Adding upstream version 124.0.1.upstream/124.0.1

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

1 files changed, 1428 insertions, 0 deletions

diff --git a/third_party/libwebrtc/call/call.cc b/third_party/libwebrtc/call/call.cc
new file mode 100644
index 0000000000..0f3699501e
--- /dev/null
+++ b/third_party/libwebrtc/call/call.cc
@@ -0,0 +1,1428 @@
+/*
+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "call/call.h"
+
+#include <string.h>
+
+#include <algorithm>
+#include <atomic>
+#include <cstdint>
+#include <map>
+#include <memory>
+#include <set>
+#include <utility>
+#include <vector>
+
+#include "absl/functional/bind_front.h"
+#include "absl/strings/string_view.h"
+#include "absl/types/optional.h"
+#include "api/media_types.h"
+#include "api/rtc_event_log/rtc_event_log.h"
+#include "api/sequence_checker.h"
+#include "api/task_queue/pending_task_safety_flag.h"
+#include "api/transport/network_control.h"
+#include "audio/audio_receive_stream.h"
+#include "audio/audio_send_stream.h"
+#include "audio/audio_state.h"
+#include "call/adaptation/broadcast_resource_listener.h"
+#include "call/bitrate_allocator.h"
+#include "call/flexfec_receive_stream_impl.h"
+#include "call/packet_receiver.h"
+#include "call/receive_time_calculator.h"
+#include "call/rtp_stream_receiver_controller.h"
+#include "call/rtp_transport_controller_send.h"
+#include "call/rtp_transport_controller_send_factory.h"
+#include "call/version.h"
+#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
+#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
+#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
+#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
+#include "logging/rtc_event_log/rtc_stream_config.h"
+#include "modules/congestion_controller/include/receive_side_congestion_controller.h"
+#include "modules/rtp_rtcp/include/flexfec_receiver.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
+#include "modules/rtp_rtcp/source/byte_io.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_util.h"
+#include "modules/video_coding/fec_controller_default.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/strings/string_builder.h"
+#include "rtc_base/system/no_unique_address.h"
+#include "rtc_base/task_utils/repeating_task.h"
+#include "rtc_base/thread_annotations.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+#include "system_wrappers/include/clock.h"
+#include "system_wrappers/include/cpu_info.h"
+#include "system_wrappers/include/metrics.h"
+#include "video/call_stats2.h"
+#include "video/send_delay_stats.h"
+#include "video/stats_counter.h"
+#include "video/video_receive_stream2.h"
+#include "video/video_send_stream.h"
+
+namespace webrtc {
+
+namespace {
+
+const int* FindKeyByValue(const std::map<int, int>& m, int v) {
+  for (const auto& kv : m) {
+    if (kv.second == v)
+      return &kv.first;
+  }
+  return nullptr;
+}
+
+std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
+    const VideoReceiveStreamInterface::Config& config) {
+  auto rtclog_config = std::make_unique<rtclog::StreamConfig>();
+  rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
+  rtclog_config->local_ssrc = config.rtp.local_ssrc;
+  rtclog_config->rtx_ssrc = config.rtp.rtx_ssrc;
+  rtclog_config->rtcp_mode = config.rtp.rtcp_mode;
+
+  for (const auto& d : config.decoders) {
+    const int* search =
+        FindKeyByValue(config.rtp.rtx_associated_payload_types, d.payload_type);
+    rtclog_config->codecs.emplace_back(d.video_format.name, d.payload_type,
+                                       search ? *search : 0);
+  }
+  return rtclog_config;
+}
+
+std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
+    const VideoSendStream::Config& config,
+    size_t ssrc_index) {
+  auto rtclog_config = std::make_unique<rtclog::StreamConfig>();
+  rtclog_config->local_ssrc = config.rtp.ssrcs[ssrc_index];
+  if (ssrc_index < config.rtp.rtx.ssrcs.size()) {
+    rtclog_config->rtx_ssrc = config.rtp.rtx.ssrcs[ssrc_index];
+  }
+  rtclog_config->rtcp_mode = config.rtp.rtcp_mode;
+  rtclog_config->rtp_extensions = config.rtp.extensions;
+
+  rtclog_config->codecs.emplace_back(config.rtp.payload_name,
+                                     config.rtp.payload_type,
+                                     config.rtp.rtx.payload_type);
+  return rtclog_config;
+}
+
+std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
+    const AudioReceiveStreamInterface::Config& config) {
+  auto rtclog_config = std::make_unique<rtclog::StreamConfig>();
+  rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
+  rtclog_config->local_ssrc = config.rtp.local_ssrc;
+  return rtclog_config;
+}
+
+TaskQueueBase* GetCurrentTaskQueueOrThread() {
+  TaskQueueBase* current = TaskQueueBase::Current();
+  if (!current)
+    current = rtc::ThreadManager::Instance()->CurrentThread();
+  return current;
+}
+
+}  // namespace
+
+namespace internal {
+
+// Wraps an injected resource in a BroadcastResourceListener and handles adding
+// and removing adapter resources to individual VideoSendStreams.
+class ResourceVideoSendStreamForwarder {
+ public:
+  ResourceVideoSendStreamForwarder(
+      rtc::scoped_refptr<webrtc::Resource> resource)
+      : broadcast_resource_listener_(resource) {
+    broadcast_resource_listener_.StartListening();
+  }
+  ~ResourceVideoSendStreamForwarder() {
+    RTC_DCHECK(adapter_resources_.empty());
+    broadcast_resource_listener_.StopListening();
+  }
+
+  rtc::scoped_refptr<webrtc::Resource> Resource() const {
+    return broadcast_resource_listener_.SourceResource();
+  }
+
+  void OnCreateVideoSendStream(VideoSendStream* video_send_stream) {
+    RTC_DCHECK(adapter_resources_.find(video_send_stream) ==
+               adapter_resources_.end());
+    auto adapter_resource =
+        broadcast_resource_listener_.CreateAdapterResource();
+    video_send_stream->AddAdaptationResource(adapter_resource);
+    adapter_resources_.insert(
+        std::make_pair(video_send_stream, adapter_resource));
+  }
+
+  void OnDestroyVideoSendStream(VideoSendStream* video_send_stream) {
+    auto it = adapter_resources_.find(video_send_stream);
+    RTC_DCHECK(it != adapter_resources_.end());
+    broadcast_resource_listener_.RemoveAdapterResource(it->second);
+    adapter_resources_.erase(it);
+  }
+
+ private:
+  BroadcastResourceListener broadcast_resource_listener_;
+  std::map<VideoSendStream*, rtc::scoped_refptr<webrtc::Resource>>
+      adapter_resources_;
+};
+
+class Call final : public webrtc::Call,
+                   public PacketReceiver,
+                   public TargetTransferRateObserver,
+                   public BitrateAllocator::LimitObserver {
+ public:
+  Call(Clock* clock,
+       const CallConfig& config,
+       std::unique_ptr<RtpTransportControllerSendInterface> transport_send,
+       TaskQueueFactory* task_queue_factory);
+  ~Call() override;
+
+  Call(const Call&) = delete;
+  Call& operator=(const Call&) = delete;
+
+  // Implements webrtc::Call.
