";
  }

  VideoEncoderFactory* encoder_factory_;
  RtpHeaderExtensionMap extensions_;
  const std::string payload_name_;
  const bool use_nack_;
  bool sent_media_;
  bool sent_flexfec_;
  const bool header_extensions_enabled_;
  const size_t num_video_streams_;
};

TEST_F(VideoSendStreamTest, SupportsFlexfecVp8) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp8Encoder(env);
      });
  FlexfecObserver test(false, false, "VP8", &encoder_factory, 1);
  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsFlexfecSimulcastVp8) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp8Encoder(env);
      });
  FlexfecObserver test(false, false, "VP8", &encoder_factory, 2);
  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsFlexfecWithNackVp8) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp8Encoder(env);
      });
  FlexfecObserver test(false, true, "VP8", &encoder_factory, 1);
  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsFlexfecWithRtpExtensionsVp8) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp8Encoder(env);
      });
  FlexfecObserver test(true, false, "VP8", &encoder_factory, 1);
  RunBaseTest(&test);
}

#if defined(RTC_ENABLE_VP9)
TEST_F(VideoSendStreamTest, SupportsFlexfecVp9) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp9Encoder(env);
      });
  FlexfecObserver test(false, false, "VP9", &encoder_factory, 1);
  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsFlexfecWithNackVp9) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return CreateVp9Encoder(env);
      });
  FlexfecObserver test(false, true, "VP9", &encoder_factory, 1);
  RunBaseTest(&test);
}
#endif  // defined(RTC_ENABLE_VP9)

TEST_F(VideoSendStreamTest, SupportsFlexfecH264) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return std::make_unique<test::FakeH264Encoder>(env);
      });
  FlexfecObserver test(false, false, "H264", &encoder_factory, 1);
  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsFlexfecWithNackH264) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [](const Environment& env, const SdpVideoFormat& format) {
        return std::make_unique<test::FakeH264Encoder>(env);
      });
  FlexfecObserver test(false, true, "H264", &encoder_factory, 1);
  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, SupportsFlexfecWithMultithreadedH264) {
  test::FunctionVideoEncoderFactory encoder_factory(
      [&](const Environment& env, const SdpVideoFormat& format) {
        return std::make_unique<test::MultithreadedFakeH264Encoder>(env);
      });

  FlexfecObserver test(false, false, "H264", &encoder_factory, 1);
  RunBaseTest(&test);
}

void VideoSendStreamTest::TestNackRetransmission(
    uint32_t retransmit_ssrc,
    uint8_t retransmit_payload_type) {
  class NackObserver : public test::SendTest {
   public:
    explicit NackObserver(uint32_t retransmit_ssrc,
                          uint8_t retransmit_payload_type)
        : SendTest(test::VideoTestConstants::kDefaultTimeout),
          send_count_(0),
          retransmit_count_(0),
          retransmit_ssrc_(retransmit_ssrc),
          retransmit_payload_type_(retransmit_payload_type) {}

   private:
    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      RtpPacket rtp_packet;
      EXPECT_TRUE(rtp_packet.Parse(packet));

      // NACK packets two times at some arbitrary points.
      const int kNackedPacketsAtOnceCount = 3;
      const int kRetransmitTarget = kNackedPacketsAtOnceCount * 2;

      // Skip padding packets because they will never be retransmitted.
      if (rtp_packet.payload_size() == 0) {
        return SEND_PACKET;
      }

      ++send_count_;

      // NACK packets at arbitrary points.
      if (send_count_ % 25 == 0) {
        RTCPSender::Configuration config;
        config.outgoing_transport = transport_adapter_.get();
        config.rtcp_report_interval = TimeDelta::Millis(kRtcpIntervalMs);
        config.local_media_ssrc =
            test::VideoTestConstants::kReceiverLocalVideoSsrc;
        RTCPSender rtcp_sender(env_, config);

        rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
        rtcp_sender.SetRemoteSSRC(test::VideoTestConstants::kVideoSendSsrcs[0]);

