/*
 *  Copyright 2016 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 "pc/rtc_stats_collector.h"

#include <cstddef>
#include <cstdint>
#include <initializer_list>
#include <map>
#include <memory>
#include <optional>
#include <string>
#include <tuple>
#include <utility>
#include <vector>

#include "absl/functional/any_invocable.h"
#include "absl/strings/str_cat.h"
#include "absl/strings/str_replace.h"
#include "api/audio/audio_device.h"
#include "api/audio/audio_processing_statistics.h"
#include "api/audio_options.h"
#include "api/candidate.h"
#include "api/data_channel_interface.h"
#include "api/dtls_transport_interface.h"
#include "api/environment/environment.h"
#include "api/make_ref_counted.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtp_parameters.h"
#include "api/rtp_transceiver_direction.h"
#include "api/scoped_refptr.h"
#include "api/stats/attribute.h"
#include "api/stats/rtc_stats.h"
#include "api/stats/rtc_stats_collector_callback.h"
#include "api/stats/rtc_stats_report.h"
#include "api/stats/rtcstats_objects.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/transport/enums.h"
#include "api/units/data_rate.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_content_type.h"
#include "api/video/video_timing.h"
#include "api/video_codecs/scalability_mode.h"
#include "call/call.h"
#include "common_video/include/quality_limitation_reason.h"
#include "json/reader.h"
#include "json/value.h"
#include "media/base/fake_media_engine.h"
#include "media/base/media_channel.h"
#include "media/base/stream_params.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
#include "p2p/base/connection_info.h"
#include "p2p/base/ice_transport_internal.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port.h"
#include "p2p/base/transport_description.h"
#include "pc/media_stream.h"
#include "pc/peer_connection_internal.h"
#include "pc/rtp_sender.h"
#include "pc/rtp_transceiver.h"
#include "pc/scoped_operations_batcher.h"
#include "pc/sctp_data_channel.h"
#include "pc/stream_collection.h"
#include "pc/test/fake_audio_track.h"
#include "pc/test/fake_data_channel_controller.h"
#include "pc/test/fake_peer_connection_for_stats.h"
#include "pc/test/fake_video_track.h"
#include "pc/test/fake_video_track_source.h"
#include "pc/test/mock_data_channel.h"
#include "pc/test/mock_rtp_receiver_internal.h"
#include "pc/test/mock_rtp_sender_internal.h"
#include "pc/test/rtc_stats_obtainer.h"
#include "pc/transport_stats.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/fake_ssl_identity.h"
#include "rtc_base/network_constants.h"
#include "rtc_base/rtc_certificate.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/ssl_certificate.h"
#include "rtc_base/ssl_fingerprint.h"
#include "rtc_base/ssl_identity.h"
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/plan_b_only.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/clock.h"
#include "test/create_test_environment.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/run_loop.h"
#include "test/time_controller/simulated_time_controller.h"

using ::testing::_;
using ::testing::IsEmpty;
using ::testing::Not;
using ::testing::NotNull;
using ::testing::Return;

namespace webrtc {

namespace {

using RunLoop = test::RunLoop;

// Fake data used by `SetupExampleStatsVoiceGraph()` to fill in remote outbound
// stats.
constexpr Timestamp kRemoteOutboundStatsTimestamp = Timestamp::Millis(123);
constexpr Timestamp kRemoteOutboundStatsRemoteTimestamp =
    Timestamp::Millis(456);
constexpr uint32_t kRemoteOutboundStatsPacketsSent = 7u;
constexpr uint64_t kRemoteOutboundStatsBytesSent = 8u;
constexpr uint64_t kRemoteOutboundStatsReportsCount = 9u;

struct CertificateInfo {
  scoped_refptr<RTCCertificate> certificate;
  std::vector<std::string> ders;
  std::vector<std::string> pems;
  std::vector<std::string> fingerprints;
};

// Return the ID for an object of the given type in a report.
// The object must be present and be unique.
template <typename T>
std::string IdForType(const RTCStatsReport* report) {
  auto stats_of_my_type = report->RTCStatsReport::GetStatsOfType<T>();
  // We cannot use ASSERT here, since we're within a function.
  EXPECT_EQ(1U, stats_of_my_type.size())
      << "Unexpected number of stats of this type";
  if (stats_of_my_type.size() == 1) {
    return stats_of_my_type[0]->id();
  } else {
    // Return something that is not going to be a valid stas ID.
    return "Type not found";
  }
}

std::unique_ptr<CertificateInfo> CreateFakeCertificateAndInfoFromDers(
    const std::vector<std::string>& ders) {
  RTC_CHECK(!ders.empty());
  std::unique_ptr<CertificateInfo> info(new CertificateInfo());
  info->ders = ders;
  for (const std::string& der : ders) {
    info->pems.push_back(SSLIdentity::DerToPem(
        "CERTIFICATE", reinterpret_cast<const unsigned char*>(der.c_str()),
        der.length()));
  }
  info->certificate = RTCCertificate::Create(
      std::unique_ptr<FakeSSLIdentity>(new FakeSSLIdentity(info->pems)));
  // Strip header/footer and newline characters of PEM strings.
  for (size_t i = 0; i < info->pems.size(); ++i) {
    absl::StrReplaceAll({{"-----BEGIN CERTIFICATE-----", ""},
                         {"-----END CERTIFICATE-----", ""},
                         {"\n", ""}},
                        &info->pems[i]);
  }
  // Fingerprints for the whole certificate chain, starting with leaf
  // certificate.
  const SSLCertChain& chain = info->certificate->GetSSLCertificateChain();
  std::unique_ptr<SSLFingerprint> fp;
  for (size_t i = 0; i < chain.GetSize(); i++) {
    fp = SSLFingerprint::Create("sha-1", chain.Get(i));
    EXPECT_TRUE(fp);
    info->fingerprints.push_back(fp->GetRfc4572Fingerprint());
  }
  EXPECT_EQ(info->ders.size(), info->fingerprints.size());
  return info;
}

std::unique_ptr<Candidate> CreateFakeCandidate(
    const std::string& hostname,
    int port,
    const std::string& protocol,
    const AdapterType adapter_type,
    IceCandidateType candidate_type,
    uint32_t priority,
    const AdapterType underlying_type_for_vpn = ADAPTER_TYPE_UNKNOWN) {
  std::unique_ptr<Candidate> candidate(new Candidate(
      ICE_CANDIDATE_COMPONENT_RTP, protocol, SocketAddress(hostname, port),
      priority, "iceusernamefragment", "" /* pwd */, candidate_type,
      0 /* generation */, "foundationIsAString"));

  candidate->set_network_type(adapter_type);
  candidate->set_underlying_type_for_vpn(underlying_type_for_vpn);
  return candidate;
}

class MockStatsCollectorCallback : public RTCStatsCollectorCallback {
 public:
  MOCK_METHOD(void,
              OnStatsDelivered,
              (const scoped_refptr<const RTCStatsReport>&),
              (override));
};

// Helper functions to create fake tracks for stats tests.
scoped_refptr<AudioTrackInterface> CreateFakeAudioTrackForStats(
    const std::string& id,
    MediaStreamTrackInterface::TrackState state,
    bool create_fake_audio_processor) {
  scoped_refptr<FakeAudioProcessor> processor = nullptr;
  if (create_fake_audio_processor) {
    processor = make_ref_counted<FakeAudioProcessor>();
    processor->stats.apm_statistics.echo_return_loss = 2.0;
    processor->stats.apm_statistics.echo_return_loss_enhancement = 3.0;
  }
  return FakeAudioTrack::Create(id, state, processor);
}

scoped_refptr<VideoTrackInterface> CreateFakeVideoTrackForStats(
    const std::string& id,
    MediaStreamTrackInterface::TrackState state,
    scoped_refptr<VideoTrackSourceInterface> source) {
  return FakeVideoTrack::Create(id, state, source);
}

scoped_refptr<MediaStreamTrackInterface> CreateFakeTrack(
    MediaType media_type,
    const std::string& track_id,
    MediaStreamTrackInterface::TrackState track_state,
    bool create_fake_audio_processor = false) {
  if (media_type == MediaType::AUDIO) {
    return CreateFakeAudioTrackForStats(track_id, track_state,
                                        create_fake_audio_processor);
  } else {
    RTC_DCHECK_EQ(media_type, MediaType::VIDEO);
    return CreateFakeVideoTrackForStats(track_id, track_state, nullptr);
  }
}

scoped_refptr<MockRtpSenderInternal> CreateMockSender(
    MediaType media_type,
    scoped_refptr<MediaStreamTrackInterface> track,
    uint32_t ssrc,
    int attachment_id,
    std::vector<std::string> local_stream_ids) {
  RTC_DCHECK(!track ||
             (track->kind() == MediaStreamTrackInterface::kAudioKind &&
              media_type == MediaType::AUDIO) ||
             (track->kind() == MediaStreamTrackInterface::kVideoKind &&
              media_type == MediaType::VIDEO));
  auto sender = make_ref_counted<MockRtpSenderInternal>();
  EXPECT_CALL(*sender, track()).WillRepeatedly(Return(track));
  EXPECT_CALL(*sender, ssrc()).WillRepeatedly(Return(ssrc));
  EXPECT_CALL(*sender, media_type()).WillRepeatedly(Return(media_type));
  EXPECT_CALL(*sender, GetParameters()).WillRepeatedly([s = sender.get()]() {
    return s->GetParametersInternal(false, false);
  });
  EXPECT_CALL(*sender, GetParametersInternal(_, _)).WillRepeatedly([ssrc]() {
    RtpParameters params;
    params.encodings.push_back(RtpEncodingParameters());
    params.encodings[0].ssrc = ssrc;
    return params;
  });
  EXPECT_CALL(*sender, AttachmentId()).WillRepeatedly(Return(attachment_id));
  EXPECT_CALL(*sender, stream_ids()).WillRepeatedly(Return(local_stream_ids));
  return sender;
}

scoped_refptr<MockRtpReceiverInternal> CreateMockReceiver(
    const scoped_refptr<MediaStreamTrackInterface>& track,
    uint32_t ssrc,
    int attachment_id) {
  auto receiver = make_ref_counted<MockRtpReceiverInternal>();
  EXPECT_CALL(*receiver, track()).WillRepeatedly(Return(track));
  EXPECT_CALL(*receiver, ssrc()).WillRepeatedly([ssrc]() { return ssrc; });
  EXPECT_CALL(*receiver, streams())
      .WillRepeatedly(
          Return(std::vector<scoped_refptr<MediaStreamInterface>>({})));

  EXPECT_CALL(*receiver, media_type())
      .WillRepeatedly(
          Return(track->kind() == MediaStreamTrackInterface::kAudioKind
                     ? MediaType::AUDIO
                     : MediaType::VIDEO));
  EXPECT_CALL(*receiver, GetParameters()).WillRepeatedly([ssrc]() {
    RtpParameters params;
    params.encodings.push_back(RtpEncodingParameters());
    params.encodings[0].ssrc = ssrc;
    return params;
  });
  EXPECT_CALL(*receiver, AttachmentId()).WillRepeatedly(Return(attachment_id));
  EXPECT_CALL(*receiver, Stop()).WillRepeatedly(Return());
  return receiver;
}

class RTCStatsCollectorWrapper {
 public:
  explicit RTCStatsCollectorWrapper(
      scoped_refptr<FakePeerConnectionForStats> pc,
      const Environment& env)
      : pc_(pc), env_(env), stats_collector_(pc_.get(), env) {}

  RTCStatsCollector& stats_collector() { return stats_collector_; }

  scoped_refptr<const RTCStatsReport> GetStatsReport(RunLoop& loop) {
    scoped_refptr<const RTCStatsReport> report;
    stats_collector_.GetStatsReport(
        RTCStatsObtainer::Create(&report, [&loop] { loop.Quit(); }));
    loop.Run();
    Timestamp after = env_.clock().CurrentTime();
    for (const RTCStats& stats : *report) {
      if (stats.type() == RTCRemoteInboundRtpStreamStats::kType ||
          stats.type() == RTCRemoteOutboundRtpStreamStats::kType) {
        // Ignore remote timestamps.
        continue;
      }
      EXPECT_LE(stats.timestamp(), after);
    }
    return report;
  }

  scoped_refptr<const RTCStatsReport> GetFreshStatsReport(RunLoop& loop) {
    stats_collector_.ClearCachedStatsReport();
    return GetStatsReport(loop);
  }

  scoped_refptr<MockRtpSenderInternal> SetupLocalTrackAndSender(
      MediaType media_type,
      const std::string& track_id,
      uint32_t ssrc,
      bool add_stream,
      int attachment_id) {
    scoped_refptr<MediaStream> local_stream;
    if (add_stream) {
      local_stream = MediaStream::Create("LocalStreamId");
      pc_->mutable_local_streams()->AddStream(local_stream);
    }
    scoped_refptr<MediaStreamTrackInterface> track;
    if (media_type == MediaType::AUDIO) {
      track = CreateFakeTrack(media_type, track_id,
                              MediaStreamTrackInterface::kLive);
      if (add_stream) {
        local_stream->AddTrack(scoped_refptr<AudioTrackInterface>(
            static_cast<AudioTrackInterface*>(track.get())));
      }
    } else {
      track = CreateFakeTrack(media_type, track_id,
                              MediaStreamTrackInterface::kLive);
      if (add_stream) {
        local_stream->AddTrack(scoped_refptr<VideoTrackInterface>(
            static_cast<VideoTrackInterface*>(track.get())));
      }
    }
    scoped_refptr<MockRtpSenderInternal> sender =
        make_ref_counted<MockRtpSenderInternal>();
    EXPECT_CALL(*sender, track()).WillRepeatedly(Return(track));
    EXPECT_CALL(*sender, ssrc()).WillRepeatedly(Return(ssrc));
    EXPECT_CALL(*sender, media_type()).WillRepeatedly([media_type]() {
      return media_type;
    });
    EXPECT_CALL(*sender, GetParameters()).WillRepeatedly([s = sender.get()]() {
      return s->GetParametersInternal(false, false);
    });
    EXPECT_CALL(*sender, GetParametersInternal(_, _)).WillRepeatedly([ssrc]() {
      RtpParameters params;
      params.encodings.push_back(RtpEncodingParameters());
      params.encodings[0].ssrc = ssrc;
      return params;
    });
    EXPECT_CALL(*sender, AttachmentId()).WillRepeatedly(Return(attachment_id));
    EXPECT_CALL(*sender, Stop()).WillRepeatedly(Return());
    EXPECT_CALL(*sender, SetSendCodecs(_)).WillRepeatedly(Return());
    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
    pc_->AddSender(sender);
    RTC_ALLOW_PLAN_B_DEPRECATION_END();
    return sender;
  }

  scoped_refptr<MockRtpReceiverInternal> SetupRemoteTrackAndReceiver(
      MediaType media_type,
      const std::string& track_id,
      const std::string& stream_id,
      uint32_t ssrc) {
    scoped_refptr<MediaStream> remote_stream = MediaStream::Create(stream_id);
    pc_->mutable_remote_streams()->AddStream(remote_stream);

    scoped_refptr<MediaStreamTrackInterface> track;
    if (media_type == MediaType::AUDIO) {
      track = CreateFakeTrack(media_type, track_id,
                              MediaStreamTrackInterface::kLive);
      remote_stream->AddTrack(scoped_refptr<AudioTrackInterface>(
          static_cast<AudioTrackInterface*>(track.get())));
    } else {
      track = CreateFakeTrack(media_type, track_id,
                              MediaStreamTrackInterface::kLive);
      remote_stream->AddTrack(scoped_refptr<VideoTrackInterface>(
          static_cast<VideoTrackInterface*>(track.get())));
    }

    scoped_refptr<MockRtpReceiverInternal> receiver =
        CreateMockReceiver(track, ssrc, 62);
    EXPECT_CALL(*receiver, streams())
        .WillRepeatedly(Return(
            std::vector<scoped_refptr<MediaStreamInterface>>({remote_stream})));
    EXPECT_CALL(*receiver, SetMediaChannel(_)).WillRepeatedly(Return());
    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
    pc_->AddReceiver(receiver);
    RTC_ALLOW_PLAN_B_DEPRECATION_END();
    return receiver;
  }

  // Attaches tracks to peer connections by configuring RTP senders and RTP
  // receivers according to the tracks' pairings with
  // |[Voice/Video][Sender/Receiver]Info| and their SSRCs. Local tracks can be
  // associated with multiple |[Voice/Video]SenderInfo|s, remote tracks can only
  // be associated with one |[Voice/Video]ReceiverInfo|.
  // Senders get assigned attachment ID "ssrc + 10".
  void CreateMockRtpSendersReceiversAndChannels(
      std::initializer_list<
          std::pair<MediaStreamTrackInterface*, VoiceSenderInfo>>
          local_audio_track_info_pairs,
      std::initializer_list<
          std::pair<MediaStreamTrackInterface*, VoiceReceiverInfo>>
          remote_audio_track_info_pairs,
      std::initializer_list<
          std::pair<MediaStreamTrackInterface*, VideoSenderInfo>>
          local_video_track_info_pairs,
      std::initializer_list<
          std::pair<MediaStreamTrackInterface*, VideoReceiverInfo>>
          remote_video_track_info_pairs,
      std::vector<std::string> local_stream_ids,
      std::vector<scoped_refptr<MediaStreamInterface>> remote_streams) {
    VoiceMediaInfo voice_media_info;
    VideoMediaInfo video_media_info;

    // Local audio tracks and voice sender infos
    for (auto& pair : local_audio_track_info_pairs) {
      MediaStreamTrackInterface* local_audio_track = pair.first;
      const VoiceSenderInfo& voice_sender_info = pair.second;
      RTC_DCHECK_EQ(local_audio_track->kind(),
                    MediaStreamTrackInterface::kAudioKind);

      voice_media_info.senders.push_back(voice_sender_info);
      scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockSender(
          MediaType::AUDIO,
          scoped_refptr<MediaStreamTrackInterface>(local_audio_track),
          voice_sender_info.local_stats[0].ssrc,
          voice_sender_info.local_stats[0].ssrc + 10, local_stream_ids);
      EXPECT_CALL(*rtp_sender, SetMediaChannel(_)).WillRepeatedly(Return());
      EXPECT_CALL(*rtp_sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
        return nullptr;
      });
      EXPECT_CALL(*rtp_sender, SetSendCodecs(_));
      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
      pc_->AddSender(rtp_sender);
      RTC_ALLOW_PLAN_B_DEPRECATION_END();
    }

    // Remote audio tracks and voice receiver infos
    for (auto& pair : remote_audio_track_info_pairs) {
      MediaStreamTrackInterface* remote_audio_track = pair.first;
      const VoiceReceiverInfo& voice_receiver_info = pair.second;
      RTC_DCHECK_EQ(remote_audio_track->kind(),
                    MediaStreamTrackInterface::kAudioKind);

      voice_media_info.receivers.push_back(voice_receiver_info);
      scoped_refptr<MockRtpReceiverInternal> rtp_receiver = CreateMockReceiver(
          scoped_refptr<MediaStreamTrackInterface>(remote_audio_track),
          voice_receiver_info.local_stats[0].ssrc,
          voice_receiver_info.local_stats[0].ssrc + 10);
      EXPECT_CALL(*rtp_receiver, streams())
          .WillRepeatedly(Return(remote_streams));
      EXPECT_CALL(*rtp_receiver, SetMediaChannel(_)).WillRepeatedly(Return());
      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
      pc_->AddReceiver(rtp_receiver);
      RTC_ALLOW_PLAN_B_DEPRECATION_END();
    }

    // Local video tracks and video sender infos
    for (auto& pair : local_video_track_info_pairs) {
      MediaStreamTrackInterface* local_video_track = pair.first;
      const VideoSenderInfo& video_sender_info = pair.second;
      RTC_DCHECK_EQ(local_video_track->kind(),
                    MediaStreamTrackInterface::kVideoKind);

      video_media_info.senders.push_back(video_sender_info);
      video_media_info.aggregated_senders.push_back(video_sender_info);
      scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockSender(
          MediaType::VIDEO,
          scoped_refptr<MediaStreamTrackInterface>(local_video_track),
          video_sender_info.local_stats[0].ssrc,
          video_sender_info.local_stats[0].ssrc + 10, local_stream_ids);
      EXPECT_CALL(*rtp_sender, SetMediaChannel(_)).WillRepeatedly(Return());
      EXPECT_CALL(*rtp_sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
        return nullptr;
      });
      EXPECT_CALL(*rtp_sender, SetSendCodecs(_));
      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
      pc_->AddSender(rtp_sender);
      RTC_ALLOW_PLAN_B_DEPRECATION_END();
    }

    // Remote video tracks and video receiver infos
    for (auto& pair : remote_video_track_info_pairs) {
      MediaStreamTrackInterface* remote_video_track = pair.first;
      const VideoReceiverInfo& video_receiver_info = pair.second;
      RTC_DCHECK_EQ(remote_video_track->kind(),
                    MediaStreamTrackInterface::kVideoKind);

      video_media_info.receivers.push_back(video_receiver_info);
      scoped_refptr<MockRtpReceiverInternal> rtp_receiver = CreateMockReceiver(
          scoped_refptr<MediaStreamTrackInterface>(remote_video_track),
          video_receiver_info.local_stats[0].ssrc,
          video_receiver_info.local_stats[0].ssrc + 10);
      EXPECT_CALL(*rtp_receiver, streams())
          .WillRepeatedly(Return(remote_streams));
      EXPECT_CALL(*rtp_receiver, SetMediaChannel(_)).WillRepeatedly(Return());
      RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
      pc_->AddReceiver(rtp_receiver);
      RTC_ALLOW_PLAN_B_DEPRECATION_END();
    }

    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
    pc_->AddVoiceChannel("audio", "transport", voice_media_info);
    pc_->AddVideoChannel("video", "transport", video_media_info);
    RTC_ALLOW_PLAN_B_DEPRECATION_END();
  }

 private:
  scoped_refptr<FakePeerConnectionForStats> pc_;
  const Environment env_;
  RTCStatsCollector stats_collector_;
};

class RTCStatsCollectorTestBase : public ::testing::Test {
 public:
  RTCStatsCollectorTestBase()
      : time_controller_(Timestamp::Zero()),
        env_(CreateTestEnvironment({.time = time_controller_.GetClock()})),
        pc_(make_ref_counted<FakePeerConnectionForStats>(env_)) {
    data_channel_controller_ =
        std::make_unique<FakeDataChannelController>(pc_->network_thread());
  }

  void SetStatsTimestampWithEnvironmentClock(bool enabled) {
    PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
    config.set_stats_timestamp_with_environment_clock(enabled);
    pc_->SetConfiguration(config);
    // RTCStatsCollector reads this flag in the constructor, so we need to
    // recreate it.
    stats_ = std::make_unique<RTCStatsCollectorWrapper>(pc_, env_);
  }

  void ExpectReportContainsCertificateInfo(
      const scoped_refptr<const RTCStatsReport>& report,
      const CertificateInfo& certinfo) {
    for (size_t i = 0; i < certinfo.fingerprints.size(); ++i) {
      RTCCertificateStats expected_certificate_stats(
          "CF" + certinfo.fingerprints[i], report->timestamp());
      expected_certificate_stats.fingerprint = certinfo.fingerprints[i];
      expected_certificate_stats.fingerprint_algorithm = "sha-1";
      expected_certificate_stats.base64_certificate = certinfo.pems[i];
      if (i + 1 < certinfo.fingerprints.size()) {
        expected_certificate_stats.issuer_certificate_id =
            "CF" + certinfo.fingerprints[i + 1];
      }
      ASSERT_TRUE(report->Get(expected_certificate_stats.id()));
      EXPECT_EQ(expected_certificate_stats,
                report->Get(expected_certificate_stats.id())
                    ->cast_to<RTCCertificateStats>());
    }
  }

  const RTCCertificateStats* GetCertificateStatsFromFingerprint(
      const scoped_refptr<const RTCStatsReport>& report,
      const std::string& fingerprint) {
    auto certificates = report->GetStatsOfType<RTCCertificateStats>();
    for (const auto* certificate : certificates) {
      if (*certificate->fingerprint == fingerprint) {
        return certificate;
      }
    }
    return nullptr;
  }

  struct ExampleStatsGraph {
    scoped_refptr<RtpSenderInternal> sender;
    scoped_refptr<RtpReceiverInternal> receiver;

    scoped_refptr<const RTCStatsReport> full_report;
    std::string send_codec_id;
    std::string recv_codec_id;
    std::string outbound_rtp_id;
    std::string inbound_rtp_id;
    std::string remote_outbound_rtp_id;
    std::string transport_id;
    std::string peer_connection_id;
    std::string media_source_id;
  };

