Repo created

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/rtp/control_handler.cc

@@ -0,0 +1,73 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/rtp/control_handler.h"
+
+#include <algorithm>
+#include <vector>
+
+#include "api/units/data_rate.h"
+#include "modules/pacing/pacing_controller.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/numerics/safe_conversions.h"
+#include "rtc_base/numerics/safe_minmax.h"
+
+namespace webrtc {
+
+void CongestionControlHandler::SetTargetRate(
+    TargetTransferRate new_target_rate) {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  RTC_CHECK(new_target_rate.at_time.IsFinite());
+  last_incoming_ = new_target_rate;
+}
+
+void CongestionControlHandler::SetNetworkAvailability(bool network_available) {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  network_available_ = network_available;
+}
+
+void CongestionControlHandler::SetPacerQueue(TimeDelta expected_queue_time) {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  pacer_expected_queue_ms_ = expected_queue_time.ms();
+}
+
+absl::optional<TargetTransferRate> CongestionControlHandler::GetUpdate() {
+  RTC_DCHECK_RUN_ON(&sequenced_checker_);
+  if (!last_incoming_.has_value())
+    return absl::nullopt;
+  TargetTransferRate new_outgoing = *last_incoming_;
+  DataRate log_target_rate = new_outgoing.target_rate;
+  bool pause_encoding = false;
+  if (!network_available_) {
+    pause_encoding = true;
+  } else if (pacer_expected_queue_ms_ >
+             PacingController::kMaxExpectedQueueLength.ms()) {
+    pause_encoding = true;
+  }
+  if (pause_encoding)
+    new_outgoing.target_rate = DataRate::Zero();
+  if (!last_reported_ ||
+      last_reported_->target_rate != new_outgoing.target_rate ||
+      (!new_outgoing.target_rate.IsZero() &&
+       (last_reported_->network_estimate.loss_rate_ratio !=
+            new_outgoing.network_estimate.loss_rate_ratio ||
+        last_reported_->network_estimate.round_trip_time !=
+            new_outgoing.network_estimate.round_trip_time))) {
+    if (encoder_paused_in_last_report_ != pause_encoding)
+      RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: "
+                       << ToString(log_target_rate) << ".";
+    encoder_paused_in_last_report_ = pause_encoding;
+    last_reported_ = new_outgoing;
+    return new_outgoing;
+  }
+  return absl::nullopt;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/rtp/control_handler.h

@@ -0,0 +1,54 @@
+/*
+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
+#define MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_
+
+#include <stdint.h>
+
+#include "absl/types/optional.h"
+#include "api/sequence_checker.h"
+#include "api/transport/network_types.h"
+#include "api/units/data_size.h"
+#include "api/units/time_delta.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace webrtc {
+// This is used to observe the network controller state and route calls to
+// the proper handler. It also keeps cached values for safe asynchronous use.
+// This makes sure that things running on the worker queue can't access state
+// in RtpTransportControllerSend, which would risk causing data race on
+// destruction unless members are properly ordered.