        RTCPSender::FeedbackState feedback_state;
        uint16_t nack_sequence_numbers[kNackedPacketsAtOnceCount];
        int nack_count = 0;
        for (uint16_t sequence_number :
             sequence_numbers_pending_retransmission_) {
          if (nack_count < kNackedPacketsAtOnceCount) {
            nack_sequence_numbers[nack_count++] = sequence_number;
          } else {
            break;
          }
        }

        EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpNack, nack_count,
                                          nack_sequence_numbers));
      }

      uint16_t sequence_number = rtp_packet.SequenceNumber();
      if (rtp_packet.Ssrc() == retransmit_ssrc_ &&
          retransmit_ssrc_ != test::VideoTestConstants::kVideoSendSsrcs[0]) {
        // Not kVideoSendSsrcs[0], assume correct RTX packet. Extract sequence
        // number.
        const uint8_t* rtx_header = rtp_packet.payload().data();
        sequence_number = (rtx_header[0] << 8) + rtx_header[1];
      }

      auto it = sequence_numbers_pending_retransmission_.find(sequence_number);
      if (it == sequence_numbers_pending_retransmission_.end()) {
        // Not currently pending retransmission. Add it to retransmission queue
        // if media and limit not reached.
        if (rtp_packet.Ssrc() == test::VideoTestConstants::kVideoSendSsrcs[0] &&
            rtp_packet.payload_size() > 0 &&
            retransmit_count_ +
                    sequence_numbers_pending_retransmission_.size() <
                kRetransmitTarget) {
          sequence_numbers_pending_retransmission_.insert(sequence_number);
          return DROP_PACKET;
        }
      } else {
        // Packet is a retransmission, remove it from queue and check if done.
        sequence_numbers_pending_retransmission_.erase(it);
        if (++retransmit_count_ == kRetransmitTarget) {
          EXPECT_EQ(retransmit_ssrc_, rtp_packet.Ssrc());
          EXPECT_EQ(retransmit_payload_type_, rtp_packet.PayloadType());
          observation_complete_.Set();
        }
      }

      return SEND_PACKET;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      transport_adapter_.reset(
          new internal::TransportAdapter(send_config->send_transport));
      transport_adapter_->Enable();
      send_config->rtp.nack.rtp_history_ms =
          test::VideoTestConstants::kNackRtpHistoryMs;
      send_config->rtp.rtx.payload_type = retransmit_payload_type_;
      if (retransmit_ssrc_ != test::VideoTestConstants::kVideoSendSsrcs[0])
        send_config->rtp.rtx.ssrcs.push_back(retransmit_ssrc_);
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait()) << "Timed out while waiting for NACK retransmission.";
    }

    const Environment env_ = CreateEnvironment();
    std::unique_ptr<internal::TransportAdapter> transport_adapter_;
    int send_count_;
    int retransmit_count_;
    const uint32_t retransmit_ssrc_;
    const uint8_t retransmit_payload_type_;
    std::set<uint16_t> sequence_numbers_pending_retransmission_;
  } test(retransmit_ssrc, retransmit_payload_type);

  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, RetransmitsNack) {
  // Normal NACKs should use the send SSRC.
  TestNackRetransmission(test::VideoTestConstants::kVideoSendSsrcs[0],
                         test::VideoTestConstants::kFakeVideoSendPayloadType);
}

TEST_F(VideoSendStreamTest, RetransmitsNackOverRtx) {
  // NACKs over RTX should use a separate SSRC.
  TestNackRetransmission(test::VideoTestConstants::kSendRtxSsrcs[0],
                         test::VideoTestConstants::kSendRtxPayloadType);
}

void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
                                                      bool with_fec) {
  // Use a fake encoder to output a frame of every size in the range [90, 290],
  // for each size making sure that the exact number of payload bytes received
  // is correct and that packets are fragmented to respect max packet size.
  static const size_t kMaxPacketSize = 128;
  static const size_t start = 90;
  static const size_t stop = 290;

