Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 815 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
new file mode 100644
index 0000000000..ff482b39b6
--- /dev/null
+++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -0,0 +1,815 @@
+/*
+ *  Copyright (c) 2012 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 "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <cstdint>
+#include <memory>
+#include <set>
+#include <string>
+#include <utility>
+
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/units/time_delta.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
+#include "modules/rtp_rtcp/source/rtp_packet_history.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/time_util.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/ntp_time.h"
+
+#ifdef _WIN32
+// Disable warning C4355: 'this' : used in base member initializer list.
+#pragma warning(disable : 4355)
+#endif
+
+namespace webrtc {
+namespace {
+constexpr TimeDelta kDefaultExpectedRetransmissionTime = TimeDelta::Millis(125);
+constexpr TimeDelta kRttUpdateInterval = TimeDelta::Millis(1000);
+
+RTCPSender::Configuration AddRtcpSendEvaluationCallback(
+    RTCPSender::Configuration config,
+    std::function<void(TimeDelta)> send_evaluation_callback) {
+  config.schedule_next_rtcp_send_evaluation_function =
+      std::move(send_evaluation_callback);
+  return config;
+}
+
+RtpPacketHistory::PaddingMode GetPaddingMode(
+    const FieldTrialsView* field_trials) {
+  if (field_trials &&
+      field_trials->IsEnabled("WebRTC-PaddingMode-RecentLargePacket")) {
+    return RtpPacketHistory::PaddingMode::kRecentLargePacket;
+  }
+  return RtpPacketHistory::PaddingMode::kPriority;
+}
+
+}  // namespace
+
+ModuleRtpRtcpImpl2::RtpSenderContext::RtpSenderContext(
+    TaskQueueBase& worker_queue,
+    const RtpRtcpInterface::Configuration& config)
+    : packet_history(config.clock, GetPaddingMode(config.field_trials)),
+      sequencer(config.local_media_ssrc,
+                config.rtx_send_ssrc,
+                /*require_marker_before_media_padding=*/!config.audio,
+                config.clock),
+      packet_sender(config, &packet_history),
+      non_paced_sender(worker_queue, &packet_sender, &sequencer),
+      packet_generator(
+          config,
+          &packet_history,
+          config.paced_sender ? config.paced_sender : &non_paced_sender) {}
+
+ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
+    : worker_queue_(TaskQueueBase::Current()),
+      rtcp_sender_(AddRtcpSendEvaluationCallback(
+          RTCPSender::Configuration::FromRtpRtcpConfiguration(configuration),
+          [this](TimeDelta duration) {
+            ScheduleRtcpSendEvaluation(duration);
+          })),
+      rtcp_receiver_(configuration, this),
+      clock_(configuration.clock),
+      packet_overhead_(28),  // IPV4 UDP.
+      nack_last_time_sent_full_ms_(0),
+      nack_last_seq_number_sent_(0),
+      rtt_stats_(configuration.rtt_stats),
+      rtt_ms_(0) {
+  RTC_DCHECK(worker_queue_);
+  rtcp_thread_checker_.Detach();
+  if (!configuration.receiver_only) {
+    rtp_sender_ =
+        std::make_unique<RtpSenderContext>(*worker_queue_, configuration);
+    rtp_sender_->sequencing_checker.Detach();
+    // Make sure rtcp sender use same timestamp offset as rtp sender.
+    rtcp_sender_.SetTimestampOffset(
+        rtp_sender_->packet_generator.TimestampOffset());
+    rtp_sender_->packet_sender.SetTimestampOffset(
+        rtp_sender_->packet_generator.TimestampOffset());
+  }
+
+  // Set default packet size limit.
+  // TODO(nisse): Kind-of duplicates
+  // webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize.