+class CongestionControlHandler {
+ public:
+  CongestionControlHandler() = default;
+
+  CongestionControlHandler(const CongestionControlHandler&) = delete;
+  CongestionControlHandler& operator=(const CongestionControlHandler&) = delete;
+
+  ~CongestionControlHandler() = default;
+
+  void SetTargetRate(TargetTransferRate new_target_rate);
+  void SetNetworkAvailability(bool network_available);
+  void SetPacerQueue(TimeDelta expected_queue_time);
+  absl::optional<TargetTransferRate> GetUpdate();
+
+ private:
+  absl::optional<TargetTransferRate> last_incoming_;
+  absl::optional<TargetTransferRate> last_reported_;
+  bool network_available_ = true;
+  bool encoder_paused_in_last_report_ = false;
+
+  int64_t pacer_expected_queue_ms_ = 0;
+
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker sequenced_checker_;
+};
+}  // namespace webrtc
+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_CONTROL_HANDLER_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc

@@ -0,0 +1,275 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
+
+#include <stdlib.h>
+
+#include <algorithm>
+#include <cmath>
+#include <utility>
+
+#include "absl/algorithm/container.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+
+namespace webrtc {
+
+constexpr TimeDelta kSendTimeHistoryWindow = TimeDelta::Seconds(60);
+
+void InFlightBytesTracker::AddInFlightPacketBytes(
+    const PacketFeedback& packet) {
+  RTC_DCHECK(packet.sent.send_time.IsFinite());
+  auto it = in_flight_data_.find(packet.network_route);
+  if (it != in_flight_data_.end()) {
+    it->second += packet.sent.size;
+  } else {
+    in_flight_data_.insert({packet.network_route, packet.sent.size});
+  }
+}
+
+void InFlightBytesTracker::RemoveInFlightPacketBytes(
+    const PacketFeedback& packet) {
+  if (packet.sent.send_time.IsInfinite())
+    return;
+  auto it = in_flight_data_.find(packet.network_route);
+  if (it != in_flight_data_.end()) {
+    RTC_DCHECK_GE(it->second, packet.sent.size);
+    it->second -= packet.sent.size;
+    if (it->second.IsZero())
+      in_flight_data_.erase(it);
+  }
+}
+
+DataSize InFlightBytesTracker::GetOutstandingData(
+    const rtc::NetworkRoute& network_route) const {
+  auto it = in_flight_data_.find(network_route);
+  if (it != in_flight_data_.end()) {
+    return it->second;
+  } else {
+    return DataSize::Zero();
+  }
+}
+
+// Comparator for consistent map with NetworkRoute as key.
+bool InFlightBytesTracker::NetworkRouteComparator::operator()(
+    const rtc::NetworkRoute& a,
+    const rtc::NetworkRoute& b) const {
+  if (a.local.network_id() != b.local.network_id())
+    return a.local.network_id() < b.local.network_id();
+  if (a.remote.network_id() != b.remote.network_id())
+    return a.remote.network_id() < b.remote.network_id();
+
+  if (a.local.adapter_id() != b.local.adapter_id())
+    return a.local.adapter_id() < b.local.adapter_id();
+  if (a.remote.adapter_id() != b.remote.adapter_id())
+    return a.remote.adapter_id() < b.remote.adapter_id();
+
+  if (a.local.uses_turn() != b.local.uses_turn())
+    return a.local.uses_turn() < b.local.uses_turn();
+  if (a.remote.uses_turn() != b.remote.uses_turn())
+    return a.remote.uses_turn() < b.remote.uses_turn();
+
+  return a.connected < b.connected;
+}
+
+TransportFeedbackAdapter::TransportFeedbackAdapter() = default;
+
+void TransportFeedbackAdapter::AddPacket(const RtpPacketSendInfo& packet_info,
+                                         size_t overhead_bytes,
+                                         Timestamp creation_time) {
+  PacketFeedback packet;
+  packet.creation_time = creation_time;
+  packet.sent.sequence_number =
+      seq_num_unwrapper_.Unwrap(packet_info.transport_sequence_number);
+  packet.sent.size = DataSize::Bytes(packet_info.length + overhead_bytes);
+  packet.sent.audio = packet_info.packet_type == RtpPacketMediaType::kAudio;
+  packet.network_route = network_route_;
+  packet.sent.pacing_info = packet_info.pacing_info;
+
+  while (!history_.empty() &&
+         creation_time - history_.begin()->second.creation_time >
+             kSendTimeHistoryWindow) {
+    // TODO(sprang): Warn if erasing (too many) old items?
+    if (history_.begin()->second.sent.sequence_number > last_ack_seq_num_)
+      in_flight_.RemoveInFlightPacketBytes(history_.begin()->second);
+    history_.erase(history_.begin());
+  }
+  history_.insert(std::make_pair(packet.sent.sequence_number, packet));
+}
+
+absl::optional<SentPacket> TransportFeedbackAdapter::ProcessSentPacket(
+    const rtc::SentPacket& sent_packet) {
+  auto send_time = Timestamp::Millis(sent_packet.send_time_ms);
+  // TODO(srte): Only use one way to indicate that packet feedback is used.