  // Observer that verifies that the expected number of packets and bytes
  // arrive for each frame size, from start_size to stop_size.
  class FrameFragmentationTest : public test::SendTest {
   public:
    FrameFragmentationTest(size_t max_packet_size,
                           size_t start_size,
                           size_t stop_size,
                           bool test_generic_packetization,
                           bool use_fec)
        : SendTest(test::VideoTestConstants::kLongTimeout),
          encoder_(stop),
          encoder_factory_(&encoder_),
          max_packet_size_(max_packet_size),
          stop_size_(stop_size),
          test_generic_packetization_(test_generic_packetization),
          use_fec_(use_fec),
          packet_count_(0),
          packets_lost_(0),
          last_packet_count_(0),
          last_packets_lost_(0),
          accumulated_size_(0),
          accumulated_payload_(0),
          fec_packet_received_(false),
          current_size_rtp_(start_size),
          current_size_frame_(static_cast<int>(start_size)) {
      // Fragmentation required, this test doesn't make sense without it.
      encoder_.SetFrameSize(start_size);
      RTC_DCHECK_GT(stop_size, max_packet_size);
      if (!test_generic_packetization_)
        encoder_.SetCodecType(kVideoCodecVP8);
    }

   private:
    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      size_t length = packet.size();
      RtpPacket rtp_packet;
      EXPECT_TRUE(rtp_packet.Parse(packet));

      EXPECT_LE(length, max_packet_size_);

      if (use_fec_ && rtp_packet.payload_size() > 0) {
        uint8_t payload_type = rtp_packet.payload()[0];
        bool is_fec =
            rtp_packet.PayloadType() ==
                test::VideoTestConstants::kRedPayloadType &&
            payload_type == test::VideoTestConstants::kUlpfecPayloadType;
        if (is_fec) {
          fec_packet_received_ = true;
          return SEND_PACKET;
        }
      }

      accumulated_size_ += length;

      if (use_fec_)
        TriggerLossReport(rtp_packet);

      if (test_generic_packetization_) {
        size_t overhead = rtp_packet.headers_size() + rtp_packet.padding_size();
        // Only remove payload header and RED header if the packet actually
        // contains payload.
        if (length > overhead) {
          overhead += (1 /* Generic header */);
          if (use_fec_)
            overhead += 1;  // RED for FEC header.
        }
        EXPECT_GE(length, overhead);
        accumulated_payload_ += length - overhead;
      }

      // Marker bit set indicates last packet of a frame.
      if (rtp_packet.Marker()) {
        if (use_fec_ && accumulated_payload_ == current_size_rtp_ - 1) {
          // With FEC enabled, frame size is incremented asynchronously, so
          // "old" frames one byte too small may arrive. Accept, but don't
          // increase expected frame size.
          accumulated_size_ = 0;
          accumulated_payload_ = 0;
          return SEND_PACKET;
        }

        EXPECT_GE(accumulated_size_, current_size_rtp_);
        if (test_generic_packetization_) {
          EXPECT_EQ(current_size_rtp_, accumulated_payload_);
        }

        // Last packet of frame; reset counters.
        accumulated_size_ = 0;
        accumulated_payload_ = 0;
        if (current_size_rtp_ == stop_size_) {
          // Done! (Don't increase size again, might arrive more @ stop_size).
          observation_complete_.Set();
        } else {
          // Increase next expected frame size. If testing with FEC, make sure
          // a FEC packet has been received for this frame size before
          // proceeding, to make sure that redundancy packets don't exceed
          // size limit.
          if (!use_fec_) {
            ++current_size_rtp_;
          } else if (fec_packet_received_) {
            fec_packet_received_ = false;
            ++current_size_rtp_;

