[fix] fix qos module

src/common/api/function_view.h

@@ -0,0 +1,127 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_FUNCTION_VIEW_H_
+#define API_FUNCTION_VIEW_H_
+
+#include <cstddef>
+#include <type_traits>
+#include <utility>
+
+// Just like std::function, FunctionView will wrap any callable and hide its
+// actual type, exposing only its signature. But unlike std::function,
+// FunctionView doesn't own its callable---it just points to it. Thus, it's a
+// good choice mainly as a function argument when the callable argument will
+// not be called again once the function has returned.
+//
+// Its constructors are implicit, so that callers won't have to convert lambdas
+// and other callables to FunctionView<Blah(Blah, Blah)> explicitly. This is
+// safe because FunctionView is only a reference to the real callable.
+//
+// Example use:
+//
+//   void SomeFunction(rtc::FunctionView<int(int)> index_transform);
+//   ...
+//   SomeFunction([](int i) { return 2 * i + 1; });
+//
+// Note: FunctionView is tiny (essentially just two pointers) and trivially
+// copyable, so it's probably cheaper to pass it by value than by const
+// reference.
+
+namespace rtc {
+
+template <typename T>
+class FunctionView;  // Undefined.
+
+template <typename RetT, typename... ArgT>
+class FunctionView<RetT(ArgT...)> final {
+ public:
+  // Constructor for lambdas and other callables; it accepts every type of
+  // argument except those noted in its enable_if call.
+  template <
+      typename F,
+      typename std::enable_if<
+          // Not for function pointers; we have another constructor for that
+          // below.
+          !std::is_function<typename std::remove_pointer<
+              typename std::remove_reference<F>::type>::type>::value &&
+
+          // Not for nullptr; we have another constructor for that below.
+          !std::is_same<std::nullptr_t,
+                        typename std::remove_cv<F>::type>::value &&
+
+          // Not for FunctionView objects; we have another constructor for that
+          // (the implicitly declared copy constructor).
+          !std::is_same<FunctionView,
+                        typename std::remove_cv<typename std::remove_reference<
+                            F>::type>::type>::value>::type* = nullptr>
+  FunctionView(F&& f)
+      : call_(CallVoidPtr<typename std::remove_reference<F>::type>) {
+    f_.void_ptr = &f;
+  }
+
+  // Constructor that accepts function pointers. If the argument is null, the
+  // result is an empty FunctionView.
+  template <
+      typename F,
+      typename std::enable_if<std::is_function<typename std::remove_pointer<
+          typename std::remove_reference<F>::type>::type>::value>::type* =
+          nullptr>
+  FunctionView(F&& f)
+      : call_(f ? CallFunPtr<typename std::remove_pointer<F>::type> : nullptr) {
+    f_.fun_ptr = reinterpret_cast<void (*)()>(f);
+  }
+
+  // Constructor that accepts nullptr. It creates an empty FunctionView.
+  template <typename F, typename std::enable_if<std::is_same<
+                            std::nullptr_t, typename std::remove_cv<F>::type>::
+                                                    value>::type* = nullptr>
+  FunctionView(F&& /* f */) : call_(nullptr) {}
+
+  // Default constructor. Creates an empty FunctionView.
+  FunctionView() : call_(nullptr) {}
+
+  RetT operator()(ArgT... args) const {
+    return call_(f_, std::forward<ArgT>(args)...);
+  }
+
+  // Returns true if we have a function, false if we don't (i.e., we're null).
+  explicit operator bool() const { return !!call_; }
+
+ private:
+  union VoidUnion {
+    void* void_ptr;
+    void (*fun_ptr)();
+  };
+
+  template <typename F>
+  static RetT CallVoidPtr(VoidUnion vu, ArgT... args) {
+    return (*static_cast<F*>(vu.void_ptr))(std::forward<ArgT>(args)...);
+  }
+  template <typename F>
+  static RetT CallFunPtr(VoidUnion vu, ArgT... args) {
+    return (reinterpret_cast<typename std::add_pointer<F>::type>(vu.fun_ptr))(
+        std::forward<ArgT>(args)...);
+  }
+
+  // A pointer to the callable thing, with type information erased. It's a
+  // union because we have to use separate types depending on if the callable
+  // thing is a function pointer or something else.
+  VoidUnion f_;
+
+  // Pointer to a dispatch function that knows the type of the callable thing
+  // that's stored in f_, and how to call it. A FunctionView object is empty
+  // (null) iff call_ is null.
+  RetT (*call_)(VoidUnion, ArgT...);
+};
+
+}  // namespace rtc
+
+#endif  // API_FUNCTION_VIEW_H_

