call/call.cc

/*
 *  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 <optional>
#include <set>
#include <string>
#include <utility>
#include <vector>

#include "absl/functional/bind_front.h"
#include "absl/strings/string_view.h"
#include "api/adaptation/resource.h"
#include "api/environment/environment.h"
#include "api/fec_controller.h"
#include "api/field_trials_view.h"
#include "api/media_types.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log/rtc_event_log.h"
#include "api/rtp_headers.h"
#include "api/scoped_refptr.h"
#include "api/sequence_checker.h"
#include "api/task_queue/pending_task_safety_flag.h"
#include "api/task_queue/task_queue_base.h"
#include "api/transport/bitrate_settings.h"
#include "api/transport/network_control.h"
#include "api/transport/network_types.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.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/call_config.h"
#include "call/flexfec_receive_stream.h"
#include "call/flexfec_receive_stream_impl.h"
#include "call/packet_receiver.h"
#include "call/payload_type.h"
#include "call/payload_type_picker.h"
#include "call/receive_stream.h"
#include "call/receive_time_calculator.h"
#include "call/rtp_config.h"
#include "call/rtp_stream_receiver_controller.h"
#include "call/rtp_transport_controller_send_factory.h"
#include "call/version.h"
#include "call/video_receive_stream.h"
#include "call/video_send_stream.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 "media/base/codec.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_rtcp_defines.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 "modules/video_coding/nack_requester.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/logging.h"
#include "rtc_base/network/sent_packet.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.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/config/video_encoder_config.h"
#include "video/decode_synchronizer.h"
#include "video/send_delay_stats.h"
#include "video/stats_counter.h"
#include "video/video_receive_stream2.h"
#include "video/video_send_stream_impl.h"

namespace webrtc {

namespace {

// In normal operation, the PTS comes from the PeerConnection.
// However, it is too much of a bother to insert it in all tests,
// so defaulting here.
class PayloadTypeSuggesterForTests : public PayloadTypeSuggester {
 public:
  PayloadTypeSuggesterForTests() = default;
  RTCErrorOr<PayloadType> SuggestPayloadType(const std::string& /* mid */,
                                             cricket::Codec codec) override {
    return payload_type_picker_.SuggestMapping(codec, nullptr);
  }
  RTCError AddLocalMapping(const std::string& /* mid */,
                           PayloadType /* payload_type */,
                           const cricket::Codec& /* codec */) override {
    return RTCError::OK();
  }

 private:
  PayloadTypePicker payload_type_picker_;
};

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(CallConfig config,
       std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
  ~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;

  PayloadTypeSuggester* GetPayloadTypeSuggester() override;
  void SetPayloadTypeSuggester(PayloadTypeSuggester* suggester) override;

  Stats GetStats() const override;

  void EnableSendCongestionControlFeedbackAccordingToRfc8888() override;
  int FeedbackAccordingToRfc8888Count() override;
  int FeedbackAccordingToTransportCcCount() 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_);
    std::optional<Timestamp> first_received_rtp_audio_timestamp_
        RTC_GUARDED_BY(sequence_checker_);
    std::optional<Timestamp> last_received_rtp_audio_timestamp_
        RTC_GUARDED_BY(sequence_checker_);
    std::optional<Timestamp> first_received_rtp_video_timestamp_
        RTC_GUARDED_BY(sequence_checker_);
    std::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(std::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};
    std::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_);

  const Environment env_;
  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_);

  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, VideoSendStreamImpl*> video_send_ssrcs_
      RTC_GUARDED_BY(worker_thread_);
  std::set<VideoSendStreamImpl*> 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_);

  // 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_;
  RepeatingTaskHandle elastic_bandwidth_allocation_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_);

  bool is_started_ RTC_GUARDED_BY(worker_thread_) = false;

  // Mechanism for proposing payload types in RTP mappings.
  PayloadTypeSuggester* pt_suggester_ = nullptr;
  std::unique_ptr<PayloadTypeSuggesterForTests> owned_pt_suggester_;

  // Sequence checker for outgoing network traffic. Could be the network thread.
  // Could also be a pacer owned thread or TQ such as the TaskQueueSender.
  RTC_NO_UNIQUE_ADDRESS SequenceChecker sent_packet_sequence_checker_;
  std::optional<rtc::SentPacket> last_sent_packet_
      RTC_GUARDED_BY(sent_packet_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_;
};
}  // namespace internal

std::string Call::Stats::ToString(int64_t time_ms) const {
  char buf[1024];
  rtc::SimpleStringBuilder ss(buf);
  ss << "Call stats: " << time_ms << ", {";
  ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
  ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
  ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
  ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
  ss << "rtt_ms: " << rtt_ms;
  ss << '}';
  return ss.str();
}

std::unique_ptr<Call> Call::Create(CallConfig config) {
  std::unique_ptr<RtpTransportControllerSendInterface> transport_send;
  if (config.rtp_transport_controller_send_factory != nullptr) {
    transport_send = config.rtp_transport_controller_send_factory->Create(
        config.ExtractTransportConfig());
  } else {
    transport_send = RtpTransportControllerSendFactory().Create(
        config.ExtractTransportConfig());
  }

  return std::make_unique<internal::Call>(std::move(config),
                                          std::move(transport_send));
}