            MutexLock lock(&mutex_);
            ++current_size_frame_;
          }
        }
      }

      return SEND_PACKET;
    }

    void TriggerLossReport(const RtpPacket& rtp_packet) {
      // Send lossy receive reports to trigger FEC enabling.
      const int kLossPercent = 5;
      if (++packet_count_ % (100 / kLossPercent) == 0) {
        packets_lost_++;
        int loss_delta = packets_lost_ - last_packets_lost_;
        int packets_delta = packet_count_ - last_packet_count_;
        last_packet_count_ = packet_count_;
        last_packets_lost_ = packets_lost_;
        uint8_t loss_ratio =
            static_cast<uint8_t>(loss_delta * 255 / packets_delta);
        FakeReceiveStatistics lossy_receive_stats(
            test::VideoTestConstants::kVideoSendSsrcs[0],
            rtp_packet.SequenceNumber(),
            packets_lost_,  // Cumulative lost.
            loss_ratio);    // Loss percent.
        RTCPSender::Configuration config;
        config.receive_statistics = &lossy_receive_stats;
        config.outgoing_transport = transport_adapter_.get();
        config.rtcp_report_interval = TimeDelta::Millis(kRtcpIntervalMs);
        config.local_media_ssrc = test::VideoTestConstants::kVideoSendSsrcs[0];
        RTCPSender rtcp_sender(env_, config);

        rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
        rtcp_sender.SetRemoteSSRC(test::VideoTestConstants::kVideoSendSsrcs[0]);

        RTCPSender::FeedbackState feedback_state;

        EXPECT_EQ(0, rtcp_sender.SendRTCP(feedback_state, kRtcpRr));
      }
    }

    void UpdateConfiguration() {
      MutexLock lock(&mutex_);
      // Increase frame size for next encoded frame, in the context of the
      // encoder thread.
      if (!use_fec_ && current_size_frame_ < static_cast<int32_t>(stop_size_)) {
        ++current_size_frame_;
      }
      encoder_.SetFrameSize(static_cast<size_t>(current_size_frame_));
    }
    void ModifySenderBitrateConfig(
        BitrateConstraints* bitrate_config) override {
      const int kMinBitrateBps = 300000;
      bitrate_config->min_bitrate_bps = kMinBitrateBps;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      transport_adapter_.reset(
          new internal::TransportAdapter(send_config->send_transport));
      transport_adapter_->Enable();
      if (use_fec_) {
        send_config->rtp.ulpfec.red_payload_type =
            test::VideoTestConstants::kRedPayloadType;
        send_config->rtp.ulpfec.ulpfec_payload_type =
            test::VideoTestConstants::kUlpfecPayloadType;
      }

      if (!test_generic_packetization_)
        send_config->rtp.payload_name = "VP8";

      send_config->encoder_settings.encoder_factory = &encoder_factory_;
      send_config->rtp.max_packet_size = kMaxPacketSize;
      encoder_.RegisterPostEncodeCallback([this]() { UpdateConfiguration(); });

      // Make sure there is at least one extension header, to make the RTP
      // header larger than the base length of 12 bytes.
      EXPECT_FALSE(send_config->rtp.extensions.empty());

      // Setup screen content disables frame dropping which makes this easier.
      EXPECT_EQ(1u, encoder_config->simulcast_layers.size());
      encoder_config->simulcast_layers[0].num_temporal_layers = 2;
      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait()) << "Timed out while observing incoming RTP packets.";
    }

    const Environment env_ = CreateEnvironment();
    std::unique_ptr<internal::TransportAdapter> transport_adapter_;
    test::ConfigurableFrameSizeEncoder encoder_;
    test::VideoEncoderProxyFactory encoder_factory_;

    const size_t max_packet_size_;
    const size_t stop_size_;
    const bool test_generic_packetization_;
    const bool use_fec_;

    uint32_t packet_count_;
    uint32_t packets_lost_;
    uint32_t last_packet_count_;
    uint32_t last_packets_lost_;
    size_t accumulated_size_;
    size_t accumulated_payload_;
    bool fec_packet_received_;

    size_t current_size_rtp_;
    Mutex mutex_;
    int current_size_frame_ RTC_GUARDED_BY(mutex_);
  };