+  const size_t kTcpOverIpv4HeaderSize = 40;
+  SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize);
+  rtt_update_task_ = RepeatingTaskHandle::DelayedStart(
+      worker_queue_, kRttUpdateInterval, [this]() {
+        PeriodicUpdate();
+        return kRttUpdateInterval;
+      });
+}
+
+ModuleRtpRtcpImpl2::~ModuleRtpRtcpImpl2() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  rtt_update_task_.Stop();
+}
+
+// static
+std::unique_ptr<ModuleRtpRtcpImpl2> ModuleRtpRtcpImpl2::Create(
+    const Configuration& configuration) {
+  RTC_DCHECK(configuration.clock);
+  RTC_DCHECK(TaskQueueBase::Current());
+  return std::make_unique<ModuleRtpRtcpImpl2>(configuration);
+}
+
+void ModuleRtpRtcpImpl2::SetRtxSendStatus(int mode) {
+  rtp_sender_->packet_generator.SetRtxStatus(mode);
+}
+
+int ModuleRtpRtcpImpl2::RtxSendStatus() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.RtxStatus() : kRtxOff;
+}
+
+void ModuleRtpRtcpImpl2::SetRtxSendPayloadType(int payload_type,
+                                               int associated_payload_type) {
+  rtp_sender_->packet_generator.SetRtxPayloadType(payload_type,
+                                                  associated_payload_type);
+}
+
+absl::optional<uint32_t> ModuleRtpRtcpImpl2::RtxSsrc() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : absl::nullopt;
+}
+
+absl::optional<uint32_t> ModuleRtpRtcpImpl2::FlexfecSsrc() const {
+  if (rtp_sender_) {
+    return rtp_sender_->packet_generator.FlexfecSsrc();
+  }
+  return absl::nullopt;
+}
+
+void ModuleRtpRtcpImpl2::IncomingRtcpPacket(
+    rtc::ArrayView<const uint8_t> rtcp_packet) {
+  RTC_DCHECK_RUN_ON(&rtcp_thread_checker_);
+  rtcp_receiver_.IncomingPacket(rtcp_packet);
+}
+
+void ModuleRtpRtcpImpl2::RegisterSendPayloadFrequency(int payload_type,
+                                                      int payload_frequency) {
+  rtcp_sender_.SetRtpClockRate(payload_type, payload_frequency);
+}
+
+int32_t ModuleRtpRtcpImpl2::DeRegisterSendPayload(const int8_t payload_type) {
+  return 0;
+}
+
+uint32_t ModuleRtpRtcpImpl2::StartTimestamp() const {
+  return rtp_sender_->packet_generator.TimestampOffset();
+}
+
+// Configure start timestamp, default is a random number.
+void ModuleRtpRtcpImpl2::SetStartTimestamp(const uint32_t timestamp) {
+  rtcp_sender_.SetTimestampOffset(timestamp);
+  rtp_sender_->packet_generator.SetTimestampOffset(timestamp);
+  rtp_sender_->packet_sender.SetTimestampOffset(timestamp);
+}
+
+uint16_t ModuleRtpRtcpImpl2::SequenceNumber() const {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  return rtp_sender_->sequencer.media_sequence_number();
+}
+
+// Set SequenceNumber, default is a random number.
+void ModuleRtpRtcpImpl2::SetSequenceNumber(const uint16_t seq_num) {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  if (rtp_sender_->sequencer.media_sequence_number() != seq_num) {
+    rtp_sender_->sequencer.set_media_sequence_number(seq_num);
+    rtp_sender_->packet_history.Clear();
+  }
+}
+
+void ModuleRtpRtcpImpl2::SetRtpState(const RtpState& rtp_state) {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  rtp_sender_->packet_generator.SetRtpState(rtp_state);
+  rtp_sender_->sequencer.SetRtpState(rtp_state);
+  rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
+  rtp_sender_->packet_sender.SetTimestampOffset(rtp_state.start_timestamp);
+}
+
+void ModuleRtpRtcpImpl2::SetRtxState(const RtpState& rtp_state) {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  rtp_sender_->packet_generator.SetRtxRtpState(rtp_state);
+  rtp_sender_->sequencer.set_rtx_sequence_number(rtp_state.sequence_number);
+}
+
+RtpState ModuleRtpRtcpImpl2::GetRtpState() const {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  RtpState state = rtp_sender_->packet_generator.GetRtpState();
+  rtp_sender_->sequencer.PopulateRtpState(state);
+  return state;
+}
+
+RtpState ModuleRtpRtcpImpl2::GetRtxState() const {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  RtpState state = rtp_sender_->packet_generator.GetRtxRtpState();
+  state.sequence_number = rtp_sender_->sequencer.rtx_sequence_number();
+  return state;
+}
+
+void ModuleRtpRtcpImpl2::SetNonSenderRttMeasurement(bool enabled) {
+  rtcp_sender_.SetNonSenderRttMeasurement(enabled);
+  rtcp_receiver_.SetNonSenderRttMeasurement(enabled);
+}
+
+uint32_t ModuleRtpRtcpImpl2::local_media_ssrc() const {
+  RTC_DCHECK_RUN_ON(&rtcp_thread_checker_);
+  RTC_DCHECK_EQ(rtcp_receiver_.local_media_ssrc(), rtcp_sender_.SSRC());
+  return rtcp_receiver_.local_media_ssrc();
+}
+
+void ModuleRtpRtcpImpl2::SetMid(absl::string_view mid) {
+  if (rtp_sender_) {
+    rtp_sender_->packet_generator.SetMid(mid);
+  }
+  // TODO(bugs.webrtc.org/4050): If we end up supporting the MID SDES item for
+  // RTCP, this will need to be passed down to the RTCPSender also.
