diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 028b0d5..b084577 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -74,6 +74,8 @@
     "histogram.h",
     "nack_module.cc",
     "nack_module.h",
+    "nack_module2.cc",
+    "nack_module2.h",
   ]
 
   deps = [
@@ -876,6 +878,7 @@
       "jitter_buffer_unittest.cc",
       "jitter_estimator_tests.cc",
       "loss_notification_controller_unittest.cc",
+      "nack_module2_unittest.cc",
       "nack_module_unittest.cc",
       "packet_buffer_unittest.cc",
       "receiver_unittest.cc",
diff --git a/modules/video_coding/nack_module2.cc b/modules/video_coding/nack_module2.cc
new file mode 100644
index 0000000..267eaeb
--- /dev/null
+++ b/modules/video_coding/nack_module2.cc
@@ -0,0 +1,349 @@
+/*
+ *  Copyright (c) 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 "modules/video_coding/nack_module2.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "api/units/timestamp.h"
+#include "modules/utility/include/process_thread.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/field_trial.h"
+
+namespace webrtc {
+
+namespace {
+const int kMaxPacketAge = 10000;
+const int kMaxNackPackets = 1000;
+const int kDefaultRttMs = 100;
+const int kMaxNackRetries = 10;
+const int kProcessFrequency = 50;
+const int kProcessIntervalMs = 1000 / kProcessFrequency;
+const int kMaxReorderedPackets = 128;
+const int kNumReorderingBuckets = 10;
+const int kDefaultSendNackDelayMs = 0;
+
+int64_t GetSendNackDelay() {
+  int64_t delay_ms = strtol(
+      webrtc::field_trial::FindFullName("WebRTC-SendNackDelayMs").c_str(),
+      nullptr, 10);
+  if (delay_ms > 0 && delay_ms <= 20) {
+    RTC_LOG(LS_INFO) << "SendNackDelay is set to " << delay_ms;
+    return delay_ms;
+  }
+  return kDefaultSendNackDelayMs;
+}
+}  // namespace
+
+NackModule2::NackInfo::NackInfo()
+    : seq_num(0), send_at_seq_num(0), sent_at_time(-1), retries(0) {}
+
+NackModule2::NackInfo::NackInfo(uint16_t seq_num,
+                                uint16_t send_at_seq_num,
+                                int64_t created_at_time)
+    : seq_num(seq_num),
+      send_at_seq_num(send_at_seq_num),
+      created_at_time(created_at_time),
+      sent_at_time(-1),
+      retries(0) {}
+
+NackModule2::BackoffSettings::BackoffSettings(TimeDelta min_retry,
+                                              TimeDelta max_rtt,
+                                              double base)
+    : min_retry_interval(min_retry), max_rtt(max_rtt), base(base) {}
+
+absl::optional<NackModule2::BackoffSettings>
+NackModule2::BackoffSettings::ParseFromFieldTrials() {
+  // Matches magic number in RTPSender::OnReceivedNack().
+  const TimeDelta kDefaultMinRetryInterval = TimeDelta::Millis(5);
+  // Upper bound on link-delay considered for exponential backoff.
+  // Selected so that cumulative delay with 1.25 base and 10 retries ends up
+  // below 3s, since above that there will be a FIR generated instead.
+  const TimeDelta kDefaultMaxRtt = TimeDelta::Millis(160);
+  // Default base for exponential backoff, adds 25% RTT delay for each retry.
+  const double kDefaultBase = 1.25;
+
+  FieldTrialParameter<bool> enabled("enabled", false);
+  FieldTrialParameter<TimeDelta> min_retry("min_retry",
+                                           kDefaultMinRetryInterval);
+  FieldTrialParameter<TimeDelta> max_rtt("max_rtt", kDefaultMaxRtt);
+  FieldTrialParameter<double> base("base", kDefaultBase);
+  ParseFieldTrial({&enabled, &min_retry, &max_rtt, &base},
+                  field_trial::FindFullName("WebRTC-ExponentialNackBackoff"));
+
+  if (enabled) {
+    return NackModule2::BackoffSettings(min_retry.Get(), max_rtt.Get(),
+                                        base.Get());
+  }
+  return absl::nullopt;
+}
+
+NackModule2::NackModule2(Clock* clock,
+                         NackSender* nack_sender,
+                         KeyFrameRequestSender* keyframe_request_sender)
+    : clock_(clock),
+      nack_sender_(nack_sender),
+      keyframe_request_sender_(keyframe_request_sender),
+      reordering_histogram_(kNumReorderingBuckets, kMaxReorderedPackets),
+      initialized_(false),
+      rtt_ms_(kDefaultRttMs),
+      newest_seq_num_(0),
+      next_process_time_ms_(-1),
+      send_nack_delay_ms_(GetSendNackDelay()),
+      backoff_settings_(BackoffSettings::ParseFromFieldTrials()) {
+  RTC_DCHECK(clock_);
+  RTC_DCHECK(nack_sender_);
+  RTC_DCHECK(keyframe_request_sender_);
+}
+
+int NackModule2::OnReceivedPacket(uint16_t seq_num, bool is_keyframe) {
+  return OnReceivedPacket(seq_num, is_keyframe, false);
+}
+
+int NackModule2::OnReceivedPacket(uint16_t seq_num,
+                                  bool is_keyframe,
+                                  bool is_recovered) {
+  rtc::CritScope lock(&crit_);
+  // TODO(philipel): When the packet includes information whether it is
+  //                 retransmitted or not, use that value instead. For
+  //                 now set it to true, which will cause the reordering
+  //                 statistics to never be updated.
+  bool is_retransmitted = true;
+
+  if (!initialized_) {
+    newest_seq_num_ = seq_num;
+    if (is_keyframe)
+      keyframe_list_.insert(seq_num);
+    initialized_ = true;
+    return 0;
+  }
+
+  // Since the |newest_seq_num_| is a packet we have actually received we know
+  // that packet has never been Nacked.
+  if (seq_num == newest_seq_num_)
+    return 0;
+
+  if (AheadOf(newest_seq_num_, seq_num)) {
+    // An out of order packet has been received.
+    auto nack_list_it = nack_list_.find(seq_num);
+    int nacks_sent_for_packet = 0;
+    if (nack_list_it != nack_list_.end()) {
+      nacks_sent_for_packet = nack_list_it->second.retries;
+      nack_list_.erase(nack_list_it);
+    }
+    if (!is_retransmitted)
+      UpdateReorderingStatistics(seq_num);
+    return nacks_sent_for_packet;
+  }
+
+  // Keep track of new keyframes.
+  if (is_keyframe)
+    keyframe_list_.insert(seq_num);
+
+  // And remove old ones so we don't accumulate keyframes.
+  auto it = keyframe_list_.lower_bound(seq_num - kMaxPacketAge);
+  if (it != keyframe_list_.begin())
+    keyframe_list_.erase(keyframe_list_.begin(), it);
+
+  if (is_recovered) {
+    recovered_list_.insert(seq_num);
+
+    // Remove old ones so we don't accumulate recovered packets.
+    auto it = recovered_list_.lower_bound(seq_num - kMaxPacketAge);
+    if (it != recovered_list_.begin())
+      recovered_list_.erase(recovered_list_.begin(), it);
+
+    // Do not send nack for packets recovered by FEC or RTX.
+    return 0;
+  }
+
+  AddPacketsToNack(newest_seq_num_ + 1, seq_num);
+  newest_seq_num_ = seq_num;
+
+  // Are there any nacks that are waiting for this seq_num.
+  std::vector<uint16_t> nack_batch = GetNackBatch(kSeqNumOnly);
+  if (!nack_batch.empty()) {
+    // This batch of NACKs is triggered externally; the initiator can
+    // batch them with other feedback messages.
+    nack_sender_->SendNack(nack_batch, /*buffering_allowed=*/true);
+  }
+
+  return 0;
+}
+
+void NackModule2::ClearUpTo(uint16_t seq_num) {
+  rtc::CritScope lock(&crit_);
+  nack_list_.erase(nack_list_.begin(), nack_list_.lower_bound(seq_num));
+  keyframe_list_.erase(keyframe_list_.begin(),
+                       keyframe_list_.lower_bound(seq_num));
+  recovered_list_.erase(recovered_list_.begin(),
+                        recovered_list_.lower_bound(seq_num));
+}
+
+void NackModule2::UpdateRtt(int64_t rtt_ms) {
+  rtc::CritScope lock(&crit_);
+  rtt_ms_ = rtt_ms;
+}
+
+void NackModule2::Clear() {
+  rtc::CritScope lock(&crit_);
+  nack_list_.clear();
+  keyframe_list_.clear();
+  recovered_list_.clear();
+}
+
+int64_t NackModule2::TimeUntilNextProcess() {
+  return std::max<int64_t>(next_process_time_ms_ - clock_->TimeInMilliseconds(),
+                           0);
+}
+
+void NackModule2::Process() {
+  if (nack_sender_) {
+    std::vector<uint16_t> nack_batch;
+    {
+      rtc::CritScope lock(&crit_);
+      nack_batch = GetNackBatch(kTimeOnly);
+    }
+
+    if (!nack_batch.empty()) {
+      // This batch of NACKs is triggered externally; there is no external
+      // initiator who can batch them with other feedback messages.
+      nack_sender_->SendNack(nack_batch, /*buffering_allowed=*/false);
+    }
+  }
+
+  // Update the next_process_time_ms_ in intervals to achieve
+  // the targeted frequency over time. Also add multiple intervals
+  // in case of a skip in time as to not make uneccessary
+  // calls to Process in order to catch up.
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (next_process_time_ms_ == -1) {
+    next_process_time_ms_ = now_ms + kProcessIntervalMs;
+  } else {
+    next_process_time_ms_ = next_process_time_ms_ + kProcessIntervalMs +
+                            (now_ms - next_process_time_ms_) /
+                                kProcessIntervalMs * kProcessIntervalMs;
+  }
+}
+
+bool NackModule2::RemovePacketsUntilKeyFrame() {
+  while (!keyframe_list_.empty()) {
+    auto it = nack_list_.lower_bound(*keyframe_list_.begin());
+
+    if (it != nack_list_.begin()) {
+      // We have found a keyframe that actually is newer than at least one
+      // packet in the nack list.
+      nack_list_.erase(nack_list_.begin(), it);
+      return true;
+    }
+
+    // If this keyframe is so old it does not remove any packets from the list,
+    // remove it from the list of keyframes and try the next keyframe.
+    keyframe_list_.erase(keyframe_list_.begin());
+  }
+  return false;
+}
+
+void NackModule2::AddPacketsToNack(uint16_t seq_num_start,
+                                   uint16_t seq_num_end) {
+  // Remove old packets.
+  auto it = nack_list_.lower_bound(seq_num_end - kMaxPacketAge);
+  nack_list_.erase(nack_list_.begin(), it);
+
+  // If the nack list is too large, remove packets from the nack list until
+  // the latest first packet of a keyframe. If the list is still too large,
+  // clear it and request a keyframe.
+  uint16_t num_new_nacks = ForwardDiff(seq_num_start, seq_num_end);
+  if (nack_list_.size() + num_new_nacks > kMaxNackPackets) {
+    while (RemovePacketsUntilKeyFrame() &&
+           nack_list_.size() + num_new_nacks > kMaxNackPackets) {
+    }
+
+    if (nack_list_.size() + num_new_nacks > kMaxNackPackets) {
+      nack_list_.clear();
+      RTC_LOG(LS_WARNING) << "NACK list full, clearing NACK"
+                             " list and requesting keyframe.";
+      keyframe_request_sender_->RequestKeyFrame();
+      return;
+    }
+  }
+
+  for (uint16_t seq_num = seq_num_start; seq_num != seq_num_end; ++seq_num) {
+    // Do not send nack for packets that are already recovered by FEC or RTX
+    if (recovered_list_.find(seq_num) != recovered_list_.end())
+      continue;
+    NackInfo nack_info(seq_num, seq_num + WaitNumberOfPackets(0.5),
+                       clock_->TimeInMilliseconds());
+    RTC_DCHECK(nack_list_.find(seq_num) == nack_list_.end());
+    nack_list_[seq_num] = nack_info;
+  }
+}
+
+std::vector<uint16_t> NackModule2::GetNackBatch(NackFilterOptions options) {
+  bool consider_seq_num = options != kTimeOnly;
+  bool consider_timestamp = options != kSeqNumOnly;
+  Timestamp now = clock_->CurrentTime();
+  std::vector<uint16_t> nack_batch;
+  auto it = nack_list_.begin();
+  while (it != nack_list_.end()) {
+    TimeDelta resend_delay = TimeDelta::Millis(rtt_ms_);
+    if (backoff_settings_) {
+      resend_delay =
+          std::max(resend_delay, backoff_settings_->min_retry_interval);
+      if (it->second.retries > 1) {
+        TimeDelta exponential_backoff =
+            std::min(TimeDelta::Millis(rtt_ms_), backoff_settings_->max_rtt) *
+            std::pow(backoff_settings_->base, it->second.retries - 1);
+        resend_delay = std::max(resend_delay, exponential_backoff);
+      }
+    }
+
+    bool delay_timed_out =
+        now.ms() - it->second.created_at_time >= send_nack_delay_ms_;
+    bool nack_on_rtt_passed =
+        now.ms() - it->second.sent_at_time >= resend_delay.ms();
+    bool nack_on_seq_num_passed =
+        it->second.sent_at_time == -1 &&
+        AheadOrAt(newest_seq_num_, it->second.send_at_seq_num);
+    if (delay_timed_out && ((consider_seq_num && nack_on_seq_num_passed) ||
+                            (consider_timestamp && nack_on_rtt_passed))) {
+      nack_batch.emplace_back(it->second.seq_num);
+      ++it->second.retries;
+      it->second.sent_at_time = now.ms();
+      if (it->second.retries >= kMaxNackRetries) {
+        RTC_LOG(LS_WARNING) << "Sequence number " << it->second.seq_num
+                            << " removed from NACK list due to max retries.";
+        it = nack_list_.erase(it);
+      } else {
+        ++it;
+      }
+      continue;
+    }
+    ++it;
+  }
+  return nack_batch;
+}
+
+void NackModule2::UpdateReorderingStatistics(uint16_t seq_num) {
+  RTC_DCHECK(AheadOf(newest_seq_num_, seq_num));
+  uint16_t diff = ReverseDiff(newest_seq_num_, seq_num);
+  reordering_histogram_.Add(diff);
+}
+
+int NackModule2::WaitNumberOfPackets(float probability) const {
+  if (reordering_histogram_.NumValues() == 0)
+    return 0;
+  return reordering_histogram_.InverseCdf(probability);
+}
+
+}  // namespace webrtc
diff --git a/modules/video_coding/nack_module2.h b/modules/video_coding/nack_module2.h
new file mode 100644
index 0000000..6518f32
--- /dev/null
+++ b/modules/video_coding/nack_module2.h
@@ -0,0 +1,129 @@
+/*
+ *  Copyright (c) 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.
+ */
+
+#ifndef MODULES_VIDEO_CODING_NACK_MODULE2_H_
+#define MODULES_VIDEO_CODING_NACK_MODULE2_H_
+
+#include <stdint.h>
+
+#include <map>
+#include <set>
+#include <vector>
+
+#include "api/units/time_delta.h"
+#include "modules/include/module.h"
+#include "modules/include/module_common_types.h"
+#include "modules/video_coding/histogram.h"
+#include "rtc_base/critical_section.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/thread_annotations.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+class NackModule2 final : public Module {
+ public:
+  NackModule2(Clock* clock,
+              NackSender* nack_sender,
+              KeyFrameRequestSender* keyframe_request_sender);
+
+  int OnReceivedPacket(uint16_t seq_num, bool is_keyframe);
+  int OnReceivedPacket(uint16_t seq_num, bool is_keyframe, bool is_recovered);
+
+  void ClearUpTo(uint16_t seq_num);
+  void UpdateRtt(int64_t rtt_ms);
+  void Clear();
+
+  // Module implementation
+  int64_t TimeUntilNextProcess() override;
+  void Process() override;
+
+ private:
+  // Which fields to consider when deciding which packet to nack in
+  // GetNackBatch.
+  enum NackFilterOptions { kSeqNumOnly, kTimeOnly, kSeqNumAndTime };
+
+  // This class holds the sequence number of the packet that is in the nack list
+  // as well as the meta data about when it should be nacked and how many times
+  // we have tried to nack this packet.
+  struct NackInfo {
+    NackInfo();
+    NackInfo(uint16_t seq_num,
+             uint16_t send_at_seq_num,
+             int64_t created_at_time);
+
+    uint16_t seq_num;
+    uint16_t send_at_seq_num;
+    int64_t created_at_time;
+    int64_t sent_at_time;
+    int retries;
+  };
+
+  struct BackoffSettings {
+    BackoffSettings(TimeDelta min_retry, TimeDelta max_rtt, double base);
+    static absl::optional<BackoffSettings> ParseFromFieldTrials();
+
+    // Min time between nacks.
+    const TimeDelta min_retry_interval;
+    // Upper bound on link-delay considered for exponential backoff.
+    const TimeDelta max_rtt;
+    // Base for the exponential backoff.
+    const double base;
+  };
+
+  void AddPacketsToNack(uint16_t seq_num_start, uint16_t seq_num_end)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Removes packets from the nack list until the next keyframe. Returns true
+  // if packets were removed.
+  bool RemovePacketsUntilKeyFrame() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
+  std::vector<uint16_t> GetNackBatch(NackFilterOptions options)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Update the reordering distribution.
+  void UpdateReorderingStatistics(uint16_t seq_num)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  // Returns how many packets we have to wait in order to receive the packet
+  // with probability |probabilty| or higher.
+  int WaitNumberOfPackets(float probability) const
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
+  rtc::CriticalSection crit_;
+  Clock* const clock_;
+  NackSender* const nack_sender_;
+  KeyFrameRequestSender* const keyframe_request_sender_;
+
+  // TODO(philipel): Some of the variables below are consistently used on a
+  // known thread (e.g. see |initialized_|). Those probably do not need
+  // synchronized access.
+  std::map<uint16_t, NackInfo, DescendingSeqNumComp<uint16_t>> nack_list_
+      RTC_GUARDED_BY(crit_);
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> keyframe_list_
+      RTC_GUARDED_BY(crit_);
+  std::set<uint16_t, DescendingSeqNumComp<uint16_t>> recovered_list_
+      RTC_GUARDED_BY(crit_);
+  video_coding::Histogram reordering_histogram_ RTC_GUARDED_BY(crit_);
+  bool initialized_ RTC_GUARDED_BY(crit_);
+  int64_t rtt_ms_ RTC_GUARDED_BY(crit_);
+  uint16_t newest_seq_num_ RTC_GUARDED_BY(crit_);
+
+  // Only touched on the process thread.
+  int64_t next_process_time_ms_;
+
+  // Adds a delay before send nack on packet received.
+  const int64_t send_nack_delay_ms_;
+
+  const absl::optional<BackoffSettings> backoff_settings_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_VIDEO_CODING_NACK_MODULE2_H_
diff --git a/modules/video_coding/nack_module2_unittest.cc b/modules/video_coding/nack_module2_unittest.cc
new file mode 100644
index 0000000..ebc28ec
--- /dev/null
+++ b/modules/video_coding/nack_module2_unittest.cc
@@ -0,0 +1,371 @@
+/*
+ *  Copyright (c) 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 "modules/video_coding/nack_module2.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <cstring>
+#include <memory>
+
+#include "system_wrappers/include/clock.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+class TestNackModule2 : public ::testing::TestWithParam<bool>,
+                        public NackSender,
+                        public KeyFrameRequestSender {
+ protected:
+  TestNackModule2()
+      : clock_(new SimulatedClock(0)),
+        field_trial_(GetParam()
+                         ? "WebRTC-ExponentialNackBackoff/enabled:true/"
+                         : "WebRTC-ExponentialNackBackoff/enabled:false/"),
+        nack_module_(clock_.get(), this, this),
+        keyframes_requested_(0) {}
+
+  void SetUp() override { nack_module_.UpdateRtt(kDefaultRttMs); }
+
+  void SendNack(const std::vector<uint16_t>& sequence_numbers,
+                bool buffering_allowed) override {
+    sent_nacks_.insert(sent_nacks_.end(), sequence_numbers.begin(),
+                       sequence_numbers.end());
+  }
+
+  void RequestKeyFrame() override { ++keyframes_requested_; }
+
+  static constexpr int64_t kDefaultRttMs = 20;
+  std::unique_ptr<SimulatedClock> clock_;
+  test::ScopedFieldTrials field_trial_;
+  NackModule2 nack_module_;
+  std::vector<uint16_t> sent_nacks_;
+  int keyframes_requested_;
+};
+
+TEST_P(TestNackModule2, NackOnePacket) {
+  nack_module_.OnReceivedPacket(1, false, false);
+  nack_module_.OnReceivedPacket(3, false, false);
+  EXPECT_EQ(1u, sent_nacks_.size());
+  EXPECT_EQ(2, sent_nacks_[0]);
+}
+
+TEST_P(TestNackModule2, WrappingSeqNum) {
+  nack_module_.OnReceivedPacket(0xfffe, false, false);
+  nack_module_.OnReceivedPacket(1, false, false);
+  EXPECT_EQ(2u, sent_nacks_.size());
+  EXPECT_EQ(0xffff, sent_nacks_[0]);
+  EXPECT_EQ(0, sent_nacks_[1]);
+}
+
+TEST_P(TestNackModule2, WrappingSeqNumClearToKeyframe) {
+  nack_module_.OnReceivedPacket(0xfffe, false, false);
+  nack_module_.OnReceivedPacket(1, false, false);
+  EXPECT_EQ(2u, sent_nacks_.size());
+  EXPECT_EQ(0xffff, sent_nacks_[0]);
+  EXPECT_EQ(0, sent_nacks_[1]);
+
+  sent_nacks_.clear();
+  nack_module_.OnReceivedPacket(2, true, false);
+  EXPECT_EQ(0u, sent_nacks_.size());
+
+  nack_module_.OnReceivedPacket(501, true, false);
+  EXPECT_EQ(498u, sent_nacks_.size());
+  for (int seq_num = 3; seq_num < 501; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 3]);
+
+  sent_nacks_.clear();
+  nack_module_.OnReceivedPacket(1001, false, false);
+  EXPECT_EQ(499u, sent_nacks_.size());
+  for (int seq_num = 502; seq_num < 1001; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 502]);
+
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module_.Process();
+  EXPECT_EQ(999u, sent_nacks_.size());
+  EXPECT_EQ(0xffff, sent_nacks_[0]);
+  EXPECT_EQ(0, sent_nacks_[1]);
+  for (int seq_num = 3; seq_num < 501; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 1]);
+  for (int seq_num = 502; seq_num < 1001; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 2]);
+
+  // Adding packet 1004 will cause the nack list to reach it's max limit.
+  // It will then clear all nacks up to the next keyframe (seq num 2),
+  // thus removing 0xffff and 0 from the nack list.
+  sent_nacks_.clear();
+  nack_module_.OnReceivedPacket(1004, false, false);
+  EXPECT_EQ(2u, sent_nacks_.size());
+  EXPECT_EQ(1002, sent_nacks_[0]);
+  EXPECT_EQ(1003, sent_nacks_[1]);
+
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module_.Process();
+  EXPECT_EQ(999u, sent_nacks_.size());
+  for (int seq_num = 3; seq_num < 501; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 3]);
+  for (int seq_num = 502; seq_num < 1001; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 4]);
+
+  // Adding packet 1007 will cause the nack module to overflow again, thus
+  // clearing everything up to 501 which is the next keyframe.
+  nack_module_.OnReceivedPacket(1007, false, false);
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module_.Process();
+  EXPECT_EQ(503u, sent_nacks_.size());
+  for (int seq_num = 502; seq_num < 1001; ++seq_num)
+    EXPECT_EQ(seq_num, sent_nacks_[seq_num - 502]);
+  EXPECT_EQ(1005, sent_nacks_[501]);
+  EXPECT_EQ(1006, sent_nacks_[502]);
+}
+
+TEST_P(TestNackModule2, DontBurstOnTimeSkip) {
+  nack_module_.Process();
+  clock_->AdvanceTimeMilliseconds(20);
+  EXPECT_EQ(0, nack_module_.TimeUntilNextProcess());
+  nack_module_.Process();
+
+  clock_->AdvanceTimeMilliseconds(100);
+  EXPECT_EQ(0, nack_module_.TimeUntilNextProcess());
+  nack_module_.Process();
+  EXPECT_EQ(20, nack_module_.TimeUntilNextProcess());
+
+  clock_->AdvanceTimeMilliseconds(19);
+  EXPECT_EQ(1, nack_module_.TimeUntilNextProcess());
+  clock_->AdvanceTimeMilliseconds(2);
+  nack_module_.Process();
+  EXPECT_EQ(19, nack_module_.TimeUntilNextProcess());
+
+  clock_->AdvanceTimeMilliseconds(19);
+  EXPECT_EQ(0, nack_module_.TimeUntilNextProcess());
+  nack_module_.Process();
+
+  clock_->AdvanceTimeMilliseconds(21);
+  EXPECT_EQ(0, nack_module_.TimeUntilNextProcess());
+  nack_module_.Process();
+  EXPECT_EQ(19, nack_module_.TimeUntilNextProcess());
+}
+
+TEST_P(TestNackModule2, ResendNack) {
+  nack_module_.OnReceivedPacket(1, false, false);
+  nack_module_.OnReceivedPacket(3, false, false);
+  size_t expected_nacks_sent = 1;
+  EXPECT_EQ(expected_nacks_sent, sent_nacks_.size());
+  EXPECT_EQ(2, sent_nacks_[0]);
+
+  if (GetParam()) {
+    // Retry has to wait at least 5ms by default.
+    nack_module_.UpdateRtt(1);
+    clock_->AdvanceTimeMilliseconds(4);
+    nack_module_.Process();  // Too early.
+    EXPECT_EQ(expected_nacks_sent, sent_nacks_.size());
+
+    clock_->AdvanceTimeMilliseconds(1);
+    nack_module_.Process();  // Now allowed.
+    EXPECT_EQ(++expected_nacks_sent, sent_nacks_.size());
+  } else {
+    nack_module_.UpdateRtt(1);
+    clock_->AdvanceTimeMilliseconds(1);
+    nack_module_.Process();  // Fast retransmit allowed.
+    EXPECT_EQ(++expected_nacks_sent, sent_nacks_.size());
+  }
+
+  // N:th try has to wait b^(N-1) * rtt by default.
+  const double b = GetParam() ? 1.25 : 1.0;
+  for (int i = 2; i < 10; ++i) {
+    // Change RTT, above the 40ms max for exponential backoff.
+    TimeDelta rtt = TimeDelta::Millis(160);  // + (i * 10 - 40)
+    nack_module_.UpdateRtt(rtt.ms());
+
+    // RTT gets capped at 160ms in backoff calculations.
+    TimeDelta expected_backoff_delay =
+        std::pow(b, i - 1) * std::min(rtt, TimeDelta::Millis(160));
+
+    // Move to one millisecond before next allowed NACK.
+    clock_->AdvanceTimeMilliseconds(expected_backoff_delay.ms() - 1);
+    nack_module_.Process();
+    EXPECT_EQ(expected_nacks_sent, sent_nacks_.size());
+
+    // Move to one millisecond after next allowed NACK.
+    // After rather than on to avoid rounding errors.
+    clock_->AdvanceTimeMilliseconds(2);
+    nack_module_.Process();  // Now allowed.
+    EXPECT_EQ(++expected_nacks_sent, sent_nacks_.size());
+  }
+
+  // Giving up after 10 tries.
+  clock_->AdvanceTimeMilliseconds(3000);
+  nack_module_.Process();
+  EXPECT_EQ(expected_nacks_sent, sent_nacks_.size());
+}
+
+TEST_P(TestNackModule2, ResendPacketMaxRetries) {
+  nack_module_.OnReceivedPacket(1, false, false);
+  nack_module_.OnReceivedPacket(3, false, false);
+  EXPECT_EQ(1u, sent_nacks_.size());
+  EXPECT_EQ(2, sent_nacks_[0]);
+
+  int backoff_factor = 1;
+  for (size_t retries = 1; retries < 10; ++retries) {
+    // Exponential backoff, so that we don't reject NACK because of time.
+    clock_->AdvanceTimeMilliseconds(backoff_factor * kDefaultRttMs);
+    backoff_factor *= 2;
+    nack_module_.Process();
+    EXPECT_EQ(retries + 1, sent_nacks_.size());
+  }
+
+  clock_->AdvanceTimeMilliseconds(backoff_factor * kDefaultRttMs);
+  nack_module_.Process();
+  EXPECT_EQ(10u, sent_nacks_.size());
+}
+
+TEST_P(TestNackModule2, TooLargeNackList) {
+  nack_module_.OnReceivedPacket(0, false, false);
+  nack_module_.OnReceivedPacket(1001, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(0, keyframes_requested_);
+  nack_module_.OnReceivedPacket(1003, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(1, keyframes_requested_);
+  nack_module_.OnReceivedPacket(1004, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(1, keyframes_requested_);
+}
+
+TEST_P(TestNackModule2, TooLargeNackListWithKeyFrame) {
+  nack_module_.OnReceivedPacket(0, false, false);
+  nack_module_.OnReceivedPacket(1, true, false);
+  nack_module_.OnReceivedPacket(1001, false, false);
+  EXPECT_EQ(999u, sent_nacks_.size());
+  EXPECT_EQ(0, keyframes_requested_);
+  nack_module_.OnReceivedPacket(1003, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(0, keyframes_requested_);
+  nack_module_.OnReceivedPacket(1005, false, false);
+  EXPECT_EQ(1000u, sent_nacks_.size());
+  EXPECT_EQ(1, keyframes_requested_);
+}
+
+TEST_P(TestNackModule2, ClearUpTo) {
+  nack_module_.OnReceivedPacket(0, false, false);
+  nack_module_.OnReceivedPacket(100, false, false);
+  EXPECT_EQ(99u, sent_nacks_.size());
+
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module_.ClearUpTo(50);
+  nack_module_.Process();
+  EXPECT_EQ(50u, sent_nacks_.size());
+  EXPECT_EQ(50, sent_nacks_[0]);
+}
+
+TEST_P(TestNackModule2, ClearUpToWrap) {
+  nack_module_.OnReceivedPacket(0xfff0, false, false);
+  nack_module_.OnReceivedPacket(0xf, false, false);
+  EXPECT_EQ(30u, sent_nacks_.size());
+
+  sent_nacks_.clear();
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module_.ClearUpTo(0);
+  nack_module_.Process();
+  EXPECT_EQ(15u, sent_nacks_.size());
+  EXPECT_EQ(0, sent_nacks_[0]);
+}
+
+TEST_P(TestNackModule2, PacketNackCount) {
+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(0, false, false));
+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(2, false, false));
+  EXPECT_EQ(1, nack_module_.OnReceivedPacket(1, false, false));
+
+  sent_nacks_.clear();
+  nack_module_.UpdateRtt(100);
+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(5, false, false));
+  clock_->AdvanceTimeMilliseconds(100);
+  nack_module_.Process();
+  clock_->AdvanceTimeMilliseconds(125);
+  nack_module_.Process();
+  EXPECT_EQ(3, nack_module_.OnReceivedPacket(3, false, false));
+  EXPECT_EQ(3, nack_module_.OnReceivedPacket(4, false, false));
+  EXPECT_EQ(0, nack_module_.OnReceivedPacket(4, false, false));
+}
+
+TEST_P(TestNackModule2, NackListFullAndNoOverlapWithKeyframes) {
+  const int kMaxNackPackets = 1000;
+  const unsigned int kFirstGap = kMaxNackPackets - 20;
+  const unsigned int kSecondGap = 200;
+  uint16_t seq_num = 0;
+  nack_module_.OnReceivedPacket(seq_num++, true, false);
+  seq_num += kFirstGap;
+  nack_module_.OnReceivedPacket(seq_num++, true, false);
+  EXPECT_EQ(kFirstGap, sent_nacks_.size());
+  sent_nacks_.clear();
+  seq_num += kSecondGap;
+  nack_module_.OnReceivedPacket(seq_num, true, false);
+  EXPECT_EQ(kSecondGap, sent_nacks_.size());
+}
+
+TEST_P(TestNackModule2, HandleFecRecoveredPacket) {
+  nack_module_.OnReceivedPacket(1, false, false);
+  nack_module_.OnReceivedPacket(4, false, true);
+  EXPECT_EQ(0u, sent_nacks_.size());
+  nack_module_.OnReceivedPacket(5, false, false);
+  EXPECT_EQ(2u, sent_nacks_.size());
+}
+
+TEST_P(TestNackModule2, SendNackWithoutDelay) {
+  nack_module_.OnReceivedPacket(0, false, false);
+  nack_module_.OnReceivedPacket(100, false, false);
+  EXPECT_EQ(99u, sent_nacks_.size());
+}
+
+INSTANTIATE_TEST_SUITE_P(WithAndWithoutBackoff,
+                         TestNackModule2,
+                         ::testing::Values(true, false));
+
+class TestNackModule2WithFieldTrial : public ::testing::Test,
+                                      public NackSender,
+                                      public KeyFrameRequestSender {
+ protected:
+  TestNackModule2WithFieldTrial()
+      : nack_delay_field_trial_("WebRTC-SendNackDelayMs/10/"),
+        clock_(new SimulatedClock(0)),
+        nack_module_(clock_.get(), this, this),
+        keyframes_requested_(0) {}
+
+  void SendNack(const std::vector<uint16_t>& sequence_numbers,
+                bool buffering_allowed) override {
+    sent_nacks_.insert(sent_nacks_.end(), sequence_numbers.begin(),
+                       sequence_numbers.end());
+  }
+
+  void RequestKeyFrame() override { ++keyframes_requested_; }
+
+  test::ScopedFieldTrials nack_delay_field_trial_;
+  std::unique_ptr<SimulatedClock> clock_;
+  NackModule2 nack_module_;
+  std::vector<uint16_t> sent_nacks_;
+  int keyframes_requested_;
+};
+
+TEST_F(TestNackModule2WithFieldTrial, SendNackWithDelay) {
+  nack_module_.OnReceivedPacket(0, false, false);
+  nack_module_.OnReceivedPacket(100, false, false);
+  EXPECT_EQ(0u, sent_nacks_.size());
+  clock_->AdvanceTimeMilliseconds(10);
+  nack_module_.OnReceivedPacket(106, false, false);
+  EXPECT_EQ(99u, sent_nacks_.size());
+  clock_->AdvanceTimeMilliseconds(10);
+  nack_module_.OnReceivedPacket(109, false, false);
+  EXPECT_EQ(104u, sent_nacks_.size());
+}
+}  // namespace webrtc
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 25df508..07b03ee 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -34,6 +34,8 @@
     "rtp_streams_synchronizer2.h",
     "rtp_video_stream_receiver.cc",
     "rtp_video_stream_receiver.h",
+    "rtp_video_stream_receiver2.cc",
+    "rtp_video_stream_receiver2.h",
     "rtp_video_stream_receiver_frame_transformer_delegate.cc",
     "rtp_video_stream_receiver_frame_transformer_delegate.h",
     "send_delay_stats.cc",
@@ -528,6 +530,7 @@
       "receive_statistics_proxy2_unittest.cc",
       "receive_statistics_proxy_unittest.cc",
       "report_block_stats_unittest.cc",
+      "rtp_video_stream_receiver2_unittest.cc",
       "rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc",
       "rtp_video_stream_receiver_unittest.cc",
       "send_delay_stats_unittest.cc",
diff --git a/video/rtp_video_stream_receiver.cc b/video/rtp_video_stream_receiver.cc
index 87691d4..53fd637 100644
--- a/video/rtp_video_stream_receiver.cc
+++ b/video/rtp_video_stream_receiver.cc
@@ -77,8 +77,6 @@
   return packet_buffer_max_size;
 }
 