  // Don't auto increment if FEC is used; continue sending frame size until
  // a FEC packet has been received.
  FrameFragmentationTest test(kMaxPacketSize, start, stop, format == kGeneric,
                              with_fec);

  RunBaseTest(&test);
}

// TODO(sprang): Is there any way of speeding up these tests?
TEST_F(VideoSendStreamTest, FragmentsGenericAccordingToMaxPacketSize) {
  TestPacketFragmentationSize(kGeneric, false);
}

TEST_F(VideoSendStreamTest, FragmentsGenericAccordingToMaxPacketSizeWithFec) {
  TestPacketFragmentationSize(kGeneric, true);
}

TEST_F(VideoSendStreamTest, FragmentsVp8AccordingToMaxPacketSize) {
  TestPacketFragmentationSize(kVP8, false);
}

TEST_F(VideoSendStreamTest, FragmentsVp8AccordingToMaxPacketSizeWithFec) {
  TestPacketFragmentationSize(kVP8, true);
}

// This test that padding stops being send after a while if the Camera stops
// producing video frames and that padding resumes if the camera restarts.
TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
  class NoPaddingWhenVideoIsMuted : public test::SendTest {
   public:
    NoPaddingWhenVideoIsMuted()
        : SendTest(test::VideoTestConstants::kDefaultTimeout),
          clock_(Clock::GetRealTimeClock()),
          capturer_(nullptr) {}

   private:
    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      MutexLock lock(&mutex_);
      last_packet_time_ms_ = clock_->TimeInMilliseconds();

      RtpPacket rtp_packet;
      rtp_packet.Parse(packet);
      const bool only_padding = rtp_packet.payload_size() == 0;

      if (test_state_ == kBeforeStopCapture) {
        // Packets are flowing, stop camera.
        capturer_->Stop();
        test_state_ = kWaitingForPadding;
      } else if (test_state_ == kWaitingForPadding && only_padding) {
        // We're still getting padding, after stopping camera.
        test_state_ = kWaitingForNoPackets;
      } else if (test_state_ == kWaitingForMediaAfterCameraRestart &&
                 !only_padding) {
        // Media packets are flowing again, stop camera a second time.
        capturer_->Stop();
        test_state_ = kWaitingForPaddingAfterCameraStopsAgain;
      } else if (test_state_ == kWaitingForPaddingAfterCameraStopsAgain &&
                 only_padding) {
        // Padding is still flowing, test ok.
        observation_complete_.Set();
      }
      return SEND_PACKET;
    }

    Action OnSendRtcp(rtc::ArrayView<const uint8_t> packet) override {
      MutexLock lock(&mutex_);
      const int kNoPacketsThresholdMs = 2000;
      if (test_state_ == kWaitingForNoPackets &&
          (last_packet_time_ms_ &&
           clock_->TimeInMilliseconds() - last_packet_time_ms_.value() >
               kNoPacketsThresholdMs)) {
        // No packets seen for `kNoPacketsThresholdMs`, restart camera.
        capturer_->Start();
        test_state_ = kWaitingForMediaAfterCameraRestart;
      }
      return SEND_PACKET;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      // Make sure padding is sent if encoder is not producing media.
      encoder_config->min_transmit_bitrate_bps = 50000;
    }

    void OnFrameGeneratorCapturerCreated(
        test::FrameGeneratorCapturer* frame_generator_capturer) override {
      MutexLock lock(&mutex_);
      capturer_ = frame_generator_capturer;
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait())
          << "Timed out while waiting for RTP packets to stop being sent.";
    }

    enum TestState {
      kBeforeStopCapture,
      kWaitingForPadding,
      kWaitingForNoPackets,
      kWaitingForMediaAfterCameraRestart,
      kWaitingForPaddingAfterCameraStopsAgain
    };