+}
+
+// TODO(pbos): Handle media and RTX streams separately (separate RTCP
+// feedbacks).
+RTCPSender::FeedbackState ModuleRtpRtcpImpl2::GetFeedbackState() {
+  // TODO(bugs.webrtc.org/11581): Called by potentially multiple threads.
+  // Mostly "Send*" methods. Make sure it's only called on the
+  // construction thread.
+
+  RTCPSender::FeedbackState state;
+  // This is called also when receiver_only is true. Hence below
+  // checks that rtp_sender_ exists.
+  if (rtp_sender_) {
+    StreamDataCounters rtp_stats;
+    StreamDataCounters rtx_stats;
+    rtp_sender_->packet_sender.GetDataCounters(&rtp_stats, &rtx_stats);
+    state.packets_sent =
+        rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
+    state.media_bytes_sent = rtp_stats.transmitted.payload_bytes +
+                             rtx_stats.transmitted.payload_bytes;
+    state.send_bitrate =
+        rtp_sender_->packet_sender.GetSendRates(clock_->CurrentTime()).Sum();
+  }
+  state.receiver = &rtcp_receiver_;
+
+  if (absl::optional<RtpRtcpInterface::SenderReportStats> last_sr =
+          rtcp_receiver_.GetSenderReportStats();
+      last_sr.has_value()) {
+    state.remote_sr = CompactNtp(last_sr->last_remote_timestamp);
+    state.last_rr = last_sr->last_arrival_timestamp;
+  }
+
+  state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
+
+  return state;
+}
+
+int32_t ModuleRtpRtcpImpl2::SetSendingStatus(const bool sending) {
+  if (rtcp_sender_.Sending() != sending) {
+    // Sends RTCP BYE when going from true to false
+    rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending);
+  }
+  return 0;
+}
+
+bool ModuleRtpRtcpImpl2::Sending() const {
+  return rtcp_sender_.Sending();
+}
+
+void ModuleRtpRtcpImpl2::SetSendingMediaStatus(const bool sending) {
+  rtp_sender_->packet_generator.SetSendingMediaStatus(sending);
+}
+
+bool ModuleRtpRtcpImpl2::SendingMedia() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.SendingMedia() : false;
+}
+
+bool ModuleRtpRtcpImpl2::IsAudioConfigured() const {
+  return rtp_sender_ ? rtp_sender_->packet_generator.IsAudioConfigured()
+                     : false;
+}
+
+void ModuleRtpRtcpImpl2::SetAsPartOfAllocation(bool part_of_allocation) {
+  RTC_CHECK(rtp_sender_);
+  rtp_sender_->packet_sender.ForceIncludeSendPacketsInAllocation(
+      part_of_allocation);
+}
+
+bool ModuleRtpRtcpImpl2::OnSendingRtpFrame(uint32_t timestamp,
+                                           int64_t capture_time_ms,
+                                           int payload_type,
+                                           bool force_sender_report) {
+  if (!Sending()) {
+    return false;
+  }
+  // TODO(bugs.webrtc.org/12873): Migrate this method and it's users to use
+  // optional Timestamps.