  // Sets up the example stats graph (see ASCII art below) for a video only
  // call. The graph is used for testing the stats selection algorithm (see
  // https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm).
  // These tests test the integration of the stats traversal algorithm inside of
  // RTCStatsCollector. See rtcstatstraveral_unittest.cc for more stats
  // traversal tests.
  ExampleStatsGraph SetupExampleStatsGraphForSelectorTests() {
    ExampleStatsGraph graph;

    // codec (send)
    graph.send_codec_id = "COTTransportName1_1";
    VideoMediaInfo video_media_info;
    RtpCodecParameters send_codec;
    send_codec.payload_type = 1;
    send_codec.clock_rate = 0;
    video_media_info.send_codecs.insert(
        std::make_pair(send_codec.payload_type, send_codec));
    // codec (recv)
    graph.recv_codec_id = "CITTransportName1_2";
    RtpCodecParameters recv_codec;
    recv_codec.payload_type = 2;
    recv_codec.clock_rate = 0;
    video_media_info.receive_codecs.insert(
        std::make_pair(recv_codec.payload_type, recv_codec));
    // outbound-rtp
    graph.outbound_rtp_id = "OTTransportName1V3";
    video_media_info.senders.push_back(VideoSenderInfo());
    video_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
    video_media_info.senders[0].local_stats[0].ssrc = 3;
    video_media_info.senders[0].codec_payload_type = send_codec.payload_type;
    video_media_info.aggregated_senders.push_back(video_media_info.senders[0]);
    // inbound-rtp
    graph.inbound_rtp_id = "ITTransportName1V4";
    video_media_info.receivers.push_back(VideoReceiverInfo());
    video_media_info.receivers[0].local_stats.push_back(SsrcReceiverInfo());
    video_media_info.receivers[0].local_stats[0].ssrc = 4;
    video_media_info.receivers[0].codec_payload_type = recv_codec.payload_type;
    video_media_info.receivers[0].packets_received = 123;
    // transport
    graph.transport_id = "TTransportName1";
    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
    pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
    RTC_ALLOW_PLAN_B_DEPRECATION_END();
    // outbound-rtp's sender
    graph.sender = stats_->SetupLocalTrackAndSender(
        MediaType::VIDEO, "LocalVideoTrackID", 3, false, 50);
    // inbound-rtp's receiver
    graph.receiver = stats_->SetupRemoteTrackAndReceiver(
        MediaType::VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 4);
    // peer-connection
    graph.peer_connection_id = "P";
    // media-source (kind: video)
    graph.media_source_id = "SV" + absl::StrCat(graph.sender->AttachmentId());

    // Expected stats graph:
    //
    //  media-source                     peer-connection
    //    ^
    //    |
    //    +--------- outbound-rtp   inbound-rtp
    //                |        |     |       |
    //                v        v     v       v
    //       codec (send)     transport     codec (recv)

    // Verify the stats graph is set up correctly.
    graph.full_report = stats_->GetStatsReport(main_thread_);
    EXPECT_EQ(graph.full_report->size(), 7u);
    EXPECT_TRUE(graph.full_report->Get(graph.send_codec_id));
    EXPECT_TRUE(graph.full_report->Get(graph.recv_codec_id));
    EXPECT_TRUE(graph.full_report->Get(graph.outbound_rtp_id));
    EXPECT_TRUE(graph.full_report->Get(graph.inbound_rtp_id));
    EXPECT_TRUE(graph.full_report->Get(graph.transport_id));
    EXPECT_TRUE(graph.full_report->Get(graph.peer_connection_id));
    EXPECT_TRUE(graph.full_report->Get(graph.media_source_id));
    const auto& outbound_rtp = graph.full_report->Get(graph.outbound_rtp_id)
                                   ->cast_to<RTCOutboundRtpStreamStats>();
    EXPECT_EQ(*outbound_rtp.media_source_id, graph.media_source_id);
    EXPECT_EQ(*outbound_rtp.codec_id, graph.send_codec_id);
    EXPECT_EQ(*outbound_rtp.transport_id, graph.transport_id);
    EXPECT_TRUE(graph.full_report->Get(graph.inbound_rtp_id));
    // We can't use an ASSERT in a function returning non-void, so just return.
    if (!graph.full_report->Get(graph.inbound_rtp_id)) {
      return graph;
    }
    const auto& inbound_rtp = graph.full_report->Get(graph.inbound_rtp_id)
                                  ->cast_to<RTCInboundRtpStreamStats>();
    EXPECT_EQ(*inbound_rtp.codec_id, graph.recv_codec_id);
    EXPECT_EQ(*inbound_rtp.transport_id, graph.transport_id);

    return graph;
  }

  // Sets up an example stats graph (see ASCII art below) for an audio only call
  // and checks that the expected stats are generated.
  ExampleStatsGraph SetupExampleStatsVoiceGraph(
      bool add_remote_outbound_stats) {
    constexpr uint32_t kLocalSsrc = 3;
    constexpr uint32_t kRemoteSsrc = 4;
    ExampleStatsGraph graph;

    // codec (send)
    graph.send_codec_id = "COTTransportName1_1";
    VoiceMediaInfo media_info;
    RtpCodecParameters send_codec;
    send_codec.payload_type = 1;
    send_codec.clock_rate = 0;
    media_info.send_codecs.insert(
        std::make_pair(send_codec.payload_type, send_codec));
    // codec (recv)
    graph.recv_codec_id = "CITTransportName1_2";
    RtpCodecParameters recv_codec;
    recv_codec.payload_type = 2;
    recv_codec.clock_rate = 0;
    media_info.receive_codecs.insert(
        std::make_pair(recv_codec.payload_type, recv_codec));
    // outbound-rtp
    graph.outbound_rtp_id = "OTTransportName1A3";
    media_info.senders.push_back(VoiceSenderInfo());
    media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
    media_info.senders[0].local_stats[0].ssrc = kLocalSsrc;
    media_info.senders[0].codec_payload_type = send_codec.payload_type;
    // inbound-rtp
    graph.inbound_rtp_id = "ITTransportName1A4";
    media_info.receivers.push_back(VoiceReceiverInfo());
    media_info.receivers[0].local_stats.push_back(SsrcReceiverInfo());
    media_info.receivers[0].local_stats[0].ssrc = kRemoteSsrc;
    media_info.receivers[0].packets_received = 123;
    media_info.receivers[0].codec_payload_type = recv_codec.payload_type;
    // remote-outbound-rtp
    if (add_remote_outbound_stats) {
      graph.remote_outbound_rtp_id = "ROA4";
      media_info.receivers[0].last_sender_report_utc_timestamp =
          kRemoteOutboundStatsTimestamp;
      media_info.receivers[0].last_sender_report_timestamp =
          kRemoteOutboundStatsTimestamp;
      media_info.receivers[0].last_sender_report_remote_utc_timestamp =
          kRemoteOutboundStatsRemoteTimestamp;
      media_info.receivers[0].sender_reports_packets_sent =
          kRemoteOutboundStatsPacketsSent;
      media_info.receivers[0].sender_reports_bytes_sent =
          kRemoteOutboundStatsBytesSent;
      media_info.receivers[0].sender_reports_reports_count =
          kRemoteOutboundStatsReportsCount;
    }
    // transport
    graph.transport_id = "TTransportName1";
    RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
    pc_->AddVoiceChannel("VoiceMid", "TransportName", media_info);
    RTC_ALLOW_PLAN_B_DEPRECATION_END();
    // outbound-rtp's sender
    graph.sender = stats_->SetupLocalTrackAndSender(
        MediaType::AUDIO, "LocalAudioTrackID", kLocalSsrc, false, 50);
    // inbound-rtp's receiver
    graph.receiver = stats_->SetupRemoteTrackAndReceiver(
        MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", kRemoteSsrc);
    // peer-connection
    graph.peer_connection_id = "P";
    // media-source (kind: video)
    graph.media_source_id = "SA" + absl::StrCat(graph.sender->AttachmentId());

    // Expected stats graph:
    //
    //  media-source                     peer-connection
    //    ^
    //    |
    //    +--------- outbound-rtp   inbound-rtp
    //                |        |     |       |
    //                v        v     v       v
    //       codec (send)     transport     codec (recv)

    // Verify the stats graph is set up correctly.
    graph.full_report = stats_->GetStatsReport(main_thread_);
    EXPECT_EQ(graph.full_report->size(), add_remote_outbound_stats ? 8u : 7u);
    EXPECT_TRUE(graph.full_report->Get(graph.send_codec_id));
    EXPECT_TRUE(graph.full_report->Get(graph.recv_codec_id));
    EXPECT_TRUE(graph.full_report->Get(graph.outbound_rtp_id));
    EXPECT_TRUE(graph.full_report->Get(graph.inbound_rtp_id));
    EXPECT_TRUE(graph.full_report->Get(graph.transport_id));
    EXPECT_TRUE(graph.full_report->Get(graph.peer_connection_id));
    EXPECT_TRUE(graph.full_report->Get(graph.media_source_id));
    // `graph.remote_outbound_rtp_id` is omitted on purpose so that expectations
    // can be added by the caller depending on what value it sets for the
    // `add_remote_outbound_stats` argument.
    const auto& outbound_rtp = graph.full_report->Get(graph.outbound_rtp_id)
                                   ->cast_to<RTCOutboundRtpStreamStats>();
    EXPECT_EQ(*outbound_rtp.media_source_id, graph.media_source_id);
    EXPECT_EQ(*outbound_rtp.codec_id, graph.send_codec_id);
    EXPECT_EQ(*outbound_rtp.transport_id, graph.transport_id);
    EXPECT_TRUE(graph.full_report->Get(graph.inbound_rtp_id));
    // We can't use ASSERT in a function with a return value.
    if (!graph.full_report->Get(graph.inbound_rtp_id)) {
      return graph;
    }
    const auto& inbound_rtp = graph.full_report->Get(graph.inbound_rtp_id)
                                  ->cast_to<RTCInboundRtpStreamStats>();
    EXPECT_EQ(*inbound_rtp.codec_id, graph.recv_codec_id);
    EXPECT_EQ(*inbound_rtp.transport_id, graph.transport_id);

    return graph;
  }

 protected:
  GlobalSimulatedTimeController time_controller_;
  Environment env_;
  RunLoop main_thread_;
  scoped_refptr<FakePeerConnectionForStats> pc_;
  std::unique_ptr<RTCStatsCollectorWrapper> stats_;
  std::unique_ptr<FakeDataChannelController> data_channel_controller_;
};

class RTCStatsCollectorTest : public RTCStatsCollectorTestBase,
                              public ::testing::WithParamInterface<bool> {
 public:
  RTCStatsCollectorTest() { SetStatsTimestampWithEnvironmentClock(GetParam()); }
};

TEST_P(RTCStatsCollectorTest, SingleCallback) {
  scoped_refptr<const RTCStatsReport> result;
  stats_->stats_collector().GetStatsReport(
      RTCStatsObtainer::Create(&result, [&] { main_thread_.Quit(); }));
  main_thread_.Run();
  EXPECT_TRUE(result);
}

TEST_P(RTCStatsCollectorTest, MultipleCallbacks) {
  scoped_refptr<const RTCStatsReport> a, b, c;
  int remaining = 3;
  auto on_complete = [&] {
    if (--remaining == 0)
      main_thread_.Quit();
  };
  stats_->stats_collector().GetStatsReport(
      RTCStatsObtainer::Create(&a, on_complete));
  stats_->stats_collector().GetStatsReport(
      RTCStatsObtainer::Create(&b, on_complete));
  stats_->stats_collector().GetStatsReport(
      RTCStatsObtainer::Create(&c, on_complete));
  main_thread_.Run();
  EXPECT_TRUE(a);
  EXPECT_TRUE(b);
  EXPECT_TRUE(c);

  EXPECT_EQ(a.get(), b.get());
  EXPECT_EQ(b.get(), c.get());
}

TEST_P(RTCStatsCollectorTest, CachedStatsReports) {
  // Caching should ensure `a` and `b` are the same report.
  scoped_refptr<const RTCStatsReport> a = stats_->GetStatsReport(main_thread_);
  scoped_refptr<const RTCStatsReport> b = stats_->GetStatsReport(main_thread_);
  EXPECT_EQ(a.get(), b.get());
  // Invalidate cache by clearing it.
  stats_->stats_collector().ClearCachedStatsReport();
  scoped_refptr<const RTCStatsReport> c = stats_->GetStatsReport(main_thread_);
  EXPECT_NE(b.get(), c.get());
  // Invalidate cache by advancing time.
  time_controller_.AdvanceTime(TimeDelta::Millis(51));
  scoped_refptr<const RTCStatsReport> d = stats_->GetStatsReport(main_thread_);
  EXPECT_TRUE(d);
  EXPECT_NE(c.get(), d.get());
}

TEST_P(RTCStatsCollectorTest, MultipleCallbacksWithInvalidatedCacheInBetween) {
  scoped_refptr<const RTCStatsReport> a, b, c;
  int remaining = 3;
  auto on_complete = [&] {
    if (--remaining == 0)
      main_thread_.Quit();
  };
  stats_->stats_collector().GetStatsReport(
      RTCStatsObtainer::Create(&a, on_complete));
  stats_->stats_collector().GetStatsReport(
      RTCStatsObtainer::Create(&b, on_complete));
  // Cache is invalidated after 50 ms.
  main_thread_.Flush();
  time_controller_.AdvanceTime(TimeDelta::Millis(51));
  stats_->stats_collector().GetStatsReport(
      RTCStatsObtainer::Create(&c, on_complete));
  main_thread_.Run();
  EXPECT_TRUE(a);
  EXPECT_TRUE(b);
  EXPECT_TRUE(c);
  EXPECT_EQ(a.get(), b.get());
  // The act of doing `AdvanceTime` processes all messages. If this was not the
  // case we might not require `c` to be fresher than `b`.
  EXPECT_NE(c.get(), b.get());
}

TEST_P(RTCStatsCollectorTest, StatsPreservedWhenChannelClearedConcurrently) {
  VoiceMediaInfo voice_media_info;
  voice_media_info.receivers.emplace_back();
  voice_media_info.receivers[0].add_ssrc(1);
  voice_media_info.receivers[0].packets_received = 123;

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  auto remote_receiver = stats_->SetupRemoteTrackAndReceiver(
      MediaType::AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
  auto local_sender = stats_->SetupLocalTrackAndSender(
      MediaType::AUDIO, "LocalAudioTrackID", 2,
      /*add_stream=*/true, /*attachment_id=*/2);

  EXPECT_CALL(*local_sender, SetSendCodecs(_)).WillRepeatedly(Return());
  EXPECT_CALL(*local_sender, SetMediaChannel(_)).WillRepeatedly(Return());
  EXPECT_CALL(*local_sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
    return nullptr;
  });
  EXPECT_CALL(*remote_receiver, SetMediaChannel(_)).WillRepeatedly(Return());

  scoped_refptr<const RTCStatsReport> report;
  bool stats_obtained = false;
  stats_->stats_collector().GetStatsReport(
      RTCStatsObtainer::Create(&report, [&] {
        stats_obtained = true;
        main_thread_.Quit();
      }));

  auto transceivers = pc_->GetTransceiversInternal();
  ASSERT_FALSE(transceivers.empty());
  auto* transceiver = transceivers[0]->internal();
  EXPECT_TRUE(transceiver->HasChannel());

  auto network_task = transceiver->GetClearChannelNetworkTask();
  pc_->network_thread()->BlockingCall([&]() { std::move(network_task)(); });

  auto delete_worker_task =
      transceiver->GetDeleteChannelWorkerTask(/*stop_senders=*/false);
  // Now the state of the transceiver is that on the signaling thread,
  // there is no channel. However, the work of actually deleting the channels
  // belongs to the worker thread task, which hasn't run yet.
  // Let's post the execution of that task so that it ends up running
  // *after* the task that the `stats_collector` has queued up for
  // gathering the stats from the channel. The stats collector should
  // be able to get the stats safely even though technically the
  // transceiver by now, doesn't have a channel.
  EXPECT_FALSE(transceiver->HasChannel());
  Event worker_task_done;
  pc_->worker_thread()->PostTask([&]() {
    std::move(delete_worker_task)();
    worker_task_done.Set();
  });

  main_thread_.Run();

  EXPECT_TRUE(report);
  auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
  EXPECT_EQ(inbound_rtps.size(), 1u);
  // Wait for the cleanup task to complete.
  worker_task_done.Wait(Event::kForever);
}

TEST_P(RTCStatsCollectorTest, ToJsonProducesParseableJson) {
  ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  std::string json_format = report->ToJson();

  Json::CharReaderBuilder builder;
  Json::Value json_value;
  std::unique_ptr<Json::CharReader> reader(builder.newCharReader());
  ASSERT_TRUE(reader->parse(json_format.c_str(),
                            json_format.c_str() + json_format.size(),
                            &json_value, nullptr));

  // A very brief sanity check on the result.
  EXPECT_EQ(report->size(), json_value.size());
}

TEST_P(RTCStatsCollectorTest, ProduceRTPStreamStatsWithStoppedTransceiver) {
  const char kTransportName[] = "transport";
  const char kMid[] = "VideoMid";

  VideoMediaInfo video_media_info;
  video_media_info.senders.emplace_back();
  video_media_info.senders[0].add_ssrc(1);

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel(kMid, kTransportName, video_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  // Get the transceiver and set it to stopped.
  std::vector<scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
      transceivers = pc_->GetTransceiversInternal();
  ASSERT_THAT(transceivers, Not(IsEmpty()));
  scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> transceiver =
      transceivers[0];
  ASSERT_TRUE(transceiver);
  transceiver->internal()->set_current_direction(
      RtpTransceiverDirection::kStopped);

  // This should not crash.
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  EXPECT_THAT(report, NotNull());
}

TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsSingle) {
  const char kTransportName[] = "transport";

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("audio", kTransportName);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  std::unique_ptr<CertificateInfo> local_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          std::vector<std::string>({"(local) single certificate"}));
  pc_->SetLocalCertificate(kTransportName, local_certinfo->certificate);

  std::unique_ptr<CertificateInfo> remote_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          std::vector<std::string>({"(remote) single certificate"}));
  pc_->SetRemoteCertChain(
      kTransportName,
      remote_certinfo->certificate->GetSSLCertificateChain().Clone());

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  ExpectReportContainsCertificateInfo(report, *local_certinfo);
  ExpectReportContainsCertificateInfo(report, *remote_certinfo);
}

// These SSRC collisions are legal.
TEST_P(RTCStatsCollectorTest, ValidSsrcCollisionDoesNotCrash) {
  // BUNDLE audio/video inbound/outbound. Unique SSRCs needed within the BUNDLE.
  VoiceMediaInfo mid1_info;
  mid1_info.receivers.emplace_back();
  mid1_info.receivers[0].add_ssrc(1);
  mid1_info.receivers[0].packets_received = 123;
  mid1_info.senders.emplace_back();
  mid1_info.senders[0].add_ssrc(2);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("Mid1", "Transport1", mid1_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  VideoMediaInfo mid2_info;
  mid2_info.receivers.emplace_back();
  mid2_info.receivers[0].add_ssrc(3);
  mid2_info.receivers[0].packets_received = 123;
  mid2_info.senders.emplace_back();
  mid2_info.senders[0].add_ssrc(4);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid2", "Transport1", mid2_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  // Now create a second BUNDLE group with SSRCs colliding with the first group
  // (but again no collisions within the group).
  VoiceMediaInfo mid3_info;
  mid3_info.receivers.emplace_back();
  mid3_info.receivers[0].add_ssrc(1);
  mid3_info.receivers[0].packets_received = 123;
  mid3_info.senders.emplace_back();
  mid3_info.senders[0].add_ssrc(2);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("Mid3", "Transport2", mid3_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  VideoMediaInfo mid4_info;
  mid4_info.receivers.emplace_back();
  mid4_info.receivers[0].add_ssrc(3);
  mid4_info.receivers[0].packets_received = 123;
  mid4_info.senders.emplace_back();
  mid4_info.senders[0].add_ssrc(4);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid4", "Transport2", mid4_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  // This should not crash (https://crbug.com/1361612).
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
  auto outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
  EXPECT_EQ(inbound_rtps.size(), 4u);
  EXPECT_EQ(outbound_rtps.size(), 4u);
}

// These SSRC collisions are illegal, so it is not clear if this setup can
// happen even when talking to a malicious endpoint, but simulate illegal SSRC
// collisions just to make sure we don't crash in even the most extreme cases.
TEST_P(RTCStatsCollectorTest, InvalidSsrcCollisionDoesNotCrash) {
  // One SSRC to rule them all.
  VoiceMediaInfo mid1_info;
  mid1_info.receivers.emplace_back();
  mid1_info.receivers[0].add_ssrc(1);
  mid1_info.senders.emplace_back();
  mid1_info.senders[0].add_ssrc(1);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("Mid1", "BundledTransport", mid1_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  VideoMediaInfo mid2_info;
  mid2_info.receivers.emplace_back();
  mid2_info.receivers[0].add_ssrc(1);
  mid2_info.senders.emplace_back();
  mid2_info.senders[0].add_ssrc(1);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid2", "BundledTransport", mid2_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  VoiceMediaInfo mid3_info;
  mid3_info.receivers.emplace_back();
  mid3_info.receivers[0].add_ssrc(1);
  mid3_info.senders.emplace_back();
  mid3_info.senders[0].add_ssrc(1);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("Mid3", "BundledTransport", mid3_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  VideoMediaInfo mid4_info;
  mid4_info.receivers.emplace_back();
  mid4_info.receivers[0].add_ssrc(1);
  mid4_info.senders.emplace_back();
  mid4_info.senders[0].add_ssrc(1);
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid4", "BundledTransport", mid4_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  // This should not crash (https://crbug.com/1361612).
  stats_->GetStatsReport(main_thread_);
  // Because this setup is illegal, there is no "right answer" to how the report
  // should look. We only care about not crashing.
}

TEST_P(RTCStatsCollectorTest, CollectRTCCodecStatsOnlyIfReferenced) {
  // Audio
  VoiceMediaInfo voice_media_info;

  RtpCodecParameters inbound_audio_codec;
  inbound_audio_codec.payload_type = 1;
  inbound_audio_codec.kind = MediaType::AUDIO;
  inbound_audio_codec.name = "opus";
  inbound_audio_codec.clock_rate = 1337;
  inbound_audio_codec.num_channels = 1;
  inbound_audio_codec.parameters = {{"minptime", "10"}, {"useinbandfec", "1"}};
  voice_media_info.receive_codecs.insert(
      std::make_pair(inbound_audio_codec.payload_type, inbound_audio_codec));

  RtpCodecParameters outbound_audio_codec;
  outbound_audio_codec.payload_type = 2;
  outbound_audio_codec.kind = MediaType::AUDIO;
  outbound_audio_codec.name = "isac";
  outbound_audio_codec.clock_rate = 1338;
  outbound_audio_codec.num_channels = 2;
  voice_media_info.send_codecs.insert(
      std::make_pair(outbound_audio_codec.payload_type, outbound_audio_codec));

  // Video
  VideoMediaInfo video_media_info;

  RtpCodecParameters inbound_video_codec;
  inbound_video_codec.payload_type = 3;
  inbound_video_codec.kind = MediaType::VIDEO;
  inbound_video_codec.name = "H264";
  inbound_video_codec.clock_rate = 1339;
  inbound_video_codec.parameters = {{"level-asymmetry-allowed", "1"},
                                    {"packetization-mode", "1"},
                                    {"profile-level-id", "42001f"}};
  video_media_info.receive_codecs.insert(
      std::make_pair(inbound_video_codec.payload_type, inbound_video_codec));

  RtpCodecParameters outbound_video_codec;
  outbound_video_codec.payload_type = 4;
  outbound_video_codec.kind = MediaType::VIDEO;
  outbound_video_codec.name = "VP8";
  outbound_video_codec.clock_rate = 1340;
  video_media_info.send_codecs.insert(
      std::make_pair(outbound_video_codec.payload_type, outbound_video_codec));

  // Ensure the above codecs are referenced.
  VoiceReceiverInfo inbound_audio_info;
  inbound_audio_info.add_ssrc(10);
  inbound_audio_info.codec_payload_type = 1;
  inbound_audio_info.packets_received = 123;
  voice_media_info.receivers.push_back(inbound_audio_info);