-}  // namespace
-
 std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
     Clock* clock,
     ReceiveStatistics* receive_statistics,
@@ -107,6 +105,8 @@
 
 static const int kPacketLogIntervalMs = 10000;
 
+}  // namespace
+
 RtpVideoStreamReceiver::RtcpFeedbackBuffer::RtcpFeedbackBuffer(
     KeyFrameRequestSender* key_frame_request_sender,
     NackSender* nack_sender,
diff --git a/video/rtp_video_stream_receiver.h b/video/rtp_video_stream_receiver.h
index 32d675c..902c118 100644
--- a/video/rtp_video_stream_receiver.h
+++ b/video/rtp_video_stream_receiver.h
@@ -69,7 +69,8 @@
                                public KeyFrameRequestSender,
                                public video_coding::OnCompleteFrameCallback,
                                public OnDecryptedFrameCallback,
-                               public OnDecryptionStatusChangeCallback {
+                               public OnDecryptionStatusChangeCallback,
+                               public RtpVideoFrameReceiver {
  public:
   // DEPRECATED due to dependency on ReceiveStatisticsProxy.
   RtpVideoStreamReceiver(
@@ -205,9 +206,11 @@
   void AddSecondarySink(RtpPacketSinkInterface* sink);
   void RemoveSecondarySink(const RtpPacketSinkInterface* sink);
 