+  PacketReceiver* Receiver() override;
+
+  webrtc::AudioSendStream* CreateAudioSendStream(
+      const webrtc::AudioSendStream::Config& config) override;
+  void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
+
+  webrtc::AudioReceiveStreamInterface* CreateAudioReceiveStream(
+      const webrtc::AudioReceiveStreamInterface::Config& config) override;
+  void DestroyAudioReceiveStream(
+      webrtc::AudioReceiveStreamInterface* receive_stream) override;
+
+  webrtc::VideoSendStream* CreateVideoSendStream(
+      webrtc::VideoSendStream::Config config,
+      VideoEncoderConfig encoder_config) override;
+  webrtc::VideoSendStream* CreateVideoSendStream(
+      webrtc::VideoSendStream::Config config,
+      VideoEncoderConfig encoder_config,
+      std::unique_ptr<FecController> fec_controller) override;
+  void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
+
+  webrtc::VideoReceiveStreamInterface* CreateVideoReceiveStream(
+      webrtc::VideoReceiveStreamInterface::Config configuration) override;
+  void DestroyVideoReceiveStream(
+      webrtc::VideoReceiveStreamInterface* receive_stream) override;
+
+  FlexfecReceiveStream* CreateFlexfecReceiveStream(
+      const FlexfecReceiveStream::Config config) override;
+  void DestroyFlexfecReceiveStream(
+      FlexfecReceiveStream* receive_stream) override;
+
+  void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) override;
+
+  RtpTransportControllerSendInterface* GetTransportControllerSend() override;
+
+  Stats GetStats() const override;
+
+  const FieldTrialsView& trials() const override;
+
+  TaskQueueBase* network_thread() const override;
+  TaskQueueBase* worker_thread() const override;
+
+  void DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) override;
+
+  void DeliverRtpPacket(
+      MediaType media_type,
+      RtpPacketReceived packet,
+      OnUndemuxablePacketHandler undemuxable_packet_handler) override;
+
+  void SignalChannelNetworkState(MediaType media, NetworkState state) override;
+
+  void OnAudioTransportOverheadChanged(
+      int transport_overhead_per_packet) override;
+
+  void OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream,
+                          uint32_t local_ssrc) override;
+  void OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
+                          uint32_t local_ssrc) override;
+  void OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
+                          uint32_t local_ssrc) override;
+
+  void OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream,
+                         absl::string_view sync_group) override;
+
+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
+
+  // Implements TargetTransferRateObserver,
+  void OnTargetTransferRate(TargetTransferRate msg) override;
+  void OnStartRateUpdate(DataRate start_rate) override;
+
+  // Implements BitrateAllocator::LimitObserver.
+  void OnAllocationLimitsChanged(BitrateAllocationLimits limits) override;
+
+  void SetClientBitratePreferences(const BitrateSettings& preferences) override;
+
+ private:
+  // Thread-compatible class that collects received packet stats and exposes
+  // them as UMA histograms on destruction.
+  class ReceiveStats {
+   public:
+    explicit ReceiveStats(Clock* clock);
+    ~ReceiveStats();
+
+    void AddReceivedRtcpBytes(int bytes);
+    void AddReceivedAudioBytes(int bytes, webrtc::Timestamp arrival_time);
+    void AddReceivedVideoBytes(int bytes, webrtc::Timestamp arrival_time);
+
+   private:
+    RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+    RateCounter received_bytes_per_second_counter_
+        RTC_GUARDED_BY(sequence_checker_);
+    RateCounter received_audio_bytes_per_second_counter_
+        RTC_GUARDED_BY(sequence_checker_);
+    RateCounter received_video_bytes_per_second_counter_
+        RTC_GUARDED_BY(sequence_checker_);
+    RateCounter received_rtcp_bytes_per_second_counter_
+        RTC_GUARDED_BY(sequence_checker_);
+    absl::optional<Timestamp> first_received_rtp_audio_timestamp_
+        RTC_GUARDED_BY(sequence_checker_);
+    absl::optional<Timestamp> last_received_rtp_audio_timestamp_
+        RTC_GUARDED_BY(sequence_checker_);
+    absl::optional<Timestamp> first_received_rtp_video_timestamp_
+        RTC_GUARDED_BY(sequence_checker_);
+    absl::optional<Timestamp> last_received_rtp_video_timestamp_
+        RTC_GUARDED_BY(sequence_checker_);
+  };
+
+  // Thread-compatible class that collects sent packet stats and exposes
+  // them as UMA histograms on destruction, provided SetFirstPacketTime was
+  // called with a non-empty packet timestamp before the destructor.
+  class SendStats {
+   public:
+    explicit SendStats(Clock* clock);
+    ~SendStats();
+
+    void SetFirstPacketTime(absl::optional<Timestamp> first_sent_packet_time);
+    void PauseSendAndPacerBitrateCounters();
+    void AddTargetBitrateSample(uint32_t target_bitrate_bps);
+    void SetMinAllocatableRate(BitrateAllocationLimits limits);
+
+   private:
+    RTC_NO_UNIQUE_ADDRESS SequenceChecker destructor_sequence_checker_;
+    RTC_NO_UNIQUE_ADDRESS SequenceChecker sequence_checker_;
+    Clock* const clock_ RTC_GUARDED_BY(destructor_sequence_checker_);
+    AvgCounter estimated_send_bitrate_kbps_counter_
+        RTC_GUARDED_BY(sequence_checker_);
+    AvgCounter pacer_bitrate_kbps_counter_ RTC_GUARDED_BY(sequence_checker_);
+    uint32_t min_allocated_send_bitrate_bps_ RTC_GUARDED_BY(sequence_checker_){
+        0};
+    absl::optional<Timestamp> first_sent_packet_time_
+        RTC_GUARDED_BY(destructor_sequence_checker_);
+  };
+
+  void DeliverRtcp(MediaType media_type, rtc::CopyOnWriteBuffer packet)
+      RTC_RUN_ON(network_thread_);
+
+  AudioReceiveStreamImpl* FindAudioStreamForSyncGroup(
+      absl::string_view sync_group) RTC_RUN_ON(worker_thread_);
+  void ConfigureSync(absl::string_view sync_group) RTC_RUN_ON(worker_thread_);
+
+  void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
+                                 MediaType media_type)
+      RTC_RUN_ON(worker_thread_);
+
+  bool RegisterReceiveStream(uint32_t ssrc, ReceiveStreamInterface* stream);
+  bool UnregisterReceiveStream(uint32_t ssrc);
+
+  void UpdateAggregateNetworkState();
+
+  // Ensure that necessary process threads are started, and any required
+  // callbacks have been registered.
+  void EnsureStarted() RTC_RUN_ON(worker_thread_);
+
+  Clock* const clock_;
+  TaskQueueFactory* const task_queue_factory_;
+  TaskQueueBase* const worker_thread_;
+  TaskQueueBase* const network_thread_;
+  const std::unique_ptr<DecodeSynchronizer> decode_sync_;
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker send_transport_sequence_checker_;
+
+  const int num_cpu_cores_;
+  const std::unique_ptr<CallStats> call_stats_;
+  const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
+  const CallConfig config_ RTC_GUARDED_BY(worker_thread_);
+  // Maps to config_.trials, can be used from any thread via `trials()`.
+  const FieldTrialsView& trials_;
+
+  NetworkState audio_network_state_ RTC_GUARDED_BY(worker_thread_);
+  NetworkState video_network_state_ RTC_GUARDED_BY(worker_thread_);
+  // TODO(bugs.webrtc.org/11993): Move aggregate_network_up_ over to the
+  // network thread.
+  bool aggregate_network_up_ RTC_GUARDED_BY(worker_thread_);
+
+  // Schedules nack periodic processing on behalf of all streams.
+  NackPeriodicProcessor nack_periodic_processor_;
+
+  // Audio, Video, and FlexFEC receive streams are owned by the client that
+  // creates them.