  VoiceSenderInfo outbound_audio_info;
  outbound_audio_info.add_ssrc(20);
  outbound_audio_info.codec_payload_type = 2;
  voice_media_info.senders.push_back(outbound_audio_info);

  VideoReceiverInfo inbound_video_info;
  inbound_video_info.add_ssrc(30);
  inbound_video_info.codec_payload_type = 3;
  inbound_video_info.packets_received = 123;
  video_media_info.receivers.push_back(inbound_video_info);

  VideoSenderInfo outbound_video_info;
  outbound_video_info.add_ssrc(40);
  outbound_video_info.codec_payload_type = 4;
  video_media_info.senders.push_back(outbound_video_info);

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  auto audio_channels =
      pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
  auto video_channels =
      pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCCodecStats expected_inbound_audio_codec(
      "CITTransportName1_1_minptime=10;useinbandfec=1", report->timestamp());
  expected_inbound_audio_codec.payload_type = 1;
  expected_inbound_audio_codec.mime_type = "audio/opus";
  expected_inbound_audio_codec.clock_rate = 1337;
  expected_inbound_audio_codec.channels = 1;
  expected_inbound_audio_codec.sdp_fmtp_line = "minptime=10;useinbandfec=1";
  expected_inbound_audio_codec.transport_id = "TTransportName1";

  RTCCodecStats expected_outbound_audio_codec("COTTransportName1_2",
                                              report->timestamp());
  expected_outbound_audio_codec.payload_type = 2;
  expected_outbound_audio_codec.mime_type = "audio/isac";
  expected_outbound_audio_codec.clock_rate = 1338;
  expected_outbound_audio_codec.channels = 2;
  expected_outbound_audio_codec.transport_id = "TTransportName1";

  RTCCodecStats expected_inbound_video_codec(
      "CITTransportName1_3_level-asymmetry-allowed=1;"
      "packetization-mode=1;profile-level-id=42001f",
      report->timestamp());
  expected_inbound_video_codec.payload_type = 3;
  expected_inbound_video_codec.mime_type = "video/H264";
  expected_inbound_video_codec.clock_rate = 1339;
  expected_inbound_video_codec.sdp_fmtp_line =
      "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f";
  expected_inbound_video_codec.transport_id = "TTransportName1";

  RTCCodecStats expected_outbound_video_codec("COTTransportName1_4",
                                              report->timestamp());
  expected_outbound_video_codec.payload_type = 4;
  expected_outbound_video_codec.mime_type = "video/VP8";
  expected_outbound_video_codec.clock_rate = 1340;
  expected_outbound_video_codec.transport_id = "TTransportName1";

  ASSERT_TRUE(report->Get(expected_inbound_audio_codec.id()));
  EXPECT_EQ(
      expected_inbound_audio_codec,
      report->Get(expected_inbound_audio_codec.id())->cast_to<RTCCodecStats>());

  ASSERT_TRUE(report->Get(expected_outbound_audio_codec.id()));
  EXPECT_EQ(expected_outbound_audio_codec,
            report->Get(expected_outbound_audio_codec.id())
                ->cast_to<RTCCodecStats>());

  ASSERT_TRUE(report->Get(expected_inbound_video_codec.id()));
  EXPECT_EQ(
      expected_inbound_video_codec,
      report->Get(expected_inbound_video_codec.id())->cast_to<RTCCodecStats>());

  ASSERT_TRUE(report->Get(expected_outbound_video_codec.id()));
  EXPECT_EQ(expected_outbound_video_codec,
            report->Get(expected_outbound_video_codec.id())
                ->cast_to<RTCCodecStats>());

  // Now remove all the RTP streams such that there are no live codecId
  // references to the codecs, this should result in none of the RTCCodecStats
  // being exposed, despite `send_codecs` and `receive_codecs` still being set.
  voice_media_info.senders.clear();
  voice_media_info.receivers.clear();
  audio_channels.first->SetStats(voice_media_info);
  audio_channels.second->SetStats(voice_media_info);
  video_media_info.senders.clear();
  video_media_info.receivers.clear();
  video_channels.first->SetStats(video_media_info);
  video_channels.second->SetStats(video_media_info);
  stats_->stats_collector().ClearCachedStatsReport();
  report = stats_->GetStatsReport(main_thread_);
  EXPECT_FALSE(report->Get(expected_inbound_audio_codec.id()));
  EXPECT_FALSE(report->Get(expected_outbound_audio_codec.id()));
  EXPECT_FALSE(report->Get(expected_inbound_video_codec.id()));
  EXPECT_FALSE(report->Get(expected_outbound_video_codec.id()));
}

TEST_P(RTCStatsCollectorTest, CodecStatsAreCollectedPerTransport) {
  // PT=10
  RtpCodecParameters outbound_codec_pt10;
  outbound_codec_pt10.payload_type = 10;
  outbound_codec_pt10.kind = MediaType::VIDEO;
  outbound_codec_pt10.name = "VP8";
  outbound_codec_pt10.clock_rate = 9000;

  // PT=11
  RtpCodecParameters outbound_codec_pt11;
  outbound_codec_pt11.payload_type = 11;
  outbound_codec_pt11.kind = MediaType::VIDEO;
  outbound_codec_pt11.name = "VP8";
  outbound_codec_pt11.clock_rate = 9000;

  // Insert codecs into `send_codecs` and ensure the PTs are referenced by RTP
  // streams.
  VideoMediaInfo info_pt10;
  info_pt10.send_codecs.insert(
      std::make_pair(outbound_codec_pt10.payload_type, outbound_codec_pt10));
  info_pt10.senders.emplace_back();
  info_pt10.senders[0].add_ssrc(42);
  info_pt10.senders[0].codec_payload_type = outbound_codec_pt10.payload_type;

  VideoMediaInfo info_pt11;
  info_pt11.send_codecs.insert(
      std::make_pair(outbound_codec_pt11.payload_type, outbound_codec_pt11));
  info_pt11.senders.emplace_back();
  info_pt11.senders[0].add_ssrc(43);
  info_pt11.senders[0].codec_payload_type = outbound_codec_pt11.payload_type;

  VideoMediaInfo info_pt10_pt11;
  info_pt10_pt11.send_codecs.insert(
      std::make_pair(outbound_codec_pt10.payload_type, outbound_codec_pt10));
  info_pt10_pt11.send_codecs.insert(
      std::make_pair(outbound_codec_pt11.payload_type, outbound_codec_pt11));
  info_pt10_pt11.senders.emplace_back();
  info_pt10_pt11.senders[0].add_ssrc(44);
  info_pt10_pt11.senders[0].codec_payload_type =
      outbound_codec_pt10.payload_type;
  info_pt10_pt11.senders.emplace_back();
  info_pt10_pt11.senders[1].add_ssrc(45);
  info_pt10_pt11.senders[1].codec_payload_type =
      outbound_codec_pt11.payload_type;

  // First two mids contain subsets, the third one contains all PTs.
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid1", "FirstTransport", info_pt10);
  pc_->AddVideoChannel("Mid2", "FirstTransport", info_pt11);
  pc_->AddVideoChannel("Mid3", "FirstTransport", info_pt10_pt11);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  // There should be no duplicate codecs because all codec references are on the
  // same transport.
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  auto codec_stats = report->GetStatsOfType<RTCCodecStats>();
  EXPECT_EQ(codec_stats.size(), 2u);

  // If a second transport is added with the same PT information, this does
  // count as different codec objects.
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid4", "SecondTransport", info_pt10_pt11);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->stats_collector().ClearCachedStatsReport();
  report = stats_->GetStatsReport(main_thread_);
  codec_stats = report->GetStatsOfType<RTCCodecStats>();
  EXPECT_EQ(codec_stats.size(), 4u);
}

TEST_P(RTCStatsCollectorTest, SamePayloadTypeButDifferentFmtpLines) {
  // PT=111, useinbandfec=0
  RtpCodecParameters inbound_codec_pt111_nofec;
  inbound_codec_pt111_nofec.payload_type = 111;
  inbound_codec_pt111_nofec.kind = MediaType::AUDIO;
  inbound_codec_pt111_nofec.name = "opus";
  inbound_codec_pt111_nofec.clock_rate = 48000;
  inbound_codec_pt111_nofec.parameters.insert(
      std::make_pair("useinbandfec", "0"));

  // PT=111, useinbandfec=1
  RtpCodecParameters inbound_codec_pt111_fec;
  inbound_codec_pt111_fec.payload_type = 111;
  inbound_codec_pt111_fec.kind = MediaType::AUDIO;
  inbound_codec_pt111_fec.name = "opus";
  inbound_codec_pt111_fec.clock_rate = 48000;
  inbound_codec_pt111_fec.parameters.insert(
      std::make_pair("useinbandfec", "1"));

  VideoMediaInfo info_nofec;
  info_nofec.receive_codecs.insert(std::make_pair(
      inbound_codec_pt111_nofec.payload_type, inbound_codec_pt111_nofec));
  info_nofec.receivers.emplace_back();
  info_nofec.receivers[0].add_ssrc(123);
  info_nofec.receivers[0].packets_received = 123;
  info_nofec.receivers[0].codec_payload_type =
      inbound_codec_pt111_nofec.payload_type;
  VideoMediaInfo info_fec;
  info_fec.receive_codecs.insert(std::make_pair(
      inbound_codec_pt111_fec.payload_type, inbound_codec_pt111_fec));
  info_fec.receivers.emplace_back();
  info_fec.receivers[0].add_ssrc(321);
  info_fec.receivers[0].packets_received = 123;
  info_fec.receivers[0].codec_payload_type =
      inbound_codec_pt111_fec.payload_type;

  // First two mids contain subsets, the third one contains all PTs.
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid1", "BundledTransport", info_nofec);
  pc_->AddVideoChannel("Mid2", "BundledTransport", info_fec);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  // Despite having the same PT we should see two codec stats because their FMTP
  // lines are different.
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  auto codec_stats = report->GetStatsOfType<RTCCodecStats>();
  EXPECT_EQ(codec_stats.size(), 2u);

  // Ensure SSRC uniqueness before the next AddVideoChannel() call. SSRCs need
  // to be unique on different m= sections when using BUNDLE.
  info_nofec.receivers[0].local_stats[0].ssrc = 12;
  info_fec.receivers[0].local_stats[0].ssrc = 21;
  // Adding more m= sections that does have the same FMTP lines does not result
  // in duplicates.
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid3", "BundledTransport", info_nofec);
  pc_->AddVideoChannel("Mid4", "BundledTransport", info_fec);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->stats_collector().ClearCachedStatsReport();
  report = stats_->GetStatsReport(main_thread_);
  codec_stats = report->GetStatsOfType<RTCCodecStats>();
  EXPECT_EQ(codec_stats.size(), 2u);

  // Same FMTP line but a different PT does count as a new codec.
  // PT=112, useinbandfec=1
  RtpCodecParameters inbound_codec_pt112_fec;
  inbound_codec_pt112_fec.payload_type = 112;
  inbound_codec_pt112_fec.kind = MediaType::AUDIO;
  inbound_codec_pt112_fec.name = "opus";
  inbound_codec_pt112_fec.clock_rate = 48000;
  inbound_codec_pt112_fec.parameters.insert(
      std::make_pair("useinbandfec", "1"));
  VideoMediaInfo info_fec_pt112;
  info_fec_pt112.receive_codecs.insert(std::make_pair(
      inbound_codec_pt112_fec.payload_type, inbound_codec_pt112_fec));
  info_fec_pt112.receivers.emplace_back();
  info_fec_pt112.receivers[0].add_ssrc(112);
  info_fec_pt112.receivers[0].packets_received = 123;
  info_fec_pt112.receivers[0].codec_payload_type =
      inbound_codec_pt112_fec.payload_type;
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid5", "BundledTransport", info_fec_pt112);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->stats_collector().ClearCachedStatsReport();
  report = stats_->GetStatsReport(main_thread_);
  codec_stats = report->GetStatsOfType<RTCCodecStats>();
  EXPECT_EQ(codec_stats.size(), 3u);
}

TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
  const char kAudioTransport[] = "audio";
  const char kVideoTransport[] = "video";

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("audio", kAudioTransport);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  std::unique_ptr<CertificateInfo> audio_local_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          std::vector<std::string>({"(local) audio"}));
  pc_->SetLocalCertificate(kAudioTransport, audio_local_certinfo->certificate);
  std::unique_ptr<CertificateInfo> audio_remote_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          std::vector<std::string>({"(remote) audio"}));
  pc_->SetRemoteCertChain(
      kAudioTransport,
      audio_remote_certinfo->certificate->GetSSLCertificateChain().Clone());

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("video", kVideoTransport);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  std::unique_ptr<CertificateInfo> video_local_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          std::vector<std::string>({"(local) video"}));
  pc_->SetLocalCertificate(kVideoTransport, video_local_certinfo->certificate);
  std::unique_ptr<CertificateInfo> video_remote_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          std::vector<std::string>({"(remote) video"}));
  pc_->SetRemoteCertChain(
      kVideoTransport,
      video_remote_certinfo->certificate->GetSSLCertificateChain().Clone());

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  ExpectReportContainsCertificateInfo(report, *audio_local_certinfo);
  ExpectReportContainsCertificateInfo(report, *audio_remote_certinfo);
  ExpectReportContainsCertificateInfo(report, *video_local_certinfo);
  ExpectReportContainsCertificateInfo(report, *video_remote_certinfo);
}

TEST_P(RTCStatsCollectorTest, CollectRTCCertificateStatsChain) {
  const char kTransportName[] = "transport";

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("audio", kTransportName);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  std::unique_ptr<CertificateInfo> local_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          {"(local) this", "(local) is", "(local) a", "(local) chain"});
  pc_->SetLocalCertificate(kTransportName, local_certinfo->certificate);

  std::unique_ptr<CertificateInfo> remote_certinfo =
      CreateFakeCertificateAndInfoFromDers({"(remote) this", "(remote) is",
                                            "(remote) another",
                                            "(remote) chain"});
  pc_->SetRemoteCertChain(
      kTransportName,
      remote_certinfo->certificate->GetSSLCertificateChain().Clone());

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  ExpectReportContainsCertificateInfo(report, *local_certinfo);
  ExpectReportContainsCertificateInfo(report, *remote_certinfo);
}

TEST_P(RTCStatsCollectorTest, CertificateStatsCache) {
  const char kTransportName[] = "transport";

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("audio", kTransportName);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  // Set local and remote cerificates.
  std::unique_ptr<CertificateInfo> initial_local_certinfo =
      CreateFakeCertificateAndInfoFromDers({"LocalCertA", "LocalCertB"});
  pc_->SetLocalCertificate(kTransportName, initial_local_certinfo->certificate);
  std::unique_ptr<CertificateInfo> initial_remote_certinfo =
      CreateFakeCertificateAndInfoFromDers({"RemoteCertA", "RemoteCertB"});
  pc_->SetRemoteCertChain(
      kTransportName,
      initial_remote_certinfo->certificate->GetSSLCertificateChain().Clone());
  ASSERT_EQ(initial_local_certinfo->fingerprints.size(), 2u);
  ASSERT_EQ(initial_remote_certinfo->fingerprints.size(), 2u);

  scoped_refptr<const RTCStatsReport> first_report =
      stats_->GetStatsReport(main_thread_);
  const auto* first_local_cert0 = GetCertificateStatsFromFingerprint(
      first_report, initial_local_certinfo->fingerprints[0]);
  const auto* first_local_cert1 = GetCertificateStatsFromFingerprint(
      first_report, initial_local_certinfo->fingerprints[1]);
  const auto* first_remote_cert0 = GetCertificateStatsFromFingerprint(
      first_report, initial_remote_certinfo->fingerprints[0]);
  const auto* first_remote_cert1 = GetCertificateStatsFromFingerprint(
      first_report, initial_remote_certinfo->fingerprints[1]);
  ASSERT_TRUE(first_local_cert0);
  ASSERT_TRUE(first_local_cert1);
  ASSERT_TRUE(first_remote_cert0);
  ASSERT_TRUE(first_remote_cert1);
  EXPECT_EQ(first_local_cert0->timestamp(), env_.clock().CurrentTime());
  EXPECT_EQ(first_local_cert1->timestamp(), env_.clock().CurrentTime());
  EXPECT_EQ(first_remote_cert0->timestamp(), env_.clock().CurrentTime());
  EXPECT_EQ(first_remote_cert1->timestamp(), env_.clock().CurrentTime());

  // Replace all certificates.
  std::unique_ptr<CertificateInfo> updated_local_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          {"UpdatedLocalCertA", "UpdatedLocalCertB"});
  pc_->SetLocalCertificate(kTransportName, updated_local_certinfo->certificate);
  std::unique_ptr<CertificateInfo> updated_remote_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          {"UpdatedRemoteCertA", "UpdatedRemoteCertB"});
  pc_->SetRemoteCertChain(
      kTransportName,
      updated_remote_certinfo->certificate->GetSSLCertificateChain().Clone());
  // This test assumes fingerprints are different for the old and new
  // certificates.
  EXPECT_NE(initial_local_certinfo->fingerprints,
            updated_local_certinfo->fingerprints);
  EXPECT_NE(initial_remote_certinfo->fingerprints,
            updated_remote_certinfo->fingerprints);

  // Advance time to ensure a fresh stats report, but don't clear the
  // certificate stats cache.
  time_controller_.AdvanceTime(TimeDelta::Seconds(1));
  scoped_refptr<const RTCStatsReport> second_report =
      stats_->GetStatsReport(main_thread_);
  // We expect to see the same certificates as before due to not clearing the
  // certificate cache.
  const auto* second_local_cert0 =
      second_report->GetAs<RTCCertificateStats>(first_local_cert0->id());
  const auto* second_local_cert1 =
      second_report->GetAs<RTCCertificateStats>(first_local_cert1->id());
  const auto* second_remote_cert0 =
      second_report->GetAs<RTCCertificateStats>(first_remote_cert0->id());
  const auto* second_remote_cert1 =
      second_report->GetAs<RTCCertificateStats>(first_remote_cert1->id());
  ASSERT_TRUE(second_local_cert0);
  ASSERT_TRUE(second_local_cert1);
  ASSERT_TRUE(second_remote_cert0);
  ASSERT_TRUE(second_remote_cert1);
  // The information in the certificate stats are obsolete.
  EXPECT_EQ(*second_local_cert0->fingerprint,
            initial_local_certinfo->fingerprints[0]);
  EXPECT_EQ(*second_local_cert1->fingerprint,
            initial_local_certinfo->fingerprints[1]);
  EXPECT_EQ(*second_remote_cert0->fingerprint,
            initial_remote_certinfo->fingerprints[0]);
  EXPECT_EQ(*second_remote_cert1->fingerprint,
            initial_remote_certinfo->fingerprints[1]);
  // But timestamps are up-to-date, because this is a fresh stats report.
  EXPECT_EQ(second_local_cert0->timestamp(), env_.clock().CurrentTime());
  EXPECT_EQ(second_local_cert1->timestamp(), env_.clock().CurrentTime());
  EXPECT_EQ(second_remote_cert0->timestamp(), env_.clock().CurrentTime());
  EXPECT_EQ(second_remote_cert1->timestamp(), env_.clock().CurrentTime());
  // The updated certificates are not part of the report yet.
  EXPECT_FALSE(GetCertificateStatsFromFingerprint(
      second_report, updated_local_certinfo->fingerprints[0]));
  EXPECT_FALSE(GetCertificateStatsFromFingerprint(
      second_report, updated_local_certinfo->fingerprints[1]));
  EXPECT_FALSE(GetCertificateStatsFromFingerprint(
      second_report, updated_remote_certinfo->fingerprints[0]));
  EXPECT_FALSE(GetCertificateStatsFromFingerprint(
      second_report, updated_remote_certinfo->fingerprints[1]));

  // Clear the cache, including the cached certificates.
  stats_->stats_collector().ClearCachedStatsReport();
  scoped_refptr<const RTCStatsReport> third_report =
      stats_->GetStatsReport(main_thread_);
  // Now the old certificates stats should be deleted.
  EXPECT_FALSE(third_report->Get(first_local_cert0->id()));
  EXPECT_FALSE(third_report->Get(first_local_cert1->id()));
  EXPECT_FALSE(third_report->Get(first_remote_cert0->id()));
  EXPECT_FALSE(third_report->Get(first_remote_cert1->id()));
  // And updated certificates exist.
  EXPECT_TRUE(GetCertificateStatsFromFingerprint(
      third_report, updated_local_certinfo->fingerprints[0]));
  EXPECT_TRUE(GetCertificateStatsFromFingerprint(
      third_report, updated_local_certinfo->fingerprints[1]));
  EXPECT_TRUE(GetCertificateStatsFromFingerprint(
      third_report, updated_remote_certinfo->fingerprints[0]));
  EXPECT_TRUE(GetCertificateStatsFromFingerprint(
      third_report, updated_remote_certinfo->fingerprints[1]));
}

TEST_P(RTCStatsCollectorTest, CollectTwoRTCDataChannelStatsWithPendingId) {
  // Note: The test assumes data channel IDs are predictable.
  // This is not a safe assumption, but in order to make it work for
  // the test, we reset the ID allocator at test start.
  SctpDataChannel::ResetInternalIdAllocatorForTesting(-1);
  pc_->AddSctpDataChannel(make_ref_counted<MockSctpDataChannel>(
      data_channel_controller_->weak_ptr(), /*id=*/-1,
      DataChannelInterface::kConnecting));
  pc_->AddSctpDataChannel(make_ref_counted<MockSctpDataChannel>(
      data_channel_controller_->weak_ptr(), /*id=*/-1,
      DataChannelInterface::kConnecting));

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  RTCDataChannelStats expected_data_channel0("D0", Timestamp::Zero());
  // Default values from MockDataChannel.
  expected_data_channel0.label = "MockSctpDataChannel";
  expected_data_channel0.protocol = "someProtocol";
  expected_data_channel0.state = "connecting";
  expected_data_channel0.messages_sent = 0;
  expected_data_channel0.bytes_sent = 0;
  expected_data_channel0.messages_received = 0;
  expected_data_channel0.bytes_received = 0;

  ASSERT_TRUE(report->Get(expected_data_channel0.id()));
  EXPECT_EQ(
      expected_data_channel0,
      report->Get(expected_data_channel0.id())->cast_to<RTCDataChannelStats>());
}

TEST_P(RTCStatsCollectorTest, CollectRTCDataChannelStats) {
  // Note: The test assumes data channel IDs are predictable.
  // This is not a safe assumption, but in order to make it work for
  // the test, we reset the ID allocator at test start.
  SctpDataChannel::ResetInternalIdAllocatorForTesting(-1);
  pc_->AddSctpDataChannel(make_ref_counted<MockSctpDataChannel>(
      data_channel_controller_->weak_ptr(), 0, "MockSctpDataChannel0",
      DataChannelInterface::kConnecting, "proto1", 1, 2, 3, 4));
  RTCDataChannelStats expected_data_channel0("D0", Timestamp::Zero());
  expected_data_channel0.label = "MockSctpDataChannel0";
  expected_data_channel0.protocol = "proto1";
  expected_data_channel0.data_channel_identifier = 0;
  expected_data_channel0.state = "connecting";
  expected_data_channel0.messages_sent = 1;
  expected_data_channel0.bytes_sent = 2;
  expected_data_channel0.messages_received = 3;
  expected_data_channel0.bytes_received = 4;

  pc_->AddSctpDataChannel(make_ref_counted<MockSctpDataChannel>(
      data_channel_controller_->weak_ptr(), 1, "MockSctpDataChannel1",
      DataChannelInterface::kOpen, "proto2", 5, 6, 7, 8));
  RTCDataChannelStats expected_data_channel1("D1", Timestamp::Zero());
  expected_data_channel1.label = "MockSctpDataChannel1";
  expected_data_channel1.protocol = "proto2";
  expected_data_channel1.data_channel_identifier = 1;
  expected_data_channel1.state = "open";
  expected_data_channel1.messages_sent = 5;
  expected_data_channel1.bytes_sent = 6;
  expected_data_channel1.messages_received = 7;
  expected_data_channel1.bytes_received = 8;

  pc_->AddSctpDataChannel(make_ref_counted<MockSctpDataChannel>(
      data_channel_controller_->weak_ptr(), 2, "MockSctpDataChannel2",
      DataChannelInterface::kClosing, "proto1", 9, 10, 11, 12));
  RTCDataChannelStats expected_data_channel2("D2", Timestamp::Zero());
  expected_data_channel2.label = "MockSctpDataChannel2";
  expected_data_channel2.protocol = "proto1";
  expected_data_channel2.data_channel_identifier = 2;
  expected_data_channel2.state = "closing";
  expected_data_channel2.messages_sent = 9;
  expected_data_channel2.bytes_sent = 10;
  expected_data_channel2.messages_received = 11;
  expected_data_channel2.bytes_received = 12;

  pc_->AddSctpDataChannel(make_ref_counted<MockSctpDataChannel>(
      data_channel_controller_->weak_ptr(), 3, "MockSctpDataChannel3",
      DataChannelInterface::kClosed, "proto3", 13, 14, 15, 16));
  RTCDataChannelStats expected_data_channel3("D3", Timestamp::Zero());
  expected_data_channel3.label = "MockSctpDataChannel3";
  expected_data_channel3.protocol = "proto3";
  expected_data_channel3.data_channel_identifier = 3;
  expected_data_channel3.state = "closed";
  expected_data_channel3.messages_sent = 13;
  expected_data_channel3.bytes_sent = 14;
  expected_data_channel3.messages_received = 15;
  expected_data_channel3.bytes_received = 16;