-  virtual void ManageFrame(std::unique_ptr<video_coding::RtpFrameObject> frame);
-
  private:
+  // Implements RtpVideoFrameReceiver.
+  void ManageFrame(
+      std::unique_ptr<video_coding::RtpFrameObject> frame) override;
+
   // Used for buffering RTCP feedback messages and sending them all together.
   // Note:
   // 1. Key frame requests and NACKs are mutually exclusive, with the
diff --git a/video/rtp_video_stream_receiver2.cc b/video/rtp_video_stream_receiver2.cc
new file mode 100644
index 0000000..98351c4
--- /dev/null
+++ b/video/rtp_video_stream_receiver2.cc
@@ -0,0 +1,1152 @@
+/*
+ *  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 "video/rtp_video_stream_receiver2.h"
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+#include <utility>
+#include <vector>
+
+#include "absl/algorithm/container.h"
+#include "absl/base/macros.h"
+#include "absl/memory/memory.h"
+#include "absl/types/optional.h"
+#include "media/base/media_constants.h"
+#include "modules/pacing/packet_router.h"
+#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/rtp_cvo.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
+#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
+#include "modules/utility/include/process_thread.h"
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/h264_sprop_parameter_sets.h"
+#include "modules/video_coding/h264_sps_pps_tracker.h"
+#include "modules/video_coding/nack_module2.h"
+#include "modules/video_coding/packet_buffer.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/location.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
+#include "system_wrappers/include/ntp_time.h"
+#include "video/receive_statistics_proxy2.h"
+
+namespace webrtc {
+
+namespace {
+// TODO(philipel): Change kPacketBufferStartSize back to 32 in M63 see:
+//                 crbug.com/752886
+constexpr int kPacketBufferStartSize = 512;
+constexpr int kPacketBufferMaxSize = 2048;
+
+int PacketBufferMaxSize() {
+  // The group here must be a positive power of 2, in which case that is used as
+  // size. All other values shall result in the default value being used.
+  const std::string group_name =
+      webrtc::field_trial::FindFullName("WebRTC-PacketBufferMaxSize");
+  int packet_buffer_max_size = kPacketBufferMaxSize;
+  if (!group_name.empty() &&
+      (sscanf(group_name.c_str(), "%d", &packet_buffer_max_size) != 1 ||
+       packet_buffer_max_size <= 0 ||
+       // Verify that the number is a positive power of 2.
+       (packet_buffer_max_size & (packet_buffer_max_size - 1)) != 0)) {
+    RTC_LOG(LS_WARNING) << "Invalid packet buffer max size: " << group_name;
+    packet_buffer_max_size = kPacketBufferMaxSize;
+  }
+  return packet_buffer_max_size;
+}
+
+std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
+    Clock* clock,
+    ReceiveStatistics* receive_statistics,
+    Transport* outgoing_transport,
+    RtcpRttStats* rtt_stats,
+    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
+    RtcpCnameCallback* rtcp_cname_callback,
+    uint32_t local_ssrc) {
+  RtpRtcp::Configuration configuration;
+  configuration.clock = clock;
+  configuration.audio = false;
+  configuration.receiver_only = true;
+  configuration.receive_statistics = receive_statistics;
+  configuration.outgoing_transport = outgoing_transport;
+  configuration.rtt_stats = rtt_stats;
+  configuration.rtcp_packet_type_counter_observer =
+      rtcp_packet_type_counter_observer;
+  configuration.rtcp_cname_callback = rtcp_cname_callback;
+  configuration.local_media_ssrc = local_ssrc;
+
+  std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
+  rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
+
+  return rtp_rtcp;
+}
+
+static const int kPacketLogIntervalMs = 10000;
+}  // namespace
+
+RtpVideoStreamReceiver2::RtcpFeedbackBuffer::RtcpFeedbackBuffer(
+    KeyFrameRequestSender* key_frame_request_sender,
+    NackSender* nack_sender,
+    LossNotificationSender* loss_notification_sender)
+    : key_frame_request_sender_(key_frame_request_sender),
+      nack_sender_(nack_sender),
+      loss_notification_sender_(loss_notification_sender),
+      request_key_frame_(false) {
+  RTC_DCHECK(key_frame_request_sender_);
+  RTC_DCHECK(nack_sender_);
+  RTC_DCHECK(loss_notification_sender_);
+}
+
+void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::RequestKeyFrame() {
+  rtc::CritScope lock(&cs_);
+  request_key_frame_ = true;
+}
+
+void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendNack(
+    const std::vector<uint16_t>& sequence_numbers,
+    bool buffering_allowed) {
+  RTC_DCHECK(!sequence_numbers.empty());
+  rtc::CritScope lock(&cs_);
+  nack_sequence_numbers_.insert(nack_sequence_numbers_.end(),
+                                sequence_numbers.cbegin(),
+                                sequence_numbers.cend());
+  if (!buffering_allowed) {
+    // Note that while *buffering* is not allowed, *batching* is, meaning that
+    // previously buffered messages may be sent along with the current message.
+    SendRtcpFeedback(ConsumeRtcpFeedbackLocked());
+  }
+}
+
+void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendLossNotification(
+    uint16_t last_decoded_seq_num,
+    uint16_t last_received_seq_num,
+    bool decodability_flag,
+    bool buffering_allowed) {
+  RTC_DCHECK(buffering_allowed);
+  rtc::CritScope lock(&cs_);
+  RTC_DCHECK(!lntf_state_)
+      << "SendLossNotification() called twice in a row with no call to "
+         "SendBufferedRtcpFeedback() in between.";
+  lntf_state_ = absl::make_optional<LossNotificationState>(
+      last_decoded_seq_num, last_received_seq_num, decodability_flag);
+}
+
+void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendBufferedRtcpFeedback() {
+  SendRtcpFeedback(ConsumeRtcpFeedback());
+}
+
+RtpVideoStreamReceiver2::RtcpFeedbackBuffer::ConsumedRtcpFeedback
+RtpVideoStreamReceiver2::RtcpFeedbackBuffer::ConsumeRtcpFeedback() {
+  rtc::CritScope lock(&cs_);
+  return ConsumeRtcpFeedbackLocked();
+}
+
+RtpVideoStreamReceiver2::RtcpFeedbackBuffer::ConsumedRtcpFeedback
+RtpVideoStreamReceiver2::RtcpFeedbackBuffer::ConsumeRtcpFeedbackLocked() {
+  ConsumedRtcpFeedback feedback;
+  std::swap(feedback.request_key_frame, request_key_frame_);
+  std::swap(feedback.nack_sequence_numbers, nack_sequence_numbers_);
+  std::swap(feedback.lntf_state, lntf_state_);
+  return feedback;
+}
+
+void RtpVideoStreamReceiver2::RtcpFeedbackBuffer::SendRtcpFeedback(
+    ConsumedRtcpFeedback feedback) {
+  if (feedback.lntf_state) {
+    // If either a NACK or a key frame request is sent, we should buffer
+    // the LNTF and wait for them (NACK or key frame request) to trigger
+    // the compound feedback message.
+    // Otherwise, the LNTF should be sent out immediately.
+    const bool buffering_allowed =
+        feedback.request_key_frame || !feedback.nack_sequence_numbers.empty();
+
+    loss_notification_sender_->SendLossNotification(
+        feedback.lntf_state->last_decoded_seq_num,
+        feedback.lntf_state->last_received_seq_num,
+        feedback.lntf_state->decodability_flag, buffering_allowed);
+  }
+
+  if (feedback.request_key_frame) {
+    key_frame_request_sender_->RequestKeyFrame();
+  } else if (!feedback.nack_sequence_numbers.empty()) {
+    nack_sender_->SendNack(feedback.nack_sequence_numbers, true);
+  }
+}
+
+RtpVideoStreamReceiver2::RtpVideoStreamReceiver2(
+    Clock* clock,
+    Transport* transport,
+    RtcpRttStats* rtt_stats,
+    PacketRouter* packet_router,
+    const VideoReceiveStream::Config* config,
+    ReceiveStatistics* rtp_receive_statistics,
+    RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
+    RtcpCnameCallback* rtcp_cname_callback,
+    ProcessThread* process_thread,
+    NackSender* nack_sender,
+    KeyFrameRequestSender* keyframe_request_sender,
+    video_coding::OnCompleteFrameCallback* complete_frame_callback,
+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer)
+    : clock_(clock),
+      config_(*config),
+      packet_router_(packet_router),
+      process_thread_(process_thread),
+      ntp_estimator_(clock),
+      rtp_header_extensions_(config_.rtp.extensions),
+      forced_playout_delay_max_ms_("max_ms", absl::nullopt),
+      forced_playout_delay_min_ms_("min_ms", absl::nullopt),
+      rtp_receive_statistics_(rtp_receive_statistics),
+      ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc,
+                                              this,
+                                              config->rtp.extensions)),
+      receiving_(false),
+      last_packet_log_ms_(-1),
+      rtp_rtcp_(CreateRtpRtcpModule(clock,
+                                    rtp_receive_statistics_,
+                                    transport,
+                                    rtt_stats,
+                                    rtcp_packet_type_counter_observer,
+                                    rtcp_cname_callback,
+                                    config_.rtp.local_ssrc)),
+      complete_frame_callback_(complete_frame_callback),
+      keyframe_request_sender_(keyframe_request_sender),
+      // TODO(bugs.webrtc.org/10336): Let |rtcp_feedback_buffer_| communicate
+      // directly with |rtp_rtcp_|.
+      rtcp_feedback_buffer_(this, nack_sender, this),
+      packet_buffer_(clock_, kPacketBufferStartSize, PacketBufferMaxSize()),
+      has_received_frame_(false),
+      frames_decryptable_(false),
+      absolute_capture_time_receiver_(clock) {
+  constexpr bool remb_candidate = true;
+  if (packet_router_)
+    packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
+
+  RTC_DCHECK(config_.rtp.rtcp_mode != RtcpMode::kOff)
+      << "A stream should not be configured with RTCP disabled. This value is "
+         "reserved for internal usage.";
+  // TODO(pbos): What's an appropriate local_ssrc for receive-only streams?
+  RTC_DCHECK(config_.rtp.local_ssrc != 0);
+  RTC_DCHECK(config_.rtp.remote_ssrc != config_.rtp.local_ssrc);
+
+  rtp_rtcp_->SetRTCPStatus(config_.rtp.rtcp_mode);
+  rtp_rtcp_->SetRemoteSSRC(config_.rtp.remote_ssrc);
+
+  static const int kMaxPacketAgeToNack = 450;
+  const int max_reordering_threshold = (config_.rtp.nack.rtp_history_ms > 0)
+                                           ? kMaxPacketAgeToNack
+                                           : kDefaultMaxReorderingThreshold;
+  rtp_receive_statistics_->SetMaxReorderingThreshold(config_.rtp.remote_ssrc,
+                                                     max_reordering_threshold);
+  // TODO(nisse): For historic reasons, we applied the above
+  // max_reordering_threshold also for RTX stats, which makes little sense since
+  // we don't NACK rtx packets. Consider deleting the below block, and rely on
+  // the default threshold.
+  if (config_.rtp.rtx_ssrc) {
+    rtp_receive_statistics_->SetMaxReorderingThreshold(
+        config_.rtp.rtx_ssrc, max_reordering_threshold);
+  }
+  if (config_.rtp.rtcp_xr.receiver_reference_time_report)
+    rtp_rtcp_->SetRtcpXrRrtrStatus(true);
+
+  ParseFieldTrial(
+      {&forced_playout_delay_max_ms_, &forced_playout_delay_min_ms_},
+      field_trial::FindFullName("WebRTC-ForcePlayoutDelay"));
+
+  process_thread_->RegisterModule(rtp_rtcp_.get(), RTC_FROM_HERE);
+
+  if (config_.rtp.lntf.enabled) {
+    loss_notification_controller_ =
+        std::make_unique<LossNotificationController>(&rtcp_feedback_buffer_,
+                                                     &rtcp_feedback_buffer_);
+  }
+
+  if (config_.rtp.nack.rtp_history_ms != 0) {
+    nack_module_ = std::make_unique<NackModule2>(clock_, &rtcp_feedback_buffer_,
+                                                 &rtcp_feedback_buffer_);
+    process_thread_->RegisterModule(nack_module_.get(), RTC_FROM_HERE);
+  }
+
+  reference_finder_ =
+      std::make_unique<video_coding::RtpFrameReferenceFinder>(this);
+
+  // Only construct the encrypted receiver if frame encryption is enabled.
+  if (config_.crypto_options.sframe.require_frame_encryption) {
+    buffered_frame_decryptor_ =
+        std::make_unique<BufferedFrameDecryptor>(this, this);
+    if (frame_decryptor != nullptr) {
+      buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor));
+    }
+  }
+
+  if (frame_transformer) {
+    frame_transformer_delegate_ = new rtc::RefCountedObject<
+        RtpVideoStreamReceiverFrameTransformerDelegate>(
+        this, std::move(frame_transformer), rtc::Thread::Current(),
+        config_.rtp.remote_ssrc);
+    frame_transformer_delegate_->Init();
+  }
+}
+
+RtpVideoStreamReceiver2::~RtpVideoStreamReceiver2() {
+  RTC_DCHECK(secondary_sinks_.empty());
+
+  if (nack_module_) {
+    process_thread_->DeRegisterModule(nack_module_.get());
+  }
+
+  process_thread_->DeRegisterModule(rtp_rtcp_.get());
+
+  if (packet_router_)
+    packet_router_->RemoveReceiveRtpModule(rtp_rtcp_.get());
+  UpdateHistograms();
+  if (frame_transformer_delegate_)
+    frame_transformer_delegate_->Reset();
+}
+
+void RtpVideoStreamReceiver2::AddReceiveCodec(
+    const VideoCodec& video_codec,
+    const std::map<std::string, std::string>& codec_params,
+    bool raw_payload) {
+  payload_type_map_.emplace(
+      video_codec.plType,
+      raw_payload ? std::make_unique<VideoRtpDepacketizerRaw>()
+                  : CreateVideoRtpDepacketizer(video_codec.codecType));
+  pt_codec_params_.emplace(video_codec.plType, codec_params);
+}
+
+absl::optional<Syncable::Info> RtpVideoStreamReceiver2::GetSyncInfo() const {
+  Syncable::Info info;
+  if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs,
+                           &info.capture_time_ntp_frac, nullptr, nullptr,
+                           &info.capture_time_source_clock) != 0) {
+    return absl::nullopt;
+  }
+  {
+    rtc::CritScope lock(&sync_info_lock_);
+    if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
+      return absl::nullopt;
+    }
+    info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
+    info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
+  }
+
+  // Leaves info.current_delay_ms uninitialized.
+  return info;
+}
+
+RtpVideoStreamReceiver2::ParseGenericDependenciesResult
+RtpVideoStreamReceiver2::ParseGenericDependenciesExtension(
+    const RtpPacketReceived& rtp_packet,
+    RTPVideoHeader* video_header) {
+  if (rtp_packet.HasExtension<RtpDependencyDescriptorExtension>()) {
+    webrtc::DependencyDescriptor dependency_descriptor;
+    if (!rtp_packet.GetExtension<RtpDependencyDescriptorExtension>(
+            video_structure_.get(), &dependency_descriptor)) {
+      // Descriptor is there, but failed to parse. Either it is invalid,
+      // or too old packet (after relevant video_structure_ changed),
+      // or too new packet (before relevant video_structure_ arrived).
+      // Drop such packet to be on the safe side.
+      // TODO(bugs.webrtc.org/10342): Stash too new packet.
+      RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc()
+                          << " Failed to parse dependency descriptor.";
+      return kDropPacket;
+    }
+    if (dependency_descriptor.attached_structure != nullptr &&
+        !dependency_descriptor.first_packet_in_frame) {
+      RTC_LOG(LS_WARNING) << "ssrc: " << rtp_packet.Ssrc()
+                          << "Invalid dependency descriptor: structure "
+                             "attached to non first packet of a frame.";
+      return kDropPacket;
+    }
+    video_header->is_first_packet_in_frame =
+        dependency_descriptor.first_packet_in_frame;
+    video_header->is_last_packet_in_frame =
+        dependency_descriptor.last_packet_in_frame;
+
+    int64_t frame_id =
+        frame_id_unwrapper_.Unwrap(dependency_descriptor.frame_number);
+    auto& generic_descriptor_info = video_header->generic.emplace();
+    generic_descriptor_info.frame_id = frame_id;
+    generic_descriptor_info.spatial_index =
+        dependency_descriptor.frame_dependencies.spatial_id;
+    generic_descriptor_info.temporal_index =
+        dependency_descriptor.frame_dependencies.temporal_id;
+    for (int fdiff : dependency_descriptor.frame_dependencies.frame_diffs) {
+      generic_descriptor_info.dependencies.push_back(frame_id - fdiff);
+    }
+    generic_descriptor_info.decode_target_indications =
+        dependency_descriptor.frame_dependencies.decode_target_indications;
+    if (dependency_descriptor.resolution) {
+      video_header->width = dependency_descriptor.resolution->Width();
+      video_header->height = dependency_descriptor.resolution->Height();
+    }
+
+    // FrameDependencyStructure is sent in dependency descriptor of the first
+    // packet of a key frame and required for parsed dependency descriptor in
+    // all the following packets until next key frame.
+    // Save it if there is a (potentially) new structure.
+    if (dependency_descriptor.attached_structure) {
+      RTC_DCHECK(dependency_descriptor.first_packet_in_frame);
+      if (video_structure_frame_id_ > frame_id) {
+        RTC_LOG(LS_WARNING)
+            << "Arrived key frame with id " << frame_id << " and structure id "
+            << dependency_descriptor.attached_structure->structure_id
+            << " is older than the latest received key frame with id "
+            << *video_structure_frame_id_ << " and structure id "
+            << video_structure_->structure_id;
+        return kDropPacket;
+      }
+      video_structure_ = std::move(dependency_descriptor.attached_structure);
+      video_structure_frame_id_ = frame_id;
+      video_header->frame_type = VideoFrameType::kVideoFrameKey;
+    } else {
+      video_header->frame_type = VideoFrameType::kVideoFrameDelta;
+    }
+    return kHasGenericDescriptor;
+  }
+
+  RtpGenericFrameDescriptor generic_frame_descriptor;
+  if (!rtp_packet.GetExtension<RtpGenericFrameDescriptorExtension00>(
+          &generic_frame_descriptor)) {
+    return kNoGenericDescriptor;
+  }
+
+  video_header->is_first_packet_in_frame =
+      generic_frame_descriptor.FirstPacketInSubFrame();
+  video_header->is_last_packet_in_frame =
+      generic_frame_descriptor.LastPacketInSubFrame();
+
+  if (generic_frame_descriptor.FirstPacketInSubFrame()) {
+    video_header->frame_type =
+        generic_frame_descriptor.FrameDependenciesDiffs().empty()
+            ? VideoFrameType::kVideoFrameKey
+            : VideoFrameType::kVideoFrameDelta;
+
+    auto& generic_descriptor_info = video_header->generic.emplace();
+    int64_t frame_id =
+        frame_id_unwrapper_.Unwrap(generic_frame_descriptor.FrameId());
+    generic_descriptor_info.frame_id = frame_id;
+    generic_descriptor_info.spatial_index =
+        generic_frame_descriptor.SpatialLayer();
+    generic_descriptor_info.temporal_index =
+        generic_frame_descriptor.TemporalLayer();
+    for (uint16_t fdiff : generic_frame_descriptor.FrameDependenciesDiffs()) {
+      generic_descriptor_info.dependencies.push_back(frame_id - fdiff);
+    }
+  }
+  video_header->width = generic_frame_descriptor.Width();
+  video_header->height = generic_frame_descriptor.Height();
+  return kHasGenericDescriptor;
+}
+
+void RtpVideoStreamReceiver2::OnReceivedPayloadData(
+    rtc::CopyOnWriteBuffer codec_payload,
+    const RtpPacketReceived& rtp_packet,
+    const RTPVideoHeader& video) {
+  RTC_DCHECK_RUN_ON(&worker_task_checker_);
+  auto packet = std::make_unique<video_coding::PacketBuffer::Packet>(
+      rtp_packet, video, ntp_estimator_.Estimate(rtp_packet.Timestamp()),
+      clock_->TimeInMilliseconds());
+
+  // Try to extrapolate absolute capture time if it is missing.
+  packet->packet_info.set_absolute_capture_time(
+      absolute_capture_time_receiver_.OnReceivePacket(
+          AbsoluteCaptureTimeReceiver::GetSource(packet->packet_info.ssrc(),
+                                                 packet->packet_info.csrcs()),
+          packet->packet_info.rtp_timestamp(),
+          // Assume frequency is the same one for all video frames.
+          kVideoPayloadTypeFrequency,
+          packet->packet_info.absolute_capture_time()));
+
+  RTPVideoHeader& video_header = packet->video_header;
+  video_header.rotation = kVideoRotation_0;
+  video_header.content_type = VideoContentType::UNSPECIFIED;
+  video_header.video_timing.flags = VideoSendTiming::kInvalid;
+  video_header.is_last_packet_in_frame |= rtp_packet.Marker();
+  video_header.frame_marking.temporal_id = kNoTemporalIdx;
+
+  if (const auto* vp9_header =
+          absl::get_if<RTPVideoHeaderVP9>(&video_header.video_type_header)) {
+    video_header.is_last_packet_in_frame |= vp9_header->end_of_frame;
+    video_header.is_first_packet_in_frame |= vp9_header->beginning_of_frame;
+  }
+
+  rtp_packet.GetExtension<VideoOrientation>(&video_header.rotation);
+  rtp_packet.GetExtension<VideoContentTypeExtension>(
+      &video_header.content_type);
+  rtp_packet.GetExtension<VideoTimingExtension>(&video_header.video_timing);
+  if (forced_playout_delay_max_ms_ && forced_playout_delay_min_ms_) {
+    video_header.playout_delay.max_ms = *forced_playout_delay_max_ms_;
+    video_header.playout_delay.min_ms = *forced_playout_delay_min_ms_;
+  } else {
+    rtp_packet.GetExtension<PlayoutDelayLimits>(&video_header.playout_delay);
+  }
+  rtp_packet.GetExtension<FrameMarkingExtension>(&video_header.frame_marking);
+
+  ParseGenericDependenciesResult generic_descriptor_state =
+      ParseGenericDependenciesExtension(rtp_packet, &video_header);
+  if (generic_descriptor_state == kDropPacket)
+    return;
+
+  // Color space should only be transmitted in the last packet of a frame,
+  // therefore, neglect it otherwise so that last_color_space_ is not reset by
+  // mistake.
+  if (video_header.is_last_packet_in_frame) {
+    video_header.color_space = rtp_packet.GetExtension<ColorSpaceExtension>();
+    if (video_header.color_space ||
+        video_header.frame_type == VideoFrameType::kVideoFrameKey) {
+      // Store color space since it's only transmitted when changed or for key
+      // frames. Color space will be cleared if a key frame is transmitted
+      // without color space information.
+      last_color_space_ = video_header.color_space;
+    } else if (last_color_space_) {