+  absl::optional<Timestamp> capture_time;
+  if (capture_time_ms > 0) {
+    capture_time = Timestamp::Millis(capture_time_ms);
+  }
+  absl::optional<int> payload_type_optional;
+  if (payload_type >= 0)
+    payload_type_optional = payload_type;
+
+  auto closure = [this, timestamp, capture_time, payload_type_optional,
+                  force_sender_report] {
+    RTC_DCHECK_RUN_ON(worker_queue_);
+    rtcp_sender_.SetLastRtpTime(timestamp, capture_time, payload_type_optional);
+    // Make sure an RTCP report isn't queued behind a key frame.
+    if (rtcp_sender_.TimeToSendRTCPReport(force_sender_report))
+      rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+  };
+  if (worker_queue_->IsCurrent()) {
+    closure();
+  } else {
+    worker_queue_->PostTask(SafeTask(task_safety_.flag(), std::move(closure)));
+  }
+  return true;
+}
+
+bool ModuleRtpRtcpImpl2::TrySendPacket(std::unique_ptr<RtpPacketToSend> packet,
+                                       const PacedPacketInfo& pacing_info) {
+  RTC_DCHECK(rtp_sender_);
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  if (!rtp_sender_->packet_generator.SendingMedia()) {
+    return false;
+  }
+  if (packet->packet_type() == RtpPacketMediaType::kPadding &&
+      packet->Ssrc() == rtp_sender_->packet_generator.SSRC() &&
+      !rtp_sender_->sequencer.CanSendPaddingOnMediaSsrc()) {
+    // New media packet preempted this generated padding packet, discard it.
+    return false;
+  }
+  bool is_flexfec =
+      packet->packet_type() == RtpPacketMediaType::kForwardErrorCorrection &&
+      packet->Ssrc() == rtp_sender_->packet_generator.FlexfecSsrc();
+  if (!is_flexfec) {
+    rtp_sender_->sequencer.Sequence(*packet);
+  }
+
+  rtp_sender_->packet_sender.SendPacket(std::move(packet), pacing_info);
+  return true;
+}
+
+void ModuleRtpRtcpImpl2::OnBatchComplete() {
+  RTC_DCHECK(rtp_sender_);
+  rtp_sender_->packet_sender.OnBatchComplete();
+}
+
+void ModuleRtpRtcpImpl2::SetFecProtectionParams(
+    const FecProtectionParams& delta_params,
+    const FecProtectionParams& key_params) {
+  RTC_DCHECK(rtp_sender_);
+  rtp_sender_->packet_sender.SetFecProtectionParameters(delta_params,
+                                                        key_params);
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+ModuleRtpRtcpImpl2::FetchFecPackets() {
+  RTC_DCHECK(rtp_sender_);
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  return rtp_sender_->packet_sender.FetchFecPackets();
+}
+
+void ModuleRtpRtcpImpl2::OnAbortedRetransmissions(
+    rtc::ArrayView<const uint16_t> sequence_numbers) {
+  RTC_DCHECK(rtp_sender_);
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  rtp_sender_->packet_sender.OnAbortedRetransmissions(sequence_numbers);
+}
+
+void ModuleRtpRtcpImpl2::OnPacketsAcknowledged(
+    rtc::ArrayView<const uint16_t> sequence_numbers) {
+  RTC_DCHECK(rtp_sender_);
+  rtp_sender_->packet_history.CullAcknowledgedPackets(sequence_numbers);
+}
+
+bool ModuleRtpRtcpImpl2::SupportsPadding() const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_generator.SupportsPadding();
+}
+
+bool ModuleRtpRtcpImpl2::SupportsRtxPayloadPadding() const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_generator.SupportsRtxPayloadPadding();
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>>
+ModuleRtpRtcpImpl2::GeneratePadding(size_t target_size_bytes) {
+  RTC_DCHECK(rtp_sender_);
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+
+  return rtp_sender_->packet_generator.GeneratePadding(
+      target_size_bytes, rtp_sender_->packet_sender.MediaHasBeenSent(),
+      rtp_sender_->sequencer.CanSendPaddingOnMediaSsrc());
+}
+
+std::vector<RtpSequenceNumberMap::Info>
+ModuleRtpRtcpImpl2::GetSentRtpPacketInfos(
+    rtc::ArrayView<const uint16_t> sequence_numbers) const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_sender.GetSentRtpPacketInfos(sequence_numbers);
+}
+
+size_t ModuleRtpRtcpImpl2::ExpectedPerPacketOverhead() const {
+  if (!rtp_sender_) {
+    return 0;
+  }
+  return rtp_sender_->packet_generator.ExpectedPerPacketOverhead();
+}
+
+void ModuleRtpRtcpImpl2::OnPacketSendingThreadSwitched() {
+  // Ownership of sequencing is being transferred to another thread.