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  ASSERT_TRUE(report->Get(expected_data_channel0.id()));
  EXPECT_EQ(
      expected_data_channel0,
      report->Get(expected_data_channel0.id())->cast_to<RTCDataChannelStats>());
  ASSERT_TRUE(report->Get(expected_data_channel1.id()));
  EXPECT_EQ(
      expected_data_channel1,
      report->Get(expected_data_channel1.id())->cast_to<RTCDataChannelStats>());
  ASSERT_TRUE(report->Get(expected_data_channel2.id()));
  EXPECT_EQ(
      expected_data_channel2,
      report->Get(expected_data_channel2.id())->cast_to<RTCDataChannelStats>());
  ASSERT_TRUE(report->Get(expected_data_channel3.id()));
  EXPECT_EQ(
      expected_data_channel3,
      report->Get(expected_data_channel3.id())->cast_to<RTCDataChannelStats>());
}

TEST_P(RTCStatsCollectorTest, CollectRTCIceCandidateStats) {
  // Candidates in the first transport stats.
  std::unique_ptr<Candidate> a_local_host = CreateFakeCandidate(
      "1.2.3.4", 5, "a_local_host's protocol", ADAPTER_TYPE_VPN,
      IceCandidateType::kHost, 0, ADAPTER_TYPE_ETHERNET);
  RTCLocalIceCandidateStats expected_a_local_host("I" + a_local_host->id(),
                                                  Timestamp::Zero());
  expected_a_local_host.transport_id = "Ta0";
  expected_a_local_host.network_type = "vpn";
  expected_a_local_host.ip = "1.2.3.4";
  expected_a_local_host.address = "1.2.3.4";
  expected_a_local_host.port = 5;
  expected_a_local_host.protocol = "a_local_host's protocol";
  expected_a_local_host.candidate_type = "host";
  expected_a_local_host.priority = 0;
  expected_a_local_host.vpn = true;
  expected_a_local_host.network_adapter_type = "ethernet";
  expected_a_local_host.foundation = "foundationIsAString";
  expected_a_local_host.username_fragment = "iceusernamefragment";

  std::unique_ptr<Candidate> a_local_tcp_slice =
      CreateFakeCandidate("1.2.3.4", 5, "tcp", ADAPTER_TYPE_VPN,
                          IceCandidateType::kHost, 0, ADAPTER_TYPE_ETHERNET);
  a_local_tcp_slice->set_network_slice(NetworkSlice::UNIFIED_COMMUNICATIONS);
  a_local_tcp_slice->set_tcptype("a_local_tcp_slice's tcptype");
  RTCLocalIceCandidateStats expected_a_local_tcp_slice(
      "I" + a_local_tcp_slice->id(), Timestamp::Zero());
  expected_a_local_tcp_slice.transport_id = "Ta0";
  expected_a_local_tcp_slice.network_type = "vpn";
  expected_a_local_tcp_slice.ip = "1.2.3.4";
  expected_a_local_tcp_slice.address = "1.2.3.4";
  expected_a_local_tcp_slice.port = 5;
  expected_a_local_tcp_slice.protocol = "tcp";
  expected_a_local_tcp_slice.candidate_type = "host";
  expected_a_local_tcp_slice.priority = 0;
  expected_a_local_tcp_slice.vpn = true;
  expected_a_local_tcp_slice.network_adapter_type = "ethernet";
  expected_a_local_tcp_slice.foundation = "foundationIsAString";
  expected_a_local_tcp_slice.username_fragment = "iceusernamefragment";
  expected_a_local_tcp_slice.tcp_type = "a_local_tcp_slice's tcptype";
  expected_a_local_tcp_slice.network_slice = "unified-communications";

  std::unique_ptr<Candidate> a_remote_srflx =
      CreateFakeCandidate("6.7.8.9", 10, "remote_srflx's protocol",
                          ADAPTER_TYPE_UNKNOWN, IceCandidateType::kSrflx, 1);
  RTCRemoteIceCandidateStats expected_a_remote_srflx("I" + a_remote_srflx->id(),
                                                     Timestamp::Zero());
  expected_a_remote_srflx.transport_id = "Ta0";
  expected_a_remote_srflx.ip = "6.7.8.9";
  expected_a_remote_srflx.address = "6.7.8.9";
  expected_a_remote_srflx.port = 10;
  expected_a_remote_srflx.protocol = "remote_srflx's protocol";
  expected_a_remote_srflx.candidate_type = "srflx";
  expected_a_remote_srflx.priority = 1;
  expected_a_remote_srflx.foundation = "foundationIsAString";
  expected_a_remote_srflx.username_fragment = "iceusernamefragment";

  std::unique_ptr<Candidate> a_local_prflx = CreateFakeCandidate(
      "11.12.13.14", 15, "a_local_prflx's protocol", ADAPTER_TYPE_CELLULAR_2G,
      IceCandidateType::kPrflx, 2);
  RTCLocalIceCandidateStats expected_a_local_prflx("I" + a_local_prflx->id(),
                                                   Timestamp::Zero());
  expected_a_local_prflx.transport_id = "Ta0";
  expected_a_local_prflx.network_type = "cellular";
  expected_a_local_prflx.ip = "11.12.13.14";
  expected_a_local_prflx.address = "11.12.13.14";
  expected_a_local_prflx.port = 15;
  expected_a_local_prflx.protocol = "a_local_prflx's protocol";
  expected_a_local_prflx.candidate_type = "prflx";
  expected_a_local_prflx.priority = 2;
  expected_a_local_prflx.vpn = false;
  expected_a_local_prflx.network_adapter_type = "cellular2g";
  expected_a_local_prflx.foundation = "foundationIsAString";
  expected_a_local_prflx.username_fragment = "iceusernamefragment";

  std::unique_ptr<Candidate> a_remote_relay =
      CreateFakeCandidate("16.17.18.19", 20, "a_remote_relay's protocol",
                          ADAPTER_TYPE_UNKNOWN, IceCandidateType::kRelay, 3);
  RTCRemoteIceCandidateStats expected_a_remote_relay("I" + a_remote_relay->id(),
                                                     Timestamp::Zero());
  expected_a_remote_relay.transport_id = "Ta0";
  expected_a_remote_relay.ip = "16.17.18.19";
  expected_a_remote_relay.address = "16.17.18.19";
  expected_a_remote_relay.port = 20;
  expected_a_remote_relay.protocol = "a_remote_relay's protocol";
  expected_a_remote_relay.candidate_type = "relay";
  expected_a_remote_relay.priority = 3;
  expected_a_remote_relay.foundation = "foundationIsAString";
  expected_a_remote_relay.username_fragment = "iceusernamefragment";

  std::unique_ptr<Candidate> a_local_relay =
      CreateFakeCandidate("16.17.18.19", 21, "a_local_relay's protocol",
                          ADAPTER_TYPE_UNKNOWN, IceCandidateType::kRelay, 1);
  a_local_relay->set_relay_protocol("tcp");
  a_local_relay->set_url("turn:url1");

  RTCLocalIceCandidateStats expected_a_local_relay("I" + a_local_relay->id(),
                                                   Timestamp::Zero());
  expected_a_local_relay.transport_id = "Ta0";
  expected_a_local_relay.network_type = "unknown";
  expected_a_local_relay.ip = "16.17.18.19";
  expected_a_local_relay.address = "16.17.18.19";
  expected_a_local_relay.port = 21;
  expected_a_local_relay.protocol = "a_local_relay's protocol";
  expected_a_local_relay.relay_protocol = "tcp";
  expected_a_local_relay.candidate_type = "relay";
  expected_a_local_relay.priority = 1;
  expected_a_local_relay.url = "turn:url1";
  expected_a_local_relay.vpn = false;
  expected_a_local_relay.network_adapter_type = "unknown";
  expected_a_local_relay.foundation = "foundationIsAString";
  expected_a_local_relay.username_fragment = "iceusernamefragment";

  std::unique_ptr<Candidate> a_local_relay_prflx =
      CreateFakeCandidate("11.12.13.20", 22, "a_local_relay_prflx's protocol",
                          ADAPTER_TYPE_UNKNOWN, IceCandidateType::kPrflx, 1);
  a_local_relay_prflx->set_relay_protocol("udp");

  RTCLocalIceCandidateStats expected_a_local_relay_prflx(
      "I" + a_local_relay_prflx->id(), Timestamp::Zero());
  expected_a_local_relay_prflx.transport_id = "Ta0";
  expected_a_local_relay_prflx.network_type = "unknown";
  expected_a_local_relay_prflx.ip = "11.12.13.20";
  expected_a_local_relay_prflx.address = "11.12.13.20";
  expected_a_local_relay_prflx.port = 22;
  expected_a_local_relay_prflx.protocol = "a_local_relay_prflx's protocol";
  expected_a_local_relay_prflx.relay_protocol = "udp";
  expected_a_local_relay_prflx.candidate_type = "prflx";
  expected_a_local_relay_prflx.priority = 1;
  expected_a_local_relay_prflx.vpn = false;
  expected_a_local_relay_prflx.network_adapter_type = "unknown";
  expected_a_local_relay_prflx.foundation = "foundationIsAString";
  expected_a_local_relay_prflx.username_fragment = "iceusernamefragment";

  // A non-paired local candidate.
  std::unique_ptr<Candidate> a_local_host_not_paired = CreateFakeCandidate(
      "1.2.3.4", 4404, "a_local_host_not_paired's protocol", ADAPTER_TYPE_VPN,
      IceCandidateType::kHost, 0, ADAPTER_TYPE_ETHERNET);
  RTCLocalIceCandidateStats expected_a_local_host_not_paired(
      "I" + a_local_host_not_paired->id(), Timestamp::Zero());
  expected_a_local_host_not_paired.transport_id = "Ta0";
  expected_a_local_host_not_paired.network_type = "vpn";
  expected_a_local_host_not_paired.ip = "1.2.3.4";
  expected_a_local_host_not_paired.address = "1.2.3.4";
  expected_a_local_host_not_paired.port = 4404;
  expected_a_local_host_not_paired.protocol =
      "a_local_host_not_paired's protocol";
  expected_a_local_host_not_paired.candidate_type = "host";
  expected_a_local_host_not_paired.priority = 0;
  expected_a_local_host_not_paired.vpn = true;
  expected_a_local_host_not_paired.network_adapter_type = "ethernet";
  expected_a_local_host_not_paired.foundation = "foundationIsAString";
  expected_a_local_host_not_paired.username_fragment = "iceusernamefragment";

  // Candidates in the second transport stats.
  std::unique_ptr<Candidate> b_local =
      CreateFakeCandidate("42.42.42.42", 42, "b_local's protocol",
                          ADAPTER_TYPE_WIFI, IceCandidateType::kHost, 42);
  RTCLocalIceCandidateStats expected_b_local("I" + b_local->id(),
                                             Timestamp::Zero());
  expected_b_local.transport_id = "Tb0";
  expected_b_local.network_type = "wifi";
  expected_b_local.ip = "42.42.42.42";
  expected_b_local.address = "42.42.42.42";
  expected_b_local.port = 42;
  expected_b_local.protocol = "b_local's protocol";
  expected_b_local.candidate_type = "host";
  expected_b_local.priority = 42;
  expected_b_local.vpn = false;
  expected_b_local.network_adapter_type = "wifi";
  expected_b_local.foundation = "foundationIsAString";
  expected_b_local.username_fragment = "iceusernamefragment";

  std::unique_ptr<Candidate> b_remote =
      CreateFakeCandidate("42.42.42.42", 42, "b_remote's protocol",
                          ADAPTER_TYPE_UNKNOWN, IceCandidateType::kHost, 42);
  RTCRemoteIceCandidateStats expected_b_remote("I" + b_remote->id(),
                                               Timestamp::Zero());
  expected_b_remote.transport_id = "Tb0";
  expected_b_remote.ip = "42.42.42.42";
  expected_b_remote.address = "42.42.42.42";
  expected_b_remote.port = 42;
  expected_b_remote.protocol = "b_remote's protocol";
  expected_b_remote.candidate_type = "host";
  expected_b_remote.priority = 42;
  expected_b_remote.foundation = "foundationIsAString";
  expected_b_remote.username_fragment = "iceusernamefragment";

  // Add candidate pairs to connection.
  TransportChannelStats a_transport_channel_stats;
  a_transport_channel_stats.ice_transport_stats.connection_infos.push_back(
      ConnectionInfo());
  a_transport_channel_stats.ice_transport_stats.connection_infos[0]
      .local_candidate = *a_local_host;
  a_transport_channel_stats.ice_transport_stats.connection_infos[0]
      .remote_candidate = *a_remote_srflx;
  a_transport_channel_stats.ice_transport_stats.connection_infos.push_back(
      ConnectionInfo());
  a_transport_channel_stats.ice_transport_stats.connection_infos[1]
      .local_candidate = *a_local_prflx;
  a_transport_channel_stats.ice_transport_stats.connection_infos[1]
      .remote_candidate = *a_remote_relay;
  a_transport_channel_stats.ice_transport_stats.connection_infos.push_back(
      ConnectionInfo());
  a_transport_channel_stats.ice_transport_stats.connection_infos[2]
      .local_candidate = *a_local_relay;
  a_transport_channel_stats.ice_transport_stats.connection_infos[2]
      .remote_candidate = *a_remote_relay;
  a_transport_channel_stats.ice_transport_stats.connection_infos.push_back(
      ConnectionInfo());
  a_transport_channel_stats.ice_transport_stats.connection_infos[3]
      .local_candidate = *a_local_relay_prflx;
  a_transport_channel_stats.ice_transport_stats.connection_infos[3]
      .remote_candidate = *a_remote_relay;
  a_transport_channel_stats.ice_transport_stats.candidate_stats_list.push_back(
      CandidateStats(*a_local_host_not_paired));
  a_transport_channel_stats.ice_transport_stats.candidate_stats_list.push_back(
      CandidateStats(*a_local_tcp_slice));

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("audio", "a");
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  pc_->SetTransportStats("a", a_transport_channel_stats);

  TransportChannelStats b_transport_channel_stats;
  b_transport_channel_stats.ice_transport_stats.connection_infos.push_back(
      ConnectionInfo());
  b_transport_channel_stats.ice_transport_stats.connection_infos[0]
      .local_candidate = *b_local;
  b_transport_channel_stats.ice_transport_stats.connection_infos[0]
      .remote_candidate = *b_remote;

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("video", "b");
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  pc_->SetTransportStats("b", b_transport_channel_stats);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  ASSERT_TRUE(report->Get(expected_a_local_host.id()));
  EXPECT_EQ(expected_a_local_host, report->Get(expected_a_local_host.id())
                                       ->cast_to<RTCLocalIceCandidateStats>());

  ASSERT_TRUE(report->Get(expected_a_local_tcp_slice.id()));
  EXPECT_EQ(expected_a_local_tcp_slice,
            report->Get(expected_a_local_tcp_slice.id())
                ->cast_to<RTCLocalIceCandidateStats>());

  ASSERT_TRUE(report->Get(expected_a_local_host_not_paired.id()));
  EXPECT_EQ(expected_a_local_host_not_paired,
            report->Get(expected_a_local_host_not_paired.id())
                ->cast_to<RTCLocalIceCandidateStats>());

  ASSERT_TRUE(report->Get(expected_a_remote_srflx.id()));
  EXPECT_EQ(expected_a_remote_srflx,
            report->Get(expected_a_remote_srflx.id())
                ->cast_to<RTCRemoteIceCandidateStats>());
  ASSERT_TRUE(report->Get(expected_a_local_prflx.id()));
  EXPECT_EQ(expected_a_local_prflx, report->Get(expected_a_local_prflx.id())
                                        ->cast_to<RTCLocalIceCandidateStats>());
  ASSERT_TRUE(report->Get(expected_a_remote_relay.id()));
  EXPECT_EQ(expected_a_remote_relay,
            report->Get(expected_a_remote_relay.id())
                ->cast_to<RTCRemoteIceCandidateStats>());
  ASSERT_TRUE(report->Get(expected_a_local_relay.id()));
  EXPECT_EQ(expected_a_local_relay, report->Get(expected_a_local_relay.id())
                                        ->cast_to<RTCLocalIceCandidateStats>());
  ASSERT_TRUE(report->Get(expected_a_local_relay_prflx.id()));
  EXPECT_EQ(expected_a_local_relay_prflx,
            report->Get(expected_a_local_relay_prflx.id())
                ->cast_to<RTCLocalIceCandidateStats>());
  ASSERT_TRUE(report->Get(expected_b_local.id()));
  EXPECT_EQ(
      expected_b_local,
      report->Get(expected_b_local.id())->cast_to<RTCLocalIceCandidateStats>());
  ASSERT_TRUE(report->Get(expected_b_remote.id()));
  EXPECT_EQ(expected_b_remote, report->Get(expected_b_remote.id())
                                   ->cast_to<RTCRemoteIceCandidateStats>());
  EXPECT_TRUE(report->Get("Ta0"));
  EXPECT_TRUE(report->Get("Tb0"));
}

TEST_P(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
  const char kTransportName[] = "transport";

  std::unique_ptr<Candidate> local_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_WIFI,
                          IceCandidateType::kHost, 42);
  local_candidate->set_username("local_iceusernamefragment");

  std::unique_ptr<Candidate> remote_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_UNKNOWN,
                          IceCandidateType::kSrflx, 42);
  remote_candidate->set_related_address(SocketAddress("192.168.2.1", 43));
  remote_candidate->set_username("remote_iceusernamefragment");

  ConnectionInfo connection_info;
  connection_info.best_connection = false;
  connection_info.local_candidate = *local_candidate;
  connection_info.remote_candidate = *remote_candidate;
  connection_info.writable = true;
  connection_info.sent_discarded_packets = 3;
  connection_info.sent_total_packets = 10;
  connection_info.packets_received = 51;
  connection_info.sent_discarded_bytes = 7;
  connection_info.sent_total_bytes = 42;
  connection_info.recv_total_bytes = 1234;
  connection_info.total_round_trip_time_ms = 0;
  connection_info.current_round_trip_time_ms = std::nullopt;
  connection_info.recv_ping_requests = 2020;
  connection_info.sent_ping_requests_total = 2222;
  connection_info.sent_ping_requests_before_first_response = 2000;
  connection_info.recv_ping_responses = 4321;
  connection_info.sent_ping_responses = 1000;
  connection_info.state = IceCandidatePairState::IN_PROGRESS;
  connection_info.priority = 5555;
  connection_info.nominated = false;
  connection_info.last_data_received = Timestamp::Millis(2500);
  connection_info.last_data_sent = Timestamp::Millis(5200);

  TransportChannelStats transport_channel_stats;
  transport_channel_stats.component = ICE_CANDIDATE_COMPONENT_RTP;
  transport_channel_stats.ice_transport_stats.connection_infos.push_back(
      connection_info);

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("video", kTransportName);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  pc_->SetTransportStats(kTransportName, transport_channel_stats);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCIceCandidatePairStats expected_pair(
      "CP" + local_candidate->id() + "_" + remote_candidate->id(),
      report->timestamp());
  expected_pair.transport_id =
      "Ttransport" + absl::StrCat(ICE_CANDIDATE_COMPONENT_RTP);
  expected_pair.local_candidate_id = "I" + local_candidate->id();
  expected_pair.remote_candidate_id = "I" + remote_candidate->id();
  expected_pair.state = "in-progress";
  expected_pair.priority = 5555;
  expected_pair.nominated = false;
  expected_pair.writable = true;
  expected_pair.packets_sent = 7;
  expected_pair.packets_received = 51;
  expected_pair.packets_discarded_on_send = 3;
  expected_pair.bytes_sent = 42;
  expected_pair.bytes_received = 1234;
  expected_pair.bytes_discarded_on_send = 7;
  expected_pair.total_round_trip_time = 0.0;
  expected_pair.requests_received = 2020;
  expected_pair.requests_sent = 2222;
  expected_pair.responses_received = 4321;
  expected_pair.responses_sent = 1000;
  expected_pair.consent_requests_sent = (2222 - 2000);
  expected_pair.last_packet_received_timestamp = 2500;
  expected_pair.last_packet_sent_timestamp = 5200;

  // `expected_pair.current_round_trip_time` should be undefined because the
  // current RTT is not set.
  // `expected_pair.available_[outgoing/incoming]_bitrate` should be undefined
  // because is is not the current pair.