+  // TODO(bugs.webrtc.org/11993): Move audio_receive_streams_,
+  // video_receive_streams_ over to the network thread.
+  std::set<AudioReceiveStreamImpl*> audio_receive_streams_
+      RTC_GUARDED_BY(worker_thread_);
+  std::set<VideoReceiveStream2*> video_receive_streams_
+      RTC_GUARDED_BY(worker_thread_);
+  // TODO(bugs.webrtc.org/7135, bugs.webrtc.org/9719): Should eventually be
+  // injected at creation, with a single object in the bundled case.
+  RtpStreamReceiverController audio_receiver_controller_
+      RTC_GUARDED_BY(worker_thread_);
+  RtpStreamReceiverController video_receiver_controller_
+      RTC_GUARDED_BY(worker_thread_);
+
+  // This extra map is used for receive processing which is
+  // independent of media type.
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker receive_11993_checker_;
+
+  // Audio and Video send streams are owned by the client that creates them.
+  // TODO(bugs.webrtc.org/11993): `audio_send_ssrcs_` and `video_send_ssrcs_`
+  // should be accessed on the network thread.
+  std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_
+      RTC_GUARDED_BY(worker_thread_);
+  std::map<uint32_t, VideoSendStream*> video_send_ssrcs_
+      RTC_GUARDED_BY(worker_thread_);
+  std::set<VideoSendStream*> video_send_streams_ RTC_GUARDED_BY(worker_thread_);
+  // True if `video_send_streams_` is empty, false if not. The atomic variable
+  // is used to decide UMA send statistics behavior and enables avoiding a
+  // PostTask().
+  std::atomic<bool> video_send_streams_empty_{true};
+
+  // Each forwarder wraps an adaptation resource that was added to the call.
+  std::vector<std::unique_ptr<ResourceVideoSendStreamForwarder>>
+      adaptation_resource_forwarders_ RTC_GUARDED_BY(worker_thread_);
+
+  using RtpStateMap = std::map<uint32_t, RtpState>;
+  RtpStateMap suspended_audio_send_ssrcs_ RTC_GUARDED_BY(worker_thread_);
+  RtpStateMap suspended_video_send_ssrcs_ RTC_GUARDED_BY(worker_thread_);
+
+  using RtpPayloadStateMap = std::map<uint32_t, RtpPayloadState>;
+  RtpPayloadStateMap suspended_video_payload_states_
+      RTC_GUARDED_BY(worker_thread_);
+
+  webrtc::RtcEventLog* const event_log_;
+
+  // TODO(bugs.webrtc.org/11993) ready to move stats access to the network
+  // thread.
+  ReceiveStats receive_stats_ RTC_GUARDED_BY(worker_thread_);
+  SendStats send_stats_ RTC_GUARDED_BY(send_transport_sequence_checker_);
+  // `last_bandwidth_bps_` and `configured_max_padding_bitrate_bps_` being
+  // atomic avoids a PostTask. The variables are used for stats gathering.
+  std::atomic<uint32_t> last_bandwidth_bps_{0};
+  std::atomic<uint32_t> configured_max_padding_bitrate_bps_{0};
+
+  ReceiveSideCongestionController receive_side_cc_;
+  RepeatingTaskHandle receive_side_cc_periodic_task_;
+
+  const std::unique_ptr<ReceiveTimeCalculator> receive_time_calculator_;
+
+  const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
+  const Timestamp start_of_call_;
+
+  // Note that `task_safety_` needs to be at a greater scope than the task queue
+  // owned by `transport_send_` since calls might arrive on the network thread
+  // while Call is being deleted and the task queue is being torn down.
+  const ScopedTaskSafety task_safety_;
+
+  // Caches transport_send_.get(), to avoid racing with destructor.
+  // Note that this is declared before transport_send_ to ensure that it is not
+  // invalidated until no more tasks can be running on the transport_send_ task
+  // queue.
+  // For more details on the background of this member variable, see:
+  // https://webrtc-review.googlesource.com/c/src/+/63023/9/call/call.cc
+  // https://bugs.chromium.org/p/chromium/issues/detail?id=992640
+  RtpTransportControllerSendInterface* const transport_send_ptr_
+      RTC_GUARDED_BY(send_transport_sequence_checker_);
+  // Declared last since it will issue callbacks from a task queue. Declaring it
+  // last ensures that it is destroyed first and any running tasks are finished.
+  const std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
+
+  bool is_started_ RTC_GUARDED_BY(worker_thread_) = false;
+
+  // Sequence checker for outgoing network traffic. Could be the network thread.
+  // Could also be a pacer owned thread or TQ such as the TaskQueuePacedSender.
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sent_packet_sequence_checker_;
+  absl::optional<rtc::SentPacket> last_sent_packet_
+      RTC_GUARDED_BY(sent_packet_sequence_checker_);
+};
+}  // namespace internal
+
+/* Mozilla: Avoid this since it could use GetRealTimeClock().
+std::unique_ptr<Call> Call::Create(const CallConfig& config) {
+  Clock* clock = Clock::GetRealTimeClock();
+  return Create(config, clock,
+                RtpTransportControllerSendFactory().Create(
+                    config.ExtractTransportConfig(), clock));
+}
+ */
+
+std::unique_ptr<Call> Call::Create(
+    const CallConfig& config,
+    Clock* clock,
+    std::unique_ptr<RtpTransportControllerSendInterface>
+        transportControllerSend) {
+  RTC_DCHECK(config.task_queue_factory);
+  return std::make_unique<internal::Call>(clock, config,
+                                          std::move(transportControllerSend),
+                                          config.task_queue_factory);
+}
+
+// This method here to avoid subclasses has to implement this method.
+// Call perf test will use Internal::Call::CreateVideoSendStream() to inject
+// FecController.
+VideoSendStream* Call::CreateVideoSendStream(
+    VideoSendStream::Config config,
+    VideoEncoderConfig encoder_config,
+    std::unique_ptr<FecController> fec_controller) {
+  return nullptr;
+}
+
+namespace internal {
+
+Call::ReceiveStats::ReceiveStats(Clock* clock)
+    : received_bytes_per_second_counter_(clock, nullptr, false),
+      received_audio_bytes_per_second_counter_(clock, nullptr, false),
+      received_video_bytes_per_second_counter_(clock, nullptr, false),
+      received_rtcp_bytes_per_second_counter_(clock, nullptr, false) {
+  sequence_checker_.Detach();
+}
+
+void Call::ReceiveStats::AddReceivedRtcpBytes(int bytes) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (received_bytes_per_second_counter_.HasSample()) {
+    // First RTP packet has been received.