    TestState test_state_ = kBeforeStopCapture;
    Clock* const clock_;
    Mutex mutex_;
    std::optional<int64_t> last_packet_time_ms_ RTC_GUARDED_BY(mutex_);
    test::FrameGeneratorCapturer* capturer_ RTC_GUARDED_BY(mutex_);
  } test;

  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, PaddingIsPrimarilyRetransmissions) {
  const int kCapacityKbps = 10000;  // 10 Mbps
  class PaddingIsPrimarilyRetransmissions : public test::EndToEndTest {
   public:
    PaddingIsPrimarilyRetransmissions()
        : EndToEndTest(test::VideoTestConstants::kDefaultTimeout),
          clock_(Clock::GetRealTimeClock()),
          padding_length_(0),
          total_length_(0),
          call_(nullptr) {}

   private:
    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
      call_ = sender_call;
    }

    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      MutexLock lock(&mutex_);

      RtpPacket rtp_packet;
      rtp_packet.Parse(packet);
      padding_length_ += rtp_packet.padding_size();
      total_length_ += packet.size();
      return SEND_PACKET;
    }

    BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
      const int kNetworkDelayMs = 50;
      BuiltInNetworkBehaviorConfig config;
      config.loss_percent = 10;
      config.link_capacity = DataRate::KilobitsPerSec(kCapacityKbps);
      config.queue_delay_ms = kNetworkDelayMs;
      return config;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      // Turn on RTX.
      send_config->rtp.rtx.payload_type =
          test::VideoTestConstants::kFakeVideoSendPayloadType;
      send_config->rtp.rtx.ssrcs.push_back(
          test::VideoTestConstants::kSendRtxSsrcs[0]);
    }

    void PerformTest() override {
      // TODO(isheriff): Some platforms do not ramp up as expected to full
      // capacity due to packet scheduling delays. Fix that before getting
      // rid of this.
      SleepMs(5000);
      {
        MutexLock lock(&mutex_);
        // Expect padding to be a small percentage of total bytes sent.
        EXPECT_LT(padding_length_, .1 * total_length_);
      }
    }

    Mutex mutex_;
    Clock* const clock_;
    size_t padding_length_ RTC_GUARDED_BY(mutex_);
    size_t total_length_ RTC_GUARDED_BY(mutex_);
    Call* call_;
  } test;

  RunBaseTest(&test);
}

// This test first observes "high" bitrate use at which point it sends a REMB to
// indicate that it should be lowered significantly. The test then observes that
// the bitrate observed is sinking well below the min-transmit-bitrate threshold
// to verify that the min-transmit bitrate respects incoming REMB.
//
// Note that the test starts at "high" bitrate and does not ramp up to "higher"
// bitrate since no receiver block or remb is sent in the initial phase.
TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
  static const int kMinTransmitBitrateBps = 400000;
  static const int kHighBitrateBps = 150000;
  static const int kRembBitrateBps = 80000;
  static const int kRembRespectedBitrateBps = 100000;
  class BitrateObserver : public test::SendTest {
   public:
    explicit BitrateObserver(TaskQueueBase* task_queue)
        : SendTest(test::VideoTestConstants::kDefaultTimeout),
          task_queue_(task_queue),
          retranmission_rate_limiter_(Clock::GetRealTimeClock(), 1000),
          stream_(nullptr),
          bitrate_capped_(false),
          task_safety_flag_(PendingTaskSafetyFlag::CreateDetached()) {}

   private:
    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      if (IsRtcpPacket(packet))
        return DROP_PACKET;