+  if (sent_packet.info.included_in_feedback || sent_packet.packet_id != -1) {
+    int64_t unwrapped_seq_num =
+        seq_num_unwrapper_.Unwrap(sent_packet.packet_id);
+    auto it = history_.find(unwrapped_seq_num);
+    if (it != history_.end()) {
+      bool packet_retransmit = it->second.sent.send_time.IsFinite();
+      it->second.sent.send_time = send_time;
+      last_send_time_ = std::max(last_send_time_, send_time);
+      // TODO(srte): Don't do this on retransmit.
+      if (!pending_untracked_size_.IsZero()) {
+        if (send_time < last_untracked_send_time_)
+          RTC_LOG(LS_WARNING)
+              << "appending acknowledged data for out of order packet. (Diff: "
+              << ToString(last_untracked_send_time_ - send_time) << " ms.)";
+        it->second.sent.prior_unacked_data += pending_untracked_size_;
+        pending_untracked_size_ = DataSize::Zero();
+      }
+      if (!packet_retransmit) {
+        if (it->second.sent.sequence_number > last_ack_seq_num_)
+          in_flight_.AddInFlightPacketBytes(it->second);
+        it->second.sent.data_in_flight = GetOutstandingData();
+        return it->second.sent;
+      }
+    }
+  } else if (sent_packet.info.included_in_allocation) {
+    if (send_time < last_send_time_) {
+      RTC_LOG(LS_WARNING) << "ignoring untracked data for out of order packet.";
+    }
+    pending_untracked_size_ +=
+        DataSize::Bytes(sent_packet.info.packet_size_bytes);
+    last_untracked_send_time_ = std::max(last_untracked_send_time_, send_time);
+  }
+  return absl::nullopt;
+}
+
+absl::optional<TransportPacketsFeedback>
+TransportFeedbackAdapter::ProcessTransportFeedback(
+    const rtcp::TransportFeedback& feedback,
+    Timestamp feedback_receive_time) {
+  if (feedback.GetPacketStatusCount() == 0) {
+    RTC_LOG(LS_INFO) << "Empty transport feedback packet received.";
+    return absl::nullopt;
+  }
+
+  TransportPacketsFeedback msg;
+  msg.feedback_time = feedback_receive_time;
+
+  msg.prior_in_flight = in_flight_.GetOutstandingData(network_route_);
+  msg.packet_feedbacks =
+      ProcessTransportFeedbackInner(feedback, feedback_receive_time);
+  if (msg.packet_feedbacks.empty())
+    return absl::nullopt;
+
+  auto it = history_.find(last_ack_seq_num_);
+  if (it != history_.end()) {
+    msg.first_unacked_send_time = it->second.sent.send_time;
+  }
+  msg.data_in_flight = in_flight_.GetOutstandingData(network_route_);
+
+  return msg;
+}
+
+void TransportFeedbackAdapter::SetNetworkRoute(
+    const rtc::NetworkRoute& network_route) {
+  network_route_ = network_route;
+}
+
+DataSize TransportFeedbackAdapter::GetOutstandingData() const {
+  return in_flight_.GetOutstandingData(network_route_);
+}
+
+std::vector<PacketResult>
+TransportFeedbackAdapter::ProcessTransportFeedbackInner(
+    const rtcp::TransportFeedback& feedback,
+    Timestamp feedback_receive_time) {
+  // Add timestamp deltas to a local time base selected on first packet arrival.
+  // This won't be the true time base, but makes it easier to manually inspect
+  // time stamps.
+  if (last_timestamp_.IsInfinite()) {
+    current_offset_ = feedback_receive_time;
+  } else {
+    // TODO(srte): We shouldn't need to do rounding here.