+    received_bytes_per_second_counter_.Add(static_cast<int>(bytes));
+    received_rtcp_bytes_per_second_counter_.Add(static_cast<int>(bytes));
+  }
+}
+
+void Call::ReceiveStats::AddReceivedAudioBytes(int bytes,
+                                               webrtc::Timestamp arrival_time) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  received_bytes_per_second_counter_.Add(bytes);
+  received_audio_bytes_per_second_counter_.Add(bytes);
+  if (!first_received_rtp_audio_timestamp_)
+    first_received_rtp_audio_timestamp_ = arrival_time;
+  last_received_rtp_audio_timestamp_ = arrival_time;
+}
+
+void Call::ReceiveStats::AddReceivedVideoBytes(int bytes,
+                                               webrtc::Timestamp arrival_time) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  received_bytes_per_second_counter_.Add(bytes);
+  received_video_bytes_per_second_counter_.Add(bytes);
+  if (!first_received_rtp_video_timestamp_)
+    first_received_rtp_video_timestamp_ = arrival_time;
+  last_received_rtp_video_timestamp_ = arrival_time;
+}
+
+Call::ReceiveStats::~ReceiveStats() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  if (first_received_rtp_audio_timestamp_) {
+    RTC_HISTOGRAM_COUNTS_100000(
+        "WebRTC.Call.TimeReceivingAudioRtpPacketsInSeconds",
+        (*last_received_rtp_audio_timestamp_ -
+         *first_received_rtp_audio_timestamp_)
+            .seconds());
+  }
+  if (first_received_rtp_video_timestamp_) {
+    RTC_HISTOGRAM_COUNTS_100000(
+        "WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds",
+        (*last_received_rtp_video_timestamp_ -
+         *first_received_rtp_video_timestamp_)
+            .seconds());
+  }
+  const int kMinRequiredPeriodicSamples = 5;
+  AggregatedStats video_bytes_per_sec =
+      received_video_bytes_per_second_counter_.GetStats();
+  if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
+                                video_bytes_per_sec.average * 8 / 1000);
+    RTC_LOG(LS_INFO) << "WebRTC.Call.VideoBitrateReceivedInBps, "
+                     << video_bytes_per_sec.ToStringWithMultiplier(8);
+  }
+  AggregatedStats audio_bytes_per_sec =
+      received_audio_bytes_per_second_counter_.GetStats();
+  if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
+                                audio_bytes_per_sec.average * 8 / 1000);
+    RTC_LOG(LS_INFO) << "WebRTC.Call.AudioBitrateReceivedInBps, "
+                     << audio_bytes_per_sec.ToStringWithMultiplier(8);
+  }
+  AggregatedStats rtcp_bytes_per_sec =
+      received_rtcp_bytes_per_second_counter_.GetStats();
+  if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
+                                rtcp_bytes_per_sec.average * 8);
+    RTC_LOG(LS_INFO) << "WebRTC.Call.RtcpBitrateReceivedInBps, "
+                     << rtcp_bytes_per_sec.ToStringWithMultiplier(8);
+  }
+  AggregatedStats recv_bytes_per_sec =
+      received_bytes_per_second_counter_.GetStats();
+  if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
+                                recv_bytes_per_sec.average * 8 / 1000);
+    RTC_LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, "
+                     << recv_bytes_per_sec.ToStringWithMultiplier(8);
+  }
+}
+
+Call::SendStats::SendStats(Clock* clock)
+    : clock_(clock),
+      estimated_send_bitrate_kbps_counter_(clock, nullptr, true),
+      pacer_bitrate_kbps_counter_(clock, nullptr, true) {
+  destructor_sequence_checker_.Detach();
+  sequence_checker_.Detach();
+}
+
+Call::SendStats::~SendStats() {
+  RTC_DCHECK_RUN_ON(&destructor_sequence_checker_);
+  if (!first_sent_packet_time_)
+    return;
+
+  TimeDelta elapsed = clock_->CurrentTime() - *first_sent_packet_time_;
+  if (elapsed.seconds() < metrics::kMinRunTimeInSeconds)
+    return;
+
+  const int kMinRequiredPeriodicSamples = 5;
+  AggregatedStats send_bitrate_stats =
+      estimated_send_bitrate_kbps_counter_.ProcessAndGetStats();
+  if (send_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
+                                send_bitrate_stats.average);
+    RTC_LOG(LS_INFO) << "WebRTC.Call.EstimatedSendBitrateInKbps, "
+                     << send_bitrate_stats.ToString();
+  }
+  AggregatedStats pacer_bitrate_stats =
+      pacer_bitrate_kbps_counter_.ProcessAndGetStats();
+  if (pacer_bitrate_stats.num_samples > kMinRequiredPeriodicSamples) {
+    RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
+                                pacer_bitrate_stats.average);
+    RTC_LOG(LS_INFO) << "WebRTC.Call.PacerBitrateInKbps, "
+                     << pacer_bitrate_stats.ToString();
+  }
+}
+
+void Call::SendStats::SetFirstPacketTime(
+    absl::optional<Timestamp> first_sent_packet_time) {
+  RTC_DCHECK_RUN_ON(&destructor_sequence_checker_);
+  first_sent_packet_time_ = first_sent_packet_time;
+}
+
+void Call::SendStats::PauseSendAndPacerBitrateCounters() {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  estimated_send_bitrate_kbps_counter_.ProcessAndPause();
+  pacer_bitrate_kbps_counter_.ProcessAndPause();
+}
+
+void Call::SendStats::AddTargetBitrateSample(uint32_t target_bitrate_bps) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  estimated_send_bitrate_kbps_counter_.Add(target_bitrate_bps / 1000);
+  // Pacer bitrate may be higher than bitrate estimate if enforcing min
+  // bitrate.
+  uint32_t pacer_bitrate_bps =
+      std::max(target_bitrate_bps, min_allocated_send_bitrate_bps_);
+  pacer_bitrate_kbps_counter_.Add(pacer_bitrate_bps / 1000);
+}
+
+void Call::SendStats::SetMinAllocatableRate(BitrateAllocationLimits limits) {
+  RTC_DCHECK_RUN_ON(&sequence_checker_);
+  min_allocated_send_bitrate_bps_ = limits.min_allocatable_rate.bps();
+}
+
+Call::Call(Clock* clock,
+           const CallConfig& config,
+           std::unique_ptr<RtpTransportControllerSendInterface> transport_send,
+           TaskQueueFactory* task_queue_factory)
+    : clock_(clock),
+      task_queue_factory_(task_queue_factory),
+      worker_thread_(GetCurrentTaskQueueOrThread()),
+      // If `network_task_queue_` was set to nullptr, network related calls
+      // must be made on `worker_thread_` (i.e. they're one and the same).
+      network_thread_(config.network_task_queue_ ? config.network_task_queue_
+                                                 : worker_thread_),
+      decode_sync_(config.metronome
+                       ? std::make_unique<DecodeSynchronizer>(clock_,
+                                                              config.metronome,
+                                                              worker_thread_)
+                       : nullptr),
+      num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
+      call_stats_(new CallStats(clock_, worker_thread_)),
+      bitrate_allocator_(new BitrateAllocator(this)),
+      config_(config),
+      trials_(*config.trials),
+      audio_network_state_(kNetworkDown),
+      video_network_state_(kNetworkDown),
+      aggregate_network_up_(false),
+      event_log_(config.event_log),
+      receive_stats_(clock_),
+      send_stats_(clock_),
+      receive_side_cc_(clock,
+                       absl::bind_front(&PacketRouter::SendCombinedRtcpPacket,
+                                        transport_send->packet_router()),
+                       absl::bind_front(&PacketRouter::SendRemb,
+                                        transport_send->packet_router()),
+                       /*network_state_estimator=*/nullptr),
+      receive_time_calculator_(
+          ReceiveTimeCalculator::CreateFromFieldTrial(*config.trials)),
+      video_send_delay_stats_(new SendDelayStats(clock_)),
+      start_of_call_(clock_->CurrentTime()),
+      transport_send_ptr_(transport_send.get()),
+      transport_send_(std::move(transport_send)) {
+  RTC_DCHECK(config.event_log != nullptr);
+  RTC_DCHECK(config.trials != nullptr);
+  RTC_DCHECK(network_thread_);
+  RTC_DCHECK(worker_thread_->IsCurrent());
+
+  receive_11993_checker_.Detach();
+  send_transport_sequence_checker_.Detach();
+  sent_packet_sequence_checker_.Detach();
+
+  // Do not remove this call; it is here to convince the compiler that the
+  // WebRTC source timestamp string needs to be in the final binary.