+      video_header.color_space = last_color_space_;
+    }
+  }
+
+  if (loss_notification_controller_) {
+    if (rtp_packet.recovered()) {
+      // TODO(bugs.webrtc.org/10336): Implement support for reordering.
+      RTC_LOG(LS_INFO)
+          << "LossNotificationController does not support reordering.";
+    } else if (generic_descriptor_state == kNoGenericDescriptor) {
+      RTC_LOG(LS_WARNING) << "LossNotificationController requires generic "
+                             "frame descriptor, but it is missing.";
+    } else {
+      if (video_header.is_first_packet_in_frame) {
+        RTC_DCHECK(video_header.generic);
+        LossNotificationController::FrameDetails frame;
+        frame.is_keyframe =
+            video_header.frame_type == VideoFrameType::kVideoFrameKey;
+        frame.frame_id = video_header.generic->frame_id;
+        frame.frame_dependencies = video_header.generic->dependencies;
+        loss_notification_controller_->OnReceivedPacket(
+            rtp_packet.SequenceNumber(), &frame);
+      } else {
+        loss_notification_controller_->OnReceivedPacket(
+            rtp_packet.SequenceNumber(), nullptr);
+      }
+    }
+  }
+
+  if (nack_module_) {
+    const bool is_keyframe =
+        video_header.is_first_packet_in_frame &&
+        video_header.frame_type == VideoFrameType::kVideoFrameKey;
+
+    packet->times_nacked = nack_module_->OnReceivedPacket(
+        rtp_packet.SequenceNumber(), is_keyframe, rtp_packet.recovered());
+  } else {
+    packet->times_nacked = -1;
+  }
+
+  if (codec_payload.size() == 0) {
+    NotifyReceiverOfEmptyPacket(packet->seq_num);
+    rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
+    return;
+  }
+
+  if (packet->codec() == kVideoCodecH264) {
+    // Only when we start to receive packets will we know what payload type
+    // that will be used. When we know the payload type insert the correct
+    // sps/pps into the tracker.
+    if (packet->payload_type != last_payload_type_) {
+      last_payload_type_ = packet->payload_type;
+      InsertSpsPpsIntoTracker(packet->payload_type);
+    }
+
+    video_coding::H264SpsPpsTracker::FixedBitstream fixed =
+        tracker_.CopyAndFixBitstream(
+            rtc::MakeArrayView(codec_payload.cdata(), codec_payload.size()),
+            &packet->video_header);
+
+    switch (fixed.action) {
+      case video_coding::H264SpsPpsTracker::kRequestKeyframe:
+        rtcp_feedback_buffer_.RequestKeyFrame();
+        rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
+        ABSL_FALLTHROUGH_INTENDED;
+      case video_coding::H264SpsPpsTracker::kDrop:
+        return;
+      case video_coding::H264SpsPpsTracker::kInsert:
+        packet->video_payload = std::move(fixed.bitstream);
+        break;
+    }
+
+  } else {
+    packet->video_payload = std::move(codec_payload);
+  }
+
+  rtcp_feedback_buffer_.SendBufferedRtcpFeedback();
+  frame_counter_.Add(packet->timestamp);
+  OnInsertedPacket(packet_buffer_.InsertPacket(std::move(packet)));
+}
+
+void RtpVideoStreamReceiver2::OnRecoveredPacket(const uint8_t* rtp_packet,
+                                                size_t rtp_packet_length) {
+  RtpPacketReceived packet;
+  if (!packet.Parse(rtp_packet, rtp_packet_length))
+    return;
+  if (packet.PayloadType() == config_.rtp.red_payload_type) {
+    RTC_LOG(LS_WARNING) << "Discarding recovered packet with RED encapsulation";
+    return;
+  }
+
+  packet.IdentifyExtensions(rtp_header_extensions_);
+  packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
+  // TODO(nisse): UlpfecReceiverImpl::ProcessReceivedFec passes both
+  // original (decapsulated) media packets and recovered packets to
+  // this callback. We need a way to distinguish, for setting
+  // packet.recovered() correctly. Ideally, move RED decapsulation out
+  // of the Ulpfec implementation.
+
+  ReceivePacket(packet);
+}
+
+// This method handles both regular RTP packets and packets recovered
+// via FlexFEC.
+void RtpVideoStreamReceiver2::OnRtpPacket(const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(&worker_task_checker_);
+
+  if (!receiving_) {
+    return;
+  }
+
+  if (!packet.recovered()) {
+    // TODO(nisse): Exclude out-of-order packets?
+    int64_t now_ms = clock_->TimeInMilliseconds();
+    {
+      rtc::CritScope cs(&sync_info_lock_);
+      last_received_rtp_timestamp_ = packet.Timestamp();
+      last_received_rtp_system_time_ms_ = now_ms;
+    }
+    // Periodically log the RTP header of incoming packets.
+    if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
+      rtc::StringBuilder ss;
+      ss << "Packet received on SSRC: " << packet.Ssrc()
+         << " with payload type: " << static_cast<int>(packet.PayloadType())
+         << ", timestamp: " << packet.Timestamp()
+         << ", sequence number: " << packet.SequenceNumber()
+         << ", arrival time: " << packet.arrival_time_ms();
+      int32_t time_offset;
+      if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
+        ss << ", toffset: " << time_offset;
+      }
+      uint32_t send_time;
+      if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
+        ss << ", abs send time: " << send_time;
+      }
+      RTC_LOG(LS_INFO) << ss.str();
+      last_packet_log_ms_ = now_ms;
+    }
+  }
+
+  ReceivePacket(packet);
+
+  // Update receive statistics after ReceivePacket.
+  // Receive statistics will be reset if the payload type changes (make sure
+  // that the first packet is included in the stats).
+  if (!packet.recovered()) {
+    rtp_receive_statistics_->OnRtpPacket(packet);
+  }
+
+  for (RtpPacketSinkInterface* secondary_sink : secondary_sinks_) {
+    secondary_sink->OnRtpPacket(packet);
+  }
+}
+
+void RtpVideoStreamReceiver2::RequestKeyFrame() {
+  // TODO(bugs.webrtc.org/10336): Allow the sender to ignore key frame requests
+  // issued by anything other than the LossNotificationController if it (the
+  // sender) is relying on LNTF alone.
+  if (keyframe_request_sender_) {
+    keyframe_request_sender_->RequestKeyFrame();
+  } else {
+    rtp_rtcp_->SendPictureLossIndication();
+  }
+}
+
+void RtpVideoStreamReceiver2::SendLossNotification(
+    uint16_t last_decoded_seq_num,
+    uint16_t last_received_seq_num,
+    bool decodability_flag,
+    bool buffering_allowed) {
+  RTC_DCHECK(config_.rtp.lntf.enabled);
+  rtp_rtcp_->SendLossNotification(last_decoded_seq_num, last_received_seq_num,
+                                  decodability_flag, buffering_allowed);
+}
+
+bool RtpVideoStreamReceiver2::IsUlpfecEnabled() const {
+  return config_.rtp.ulpfec_payload_type != -1;
+}
+
+bool RtpVideoStreamReceiver2::IsRetransmissionsEnabled() const {
+  return config_.rtp.nack.rtp_history_ms > 0;
+}
+
+void RtpVideoStreamReceiver2::RequestPacketRetransmit(
+    const std::vector<uint16_t>& sequence_numbers) {
+  rtp_rtcp_->SendNack(sequence_numbers);
+}
+
+bool RtpVideoStreamReceiver2::IsDecryptable() const {
+  return frames_decryptable_.load();
+}
+
+void RtpVideoStreamReceiver2::OnInsertedPacket(
+    video_coding::PacketBuffer::InsertResult result) {
+  video_coding::PacketBuffer::Packet* first_packet = nullptr;
+  int max_nack_count;
+  int64_t min_recv_time;
+  int64_t max_recv_time;
+  std::vector<rtc::ArrayView<const uint8_t>> payloads;
+  RtpPacketInfos::vector_type packet_infos;
+
+  bool frame_boundary = true;
+  for (auto& packet : result.packets) {
+    // PacketBuffer promisses frame boundaries are correctly set on each
+    // packet. Document that assumption with the DCHECKs.
+    RTC_DCHECK_EQ(frame_boundary, packet->is_first_packet_in_frame());
+    if (packet->is_first_packet_in_frame()) {
+      first_packet = packet.get();
+      max_nack_count = packet->times_nacked;
+      min_recv_time = packet->packet_info.receive_time_ms();
+      max_recv_time = packet->packet_info.receive_time_ms();
+      payloads.clear();
+      packet_infos.clear();
+    } else {
+      max_nack_count = std::max(max_nack_count, packet->times_nacked);
+      min_recv_time =
+          std::min(min_recv_time, packet->packet_info.receive_time_ms());
+      max_recv_time =
+          std::max(max_recv_time, packet->packet_info.receive_time_ms());
+    }
+    payloads.emplace_back(packet->video_payload);
+    packet_infos.push_back(packet->packet_info);
+
+    frame_boundary = packet->is_last_packet_in_frame();
+    if (packet->is_last_packet_in_frame()) {
+      auto depacketizer_it = payload_type_map_.find(first_packet->payload_type);
+      RTC_CHECK(depacketizer_it != payload_type_map_.end());
+
+      rtc::scoped_refptr<EncodedImageBuffer> bitstream =
+          depacketizer_it->second->AssembleFrame(payloads);
+      if (!bitstream) {
+        // Failed to assemble a frame. Discard and continue.
+        continue;
+      }
+
+      const video_coding::PacketBuffer::Packet& last_packet = *packet;
+      OnAssembledFrame(std::make_unique<video_coding::RtpFrameObject>(
+          first_packet->seq_num,                    //
+          last_packet.seq_num,                      //
+          last_packet.marker_bit,                   //
+          max_nack_count,                           //
+          min_recv_time,                            //
+          max_recv_time,                            //
+          first_packet->timestamp,                  //
+          first_packet->ntp_time_ms,                //
+          last_packet.video_header.video_timing,    //
+          first_packet->payload_type,               //
+          first_packet->codec(),                    //
+          last_packet.video_header.rotation,        //
+          last_packet.video_header.content_type,    //
+          first_packet->video_header,               //
+          last_packet.video_header.color_space,     //
+          RtpPacketInfos(std::move(packet_infos)),  //
+          std::move(bitstream)));
+    }
+  }
+  RTC_DCHECK(frame_boundary);
+  if (result.buffer_cleared) {
+    RequestKeyFrame();
+  }
+}
+
+void RtpVideoStreamReceiver2::OnAssembledFrame(
+    std::unique_ptr<video_coding::RtpFrameObject> frame) {
+  RTC_DCHECK_RUN_ON(&network_tc_);
+  RTC_DCHECK(frame);
+
+  const absl::optional<RTPVideoHeader::GenericDescriptorInfo>& descriptor =
+      frame->GetRtpVideoHeader().generic;
+
+  if (loss_notification_controller_ && descriptor) {
+    loss_notification_controller_->OnAssembledFrame(
+        frame->first_seq_num(), descriptor->frame_id,
+        absl::c_linear_search(descriptor->decode_target_indications,
+                              DecodeTargetIndication::kDiscardable),
+        descriptor->dependencies);
+  }
+
+  // If frames arrive before a key frame, they would not be decodable.
+  // In that case, request a key frame ASAP.
+  if (!has_received_frame_) {
+    if (frame->FrameType() != VideoFrameType::kVideoFrameKey) {
+      // |loss_notification_controller_|, if present, would have already
+      // requested a key frame when the first packet for the non-key frame
+      // had arrived, so no need to replicate the request.
+      if (!loss_notification_controller_) {
+        RequestKeyFrame();
+      }
+    }
+    has_received_frame_ = true;
+  }
+
+  rtc::CritScope lock(&reference_finder_lock_);
+  // Reset |reference_finder_| if |frame| is new and the codec have changed.
+  if (current_codec_) {
+    bool frame_is_newer =
+        AheadOf(frame->Timestamp(), last_assembled_frame_rtp_timestamp_);
+
+    if (frame->codec_type() != current_codec_) {
+      if (frame_is_newer) {
+        // When we reset the |reference_finder_| we don't want new picture ids
+        // to overlap with old picture ids. To ensure that doesn't happen we
+        // start from the |last_completed_picture_id_| and add an offset in case
+        // of reordering.
+        reference_finder_ =
+            std::make_unique<video_coding::RtpFrameReferenceFinder>(
+                this, last_completed_picture_id_ +
+                          std::numeric_limits<uint16_t>::max());
+        current_codec_ = frame->codec_type();
+      } else {
+        // Old frame from before the codec switch, discard it.
+        return;
+      }
+    }
+
+    if (frame_is_newer) {
+      last_assembled_frame_rtp_timestamp_ = frame->Timestamp();
+    }
+  } else {
+    current_codec_ = frame->codec_type();
+    last_assembled_frame_rtp_timestamp_ = frame->Timestamp();
+  }
+
+  if (buffered_frame_decryptor_ != nullptr) {
+    buffered_frame_decryptor_->ManageEncryptedFrame(std::move(frame));
+  } else if (frame_transformer_delegate_) {
+    frame_transformer_delegate_->TransformFrame(std::move(frame));
+  } else {
+    reference_finder_->ManageFrame(std::move(frame));
+  }
+}
+
+void RtpVideoStreamReceiver2::OnCompleteFrame(
+    std::unique_ptr<video_coding::EncodedFrame> frame) {
+  {
+    rtc::CritScope lock(&last_seq_num_cs_);
+    video_coding::RtpFrameObject* rtp_frame =
+        static_cast<video_coding::RtpFrameObject*>(frame.get());
+    last_seq_num_for_pic_id_[rtp_frame->id.picture_id] =
+        rtp_frame->last_seq_num();
+  }
+  last_completed_picture_id_ =
+      std::max(last_completed_picture_id_, frame->id.picture_id);
+  complete_frame_callback_->OnCompleteFrame(std::move(frame));
+}
+
+void RtpVideoStreamReceiver2::OnDecryptedFrame(
+    std::unique_ptr<video_coding::RtpFrameObject> frame) {
+  rtc::CritScope lock(&reference_finder_lock_);
+  reference_finder_->ManageFrame(std::move(frame));
+}
+
+void RtpVideoStreamReceiver2::OnDecryptionStatusChange(
+    FrameDecryptorInterface::Status status) {
+  frames_decryptable_.store(
+      (status == FrameDecryptorInterface::Status::kOk) ||
+      (status == FrameDecryptorInterface::Status::kRecoverable));
+}
+
+void RtpVideoStreamReceiver2::SetFrameDecryptor(
+    rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor) {
+  RTC_DCHECK_RUN_ON(&network_tc_);
+  if (buffered_frame_decryptor_ == nullptr) {
+    buffered_frame_decryptor_ =
+        std::make_unique<BufferedFrameDecryptor>(this, this);
+  }
+  buffered_frame_decryptor_->SetFrameDecryptor(std::move(frame_decryptor));
+}
+
+void RtpVideoStreamReceiver2::SetDepacketizerToDecoderFrameTransformer(
+    rtc::scoped_refptr<FrameTransformerInterface> frame_transformer) {
+  RTC_DCHECK_RUN_ON(&network_tc_);
+  frame_transformer_delegate_ =
+      new rtc::RefCountedObject<RtpVideoStreamReceiverFrameTransformerDelegate>(
+          this, std::move(frame_transformer), rtc::Thread::Current(),
+          config_.rtp.remote_ssrc);
+  frame_transformer_delegate_->Init();
+}
+
+void RtpVideoStreamReceiver2::UpdateRtt(int64_t max_rtt_ms) {
+  if (nack_module_)
+    nack_module_->UpdateRtt(max_rtt_ms);
+}
+
+absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedPacketMs() const {
+  return packet_buffer_.LastReceivedPacketMs();
+}
+
+absl::optional<int64_t> RtpVideoStreamReceiver2::LastReceivedKeyframePacketMs()
+    const {
+  return packet_buffer_.LastReceivedKeyframePacketMs();
+}
+
+void RtpVideoStreamReceiver2::AddSecondarySink(RtpPacketSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(&worker_task_checker_);
+  RTC_DCHECK(!absl::c_linear_search(secondary_sinks_, sink));
+  secondary_sinks_.push_back(sink);
+}
+
+void RtpVideoStreamReceiver2::RemoveSecondarySink(
+    const RtpPacketSinkInterface* sink) {
+  RTC_DCHECK_RUN_ON(&worker_task_checker_);
+  auto it = absl::c_find(secondary_sinks_, sink);
+  if (it == secondary_sinks_.end()) {
+    // We might be rolling-back a call whose setup failed mid-way. In such a
+    // case, it's simpler to remove "everything" rather than remember what
+    // has already been added.
+    RTC_LOG(LS_WARNING) << "Removal of unknown sink.";
+    return;
+  }
+  secondary_sinks_.erase(it);
+}
+
+void RtpVideoStreamReceiver2::ManageFrame(
+    std::unique_ptr<video_coding::RtpFrameObject> frame) {
+  rtc::CritScope lock(&reference_finder_lock_);
+  reference_finder_->ManageFrame(std::move(frame));
+}
+
+void RtpVideoStreamReceiver2::ReceivePacket(const RtpPacketReceived& packet) {
+  if (packet.payload_size() == 0) {
+    // Padding or keep-alive packet.
+    // TODO(nisse): Could drop empty packets earlier, but need to figure out how
+    // they should be counted in stats.
+    NotifyReceiverOfEmptyPacket(packet.SequenceNumber());
+    return;
+  }
+  if (packet.PayloadType() == config_.rtp.red_payload_type) {
+    ParseAndHandleEncapsulatingHeader(packet);
+    return;
+  }
+
+  const auto type_it = payload_type_map_.find(packet.PayloadType());
+  if (type_it == payload_type_map_.end()) {
+    return;
+  }
+  absl::optional<VideoRtpDepacketizer::ParsedRtpPayload> parsed_payload =
+      type_it->second->Parse(packet.PayloadBuffer());
+  if (parsed_payload == absl::nullopt) {
+    RTC_LOG(LS_WARNING) << "Failed parsing payload.";
+    return;
+  }
+
+  OnReceivedPayloadData(std::move(parsed_payload->video_payload), packet,
+                        parsed_payload->video_header);
+}
+
+void RtpVideoStreamReceiver2::ParseAndHandleEncapsulatingHeader(
+    const RtpPacketReceived& packet) {
+  RTC_DCHECK_RUN_ON(&worker_task_checker_);
+  if (packet.PayloadType() == config_.rtp.red_payload_type &&
+      packet.payload_size() > 0) {
+    if (packet.payload()[0] == config_.rtp.ulpfec_payload_type) {
+      // Notify video_receiver about received FEC packets to avoid NACKing these
+      // packets.
+      NotifyReceiverOfEmptyPacket(packet.SequenceNumber());
+    }
+    if (!ulpfec_receiver_->AddReceivedRedPacket(
+            packet, config_.rtp.ulpfec_payload_type)) {
+      return;
+    }
+    ulpfec_receiver_->ProcessReceivedFec();
+  }
+}
+
+// In the case of a video stream without picture ids and no rtx the
+// RtpFrameReferenceFinder will need to know about padding to
+// correctly calculate frame references.
+void RtpVideoStreamReceiver2::NotifyReceiverOfEmptyPacket(uint16_t seq_num) {
+  {
+    rtc::CritScope lock(&reference_finder_lock_);
+    reference_finder_->PaddingReceived(seq_num);
+  }
+  OnInsertedPacket(packet_buffer_.InsertPadding(seq_num));
+  if (nack_module_) {
+    nack_module_->OnReceivedPacket(seq_num, /* is_keyframe = */ false,
+                                   /* is _recovered = */ false);
+  }
+  if (loss_notification_controller_) {
+    // TODO(bugs.webrtc.org/10336): Handle empty packets.
+    RTC_LOG(LS_WARNING)
+        << "LossNotificationController does not expect empty packets.";
+  }
+}
+
+bool RtpVideoStreamReceiver2::DeliverRtcp(const uint8_t* rtcp_packet,
+                                          size_t rtcp_packet_length) {
+  RTC_DCHECK_RUN_ON(&worker_task_checker_);
+
+  if (!receiving_) {
+    return false;
+  }
+
+  rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
+
+  int64_t rtt = 0;
+  rtp_rtcp_->RTT(config_.rtp.remote_ssrc, &rtt, nullptr, nullptr, nullptr);
+  if (rtt == 0) {
+    // Waiting for valid rtt.
+    return true;
+  }
+  uint32_t ntp_secs = 0;
+  uint32_t ntp_frac = 0;
+  uint32_t rtp_timestamp = 0;
+  uint32_t recieved_ntp_secs = 0;
+  uint32_t recieved_ntp_frac = 0;
+  if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, &recieved_ntp_secs,
+                           &recieved_ntp_frac, &rtp_timestamp) != 0) {
+    // Waiting for RTCP.
+    return true;
+  }
+  NtpTime recieved_ntp(recieved_ntp_secs, recieved_ntp_frac);
+  int64_t time_since_recieved =
+      clock_->CurrentNtpInMilliseconds() - recieved_ntp.ToMs();
+  // Don't use old SRs to estimate time.
+  if (time_since_recieved <= 1) {
+    ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
+    absl::optional<int64_t> remote_to_local_clock_offset_ms =
+        ntp_estimator_.EstimateRemoteToLocalClockOffsetMs();
+    if (remote_to_local_clock_offset_ms.has_value()) {
+      absolute_capture_time_receiver_.SetRemoteToLocalClockOffset(
+          Int64MsToQ32x32(*remote_to_local_clock_offset_ms));
+    }
+  }
+
+  return true;
+}
+
+void RtpVideoStreamReceiver2::FrameContinuous(int64_t picture_id) {
+  if (!nack_module_)
+    return;
+
+  int seq_num = -1;
+  {
+    rtc::CritScope lock(&last_seq_num_cs_);
+    auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
+    if (seq_num_it != last_seq_num_for_pic_id_.end())
+      seq_num = seq_num_it->second;
+  }
+  if (seq_num != -1)
+    nack_module_->ClearUpTo(seq_num);
+}
+
+void RtpVideoStreamReceiver2::FrameDecoded(int64_t picture_id) {
+  int seq_num = -1;
+  {
+    rtc::CritScope lock(&last_seq_num_cs_);
+    auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
+    if (seq_num_it != last_seq_num_for_pic_id_.end()) {
+      seq_num = seq_num_it->second;
+      last_seq_num_for_pic_id_.erase(last_seq_num_for_pic_id_.begin(),
+                                     ++seq_num_it);
+    }
+  }
+  if (seq_num != -1) {
+    packet_buffer_.ClearTo(seq_num);
+    rtc::CritScope lock(&reference_finder_lock_);
+    reference_finder_->ClearTo(seq_num);
+  }
+}
+
+void RtpVideoStreamReceiver2::SignalNetworkState(NetworkState state) {
+  rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
+                                               : RtcpMode::kOff);
+}
+
+void RtpVideoStreamReceiver2::StartReceive() {
+  RTC_DCHECK_RUN_ON(&worker_task_checker_);
+  receiving_ = true;
+}
+
+void RtpVideoStreamReceiver2::StopReceive() {
+  RTC_DCHECK_RUN_ON(&worker_task_checker_);
+  receiving_ = false;
+}
+
+void RtpVideoStreamReceiver2::UpdateHistograms() {
+  FecPacketCounter counter = ulpfec_receiver_->GetPacketCounter();
+  if (counter.first_packet_time_ms == -1)
+    return;
+
+  int64_t elapsed_sec =
+      (clock_->TimeInMilliseconds() - counter.first_packet_time_ms) / 1000;
+  if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+    return;
+
+  if (counter.num_packets > 0) {
+    RTC_HISTOGRAM_PERCENTAGE(
+        "WebRTC.Video.ReceivedFecPacketsInPercent",
+        static_cast<int>(counter.num_fec_packets * 100 / counter.num_packets));
+  }
+  if (counter.num_fec_packets > 0) {
+    RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.RecoveredMediaPacketsInPercentOfFec",
+                             static_cast<int>(counter.num_recovered_packets *
+                                              100 / counter.num_fec_packets));