+    const TimeDelta delta = feedback.GetBaseDelta(last_timestamp_)
+                                .RoundDownTo(TimeDelta::Millis(1));
+    // Protect against assigning current_offset_ negative value.
+    if (delta < Timestamp::Zero() - current_offset_) {
+      RTC_LOG(LS_WARNING) << "Unexpected feedback timestamp received.";
+      current_offset_ = feedback_receive_time;
+    } else {
+      current_offset_ += delta;
+    }
+  }
+  last_timestamp_ = feedback.BaseTime();
+
+  std::vector<PacketResult> packet_result_vector;
+  packet_result_vector.reserve(feedback.GetPacketStatusCount());
+
+  size_t failed_lookups = 0;
+  size_t ignored = 0;
+
+  feedback.ForAllPackets(
+      [&](uint16_t sequence_number, TimeDelta delta_since_base) {
+        int64_t seq_num = seq_num_unwrapper_.Unwrap(sequence_number);
+
+        if (seq_num > last_ack_seq_num_) {
+          // Starts at history_.begin() if last_ack_seq_num_ < 0, since any
+          // valid sequence number is >= 0.
+          for (auto it = history_.upper_bound(last_ack_seq_num_);
+               it != history_.upper_bound(seq_num); ++it) {
+            in_flight_.RemoveInFlightPacketBytes(it->second);
+          }
+          last_ack_seq_num_ = seq_num;
+        }
+
+        auto it = history_.find(seq_num);
+        if (it == history_.end()) {
+          ++failed_lookups;
+          return;
+        }
+
+        if (it->second.sent.send_time.IsInfinite()) {
+          // TODO(srte): Fix the tests that makes this happen and make this a
+          // DCHECK.
+          RTC_DLOG(LS_ERROR)
+              << "Received feedback before packet was indicated as sent";
+          return;
+        }
+
+        PacketFeedback packet_feedback = it->second;
+        if (delta_since_base.IsFinite()) {
+          packet_feedback.receive_time =
+              current_offset_ +
+              delta_since_base.RoundDownTo(TimeDelta::Millis(1));
+          // Note: Lost packets are not removed from history because they might
+          // be reported as received by a later feedback.
+          history_.erase(it);
+        }
+        if (packet_feedback.network_route == network_route_) {
+          PacketResult result;
+          result.sent_packet = packet_feedback.sent;
+          result.receive_time = packet_feedback.receive_time;
+          packet_result_vector.push_back(result);
+        } else {
+          ++ignored;
+        }
+      });
+
+  if (failed_lookups > 0) {
+    RTC_LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
+                        << " packet" << (failed_lookups > 1 ? "s" : "")
+                        << ". Send time history too small?";
+  }
+  if (ignored > 0) {
+    RTC_LOG(LS_INFO) << "Ignoring " << ignored
+                     << " packets because they were sent on a different route.";
+  }
+
+  return packet_result_vector;
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h

@@ -0,0 +1,102 @@
+/*
+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
+#define MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
+
+#include <deque>
+#include <map>
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "api/transport/network_types.h"
+#include "api/units/timestamp.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/network/sent_packet.h"
+#include "rtc_base/network_route.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/thread_annotations.h"
+
+namespace webrtc {
+
+struct PacketFeedback {
+  PacketFeedback() = default;
+  // Time corresponding to when this object was created.
+  Timestamp creation_time = Timestamp::MinusInfinity();
+  SentPacket sent;
+  // Time corresponding to when the packet was received. Timestamped with the
+  // receiver's clock. For unreceived packet, Timestamp::PlusInfinity() is
+  // used.
+  Timestamp receive_time = Timestamp::PlusInfinity();
+
+  // The network route that this packet is associated with.