+  LoadWebRTCVersionInRegister();
+
+  call_stats_->RegisterStatsObserver(&receive_side_cc_);
+
+  ReceiveSideCongestionController* receive_side_cc = &receive_side_cc_;
+  receive_side_cc_periodic_task_ = RepeatingTaskHandle::Start(
+      worker_thread_,
+      [receive_side_cc] { return receive_side_cc->MaybeProcess(); },
+      TaskQueueBase::DelayPrecision::kLow, clock_);
+}
+
+Call::~Call() {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  RTC_CHECK(audio_send_ssrcs_.empty());
+  RTC_CHECK(video_send_ssrcs_.empty());
+  RTC_CHECK(video_send_streams_.empty());
+  RTC_CHECK(audio_receive_streams_.empty());
+  RTC_CHECK(video_receive_streams_.empty());
+
+  receive_side_cc_periodic_task_.Stop();
+  call_stats_->DeregisterStatsObserver(&receive_side_cc_);
+  send_stats_.SetFirstPacketTime(transport_send_->GetFirstPacketTime());
+
+  RTC_HISTOGRAM_COUNTS_100000(
+      "WebRTC.Call.LifetimeInSeconds",
+      (clock_->CurrentTime() - start_of_call_).seconds());
+}
+
+void Call::EnsureStarted() {
+  if (is_started_) {
+    return;
+  }
+  is_started_ = true;
+
+  call_stats_->EnsureStarted();
+
+  // This call seems to kick off a number of things, so probably better left
+  // off being kicked off on request rather than in the ctor.
+  transport_send_->RegisterTargetTransferRateObserver(this);
+
+  transport_send_->EnsureStarted();
+}
+
+void Call::SetClientBitratePreferences(const BitrateSettings& preferences) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  GetTransportControllerSend()->SetClientBitratePreferences(preferences);
+}
+
+PacketReceiver* Call::Receiver() {
+  return this;
+}
+
+webrtc::AudioSendStream* Call::CreateAudioSendStream(
+    const webrtc::AudioSendStream::Config& config) {
+  TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  EnsureStarted();
+
+  // Stream config is logged in AudioSendStream::ConfigureStream, as it may
+  // change during the stream's lifetime.
+  absl::optional<RtpState> suspended_rtp_state;
+  {
+    const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
+    if (iter != suspended_audio_send_ssrcs_.end()) {
+      suspended_rtp_state.emplace(iter->second);
+    }
+  }
+
+  AudioSendStream* send_stream = new AudioSendStream(
+      clock_, config, config_.audio_state, task_queue_factory_,
+      transport_send_.get(), bitrate_allocator_.get(), event_log_,
+      call_stats_->AsRtcpRttStats(), suspended_rtp_state, trials());
+  RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
+             audio_send_ssrcs_.end());
+  audio_send_ssrcs_[config.rtp.ssrc] = send_stream;
+
+  // TODO(bugs.webrtc.org/11993): call AssociateSendStream and
+  // UpdateAggregateNetworkState asynchronously on the network thread.
+  for (AudioReceiveStreamImpl* stream : audio_receive_streams_) {
+    if (stream->local_ssrc() == config.rtp.ssrc) {
+      stream->AssociateSendStream(send_stream);
+    }
+  }
+
+  UpdateAggregateNetworkState();
+
+  return send_stream;
+}
+
+void Call::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyAudioSendStream");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK(send_stream != nullptr);
+
+  send_stream->Stop();
+
+  const uint32_t ssrc = send_stream->GetConfig().rtp.ssrc;
+  webrtc::internal::AudioSendStream* audio_send_stream =
+      static_cast<webrtc::internal::AudioSendStream*>(send_stream);
+  suspended_audio_send_ssrcs_[ssrc] = audio_send_stream->GetRtpState();
+
+  size_t num_deleted = audio_send_ssrcs_.erase(ssrc);
+  RTC_DCHECK_EQ(1, num_deleted);
+
+  // TODO(bugs.webrtc.org/11993): call AssociateSendStream and
+  // UpdateAggregateNetworkState asynchronously on the network thread.
+  for (AudioReceiveStreamImpl* stream : audio_receive_streams_) {
+    if (stream->local_ssrc() == ssrc) {
+      stream->AssociateSendStream(nullptr);
+    }
+  }
+
+  UpdateAggregateNetworkState();
+
+  delete send_stream;
+}
+
+webrtc::AudioReceiveStreamInterface* Call::CreateAudioReceiveStream(
+    const webrtc::AudioReceiveStreamInterface::Config& config) {
+  TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  EnsureStarted();
+  event_log_->Log(std::make_unique<RtcEventAudioReceiveStreamConfig>(
+      CreateRtcLogStreamConfig(config)));
+
+  AudioReceiveStreamImpl* receive_stream = new AudioReceiveStreamImpl(
+      clock_, transport_send_->packet_router(), config_.neteq_factory, config,
+      config_.audio_state, event_log_);
+  audio_receive_streams_.insert(receive_stream);
+
+  // TODO(bugs.webrtc.org/11993): Make the registration on the network thread
+  // (asynchronously). The registration and `audio_receiver_controller_` need
+  // to live on the network thread.
+  receive_stream->RegisterWithTransport(&audio_receiver_controller_);
+
+  ConfigureSync(config.sync_group);
+
+  auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc);
+  if (it != audio_send_ssrcs_.end()) {
+    receive_stream->AssociateSendStream(it->second);
+  }
+
+  UpdateAggregateNetworkState();
+  return receive_stream;
+}
+
+void Call::DestroyAudioReceiveStream(
+    webrtc::AudioReceiveStreamInterface* receive_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyAudioReceiveStream");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK(receive_stream != nullptr);
+  webrtc::AudioReceiveStreamImpl* audio_receive_stream =
+      static_cast<webrtc::AudioReceiveStreamImpl*>(receive_stream);
+
+  // TODO(bugs.webrtc.org/11993): Access the map, rtp config, call ConfigureSync
+  // and UpdateAggregateNetworkState on the network thread. The call to
+  // `UnregisterFromTransport` should also happen on the network thread.
+  audio_receive_stream->UnregisterFromTransport();
+
+  uint32_t ssrc = audio_receive_stream->remote_ssrc();
+  receive_side_cc_.RemoveStream(ssrc);
+
+  audio_receive_streams_.erase(audio_receive_stream);
+
+  // After calling erase(), call ConfigureSync. This will clear associated
+  // video streams or associate them with a different audio stream if one exists
+  // for this sync_group.
+  ConfigureSync(audio_receive_stream->sync_group());
+
+  UpdateAggregateNetworkState();
+  // TODO(bugs.webrtc.org/11993): Consider if deleting `audio_receive_stream`
+  // on the network thread would be better or if we'd need to tear down the
+  // state in two phases.
+  delete audio_receive_stream;
+}
+
+// This method can be used for Call tests with external fec controller factory.
+webrtc::VideoSendStream* Call::CreateVideoSendStream(
+    webrtc::VideoSendStream::Config config,
+    VideoEncoderConfig encoder_config,
+    std::unique_ptr<FecController> fec_controller) {
+  TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  EnsureStarted();
+
+  video_send_delay_stats_->AddSsrcs(config);
+  for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
+       ++ssrc_index) {
+    event_log_->Log(std::make_unique<RtcEventVideoSendStreamConfig>(
+        CreateRtcLogStreamConfig(config, ssrc_index)));
+  }
+
+  // TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if
+  // the call has already started.