// 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(
    std::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(CallConfig config,
           std::unique_ptr<RtpTransportControllerSendInterface> transport_send)
    : env_(config.env),
      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.decode_metronome
              ? std::make_unique<DecodeSynchronizer>(&env_.clock(),
                                                     config.decode_metronome,
                                                     worker_thread_)
              : nullptr),
      num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
      call_stats_(new CallStats(&env_.clock(), worker_thread_)),
      bitrate_allocator_(new BitrateAllocator(
          this,
          GetElasticRateAllocationFieldTrialParameter(env_.field_trials()))),
      config_(std::move(config)),
      audio_network_state_(kNetworkDown),
      video_network_state_(kNetworkDown),
      aggregate_network_up_(false),
      receive_stats_(&env_.clock()),
      send_stats_(&env_.clock()),
      receive_side_cc_(env_,
                       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(env_.field_trials())),
      video_send_delay_stats_(new SendDelayStats(&env_.clock())),
      start_of_call_(env_.clock().CurrentTime()),
      transport_send_ptr_(transport_send.get()),
      transport_send_(std::move(transport_send)) {
  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, &env_.clock());

  // TODO(b/350555527): Remove after experiment
  if (GetElasticRateAllocationFieldTrialParameter(env_.field_trials()) !=
      DataRate::Zero()) {
    elastic_bandwidth_allocation_task_ = RepeatingTaskHandle::Start(
        worker_thread_,
        [this] {
          TimeDelta next_schedule_interval = TimeDelta::Millis(25);
          if (bitrate_allocator_) {
            if (!bitrate_allocator_->RecomputeAllocationIfNeeded())
              next_schedule_interval = TimeDelta::Millis(300);
          }
          return next_schedule_interval;
        },
        TaskQueueBase::DelayPrecision::kLow, &env_.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();
  elastic_bandwidth_allocation_task_.Stop();
  call_stats_->DeregisterStatsObserver(&receive_side_cc_);
  send_stats_.SetFirstPacketTime(transport_send_->GetFirstPacketTime());

  RTC_HISTOGRAM_COUNTS_100000(
      "WebRTC.Call.LifetimeInSeconds",
      (env_.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.
  std::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(env_, config, config_.audio_state,
                          transport_send_.get(), bitrate_allocator_.get(),
                          call_stats_->AsRtcpRttStats(), suspended_rtp_state);
  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();
  env_.event_log().Log(std::make_unique<RtcEventAudioReceiveStreamConfig>(
      CreateRtcLogStreamConfig(config)));

  AudioReceiveStreamImpl* receive_stream = new AudioReceiveStreamImpl(
      env_, transport_send_->packet_router(), config_.neteq_factory, config,
      config_.audio_state);
  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) {
    env_.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;

  VideoSendStreamImpl* send_stream = new VideoSendStreamImpl(
      env_, num_cpu_cores_, call_stats_->AsRtcpRttStats(),
      transport_send_.get(), config_.encode_metronome, bitrate_allocator_.get(),
      video_send_delay_stats_.get(), std::move(config),
      std::move(encoder_config), suspended_video_send_ssrcs_,
      suspended_video_payload_states_, std::move(fec_controller));

  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(env_)
          : std::make_unique<FecControllerDefault>(env_);
  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_);

  VideoSendStreamImpl* send_stream_impl =
      static_cast<VideoSendStreamImpl*>(send_stream);

  auto it = video_send_ssrcs_.begin();
  while (it != video_send_ssrcs_.end()) {
    if (it->second == static_cast<VideoSendStreamImpl*>(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);

  VideoSendStreamImpl::RtpStateMap rtp_states;
  VideoSendStreamImpl::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();

  env_.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(
      env_, this, num_cpu_cores_, transport_send_->packet_router(),
      std::move(configuration), call_stats_.get(),
      std::make_unique<VCMTiming>(&env_.clock(), trials()),
      &nack_periodic_processor_, decode_sync_.get());
  // 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(
      env_, 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();
}

PayloadTypeSuggester* Call::GetPayloadTypeSuggester() {
  // TODO: https://issues.webrtc.org/360058654 - make mandatory at
  // initialization. Currently, only some channels use it.
  RTC_DCHECK_RUN_ON(worker_thread_);
  if (!pt_suggester_) {
    // Make something that will work most of the time for testing.
    owned_pt_suggester_ = std::make_unique<PayloadTypeSuggesterForTests>();
    SetPayloadTypeSuggester(owned_pt_suggester_.get());
  }
  return pt_suggester_;
}

void Call::SetPayloadTypeSuggester(PayloadTypeSuggester* suggester) {
  RTC_CHECK(!pt_suggester_)
      << "SetPayloadTypeSuggester can be called only once";
  pt_suggester_ = suggester;
}

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;
}

void Call::EnableSendCongestionControlFeedbackAccordingToRfc8888() {
  receive_side_cc_.EnableSendCongestionControlFeedbackAccordingToRfc8888();
}

int Call::FeedbackAccordingToRfc8888Count() {
  return transport_send_->ReceivedCongestionControlFeedbackCount();
}

int Call::FeedbackAccordingToTransportCcCount() {
  return transport_send_->ReceivedTransportCcFeedbackCount();
}

const FieldTrialsView& Call::trials() const {
  return env_.field_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,
                                        env_.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 (VideoSendStreamImpl* 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) {
    env_.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(),
        env_.clock().TimeInMicroseconds());
    packet.set_arrival_time(Timestamp::Micros(packet_time_us));
  }

  NotifyBweOfReceivedPacket(packet, media_type);

  env_.event_log().Log(std::make_unique<RtcEventRtpPacketIncoming>(packet));
  if (media_type != MediaType::AUDIO && media_type != MediaType::VIDEO) {
    return;
  }

  const TimeDelta nw_to_deliver_delay =
      env_.clock().CurrentTime() - packet.arrival_time();
  RTC_HISTOGRAM_COUNTS_100000("WebRTC.TimeFromNetworkToDeliverRtpPacketUs",
                              nw_to_deliver_delay.us());

  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