+  rtp_sender_->sequencing_checker.Detach();
+}
+
+size_t ModuleRtpRtcpImpl2::MaxRtpPacketSize() const {
+  RTC_DCHECK(rtp_sender_);
+  return rtp_sender_->packet_generator.MaxRtpPacketSize();
+}
+
+void ModuleRtpRtcpImpl2::SetMaxRtpPacketSize(size_t rtp_packet_size) {
+  RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE)
+      << "rtp packet size too large: " << rtp_packet_size;
+  RTC_DCHECK_GT(rtp_packet_size, packet_overhead_)
+      << "rtp packet size too small: " << rtp_packet_size;
+
+  rtcp_sender_.SetMaxRtpPacketSize(rtp_packet_size);
+  if (rtp_sender_) {
+    rtp_sender_->packet_generator.SetMaxRtpPacketSize(rtp_packet_size);
+  }
+}
+
+RtcpMode ModuleRtpRtcpImpl2::RTCP() const {
+  return rtcp_sender_.Status();
+}
+
+// Configure RTCP status i.e on/off.
+void ModuleRtpRtcpImpl2::SetRTCPStatus(const RtcpMode method) {
+  rtcp_sender_.SetRTCPStatus(method);
+}
+
+int32_t ModuleRtpRtcpImpl2::SetCNAME(absl::string_view c_name) {
+  return rtcp_sender_.SetCNAME(c_name);
+}
+
+absl::optional<TimeDelta> ModuleRtpRtcpImpl2::LastRtt() const {
+  absl::optional<TimeDelta> rtt = rtcp_receiver_.LastRtt();
+  if (!rtt.has_value()) {
+    MutexLock lock(&mutex_rtt_);
+    if (rtt_ms_ > 0) {
+      rtt = TimeDelta::Millis(rtt_ms_);
+    }
+  }
+  return rtt;
+}
+
+TimeDelta ModuleRtpRtcpImpl2::ExpectedRetransmissionTime() const {
+  int64_t expected_retransmission_time_ms = rtt_ms();
+  if (expected_retransmission_time_ms > 0) {
+    return TimeDelta::Millis(expected_retransmission_time_ms);
+  }
+  // No rtt available (`kRttUpdateInterval` not yet passed?), so try to
+  // poll avg_rtt_ms directly from rtcp receiver.
+  if (absl::optional<TimeDelta> rtt = rtcp_receiver_.AverageRtt()) {
+    return *rtt;
+  }
+  return kDefaultExpectedRetransmissionTime;
+}
+
+// Force a send of an RTCP packet.
+// Normal SR and RR are triggered via the process function.
+int32_t ModuleRtpRtcpImpl2::SendRTCP(RTCPPacketType packet_type) {
+  return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type);
+}
+
+void ModuleRtpRtcpImpl2::GetSendStreamDataCounters(
+    StreamDataCounters* rtp_counters,
+    StreamDataCounters* rtx_counters) const {
+  rtp_sender_->packet_sender.GetDataCounters(rtp_counters, rtx_counters);
+}
+
+// Received RTCP report.