      RtpPacket rtp_packet;
      RTC_CHECK(rtp_packet.Parse(packet));
      const uint32_t ssrc = rtp_packet.Ssrc();
      RTC_DCHECK(stream_);

      task_queue_->PostTask(SafeTask(task_safety_flag_, [this, ssrc]() {
        VideoSendStream::Stats stats = stream_->GetStats();
        if (!stats.substreams.empty()) {
          EXPECT_EQ(1u, stats.substreams.size());
          int total_bitrate_bps =
              stats.substreams.begin()->second.total_bitrate_bps;
          test::GetGlobalMetricsLogger()->LogSingleValueMetric(
              "bitrate_stats_min_transmit_bitrate_low_remb", "bitrate_bps",
              static_cast<size_t>(total_bitrate_bps) / 1000.0,
              test::Unit::kKilobitsPerSecond,
              test::ImprovementDirection::kNeitherIsBetter);
          if (total_bitrate_bps > kHighBitrateBps) {
            rtp_rtcp_->SetRemb(kRembBitrateBps, {ssrc});
            bitrate_capped_ = true;
          } else if (bitrate_capped_ &&
                     total_bitrate_bps < kRembRespectedBitrateBps) {
            observation_complete_.Set();
          }
        }
      }));

      // Packets don't have to be delivered since the test is the receiver.
      return DROP_PACKET;
    }

    void OnVideoStreamsCreated(VideoSendStream* send_stream,
                               const std::vector<VideoReceiveStreamInterface*>&
                                   receive_streams) override {
      stream_ = send_stream;
      RtpRtcpInterface::Configuration config;
      config.outgoing_transport = feedback_transport_.get();
      config.retransmission_rate_limiter = &retranmission_rate_limiter_;
      rtp_rtcp_ = std::make_unique<ModuleRtpRtcpImpl2>(env_, config);
      rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      feedback_transport_.reset(
          new internal::TransportAdapter(send_config->send_transport));
      feedback_transport_->Enable();
      encoder_config->min_transmit_bitrate_bps = kMinTransmitBitrateBps;
    }

    void OnStreamsStopped() override {
      task_safety_flag_->SetNotAlive();
      rtp_rtcp_.reset();
    }

    void PerformTest() override {
      EXPECT_TRUE(Wait())
          << "Timeout while waiting for low bitrate stats after REMB.";
    }

    TaskQueueBase* const task_queue_;
    const Environment env_ = CreateEnvironment();
    std::unique_ptr<ModuleRtpRtcpImpl2> rtp_rtcp_;
    std::unique_ptr<internal::TransportAdapter> feedback_transport_;
    RateLimiter retranmission_rate_limiter_;
    VideoSendStream* stream_;
    bool bitrate_capped_;
    rtc::scoped_refptr<PendingTaskSafetyFlag> task_safety_flag_;
  } test(task_queue());

  RunBaseTest(&test);
}

TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
  static const int kStartBitrateBps = 300000;
  static const int kNewMaxBitrateBps = 1234567;
  static const uint8_t kExtensionId = kTransportSequenceNumberExtensionId;
  class ChangingNetworkRouteTest : public test::EndToEndTest {
   public:
    explicit ChangingNetworkRouteTest(TaskQueueBase* task_queue)
        : EndToEndTest(test::VideoTestConstants::kDefaultTimeout),
          task_queue_(task_queue),
          call_(nullptr) {
      module_process_thread_.Detach();
      task_queue_thread_.Detach();
      extensions_.Register<TransportSequenceNumber>(kExtensionId);
    }

    ~ChangingNetworkRouteTest() {
      // Block until all already posted tasks run to avoid 'use after free'
      // when such task accesses `this`.
      SendTask(task_queue_, [] {});
    }

    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
      RTC_DCHECK_RUN_ON(&task_queue_thread_);
      RTC_DCHECK(!call_);
      call_ = sender_call;
    }

    void ModifyVideoConfigs(
        VideoSendStream::Config* send_config,
        std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
        VideoEncoderConfig* encoder_config) override {
      RTC_DCHECK_RUN_ON(&task_queue_thread_);
      send_config->rtp.extensions.clear();
      send_config->rtp.extensions.push_back(RtpExtension(
          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
    }

    void ModifyAudioConfigs(AudioSendStream::Config* send_config,
                            std::vector<AudioReceiveStreamInterface::Config>*
                                receive_configs) override {
      RTC_DCHECK_RUN_ON(&task_queue_thread_);
      send_config->rtp.extensions.clear();
      send_config->rtp.extensions.push_back(RtpExtension(
          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
    }

    Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
      RTC_DCHECK_RUN_ON(&module_process_thread_);
      task_queue_->PostTask([this]() {
        RTC_DCHECK_RUN_ON(&task_queue_thread_);
        if (!call_)
          return;
        Call::Stats stats = call_->GetStats();
        if (stats.send_bandwidth_bps > kStartBitrateBps)
          observation_complete_.Set();
      });
      return SEND_PACKET;
    }

    void OnStreamsStopped() override {
      RTC_DCHECK_RUN_ON(&task_queue_thread_);
      call_ = nullptr;
    }

    void PerformTest() override {
      rtc::NetworkRoute new_route;
      new_route.connected = true;
      new_route.local = rtc::RouteEndpoint::CreateWithNetworkId(10);
      new_route.remote = rtc::RouteEndpoint::CreateWithNetworkId(20);
      BitrateConstraints bitrate_config;

      SendTask(task_queue_, [this, &new_route, &bitrate_config]() {
        RTC_DCHECK_RUN_ON(&task_queue_thread_);
        call_->GetTransportControllerSend()->OnNetworkRouteChanged("transport",
                                                                   new_route);
        bitrate_config.start_bitrate_bps = kStartBitrateBps;
        call_->GetTransportControllerSend()->SetSdpBitrateParameters(
            bitrate_config);
      });

      EXPECT_TRUE(Wait())
          << "Timed out while waiting for start bitrate to be exceeded.";

      SendTask(task_queue_, [this, &new_route, &bitrate_config]() {
        RTC_DCHECK_RUN_ON(&task_queue_thread_);
        bitrate_config.start_bitrate_bps = -1;
        bitrate_config.max_bitrate_bps = kNewMaxBitrateBps;
        call_->GetTransportControllerSend()->SetSdpBitrateParameters(
            bitrate_config);
        // TODO(holmer): We should set the last sent packet id here and
        // verify that we correctly ignore any packet loss reported prior to
        // that id.
        new_route.local = rtc::RouteEndpoint::CreateWithNetworkId(
            new_route.local.network_id() + 1);
        call_->GetTransportControllerSend()->OnNetworkRouteChanged("transport",
                                                                   new_route);
        EXPECT_GE(call_->GetStats().send_bandwidth_bps, kStartBitrateBps);
      });
    }

   private:
    webrtc::SequenceChecker module_process_thread_;
    webrtc::SequenceChecker task_queue_thread_;
    TaskQueueBase* const task_queue_;
    RtpHeaderExtensionMap extensions_;
    Call* call_ RTC_GUARDED_BY(task_queue_thread_);
  } test(task_queue());

  RunBaseTest(&test);
}

// Test that if specified, relay cap is lifted on transition to direct
// connection.
// TODO(https://bugs.webrtc.org/13353): Test disabled  due to flakiness.
TEST_F(VideoSendStreamTest, DISABLED_RelayToDirectRoute) {
  static const int kStartBitrateBps = 300000;
  static const int kRelayBandwidthCapBps = 800000;
  static const int kMinPacketsToSend = 100;
  webrtc::test::ScopedKeyValueConfig field_trials(
      field_trials_, "WebRTC-Bwe-NetworkRouteConstraints/relay_cap:" +
                         std::to_string(kRelayBandwidthCapBps) + "bps/");

  class RelayToDirectRouteTest : public test::EndToEndTest {
   public:
    explicit RelayToDirectRouteTest(TaskQueueBase* task_queue)
        : EndToEndTest(test::VideoTestConstants::kDefaultTimeout),
          task_queue_(task_queue),
          call_(nullptr),
          packets_sent_(0),
          relayed_phase_(true) {
      module_process_thread_.Detach();
      task_queue_thread_.Detach();
    }

    ~RelayToDirectRouteTest() {
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