+  }
+  if (config_.rtp.ulpfec_payload_type != -1) {
+    RTC_HISTOGRAM_COUNTS_10000(
+        "WebRTC.Video.FecBitrateReceivedInKbps",
+        static_cast<int>(counter.num_bytes * 8 / elapsed_sec / 1000));
+  }
+}
+
+void RtpVideoStreamReceiver2::InsertSpsPpsIntoTracker(uint8_t payload_type) {
+  auto codec_params_it = pt_codec_params_.find(payload_type);
+  if (codec_params_it == pt_codec_params_.end())
+    return;
+
+  RTC_LOG(LS_INFO) << "Found out of band supplied codec parameters for"
+                      " payload type: "
+                   << static_cast<int>(payload_type);
+
+  H264SpropParameterSets sprop_decoder;
+  auto sprop_base64_it =
+      codec_params_it->second.find(cricket::kH264FmtpSpropParameterSets);
+
+  if (sprop_base64_it == codec_params_it->second.end())
+    return;
+
+  if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
+    return;
+
+  tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
+                             sprop_decoder.pps_nalu());
+}
+
+}  // namespace webrtc
diff --git a/video/rtp_video_stream_receiver2.h b/video/rtp_video_stream_receiver2.h
new file mode 100644
index 0000000..3026e1d
--- /dev/null
+++ b/video/rtp_video_stream_receiver2.h
@@ -0,0 +1,386 @@
+/*
+ *  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.
+ */
+
+#ifndef VIDEO_RTP_VIDEO_STREAM_RECEIVER2_H_
+#define VIDEO_RTP_VIDEO_STREAM_RECEIVER2_H_
+
+#include <atomic>
+#include <list>
+#include <map>
+#include <memory>
+#include <string>
+#include <vector>
+
+#include "absl/types/optional.h"
+#include "api/array_view.h"
+#include "api/crypto/frame_decryptor_interface.h"
+#include "api/video/color_space.h"
+#include "api/video_codecs/video_codec.h"
+#include "call/rtp_packet_sink_interface.h"
+#include "call/syncable.h"
+#include "call/video_receive_stream.h"
+#include "modules/rtp_rtcp/include/receive_statistics.h"
+#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/absolute_capture_time_receiver.h"
+#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_video_header.h"
+#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
+#include "modules/video_coding/h264_sps_pps_tracker.h"
+#include "modules/video_coding/loss_notification_controller.h"
+#include "modules/video_coding/packet_buffer.h"
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+#include "modules/video_coding/unique_timestamp_counter.h"
+#include "rtc_base/constructor_magic.h"
+#include "rtc_base/critical_section.h"
+#include "rtc_base/experiments/field_trial_parser.h"
+#include "rtc_base/numerics/sequence_number_util.h"
+#include "rtc_base/synchronization/sequence_checker.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/thread_checker.h"
+#include "video/buffered_frame_decryptor.h"
+#include "video/rtp_video_stream_receiver_frame_transformer_delegate.h"
+
+namespace webrtc {
+
+class NackModule2;
+class PacketRouter;
+class ProcessThread;
+class ReceiveStatistics;
+class RtcpRttStats;
+class RtpPacketReceived;
+class Transport;
+class UlpfecReceiver;
+
+class RtpVideoStreamReceiver2 : public LossNotificationSender,
+                                public RecoveredPacketReceiver,
+                                public RtpPacketSinkInterface,
+                                public KeyFrameRequestSender,
+                                public video_coding::OnCompleteFrameCallback,
+                                public OnDecryptedFrameCallback,
+                                public OnDecryptionStatusChangeCallback,
+                                public RtpVideoFrameReceiver {
+ public:
+  RtpVideoStreamReceiver2(
+      Clock* clock,
+      Transport* transport,
+      RtcpRttStats* rtt_stats,
+      // The packet router is optional; if provided, the RtpRtcp module for this
+      // stream is registered as a candidate for sending REMB and transport
+      // feedback.
+      PacketRouter* packet_router,
+      const VideoReceiveStream::Config* config,
+      ReceiveStatistics* rtp_receive_statistics,
+      RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer,
+      RtcpCnameCallback* rtcp_cname_callback,
+      ProcessThread* process_thread,
+      NackSender* nack_sender,
+      // The KeyFrameRequestSender is optional; if not provided, key frame
+      // requests are sent via the internal RtpRtcp module.
+      KeyFrameRequestSender* keyframe_request_sender,
+      video_coding::OnCompleteFrameCallback* complete_frame_callback,
+      rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor,
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+  ~RtpVideoStreamReceiver2() override;
+
+  void AddReceiveCodec(const VideoCodec& video_codec,
+                       const std::map<std::string, std::string>& codec_params,
+                       bool raw_payload);
+
+  void StartReceive();
+  void StopReceive();
+
+  // Produces the transport-related timestamps; current_delay_ms is left unset.
+  absl::optional<Syncable::Info> GetSyncInfo() const;
+
+  bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
+
+  void FrameContinuous(int64_t seq_num);
+
+  void FrameDecoded(int64_t seq_num);
+
+  void SignalNetworkState(NetworkState state);
+
+  // Returns number of different frames seen.
+  int GetUniqueFramesSeen() const {
+    RTC_DCHECK_RUN_ON(&worker_task_checker_);
+    return frame_counter_.GetUniqueSeen();
+  }
+
+  // Implements RtpPacketSinkInterface.
+  void OnRtpPacket(const RtpPacketReceived& packet) override;
+
+  // TODO(philipel): Stop using VCMPacket in the new jitter buffer and then
+  //                 remove this function. Public only for tests.
+  void OnReceivedPayloadData(rtc::CopyOnWriteBuffer codec_payload,
+                             const RtpPacketReceived& rtp_packet,
+                             const RTPVideoHeader& video);
+
+  // Implements RecoveredPacketReceiver.
+  void OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
+
+  // Send an RTCP keyframe request.
+  void RequestKeyFrame() override;
+
+  // Implements LossNotificationSender.
+  void SendLossNotification(uint16_t last_decoded_seq_num,
+                            uint16_t last_received_seq_num,
+                            bool decodability_flag,
+                            bool buffering_allowed) override;
+
+  bool IsUlpfecEnabled() const;
+  bool IsRetransmissionsEnabled() const;
+
+  // Returns true if a decryptor is attached and frames can be decrypted.
+  // Updated by OnDecryptionStatusChangeCallback. Note this refers to Frame
+  // Decryption not SRTP.
+  bool IsDecryptable() const;
+
+  // Don't use, still experimental.
+  void RequestPacketRetransmit(const std::vector<uint16_t>& sequence_numbers);
+
+  // Implements OnCompleteFrameCallback.
+  void OnCompleteFrame(
+      std::unique_ptr<video_coding::EncodedFrame> frame) override;
+
+  // Implements OnDecryptedFrameCallback.
+  void OnDecryptedFrame(
+      std::unique_ptr<video_coding::RtpFrameObject> frame) override;
+
+  // Implements OnDecryptionStatusChangeCallback.
+  void OnDecryptionStatusChange(
+      FrameDecryptorInterface::Status status) override;
+
+  // Optionally set a frame decryptor after a stream has started. This will not
+  // reset the decoder state.
+  void SetFrameDecryptor(
+      rtc::scoped_refptr<FrameDecryptorInterface> frame_decryptor);
+
+  // Sets a frame transformer after a stream has started, if no transformer
+  // has previously been set. Does not reset the decoder state.
+  void SetDepacketizerToDecoderFrameTransformer(
+      rtc::scoped_refptr<FrameTransformerInterface> frame_transformer);
+
+  // Called by VideoReceiveStream when stats are updated.
+  void UpdateRtt(int64_t max_rtt_ms);
+
+  absl::optional<int64_t> LastReceivedPacketMs() const;
+  absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
+
+  // RtpDemuxer only forwards a given RTP packet to one sink. However, some
+  // sinks, such as FlexFEC, might wish to be informed of all of the packets
+  // a given sink receives (or any set of sinks). They may do so by registering
+  // themselves as secondary sinks.
+  void AddSecondarySink(RtpPacketSinkInterface* sink);
+  void RemoveSecondarySink(const RtpPacketSinkInterface* sink);
+
+ private:
+  // Implements RtpVideoFrameReceiver.
+  void ManageFrame(
+      std::unique_ptr<video_coding::RtpFrameObject> frame) override;
+
+  // Used for buffering RTCP feedback messages and sending them all together.
+  // Note:
+  // 1. Key frame requests and NACKs are mutually exclusive, with the
+  //    former taking precedence over the latter.
+  // 2. Loss notifications are orthogonal to either. (That is, may be sent
+  //    alongside either.)
+  class RtcpFeedbackBuffer : public KeyFrameRequestSender,
+                             public NackSender,
+                             public LossNotificationSender {
+   public:
+    RtcpFeedbackBuffer(KeyFrameRequestSender* key_frame_request_sender,
+                       NackSender* nack_sender,
+                       LossNotificationSender* loss_notification_sender);
+
+    ~RtcpFeedbackBuffer() override = default;
+
+    // KeyFrameRequestSender implementation.
+    void RequestKeyFrame() RTC_LOCKS_EXCLUDED(cs_) override;
+
+    // NackSender implementation.
+    void SendNack(const std::vector<uint16_t>& sequence_numbers,
+                  bool buffering_allowed) RTC_LOCKS_EXCLUDED(cs_) override;
+
+    // LossNotificationSender implementation.
+    void SendLossNotification(uint16_t last_decoded_seq_num,
+                              uint16_t last_received_seq_num,
+                              bool decodability_flag,
+                              bool buffering_allowed)
+        RTC_LOCKS_EXCLUDED(cs_) override;
+
+    // Send all RTCP feedback messages buffered thus far.
+    void SendBufferedRtcpFeedback() RTC_LOCKS_EXCLUDED(cs_);
+
+   private:
+    // LNTF-related state.
+    struct LossNotificationState {
+      LossNotificationState(uint16_t last_decoded_seq_num,
+                            uint16_t last_received_seq_num,
+                            bool decodability_flag)
+          : last_decoded_seq_num(last_decoded_seq_num),
+            last_received_seq_num(last_received_seq_num),
+            decodability_flag(decodability_flag) {}
+
+      uint16_t last_decoded_seq_num;
+      uint16_t last_received_seq_num;
+      bool decodability_flag;
+    };
+    struct ConsumedRtcpFeedback {
+      bool request_key_frame = false;
+      std::vector<uint16_t> nack_sequence_numbers;
+      absl::optional<LossNotificationState> lntf_state;
+    };
+
+    ConsumedRtcpFeedback ConsumeRtcpFeedback() RTC_LOCKS_EXCLUDED(cs_);
+    ConsumedRtcpFeedback ConsumeRtcpFeedbackLocked()
+        RTC_EXCLUSIVE_LOCKS_REQUIRED(cs_);
+    // This method is called both with and without cs_ held.
+    void SendRtcpFeedback(ConsumedRtcpFeedback feedback);
+
+    KeyFrameRequestSender* const key_frame_request_sender_;
+    NackSender* const nack_sender_;
+    LossNotificationSender* const loss_notification_sender_;
+
+    // NACKs are accessible from two threads due to nack_module_ being a module.
+    rtc::CriticalSection cs_;
+
+    // Key-frame-request-related state.
+    bool request_key_frame_ RTC_GUARDED_BY(cs_);
+
+    // NACK-related state.
+    std::vector<uint16_t> nack_sequence_numbers_ RTC_GUARDED_BY(cs_);
+
+    absl::optional<LossNotificationState> lntf_state_ RTC_GUARDED_BY(cs_);
+  };
+  enum ParseGenericDependenciesResult {
+    kDropPacket,
+    kHasGenericDescriptor,
+    kNoGenericDescriptor
+  };
+
+  // Entry point doing non-stats work for a received packet. Called
+  // for the same packet both before and after RED decapsulation.
+  void ReceivePacket(const RtpPacketReceived& packet);
+  // Parses and handles RED headers.
+  // This function assumes that it's being called from only one thread.
+  void ParseAndHandleEncapsulatingHeader(const RtpPacketReceived& packet);
+  void NotifyReceiverOfEmptyPacket(uint16_t seq_num);
+  void UpdateHistograms();
+  bool IsRedEnabled() const;
+  void InsertSpsPpsIntoTracker(uint8_t payload_type);
+  void OnInsertedPacket(video_coding::PacketBuffer::InsertResult result);
+  ParseGenericDependenciesResult ParseGenericDependenciesExtension(
+      const RtpPacketReceived& rtp_packet,
+      RTPVideoHeader* video_header) RTC_RUN_ON(worker_task_checker_);
+  void OnAssembledFrame(std::unique_ptr<video_coding::RtpFrameObject> frame);
+
+  Clock* const clock_;
+  // Ownership of this object lies with VideoReceiveStream, which owns |this|.
+  const VideoReceiveStream::Config& config_;
+  PacketRouter* const packet_router_;
+  ProcessThread* const process_thread_;
+
+  RemoteNtpTimeEstimator ntp_estimator_;
+
+  RtpHeaderExtensionMap rtp_header_extensions_;
+  // Set by the field trial WebRTC-ForcePlayoutDelay to override any playout
+  // delay that is specified in the received packets.
+  FieldTrialOptional<int> forced_playout_delay_max_ms_;
+  FieldTrialOptional<int> forced_playout_delay_min_ms_;
+  ReceiveStatistics* const rtp_receive_statistics_;
+  std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
+
+  SequenceChecker worker_task_checker_;
+  bool receiving_ RTC_GUARDED_BY(worker_task_checker_);
+  int64_t last_packet_log_ms_ RTC_GUARDED_BY(worker_task_checker_);
+
+  const std::unique_ptr<RtpRtcp> rtp_rtcp_;
+
+  video_coding::OnCompleteFrameCallback* complete_frame_callback_;
+  KeyFrameRequestSender* const keyframe_request_sender_;
+
+  RtcpFeedbackBuffer rtcp_feedback_buffer_;
+  std::unique_ptr<NackModule2> nack_module_;
+  std::unique_ptr<LossNotificationController> loss_notification_controller_;
+
+  video_coding::PacketBuffer packet_buffer_;
+  UniqueTimestampCounter frame_counter_ RTC_GUARDED_BY(worker_task_checker_);
+  SeqNumUnwrapper<uint16_t> frame_id_unwrapper_
+      RTC_GUARDED_BY(worker_task_checker_);
+
+  // Video structure provided in the dependency descriptor in a first packet
+  // of a key frame. It is required to parse dependency descriptor in the
+  // following delta packets.
+  std::unique_ptr<FrameDependencyStructure> video_structure_
+      RTC_GUARDED_BY(worker_task_checker_);
+  // Frame id of the last frame with the attached video structure.
+  // absl::nullopt when `video_structure_ == nullptr`;
+  absl::optional<int64_t> video_structure_frame_id_
+      RTC_GUARDED_BY(worker_task_checker_);
+
+  rtc::CriticalSection reference_finder_lock_;
+  std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_
+      RTC_GUARDED_BY(reference_finder_lock_);
+  absl::optional<VideoCodecType> current_codec_;
+  uint32_t last_assembled_frame_rtp_timestamp_;
+
+  rtc::CriticalSection last_seq_num_cs_;
+  std::map<int64_t, uint16_t> last_seq_num_for_pic_id_
+      RTC_GUARDED_BY(last_seq_num_cs_);
+  video_coding::H264SpsPpsTracker tracker_;
+
+  // Maps payload id to the depacketizer.
+  std::map<uint8_t, std::unique_ptr<VideoRtpDepacketizer>> payload_type_map_;
+
+  // TODO(johan): Remove pt_codec_params_ once
+  // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved.
+  // Maps a payload type to a map of out-of-band supplied codec parameters.
+  std::map<uint8_t, std::map<std::string, std::string>> pt_codec_params_;
+  int16_t last_payload_type_ = -1;
+
+  bool has_received_frame_;
+
+  std::vector<RtpPacketSinkInterface*> secondary_sinks_
+      RTC_GUARDED_BY(worker_task_checker_);
+
+  // Info for GetSyncInfo is updated on network or worker thread, and queried on
+  // the worker thread.
+  rtc::CriticalSection sync_info_lock_;
+  absl::optional<uint32_t> last_received_rtp_timestamp_
+      RTC_GUARDED_BY(sync_info_lock_);
+  absl::optional<int64_t> last_received_rtp_system_time_ms_
+      RTC_GUARDED_BY(sync_info_lock_);
+
+  // Used to validate the buffered frame decryptor is always run on the correct
+  // thread.
+  rtc::ThreadChecker network_tc_;
+  // Handles incoming encrypted frames and forwards them to the
+  // rtp_reference_finder if they are decryptable.
+  std::unique_ptr<BufferedFrameDecryptor> buffered_frame_decryptor_
+      RTC_PT_GUARDED_BY(network_tc_);
+  std::atomic<bool> frames_decryptable_;
+  absl::optional<ColorSpace> last_color_space_;
+
+  AbsoluteCaptureTimeReceiver absolute_capture_time_receiver_
+      RTC_GUARDED_BY(worker_task_checker_);
+
+  int64_t last_completed_picture_id_ = 0;
+
+  rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate>
+      frame_transformer_delegate_;
+};
+
+}  // namespace webrtc
+
+#endif  // VIDEO_RTP_VIDEO_STREAM_RECEIVER2_H_
diff --git a/video/rtp_video_stream_receiver2_unittest.cc b/video/rtp_video_stream_receiver2_unittest.cc
new file mode 100644
index 0000000..c8584fc
--- /dev/null
+++ b/video/rtp_video_stream_receiver2_unittest.cc
@@ -0,0 +1,1215 @@
+/*
+ *  Copyright 2017 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 "video/rtp_video_stream_receiver2.h"
+
+#include <memory>
+#include <utility>
+
+#include "api/video/video_codec_type.h"
+#include "api/video/video_frame_type.h"
+#include "common_video/h264/h264_common.h"
+#include "media/base/media_constants.h"
+#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
+#include "modules/rtp_rtcp/source/rtp_format.h"
+#include "modules/rtp_rtcp/source/rtp_format_vp9.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
+#include "modules/utility/include/process_thread.h"
+#include "modules/video_coding/frame_object.h"
+#include "modules/video_coding/include/video_coding_defines.h"
+#include "modules/video_coding/rtp_frame_reference_finder.h"
+#include "rtc_base/byte_buffer.h"
+#include "rtc_base/logging.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/field_trial.h"
+#include "test/field_trial.h"
+#include "test/gmock.h"
+#include "test/gtest.h"
+#include "test/mock_frame_transformer.h"
+
+using ::testing::_;
+using ::testing::ElementsAre;
+using ::testing::Invoke;
+using ::testing::SizeIs;
+using ::testing::Values;
+
+namespace webrtc {
+
+namespace {
+
+const uint8_t kH264StartCode[] = {0x00, 0x00, 0x00, 0x01};
+
+std::vector<uint64_t> GetAbsoluteCaptureTimestamps(
+    const video_coding::EncodedFrame* frame) {
+  std::vector<uint64_t> result;
+  for (const auto& packet_info : frame->PacketInfos()) {
+    if (packet_info.absolute_capture_time()) {
+      result.push_back(
+          packet_info.absolute_capture_time()->absolute_capture_timestamp);
+    }
+  }
+  return result;
+}
+
+RTPVideoHeader GetGenericVideoHeader(VideoFrameType frame_type) {
+  RTPVideoHeader video_header;
+  video_header.is_first_packet_in_frame = true;
+  video_header.is_last_packet_in_frame = true;
+  video_header.codec = kVideoCodecGeneric;
+  video_header.frame_type = frame_type;
+  return video_header;
+}
+
+class MockTransport : public Transport {
+ public:
+  MOCK_METHOD(bool,
+              SendRtp,
+              (const uint8_t*, size_t length, const PacketOptions& options),
+              (override));
+  MOCK_METHOD(bool, SendRtcp, (const uint8_t*, size_t length), (override));
+};
+
+class MockNackSender : public NackSender {
+ public:
+  MOCK_METHOD(void,
+              SendNack,
+              (const std::vector<uint16_t>& sequence_numbers,
+               bool buffering_allowed),
+              (override));
+};
+
+class MockKeyFrameRequestSender : public KeyFrameRequestSender {
+ public:
+  MOCK_METHOD(void, RequestKeyFrame, (), (override));
+};
+
+class MockOnCompleteFrameCallback
+    : public video_coding::OnCompleteFrameCallback {
+ public:
+  MOCK_METHOD(void, DoOnCompleteFrame, (video_coding::EncodedFrame*), ());
+  MOCK_METHOD(void,
+              DoOnCompleteFrameFailNullptr,
+              (video_coding::EncodedFrame*),
+              ());
+  MOCK_METHOD(void,
+              DoOnCompleteFrameFailLength,
+              (video_coding::EncodedFrame*),
+              ());
+  MOCK_METHOD(void,
+              DoOnCompleteFrameFailBitstream,
+              (video_coding::EncodedFrame*),
+              ());
+  void OnCompleteFrame(
+      std::unique_ptr<video_coding::EncodedFrame> frame) override {
+    if (!frame) {
+      DoOnCompleteFrameFailNullptr(nullptr);
+      return;
+    }
+    EXPECT_EQ(buffer_.Length(), frame->size());
+    if (buffer_.Length() != frame->size()) {
+      DoOnCompleteFrameFailLength(frame.get());
+      return;
+    }
+    if (frame->size() != buffer_.Length() ||
+        memcmp(buffer_.Data(), frame->data(), buffer_.Length()) != 0) {
+      DoOnCompleteFrameFailBitstream(frame.get());
+      return;
+    }
+    DoOnCompleteFrame(frame.get());
+  }
+
+  void ClearExpectedBitstream() { buffer_.Clear(); }
+
+  void AppendExpectedBitstream(const uint8_t data[], size_t size_in_bytes) {
+    // TODO(Johan): Let rtc::ByteBuffer handle uint8_t* instead of char*.
+    buffer_.WriteBytes(reinterpret_cast<const char*>(data), size_in_bytes);
+  }
+  rtc::ByteBufferWriter buffer_;
+};
+
+class MockRtpPacketSink : public RtpPacketSinkInterface {
+ public:
+  MOCK_METHOD(void, OnRtpPacket, (const RtpPacketReceived&), (override));
+};
+
+constexpr uint32_t kSsrc = 111;
+constexpr uint16_t kSequenceNumber = 222;
+constexpr int kPayloadType = 100;
+constexpr int kRedPayloadType = 125;
+
+std::unique_ptr<RtpPacketReceived> CreateRtpPacketReceived() {
+  auto packet = std::make_unique<RtpPacketReceived>();
+  packet->SetSsrc(kSsrc);
+  packet->SetSequenceNumber(kSequenceNumber);
+  packet->SetPayloadType(kPayloadType);
+  return packet;
+}
+
+MATCHER_P(SamePacketAs, other, "") {
+  return arg.Ssrc() == other.Ssrc() &&
+         arg.SequenceNumber() == other.SequenceNumber();
+}
+
+}  // namespace
+
+class RtpVideoStreamReceiver2Test : public ::testing::Test {
+ public:
+  RtpVideoStreamReceiver2Test() : RtpVideoStreamReceiver2Test("") {}
+  explicit RtpVideoStreamReceiver2Test(std::string field_trials)
+      : override_field_trials_(field_trials),
+        config_(CreateConfig()),
+        process_thread_(ProcessThread::Create("TestThread")) {
+    rtp_receive_statistics_ =
+        ReceiveStatistics::Create(Clock::GetRealTimeClock());
+    rtp_video_stream_receiver_ = std::make_unique<RtpVideoStreamReceiver2>(
+        Clock::GetRealTimeClock(), &mock_transport_, nullptr, nullptr, &config_,