+void ModuleRtpRtcpImpl2::RemoteRTCPSenderInfo(
+    uint32_t* packet_count, uint32_t* octet_count, int64_t* ntp_timestamp_ms,
+    int64_t* remote_ntp_timestamp_ms) const {
+  return rtcp_receiver_.RemoteRTCPSenderInfo(
+      packet_count, octet_count, ntp_timestamp_ms, remote_ntp_timestamp_ms);
+}
+
+std::vector<ReportBlockData> ModuleRtpRtcpImpl2::GetLatestReportBlockData()
+    const {
+  return rtcp_receiver_.GetLatestReportBlockData();
+}
+
+absl::optional<RtpRtcpInterface::SenderReportStats>
+ModuleRtpRtcpImpl2::GetSenderReportStats() const {
+  return rtcp_receiver_.GetSenderReportStats();
+}
+
+absl::optional<RtpRtcpInterface::NonSenderRttStats>
+ModuleRtpRtcpImpl2::GetNonSenderRttStats() const {
+  RTCPReceiver::NonSenderRttStats non_sender_rtt_stats =
+      rtcp_receiver_.GetNonSenderRTT();
+  return {{
+      non_sender_rtt_stats.round_trip_time(),
+      non_sender_rtt_stats.total_round_trip_time(),
+      non_sender_rtt_stats.round_trip_time_measurements(),
+  }};
+}
+
+// (REMB) Receiver Estimated Max Bitrate.
+void ModuleRtpRtcpImpl2::SetRemb(int64_t bitrate_bps,
+                                 std::vector<uint32_t> ssrcs) {
+  rtcp_sender_.SetRemb(bitrate_bps, std::move(ssrcs));
+}
+
+void ModuleRtpRtcpImpl2::UnsetRemb() {
+  rtcp_sender_.UnsetRemb();
+}
+
+void ModuleRtpRtcpImpl2::SetExtmapAllowMixed(bool extmap_allow_mixed) {
+  rtp_sender_->packet_generator.SetExtmapAllowMixed(extmap_allow_mixed);
+}
+
+void ModuleRtpRtcpImpl2::RegisterRtpHeaderExtension(absl::string_view uri,
+                                                    int id) {
+  bool registered =
+      rtp_sender_->packet_generator.RegisterRtpHeaderExtension(uri, id);
+  RTC_CHECK(registered);
+}
+
+void ModuleRtpRtcpImpl2::DeregisterSendRtpHeaderExtension(
+    absl::string_view uri) {
+  rtp_sender_->packet_generator.DeregisterRtpHeaderExtension(uri);
+}
+
+void ModuleRtpRtcpImpl2::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
+  rtcp_sender_.SetTmmbn(std::move(bounding_set));
+}
+
+// Send a Negative acknowledgment packet.
+int32_t ModuleRtpRtcpImpl2::SendNACK(const uint16_t* nack_list,
+                                     const uint16_t size) {
+  uint16_t nack_length = size;
+  uint16_t start_id = 0;
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (TimeToSendFullNackList(now_ms)) {
+    nack_last_time_sent_full_ms_ = now_ms;
+  } else {
+    // Only send extended list.
+    if (nack_last_seq_number_sent_ == nack_list[size - 1]) {
+      // Last sequence number is the same, do not send list.
+      return 0;
+    }
+    // Send new sequence numbers.
+    for (int i = 0; i < size; ++i) {
+      if (nack_last_seq_number_sent_ == nack_list[i]) {
+        start_id = i + 1;
+        break;
+      }
+    }
+    nack_length = size - start_id;
+  }
+
+  // Our RTCP NACK implementation is limited to kRtcpMaxNackFields sequence
+  // numbers per RTCP packet.
+  if (nack_length > kRtcpMaxNackFields) {
+    nack_length = kRtcpMaxNackFields;
+  }
+  nack_last_seq_number_sent_ = nack_list[start_id + nack_length - 1];
+
+  return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, nack_length,
+                               &nack_list[start_id]);
+}
+
+void ModuleRtpRtcpImpl2::SendNack(
+    const std::vector<uint16_t>& sequence_numbers) {
+  rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, sequence_numbers.size(),
+                        sequence_numbers.data());
+}
+
+bool ModuleRtpRtcpImpl2::TimeToSendFullNackList(int64_t now) const {
+  // Use RTT from RtcpRttStats class if provided.
+  int64_t rtt = rtt_ms();
+  if (rtt == 0) {
+    if (absl::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
+      rtt = average_rtt->ms();
+    }
+  }
+
+  const int64_t kStartUpRttMs = 100;
+  int64_t wait_time = 5 + ((rtt * 3) >> 1);  // 5 + RTT * 1.5.