src/common/api/transport/ecn_marking.h

@@ -7,13 +7,11 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
-#ifndef RTC_BASE_NETWORK_ECN_MARKING_H_
-#define RTC_BASE_NETWORK_ECN_MARKING_H_
 
-// // TODO: bugs.webrtc.org/42225697 - delete this file.
-#include "ecn_marking.h"
+#ifndef API_TRANSPORT_ECN_MARKING_H_
+#define API_TRANSPORT_ECN_MARKING_H_
 
-namespace rtc {
+namespace webrtc {
 
 // TODO: bugs.webrtc.org/42225697 - L4S support is slowly being developed.
 // Help is appreciated.
@@ -39,6 +37,6 @@ enum class EcnMarking {
   kCe = 3,      // Congestion experienced
 };
 
-}  // namespace rtc
+}  // namespace webrtc
 
-#endif  // RTC_BASE_NETWORK_ECN_MARKING_H_
+#endif  // API_TRANSPORT_ECN_MARKING_H_

src/common/api/transport/network_control.h

@@ -1,124 +0,0 @@
-/*
- *  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 API_TRANSPORT_NETWORK_CONTROL_H_
-#define API_TRANSPORT_NETWORK_CONTROL_H_
-
-#include <memory>
-#include <optional>
-
-#include "api/transport/network_types.h"
-#include "api/units/data_rate.h"
-#include "api/units/time_delta.h"
-
-namespace webrtc {
-
-class TargetTransferRateObserver {
- public:
-  virtual ~TargetTransferRateObserver() = default;
-  // Called to indicate target transfer rate as well as giving information about
-  // the current estimate of network parameters.
-  virtual void OnTargetTransferRate(TargetTransferRate) = 0;
-  // Called to provide updates to the expected target rate in case it changes
-  // before the first call to OnTargetTransferRate.
-  virtual void OnStartRateUpdate(DataRate) {}
-};
-
-// Configuration sent to factory create function. The parameters here are
-// optional to use for a network controller implementation.
-struct NetworkControllerConfig {
-  explicit NetworkControllerConfig() {}
-
-  // The initial constraints to start with, these can be changed at any later
-  // time by calls to OnTargetRateConstraints. Note that the starting rate
-  // has to be set initially to provide a starting state for the network
-  // controller, even though the field is marked as optional.
-  TargetRateConstraints constraints;
-  // Initial stream specific configuration, these are changed at any later time
-  // by calls to OnStreamsConfig.
-  StreamsConfig stream_based_config;
-};
-
-// NetworkControllerInterface is implemented by network controllers. A network
-// controller is a class that uses information about network state and traffic
-// to estimate network parameters such as round trip time and bandwidth. Network
-// controllers does not guarantee thread safety, the interface must be used in a
-// non-concurrent fashion.
-class NetworkControllerInterface {
- public:
-  virtual ~NetworkControllerInterface() = default;
-
-  // Called when network availabilty changes.
-  virtual NetworkControlUpdate OnNetworkAvailability(NetworkAvailability) = 0;
-  // Called when the receiving or sending endpoint changes address.
-  virtual NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange) = 0;
-  // Called periodically with a periodicy as specified by
-  // NetworkControllerFactoryInterface::GetProcessInterval.
-  virtual NetworkControlUpdate OnProcessInterval(ProcessInterval) = 0;
-  // Called when remotely calculated bitrate is received.
-  virtual NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport) = 0;