+  // Copy ssrcs from `config` since `config` is moved.
+  std::vector<uint32_t> ssrcs = config.rtp.ssrcs;
+
+  VideoSendStream* send_stream = new VideoSendStream(
+      clock_, num_cpu_cores_, task_queue_factory_, network_thread_,
+      call_stats_->AsRtcpRttStats(), transport_send_.get(),
+      bitrate_allocator_.get(), video_send_delay_stats_.get(), event_log_,
+      std::move(config), std::move(encoder_config), suspended_video_send_ssrcs_,
+      suspended_video_payload_states_, std::move(fec_controller),
+      *config_.trials);
+
+  for (uint32_t ssrc : ssrcs) {
+    RTC_DCHECK(video_send_ssrcs_.find(ssrc) == video_send_ssrcs_.end());
+    video_send_ssrcs_[ssrc] = send_stream;
+  }
+  video_send_streams_.insert(send_stream);
+  video_send_streams_empty_.store(false, std::memory_order_relaxed);
+
+  // Forward resources that were previously added to the call to the new stream.
+  for (const auto& resource_forwarder : adaptation_resource_forwarders_) {
+    resource_forwarder->OnCreateVideoSendStream(send_stream);
+  }
+
+  UpdateAggregateNetworkState();
+
+  return send_stream;
+}
+
+webrtc::VideoSendStream* Call::CreateVideoSendStream(
+    webrtc::VideoSendStream::Config config,
+    VideoEncoderConfig encoder_config) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  if (config_.fec_controller_factory) {
+    RTC_LOG(LS_INFO) << "External FEC Controller will be used.";
+  }
+  std::unique_ptr<FecController> fec_controller =
+      config_.fec_controller_factory
+          ? config_.fec_controller_factory->CreateFecController()
+          : std::make_unique<FecControllerDefault>(clock_);
+  return CreateVideoSendStream(std::move(config), std::move(encoder_config),
+                               std::move(fec_controller));
+}
+
+void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
+  RTC_DCHECK(send_stream != nullptr);
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  VideoSendStream* send_stream_impl =
+      static_cast<VideoSendStream*>(send_stream);
+
+  auto it = video_send_ssrcs_.begin();
+  while (it != video_send_ssrcs_.end()) {
+    if (it->second == static_cast<VideoSendStream*>(send_stream)) {
+      send_stream_impl = it->second;
+      video_send_ssrcs_.erase(it++);
+    } else {
+      ++it;
+    }
+  }
+
+  // Stop forwarding resources to the stream being destroyed.
+  for (const auto& resource_forwarder : adaptation_resource_forwarders_) {
+    resource_forwarder->OnDestroyVideoSendStream(send_stream_impl);
+  }
+  video_send_streams_.erase(send_stream_impl);
+  if (video_send_streams_.empty())
+    video_send_streams_empty_.store(true, std::memory_order_relaxed);
+
+  VideoSendStream::RtpStateMap rtp_states;
+  VideoSendStream::RtpPayloadStateMap rtp_payload_states;
+  send_stream_impl->StopPermanentlyAndGetRtpStates(&rtp_states,
+                                                   &rtp_payload_states);
+  for (const auto& kv : rtp_states) {
+    suspended_video_send_ssrcs_[kv.first] = kv.second;
+  }
+  for (const auto& kv : rtp_payload_states) {
+    suspended_video_payload_states_[kv.first] = kv.second;
+  }
+
+  UpdateAggregateNetworkState();
+  // TODO(tommi): consider deleting on the same thread as runs
+  // StopPermanentlyAndGetRtpStates.
+  delete send_stream_impl;
+}
+
+webrtc::VideoReceiveStreamInterface* Call::CreateVideoReceiveStream(
+    webrtc::VideoReceiveStreamInterface::Config configuration) {
+  TRACE_EVENT0("webrtc", "Call::CreateVideoReceiveStream");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  EnsureStarted();
+
+  event_log_->Log(std::make_unique<RtcEventVideoReceiveStreamConfig>(
+      CreateRtcLogStreamConfig(configuration)));
+
+  // TODO(bugs.webrtc.org/11993): Move the registration between `receive_stream`
+  // and `video_receiver_controller_` out of VideoReceiveStream2 construction
+  // and set it up asynchronously on the network thread (the registration and
+  // `video_receiver_controller_` need to live on the network thread).
+  // TODO(crbug.com/1381982): Re-enable decode synchronizer once the Chromium
+  // API has adapted to the new Metronome interface.
+  VideoReceiveStream2* receive_stream = new VideoReceiveStream2(
+      task_queue_factory_, this, num_cpu_cores_,
+      transport_send_->packet_router(), std::move(configuration),
+      call_stats_.get(), clock_, std::make_unique<VCMTiming>(clock_, trials()),
+      &nack_periodic_processor_, decode_sync_.get(), event_log_);
+  // TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
+  // thread.
+  receive_stream->RegisterWithTransport(&video_receiver_controller_);
+  video_receive_streams_.insert(receive_stream);
+
+  ConfigureSync(receive_stream->sync_group());
+
+  receive_stream->SignalNetworkState(video_network_state_);
+  UpdateAggregateNetworkState();
+  return receive_stream;
+}
+
+void Call::DestroyVideoReceiveStream(
+    webrtc::VideoReceiveStreamInterface* receive_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyVideoReceiveStream");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK(receive_stream != nullptr);
+  VideoReceiveStream2* receive_stream_impl =
+      static_cast<VideoReceiveStream2*>(receive_stream);
+  // TODO(bugs.webrtc.org/11993): Unregister on the network thread.
+  receive_stream_impl->UnregisterFromTransport();
+  video_receive_streams_.erase(receive_stream_impl);
+  ConfigureSync(receive_stream_impl->sync_group());
+
+  receive_side_cc_.RemoveStream(receive_stream_impl->remote_ssrc());
+
+  UpdateAggregateNetworkState();
+  delete receive_stream_impl;
+}
+
+FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
+    const FlexfecReceiveStream::Config config) {
+  TRACE_EVENT0("webrtc", "Call::CreateFlexfecReceiveStream");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  // Unlike the video and audio receive streams, FlexfecReceiveStream implements
+  // RtpPacketSinkInterface itself, and hence its constructor passes its `this`
+  // pointer to video_receiver_controller_->CreateStream(). Calling the
+  // constructor while on the worker thread ensures that we don't call
+  // OnRtpPacket until the constructor is finished and the object is
+  // in a valid state, since OnRtpPacket runs on the same thread.
+  FlexfecReceiveStreamImpl* receive_stream = new FlexfecReceiveStreamImpl(
+      clock_, std::move(config), &video_receiver_controller_,
+      call_stats_->AsRtcpRttStats());
+
+  // TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
+  // thread.
+  receive_stream->RegisterWithTransport(&video_receiver_controller_);
+  // TODO(brandtr): Store config in RtcEventLog here.
+
+  return receive_stream;
+}
+
+void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
+  TRACE_EVENT0("webrtc", "Call::DestroyFlexfecReceiveStream");
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  FlexfecReceiveStreamImpl* receive_stream_impl =
+      static_cast<FlexfecReceiveStreamImpl*>(receive_stream);
+  // TODO(bugs.webrtc.org/11993): Unregister on the network thread.
+  receive_stream_impl->UnregisterFromTransport();
+
+  auto ssrc = receive_stream_impl->remote_ssrc();
+  // Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
+  // destroyed.