+  if (rtt == 0) {
+    wait_time = kStartUpRttMs;
+  }
+
+  // Send a full NACK list once within every `wait_time`.
+  return now - nack_last_time_sent_full_ms_ > wait_time;
+}
+
+// Store the sent packets, needed to answer to Negative acknowledgment requests.
+void ModuleRtpRtcpImpl2::SetStorePacketsStatus(const bool enable,
+                                               const uint16_t number_to_store) {
+  rtp_sender_->packet_history.SetStorePacketsStatus(
+      enable ? RtpPacketHistory::StorageMode::kStoreAndCull
+             : RtpPacketHistory::StorageMode::kDisabled,
+      number_to_store);
+}
+
+bool ModuleRtpRtcpImpl2::StorePackets() const {
+  return rtp_sender_->packet_history.GetStorageMode() !=
+         RtpPacketHistory::StorageMode::kDisabled;
+}
+
+void ModuleRtpRtcpImpl2::SendCombinedRtcpPacket(
+    std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
+  rtcp_sender_.SendCombinedRtcpPacket(std::move(rtcp_packets));
+}
+
+int32_t ModuleRtpRtcpImpl2::SendLossNotification(uint16_t last_decoded_seq_num,
+                                                 uint16_t last_received_seq_num,
+                                                 bool decodability_flag,
+                                                 bool buffering_allowed) {
+  return rtcp_sender_.SendLossNotification(
+      GetFeedbackState(), last_decoded_seq_num, last_received_seq_num,
+      decodability_flag, buffering_allowed);
+}
+
+void ModuleRtpRtcpImpl2::SetRemoteSSRC(const uint32_t ssrc) {
+  // Inform about the incoming SSRC.
+  rtcp_sender_.SetRemoteSSRC(ssrc);
+  rtcp_receiver_.SetRemoteSSRC(ssrc);
+}
+
+void ModuleRtpRtcpImpl2::SetLocalSsrc(uint32_t local_ssrc) {
+  RTC_DCHECK_RUN_ON(&rtcp_thread_checker_);
+  rtcp_receiver_.set_local_media_ssrc(local_ssrc);
+  rtcp_sender_.SetSsrc(local_ssrc);
+}
+
+RtpSendRates ModuleRtpRtcpImpl2::GetSendRates() const {
+  RTC_DCHECK_RUN_ON(&rtp_sender_->sequencing_checker);
+  return rtp_sender_->packet_sender.GetSendRates(clock_->CurrentTime());
+}
+
+void ModuleRtpRtcpImpl2::OnRequestSendReport() {
+  SendRTCP(kRtcpSr);
+}
+
+void ModuleRtpRtcpImpl2::OnReceivedNack(
+    const std::vector<uint16_t>& nack_sequence_numbers) {
+  if (!rtp_sender_)
+    return;
+
+  if (!StorePackets() || nack_sequence_numbers.empty()) {
+    return;
+  }
+  // Use RTT from RtcpRttStats class if provided.
+  int64_t rtt = rtt_ms();
+  if (rtt == 0) {
+    if (absl::optional<TimeDelta> average_rtt = rtcp_receiver_.AverageRtt()) {
+      rtt = average_rtt->ms();
+    }
+  }
+  rtp_sender_->packet_generator.OnReceivedNack(nack_sequence_numbers, rtt);
+}
+
+void ModuleRtpRtcpImpl2::OnReceivedRtcpReportBlocks(
+    rtc::ArrayView<const ReportBlockData> report_blocks) {
+  if (rtp_sender_) {
+    uint32_t ssrc = SSRC();
+    absl::optional<uint32_t> rtx_ssrc;
+    if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) {
+      rtx_ssrc = rtp_sender_->packet_generator.RtxSsrc();
+    }
+
+    for (const ReportBlockData& report_block : report_blocks) {
+      if (ssrc == report_block.source_ssrc()) {
+        rtp_sender_->packet_generator.OnReceivedAckOnSsrc(
+            report_block.extended_highest_sequence_number());
+      } else if (rtx_ssrc == report_block.source_ssrc()) {
+        rtp_sender_->packet_generator.OnReceivedAckOnRtxSsrc(
+            report_block.extended_highest_sequence_number());
+      }
+    }
+  }
+}
+
+void ModuleRtpRtcpImpl2::set_rtt_ms(int64_t rtt_ms) {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  {