-  // Called round trip time has been calculated by protocol specific mechanisms.
-  virtual NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate) = 0;
-  // Called when a packet is sent on the network.
-  virtual NetworkControlUpdate OnSentPacket(SentPacket) = 0;
-  // Called when a packet is received from the remote client.
-  virtual NetworkControlUpdate OnReceivedPacket(ReceivedPacket) = 0;
-  // Called when the stream specific configuration has been updated.
-  virtual NetworkControlUpdate OnStreamsConfig(StreamsConfig) = 0;
-  // Called when target transfer rate constraints has been changed.
-  virtual NetworkControlUpdate OnTargetRateConstraints(
-      TargetRateConstraints) = 0;
-  // Called when a protocol specific calculation of packet loss has been made.
-  virtual NetworkControlUpdate OnTransportLossReport(TransportLossReport) = 0;
-  // Called with per packet feedback regarding receive time.
-  virtual NetworkControlUpdate OnTransportPacketsFeedback(
-      TransportPacketsFeedback) = 0;
-  // Called with network state estimate updates.
-  virtual NetworkControlUpdate OnNetworkStateEstimate(NetworkStateEstimate) = 0;
-};
-
-// NetworkControllerFactoryInterface is an interface for creating a network
-// controller.
-class NetworkControllerFactoryInterface {
- public:
-  virtual ~NetworkControllerFactoryInterface() = default;
-
-  // Used to create a new network controller, requires an observer to be
-  // provided to handle callbacks.
-  virtual std::unique_ptr<NetworkControllerInterface> Create(
-      NetworkControllerConfig config) = 0;
-  // Returns the interval by which the network controller expects
-  // OnProcessInterval calls.
-  virtual TimeDelta GetProcessInterval() const = 0;
-};
-
-// Under development, subject to change without notice.
-class NetworkStateEstimator {
- public:
-  // Gets the current best estimate according to the estimator.
-  virtual std::optional<NetworkStateEstimate> GetCurrentEstimate() = 0;
-  // Called with per packet feedback regarding receive time.
-  // Used when the NetworkStateEstimator runs in the sending endpoint.
-  virtual void OnTransportPacketsFeedback(const TransportPacketsFeedback&) = 0;
-  // Called with per packet feedback regarding receive time.
-  // Used when the NetworkStateEstimator runs in the receiving endpoint.
-  virtual void OnReceivedPacket(const PacketResult&) {}
-  // Called when the receiving or sending endpoint changes address.
-  virtual void OnRouteChange(const NetworkRouteChange&) = 0;
-  virtual ~NetworkStateEstimator() = default;
-};
-class NetworkStateEstimatorFactory {
- public:
-  virtual std::unique_ptr<NetworkStateEstimator> Create() = 0;
-  virtual ~NetworkStateEstimatorFactory() = default;
-};
-}  // namespace webrtc
-
-#endif  // API_TRANSPORT_NETWORK_CONTROL_H_

src/common/api/transport/network_types.h

@@ -16,11 +16,11 @@
 #include <optional>
 #include <vector>
 
+#include "api/transport/ecn_marking.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 "rtc_base/network/ecn_marking.h"
 
 namespace webrtc {
 
@@ -167,7 +167,7 @@ struct PacketResult {
 
   SentPacket sent_packet;
   Timestamp receive_time = Timestamp::PlusInfinity();
-  rtc::EcnMarking ecn = rtc::EcnMarking::kNotEct;
+  EcnMarking ecn = EcnMarking::kNotEct;
 };
 