+        rtp_receive_statistics_.get(), nullptr, nullptr, process_thread_.get(),
+        &mock_nack_sender_, &mock_key_frame_request_sender_,
+        &mock_on_complete_frame_callback_, nullptr, nullptr);
+    VideoCodec codec;
+    codec.plType = kPayloadType;
+    codec.codecType = kVideoCodecGeneric;
+    rtp_video_stream_receiver_->AddReceiveCodec(codec, {},
+                                                /*raw_payload=*/false);
+  }
+
+  RTPVideoHeader GetDefaultH264VideoHeader() {
+    RTPVideoHeader video_header;
+    video_header.codec = kVideoCodecH264;
+    video_header.video_type_header.emplace<RTPVideoHeaderH264>();
+    return video_header;
+  }
+
+  // TODO(Johan): refactor h264_sps_pps_tracker_unittests.cc to avoid duplicate
+  // code.
+  void AddSps(RTPVideoHeader* video_header,
+              uint8_t sps_id,
+              rtc::CopyOnWriteBuffer* data) {
+    NaluInfo info;
+    info.type = H264::NaluType::kSps;
+    info.sps_id = sps_id;
+    info.pps_id = -1;
+    data->AppendData({H264::NaluType::kSps, sps_id});
+    auto& h264 = absl::get<RTPVideoHeaderH264>(video_header->video_type_header);
+    h264.nalus[h264.nalus_length++] = info;
+  }
+
+  void AddPps(RTPVideoHeader* video_header,
+              uint8_t sps_id,
+              uint8_t pps_id,
+              rtc::CopyOnWriteBuffer* data) {
+    NaluInfo info;
+    info.type = H264::NaluType::kPps;
+    info.sps_id = sps_id;
+    info.pps_id = pps_id;
+    data->AppendData({H264::NaluType::kPps, pps_id});
+    auto& h264 = absl::get<RTPVideoHeaderH264>(video_header->video_type_header);
+    h264.nalus[h264.nalus_length++] = info;
+  }
+
+  void AddIdr(RTPVideoHeader* video_header, int pps_id) {
+    NaluInfo info;
+    info.type = H264::NaluType::kIdr;
+    info.sps_id = -1;
+    info.pps_id = pps_id;
+    auto& h264 = absl::get<RTPVideoHeaderH264>(video_header->video_type_header);
+    h264.nalus[h264.nalus_length++] = info;
+  }
+
+ protected:
+  static VideoReceiveStream::Config CreateConfig() {
+    VideoReceiveStream::Config config(nullptr);
+    config.rtp.remote_ssrc = 1111;
+    config.rtp.local_ssrc = 2222;
+    config.rtp.red_payload_type = kRedPayloadType;
+    return config;
+  }
+
+  const webrtc::test::ScopedFieldTrials override_field_trials_;
+  VideoReceiveStream::Config config_;
+  MockNackSender mock_nack_sender_;
+  MockKeyFrameRequestSender mock_key_frame_request_sender_;
+  MockTransport mock_transport_;
+  MockOnCompleteFrameCallback mock_on_complete_frame_callback_;
+  std::unique_ptr<ProcessThread> process_thread_;
+  std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
+  std::unique_ptr<RtpVideoStreamReceiver2> rtp_video_stream_receiver_;
+};
+
+TEST_F(RtpVideoStreamReceiver2Test, CacheColorSpaceFromLastPacketOfKeyframe) {
+  // Test that color space is cached from the last packet of a key frame and
+  // that it's not reset by padding packets without color space.
+  constexpr int kVp9PayloadType = 99;
+  const ColorSpace kColorSpace(
+      ColorSpace::PrimaryID::kFILM, ColorSpace::TransferID::kBT2020_12,
+      ColorSpace::MatrixID::kBT2020_NCL, ColorSpace::RangeID::kFull);
+  const std::vector<uint8_t> kKeyFramePayload = {0, 1, 2, 3, 4, 5,
+                                                 6, 7, 8, 9, 10};
+  const std::vector<uint8_t> kDeltaFramePayload = {0, 1, 2, 3, 4};
+
+  // Anonymous helper class that generates received packets.
+  class {
+   public:
+    void SetPayload(const std::vector<uint8_t>& payload,
+                    VideoFrameType video_frame_type) {
+      video_frame_type_ = video_frame_type;
+      RtpPacketizer::PayloadSizeLimits pay_load_size_limits;
+      // Reduce max payload length to make sure the key frame generates two
+      // packets.
+      pay_load_size_limits.max_payload_len = 8;
+      RTPVideoHeaderVP9 rtp_video_header_vp9;
+      rtp_video_header_vp9.InitRTPVideoHeaderVP9();
+      rtp_video_header_vp9.inter_pic_predicted =
+          (video_frame_type == VideoFrameType::kVideoFrameDelta);
+      rtp_packetizer_ = std::make_unique<RtpPacketizerVp9>(
+          payload, pay_load_size_limits, rtp_video_header_vp9);
+    }
+
+    size_t NumPackets() { return rtp_packetizer_->NumPackets(); }
+    void SetColorSpace(const ColorSpace& color_space) {
+      color_space_ = color_space;
+    }
+
+    RtpPacketReceived NextPacket() {
+      RtpHeaderExtensionMap extension_map;
+      extension_map.Register<ColorSpaceExtension>(1);
+      RtpPacketToSend packet_to_send(&extension_map);
+      packet_to_send.SetSequenceNumber(sequence_number_++);
+      packet_to_send.SetSsrc(kSsrc);
+      packet_to_send.SetPayloadType(kVp9PayloadType);
+      bool include_color_space =
+          (rtp_packetizer_->NumPackets() == 1u &&
+           video_frame_type_ == VideoFrameType::kVideoFrameKey);
+      if (include_color_space) {
+        EXPECT_TRUE(
+            packet_to_send.SetExtension<ColorSpaceExtension>(color_space_));
+      }
+      rtp_packetizer_->NextPacket(&packet_to_send);
+
+      RtpPacketReceived received_packet(&extension_map);
+      received_packet.Parse(packet_to_send.data(), packet_to_send.size());
+      return received_packet;
+    }
+
+   private:
+    uint16_t sequence_number_ = 0;
+    VideoFrameType video_frame_type_;
+    ColorSpace color_space_;
+    std::unique_ptr<RtpPacketizer> rtp_packetizer_;
+  } received_packet_generator;
+  received_packet_generator.SetColorSpace(kColorSpace);
+
+  // Prepare the receiver for VP9.
+  VideoCodec codec;
+  codec.plType = kVp9PayloadType;
+  codec.codecType = kVideoCodecVP9;
+  std::map<std::string, std::string> codec_params;
+  rtp_video_stream_receiver_->AddReceiveCodec(codec, codec_params,
+                                              /*raw_payload=*/false);
+
+  // Generate key frame packets.
+  received_packet_generator.SetPayload(kKeyFramePayload,
+                                       VideoFrameType::kVideoFrameKey);
+  EXPECT_EQ(received_packet_generator.NumPackets(), 2u);
+  RtpPacketReceived key_frame_packet1 = received_packet_generator.NextPacket();
+  RtpPacketReceived key_frame_packet2 = received_packet_generator.NextPacket();
+
+  // Generate delta frame packet.
+  received_packet_generator.SetPayload(kDeltaFramePayload,
+                                       VideoFrameType::kVideoFrameDelta);
+  EXPECT_EQ(received_packet_generator.NumPackets(), 1u);
+  RtpPacketReceived delta_frame_packet = received_packet_generator.NextPacket();
+
+  rtp_video_stream_receiver_->StartReceive();
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(
+      kKeyFramePayload.data(), kKeyFramePayload.size());
+
+  // Send the key frame and expect a callback with color space information.
+  EXPECT_FALSE(key_frame_packet1.GetExtension<ColorSpaceExtension>());
+  EXPECT_TRUE(key_frame_packet2.GetExtension<ColorSpaceExtension>());
+  rtp_video_stream_receiver_->OnRtpPacket(key_frame_packet1);
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_))
+      .WillOnce(Invoke([kColorSpace](video_coding::EncodedFrame* frame) {
+        ASSERT_TRUE(frame->EncodedImage().ColorSpace());
+        EXPECT_EQ(*frame->EncodedImage().ColorSpace(), kColorSpace);
+      }));
+  rtp_video_stream_receiver_->OnRtpPacket(key_frame_packet2);
+  // Resend the first key frame packet to simulate padding for example.
+  rtp_video_stream_receiver_->OnRtpPacket(key_frame_packet1);
+
+  mock_on_complete_frame_callback_.ClearExpectedBitstream();
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(
+      kDeltaFramePayload.data(), kDeltaFramePayload.size());
+
+  // Expect delta frame to have color space set even though color space not
+  // included in the RTP packet.
+  EXPECT_FALSE(delta_frame_packet.GetExtension<ColorSpaceExtension>());
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_))
+      .WillOnce(Invoke([kColorSpace](video_coding::EncodedFrame* frame) {
+        ASSERT_TRUE(frame->EncodedImage().ColorSpace());
+        EXPECT_EQ(*frame->EncodedImage().ColorSpace(), kColorSpace);
+      }));
+  rtp_video_stream_receiver_->OnRtpPacket(delta_frame_packet);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, GenericKeyFrame) {
+  RtpPacketReceived rtp_packet;
+  rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtp_packet.SetSequenceNumber(1);
+  RTPVideoHeader video_header =
+      GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
+                                                           data.size());
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, PacketInfoIsPropagatedIntoVideoFrames) {
+  constexpr uint64_t kAbsoluteCaptureTimestamp = 12;
+  constexpr int kId0 = 1;
+
+  RtpHeaderExtensionMap extension_map;
+  extension_map.Register<AbsoluteCaptureTimeExtension>(kId0);
+  RtpPacketReceived rtp_packet(&extension_map);
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
+  rtp_packet.SetSequenceNumber(1);
+  rtp_packet.SetTimestamp(1);
+  rtp_packet.SetSsrc(kSsrc);
+  rtp_packet.SetExtension<AbsoluteCaptureTimeExtension>(
+      AbsoluteCaptureTime{kAbsoluteCaptureTimestamp,
+                          /*estimated_capture_clock_offset=*/absl::nullopt});
+
+  RTPVideoHeader video_header =
+      GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
+                                                           data.size());
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_))
+      .WillOnce(Invoke(
+          [kAbsoluteCaptureTimestamp](video_coding::EncodedFrame* frame) {
+            EXPECT_THAT(GetAbsoluteCaptureTimestamps(frame),
+                        ElementsAre(kAbsoluteCaptureTimestamp));
+          }));
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test,
+       MissingAbsoluteCaptureTimeIsFilledWithExtrapolatedValue) {
+  constexpr uint64_t kAbsoluteCaptureTimestamp = 12;
+  constexpr int kId0 = 1;
+
+  RtpHeaderExtensionMap extension_map;
+  extension_map.Register<AbsoluteCaptureTimeExtension>(kId0);
+  RtpPacketReceived rtp_packet(&extension_map);
+  rtp_packet.SetPayloadType(kPayloadType);
+
+  rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
+  uint16_t sequence_number = 1;
+  uint32_t rtp_timestamp = 1;
+  rtp_packet.SetSequenceNumber(sequence_number);
+  rtp_packet.SetTimestamp(rtp_timestamp);
+  rtp_packet.SetSsrc(kSsrc);
+  rtp_packet.SetExtension<AbsoluteCaptureTimeExtension>(
+      AbsoluteCaptureTime{kAbsoluteCaptureTimestamp,
+                          /*estimated_capture_clock_offset=*/absl::nullopt});
+
+  RTPVideoHeader video_header =
+      GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
+                                                           data.size());
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+
+  // Rtp packet without absolute capture time.
+  rtp_packet = RtpPacketReceived(&extension_map);
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtp_packet.SetSequenceNumber(++sequence_number);
+  rtp_packet.SetTimestamp(++rtp_timestamp);
+  rtp_packet.SetSsrc(kSsrc);
+
+  // There is no absolute capture time in the second packet.
+  // Expect rtp video stream receiver to extrapolate it for the resulting video
+  // frame using absolute capture time from the previous packet.
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_))
+      .WillOnce(Invoke([](video_coding::EncodedFrame* frame) {
+        EXPECT_THAT(GetAbsoluteCaptureTimestamps(frame), SizeIs(1));
+      }));
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test,
+       NoInfiniteRecursionOnEncapsulatedRedPacket) {
+  const std::vector<uint8_t> data({
+      0x80,              // RTP version.
+      kRedPayloadType,   // Payload type.
+      0, 0, 0, 0, 0, 0,  // Don't care.
+      0, 0, 0x4, 0x57,   // SSRC
+      kRedPayloadType,   // RED header.
+      0, 0, 0, 0, 0      // Don't care.
+  });
+  RtpPacketReceived packet;
+  EXPECT_TRUE(packet.Parse(data.data(), data.size()));
+  rtp_video_stream_receiver_->StartReceive();
+  rtp_video_stream_receiver_->OnRtpPacket(packet);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test,
+       DropsPacketWithRedPayloadTypeAndEmptyPayload) {
+  const uint8_t kRedPayloadType = 125;
+  config_.rtp.red_payload_type = kRedPayloadType;
+  SetUp();  // re-create rtp_video_stream_receiver with red payload type.
+  // clang-format off
+  const uint8_t data[] = {
+      0x80,              // RTP version.
+      kRedPayloadType,   // Payload type.
+      0, 0, 0, 0, 0, 0,  // Don't care.
+      0, 0, 0x4, 0x57,   // SSRC
+      // Empty rtp payload.
+  };
+  // clang-format on
+  RtpPacketReceived packet;
+  // Manually convert to CopyOnWriteBuffer to be sure capacity == size
+  // and asan bot can catch read buffer overflow.
+  EXPECT_TRUE(packet.Parse(rtc::CopyOnWriteBuffer(data)));
+  rtp_video_stream_receiver_->StartReceive();
+  rtp_video_stream_receiver_->OnRtpPacket(packet);
+  // Expect asan doesn't find anything.
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, GenericKeyFrameBitstreamError) {
+  RtpPacketReceived rtp_packet;
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
+  rtp_packet.SetSequenceNumber(1);
+  RTPVideoHeader video_header =
+      GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
+  constexpr uint8_t expected_bitsteam[] = {1, 2, 3, 0xff};
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(
+      expected_bitsteam, sizeof(expected_bitsteam));
+  EXPECT_CALL(mock_on_complete_frame_callback_,
+              DoOnCompleteFrameFailBitstream(_));
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+}
+
+class RtpVideoStreamReceiver2TestH264
+    : public RtpVideoStreamReceiver2Test,
+      public ::testing::WithParamInterface<std::string> {
+ protected:
+  RtpVideoStreamReceiver2TestH264() : RtpVideoStreamReceiver2Test(GetParam()) {}
+};
+
+INSTANTIATE_TEST_SUITE_P(SpsPpsIdrIsKeyframe,
+                         RtpVideoStreamReceiver2TestH264,
+                         Values("", "WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/"));
+
+// Fails on MSAN: https://bugs.chromium.org/p/webrtc/issues/detail?id=11376.
+#if defined(MEMORY_SANITIZER)
+#define MAYBE_InBandSpsPps DISABLED_InBandSpsPps
+#else
+#define MAYBE_InBandSpsPps InBandSpsPps
+#endif
+TEST_P(RtpVideoStreamReceiver2TestH264, MAYBE_InBandSpsPps) {
+  rtc::CopyOnWriteBuffer sps_data;
+  RtpPacketReceived rtp_packet;
+  RTPVideoHeader sps_video_header = GetDefaultH264VideoHeader();
+  AddSps(&sps_video_header, 0, &sps_data);
+  rtp_packet.SetSequenceNumber(0);
+  rtp_packet.SetPayloadType(kPayloadType);
+  sps_video_header.is_first_packet_in_frame = true;
+  sps_video_header.frame_type = VideoFrameType::kEmptyFrame;
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(
+      kH264StartCode, sizeof(kH264StartCode));
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(sps_data.data(),
+                                                           sps_data.size());
+  rtp_video_stream_receiver_->OnReceivedPayloadData(sps_data, rtp_packet,
+                                                    sps_video_header);
+
+  rtc::CopyOnWriteBuffer pps_data;
+  RTPVideoHeader pps_video_header = GetDefaultH264VideoHeader();
+  AddPps(&pps_video_header, 0, 1, &pps_data);
+  rtp_packet.SetSequenceNumber(1);
+  pps_video_header.is_first_packet_in_frame = true;
+  pps_video_header.frame_type = VideoFrameType::kEmptyFrame;
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(
+      kH264StartCode, sizeof(kH264StartCode));
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(pps_data.data(),
+                                                           pps_data.size());
+  rtp_video_stream_receiver_->OnReceivedPayloadData(pps_data, rtp_packet,
+                                                    pps_video_header);
+
+  rtc::CopyOnWriteBuffer idr_data;
+  RTPVideoHeader idr_video_header = GetDefaultH264VideoHeader();
+  AddIdr(&idr_video_header, 1);
+  rtp_packet.SetSequenceNumber(2);
+  idr_video_header.is_first_packet_in_frame = true;
+  idr_video_header.is_last_packet_in_frame = true;
+  idr_video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  const uint8_t idr[] = {0x65, 1, 2, 3};
+  idr_data.AppendData(idr);
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(
+      kH264StartCode, sizeof(kH264StartCode));
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(idr_data.data(),
+                                                           idr_data.size());
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
+  rtp_video_stream_receiver_->OnReceivedPayloadData(idr_data, rtp_packet,
+                                                    idr_video_header);
+}
+
+TEST_P(RtpVideoStreamReceiver2TestH264, OutOfBandFmtpSpsPps) {
+  constexpr int kPayloadType = 99;
+  VideoCodec codec;
+  codec.plType = kPayloadType;
+  std::map<std::string, std::string> codec_params;
+  // Example parameter sets from https://tools.ietf.org/html/rfc3984#section-8.2
+  // .
+  codec_params.insert(
+      {cricket::kH264FmtpSpropParameterSets, "Z0IACpZTBYmI,aMljiA=="});
+  rtp_video_stream_receiver_->AddReceiveCodec(codec, codec_params,
+                                              /*raw_payload=*/false);
+  const uint8_t binary_sps[] = {0x67, 0x42, 0x00, 0x0a, 0x96,
+                                0x53, 0x05, 0x89, 0x88};
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(
+      kH264StartCode, sizeof(kH264StartCode));
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(binary_sps,
+                                                           sizeof(binary_sps));
+  const uint8_t binary_pps[] = {0x68, 0xc9, 0x63, 0x88};
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(
+      kH264StartCode, sizeof(kH264StartCode));
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(binary_pps,
+                                                           sizeof(binary_pps));
+
+  RtpPacketReceived rtp_packet;
+  RTPVideoHeader video_header = GetDefaultH264VideoHeader();
+  AddIdr(&video_header, 0);
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtp_packet.SetSequenceNumber(2);
+  video_header.is_first_packet_in_frame = true;
+  video_header.is_last_packet_in_frame = true;
+  video_header.codec = kVideoCodecH264;
+  video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  rtc::CopyOnWriteBuffer data({1, 2, 3});
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(
+      kH264StartCode, sizeof(kH264StartCode));
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
+                                                           data.size());
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, PaddingInMediaStream) {
+  RtpPacketReceived rtp_packet;
+  RTPVideoHeader video_header = GetDefaultH264VideoHeader();
+  rtc::CopyOnWriteBuffer data({1, 2, 3});
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtp_packet.SetSequenceNumber(2);
+  video_header.is_first_packet_in_frame = true;
+  video_header.is_last_packet_in_frame = true;
+  video_header.codec = kVideoCodecGeneric;
+  video_header.frame_type = VideoFrameType::kVideoFrameKey;
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
+                                                           data.size());
+
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+
+  rtp_packet.SetSequenceNumber(3);
+  rtp_video_stream_receiver_->OnReceivedPayloadData({}, rtp_packet,
+                                                    video_header);
+
+  rtp_packet.SetSequenceNumber(4);
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
+  video_header.frame_type = VideoFrameType::kVideoFrameDelta;
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+
+  rtp_packet.SetSequenceNumber(6);
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
+  rtp_packet.SetSequenceNumber(5);
+  rtp_video_stream_receiver_->OnReceivedPayloadData({}, rtp_packet,
+                                                    video_header);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, RequestKeyframeIfFirstFrameIsDelta) {
+  RtpPacketReceived rtp_packet;
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
+  rtp_packet.SetSequenceNumber(1);
+  RTPVideoHeader video_header =
+      GetGenericVideoHeader(VideoFrameType::kVideoFrameDelta);
+  EXPECT_CALL(mock_key_frame_request_sender_, RequestKeyFrame());
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, RequestKeyframeWhenPacketBufferGetsFull) {
+  constexpr int kPacketBufferMaxSize = 2048;
+
+  RtpPacketReceived rtp_packet;
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
+  RTPVideoHeader video_header =
+      GetGenericVideoHeader(VideoFrameType::kVideoFrameDelta);
+  // Incomplete frames so that the packet buffer is filling up.
+  video_header.is_last_packet_in_frame = false;
+  uint16_t start_sequence_number = 1234;
+  rtp_packet.SetSequenceNumber(start_sequence_number);
+  while (rtp_packet.SequenceNumber() - start_sequence_number <
+         kPacketBufferMaxSize) {
+    rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                      video_header);
+    rtp_packet.SetSequenceNumber(rtp_packet.SequenceNumber() + 2);
+  }
+
+  EXPECT_CALL(mock_key_frame_request_sender_, RequestKeyFrame());
+  rtp_video_stream_receiver_->OnReceivedPayloadData(data, rtp_packet,
+                                                    video_header);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, SecondarySinksGetRtpNotifications) {
+  rtp_video_stream_receiver_->StartReceive();
+
+  MockRtpPacketSink secondary_sink_1;
+  MockRtpPacketSink secondary_sink_2;
+
+  rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink_1);
+  rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink_2);
+
+  auto rtp_packet = CreateRtpPacketReceived();
+  EXPECT_CALL(secondary_sink_1, OnRtpPacket(SamePacketAs(*rtp_packet)));
+  EXPECT_CALL(secondary_sink_2, OnRtpPacket(SamePacketAs(*rtp_packet)));
+
+  rtp_video_stream_receiver_->OnRtpPacket(*rtp_packet);
+
+  // Test tear-down.
+  rtp_video_stream_receiver_->StopReceive();
+  rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink_1);
+  rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink_2);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test,
+       RemovedSecondarySinksGetNoRtpNotifications) {
+  rtp_video_stream_receiver_->StartReceive();
+