  ASSERT_TRUE(report->Get(expected_pair.id()));
  EXPECT_EQ(
      expected_pair,
      report->Get(expected_pair.id())->cast_to<RTCIceCandidatePairStats>());
  EXPECT_TRUE(report->Get(*expected_pair.transport_id));

  // Set nominated and "GetStats" again.
  transport_channel_stats.ice_transport_stats.connection_infos[0].nominated =
      true;
  pc_->SetTransportStats(kTransportName, transport_channel_stats);
  report = stats_->GetFreshStatsReport(main_thread_);
  expected_pair.nominated = true;
  ASSERT_TRUE(report->Get(expected_pair.id()));
  EXPECT_EQ(
      expected_pair,
      report->Get(expected_pair.id())->cast_to<RTCIceCandidatePairStats>());
  EXPECT_TRUE(report->Get(*expected_pair.transport_id));

  // Set round trip times and "GetStats" again.
  transport_channel_stats.ice_transport_stats.connection_infos[0]
      .total_round_trip_time_ms = 7331;
  transport_channel_stats.ice_transport_stats.connection_infos[0]
      .current_round_trip_time_ms = 1337;
  pc_->SetTransportStats(kTransportName, transport_channel_stats);
  report = stats_->GetFreshStatsReport(main_thread_);
  expected_pair.total_round_trip_time = 7.331;
  expected_pair.current_round_trip_time = 1.337;
  ASSERT_TRUE(report->Get(expected_pair.id()));
  EXPECT_EQ(
      expected_pair,
      report->Get(expected_pair.id())->cast_to<RTCIceCandidatePairStats>());
  EXPECT_TRUE(report->Get(*expected_pair.transport_id));

  // Make pair the current pair, clear bandwidth and "GetStats" again.
  transport_channel_stats.ice_transport_stats.connection_infos[0]
      .best_connection = true;
  pc_->SetTransportStats(kTransportName, transport_channel_stats);
  report = stats_->GetFreshStatsReport(main_thread_);
  // |expected_pair.available_[outgoing/incoming]_bitrate| should still be
  // undefined because bandwidth is not set.
  ASSERT_TRUE(report->Get(expected_pair.id()));
  EXPECT_EQ(
      expected_pair,
      report->Get(expected_pair.id())->cast_to<RTCIceCandidatePairStats>());
  EXPECT_TRUE(report->Get(*expected_pair.transport_id));

  // Set bandwidth and "GetStats" again.
  Call::Stats call_stats;
  const int kSendBandwidth = 888;
  call_stats.send_bandwidth_bps = kSendBandwidth;
  const int kRecvBandwidth = 999;
  call_stats.recv_bandwidth_bps = kRecvBandwidth;
  pc_->SetCallStats(call_stats);
  report = stats_->GetFreshStatsReport(main_thread_);
  expected_pair.available_outgoing_bitrate = kSendBandwidth;
  expected_pair.available_incoming_bitrate = kRecvBandwidth;
  ASSERT_TRUE(report->Get(expected_pair.id()));
  EXPECT_EQ(
      expected_pair,
      report->Get(expected_pair.id())->cast_to<RTCIceCandidatePairStats>());
  EXPECT_TRUE(report->Get(*expected_pair.transport_id));

  RTCLocalIceCandidateStats expected_local_candidate(
      *expected_pair.local_candidate_id, report->timestamp());
  expected_local_candidate.transport_id = *expected_pair.transport_id;
  expected_local_candidate.network_type = "wifi";
  expected_local_candidate.ip = "42.42.42.42";
  expected_local_candidate.address = "42.42.42.42";
  expected_local_candidate.port = 42;
  expected_local_candidate.protocol = "protocol";
  expected_local_candidate.candidate_type = "host";
  expected_local_candidate.priority = 42;
  expected_local_candidate.foundation = "foundationIsAString";
  expected_local_candidate.username_fragment = "local_iceusernamefragment";
  expected_local_candidate.vpn = false;
  expected_local_candidate.network_adapter_type = "wifi";
  ASSERT_TRUE(report->Get(expected_local_candidate.id()));
  EXPECT_EQ(expected_local_candidate,
            report->Get(expected_local_candidate.id())
                ->cast_to<RTCLocalIceCandidateStats>());

  RTCRemoteIceCandidateStats expected_remote_candidate(
      *expected_pair.remote_candidate_id, report->timestamp());
  expected_remote_candidate.transport_id = *expected_pair.transport_id;
  expected_remote_candidate.ip = "42.42.42.42";
  expected_remote_candidate.address = "42.42.42.42";
  expected_remote_candidate.port = 42;
  expected_remote_candidate.protocol = "protocol";
  expected_remote_candidate.candidate_type = "srflx";
  expected_remote_candidate.priority = 42;
  expected_remote_candidate.foundation = "foundationIsAString";
  expected_remote_candidate.username_fragment = "remote_iceusernamefragment";
  expected_remote_candidate.related_address = "192.168.2.1";
  expected_remote_candidate.related_port = 43;
  ASSERT_TRUE(report->Get(expected_remote_candidate.id()));
  EXPECT_EQ(expected_remote_candidate,
            report->Get(expected_remote_candidate.id())
                ->cast_to<RTCRemoteIceCandidateStats>());
}

TEST_P(RTCStatsCollectorTest, CollectRTCPeerConnectionStats) {
  {
    scoped_refptr<const RTCStatsReport> report =
        stats_->GetStatsReport(main_thread_);
    RTCPeerConnectionStats expected("P", report->timestamp());
    expected.data_channels_opened = 0;
    expected.data_channels_closed = 0;
    ASSERT_TRUE(report->Get("P"));
    EXPECT_EQ(expected, report->Get("P")->cast_to<RTCPeerConnectionStats>());
  }

  ScopedTaskSafety signaling_safety;

  FakeDataChannelController controller(pc_->network_thread());
  scoped_refptr<SctpDataChannel> dummy_channel_a = SctpDataChannel::Create(
      controller.weak_ptr(), "DummyChannelA", false, InternalDataChannelInit(),
      signaling_safety.flag(), Thread::Current(), Thread::Current());
  scoped_refptr<SctpDataChannel> dummy_channel_b = SctpDataChannel::Create(
      controller.weak_ptr(), "DummyChannelB", false, InternalDataChannelInit(),
      signaling_safety.flag(), Thread::Current(), Thread::Current());

  stats_->stats_collector().OnSctpDataChannelStateChanged(
      dummy_channel_a->internal_id(), DataChannelInterface::DataState::kOpen);
  // Closing a channel that is not opened should not affect the counts.
  stats_->stats_collector().OnSctpDataChannelStateChanged(
      dummy_channel_b->internal_id(), DataChannelInterface::DataState::kClosed);

  {
    scoped_refptr<const RTCStatsReport> report =
        stats_->GetFreshStatsReport(main_thread_);
    RTCPeerConnectionStats expected("P", report->timestamp());
    expected.data_channels_opened = 1;
    expected.data_channels_closed = 0;
    ASSERT_TRUE(report->Get("P"));
    EXPECT_EQ(expected, report->Get("P")->cast_to<RTCPeerConnectionStats>());
  }

  stats_->stats_collector().OnSctpDataChannelStateChanged(
      dummy_channel_b->internal_id(), DataChannelInterface::DataState::kOpen);
  stats_->stats_collector().OnSctpDataChannelStateChanged(
      dummy_channel_b->internal_id(), DataChannelInterface::DataState::kClosed);

  {
    scoped_refptr<const RTCStatsReport> report =
        stats_->GetFreshStatsReport(main_thread_);
    RTCPeerConnectionStats expected("P", report->timestamp());
    expected.data_channels_opened = 2;
    expected.data_channels_closed = 1;
    ASSERT_TRUE(report->Get("P"));
    EXPECT_EQ(expected, report->Get("P")->cast_to<RTCPeerConnectionStats>());
  }

  // Re-opening a data channel (or opening a new data channel that is re-using
  // the same address in memory) should increase the opened count.
  stats_->stats_collector().OnSctpDataChannelStateChanged(
      dummy_channel_b->internal_id(), DataChannelInterface::DataState::kOpen);

  {
    scoped_refptr<const RTCStatsReport> report =
        stats_->GetFreshStatsReport(main_thread_);
    RTCPeerConnectionStats expected("P", report->timestamp());
    expected.data_channels_opened = 3;
    expected.data_channels_closed = 1;
    ASSERT_TRUE(report->Get("P"));
    EXPECT_EQ(expected, report->Get("P")->cast_to<RTCPeerConnectionStats>());
  }

  stats_->stats_collector().OnSctpDataChannelStateChanged(
      dummy_channel_a->internal_id(), DataChannelInterface::DataState::kClosed);
  stats_->stats_collector().OnSctpDataChannelStateChanged(
      dummy_channel_b->internal_id(), DataChannelInterface::DataState::kClosed);

  {
    scoped_refptr<const RTCStatsReport> report =
        stats_->GetFreshStatsReport(main_thread_);
    RTCPeerConnectionStats expected("P", report->timestamp());
    expected.data_channels_opened = 3;
    expected.data_channels_closed = 3;
    ASSERT_TRUE(report->Get("P"));
    EXPECT_EQ(expected, report->Get("P")->cast_to<RTCPeerConnectionStats>());
  }
}

TEST_P(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio) {
  VoiceMediaInfo voice_media_info;

  voice_media_info.receivers.push_back(VoiceReceiverInfo());
  voice_media_info.receivers[0].local_stats.push_back(SsrcReceiverInfo());
  voice_media_info.receivers[0].local_stats[0].ssrc = 1;
  voice_media_info.receivers[0].packets_lost = -1;  // Signed per RFC3550
  voice_media_info.receivers[0].packets_discarded = 7788;
  voice_media_info.receivers[0].packets_received = 2;
  voice_media_info.receivers[0].packets_received_with_ect1 = 7;
  voice_media_info.receivers[0].packets_received_with_ce = 5;
  voice_media_info.receivers[0].nacks_sent = 5;
  voice_media_info.receivers[0].fec_packets_discarded = 5566;
  voice_media_info.receivers[0].fec_packets_received = 6677;
  voice_media_info.receivers[0].payload_bytes_received = 3;
  voice_media_info.receivers[0].header_and_padding_bytes_received = 4;
  voice_media_info.receivers[0].codec_payload_type = 42;
  voice_media_info.receivers[0].jitter_ms = 4500;
  voice_media_info.receivers[0].jitter_buffer_delay_seconds = 1.0;
  voice_media_info.receivers[0].jitter_buffer_target_delay_seconds = 1.1;
  voice_media_info.receivers[0].jitter_buffer_minimum_delay_seconds = 0.999;
  voice_media_info.receivers[0].jitter_buffer_emitted_count = 2;
  voice_media_info.receivers[0].total_samples_received = 3;
  voice_media_info.receivers[0].concealed_samples = 4;
  voice_media_info.receivers[0].silent_concealed_samples = 5;
  voice_media_info.receivers[0].concealment_events = 6;
  voice_media_info.receivers[0].inserted_samples_for_deceleration = 7;
  voice_media_info.receivers[0].removed_samples_for_acceleration = 8;
  voice_media_info.receivers[0].audio_level = 14442;  // [0,32767]
  voice_media_info.receivers[0].total_output_energy = 10.0;
  voice_media_info.receivers[0].total_output_duration = 11.0;
  voice_media_info.receivers[0].jitter_buffer_flushes = 7;
  voice_media_info.receivers[0].delayed_packet_outage_samples = 15;
  voice_media_info.receivers[0].relative_packet_arrival_delay_seconds = 16;
  voice_media_info.receivers[0].interruption_count = 7788;
  voice_media_info.receivers[0].total_interruption_duration_ms = 778899;
  voice_media_info.receivers[0].last_packet_received = std::nullopt;
  voice_media_info.receivers[0].total_processing_delay_seconds = 0.123;

  RtpCodecParameters codec_parameters;
  codec_parameters.payload_type = 42;
  codec_parameters.kind = MediaType::AUDIO;
  codec_parameters.name = "dummy";
  codec_parameters.clock_rate = 0;
  voice_media_info.receive_codecs.insert(
      std::make_pair(codec_parameters.payload_type, codec_parameters));

  Call::Stats call_stats;
  call_stats.sent_ccfb_stats_per_ssrc[1] = {
      .num_packets_reported_lost = 222, .num_packets_reported_recovered = 200};
  pc_->SetCallStats(call_stats);

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  auto voice_media_channels =
      pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->SetupRemoteTrackAndReceiver(MediaType::AUDIO, "RemoteAudioTrackID",
                                      "RemoteStreamId", 1);

  // Needed for playoutId to be populated.
  pc_->SetAudioDeviceStats(AudioDeviceModule::Stats());
  pc_->GetTransceiversInternal()[0]->internal()->set_current_direction(
      RtpTransceiverDirection::kSendRecv);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCInboundRtpStreamStats expected_audio("ITTransportName1A1",
                                          report->timestamp());
  expected_audio.ssrc = 1;
  expected_audio.kind = "audio";
  expected_audio.track_identifier = "RemoteAudioTrackID";
  expected_audio.mid = "AudioMid";
  expected_audio.transport_id = "TTransportName1";
  expected_audio.codec_id = "CITTransportName1_42";
  expected_audio.packets_received = 2;
  expected_audio.packets_received_with_ect1 = 7;
  expected_audio.packets_received_with_ce = 5;
  expected_audio.packets_reported_as_lost = 222;
  expected_audio.packets_reported_as_lost_but_recovered = 200;
  expected_audio.nack_count = 5;
  expected_audio.fec_packets_discarded = 5566;
  expected_audio.fec_packets_received = 6677;
  expected_audio.bytes_received = 3;
  expected_audio.header_bytes_received = 4;
  expected_audio.packets_lost = -1;
  expected_audio.packets_discarded = 7788;
  // `expected_audio.last_packet_received_timestamp` should be undefined.
  expected_audio.jitter = 4.5;
  expected_audio.jitter_buffer_delay = 1.0;
  expected_audio.jitter_buffer_target_delay = 1.1;
  expected_audio.jitter_buffer_minimum_delay = 0.999;
  expected_audio.jitter_buffer_emitted_count = 2;
  expected_audio.total_samples_received = 3;
  expected_audio.concealed_samples = 4;
  expected_audio.silent_concealed_samples = 5;
  expected_audio.concealment_events = 6;
  expected_audio.inserted_samples_for_deceleration = 7;
  expected_audio.removed_samples_for_acceleration = 8;
  expected_audio.audio_level = 14442.0 / 32767.0;  // [0,1]
  expected_audio.total_audio_energy = 10.0;
  expected_audio.total_samples_duration = 11.0;
  expected_audio.jitter_buffer_flushes = 7;
  expected_audio.delayed_packet_outage_samples = 15;
  expected_audio.relative_packet_arrival_delay = 16;
  expected_audio.interruption_count = 7788;
  expected_audio.total_interruption_duration = 778.899;
  expected_audio.total_processing_delay = 0.123;
  expected_audio.playout_id = "AP";

  ASSERT_TRUE(report->Get(expected_audio.id()));
  EXPECT_EQ(
      report->Get(expected_audio.id())->cast_to<RTCInboundRtpStreamStats>(),
      expected_audio);

  // Set previously undefined values and "GetStats" again.
  voice_media_info.receivers[0].last_packet_received = Timestamp::Seconds(3);
  expected_audio.last_packet_received_timestamp = 3000.0;
  voice_media_info.receivers[0].estimated_playout_ntp_timestamp_ms = 4567;
  expected_audio.estimated_playout_timestamp = 4567;
  voice_media_channels.first->SetStats(voice_media_info);
  voice_media_channels.second->SetStats(voice_media_info);

  report = stats_->GetFreshStatsReport(main_thread_);

  ASSERT_TRUE(report->Get(expected_audio.id()));
  EXPECT_EQ(
      report->Get(expected_audio.id())->cast_to<RTCInboundRtpStreamStats>(),
      expected_audio);
  EXPECT_TRUE(report->Get(*expected_audio.transport_id));
  EXPECT_TRUE(report->Get(*expected_audio.codec_id));
}

TEST_P(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio_PlayoutId) {
  VoiceMediaInfo voice_media_info;

  voice_media_info.receivers.push_back(VoiceReceiverInfo());
  voice_media_info.receivers[0].local_stats.push_back(SsrcReceiverInfo());
  voice_media_info.receivers[0].local_stats[0].ssrc = 1;
  voice_media_info.receivers[0].packets_received = 123;

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->SetupRemoteTrackAndReceiver(MediaType::AUDIO, "RemoteAudioTrackID",
                                      "RemoteStreamId", 1);
  // Needed for playoutId to be populated.
  pc_->SetAudioDeviceStats(AudioDeviceModule::Stats());

  {
    // We do not expect a playout id when only sending.
    pc_->GetTransceiversInternal()[0]->internal()->set_current_direction(
        RtpTransceiverDirection::kSendOnly);
    scoped_refptr<const RTCStatsReport> report =
        stats_->GetStatsReport(main_thread_);
    ASSERT_TRUE(report->Get("ITTransportName1A1"));
    auto stats =
        report->Get("ITTransportName1A1")->cast_to<RTCInboundRtpStreamStats>();
    ASSERT_FALSE(stats.playout_id.has_value());
  }
  {
    // We do expect a playout id when receiving.
    pc_->GetTransceiversInternal()[0]->internal()->set_current_direction(
        RtpTransceiverDirection::kRecvOnly);
    scoped_refptr<const RTCStatsReport> report =
        stats_->GetFreshStatsReport(main_thread_);
    ASSERT_TRUE(report->Get("ITTransportName1A1"));
    auto stats =
        report->Get("ITTransportName1A1")->cast_to<RTCInboundRtpStreamStats>();
    ASSERT_TRUE(stats.playout_id.has_value());
    EXPECT_EQ(*stats.playout_id, "AP");
  }
}

TEST_P(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) {
  VideoMediaInfo video_media_info;

  video_media_info.receivers.push_back(VideoReceiverInfo());
  video_media_info.receivers[0].local_stats.push_back(SsrcReceiverInfo());
  video_media_info.receivers[0].local_stats[0].ssrc = 1;
  video_media_info.receivers[0].packets_received = 2;
  video_media_info.receivers[0].packets_received_with_ect1 = 7;
  video_media_info.receivers[0].packets_received_with_ce = 5;
  video_media_info.receivers[0].packets_lost = 42;
  video_media_info.receivers[0].payload_bytes_received = 3;
  video_media_info.receivers[0].header_and_padding_bytes_received = 12;
  video_media_info.receivers[0].codec_payload_type = 42;
  video_media_info.receivers[0].firs_sent = 5;
  video_media_info.receivers[0].plis_sent = 6;
  video_media_info.receivers[0].nacks_sent = 7;
  video_media_info.receivers[0].frames_received = 8;
  video_media_info.receivers[0].frames_decoded = 9;
  video_media_info.receivers[0].key_frames_decoded = 3;
  video_media_info.receivers[0].frames_dropped = 13;
  video_media_info.receivers[0].qp_sum = std::nullopt;
  video_media_info.receivers[0].corruption_score_sum = std::nullopt;
  video_media_info.receivers[0].corruption_score_squared_sum = std::nullopt;
  video_media_info.receivers[0].total_decode_time = TimeDelta::Seconds(9);
  video_media_info.receivers[0].total_processing_delay = TimeDelta::Millis(600);
  video_media_info.receivers[0].total_assembly_time = TimeDelta::Millis(500);
  video_media_info.receivers[0].frames_assembled_from_multiple_packets = 23;
  video_media_info.receivers[0].total_inter_frame_delay = 0.123;
  video_media_info.receivers[0].total_squared_inter_frame_delay = 0.00456;
  video_media_info.receivers[0].pause_count = 2;
  video_media_info.receivers[0].total_pauses_duration_ms = 10000;
  video_media_info.receivers[0].freeze_count = 3;
  video_media_info.receivers[0].total_freezes_duration_ms = 1000;
  video_media_info.receivers[0].jitter_ms = 1199;
  video_media_info.receivers[0].jitter_buffer_delay_seconds = 3.456;
  video_media_info.receivers[0].jitter_buffer_target_delay_seconds = 1.1;
  video_media_info.receivers[0].jitter_buffer_minimum_delay_seconds = 0.999;
  video_media_info.receivers[0].jitter_buffer_emitted_count = 13;
  video_media_info.receivers[0].last_packet_received = std::nullopt;
  video_media_info.receivers[0].content_type = VideoContentType::UNSPECIFIED;
  video_media_info.receivers[0].estimated_playout_ntp_timestamp_ms =
      std::nullopt;
  video_media_info.receivers[0].decoder_implementation_name = std::nullopt;
  video_media_info.receivers[0].min_playout_delay_ms = 50;
  video_media_info.receivers[0].power_efficient_decoder = false;
  video_media_info.receivers[0].retransmitted_packets_received = 17;
  video_media_info.receivers[0].retransmitted_bytes_received = 62;
  video_media_info.receivers[0].fec_packets_received = 32;
  video_media_info.receivers[0].fec_bytes_received = 54;
  video_media_info.receivers[0].ssrc_groups.push_back(
      {kFidSsrcGroupSemantics, {1, 4404}});
  video_media_info.receivers[0].ssrc_groups.push_back(
      {kFecFrSsrcGroupSemantics, {1, 5505}});

  // Note: these two values intentionally differ,
  // only the decoded one should show up.
  video_media_info.receivers[0].framerate_received = 15;
  video_media_info.receivers[0].framerate_decoded = 5;

  RtpCodecParameters codec_parameters;
  codec_parameters.payload_type = 42;
  codec_parameters.kind = MediaType::VIDEO;
  codec_parameters.name = "dummy";
  codec_parameters.clock_rate = 0;
  video_media_info.receive_codecs.insert(
      std::make_pair(codec_parameters.payload_type, codec_parameters));

  Call::Stats call_stats;
  call_stats.sent_ccfb_stats_per_ssrc[1] = {
      .num_packets_reported_lost = 222, .num_packets_reported_recovered = 200};
  pc_->SetCallStats(call_stats);

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  auto video_media_channels =
      pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->SetupRemoteTrackAndReceiver(MediaType::VIDEO, "RemoteVideoTrackID",
                                      "RemoteStreamId", 1);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCInboundRtpStreamStats expected_video("ITTransportName1V1",
                                          report->timestamp());
  expected_video.ssrc = 1;
  expected_video.kind = "video";
  expected_video.track_identifier = "RemoteVideoTrackID";
  expected_video.mid = "VideoMid";
  expected_video.transport_id = "TTransportName1";
  expected_video.codec_id = "CITTransportName1_42";
  expected_video.fir_count = 5;
  expected_video.pli_count = 6;
  expected_video.nack_count = 7;
  expected_video.packets_received = 2;
  expected_video.packets_received_with_ect1 = 7;
  expected_video.packets_received_with_ce = 5;
  expected_video.packets_reported_as_lost = 222;
  expected_video.packets_reported_as_lost_but_recovered = 200;
  expected_video.bytes_received = 3;
  expected_video.header_bytes_received = 12;
  expected_video.packets_lost = 42;
  expected_video.frames_received = 8;
  expected_video.frames_decoded = 9;
  expected_video.key_frames_decoded = 3;
  expected_video.frames_dropped = 13;
  // `expected_video.qp_sum` should be undefined.
  // `corruption_score` related metrics should be undefined.
  expected_video.total_decode_time = 9.0;
  expected_video.total_processing_delay = 0.6;
  expected_video.total_assembly_time = 0.5;
  expected_video.frames_assembled_from_multiple_packets = 23;
  expected_video.total_inter_frame_delay = 0.123;
  expected_video.total_squared_inter_frame_delay = 0.00456;
  expected_video.pause_count = 2;
  expected_video.total_pauses_duration = 10;
  expected_video.freeze_count = 3;
  expected_video.total_freezes_duration = 1;
  expected_video.jitter = 1.199;
  expected_video.jitter_buffer_delay = 3.456;
  expected_video.jitter_buffer_target_delay = 1.1;
  expected_video.jitter_buffer_minimum_delay = 0.999;
  expected_video.jitter_buffer_emitted_count = 13;
  // `expected_video.last_packet_received_timestamp` should be undefined.
  // `expected_video.content_type` should be undefined.
  // `expected_video.decoder_implementation` should be undefined.
  expected_video.min_playout_delay = 0.05;
  expected_video.frames_per_second = 5;
  expected_video.power_efficient_decoder = false;
  expected_video.retransmitted_packets_received = 17;
  expected_video.retransmitted_bytes_received = 62;
  expected_video.fec_packets_received = 32;
  expected_video.fec_bytes_received = 54;
  expected_video.rtx_ssrc = 4404;
  expected_video.fec_ssrc = 5505;

  ASSERT_TRUE(report->Get(expected_video.id()));
  EXPECT_EQ(
      report->Get(expected_video.id())->cast_to<RTCInboundRtpStreamStats>(),
      expected_video);

  // Set previously undefined values and "GetStats" again.
  video_media_info.receivers[0].qp_sum = 9;
  expected_video.qp_sum = 9;
  video_media_info.receivers[0].corruption_score_sum = 0.5;
  video_media_info.receivers[0].corruption_score_squared_sum = 0.25;
  video_media_info.receivers[0].corruption_score_count = 5;
  expected_video.total_corruption_probability = 0.5;
  expected_video.total_squared_corruption_probability = 0.25;
  expected_video.corruption_measurements = 5;
  video_media_info.receivers[0].last_packet_received = Timestamp::Seconds(1);
  expected_video.last_packet_received_timestamp = 1000.0;
  video_media_info.receivers[0].content_type = VideoContentType::SCREENSHARE;
  expected_video.content_type = "screenshare";
  video_media_info.receivers[0].estimated_playout_ntp_timestamp_ms = 1234;
  expected_video.estimated_playout_timestamp = 1234;
  video_media_info.receivers[0].decoder_implementation_name = "libfoodecoder";
  expected_video.decoder_implementation = "libfoodecoder";
  video_media_info.receivers[0].power_efficient_decoder = true;
  expected_video.power_efficient_decoder = true;
  video_media_channels.first->SetStats(video_media_info);
  video_media_channels.second->SetStats(video_media_info);

  report = stats_->GetFreshStatsReport(main_thread_);

  ASSERT_TRUE(report->Get(expected_video.id()));
  EXPECT_EQ(
      report->Get(expected_video.id())->cast_to<RTCInboundRtpStreamStats>(),
      expected_video);
  EXPECT_TRUE(report->Get(*expected_video.transport_id));
  EXPECT_TRUE(report->Get(*expected_video.codec_id));
}

TEST_P(RTCStatsCollectorTest, CollectRTCAudioPlayoutStats) {
  AudioDeviceModule::Stats audio_device_stats;
  audio_device_stats.synthesized_samples_duration_s = 1;
  audio_device_stats.synthesized_samples_events = 2;
  audio_device_stats.total_samples_count = 3;
  audio_device_stats.total_samples_duration_s = 4;
  audio_device_stats.total_playout_delay_s = 5;
  pc_->SetAudioDeviceStats(audio_device_stats);

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("AudioMid", "TransportName", {});
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->SetupRemoteTrackAndReceiver(MediaType::AUDIO, "RemoteAudioTrackID",
                                      "RemoteStreamId", 1);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  auto stats_of_track_type = report->GetStatsOfType<RTCAudioPlayoutStats>();
  ASSERT_EQ(1U, stats_of_track_type.size());

  RTCAudioPlayoutStats expected_stats("AP", report->timestamp());
  expected_stats.kind = "audio";
  expected_stats.synthesized_samples_duration = 1;
  expected_stats.synthesized_samples_events = 2;
  expected_stats.total_samples_count = 3;
  expected_stats.total_samples_duration = 4;
  expected_stats.total_playout_delay = 5;