 struct TransportPacketsFeedback {

src/common/api/video/video_timing.cc

@@ -0,0 +1,118 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/video/video_timing.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <string>
+
+#include "api/array_view.h"
+#include "api/units/time_delta.h"
+#include "log.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+uint16_t VideoSendTiming::GetDeltaCappedMs(int64_t base_ms, int64_t time_ms) {
+  if (time_ms < base_ms) {
+    LOG_ERROR("Delta {} ms expected to be positive", (time_ms - base_ms));
+  }
+  return rtc::saturated_cast<uint16_t>(time_ms - base_ms);
+}
+
+uint16_t VideoSendTiming::GetDeltaCappedMs(TimeDelta delta) {
+  if (delta < TimeDelta::Zero()) {
+    LOG_ERROR("Delta {} ms expected to be positive", delta.ms());
+  }
+  return rtc::saturated_cast<uint16_t>(delta.ms());
+}
+
+TimingFrameInfo::TimingFrameInfo()
+    : rtp_timestamp(0),
+      capture_time_ms(-1),
+      encode_start_ms(-1),
+      encode_finish_ms(-1),
+      packetization_finish_ms(-1),
+      pacer_exit_ms(-1),
+      network_timestamp_ms(-1),
+      network2_timestamp_ms(-1),
+      receive_start_ms(-1),
+      receive_finish_ms(-1),
+      decode_start_ms(-1),
+      decode_finish_ms(-1),
+      render_time_ms(-1),
+      flags(VideoSendTiming::kNotTriggered) {}
+
+int64_t TimingFrameInfo::EndToEndDelay() const {
+  return capture_time_ms >= 0 ? decode_finish_ms - capture_time_ms : -1;
+}
+
+bool TimingFrameInfo::IsLongerThan(const TimingFrameInfo& other) const {
+  int64_t other_delay = other.EndToEndDelay();
+  return other_delay == -1 || EndToEndDelay() > other_delay;
+}
+
+bool TimingFrameInfo::operator<(const TimingFrameInfo& other) const {
+  return other.IsLongerThan(*this);
+}
+
+bool TimingFrameInfo::operator<=(const TimingFrameInfo& other) const {
+  return !IsLongerThan(other);
+}
+
+bool TimingFrameInfo::IsOutlier() const {
+  return !IsInvalid() && (flags & VideoSendTiming::kTriggeredBySize);
+}
+
+bool TimingFrameInfo::IsTimerTriggered() const {
+  return !IsInvalid() && (flags & VideoSendTiming::kTriggeredByTimer);
+}
+
+bool TimingFrameInfo::IsInvalid() const {
+  return flags == VideoSendTiming::kInvalid;
+}
+
+std::string TimingFrameInfo::ToString() const {
+  if (IsInvalid()) {
+    return "";
+  }
+
+  std::ostringstream oss;
+  oss << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms
+      << ',' << encode_finish_ms << ',' << packetization_finish_ms << ','
+      << pacer_exit_ms << ',' << network_timestamp_ms << ','
+      << network2_timestamp_ms << ',' << receive_start_ms << ','
+      << receive_finish_ms << ',' << decode_start_ms << ',' << decode_finish_ms
+      << ',' << render_time_ms << ',' << IsOutlier() << ','
+      << IsTimerTriggered();
+
+  return oss.str();
+}
+
+VideoPlayoutDelay::VideoPlayoutDelay(TimeDelta min, TimeDelta max)
+    : min_(std::clamp(min, TimeDelta::Zero(), kMax)),
+      max_(std::clamp(max, min_, kMax)) {
+  if (!(TimeDelta::Zero() <= min && min <= max && max <= kMax)) {
+    LOG_ERROR("Invalid video playout delay: [{},{}]. Clamped to [{},{}]", min,
+              max, this->min(), this->max());
+  }
+}
+
+bool VideoPlayoutDelay::Set(TimeDelta min, TimeDelta max) {
+  if (TimeDelta::Zero() <= min && min <= max && max <= kMax) {
+    min_ = min;
+    max_ = max;
+    return true;
+  }
+  return false;
+}
+
+}  // namespace webrtc