+  MockRtpPacketSink secondary_sink;
+
+  rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
+  rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink);
+
+  auto rtp_packet = CreateRtpPacketReceived();
+
+  EXPECT_CALL(secondary_sink, OnRtpPacket(_)).Times(0);
+
+  rtp_video_stream_receiver_->OnRtpPacket(*rtp_packet);
+
+  // Test tear-down.
+  rtp_video_stream_receiver_->StopReceive();
+}
+
+TEST_F(RtpVideoStreamReceiver2Test,
+       OnlyRemovedSecondarySinksExcludedFromNotifications) {
+  rtp_video_stream_receiver_->StartReceive();
+
+  MockRtpPacketSink kept_secondary_sink;
+  MockRtpPacketSink removed_secondary_sink;
+
+  rtp_video_stream_receiver_->AddSecondarySink(&kept_secondary_sink);
+  rtp_video_stream_receiver_->AddSecondarySink(&removed_secondary_sink);
+  rtp_video_stream_receiver_->RemoveSecondarySink(&removed_secondary_sink);
+
+  auto rtp_packet = CreateRtpPacketReceived();
+  EXPECT_CALL(kept_secondary_sink, OnRtpPacket(SamePacketAs(*rtp_packet)));
+
+  rtp_video_stream_receiver_->OnRtpPacket(*rtp_packet);
+
+  // Test tear-down.
+  rtp_video_stream_receiver_->StopReceive();
+  rtp_video_stream_receiver_->RemoveSecondarySink(&kept_secondary_sink);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test,
+       SecondariesOfNonStartedStreamGetNoNotifications) {
+  // Explicitly showing that the stream is not in the |started| state,
+  // regardless of whether streams start out |started| or |stopped|.
+  rtp_video_stream_receiver_->StopReceive();
+
+  MockRtpPacketSink secondary_sink;
+  rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
+
+  auto rtp_packet = CreateRtpPacketReceived();
+  EXPECT_CALL(secondary_sink, OnRtpPacket(_)).Times(0);
+
+  rtp_video_stream_receiver_->OnRtpPacket(*rtp_packet);
+
+  // Test tear-down.
+  rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, ParseGenericDescriptorOnePacket) {
+  const std::vector<uint8_t> data = {0, 1, 2, 3, 4};
+  const int kSpatialIndex = 1;
+
+  rtp_video_stream_receiver_->StartReceive();
+
+  RtpHeaderExtensionMap extension_map;
+  extension_map.Register<RtpGenericFrameDescriptorExtension00>(5);
+  RtpPacketReceived rtp_packet(&extension_map);
+  rtp_packet.SetPayloadType(kPayloadType);
+
+  RtpGenericFrameDescriptor generic_descriptor;
+  generic_descriptor.SetFirstPacketInSubFrame(true);
+  generic_descriptor.SetLastPacketInSubFrame(true);
+  generic_descriptor.SetFrameId(100);
+  generic_descriptor.SetSpatialLayersBitmask(1 << kSpatialIndex);
+  generic_descriptor.AddFrameDependencyDiff(90);
+  generic_descriptor.AddFrameDependencyDiff(80);
+  ASSERT_TRUE(rtp_packet.SetExtension<RtpGenericFrameDescriptorExtension00>(
+      generic_descriptor));
+
+  uint8_t* payload = rtp_packet.SetPayloadSize(data.size());
+  memcpy(payload, data.data(), data.size());
+  // The first byte is the header, so we ignore the first byte of |data|.
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data() + 1,
+                                                           data.size() - 1);
+
+  rtp_packet.SetMarker(true);
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtp_packet.SetSequenceNumber(1);
+
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
+      .WillOnce(Invoke([kSpatialIndex](video_coding::EncodedFrame* frame) {
+        EXPECT_EQ(frame->num_references, 2U);
+        EXPECT_EQ(frame->references[0], frame->id.picture_id - 90);
+        EXPECT_EQ(frame->references[1], frame->id.picture_id - 80);
+        EXPECT_EQ(frame->id.spatial_layer, kSpatialIndex);
+        EXPECT_THAT(frame->PacketInfos(), SizeIs(1));
+      }));
+
+  rtp_video_stream_receiver_->OnRtpPacket(rtp_packet);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, ParseGenericDescriptorTwoPackets) {
+  const std::vector<uint8_t> data = {0, 1, 2, 3, 4};
+  const int kSpatialIndex = 1;
+
+  rtp_video_stream_receiver_->StartReceive();
+
+  RtpHeaderExtensionMap extension_map;
+  extension_map.Register<RtpGenericFrameDescriptorExtension00>(5);
+  RtpPacketReceived first_packet(&extension_map);
+
+  RtpGenericFrameDescriptor first_packet_descriptor;
+  first_packet_descriptor.SetFirstPacketInSubFrame(true);
+  first_packet_descriptor.SetLastPacketInSubFrame(false);
+  first_packet_descriptor.SetFrameId(100);
+  first_packet_descriptor.SetSpatialLayersBitmask(1 << kSpatialIndex);
+  first_packet_descriptor.SetResolution(480, 360);
+  ASSERT_TRUE(first_packet.SetExtension<RtpGenericFrameDescriptorExtension00>(
+      first_packet_descriptor));
+
+  uint8_t* first_packet_payload = first_packet.SetPayloadSize(data.size());
+  memcpy(first_packet_payload, data.data(), data.size());
+  // The first byte is the header, so we ignore the first byte of |data|.
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data() + 1,
+                                                           data.size() - 1);
+
+  first_packet.SetPayloadType(kPayloadType);
+  first_packet.SetSequenceNumber(1);
+  rtp_video_stream_receiver_->OnRtpPacket(first_packet);
+
+  RtpPacketReceived second_packet(&extension_map);
+  RtpGenericFrameDescriptor second_packet_descriptor;
+  second_packet_descriptor.SetFirstPacketInSubFrame(false);
+  second_packet_descriptor.SetLastPacketInSubFrame(true);
+  ASSERT_TRUE(second_packet.SetExtension<RtpGenericFrameDescriptorExtension00>(
+      second_packet_descriptor));
+
+  second_packet.SetMarker(true);
+  second_packet.SetPayloadType(kPayloadType);
+  second_packet.SetSequenceNumber(2);
+
+  uint8_t* second_packet_payload = second_packet.SetPayloadSize(data.size());
+  memcpy(second_packet_payload, data.data(), data.size());
+  // The first byte is the header, so we ignore the first byte of |data|.
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data() + 1,
+                                                           data.size() - 1);
+
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
+      .WillOnce(Invoke([kSpatialIndex](video_coding::EncodedFrame* frame) {
+        EXPECT_EQ(frame->num_references, 0U);
+        EXPECT_EQ(frame->id.spatial_layer, kSpatialIndex);
+        EXPECT_EQ(frame->EncodedImage()._encodedWidth, 480u);
+        EXPECT_EQ(frame->EncodedImage()._encodedHeight, 360u);
+        EXPECT_THAT(frame->PacketInfos(), SizeIs(2));
+      }));
+
+  rtp_video_stream_receiver_->OnRtpPacket(second_packet);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, ParseGenericDescriptorRawPayload) {
+  const std::vector<uint8_t> data = {0, 1, 2, 3, 4};
+  const int kRawPayloadType = 123;
+
+  VideoCodec codec;
+  codec.plType = kRawPayloadType;
+  rtp_video_stream_receiver_->AddReceiveCodec(codec, {}, /*raw_payload=*/true);
+  rtp_video_stream_receiver_->StartReceive();
+
+  RtpHeaderExtensionMap extension_map;
+  extension_map.Register<RtpGenericFrameDescriptorExtension00>(5);
+  RtpPacketReceived rtp_packet(&extension_map);
+
+  RtpGenericFrameDescriptor generic_descriptor;
+  generic_descriptor.SetFirstPacketInSubFrame(true);
+  generic_descriptor.SetLastPacketInSubFrame(true);
+  ASSERT_TRUE(rtp_packet.SetExtension<RtpGenericFrameDescriptorExtension00>(
+      generic_descriptor));
+
+  uint8_t* payload = rtp_packet.SetPayloadSize(data.size());
+  memcpy(payload, data.data(), data.size());
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
+                                                           data.size());
+
+  rtp_packet.SetMarker(true);
+  rtp_packet.SetPayloadType(kRawPayloadType);
+  rtp_packet.SetSequenceNumber(1);
+
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame);
+  rtp_video_stream_receiver_->OnRtpPacket(rtp_packet);
+}
+
+TEST_F(RtpVideoStreamReceiver2Test, UnwrapsFrameId) {
+  const std::vector<uint8_t> data = {0, 1, 2, 3, 4};
+  const int kPayloadType = 123;
+
+  VideoCodec codec;
+  codec.plType = kPayloadType;
+  rtp_video_stream_receiver_->AddReceiveCodec(codec, {}, /*raw_payload=*/true);
+  rtp_video_stream_receiver_->StartReceive();
+  RtpHeaderExtensionMap extension_map;
+  extension_map.Register<RtpGenericFrameDescriptorExtension00>(5);
+
+  uint16_t rtp_sequence_number = 1;
+  auto inject_packet = [&](uint16_t wrapped_frame_id) {
+    RtpPacketReceived rtp_packet(&extension_map);
+
+    RtpGenericFrameDescriptor generic_descriptor;
+    generic_descriptor.SetFirstPacketInSubFrame(true);
+    generic_descriptor.SetLastPacketInSubFrame(true);
+    generic_descriptor.SetFrameId(wrapped_frame_id);
+    ASSERT_TRUE(rtp_packet.SetExtension<RtpGenericFrameDescriptorExtension00>(
+        generic_descriptor));
+
+    uint8_t* payload = rtp_packet.SetPayloadSize(data.size());
+    ASSERT_TRUE(payload);
+    memcpy(payload, data.data(), data.size());
+    mock_on_complete_frame_callback_.ClearExpectedBitstream();
+    mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
+                                                             data.size());
+    rtp_packet.SetMarker(true);
+    rtp_packet.SetPayloadType(kPayloadType);
+    rtp_packet.SetSequenceNumber(++rtp_sequence_number);
+    rtp_video_stream_receiver_->OnRtpPacket(rtp_packet);
+  };
+
+  int64_t first_picture_id;
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
+      .WillOnce([&](video_coding::EncodedFrame* frame) {
+        first_picture_id = frame->id.picture_id;
+      });
+  inject_packet(/*wrapped_frame_id=*/0xffff);
+
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
+      .WillOnce([&](video_coding::EncodedFrame* frame) {
+        EXPECT_EQ(frame->id.picture_id - first_picture_id, 3);
+      });
+  inject_packet(/*wrapped_frame_id=*/0x0002);
+}
+
+class RtpVideoStreamReceiver2DependencyDescriptorTest
+    : public RtpVideoStreamReceiver2Test {
+ public:
+  RtpVideoStreamReceiver2DependencyDescriptorTest() {
+    VideoCodec codec;
+    codec.plType = payload_type_;
+    rtp_video_stream_receiver_->AddReceiveCodec(codec, {},
+                                                /*raw_payload=*/true);
+    extension_map_.Register<RtpDependencyDescriptorExtension>(7);
+    rtp_video_stream_receiver_->StartReceive();
+  }
+
+  // Returns some valid structure for the DependencyDescriptors.
+  // First template of that structure always fit for a key frame.
+  static FrameDependencyStructure CreateStreamStructure() {
+    FrameDependencyStructure stream_structure;
+    stream_structure.num_decode_targets = 1;
+    stream_structure.templates = {
+        GenericFrameInfo::Builder().Dtis("S").Build(),
+        GenericFrameInfo::Builder().Dtis("S").Fdiffs({1}).Build(),
+    };
+    return stream_structure;
+  }
+
+  void InjectPacketWith(const FrameDependencyStructure& stream_structure,
+                        const DependencyDescriptor& dependency_descriptor) {
+    const std::vector<uint8_t> data = {0, 1, 2, 3, 4};
+    RtpPacketReceived rtp_packet(&extension_map_);
+    ASSERT_TRUE(rtp_packet.SetExtension<RtpDependencyDescriptorExtension>(
+        stream_structure, dependency_descriptor));
+    uint8_t* payload = rtp_packet.SetPayloadSize(data.size());
+    ASSERT_TRUE(payload);
+    memcpy(payload, data.data(), data.size());
+    mock_on_complete_frame_callback_.ClearExpectedBitstream();
+    mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
+                                                             data.size());
+    rtp_packet.SetMarker(true);
+    rtp_packet.SetPayloadType(payload_type_);
+    rtp_packet.SetSequenceNumber(++rtp_sequence_number_);
+    rtp_video_stream_receiver_->OnRtpPacket(rtp_packet);
+  }
+
+ private:
+  const int payload_type_ = 123;
+  RtpHeaderExtensionMap extension_map_;
+  uint16_t rtp_sequence_number_ = 321;
+};
+
+TEST_F(RtpVideoStreamReceiver2DependencyDescriptorTest, UnwrapsFrameId) {
+  FrameDependencyStructure stream_structure = CreateStreamStructure();
+
+  DependencyDescriptor keyframe_descriptor;
+  keyframe_descriptor.attached_structure =
+      std::make_unique<FrameDependencyStructure>(stream_structure);
+  keyframe_descriptor.frame_dependencies = stream_structure.templates[0];
+  keyframe_descriptor.frame_number = 0xfff0;
+  // DependencyDescriptor doesn't support reordering delta frame before
+  // keyframe. Thus feed a key frame first, then test reodered delta frames.
+  int64_t first_picture_id;
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
+      .WillOnce([&](video_coding::EncodedFrame* frame) {
+        first_picture_id = frame->id.picture_id;
+      });
+  InjectPacketWith(stream_structure, keyframe_descriptor);
+
+  DependencyDescriptor deltaframe1_descriptor;
+  deltaframe1_descriptor.frame_dependencies = stream_structure.templates[1];
+  deltaframe1_descriptor.frame_number = 0xfffe;
+
+  DependencyDescriptor deltaframe2_descriptor;
+  deltaframe1_descriptor.frame_dependencies = stream_structure.templates[1];
+  deltaframe2_descriptor.frame_number = 0x0002;
+
+  // Parser should unwrap frame ids correctly even if packets were reordered by
+  // the network.
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
+      .WillOnce([&](video_coding::EncodedFrame* frame) {
+        // 0x0002 - 0xfff0
+        EXPECT_EQ(frame->id.picture_id - first_picture_id, 18);
+      })
+      .WillOnce([&](video_coding::EncodedFrame* frame) {
+        // 0xfffe - 0xfff0
+        EXPECT_EQ(frame->id.picture_id - first_picture_id, 14);
+      });
+  InjectPacketWith(stream_structure, deltaframe2_descriptor);
+  InjectPacketWith(stream_structure, deltaframe1_descriptor);
+}
+
+TEST_F(RtpVideoStreamReceiver2DependencyDescriptorTest,
+       DropsLateDeltaFramePacketWithDependencyDescriptorExtension) {
+  FrameDependencyStructure stream_structure1 = CreateStreamStructure();
+  FrameDependencyStructure stream_structure2 = CreateStreamStructure();
+  // Make sure template ids for these two structures do not collide:
+  // adjust structure_id (that is also used as template id offset).
+  stream_structure1.structure_id = 13;
+  stream_structure2.structure_id =
+      stream_structure1.structure_id + stream_structure1.templates.size();
+
+  DependencyDescriptor keyframe1_descriptor;
+  keyframe1_descriptor.attached_structure =
+      std::make_unique<FrameDependencyStructure>(stream_structure1);
+  keyframe1_descriptor.frame_dependencies = stream_structure1.templates[0];
+  keyframe1_descriptor.frame_number = 1;
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame);
+  InjectPacketWith(stream_structure1, keyframe1_descriptor);
+
+  // Pass in 2nd key frame with different structure.
+  DependencyDescriptor keyframe2_descriptor;
+  keyframe2_descriptor.attached_structure =
+      std::make_unique<FrameDependencyStructure>(stream_structure2);
+  keyframe2_descriptor.frame_dependencies = stream_structure2.templates[0];
+  keyframe2_descriptor.frame_number = 3;
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame);
+  InjectPacketWith(stream_structure2, keyframe2_descriptor);
+
+  // Pass in late delta frame that uses structure of the 1st key frame.
+  DependencyDescriptor deltaframe_descriptor;
+  deltaframe_descriptor.frame_dependencies = stream_structure1.templates[0];
+  deltaframe_descriptor.frame_number = 2;
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame).Times(0);
+  InjectPacketWith(stream_structure1, deltaframe_descriptor);
+}
+
+TEST_F(RtpVideoStreamReceiver2DependencyDescriptorTest,
+       DropsLateKeyFramePacketWithDependencyDescriptorExtension) {
+  FrameDependencyStructure stream_structure1 = CreateStreamStructure();
+  FrameDependencyStructure stream_structure2 = CreateStreamStructure();
+  // Make sure template ids for these two structures do not collide:
+  // adjust structure_id (that is also used as template id offset).
+  stream_structure1.structure_id = 13;
+  stream_structure2.structure_id =
+      stream_structure1.structure_id + stream_structure1.templates.size();
+
+  DependencyDescriptor keyframe1_descriptor;
+  keyframe1_descriptor.attached_structure =
+      std::make_unique<FrameDependencyStructure>(stream_structure1);
+  keyframe1_descriptor.frame_dependencies = stream_structure1.templates[0];
+  keyframe1_descriptor.frame_number = 1;
+
+  DependencyDescriptor keyframe2_descriptor;
+  keyframe2_descriptor.attached_structure =
+      std::make_unique<FrameDependencyStructure>(stream_structure2);
+  keyframe2_descriptor.frame_dependencies = stream_structure2.templates[0];
+  keyframe2_descriptor.frame_number = 3;
+
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
+      .WillOnce([&](video_coding::EncodedFrame* frame) {
+        EXPECT_EQ(frame->id.picture_id & 0xFFFF, 3);
+      });
+  InjectPacketWith(stream_structure2, keyframe2_descriptor);
+  InjectPacketWith(stream_structure1, keyframe1_descriptor);
+
+  // Pass in delta frame that uses structure of the 2nd key frame. Late key
+  // frame shouldn't block it.
+  DependencyDescriptor deltaframe_descriptor;
+  deltaframe_descriptor.frame_dependencies = stream_structure2.templates[0];
+  deltaframe_descriptor.frame_number = 4;
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame)
+      .WillOnce([&](video_coding::EncodedFrame* frame) {
+        EXPECT_EQ(frame->id.picture_id & 0xFFFF, 4);
+      });
+  InjectPacketWith(stream_structure2, deltaframe_descriptor);
+}
+
+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
+TEST_F(RtpVideoStreamReceiver2Test, RepeatedSecondarySinkDisallowed) {
+  MockRtpPacketSink secondary_sink;
+
+  rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink);
+  EXPECT_DEATH(rtp_video_stream_receiver_->AddSecondarySink(&secondary_sink),
+               "");
+
+  // Test tear-down.
+  rtp_video_stream_receiver_->RemoveSecondarySink(&secondary_sink);
+}
+#endif
+
+TEST_F(RtpVideoStreamReceiver2Test, TransformFrame) {
+  rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
+      new rtc::RefCountedObject<testing::NiceMock<MockFrameTransformer>>();
+  EXPECT_CALL(*mock_frame_transformer,
+              RegisterTransformedFrameSinkCallback(_, config_.rtp.remote_ssrc));
+  auto receiver = std::make_unique<RtpVideoStreamReceiver2>(
+      Clock::GetRealTimeClock(), &mock_transport_, nullptr, nullptr, &config_,
+      rtp_receive_statistics_.get(), nullptr, nullptr, process_thread_.get(),
+      &mock_nack_sender_, nullptr, &mock_on_complete_frame_callback_, nullptr,
+      mock_frame_transformer);
+  VideoCodec video_codec;
+  video_codec.plType = kPayloadType;
+  video_codec.codecType = kVideoCodecGeneric;
+  receiver->AddReceiveCodec(video_codec, {}, /*raw_payload=*/false);
+
+  RtpPacketReceived rtp_packet;
+  rtp_packet.SetPayloadType(kPayloadType);
+  rtc::CopyOnWriteBuffer data({1, 2, 3, 4});
+  rtp_packet.SetSequenceNumber(1);
+  RTPVideoHeader video_header =
+      GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
+                                                           data.size());
+  EXPECT_CALL(*mock_frame_transformer, Transform(_));
+  receiver->OnReceivedPayloadData(data, rtp_packet, video_header);
+
+  EXPECT_CALL(*mock_frame_transformer,
+              UnregisterTransformedFrameSinkCallback(config_.rtp.remote_ssrc));
+  receiver = nullptr;
+}
+
+// Test default behavior and when playout delay is overridden by field trial.
+const PlayoutDelay kTransmittedPlayoutDelay = {100, 200};
+const PlayoutDelay kForcedPlayoutDelay = {70, 90};
+struct PlayoutDelayOptions {
+  std::string field_trial;
+  PlayoutDelay expected_delay;
+};
+const PlayoutDelayOptions kDefaultBehavior = {
+    /*field_trial=*/"", /*expected_delay=*/kTransmittedPlayoutDelay};
+const PlayoutDelayOptions kOverridePlayoutDelay = {
+    /*field_trial=*/"WebRTC-ForcePlayoutDelay/min_ms:70,max_ms:90/",
+    /*expected_delay=*/kForcedPlayoutDelay};
+
+class RtpVideoStreamReceiver2TestPlayoutDelay
+    : public RtpVideoStreamReceiver2Test,
+      public ::testing::WithParamInterface<PlayoutDelayOptions> {
+ protected:
+  RtpVideoStreamReceiver2TestPlayoutDelay()
+      : RtpVideoStreamReceiver2Test(GetParam().field_trial) {}
+};
+
+INSTANTIATE_TEST_SUITE_P(PlayoutDelay,
+                         RtpVideoStreamReceiver2TestPlayoutDelay,
+                         Values(kDefaultBehavior, kOverridePlayoutDelay));
+
+TEST_P(RtpVideoStreamReceiver2TestPlayoutDelay, PlayoutDelay) {
+  rtc::CopyOnWriteBuffer payload_data({1, 2, 3, 4});
+  RtpHeaderExtensionMap extension_map;
+  extension_map.Register<PlayoutDelayLimits>(1);
+  RtpPacketToSend packet_to_send(&extension_map);
+  packet_to_send.SetPayloadType(kPayloadType);
+  packet_to_send.SetSequenceNumber(1);
+
+  // Set playout delay on outgoing packet.
+  EXPECT_TRUE(packet_to_send.SetExtension<PlayoutDelayLimits>(
+      kTransmittedPlayoutDelay));
+  uint8_t* payload = packet_to_send.AllocatePayload(payload_data.size());
+  memcpy(payload, payload_data.data(), payload_data.size());
+
+  RtpPacketReceived received_packet(&extension_map);
+  received_packet.Parse(packet_to_send.data(), packet_to_send.size());
+
+  RTPVideoHeader video_header =
+      GetGenericVideoHeader(VideoFrameType::kVideoFrameKey);
+  mock_on_complete_frame_callback_.AppendExpectedBitstream(payload_data.data(),
+                                                           payload_data.size());
+  // Expect the playout delay of encoded frame to be the same as the transmitted
+  // playout delay unless it was overridden by a field trial.
+  EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_))
+      .WillOnce(Invoke([expected_playout_delay = GetParam().expected_delay](
+                           video_coding::EncodedFrame* frame) {
+        EXPECT_EQ(frame->EncodedImage().playout_delay_, expected_playout_delay);
+      }));
+  rtp_video_stream_receiver_->OnReceivedPayloadData(
+      received_packet.PayloadBuffer(), received_packet, video_header);
+}
+
+}  // namespace webrtc
diff --git a/video/rtp_video_stream_receiver_frame_transformer_delegate.cc b/video/rtp_video_stream_receiver_frame_transformer_delegate.cc
index 753dbca..31eb344 100644
--- a/video/rtp_video_stream_receiver_frame_transformer_delegate.cc
+++ b/video/rtp_video_stream_receiver_frame_transformer_delegate.cc
@@ -17,7 +17,6 @@
 #include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
 #include "rtc_base/task_utils/to_queued_task.h"
 #include "rtc_base/thread.h"
-#include "video/rtp_video_stream_receiver.h"
 