+  receive_side_cc_.RemoveStream(ssrc);
+
+  delete receive_stream_impl;
+}
+
+void Call::AddAdaptationResource(rtc::scoped_refptr<Resource> resource) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  adaptation_resource_forwarders_.push_back(
+      std::make_unique<ResourceVideoSendStreamForwarder>(resource));
+  const auto& resource_forwarder = adaptation_resource_forwarders_.back();
+  for (VideoSendStream* send_stream : video_send_streams_) {
+    resource_forwarder->OnCreateVideoSendStream(send_stream);
+  }
+}
+
+RtpTransportControllerSendInterface* Call::GetTransportControllerSend() {
+  return transport_send_.get();
+}
+
+Call::Stats Call::GetStats() const {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  Stats stats;
+  // TODO(srte): It is unclear if we only want to report queues if network is
+  // available.
+  stats.pacer_delay_ms =
+      aggregate_network_up_ ? transport_send_->GetPacerQueuingDelayMs() : 0;
+
+  stats.rtt_ms = call_stats_->LastProcessedRtt();
+
+  // Fetch available send/receive bitrates.
+  stats.recv_bandwidth_bps = receive_side_cc_.LatestReceiveSideEstimate().bps();
+  stats.send_bandwidth_bps =
+      last_bandwidth_bps_.load(std::memory_order_relaxed);
+  stats.max_padding_bitrate_bps =
+      configured_max_padding_bitrate_bps_.load(std::memory_order_relaxed);
+
+  return stats;
+}
+
+const FieldTrialsView& Call::trials() const {
+  return trials_;
+}
+
+TaskQueueBase* Call::network_thread() const {
+  return network_thread_;
+}
+
+TaskQueueBase* Call::worker_thread() const {
+  return worker_thread_;
+}
+
+void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  RTC_DCHECK(media == MediaType::AUDIO || media == MediaType::VIDEO);
+
+  auto closure = [this, media, state]() {
+    // TODO(bugs.webrtc.org/11993): Move this over to the network thread.
+    RTC_DCHECK_RUN_ON(worker_thread_);
+    if (media == MediaType::AUDIO) {
+      audio_network_state_ = state;
+    } else {
+      RTC_DCHECK_EQ(media, MediaType::VIDEO);
+      video_network_state_ = state;
+    }
+
+    // TODO(tommi): Is it necessary to always do this, including if there
+    // was no change in state?
+    UpdateAggregateNetworkState();
+
+    // TODO(tommi): Is it right to do this if media == AUDIO?
+    for (VideoReceiveStream2* video_receive_stream : video_receive_streams_) {
+      video_receive_stream->SignalNetworkState(video_network_state_);
+    }
+  };
+
+  if (network_thread_ == worker_thread_) {
+    closure();
+  } else {
+    // TODO(bugs.webrtc.org/11993): Remove workaround when we no longer need to
+    // post to the worker thread.
+    worker_thread_->PostTask(SafeTask(task_safety_.flag(), std::move(closure)));
+  }
+}
+
+void Call::OnAudioTransportOverheadChanged(int transport_overhead_per_packet) {
+  RTC_DCHECK_RUN_ON(network_thread_);
+  worker_thread_->PostTask(
+      SafeTask(task_safety_.flag(), [this, transport_overhead_per_packet]() {
+        // TODO(bugs.webrtc.org/11993): Move this over to the network thread.
+        RTC_DCHECK_RUN_ON(worker_thread_);
+        for (auto& kv : audio_send_ssrcs_) {
+          kv.second->SetTransportOverhead(transport_overhead_per_packet);
+        }
+      }));
+}
+
+void Call::UpdateAggregateNetworkState() {
+  // TODO(bugs.webrtc.org/11993): Move this over to the network thread.
+  // RTC_DCHECK_RUN_ON(network_thread_);
+
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  bool have_audio =
+      !audio_send_ssrcs_.empty() || !audio_receive_streams_.empty();
+  bool have_video =
+      !video_send_ssrcs_.empty() || !video_receive_streams_.empty();
+
+  bool aggregate_network_up =
+      ((have_video && video_network_state_ == kNetworkUp) ||
+       (have_audio && audio_network_state_ == kNetworkUp));
+
+  if (aggregate_network_up != aggregate_network_up_) {
+    RTC_LOG(LS_INFO)
+        << "UpdateAggregateNetworkState: aggregate_state change to "
+        << (aggregate_network_up ? "up" : "down");
+  } else {
+    RTC_LOG(LS_VERBOSE)
+        << "UpdateAggregateNetworkState: aggregate_state remains at "
+        << (aggregate_network_up ? "up" : "down");
+  }
+  aggregate_network_up_ = aggregate_network_up;
+
+  transport_send_->OnNetworkAvailability(aggregate_network_up);
+}
+
+void Call::OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream,
+                              uint32_t local_ssrc) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  webrtc::AudioReceiveStreamImpl& receive_stream =
+      static_cast<webrtc::AudioReceiveStreamImpl&>(stream);
+
+  receive_stream.SetLocalSsrc(local_ssrc);
+  auto it = audio_send_ssrcs_.find(local_ssrc);
+  receive_stream.AssociateSendStream(it != audio_send_ssrcs_.end() ? it->second
+                                                                   : nullptr);
+}
+
+void Call::OnLocalSsrcUpdated(VideoReceiveStreamInterface& stream,
+                              uint32_t local_ssrc) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  static_cast<VideoReceiveStream2&>(stream).SetLocalSsrc(local_ssrc);
+}
+
+void Call::OnLocalSsrcUpdated(FlexfecReceiveStream& stream,
+                              uint32_t local_ssrc) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  static_cast<FlexfecReceiveStreamImpl&>(stream).SetLocalSsrc(local_ssrc);
+}
+
+void Call::OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream,
+                             absl::string_view sync_group) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  webrtc::AudioReceiveStreamImpl& receive_stream =
+      static_cast<webrtc::AudioReceiveStreamImpl&>(stream);
+  receive_stream.SetSyncGroup(sync_group);
+  ConfigureSync(sync_group);
+}
+
+void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
+  RTC_DCHECK_RUN_ON(&sent_packet_sequence_checker_);
+  // When bundling is in effect, multiple senders may be sharing the same
+  // transport. It means every |sent_packet| will be multiply notified from
+  // different channels, WebRtcVoiceMediaChannel or WebRtcVideoChannel. Record
+  // |last_sent_packet_| to deduplicate redundant notifications to downstream.
+  // (https://crbug.com/webrtc/13437): Pass all packets without a |packet_id| to
+  // downstream.
+  if (last_sent_packet_.has_value() && last_sent_packet_->packet_id != -1 &&
+      last_sent_packet_->packet_id == sent_packet.packet_id &&
+      last_sent_packet_->send_time_ms == sent_packet.send_time_ms) {
+    return;
+  }
+  last_sent_packet_ = sent_packet;
+
+  // In production and with most tests, this method will be called on the
+  // network thread. However some test classes such as DirectTransport don't
+  // incorporate a network thread. This means that tests for RtpSenderEgress
+  // and ModuleRtpRtcpImpl2 that use DirectTransport, will call this method
+  // on a ProcessThread. This is alright as is since we forward the call to
+  // implementations that either just do a PostTask or use locking.