src/common/api/video/video_timing.h

@@ -0,0 +1,149 @@
+/*
+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_VIDEO_VIDEO_TIMING_H_
+#define API_VIDEO_VIDEO_TIMING_H_
+
+#include <stdint.h>
+
+#include <limits>
+#include <string>
+
+#include "api/units/time_delta.h"
+
+namespace webrtc {
+
+// Video timing timestamps in ms counted from capture_time_ms of a frame.
+// This structure represents data sent in video-timing RTP header extension.
+struct VideoSendTiming {
+  enum TimingFrameFlags : uint8_t {
+    kNotTriggered = 0,  // Timing info valid, but not to be transmitted.
+                        // Used on send-side only.
+    kTriggeredByTimer = 1 << 0,  // Frame marked for tracing by periodic timer.
+    kTriggeredBySize = 1 << 1,   // Frame marked for tracing due to size.
+    kInvalid = std::numeric_limits<uint8_t>::max()  // Invalid, ignore!
+  };
+
+  // Returns |time_ms - base_ms| capped at max 16-bit value.
+  // Used to fill this data structure as per
+  // https://webrtc.org/experiments/rtp-hdrext/video-timing/ extension stores
+  // 16-bit deltas of timestamps from packet capture time.
+  static uint16_t GetDeltaCappedMs(int64_t base_ms, int64_t time_ms);
+  static uint16_t GetDeltaCappedMs(TimeDelta delta);
+
+  uint16_t encode_start_delta_ms;
+  uint16_t encode_finish_delta_ms;
+  uint16_t packetization_finish_delta_ms;
+  uint16_t pacer_exit_delta_ms;
+  uint16_t network_timestamp_delta_ms;
+  uint16_t network2_timestamp_delta_ms;
+  uint8_t flags = TimingFrameFlags::kInvalid;
+};
+
+// Used to report precise timings of a 'timing frames'. Contains all important
+// timestamps for a lifetime of that specific frame. Reported as a string via
+// GetStats(). Only frame which took the longest between two GetStats calls is
+// reported.
+struct TimingFrameInfo {
+  TimingFrameInfo();
+
+  // Returns end-to-end delay of a frame, if sender and receiver timestamps are
+  // synchronized, -1 otherwise.
+  int64_t EndToEndDelay() const;
+
+  // Returns true if current frame took longer to process than `other` frame.
+  // If other frame's clocks are not synchronized, current frame is always
+  // preferred.
+  bool IsLongerThan(const TimingFrameInfo& other) const;
+
+  // Returns true if flags are set to indicate this frame was marked for tracing
+  // due to the size being outside some limit.
+  bool IsOutlier() const;
+
+  // Returns true if flags are set to indicate this frame was marked fro tracing
+  // due to cyclic timer.
+  bool IsTimerTriggered() const;
+
+  // Returns true if the timing data is marked as invalid, in which case it
+  // should be ignored.
+  bool IsInvalid() const;
+
+  std::string ToString() const;
+
+  bool operator<(const TimingFrameInfo& other) const;
+
+  bool operator<=(const TimingFrameInfo& other) const;
+
+  uint32_t rtp_timestamp;  // Identifier of a frame.
+  // All timestamps below are in local monotonous clock of a receiver.
+  // If sender clock is not yet estimated, sender timestamps
+  // (capture_time_ms ... pacer_exit_ms) are negative values, still
+  // relatively correct.
+  int64_t capture_time_ms;          // Captrue time of a frame.
+  int64_t encode_start_ms;          // Encode start time.
+  int64_t encode_finish_ms;         // Encode completion time.
+  int64_t packetization_finish_ms;  // Time when frame was passed to pacer.
+  int64_t pacer_exit_ms;  // Time when last packet was pushed out of pacer.
+  // Two in-network RTP processor timestamps: meaning is application specific.
+  int64_t network_timestamp_ms;
+  int64_t network2_timestamp_ms;
+  int64_t receive_start_ms;   // First received packet time.
+  int64_t receive_finish_ms;  // Last received packet time.
+  int64_t decode_start_ms;    // Decode start time.
+  int64_t decode_finish_ms;   // Decode completion time.
+  int64_t render_time_ms;     // Proposed render time to insure smooth playback.
+
+  uint8_t flags;  // Flags indicating validity and/or why tracing was triggered.
+};
+
+// Minimum and maximum playout delay values from capture to render.
+// These are best effort values.
+//
+// min = max = 0 indicates that the receiver should try and render
+// frame as soon as possible.
+//
+// min = x, max = y indicates that the receiver is free to adapt
+// in the range (x, y) based on network jitter.
+// This class ensures invariant 0 <= min <= max <= kMax.
+class VideoPlayoutDelay {
+ public:
+  // Maximum supported value for the delay limit.
+  static constexpr TimeDelta kMax = TimeDelta::Millis(10) * 0xFFF;
+
+  // Creates delay limits that indicates receiver should try to render frame
+  // as soon as possible.
+  static VideoPlayoutDelay Minimal() {
+    return VideoPlayoutDelay(TimeDelta::Zero(), TimeDelta::Zero());
+  }
+
+  // Creates valid, but unspecified limits.
+  VideoPlayoutDelay() = default;
+  VideoPlayoutDelay(const VideoPlayoutDelay&) = default;
+  VideoPlayoutDelay& operator=(const VideoPlayoutDelay&) = default;
+  VideoPlayoutDelay(TimeDelta min, TimeDelta max);
+
+  bool Set(TimeDelta min, TimeDelta max);
+
+  TimeDelta min() const { return min_; }
+  TimeDelta max() const { return max_; }
+
+  friend bool operator==(const VideoPlayoutDelay& lhs,
+                         const VideoPlayoutDelay& rhs) {
+    return lhs.min_ == rhs.min_ && lhs.max_ == rhs.max_;
+  }
+
+ private:
+  TimeDelta min_ = TimeDelta::Zero();
+  TimeDelta max_ = kMax;
+};
+
+}  // namespace webrtc
+
+#endif  // API_VIDEO_VIDEO_TIMING_H_