  ASSERT_TRUE(report->Get(expected_stats.id()));
  EXPECT_EQ(report->Get(expected_stats.id())->cast_to<RTCAudioPlayoutStats>(),
            expected_stats);
}

TEST_P(RTCStatsCollectorTest, CollectGoogTimingFrameInfo) {
  VideoMediaInfo video_media_info;

  video_media_info.receivers.push_back(VideoReceiverInfo());
  video_media_info.receivers[0].local_stats.push_back(SsrcReceiverInfo());
  video_media_info.receivers[0].local_stats[0].ssrc = 1;
  video_media_info.receivers[0].packets_received = 123;
  TimingFrameInfo timing_frame_info;
  timing_frame_info.rtp_timestamp = 1;
  timing_frame_info.capture_time_ms = 2;
  timing_frame_info.encode_start_ms = 3;
  timing_frame_info.encode_finish_ms = 4;
  timing_frame_info.packetization_finish_ms = 5;
  timing_frame_info.pacer_exit_ms = 6;
  timing_frame_info.network_timestamp_ms = 7;
  timing_frame_info.network2_timestamp_ms = 8;
  timing_frame_info.receive_start_ms = 9;
  timing_frame_info.receive_finish_ms = 10;
  timing_frame_info.decode_start_ms = 11;
  timing_frame_info.decode_finish_ms = 12;
  timing_frame_info.render_time_ms = 13;
  timing_frame_info.flags = 14;
  video_media_info.receivers[0].timing_frame_info = timing_frame_info;

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("Mid0", "Transport0", video_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->SetupRemoteTrackAndReceiver(MediaType::VIDEO, "RemoteVideoTrackID",
                                      "RemoteStreamId", 1);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
  ASSERT_EQ(inbound_rtps.size(), 1u);
  ASSERT_TRUE(inbound_rtps[0]->goog_timing_frame_info.has_value());
  EXPECT_EQ(*inbound_rtps[0]->goog_timing_frame_info,
            "1,2,3,4,5,6,7,8,9,10,11,12,13,1,0");
}

TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Audio) {
  VoiceMediaInfo voice_media_info;

  voice_media_info.senders.push_back(VoiceSenderInfo());
  voice_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
  voice_media_info.senders[0].local_stats[0].ssrc = 1;
  voice_media_info.senders[0].packets_sent = 2;
  voice_media_info.senders[0].packets_sent_with_ect1 = 2;
  voice_media_info.senders[0].total_packet_send_delay = TimeDelta::Seconds(1);
  voice_media_info.senders[0].retransmitted_packets_sent = 20;
  voice_media_info.senders[0].payload_bytes_sent = 3;
  voice_media_info.senders[0].header_and_padding_bytes_sent = 12;
  voice_media_info.senders[0].retransmitted_bytes_sent = 30;
  voice_media_info.senders[0].nacks_received = 31;
  voice_media_info.senders[0].target_bitrate = DataRate::BitsPerSec(32'000);
  voice_media_info.senders[0].codec_payload_type = 42;
  voice_media_info.senders[0].active = true;

  RtpCodecParameters codec_parameters;
  codec_parameters.payload_type = 42;
  codec_parameters.kind = MediaType::AUDIO;
  codec_parameters.name = "dummy";
  codec_parameters.clock_rate = 0;
  voice_media_info.send_codecs.insert(
      std::make_pair(codec_parameters.payload_type, codec_parameters));

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->SetupLocalTrackAndSender(MediaType::AUDIO, "LocalAudioTrackID", 1,
                                   true,
                                   /*attachment_id=*/50);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCOutboundRtpStreamStats expected_audio("OTTransportName1A1",
                                           report->timestamp());
  expected_audio.media_source_id = "SA50";
  // `expected_audio.remote_id` should be undefined.
  expected_audio.mid = "AudioMid";
  expected_audio.ssrc = 1;
  expected_audio.kind = "audio";
  expected_audio.transport_id = "TTransportName1";
  expected_audio.codec_id = "COTTransportName1_42";
  expected_audio.packets_sent = 2;
  expected_audio.packets_sent_with_ect1 = 2;
  expected_audio.total_packet_send_delay = 1;
  expected_audio.retransmitted_packets_sent = 20;
  expected_audio.bytes_sent = 3;
  expected_audio.header_bytes_sent = 12;
  expected_audio.retransmitted_bytes_sent = 30;
  expected_audio.nack_count = 31;
  expected_audio.target_bitrate = 32000;
  expected_audio.active = true;

  ASSERT_TRUE(report->Get(expected_audio.id()));
  EXPECT_EQ(
      report->Get(expected_audio.id())->cast_to<RTCOutboundRtpStreamStats>(),
      expected_audio);

  ASSERT_TRUE(report->Get(expected_audio.id()));
  EXPECT_EQ(
      report->Get(expected_audio.id())->cast_to<RTCOutboundRtpStreamStats>(),
      expected_audio);
  EXPECT_TRUE(report->Get(*expected_audio.transport_id));
  EXPECT_TRUE(report->Get(*expected_audio.codec_id));
}

TEST_P(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
  VideoMediaInfo video_media_info;

  video_media_info.senders.push_back(VideoSenderInfo());
  video_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
  video_media_info.senders[0].local_stats[0].ssrc = 1;
  video_media_info.senders[0].rid = "q";
  video_media_info.senders[0].encoding_index = 0;
  video_media_info.senders[0].firs_received = 2;
  video_media_info.senders[0].plis_received = 3;
  video_media_info.senders[0].nacks_received = 4;
  video_media_info.senders[0].packets_sent = 5;
  video_media_info.senders[0].packets_sent_with_ect1 = 3;
  video_media_info.senders[0].retransmitted_packets_sent = 50;
  video_media_info.senders[0].payload_bytes_sent = 6;
  video_media_info.senders[0].header_and_padding_bytes_sent = 12;
  video_media_info.senders[0].retransmitted_bytes_sent = 60;
  video_media_info.senders[0].codec_payload_type = 42;
  video_media_info.senders[0].frames_encoded = 8;
  video_media_info.senders[0].key_frames_encoded = 3;
  video_media_info.senders[0].total_encode_time_ms = 9000;
  video_media_info.senders[0].total_encoded_bytes_target = 1234;
  video_media_info.senders[0].total_packet_send_delay = TimeDelta::Seconds(10);
  video_media_info.senders[0].quality_limitation_reason =
      QualityLimitationReason::kBandwidth;
  video_media_info.senders[0]
      .quality_limitation_durations_ms[QualityLimitationReason::kBandwidth] =
      300;
  video_media_info.senders[0].quality_limitation_resolution_changes = 56u;
  video_media_info.senders[0].qp_sum = std::nullopt;
  video_media_info.senders[0].psnr_measurements = 0;
  video_media_info.senders[0].content_type = VideoContentType::UNSPECIFIED;
  video_media_info.senders[0].encoder_implementation_name = std::nullopt;
  video_media_info.senders[0].power_efficient_encoder = false;
  video_media_info.senders[0].send_frame_width = 200;
  video_media_info.senders[0].send_frame_height = 100;
  video_media_info.senders[0].framerate_sent = 10;
  video_media_info.senders[0].frames_sent = 5;
  video_media_info.senders[0].huge_frames_sent = 2;
  video_media_info.senders[0].active = false;
  video_media_info.senders[0].scalability_mode = ScalabilityMode::kL3T3_KEY;
  video_media_info.senders[0].ssrc_groups.push_back(
      {kFidSsrcGroupSemantics, {1, 4404}});
  video_media_info.aggregated_senders.push_back(video_media_info.senders[0]);
  RtpCodecParameters codec_parameters;
  codec_parameters.payload_type = 42;
  codec_parameters.kind = MediaType::AUDIO;
  codec_parameters.name = "dummy";
  codec_parameters.clock_rate = 0;
  video_media_info.send_codecs.insert(
      std::make_pair(codec_parameters.payload_type, codec_parameters));
  // Copy the sender info into a second simulcast layer and update ssrc, rid and
  // simulcast index to be unique.
  video_media_info.senders.push_back(video_media_info.senders[0]);
  video_media_info.senders[1].local_stats[0].ssrc = 2;
  video_media_info.senders[1].rid = "h";
  video_media_info.senders[1].encoding_index = 1;

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  auto video_media_channels =
      pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  auto video_sender = stats_->SetupLocalTrackAndSender(
      MediaType::VIDEO, "LocalVideoTrackID", 1, true,
      /*attachment_id=*/50);

  // Set up the fake video channel to return the expected RtpParameters
  // containing both simulcast encodings for the primary SSRC.
  StreamParams sp;
  sp.add_ssrc(1);
  sp.add_ssrc(2);
  sp.ssrc_groups.push_back(SsrcGroup(kSimSsrcGroupSemantics, {1, 2}));
  video_media_channels.first->AddSendStream(sp);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  auto stats_of_my_type = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
  ASSERT_EQ(2U, stats_of_my_type.size());
  std::string id_of_first_ssrc;
  for (const RTCOutboundRtpStreamStats* outbound_rtp : stats_of_my_type) {
    if (outbound_rtp->ssrc.value_or(0) == 1) {
      id_of_first_ssrc = outbound_rtp->id();
      break;
    }
  }

  RTCOutboundRtpStreamStats expected_video(id_of_first_ssrc,
                                           report->timestamp());
  expected_video.media_source_id = "SV50";
  // `expected_video.remote_id` should be undefined.
  expected_video.mid = "VideoMid";
  expected_video.rid = "q";
  expected_video.encoding_index = 0;
  expected_video.ssrc = 1;
  expected_video.kind = "video";
  expected_video.transport_id = "TTransportName1";
  expected_video.codec_id = "COTTransportName1_42";
  expected_video.fir_count = 2;
  expected_video.pli_count = 3;
  expected_video.nack_count = 4;
  expected_video.packets_sent = 5;
  expected_video.packets_sent_with_ect1 = 3;
  expected_video.retransmitted_packets_sent = 50;
  expected_video.bytes_sent = 6;
  expected_video.header_bytes_sent = 12;
  expected_video.retransmitted_bytes_sent = 60;
  expected_video.frames_encoded = 8;
  expected_video.key_frames_encoded = 3;
  expected_video.total_encode_time = 9.0;
  expected_video.total_encoded_bytes_target = 1234;
  expected_video.total_packet_send_delay = 10.0;
  expected_video.quality_limitation_reason = "bandwidth";
  expected_video.quality_limitation_durations = std::map<std::string, double>{
      std::pair<std::string, double>{"bandwidth", 0.3},
  };
  expected_video.quality_limitation_resolution_changes = 56u;
  expected_video.frame_width = 200u;
  expected_video.frame_height = 100u;
  expected_video.frames_per_second = 10.0;
  expected_video.frames_sent = 5;
  expected_video.huge_frames_sent = 2;
  expected_video.active = false;
  expected_video.power_efficient_encoder = false;
  expected_video.scalability_mode = "L3T3_KEY";
  expected_video.rtx_ssrc = 4404;
  // `expected_video.content_type` should be undefined.
  // `expected_video.qp_sum` should be undefined.
  // `expected_video.psnr_sum` should be undefined.
  // `expected_video.psnr_measurements` should be undefined.
  // `expected_video.encoder_implementation` should be undefined.
  ASSERT_TRUE(report->Get(expected_video.id()));

  EXPECT_EQ(
      report->Get(expected_video.id())->cast_to<RTCOutboundRtpStreamStats>(),
      expected_video);

  // Find the sub-stats for the second simulcast outbound RTP stream and verify
  // that its `media_source_id` is set.
  std::string id_of_second_ssrc;
  for (const RTCOutboundRtpStreamStats* outbound_rtp : stats_of_my_type) {
    if (outbound_rtp->ssrc.value_or(0) == 2) {
      id_of_second_ssrc = outbound_rtp->id();
      break;
    }
  }
  ASSERT_TRUE(report->Get(id_of_second_ssrc));
  auto& second_video =
      report->Get(id_of_second_ssrc)->cast_to<RTCOutboundRtpStreamStats>();
  EXPECT_TRUE(second_video.media_source_id.has_value());
  EXPECT_EQ(*second_video.media_source_id, "SV50");

  // Set previously undefined values and "GetStats" again.
  video_media_info.senders[0].qp_sum = 9;
  expected_video.qp_sum = 9;
  video_media_info.senders[0].psnr_sum = {.y = 29, .u = 30, .v = 31};
  video_media_info.senders[0].psnr_measurements = 1;
  expected_video.psnr_sum = {{"y", 29}, {"u", 30}, {"v", 31}};
  expected_video.psnr_measurements = 1;
  video_media_info.senders[0].content_type = VideoContentType::SCREENSHARE;
  expected_video.content_type = "screenshare";
  video_media_info.senders[0].encoder_implementation_name = "libfooencoder";
  video_media_info.aggregated_senders[0] = video_media_info.senders[0];
  expected_video.encoder_implementation = "libfooencoder";
  video_media_info.senders[0].power_efficient_encoder = true;
  expected_video.power_efficient_encoder = true;
  video_media_channels.first->SetStats(video_media_info);
  video_media_channels.second->SetStats(video_media_info);

  report = stats_->GetFreshStatsReport(main_thread_);

  ASSERT_TRUE(report->Get(expected_video.id()));
  EXPECT_EQ(
      report->Get(expected_video.id())->cast_to<RTCOutboundRtpStreamStats>(),
      expected_video);
  EXPECT_TRUE(report->Get(*expected_video.transport_id));
  EXPECT_TRUE(report->Get(*expected_video.codec_id));
}

TEST_P(RTCStatsCollectorTest, CollectRTCTransportStats) {
  const char kTransportName[] = "transport";

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("audio", kTransportName);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  std::unique_ptr<Candidate> rtp_local_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_WIFI,
                          IceCandidateType::kHost, 42);
  std::unique_ptr<Candidate> rtp_remote_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_UNKNOWN,
                          IceCandidateType::kHost, 42);
  std::unique_ptr<Candidate> rtcp_local_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_WIFI,
                          IceCandidateType::kHost, 42);
  std::unique_ptr<Candidate> rtcp_remote_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_UNKNOWN,
                          IceCandidateType::kHost, 42);

  ConnectionInfo rtp_connection_info;
  rtp_connection_info.best_connection = false;
  rtp_connection_info.local_candidate = *rtp_local_candidate;
  rtp_connection_info.remote_candidate = *rtp_remote_candidate;
  rtp_connection_info.sent_total_bytes = 42;
  rtp_connection_info.recv_total_bytes = 1337;
  rtp_connection_info.sent_total_packets = 3;
  rtp_connection_info.sent_discarded_packets = 2;
  rtp_connection_info.packets_received = 4;
  TransportChannelStats rtp_transport_channel_stats;
  rtp_transport_channel_stats.component = ICE_CANDIDATE_COMPONENT_RTP;
  rtp_transport_channel_stats.ice_transport_stats.connection_infos.push_back(
      rtp_connection_info);
  rtp_transport_channel_stats.dtls_state = DtlsTransportState::kNew;
  rtp_transport_channel_stats.ice_transport_stats.bytes_sent = 42;
  rtp_transport_channel_stats.ice_transport_stats.packets_sent = 1;
  rtp_transport_channel_stats.ice_transport_stats.bytes_received = 1337;
  rtp_transport_channel_stats.ice_transport_stats.packets_received = 4;
  rtp_transport_channel_stats.ice_transport_stats
      .selected_candidate_pair_changes = 1;
  rtp_transport_channel_stats.ice_transport_stats.ice_local_username_fragment =
      "thelocalufrag";
  pc_->SetTransportStats(kTransportName, {rtp_transport_channel_stats});

  // Get stats without RTCP, an active connection or certificates.
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCTransportStats expected_rtp_transport(
      "Ttransport" + absl::StrCat(ICE_CANDIDATE_COMPONENT_RTP),
      report->timestamp());
  expected_rtp_transport.bytes_sent = 42;
  expected_rtp_transport.packets_sent = 1;
  expected_rtp_transport.bytes_received = 1337;
  expected_rtp_transport.packets_received = 4;
  expected_rtp_transport.dtls_state = "new";
  expected_rtp_transport.dtls_role = "unknown";
  expected_rtp_transport.selected_candidate_pair_changes = 1;
  expected_rtp_transport.ice_role = "unknown";
  expected_rtp_transport.ice_local_username_fragment = "thelocalufrag";
  expected_rtp_transport.ice_state = "new";

  ASSERT_TRUE(report->Get(expected_rtp_transport.id()));
  EXPECT_EQ(
      expected_rtp_transport,
      report->Get(expected_rtp_transport.id())->cast_to<RTCTransportStats>());

  ConnectionInfo rtcp_connection_info;
  rtcp_connection_info.best_connection = false;
  rtcp_connection_info.local_candidate = *rtcp_local_candidate;
  rtcp_connection_info.remote_candidate = *rtcp_remote_candidate;
  rtcp_connection_info.sent_total_bytes = 1337;
  rtcp_connection_info.recv_total_bytes = 42;
  rtcp_connection_info.sent_total_packets = 3;
  rtcp_connection_info.sent_discarded_packets = 2;
  rtcp_connection_info.packets_received = 4;
  TransportChannelStats rtcp_transport_channel_stats;
  rtcp_transport_channel_stats.component = ICE_CANDIDATE_COMPONENT_RTCP;
  rtcp_transport_channel_stats.ice_transport_stats.connection_infos.push_back(
      rtcp_connection_info);
  rtcp_transport_channel_stats.dtls_state = DtlsTransportState::kConnecting;
  rtcp_transport_channel_stats.ice_transport_stats.bytes_sent = 1337;
  rtcp_transport_channel_stats.ice_transport_stats.packets_sent = 1;
  rtcp_transport_channel_stats.ice_transport_stats.bytes_received = 42;
  rtcp_transport_channel_stats.ice_transport_stats.packets_received = 4;
  rtcp_transport_channel_stats.ice_transport_stats.ice_local_username_fragment =
      "thelocalufrag";
  rtcp_transport_channel_stats.ice_transport_stats.ice_state =
      IceTransportState::kChecking;
  pc_->SetTransportStats(kTransportName, {rtp_transport_channel_stats,
                                          rtcp_transport_channel_stats});

  // Get stats with RTCP and without an active connection or certificates.
  report = stats_->GetFreshStatsReport(main_thread_);

  RTCTransportStats expected_rtcp_transport(
      "Ttransport" + absl::StrCat(ICE_CANDIDATE_COMPONENT_RTCP),
      report->timestamp());
  expected_rtcp_transport.bytes_sent = 1337;
  expected_rtcp_transport.packets_sent = 1;
  expected_rtcp_transport.bytes_received = 42;
  expected_rtcp_transport.packets_received = 4;
  expected_rtcp_transport.dtls_state = "connecting";
  expected_rtcp_transport.dtls_role = "unknown";
  expected_rtcp_transport.selected_candidate_pair_changes = 0;
  expected_rtcp_transport.ice_role = "unknown";
  expected_rtcp_transport.ice_local_username_fragment = "thelocalufrag";
  expected_rtcp_transport.ice_state = "checking";

  expected_rtp_transport.rtcp_transport_stats_id = expected_rtcp_transport.id();
  ASSERT_TRUE(report->Get(expected_rtp_transport.id()));
  EXPECT_EQ(
      expected_rtp_transport,
      report->Get(expected_rtp_transport.id())->cast_to<RTCTransportStats>());
  ASSERT_TRUE(report->Get(expected_rtcp_transport.id()));
  EXPECT_EQ(
      expected_rtcp_transport,
      report->Get(expected_rtcp_transport.id())->cast_to<RTCTransportStats>());

  // Get stats with an active connection (selected candidate pair).
  rtcp_transport_channel_stats.ice_transport_stats.connection_infos[0]
      .best_connection = true;
  pc_->SetTransportStats(kTransportName, {rtp_transport_channel_stats,
                                          rtcp_transport_channel_stats});

  report = stats_->GetFreshStatsReport(main_thread_);

  expected_rtcp_transport.selected_candidate_pair_id =
      "CP" + rtcp_local_candidate->id() + "_" + rtcp_remote_candidate->id();

  ASSERT_TRUE(report->Get(expected_rtp_transport.id()));
  EXPECT_EQ(
      expected_rtp_transport,
      report->Get(expected_rtp_transport.id())->cast_to<RTCTransportStats>());
  ASSERT_TRUE(report->Get(expected_rtcp_transport.id()));
  EXPECT_EQ(
      expected_rtcp_transport,
      report->Get(expected_rtcp_transport.id())->cast_to<RTCTransportStats>());

  // Get stats with certificates.
  std::unique_ptr<CertificateInfo> local_certinfo =
      CreateFakeCertificateAndInfoFromDers({"(local) local", "(local) chain"});
  pc_->SetLocalCertificate(kTransportName, local_certinfo->certificate);
  std::unique_ptr<CertificateInfo> remote_certinfo =
      CreateFakeCertificateAndInfoFromDers(
          {"(remote) local", "(remote) chain"});
  pc_->SetRemoteCertChain(
      kTransportName,
      remote_certinfo->certificate->GetSSLCertificateChain().Clone());

  report = stats_->GetFreshStatsReport(main_thread_);

  expected_rtp_transport.local_certificate_id =
      "CF" + local_certinfo->fingerprints[0];
  expected_rtp_transport.remote_certificate_id =
      "CF" + remote_certinfo->fingerprints[0];

  expected_rtcp_transport.local_certificate_id =
      *expected_rtp_transport.local_certificate_id;
  expected_rtcp_transport.remote_certificate_id =
      *expected_rtp_transport.remote_certificate_id;

  ASSERT_TRUE(report->Get(expected_rtp_transport.id()));
  EXPECT_EQ(
      expected_rtp_transport,
      report->Get(expected_rtp_transport.id())->cast_to<RTCTransportStats>());
  ASSERT_TRUE(report->Get(expected_rtcp_transport.id()));
  EXPECT_EQ(
      expected_rtcp_transport,
      report->Get(expected_rtcp_transport.id())->cast_to<RTCTransportStats>());
}

TEST_P(RTCStatsCollectorTest, CollectRTCTransportStatsWithCrypto) {
  const char kTransportName[] = "transport";

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("audio", kTransportName);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  std::unique_ptr<Candidate> rtp_local_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_WIFI,
                          IceCandidateType::kHost, 42);
  std::unique_ptr<Candidate> rtp_remote_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_UNKNOWN,
                          IceCandidateType::kHost, 42);
  std::unique_ptr<Candidate> rtcp_local_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_WIFI,
                          IceCandidateType::kHost, 42);
  std::unique_ptr<Candidate> rtcp_remote_candidate =
      CreateFakeCandidate("42.42.42.42", 42, "protocol", ADAPTER_TYPE_UNKNOWN,
                          IceCandidateType::kHost, 42);

  ConnectionInfo rtp_connection_info;
  rtp_connection_info.best_connection = false;
  rtp_connection_info.local_candidate = *rtp_local_candidate;
  rtp_connection_info.remote_candidate = *rtp_remote_candidate;
  TransportChannelStats rtp_transport_channel_stats;
  rtp_transport_channel_stats.component = ICE_CANDIDATE_COMPONENT_RTP;
  rtp_transport_channel_stats.ice_transport_stats.connection_infos.push_back(
      rtp_connection_info);
  // The state must be connected in order for crypto parameters to show up.
  rtp_transport_channel_stats.dtls_state = DtlsTransportState::kConnected;
  rtp_transport_channel_stats.ice_transport_stats
      .selected_candidate_pair_changes = 1;
  rtp_transport_channel_stats.ice_transport_stats.ice_role =
      ICEROLE_CONTROLLING;
  rtp_transport_channel_stats.ice_transport_stats.ice_local_username_fragment =
      "thelocalufrag";
  rtp_transport_channel_stats.ice_transport_stats.ice_state =
      IceTransportState::kConnected;
  rtp_transport_channel_stats.ssl_version_bytes = 0x0203;
  rtp_transport_channel_stats.dtls_role = SSL_CLIENT;
  rtp_transport_channel_stats.tls_cipher_suite_name =
      "TLS_RSA_WITH_AES_128_CBC_SHA";
  rtp_transport_channel_stats.srtp_crypto_suite = kSrtpAes128CmSha1_80;
  pc_->SetTransportStats(kTransportName, {rtp_transport_channel_stats});

  Call::Stats call_stats;
  call_stats.ccfb_messages_received = 5;
  pc_->SetCallStats(call_stats);