+  video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
+                                        clock_->CurrentTime());
+  transport_send_->OnSentPacket(sent_packet);
+}
+
+void Call::OnStartRateUpdate(DataRate start_rate) {
+  RTC_DCHECK_RUN_ON(&send_transport_sequence_checker_);
+  bitrate_allocator_->UpdateStartRate(start_rate.bps<uint32_t>());
+}
+
+void Call::OnTargetTransferRate(TargetTransferRate msg) {
+  RTC_DCHECK_RUN_ON(&send_transport_sequence_checker_);
+
+  uint32_t target_bitrate_bps = msg.target_rate.bps();
+  // For controlling the rate of feedback messages.
+  receive_side_cc_.OnBitrateChanged(target_bitrate_bps);
+  bitrate_allocator_->OnNetworkEstimateChanged(msg);
+
+  last_bandwidth_bps_.store(target_bitrate_bps, std::memory_order_relaxed);
+
+  // Ignore updates if bitrate is zero (the aggregate network state is
+  // down) or if we're not sending video.
+  // Using `video_send_streams_empty_` is racy but as the caller can't
+  // reasonably expect synchronize with changes in `video_send_streams_` (being
+  // on `send_transport_sequence_checker`), we can avoid a PostTask this way.
+  if (target_bitrate_bps == 0 ||
+      video_send_streams_empty_.load(std::memory_order_relaxed)) {
+    send_stats_.PauseSendAndPacerBitrateCounters();
+  } else {
+    send_stats_.AddTargetBitrateSample(target_bitrate_bps);
+  }
+}
+
+void Call::OnAllocationLimitsChanged(BitrateAllocationLimits limits) {
+  RTC_DCHECK_RUN_ON(&send_transport_sequence_checker_);
+
+  transport_send_ptr_->SetAllocatedSendBitrateLimits(limits);
+  send_stats_.SetMinAllocatableRate(limits);
+  configured_max_padding_bitrate_bps_.store(limits.max_padding_rate.bps(),
+                                            std::memory_order_relaxed);
+}
+
+AudioReceiveStreamImpl* Call::FindAudioStreamForSyncGroup(
+    absl::string_view sync_group) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK_RUN_ON(&receive_11993_checker_);
+  if (!sync_group.empty()) {
+    for (AudioReceiveStreamImpl* stream : audio_receive_streams_) {
+      if (stream->sync_group() == sync_group)
+        return stream;
+    }
+  }
+
+  return nullptr;
+}
+
+void Call::ConfigureSync(absl::string_view sync_group) {
+  // TODO(bugs.webrtc.org/11993): Expect to be called on the network thread.
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  // `audio_stream` may be nullptr when clearing the audio stream for a group.
+  AudioReceiveStreamImpl* audio_stream =
+      FindAudioStreamForSyncGroup(sync_group);
+
+  size_t num_synced_streams = 0;
+  for (VideoReceiveStream2* video_stream : video_receive_streams_) {
+    if (video_stream->sync_group() != sync_group)
+      continue;
+    ++num_synced_streams;
+    // TODO(bugs.webrtc.org/4762): Support synchronizing more than one A/V pair.
+    // Attempting to sync more than one audio/video pair within the same sync
+    // group is not supported in the current implementation.
+    // Only sync the first A/V pair within this sync group.
+    if (num_synced_streams == 1) {
+      // sync_audio_stream may be null and that's ok.
+      video_stream->SetSync(audio_stream);
+    } else {
+      video_stream->SetSync(nullptr);
+    }
+  }
+}
+
+void Call::DeliverRtcpPacket(rtc::CopyOnWriteBuffer packet) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK(IsRtcpPacket(packet));
+  TRACE_EVENT0("webrtc", "Call::DeliverRtcp");
+
+  receive_stats_.AddReceivedRtcpBytes(static_cast<int>(packet.size()));
+  bool rtcp_delivered = false;
+  for (VideoReceiveStream2* stream : video_receive_streams_) {
+    if (stream->DeliverRtcp(packet.cdata(), packet.size()))
+      rtcp_delivered = true;
+  }
+
+  for (AudioReceiveStreamImpl* stream : audio_receive_streams_) {
+    stream->DeliverRtcp(packet.cdata(), packet.size());
+    rtcp_delivered = true;
+  }
+
+  for (VideoSendStream* stream : video_send_streams_) {
+    stream->DeliverRtcp(packet.cdata(), packet.size());
+    rtcp_delivered = true;
+  }
+
+  for (auto& kv : audio_send_ssrcs_) {
+    kv.second->DeliverRtcp(packet.cdata(), packet.size());
+    rtcp_delivered = true;
+  }
+
+  if (rtcp_delivered) {
+    event_log_->Log(std::make_unique<RtcEventRtcpPacketIncoming>(packet));
+  }
+}
+
+void Call::DeliverRtpPacket(
+    MediaType media_type,
+    RtpPacketReceived packet,
+    OnUndemuxablePacketHandler undemuxable_packet_handler) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+  RTC_DCHECK(packet.arrival_time().IsFinite());
+
+  if (receive_time_calculator_) {
+    int64_t packet_time_us = packet.arrival_time().us();
+    // Repair packet_time_us for clock resets by comparing a new read of
+    // the same clock (TimeUTCMicros) to a monotonic clock reading.
+    packet_time_us = receive_time_calculator_->ReconcileReceiveTimes(
+        packet_time_us, rtc::TimeUTCMicros(), clock_->TimeInMicroseconds());
+    packet.set_arrival_time(Timestamp::Micros(packet_time_us));
+  }
+
+  NotifyBweOfReceivedPacket(packet, media_type);
+
+  event_log_->Log(std::make_unique<RtcEventRtpPacketIncoming>(packet));
+  if (media_type != MediaType::AUDIO && media_type != MediaType::VIDEO) {
+    return;
+  }
+
+  RtpStreamReceiverController& receiver_controller =
+      media_type == MediaType::AUDIO ? audio_receiver_controller_
+                                     : video_receiver_controller_;
+
+  if (!receiver_controller.OnRtpPacket(packet)) {
+    // Demuxing failed.  Allow the caller to create a
+    // receive stream in order to handle unsignalled SSRCs and try again.
+    // Note that we dont want to call NotifyBweOfReceivedPacket twice per
+    // packet.
+    if (!undemuxable_packet_handler(packet)) {
+      return;
+    }
+    if (!receiver_controller.OnRtpPacket(packet)) {
+      RTC_LOG(LS_INFO) << "Failed to demux packet " << packet.Ssrc();
+      return;
+    }
+  }
+
+  // RateCounters expect input parameter as int, save it as int,
+  // instead of converting each time it is passed to RateCounter::Add below.
+  int length = static_cast<int>(packet.size());
+  if (media_type == MediaType::AUDIO) {
+    receive_stats_.AddReceivedAudioBytes(length, packet.arrival_time());
+  }
+  if (media_type == MediaType::VIDEO) {
+    receive_stats_.AddReceivedVideoBytes(length, packet.arrival_time());
+  }
+}
+
+void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
+                                     MediaType media_type) {
+  RTC_DCHECK_RUN_ON(worker_thread_);
+
+  ReceivedPacket packet_msg;
+  packet_msg.size = DataSize::Bytes(packet.payload_size());
+  packet_msg.receive_time = packet.arrival_time();
+  uint32_t time_24;
+  if (packet.GetExtension<AbsoluteSendTime>(&time_24)) {
+    packet_msg.send_time = AbsoluteSendTime::ToTimestamp(time_24);
+  }
+  transport_send_->OnReceivedPacket(packet_msg);
+
+  receive_side_cc_.OnReceivedPacket(packet, media_type);
+}
+
+}  // namespace internal
+
+}  // namespace webrtc