+  rtc::NetworkRoute network_route;
+};
+
+class InFlightBytesTracker {
+ public:
+  void AddInFlightPacketBytes(const PacketFeedback& packet);
+  void RemoveInFlightPacketBytes(const PacketFeedback& packet);
+  DataSize GetOutstandingData(const rtc::NetworkRoute& network_route) const;
+
+ private:
+  struct NetworkRouteComparator {
+    bool operator()(const rtc::NetworkRoute& a,
+                    const rtc::NetworkRoute& b) const;
+  };
+  std::map<rtc::NetworkRoute, DataSize, NetworkRouteComparator> in_flight_data_;
+};
+
+class TransportFeedbackAdapter {
+ public:
+  TransportFeedbackAdapter();
+
+  void AddPacket(const RtpPacketSendInfo& packet_info,
+                 size_t overhead_bytes,
+                 Timestamp creation_time);
+  absl::optional<SentPacket> ProcessSentPacket(
+      const rtc::SentPacket& sent_packet);
+
+  absl::optional<TransportPacketsFeedback> ProcessTransportFeedback(
+      const rtcp::TransportFeedback& feedback,
+      Timestamp feedback_receive_time);
+
+  void SetNetworkRoute(const rtc::NetworkRoute& network_route);
+
+  DataSize GetOutstandingData() const;
+
+ private:
+  enum class SendTimeHistoryStatus { kNotAdded, kOk, kDuplicate };
+
+  std::vector<PacketResult> ProcessTransportFeedbackInner(
+      const rtcp::TransportFeedback& feedback,
+      Timestamp feedback_receive_time);
+
+  DataSize pending_untracked_size_ = DataSize::Zero();
+  Timestamp last_send_time_ = Timestamp::MinusInfinity();
+  Timestamp last_untracked_send_time_ = Timestamp::MinusInfinity();
+  RtpSequenceNumberUnwrapper seq_num_unwrapper_;
+  std::map<int64_t, PacketFeedback> history_;
+
+  // Sequence numbers are never negative, using -1 as it always < a real
+  // sequence number.
+  int64_t last_ack_seq_num_ = -1;
+  InFlightBytesTracker in_flight_;
+
+  Timestamp current_offset_ = Timestamp::MinusInfinity();
+  Timestamp last_timestamp_ = Timestamp::MinusInfinity();
+
+  rtc::NetworkRoute network_route_;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.cc

@@ -0,0 +1,95 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#include "modules/congestion_controller/rtp/transport_feedback_demuxer.h"
+
+#include "absl/algorithm/container.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+
+namespace webrtc {
+namespace {
+static const size_t kMaxPacketsInHistory = 5000;
+}
+
+TransportFeedbackDemuxer::TransportFeedbackDemuxer() {
+  // In case the construction thread is different from where the registration
+  // and callbacks occur, detach from the construction thread.
+  observer_checker_.Detach();
+}
+
+void TransportFeedbackDemuxer::RegisterStreamFeedbackObserver(
+    std::vector<uint32_t> ssrcs,
+    StreamFeedbackObserver* observer) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+  RTC_DCHECK(observer);
+  RTC_DCHECK(absl::c_find_if(observers_, [=](const auto& pair) {
+               return pair.second == observer;
+             }) == observers_.end());
+  observers_.push_back({ssrcs, observer});
+}
+
+void TransportFeedbackDemuxer::DeRegisterStreamFeedbackObserver(
+    StreamFeedbackObserver* observer) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+  RTC_DCHECK(observer);
+  const auto it = absl::c_find_if(
+      observers_, [=](const auto& pair) { return pair.second == observer; });
+  RTC_DCHECK(it != observers_.end());
+  observers_.erase(it);
+}
+
+void TransportFeedbackDemuxer::AddPacket(const RtpPacketSendInfo& packet_info) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+
+  StreamFeedbackObserver::StreamPacketInfo info;
+  info.ssrc = packet_info.media_ssrc;
+  info.rtp_sequence_number = packet_info.rtp_sequence_number;
+  info.received = false;
+  info.is_retransmission =
+      packet_info.packet_type == RtpPacketMediaType::kRetransmission;