  // Get stats
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCTransportStats expected_rtp_transport(
      "Ttransport" + absl::StrCat(ICE_CANDIDATE_COMPONENT_RTP),
      report->timestamp());
  expected_rtp_transport.dtls_state = "connected";
  expected_rtp_transport.selected_candidate_pair_changes = 1;
  expected_rtp_transport.ice_role = "unknown";
  expected_rtp_transport.bytes_sent = 0;
  expected_rtp_transport.bytes_received = 0;
  expected_rtp_transport.packets_sent = 0;
  expected_rtp_transport.packets_received = 0;
  expected_rtp_transport.ice_role = "controlling";
  expected_rtp_transport.ice_local_username_fragment = "thelocalufrag";
  expected_rtp_transport.ice_state = "connected";
  // Crypto parameters
  expected_rtp_transport.tls_version = "0203";
  expected_rtp_transport.dtls_role = "client";
  expected_rtp_transport.dtls_cipher = "TLS_RSA_WITH_AES_128_CBC_SHA";
  expected_rtp_transport.srtp_cipher = "AES_CM_128_HMAC_SHA1_80";
  // CCFB stats.
  expected_rtp_transport.ccfb_messages_received = 5;

  ASSERT_TRUE(report->Get(expected_rtp_transport.id()));
  EXPECT_EQ(
      expected_rtp_transport,
      report->Get(expected_rtp_transport.id())->cast_to<RTCTransportStats>());
}

TEST_P(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRtpStreamStats_Audio) {
  VoiceMediaInfo voice_media_info;

  voice_media_info.senders.push_back(VoiceSenderInfo());
  voice_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
  voice_media_info.senders[0].local_stats[0].ssrc = 1;
  voice_media_info.senders[0].packets_sent = 2;
  voice_media_info.senders[0].packets_sent_with_ect1 = 0;
  voice_media_info.senders[0].total_packet_send_delay = TimeDelta::Seconds(0.5);
  voice_media_info.senders[0].retransmitted_packets_sent = 20;
  voice_media_info.senders[0].payload_bytes_sent = 3;
  voice_media_info.senders[0].header_and_padding_bytes_sent = 4;
  voice_media_info.senders[0].retransmitted_bytes_sent = 30;
  voice_media_info.senders[0].nacks_received = 31;
  voice_media_info.senders[0].codec_payload_type = 42;
  voice_media_info.senders[0].active = true;

  RtpCodecParameters codec_parameters;
  codec_parameters.payload_type = 42;
  codec_parameters.kind = MediaType::AUDIO;
  codec_parameters.name = "dummy";
  codec_parameters.clock_rate = 0;
  voice_media_info.send_codecs.insert(
      std::make_pair(codec_parameters.payload_type, codec_parameters));

  // Emulates the case where AddTrack is used without an associated MediaStream
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->SetupLocalTrackAndSender(MediaType::AUDIO, "LocalAudioTrackID", 1,
                                   false,
                                   /*attachment_id=*/50);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCOutboundRtpStreamStats expected_audio("OTTransportName1A1",
                                           report->timestamp());
  expected_audio.media_source_id = "SA50";
  expected_audio.mid = "AudioMid";
  expected_audio.ssrc = 1;
  expected_audio.kind = "audio";
  expected_audio.transport_id = "TTransportName1";
  expected_audio.codec_id = "COTTransportName1_42";
  expected_audio.packets_sent = 2;
  expected_audio.packets_sent_with_ect1 = 0;
  expected_audio.total_packet_send_delay = 0.5;
  expected_audio.retransmitted_packets_sent = 20;
  expected_audio.bytes_sent = 3;
  expected_audio.header_bytes_sent = 4;
  expected_audio.retransmitted_bytes_sent = 30;
  expected_audio.nack_count = 31;
  expected_audio.active = true;

  ASSERT_TRUE(report->Get(expected_audio.id()));
  EXPECT_EQ(
      report->Get(expected_audio.id())->cast_to<RTCOutboundRtpStreamStats>(),
      expected_audio);
  EXPECT_TRUE(report->Get(*expected_audio.transport_id));
  EXPECT_TRUE(report->Get(*expected_audio.codec_id));
}

TEST_P(RTCStatsCollectorTest, RTCAudioSourceStatsCollectedForSenderWithTrack) {
  const uint32_t kSsrc = 4;
  const int kAttachmentId = 42;

  VoiceMediaInfo voice_media_info;
  voice_media_info.senders.push_back(VoiceSenderInfo());
  voice_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
  voice_media_info.senders[0].local_stats[0].ssrc = kSsrc;
  voice_media_info.senders[0].audio_level = 32767;  // [0,32767]
  voice_media_info.senders[0].total_input_energy = 2.0;
  voice_media_info.senders[0].total_input_duration = 3.0;
  voice_media_info.senders[0].apm_statistics.echo_return_loss = 42.0;
  voice_media_info.senders[0].apm_statistics.echo_return_loss_enhancement =
      52.0;
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  stats_->SetupLocalTrackAndSender(MediaType::AUDIO, "LocalAudioTrackID", kSsrc,
                                   false, kAttachmentId);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCAudioSourceStats expected_audio("SA42", report->timestamp());
  expected_audio.track_identifier = "LocalAudioTrackID";
  expected_audio.kind = "audio";
  expected_audio.audio_level = 1.0;  // [0,1]
  expected_audio.total_audio_energy = 2.0;
  expected_audio.total_samples_duration = 3.0;
  expected_audio.echo_return_loss = 42.0;
  expected_audio.echo_return_loss_enhancement = 52.0;

  ASSERT_TRUE(report->Get(expected_audio.id()));
  EXPECT_EQ(report->Get(expected_audio.id())->cast_to<RTCAudioSourceStats>(),
            expected_audio);
}

TEST_P(RTCStatsCollectorTest, RTCVideoSourceStatsCollectedForSenderWithTrack) {
  const uint32_t kSsrc = 4;
  const int kAttachmentId = 42;
  const int kVideoSourceWidth = 12;
  const int kVideoSourceHeight = 34;

  VideoMediaInfo video_media_info;
  video_media_info.aggregated_senders.push_back(VideoSenderInfo());
  video_media_info.senders.push_back(VideoSenderInfo());
  video_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
  video_media_info.senders[0].local_stats[0].ssrc = kSsrc;
  video_media_info.senders[0].framerate_input = 29.0;
  video_media_info.aggregated_senders[0].local_stats.push_back(
      SsrcSenderInfo());
  video_media_info.aggregated_senders[0].local_stats[0].ssrc = kSsrc;
  video_media_info.aggregated_senders[0].framerate_input = 29.0;
  video_media_info.aggregated_senders[0].frames = 10001;
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  auto video_source = FakeVideoTrackSource::Create(false);
  video_source->SetSize(kVideoSourceWidth, kVideoSourceHeight);
  auto video_track = CreateFakeVideoTrackForStats(
      "LocalVideoTrackID", MediaStreamTrackInterface::kLive, video_source);
  scoped_refptr<MockRtpSenderInternal> sender =
      CreateMockSender(MediaType::VIDEO, video_track, kSsrc, kAttachmentId, {});
  EXPECT_CALL(*sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
    return nullptr;
  });
  EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
  EXPECT_CALL(*sender, SetSendCodecs(_));
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddSender(sender);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCVideoSourceStats expected_video("SV42", report->timestamp());
  expected_video.track_identifier = "LocalVideoTrackID";
  expected_video.kind = "video";
  expected_video.width = kVideoSourceWidth;
  expected_video.height = kVideoSourceHeight;
  expected_video.frames_per_second = 29.0;
  expected_video.frames = 10001;

  ASSERT_TRUE(report->Get(expected_video.id()));
  EXPECT_EQ(report->Get(expected_video.id())->cast_to<RTCVideoSourceStats>(),
            expected_video);
}

// This test exercises the current behavior and code path, but the correct
// behavior is to report frame rate even if we have no SSRC.
// TODO(hbos): When we know the frame rate even if we have no SSRC, update the
// expectations of this test.
TEST_P(RTCStatsCollectorTest,
       RTCVideoSourceStatsMissingFrameRateWhenSenderHasNoSsrc) {
  // TODO(https://crbug.com/webrtc/8694): When 0 is no longer a magic value for
  // "none", update this test.
  const uint32_t kNoSsrc = 0;
  const int kAttachmentId = 42;
  const int kVideoSourceWidth = 12;
  const int kVideoSourceHeight = 34;

  VideoMediaInfo video_media_info;
  video_media_info.senders.push_back(VideoSenderInfo());
  video_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
  video_media_info.senders[0].framerate_input = 29.0;
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  auto video_source = FakeVideoTrackSource::Create(false);
  video_source->SetSize(kVideoSourceWidth, kVideoSourceHeight);
  auto video_track = CreateFakeVideoTrackForStats(
      "LocalVideoTrackID", MediaStreamTrackInterface::kLive, video_source);
  scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
      MediaType::VIDEO, video_track, kNoSsrc, kAttachmentId, {});
  EXPECT_CALL(*sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
    return nullptr;
  });
  EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
  EXPECT_CALL(*sender, SetSendCodecs(_));
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddSender(sender);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  ASSERT_TRUE(report->Get("SV42"));
  auto video_stats = report->Get("SV42")->cast_to<RTCVideoSourceStats>();
  EXPECT_FALSE(video_stats.frames_per_second.has_value());
  EXPECT_FALSE(video_stats.frames.has_value());
}

// The track not having a source is not expected to be true in practise, but
// this is true in some tests relying on fakes. This test covers that code path.
TEST_P(RTCStatsCollectorTest,
       RTCVideoSourceStatsMissingResolutionWhenTrackHasNoSource) {
  const uint32_t kSsrc = 4;
  const int kAttachmentId = 42;

  VideoMediaInfo video_media_info;
  video_media_info.senders.push_back(VideoSenderInfo());
  video_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
  video_media_info.senders[0].local_stats[0].ssrc = kSsrc;
  video_media_info.senders[0].framerate_input = 29.0;
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  auto video_track = CreateFakeVideoTrackForStats(
      "LocalVideoTrackID", MediaStreamTrackInterface::kLive,
      /*source=*/nullptr);
  scoped_refptr<MockRtpSenderInternal> sender =
      CreateMockSender(MediaType::VIDEO, video_track, kSsrc, kAttachmentId, {});
  EXPECT_CALL(*sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
    return nullptr;
  });
  EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
  EXPECT_CALL(*sender, SetSendCodecs(_));
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddSender(sender);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  ASSERT_TRUE(report->Get("SV42"));
  auto video_stats = report->Get("SV42")->cast_to<RTCVideoSourceStats>();
  EXPECT_FALSE(video_stats.width.has_value());
  EXPECT_FALSE(video_stats.height.has_value());
}

TEST_P(RTCStatsCollectorTest,
       RTCAudioSourceStatsNotCollectedForSenderWithoutTrack) {
  const uint32_t kSsrc = 4;
  const int kAttachmentId = 42;

  VoiceMediaInfo voice_media_info;
  voice_media_info.senders.push_back(VoiceSenderInfo());
  voice_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
  voice_media_info.senders[0].local_stats[0].ssrc = kSsrc;
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVoiceChannel("AudioMid", "TransportName", voice_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();
  scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
      MediaType::AUDIO, /*track=*/nullptr, kSsrc, kAttachmentId, {});
  EXPECT_CALL(*sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
    return nullptr;
  });
  EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
  EXPECT_CALL(*sender, SetSendCodecs(_));
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddSender(sender);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  EXPECT_FALSE(report->Get("SA42"));
}

// Parameterized tests on webrtc::MediaType (audio or video) and a boolean
// indicating if the stats timestamp should use the environment clock.
class RTCStatsCollectorTestWithParamKind
    : public RTCStatsCollectorTestBase,
      public ::testing::WithParamInterface<std::tuple<MediaType, bool>> {
 public:
  RTCStatsCollectorTestWithParamKind() : media_type_(std::get<0>(GetParam())) {
    SetStatsTimestampWithEnvironmentClock(std::get<1>(GetParam()));
  }

  std::string MediaTypeCharStr() const {
    switch (media_type_) {
      case MediaType::AUDIO:
        return "A";
      case MediaType::VIDEO:
        return "V";
      default:
        RTC_DCHECK_NOTREACHED();
        return "?";
    }
  }

  std::string MediaTypeKind() const {
    switch (media_type_) {
      case MediaType::AUDIO:
        return "audio";
      case MediaType::VIDEO:
        return "video";
      default:
        RTC_DCHECK_NOTREACHED();
        return "";
    }
  }

  // Adds a sender and channel of the appropriate kind, creating a sender info
  // with the report block's `source_ssrc` and report block data.
  void AddSenderInfoAndMediaChannel(
      std::string transport_name,
      const std::vector<ReportBlockData>& report_block_datas,
      std::optional<RtpCodecParameters> codec) {
    switch (media_type_) {
      case MediaType::AUDIO: {
        VoiceMediaInfo voice_media_info;
        for (const auto& report_block_data : report_block_datas) {
          VoiceSenderInfo sender;
          sender.local_stats.push_back(SsrcSenderInfo());
          sender.local_stats[0].ssrc = report_block_data.source_ssrc();
          if (codec.has_value()) {
            sender.codec_payload_type = codec->payload_type;
            voice_media_info.send_codecs.insert(
                std::make_pair(codec->payload_type, *codec));
          }
          sender.report_block_datas.push_back(report_block_data);
          voice_media_info.senders.push_back(sender);
        }
        RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
        pc_->AddVoiceChannel("mid", transport_name, voice_media_info);
        RTC_ALLOW_PLAN_B_DEPRECATION_END();
        return;
      }
      case MediaType::VIDEO: {
        VideoMediaInfo video_media_info;
        for (const auto& report_block_data : report_block_datas) {
          VideoSenderInfo sender;
          sender.local_stats.push_back(SsrcSenderInfo());
          sender.local_stats[0].ssrc = report_block_data.source_ssrc();
          if (codec.has_value()) {
            sender.codec_payload_type = codec->payload_type;
            video_media_info.send_codecs.insert(
                std::make_pair(codec->payload_type, *codec));
          }
          sender.report_block_datas.push_back(report_block_data);
          video_media_info.aggregated_senders.push_back(sender);
          video_media_info.senders.push_back(sender);
        }
        RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
        pc_->AddVideoChannel("mid", transport_name, video_media_info);
        RTC_ALLOW_PLAN_B_DEPRECATION_END();
        return;
      }
      case MediaType::DATA:
      default:
        RTC_DCHECK_NOTREACHED();
    }
  }

 protected:
  MediaType media_type_;
};

// Verifies RTCRemoteInboundRtpStreamStats members that don't require
// RTCCodecStats (codecId, jitter) and without setting up an RTCP transport.
TEST_P(RTCStatsCollectorTestWithParamKind,
       RTCRemoteInboundRtpStreamStatsCollectedFromReportBlock) {
  constexpr TimeDelta kReportBlockTimestampUtcOffset =
      TimeDelta::Micros(123456789);
  const Timestamp kReportBlockTimestampUtc =
      time_controller_.GetClock()->CurrentTime() +
      kReportBlockTimestampUtcOffset;
  const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);
  const uint8_t kFractionLost = 12;
  const TimeDelta kRoundTripTimeSample1 = TimeDelta::Millis(1'234);
  const TimeDelta kRoundTripTimeSample2 = TimeDelta::Seconds(13);

  // The report block's timestamp cannot be from the future, set the fake clock
  // to match.
  time_controller_.AdvanceTime(kReportBlockTimestampUtcOffset);
  uint32_t ssrcs[] = {12, 13};
  std::vector<ReportBlockData> report_block_datas;
  Call::Stats call_stats;
  for (uint32_t ssrc : ssrcs) {
    rtcp::ReportBlock report_block;
    // The remote-inbound-rtp SSRC and the outbound-rtp SSRC is the same as the
    // `source_ssrc`, "SSRC of the RTP packet sender".
    report_block.SetMediaSsrc(ssrc);
    report_block.SetCumulativeLost(7);
    report_block.SetFractionLost(kFractionLost);
    ReportBlockData report_block_data;
    report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
                                     kReportBlockTimestamp);
    report_block_data.AddRoundTripTimeSample(kRoundTripTimeSample1);
    // Only the last sample should be exposed as the
    // `RTCRemoteInboundRtpStreamStats::round_trip_time`.
    report_block_data.AddRoundTripTimeSample(kRoundTripTimeSample2);
    report_block_datas.push_back(report_block_data);

    call_stats.received_ccfb_stats_per_ssrc[ssrc] = {
        .num_packets_received_with_ect1 = 100,
        .num_packets_received_with_ce = 10,
        .num_packets_reported_as_lost = 30,
        .num_packets_reported_as_lost_but_recovered = 20,
        .num_packets_with_bleached_ect1_marking = 1};
  }
  AddSenderInfoAndMediaChannel("TransportName", report_block_datas,
                               std::nullopt);
  pc_->SetCallStats(call_stats);

  const Timestamp expected_timestamp = std::get<1>(GetParam())
                                           ? kReportBlockTimestamp
                                           : kReportBlockTimestampUtc;
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  for (uint32_t ssrc : ssrcs) {
    std::string stream_id = "" + std::to_string(ssrc);
    RTCRemoteInboundRtpStreamStats expected_remote_inbound_rtp(
        "RI" + MediaTypeCharStr() + stream_id, expected_timestamp);
    expected_remote_inbound_rtp.ssrc = ssrc;
    expected_remote_inbound_rtp.fraction_lost =
        static_cast<double>(kFractionLost) / (1 << 8);
    expected_remote_inbound_rtp.kind = MediaTypeKind();
    expected_remote_inbound_rtp.transport_id =
        "TTransportName1";  // 1 for RTP (we have no RTCP
                            // transport)
    expected_remote_inbound_rtp.packets_lost = 7;
    expected_remote_inbound_rtp.local_id =
        "OTTransportName1" + MediaTypeCharStr() + stream_id;
    expected_remote_inbound_rtp.round_trip_time =
        kRoundTripTimeSample2.seconds<double>();
    expected_remote_inbound_rtp.total_round_trip_time =
        (kRoundTripTimeSample1 + kRoundTripTimeSample2).seconds<double>();
    expected_remote_inbound_rtp.round_trip_time_measurements = 2;
    expected_remote_inbound_rtp.packets_received_with_ect1 = 100;
    expected_remote_inbound_rtp.packets_received_with_ce = 10;
    expected_remote_inbound_rtp.packets_reported_as_lost = 30;
    expected_remote_inbound_rtp.packets_reported_as_lost_but_recovered = 20;
    expected_remote_inbound_rtp.packets_with_bleached_ect1_marking = 1;
    // This test does not set up RTCCodecStats, so `codec_id` and `jitter` are
    // expected to be missing. These are tested separately.

    ASSERT_TRUE(report->Get(expected_remote_inbound_rtp.id()));
    const auto& remote_inbound_rtp =
        report->Get(expected_remote_inbound_rtp.id())
            ->cast_to<RTCRemoteInboundRtpStreamStats>();
    EXPECT_EQ(remote_inbound_rtp, expected_remote_inbound_rtp);
    EXPECT_EQ(remote_inbound_rtp.timestamp(), expected_timestamp);
    EXPECT_TRUE(report->Get(*expected_remote_inbound_rtp.transport_id));
    ASSERT_TRUE(report->Get(*expected_remote_inbound_rtp.local_id));
    // Lookup works in both directions.
    EXPECT_EQ(*report->Get(*expected_remote_inbound_rtp.local_id)
                   ->cast_to<RTCOutboundRtpStreamStats>()
                   .remote_id,
              expected_remote_inbound_rtp.id());
  }
}

TEST_P(RTCStatsCollectorTestWithParamKind,
       RTCRemoteInboundRtpStreamStatsRttMissingBeforeMeasurement) {
  const Timestamp kReportBlockTimestampUtc = Timestamp::Micros(123456789);
  const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);

  rtcp::ReportBlock report_block;
  // The remote-inbound-rtp SSRC and the outbound-rtp SSRC is the same as the
  // `source_ssrc`, "SSRC of the RTP packet sender".
  report_block.SetMediaSsrc(12);
  ReportBlockData report_block_data;  // AddRoundTripTimeSample() not called.
  report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
                                   kReportBlockTimestamp);

  AddSenderInfoAndMediaChannel("TransportName", {report_block_data},
                               std::nullopt);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  std::string remote_inbound_rtp_id = "RI" + MediaTypeCharStr() + "12";
  ASSERT_TRUE(report->Get(remote_inbound_rtp_id));
  auto& remote_inbound_rtp = report->Get(remote_inbound_rtp_id)
                                 ->cast_to<RTCRemoteInboundRtpStreamStats>();

  EXPECT_TRUE(remote_inbound_rtp.round_trip_time_measurements.has_value());
  EXPECT_EQ(0, *remote_inbound_rtp.round_trip_time_measurements);
  EXPECT_FALSE(remote_inbound_rtp.round_trip_time.has_value());
}

TEST_P(RTCStatsCollectorTestWithParamKind,
       RTCRemoteInboundRtpStreamStatsWithTimestampFromReportBlock) {
  const Timestamp kReportBlockTimestampUtc = Timestamp::Micros(123456789);
  const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);
  time_controller_.AdvanceTime(kReportBlockTimestampUtc -
                               time_controller_.GetClock()->CurrentTime());

  rtcp::ReportBlock report_block;
  // The remote-inbound-rtp SSRC and the outbound-rtp SSRC is the same as the
  // `source_ssrc`, "SSRC of the RTP packet sender".
  report_block.SetMediaSsrc(12);
  ReportBlockData report_block_data;
  report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
                                   kReportBlockTimestamp);

  AddSenderInfoAndMediaChannel("TransportName", {report_block_data},
                               std::nullopt);

  // Advance time, it should be OK to have fresher reports than report blocks.
  time_controller_.AdvanceTime(TimeDelta::Micros(1234));

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  std::string remote_inbound_rtp_id = "RI" + MediaTypeCharStr() + "12";
  ASSERT_TRUE(report->Get(remote_inbound_rtp_id));
  auto& remote_inbound_rtp = report->Get(remote_inbound_rtp_id)
                                 ->cast_to<RTCRemoteInboundRtpStreamStats>();

  // Even though the report time is different, the remote-inbound-rtp timestamp
  // is of the time that the report block was received.
  EXPECT_EQ(report->timestamp(),
            kReportBlockTimestampUtc + TimeDelta::Micros(1234));
  if (std::get<1>(GetParam())) {
    EXPECT_EQ(remote_inbound_rtp.timestamp(), kReportBlockTimestamp);
  } else {
    EXPECT_EQ(remote_inbound_rtp.timestamp(), kReportBlockTimestampUtc);
  }
}

TEST_P(RTCStatsCollectorTestWithParamKind,
       RTCRemoteInboundRtpStreamStatsWithCodecBasedMembers) {
  const Timestamp kReportBlockTimestampUtc = Timestamp::Micros(123456789);
  const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);
  time_controller_.AdvanceTime(kReportBlockTimestampUtc -
                               time_controller_.GetClock()->CurrentTime());

  rtcp::ReportBlock report_block;
  // The remote-inbound-rtp SSRC and the outbound-rtp SSRC is the same as the
  // `source_ssrc`, "SSRC of the RTP packet sender".
  report_block.SetMediaSsrc(12);
  report_block.SetJitter(5000);
  ReportBlockData report_block_data;
  report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
                                   kReportBlockTimestamp);

  RtpCodecParameters codec;
  codec.payload_type = 3;
  codec.kind = media_type_;
  codec.clock_rate = 1000;

  AddSenderInfoAndMediaChannel("TransportName", {report_block_data}, codec);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  std::string remote_inbound_rtp_id = "RI" + MediaTypeCharStr() + "12";
  ASSERT_TRUE(report->Get(remote_inbound_rtp_id));
  auto& remote_inbound_rtp = report->Get(remote_inbound_rtp_id)
                                 ->cast_to<RTCRemoteInboundRtpStreamStats>();

  EXPECT_TRUE(remote_inbound_rtp.codec_id.has_value());
  EXPECT_TRUE(report->Get(*remote_inbound_rtp.codec_id));

  EXPECT_TRUE(remote_inbound_rtp.jitter.has_value());
  // The jitter (in seconds) is the report block's jitter divided by the codec's
  // clock rate.
  EXPECT_EQ(5.0, *remote_inbound_rtp.jitter);
}

TEST_P(RTCStatsCollectorTestWithParamKind,
       RTCRemoteInboundRtpStreamStatsWithRtcpTransport) {
  const Timestamp kReportBlockTimestampUtc = Timestamp::Micros(123456789);
  const Timestamp kReportBlockTimestamp = Timestamp::Micros(12345678);
  time_controller_.AdvanceTime(kReportBlockTimestampUtc -
                               time_controller_.GetClock()->CurrentTime());

  rtcp::ReportBlock report_block;
  // The remote-inbound-rtp SSRC and the outbound-rtp SSRC is the same as the
  // `source_ssrc`, "SSRC of the RTP packet sender".
  report_block.SetMediaSsrc(12);
  ReportBlockData report_block_data;
  report_block_data.SetReportBlock(0, report_block, kReportBlockTimestampUtc,
                                   kReportBlockTimestamp);