+    MutexLock lock(&mutex_rtt_);
+    rtt_ms_ = rtt_ms;
+  }
+  if (rtp_sender_) {
+    rtp_sender_->packet_history.SetRtt(TimeDelta::Millis(rtt_ms));
+  }
+}
+
+int64_t ModuleRtpRtcpImpl2::rtt_ms() const {
+  MutexLock lock(&mutex_rtt_);
+  return rtt_ms_;
+}
+
+void ModuleRtpRtcpImpl2::SetVideoBitrateAllocation(
+    const VideoBitrateAllocation& bitrate) {
+  rtcp_sender_.SetVideoBitrateAllocation(bitrate);
+}
+
+RTPSender* ModuleRtpRtcpImpl2::RtpSender() {
+  return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
+}
+
+const RTPSender* ModuleRtpRtcpImpl2::RtpSender() const {
+  return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr;
+}
+
+void ModuleRtpRtcpImpl2::PeriodicUpdate() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+
+  Timestamp check_since = clock_->CurrentTime() - kRttUpdateInterval;
+  absl::optional<TimeDelta> rtt =
+      rtcp_receiver_.OnPeriodicRttUpdate(check_since, rtcp_sender_.Sending());
+  if (rtt) {
+    if (rtt_stats_) {
+      rtt_stats_->OnRttUpdate(rtt->ms());
+    }
+    set_rtt_ms(rtt->ms());
+  }
+}
+
+void ModuleRtpRtcpImpl2::MaybeSendRtcp() {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  if (rtcp_sender_.TimeToSendRTCPReport())
+    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
+}
+
+// TODO(bugs.webrtc.org/12889): Consider removing this function when the issue
+// is resolved.
+void ModuleRtpRtcpImpl2::MaybeSendRtcpAtOrAfterTimestamp(
+    Timestamp execution_time) {
+  RTC_DCHECK_RUN_ON(worker_queue_);
+  Timestamp now = clock_->CurrentTime();
+  if (now >= execution_time) {
+    MaybeSendRtcp();
+    return;
+  }
+
+  TimeDelta delta = execution_time - now;
+  // TaskQueue may run task 1ms earlier, so don't print warning if in this case.
+  if (delta > TimeDelta::Millis(1)) {
+    RTC_DLOG(LS_WARNING) << "BUGBUG: Task queue scheduled delayed call "
+                         << delta << " too early.";
+  }
+
+  ScheduleMaybeSendRtcpAtOrAfterTimestamp(execution_time, delta);
+}
+
+void ModuleRtpRtcpImpl2::ScheduleRtcpSendEvaluation(TimeDelta duration) {
+  // We end up here under various sequences including the worker queue, and
+  // the RTCPSender lock is held.
+  // We're assuming that the fact that RTCPSender executes under other sequences
+  // than the worker queue on which it's created on implies that external
+  // synchronization is present and removes this activity before destruction.
+  if (duration.IsZero()) {
+    worker_queue_->PostTask(SafeTask(task_safety_.flag(), [this] {
+      RTC_DCHECK_RUN_ON(worker_queue_);
+      MaybeSendRtcp();
+    }));
+  } else {
+    Timestamp execution_time = clock_->CurrentTime() + duration;
+    ScheduleMaybeSendRtcpAtOrAfterTimestamp(execution_time, duration);
+  }
+}
+
+void ModuleRtpRtcpImpl2::ScheduleMaybeSendRtcpAtOrAfterTimestamp(
+    Timestamp execution_time,
+    TimeDelta duration) {
+  // We end up here under various sequences including the worker queue, and
+  // the RTCPSender lock is held.
+  // See note in ScheduleRtcpSendEvaluation about why `worker_queue_` can be
+  // accessed.
+  worker_queue_->PostDelayedTask(
+      SafeTask(task_safety_.flag(),
+               [this, execution_time] {
+                 RTC_DCHECK_RUN_ON(worker_queue_);
+                 MaybeSendRtcpAtOrAfterTimestamp(execution_time);
+               }),
+      duration.RoundUpTo(TimeDelta::Millis(1)));
+}
+
+}  // namespace webrtc