 namespace webrtc {
 
@@ -69,7 +68,7 @@
 
 RtpVideoStreamReceiverFrameTransformerDelegate::
     RtpVideoStreamReceiverFrameTransformerDelegate(
-        RtpVideoStreamReceiver* receiver,
+        RtpVideoFrameReceiver* receiver,
         rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
         rtc::Thread* network_thread,
         uint32_t ssrc)
diff --git a/video/rtp_video_stream_receiver_frame_transformer_delegate.h b/video/rtp_video_stream_receiver_frame_transformer_delegate.h
index 0a106c9..e687e7f 100644
--- a/video/rtp_video_stream_receiver_frame_transformer_delegate.h
+++ b/video/rtp_video_stream_receiver_frame_transformer_delegate.h
@@ -20,7 +20,16 @@
 
 namespace webrtc {
 
-class RtpVideoStreamReceiver;
+// Called back by RtpVideoStreamReceiverFrameTransformerDelegate on the network
+// thread after transformation.
+class RtpVideoFrameReceiver {
+ public:
+  virtual void ManageFrame(
+      std::unique_ptr<video_coding::RtpFrameObject> frame) = 0;
+
+ protected:
+  virtual ~RtpVideoFrameReceiver() = default;
+};
 
 // Delegates calls to FrameTransformerInterface to transform frames, and to
 // RtpVideoStreamReceiver to manage transformed frames on the |network_thread_|.
@@ -28,7 +37,7 @@
     : public TransformedFrameCallback {
  public:
   RtpVideoStreamReceiverFrameTransformerDelegate(
-      RtpVideoStreamReceiver* receiver,
+      RtpVideoFrameReceiver* receiver,
       rtc::scoped_refptr<FrameTransformerInterface> frame_transformer,
       rtc::Thread* network_thread,
       uint32_t ssrc);
@@ -44,7 +53,7 @@
   void OnTransformedFrame(
       std::unique_ptr<TransformableFrameInterface> frame) override;
 
-  // Delegates the call to RtpVideoReceiver::ManageFrame on the
+  // Delegates the call to RtpVideoFrameReceiver::ManageFrame on the
   // |network_thread_|.
   void ManageFrame(std::unique_ptr<TransformableFrameInterface> frame);
 
@@ -53,7 +62,7 @@
 
  private:
   SequenceChecker network_sequence_checker_;
-  RtpVideoStreamReceiver* receiver_ RTC_GUARDED_BY(network_sequence_checker_);
+  RtpVideoFrameReceiver* receiver_ RTC_GUARDED_BY(network_sequence_checker_);
   rtc::scoped_refptr<FrameTransformerInterface> frame_transformer_
       RTC_GUARDED_BY(network_sequence_checker_);
   rtc::Thread* const network_thread_;
diff --git a/video/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc b/video/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc
index b427137..a411ca6 100644
--- a/video/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc
+++ b/video/rtp_video_stream_receiver_frame_transformer_delegate_unittest.cc
@@ -21,11 +21,11 @@
 #include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
 #include "modules/utility/include/process_thread.h"
 #include "rtc_base/event.h"
+#include "rtc_base/ref_counted_object.h"
 #include "rtc_base/task_utils/to_queued_task.h"
 #include "test/gmock.h"
 #include "test/gtest.h"
 #include "test/mock_frame_transformer.h"
-#include "video/rtp_video_stream_receiver.h"
 
 namespace webrtc {
 namespace {
@@ -47,70 +47,10 @@
   return CreateRtpFrameObject(RTPVideoHeader());
 }
 
-class FakeTransport : public Transport {
+class TestRtpVideoFrameReceiver : public RtpVideoFrameReceiver {
  public:
-  bool SendRtp(const uint8_t* packet,
-               size_t length,
-               const PacketOptions& options) {
-    return true;
-  }
-  bool SendRtcp(const uint8_t* packet, size_t length) { return true; }
-};
-
-class FakeNackSender : public NackSender {
- public:
-  void SendNack(const std::vector<uint16_t>& sequence_numbers) {}
-  void SendNack(const std::vector<uint16_t>& sequence_numbers,
-                bool buffering_allowed) {}
-};
-
-class FakeOnCompleteFrameCallback
-    : public video_coding::OnCompleteFrameCallback {
- public:
-  void OnCompleteFrame(
-      std::unique_ptr<video_coding::EncodedFrame> frame) override {}
-};
-
-class TestRtpVideoStreamReceiverInitializer {
- public:
-  TestRtpVideoStreamReceiverInitializer()
-      : test_config_(nullptr),
-        test_process_thread_(ProcessThread::Create("TestThread")) {
-    test_config_.rtp.remote_ssrc = 1111;
-    test_config_.rtp.local_ssrc = 2222;
-    test_rtp_receive_statistics_ =
-        ReceiveStatistics::Create(Clock::GetRealTimeClock());
-  }
-
- protected:
-  VideoReceiveStream::Config test_config_;
-  FakeTransport fake_transport_;
-  FakeNackSender fake_nack_sender_;
-  FakeOnCompleteFrameCallback fake_on_complete_frame_callback_;
-  std::unique_ptr<ProcessThread> test_process_thread_;
-  std::unique_ptr<ReceiveStatistics> test_rtp_receive_statistics_;
-};
-
-class TestRtpVideoStreamReceiver : public TestRtpVideoStreamReceiverInitializer,
-                                   public RtpVideoStreamReceiver {
- public:
-  TestRtpVideoStreamReceiver()
-      : TestRtpVideoStreamReceiverInitializer(),
-        RtpVideoStreamReceiver(Clock::GetRealTimeClock(),
-                               &fake_transport_,
-                               nullptr,
-                               nullptr,
-                               &test_config_,
-                               test_rtp_receive_statistics_.get(),
-                               nullptr,
-                               nullptr,
-                               test_process_thread_.get(),
-                               &fake_nack_sender_,
-                               nullptr,
-                               &fake_on_complete_frame_callback_,
-                               nullptr,
-                               nullptr) {}
-  ~TestRtpVideoStreamReceiver() override = default;
+  TestRtpVideoFrameReceiver() {}
+  ~TestRtpVideoFrameReceiver() override = default;
 
   MOCK_METHOD(void,
               ManageFrame,
@@ -120,7 +60,7 @@
 
 TEST(RtpVideoStreamReceiverFrameTransformerDelegateTest,
      RegisterTransformedFrameCallbackSinkOnInit) {
-  TestRtpVideoStreamReceiver receiver;
+  TestRtpVideoFrameReceiver receiver;
   rtc::scoped_refptr<MockFrameTransformer> frame_transformer(
       new rtc::RefCountedObject<MockFrameTransformer>());
   rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate> delegate(
@@ -134,7 +74,7 @@
 
 TEST(RtpVideoStreamReceiverFrameTransformerDelegateTest,
      UnregisterTransformedFrameSinkCallbackOnReset) {
-  TestRtpVideoStreamReceiver receiver;
+  TestRtpVideoFrameReceiver receiver;
   rtc::scoped_refptr<MockFrameTransformer> frame_transformer(
       new rtc::RefCountedObject<MockFrameTransformer>());
   rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate> delegate(
@@ -146,7 +86,7 @@
 }
 
 TEST(RtpVideoStreamReceiverFrameTransformerDelegateTest, TransformFrame) {
-  TestRtpVideoStreamReceiver receiver;
+  TestRtpVideoFrameReceiver receiver;
   rtc::scoped_refptr<MockFrameTransformer> frame_transformer(
       new rtc::RefCountedObject<testing::NiceMock<MockFrameTransformer>>());
   rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate> delegate(
@@ -160,7 +100,7 @@
 
 TEST(RtpVideoStreamReceiverFrameTransformerDelegateTest,
      ManageFrameOnTransformedFrame) {
-  TestRtpVideoStreamReceiver receiver;
+  TestRtpVideoFrameReceiver receiver;
   rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer(
       new rtc::RefCountedObject<NiceMock<MockFrameTransformer>>());
   rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate> delegate =
@@ -186,7 +126,7 @@
 
 TEST(RtpVideoStreamReceiverFrameTransformerDelegateTest,
      TransformableFrameMetadataHasCorrectValue) {
-  TestRtpVideoStreamReceiver receiver;
+  TestRtpVideoFrameReceiver receiver;
   rtc::scoped_refptr<MockFrameTransformer> mock_frame_transformer =
       new rtc::RefCountedObject<NiceMock<MockFrameTransformer>>();
   rtc::scoped_refptr<RtpVideoStreamReceiverFrameTransformerDelegate> delegate =
diff --git a/video/video_receive_stream2.cc b/video/video_receive_stream2.cc
index b1b482d..1470123 100644
--- a/video/video_receive_stream2.cc
+++ b/video/video_receive_stream2.cc
@@ -49,7 +49,7 @@
 #include "system_wrappers/include/field_trial.h"
 #include "video/call_stats2.h"
 #include "video/frame_dumping_decoder.h"
-#include "video/receive_statistics_proxy.h"
+#include "video/receive_statistics_proxy2.h"
 
 namespace webrtc {
 
diff --git a/video/video_receive_stream2.h b/video/video_receive_stream2.h
index f8cd65d..0eab5dd 100644
--- a/video/video_receive_stream2.h
+++ b/video/video_receive_stream2.h
@@ -30,7 +30,7 @@
 #include "system_wrappers/include/clock.h"
 #include "video/receive_statistics_proxy2.h"
 #include "video/rtp_streams_synchronizer2.h"
-#include "video/rtp_video_stream_receiver.h"
+#include "video/rtp_video_stream_receiver2.h"
 #include "video/transport_adapter.h"
 #include "video/video_stream_decoder2.h"
 
@@ -199,7 +199,7 @@
   std::unique_ptr<VCMTiming> timing_;  // Jitter buffer experiment.
   VideoReceiver2 video_receiver_;
   std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> incoming_video_stream_;
-  RtpVideoStreamReceiver rtp_video_stream_receiver_;
+  RtpVideoStreamReceiver2 rtp_video_stream_receiver_;
   std::unique_ptr<VideoStreamDecoder> video_stream_decoder_;
   RtpStreamsSynchronizer rtp_stream_sync_;