  TransportChannelStats rtp_transport_channel_stats;
  rtp_transport_channel_stats.component = ICE_CANDIDATE_COMPONENT_RTP;
  rtp_transport_channel_stats.dtls_state = DtlsTransportState::kNew;
  TransportChannelStats rtcp_transport_channel_stats;
  rtcp_transport_channel_stats.component = ICE_CANDIDATE_COMPONENT_RTCP;
  rtcp_transport_channel_stats.dtls_state = DtlsTransportState::kNew;
  pc_->SetTransportStats("TransportName", {rtp_transport_channel_stats,
                                           rtcp_transport_channel_stats});
  AddSenderInfoAndMediaChannel("TransportName", {report_block_data},
                               std::nullopt);

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  std::string remote_inbound_rtp_id = "RI" + MediaTypeCharStr() + "12";
  ASSERT_TRUE(report->Get(remote_inbound_rtp_id));
  auto& remote_inbound_rtp = report->Get(remote_inbound_rtp_id)
                                 ->cast_to<RTCRemoteInboundRtpStreamStats>();

  EXPECT_TRUE(remote_inbound_rtp.transport_id.has_value());
  EXPECT_EQ("TTransportName2",  // 2 for RTCP
            *remote_inbound_rtp.transport_id);
  EXPECT_TRUE(report->Get(*remote_inbound_rtp.transport_id));
}

INSTANTIATE_TEST_SUITE_P(All, RTCStatsCollectorTest, ::testing::Bool());

INSTANTIATE_TEST_SUITE_P(All,
                         RTCStatsCollectorTestWithParamKind,
                         ::testing::Combine(::testing::Values(MediaType::AUDIO,
                                                              MediaType::VIDEO),
                                            ::testing::Bool()));

// Checks that no remote outbound stats are collected if not available in
// `VoiceMediaInfo`.
TEST_P(RTCStatsCollectorTest,
       RTCRemoteOutboundRtpAudioStreamStatsNotCollected) {
  ExampleStatsGraph graph =
      SetupExampleStatsVoiceGraph(/*add_remote_outbound_stats=*/false);
  EXPECT_FALSE(graph.full_report->Get(graph.remote_outbound_rtp_id));
  // Also check that no other remote outbound report is created (in case the
  // expected ID is incorrect).
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  ASSERT_NE(report->begin(), report->end())
      << "No reports have been generated.";
  for (const auto& stats : *report) {
    SCOPED_TRACE(stats.id());
    EXPECT_NE(stats.type(), RTCRemoteOutboundRtpStreamStats::kType);
  }
}

// Checks that the remote outbound stats are collected when available in
// `VoiceMediaInfo`.
TEST_P(RTCStatsCollectorTest, RTCRemoteOutboundRtpAudioStreamStatsCollected) {
  ExampleStatsGraph graph =
      SetupExampleStatsVoiceGraph(/*add_remote_outbound_stats=*/true);
  ASSERT_TRUE(graph.full_report->Get(graph.remote_outbound_rtp_id));
  const auto& remote_outbound_rtp =
      graph.full_report->Get(graph.remote_outbound_rtp_id)
          ->cast_to<RTCRemoteOutboundRtpStreamStats>();
  EXPECT_EQ(remote_outbound_rtp.timestamp(), kRemoteOutboundStatsTimestamp);
  EXPECT_FLOAT_EQ(*remote_outbound_rtp.remote_timestamp,
                  kRemoteOutboundStatsRemoteTimestamp.ms<double>());
  EXPECT_EQ(*remote_outbound_rtp.packets_sent, kRemoteOutboundStatsPacketsSent);
  EXPECT_EQ(*remote_outbound_rtp.bytes_sent, kRemoteOutboundStatsBytesSent);
  EXPECT_EQ(*remote_outbound_rtp.reports_sent,
            kRemoteOutboundStatsReportsCount);
}

TEST_P(RTCStatsCollectorTest,
       RTCVideoSourceStatsNotCollectedForSenderWithoutTrack) {
  const uint32_t kSsrc = 4;
  const int kAttachmentId = 42;

  VideoMediaInfo video_media_info;
  video_media_info.senders.push_back(VideoSenderInfo());
  video_media_info.senders[0].local_stats.push_back(SsrcSenderInfo());
  video_media_info.senders[0].local_stats[0].ssrc = kSsrc;
  video_media_info.senders[0].framerate_input = 29.0;
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddVideoChannel("VideoMid", "TransportName", video_media_info);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  scoped_refptr<MockRtpSenderInternal> sender = CreateMockSender(
      MediaType::VIDEO, /*track=*/nullptr, kSsrc, kAttachmentId, {});
  EXPECT_CALL(*sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
    return nullptr;
  });
  EXPECT_CALL(*sender, SetMediaChannel(_)).WillRepeatedly(Return());
  EXPECT_CALL(*sender, SetSendCodecs(_));
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddSender(sender);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  EXPECT_FALSE(report->Get("SV42"));
}

// Test collecting echo return loss stats from the audio processor attached to
// the track, rather than the voice sender info.
TEST_P(RTCStatsCollectorTest, CollectEchoReturnLossFromTrackAudioProcessor) {
  scoped_refptr<MediaStream> local_stream =
      MediaStream::Create("LocalStreamId");
  pc_->mutable_local_streams()->AddStream(local_stream);

  // Local audio track
  scoped_refptr<MediaStreamTrackInterface> local_audio_track = CreateFakeTrack(
      MediaType::AUDIO, "LocalAudioTrackID", MediaStreamTrackInterface::kEnded,
      /*create_fake_audio_processor=*/true);
  local_stream->AddTrack(scoped_refptr<AudioTrackInterface>(
      static_cast<AudioTrackInterface*>(local_audio_track.get())));

  VoiceSenderInfo voice_sender_info_ssrc1;
  voice_sender_info_ssrc1.local_stats.push_back(SsrcSenderInfo());
  voice_sender_info_ssrc1.local_stats[0].ssrc = 1;

  stats_->CreateMockRtpSendersReceiversAndChannels(
      {std::make_pair(local_audio_track.get(), voice_sender_info_ssrc1)}, {},
      {}, {}, {local_stream->id()}, {});

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  RTCAudioSourceStats expected_audio("SA11", report->timestamp());
  expected_audio.track_identifier = "LocalAudioTrackID";
  expected_audio.kind = "audio";
  expected_audio.audio_level = 0;
  expected_audio.total_audio_energy = 0;
  expected_audio.total_samples_duration = 0;
  expected_audio.echo_return_loss = 2.0;
  expected_audio.echo_return_loss_enhancement = 3.0;

  ASSERT_TRUE(report->Get(expected_audio.id()));
  EXPECT_EQ(report->Get(expected_audio.id())->cast_to<RTCAudioSourceStats>(),
            expected_audio);
}

TEST_P(RTCStatsCollectorTest, GetStatsWithSenderSelector) {
  ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
  // Expected stats graph when filtered by sender:
  //
  //  media-source
  //    ^
  //    |
  //    +--------- outbound-rtp
  //                |        |
  //                v        v
  //       codec (send)     transport
  scoped_refptr<const RTCStatsReport> sender_report;
  stats_->stats_collector().GetStatsReport(
      graph.sender,
      RTCStatsObtainer::Create(&sender_report, [&] { main_thread_.Quit(); }));
  main_thread_.Run();
  EXPECT_TRUE(sender_report);
  EXPECT_EQ(sender_report->timestamp(), graph.full_report->timestamp());
  EXPECT_EQ(sender_report->size(), 4u);
  EXPECT_TRUE(sender_report->Get(graph.send_codec_id));
  EXPECT_FALSE(sender_report->Get(graph.recv_codec_id));
  EXPECT_TRUE(sender_report->Get(graph.outbound_rtp_id));
  EXPECT_FALSE(sender_report->Get(graph.inbound_rtp_id));
  EXPECT_TRUE(sender_report->Get(graph.transport_id));
  EXPECT_FALSE(sender_report->Get(graph.peer_connection_id));
  EXPECT_TRUE(sender_report->Get(graph.media_source_id));
}

TEST_P(RTCStatsCollectorTest, GetStatsWithReceiverSelector) {
  ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
  // Expected stats graph when filtered by receiver:
  //
  //
  //
  //                              inbound-rtp
  //                               |       |
  //                               v       v
  //                        transport     codec (recv)
  scoped_refptr<const RTCStatsReport> receiver_report;
  stats_->stats_collector().GetStatsReport(
      graph.receiver,
      RTCStatsObtainer::Create(&receiver_report, [&] { main_thread_.Quit(); }));
  main_thread_.Run();
  EXPECT_TRUE(receiver_report);
  EXPECT_EQ(receiver_report->size(), 3u);
  EXPECT_EQ(receiver_report->timestamp(), graph.full_report->timestamp());
  EXPECT_FALSE(receiver_report->Get(graph.send_codec_id));
  EXPECT_TRUE(receiver_report->Get(graph.recv_codec_id));
  EXPECT_FALSE(receiver_report->Get(graph.outbound_rtp_id));
  EXPECT_TRUE(receiver_report->Get(graph.inbound_rtp_id));
  EXPECT_TRUE(receiver_report->Get(graph.transport_id));
  EXPECT_FALSE(receiver_report->Get(graph.peer_connection_id));
  EXPECT_FALSE(receiver_report->Get(graph.media_source_id));
}

TEST_P(RTCStatsCollectorTest,
       GetStatsDoesNotBlockWhenFetchingSenderParameters) {
  // We must create a new FakePeerConnectionForStats with a separate worker
  // thread because the test base uses the current thread for all three, which
  // bypasses the thread hop and avoids the
  // RTC_DCHECK_DISALLOW_THREAD_BLOCKING_CALLS check.
  auto worker_thread = Thread::Create();
  worker_thread->Start();
  pc_ = make_ref_counted<FakePeerConnectionForStats>(env_, worker_thread.get());
  SetStatsTimestampWithEnvironmentClock(GetParam());
  // Create a track and a real sender to ensure we test the actual blocking
  // behavior in RtpSenderBase::GetParametersInternal.
  scoped_refptr<MediaStreamTrackInterface> track = CreateFakeTrack(
      MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive);

  auto fake_media_channel = std::make_unique<FakeVoiceMediaSendChannel>(
      webrtc::AudioOptions(), pc_->network_thread());

  scoped_refptr<AudioRtpSender> sender = AudioRtpSender::Create(
      env_, pc_->signaling_thread(), pc_->worker_thread(), "sender_id",
      /*stats=*/nullptr, /*set_streams_observer=*/nullptr,
      /*enable_sframe_at_owner=*/nullptr,
      /*media_channel=*/nullptr);

  {
    ScopedOperationsBatcher worker_tasks(pc_->worker_thread());
    worker_tasks.Add([&]() {
      fake_media_channel->AddSendStream(StreamParams::CreateLegacy(1234));
      sender->SetMediaChannel(fake_media_channel.get());
    });
    worker_tasks.AddWithFinalizer(sender->SetSsrcTask(1234));
  }
  sender->SetTrack(track.get());

  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddSender(sender);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  scoped_refptr<const RTCStatsReport> report;
  stats_->stats_collector().GetStatsReport(
      RTCStatsObtainer::Create(&report, [this] { main_thread_.Quit(); }));
  main_thread_.Run();

  sender->Stop();
  sender = nullptr;
  stats_.reset();
  pc_ = nullptr;
}

TEST_P(RTCStatsCollectorTest, GetStatsWithNullSenderSelector) {
  ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
  scoped_refptr<const RTCStatsReport> empty_report;
  stats_->stats_collector().GetStatsReport(
      scoped_refptr<RtpSenderInternal>(nullptr),
      RTCStatsObtainer::Create(&empty_report, [&] { main_thread_.Quit(); }));
  main_thread_.Run();
  EXPECT_TRUE(empty_report);
  EXPECT_EQ(empty_report->timestamp(), graph.full_report->timestamp());
  EXPECT_EQ(empty_report->size(), 0u);
}

TEST_P(RTCStatsCollectorTest, GetStatsWithNullReceiverSelector) {
  ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
  scoped_refptr<const RTCStatsReport> empty_report;
  stats_->stats_collector().GetStatsReport(
      scoped_refptr<RtpReceiverInternal>(nullptr),
      RTCStatsObtainer::Create(&empty_report, [&] { main_thread_.Quit(); }));
  main_thread_.Run();
  EXPECT_TRUE(empty_report);
  EXPECT_EQ(empty_report->timestamp(), graph.full_report->timestamp());
  EXPECT_EQ(empty_report->size(), 0u);
}

// Before SetLocalDescription() senders don't have an SSRC.
// To simulate this case we create a mock sender with SSRC=0.
TEST_P(RTCStatsCollectorTest, RtpIsMissingWhileSsrcIsZero) {
  scoped_refptr<MediaStreamTrackInterface> track = CreateFakeTrack(
      MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive);
  scoped_refptr<MockRtpSenderInternal> sender =
      CreateMockSender(MediaType::AUDIO, track, 0, 49, {});
  EXPECT_CALL(*sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
    return nullptr;
  });
  EXPECT_CALL(*sender, SetSendCodecs(_));
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddSender(sender);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);

  auto outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
  EXPECT_TRUE(outbound_rtps.empty());
}

// We may also be in a case where the SSRC has been assigned but no
// `voice_sender_info` stats exist yet.
TEST_P(RTCStatsCollectorTest, DoNotCrashIfSsrcIsKnownButInfosAreStillMissing) {
  scoped_refptr<MediaStreamTrackInterface> track = CreateFakeTrack(
      MediaType::AUDIO, "audioTrack", MediaStreamTrackInterface::kLive);
  scoped_refptr<MockRtpSenderInternal> sender =
      CreateMockSender(MediaType::AUDIO, track, 4711, 49, {});
  EXPECT_CALL(*sender, DetachTrackAndGetStopTask()).WillRepeatedly([] {
    return nullptr;
  });
  EXPECT_CALL(*sender, SetSendCodecs(_));
  RTC_ALLOW_PLAN_B_DEPRECATION_BEGIN();
  pc_->AddSender(sender);
  RTC_ALLOW_PLAN_B_DEPRECATION_END();

  // We do not generate any matching voice_sender_info stats.
  scoped_refptr<const RTCStatsReport> report =
      stats_->GetStatsReport(main_thread_);
  auto outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
  EXPECT_TRUE(outbound_rtps.empty());
}

// Used for test below, to test calling GetStatsReport during a callback.
class RecursiveCallback : public RTCStatsCollectorCallback {
 public:
  explicit RecursiveCallback(RTCStatsCollectorWrapper* stats,
                             RunLoop& loop,
                             absl::AnyInvocable<void() &&> on_complete)
      : stats_(stats), loop_(loop), on_complete_(std::move(on_complete)) {}

  void OnStatsDelivered(
      const scoped_refptr<const RTCStatsReport>& report) override {
    stats_->GetStatsReport(loop_);
    called_ = true;
    if (on_complete_)
      std::move(on_complete_)();
  }

  bool called() const { return called_; }

 private:
  RTCStatsCollectorWrapper* stats_;
  RunLoop& loop_;
  absl::AnyInvocable<void() &&> on_complete_;
  bool called_ = false;
};

// Test that nothing bad happens if a callback causes GetStatsReport to be
// called again recursively. Regression test for crbug.com/webrtc/8973.
TEST_P(RTCStatsCollectorTest, DoNotCrashWhenGetStatsCalledDuringCallback) {
  int remaining = 2;
  auto on_complete = [&] {
    if (--remaining == 0)
      main_thread_.Quit();
  };
  auto callback1 = make_ref_counted<RecursiveCallback>(
      stats_.get(), main_thread_, on_complete);
  auto callback2 = make_ref_counted<RecursiveCallback>(
      stats_.get(), main_thread_, on_complete);
  stats_->stats_collector().GetStatsReport(callback1);
  stats_->stats_collector().GetStatsReport(callback2);
  main_thread_.Run();
  EXPECT_TRUE(callback1->called());
  EXPECT_TRUE(callback2->called());
}

class RTCTestStats : public RTCStats {
 public:
  WEBRTC_RTCSTATS_DECL(RTCTestStats);

  RTCTestStats(const std::string& id, Timestamp timestamp)
      : RTCStats(id, timestamp) {}

  std::optional<int32_t> dummy_stat;
};

WEBRTC_RTCSTATS_IMPL(RTCTestStats,
                     RTCStats,
                     "test-stats",
                     AttributeInit("dummyStat", &dummy_stat))

// Overrides the stats collection to verify thread usage and that the resulting
// partial reports are merged.
class FakeRTCStatsCollector final : public RTCStatsCollector {
 public:
  static std::unique_ptr<FakeRTCStatsCollector> Create(
      PeerConnectionInternal* pc,
      const Environment& env,
      int64_t cache_lifetime_us = 50 * kNumMicrosecsPerMillisec) {
    return std::make_unique<FakeRTCStatsCollector>(pc, env, cache_lifetime_us);
  }

  FakeRTCStatsCollector(PeerConnectionInternal* pc,
                        const Environment& env,
                        int64_t cache_lifetime_us)
      : RTCStatsCollector(pc, env, cache_lifetime_us),
        signaling_thread_(pc->signaling_thread()),
        network_thread_(pc->network_thread()),
        callback_(make_ref_counted<Callback>(this)) {}

  class Callback : public RTCStatsCollectorCallback {
   public:
    explicit Callback(FakeRTCStatsCollector* collector)
        : flag_(collector->safety_.flag()), collector_(collector) {}

    void OnStatsDelivered(
        const scoped_refptr<const RTCStatsReport>& report) override {
      if (!flag_->alive())
        return;
      collector_->OnStatsDelivered(report);
    }

   private:
    scoped_refptr<PendingTaskSafetyFlag> flag_;
    FakeRTCStatsCollector* collector_;
  };

  scoped_refptr<Callback> callback() { return callback_; }

  void OnStatsDelivered(const scoped_refptr<const RTCStatsReport>& report) {
    EXPECT_TRUE(signaling_thread_->IsCurrent());
    MutexLock lock(&lock_);
    delivered_report_ = report;
  }

  bool HasVerifiedResults() {
    EXPECT_TRUE(signaling_thread_->IsCurrent());
    MutexLock lock(&lock_);
    if (!delivered_report_)
      return false;
    EXPECT_EQ(produced_on_signaling_thread_, 1);
    EXPECT_TRUE(delivered_report_->Get("SignalingThreadStats"));

    produced_on_signaling_thread_ = 0;
    delivered_report_ = nullptr;
    return true;
  }

 protected:
  void ProducePartialResultsOnSignalingThreadImpl(
      Timestamp timestamp,
      const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
      const std::vector<TransceiverReferences>& transceiver_references,
      const std::optional<AudioDeviceModule::Stats>& audio_device_stats,
      RTCStatsReport* partial_report) override {
    EXPECT_TRUE(signaling_thread_->IsCurrent());
    {
      MutexLock lock(&lock_);
      EXPECT_FALSE(delivered_report_);
      ++produced_on_signaling_thread_;
    }

    partial_report->AddStats(std::unique_ptr<const RTCStats>(
        new RTCTestStats("SignalingThreadStats", timestamp)));
  }

 private:
  Thread* const signaling_thread_;
  Thread* const network_thread_;
  ScopedTaskSafety safety_;
  scoped_refptr<Callback> callback_;

  Mutex lock_;
  scoped_refptr<const RTCStatsReport> delivered_report_;
  int produced_on_signaling_thread_ = 0;
};

// Simple test that verifies that GetStatsReport() can be called and async
// results delivered on the same thread.
// This covers the following steps:
// * Request stats
// * Task posted from signaling to network thread.
// * Task posted from network thread back to signaling
// * Issue the `OnStatsDelivered` callback on the signaling thread.
TEST(RTCStatsCollectorSafetyTest, WaitPendingRequestGetsCallback) {
  RunLoop loop;
  auto env = CreateTestEnvironment();
  auto pc = make_ref_counted<FakePeerConnectionForStats>(env);
  RTCStatsCollectorWrapper wrapper(pc, env);
  auto callback = make_ref_counted<MockStatsCollectorCallback>();
  EXPECT_CALL(*callback, OnStatsDelivered(_)).WillOnce([&] { loop.Quit(); });
  wrapper.stats_collector().GetStatsReport(callback);
  loop.Run();
}

// Similar to WaitPendingRequestGetsCallback except that we call
// CancelPendingRequestAndGetShutdownTask to make sure the callback tasks won't
// run (and callbacks not issued). This covers the following steps:
// * Request stats
// * Task posted from signaling to network thread.
// * Task posted from network thread back to signaling
// * The task for the signaling thread will be dropped, no callback.
TEST(RTCStatsCollectorSafetyTest, CancelPendingRequestReturnsImmediately) {
  RunLoop loop;
  std::unique_ptr<Thread> worker_and_network = Thread::Create();
  worker_and_network->Start();

  auto env = CreateTestEnvironment();
  auto pc = make_ref_counted<FakePeerConnectionForStats>(
      env, worker_and_network.get(), worker_and_network.get());
  RTCStatsCollectorWrapper wrapper(pc, env);
  auto callback = make_ref_counted<MockStatsCollectorCallback>();

  // The callback that should not be called.
  EXPECT_CALL(*callback, OnStatsDelivered(_)).Times(0);

  // Let's pause the network/worker threads.
  Event blocker;
  worker_and_network->PostTask([&]() { blocker.Wait(Event::kForever); });

  // Issue the GetStats call. At this point, cancellation has not been made,
  // this posts a task to the worker/network threads which will be blocked by
  // the above task.
  wrapper.stats_collector().GetStatsReport(callback);
  ScopedOperationsBatcher network_tasks(worker_and_network.get());
  ScopedOperationsBatcher worker_tasks(worker_and_network.get());

  // Now cancel any ongoing stats gathering operations.
  // This should cancel the outstanding operations and invoke pending callbacks
  // immediately.
  wrapper.stats_collector().CancelPendingRequestAndGetShutdownTasks(
      network_tasks, worker_tasks);

  // Resume the network and worker threads.
  blocker.Set();

  // Run the cleanup tasks.
  network_tasks.Run();
  worker_tasks.Run();
}

// This covers the following steps:
// * Mark the network thread as not alive (via CancelPendingRequest)
// * Request stats
// * Task posted from signaling to network thread.
// * The task for the network thread will be dropped, no further work done.
TEST(RTCStatsCollectorSafetyTest, NetworkThreadSafetyPreventsCallback) {
  RunLoop loop;
  auto env = CreateTestEnvironment();
  auto pc = make_ref_counted<FakePeerConnectionForStats>(env);
  RTCStatsCollectorWrapper wrapper(pc, env);
  auto callback = make_ref_counted<MockStatsCollectorCallback>();
  EXPECT_CALL(*callback, OnStatsDelivered(_)).Times(0);
  // Start by canceling any ongoing tasks. There aren't actually any ongoing
  // tasks, but this gives us the network cleanup task.
  ScopedOperationsBatcher network_tasks(pc->network_thread());
  ScopedOperationsBatcher worker_tasks(pc->worker_thread());
  wrapper.stats_collector().CancelPendingRequestAndGetShutdownTasks(
      network_tasks, worker_tasks);
  // Clean up the state on the network thread. This will have the effect of
  // dropping any tasks targeting the network thread.
  network_tasks.Run();
  worker_tasks.Run();
  // Now, attempt to get a stats report. This will try to post a task to the
  // network thread, which will be dropped.
  wrapper.stats_collector().GetStatsReport(callback);

  loop.Flush();
}

TEST(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
  GlobalSimulatedTimeController time_controller{Timestamp::Seconds(1234)};
  Environment env = CreateTestEnvironment({.time = &time_controller});
  scoped_refptr<FakePeerConnectionForStats> pc =
      make_ref_counted<FakePeerConnectionForStats>(env);
  std::unique_ptr<FakeRTCStatsCollector> stats_collector =
      FakeRTCStatsCollector::Create(pc.get(), env);
  stats_collector->GetStatsReport(stats_collector->callback());
  time_controller.AdvanceTime(TimeDelta::Zero());
  EXPECT_TRUE(stats_collector->HasVerifiedResults());
}

}  // namespace

}  // namespace webrtc