+  history_.insert(
+      {seq_num_unwrapper_.Unwrap(packet_info.transport_sequence_number), info});
+
+  while (history_.size() > kMaxPacketsInHistory) {
+    history_.erase(history_.begin());
+  }
+}
+
+void TransportFeedbackDemuxer::OnTransportFeedback(
+    const rtcp::TransportFeedback& feedback) {
+  RTC_DCHECK_RUN_ON(&observer_checker_);
+
+  std::vector<StreamFeedbackObserver::StreamPacketInfo> stream_feedbacks;
+  feedback.ForAllPackets(
+      [&](uint16_t sequence_number, TimeDelta delta_since_base) {
+        RTC_DCHECK_RUN_ON(&observer_checker_);
+        auto it = history_.find(seq_num_unwrapper_.PeekUnwrap(sequence_number));
+        if (it != history_.end()) {
+          auto packet_info = it->second;
+          packet_info.received = delta_since_base.IsFinite();
+          stream_feedbacks.push_back(std::move(packet_info));
+          if (delta_since_base.IsFinite())
+            history_.erase(it);
+        }
+      });
+
+  for (auto& observer : observers_) {
+    std::vector<StreamFeedbackObserver::StreamPacketInfo> selected_feedback;
+    for (const auto& packet_info : stream_feedbacks) {
+      if (absl::c_count(observer.first, packet_info.ssrc) > 0) {
+        selected_feedback.push_back(packet_info);
+      }
+    }
+    if (!selected_feedback.empty()) {
+      observer.second->OnPacketFeedbackVector(std::move(selected_feedback));
+    }
+  }
+}
+
+}  // namespace webrtc

TMessagesProj/jni/voip/webrtc/modules/congestion_controller/rtp/transport_feedback_demuxer.h

@@ -0,0 +1,62 @@
+/*
+ *  Copyright (c) 2019 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_
+#define MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_
+
+#include <map>
+#include <utility>
+#include <vector>
+
+#include "api/sequence_checker.h"
+#include "modules/include/module_common_types_public.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "rtc_base/numerics/sequence_number_unwrapper.h"
+#include "rtc_base/system/no_unique_address.h"
+
+namespace webrtc {
+
+// Implementation of StreamFeedbackProvider that provides a way for
+// implementations of StreamFeedbackObserver to register for feedback callbacks
+// for a given set of SSRCs.
+// Registration methods need to be called from the same execution context
+// (thread or task queue) and callbacks to
+// StreamFeedbackObserver::OnPacketFeedbackVector will be made in that same
+// context.
+// TODO(tommi): This appears to be the only implementation of this interface.
+// Do we need the interface?
+class TransportFeedbackDemuxer final : public StreamFeedbackProvider {
+ public:
+  TransportFeedbackDemuxer();
+
+  // Implements StreamFeedbackProvider interface
+  void RegisterStreamFeedbackObserver(
+      std::vector<uint32_t> ssrcs,
+      StreamFeedbackObserver* observer) override;
+  void DeRegisterStreamFeedbackObserver(
+      StreamFeedbackObserver* observer) override;
+  void AddPacket(const RtpPacketSendInfo& packet_info);
+  void OnTransportFeedback(const rtcp::TransportFeedback& feedback);
+
+ private:
+  RTC_NO_UNIQUE_ADDRESS SequenceChecker observer_checker_;
+  RtpSequenceNumberUnwrapper seq_num_unwrapper_
+      RTC_GUARDED_BY(&observer_checker_);
+  std::map<int64_t, StreamFeedbackObserver::StreamPacketInfo> history_
+      RTC_GUARDED_BY(&observer_checker_);
+
+  // Maps a set of ssrcs to corresponding observer. Vectors are used rather than
+  // set/map to ensure that the processing order is consistent independently of
+  // the randomized ssrcs.
+  std::vector<std::pair<std::vector<uint32_t>, StreamFeedbackObserver*>>
+      observers_ RTC_GUARDED_BY(&observer_checker_);
+};
+}  // namespace webrtc
+
+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_DEMUXER_H_