src/common/rtc_base/network/sent_packet.cc

@@ -0,0 +1,27 @@
+/*
+ *  Copyright 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 "rtc_base/network/sent_packet.h"
+
+namespace rtc {
+
+PacketInfo::PacketInfo() = default;
+PacketInfo::PacketInfo(const PacketInfo& info) = default;
+PacketInfo::~PacketInfo() = default;
+
+SentPacket::SentPacket() = default;
+SentPacket::SentPacket(int64_t packet_id, int64_t send_time_ms)
+    : packet_id(packet_id), send_time_ms(send_time_ms) {}
+SentPacket::SentPacket(int64_t packet_id,
+                       int64_t send_time_ms,
+                       const rtc::PacketInfo& info)
+    : packet_id(packet_id), send_time_ms(send_time_ms), info(info) {}
+
+}  // namespace rtc

src/common/rtc_base/network/sent_packet.h

@@ -0,0 +1,70 @@
+/*
+ *  Copyright 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 RTC_BASE_NETWORK_SENT_PACKET_H_
+#define RTC_BASE_NETWORK_SENT_PACKET_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <optional>
+
+namespace rtc {
+
+enum class PacketType {
+  kUnknown,
+  kData,
+  kIceConnectivityCheck,
+  kIceConnectivityCheckResponse,
+  kStunMessage,
+  kTurnMessage,
+};
+
+enum class PacketInfoProtocolType {
+  kUnknown,
+  kUdp,
+  kTcp,
+  kSsltcp,
+  kTls,
+};
+
+struct PacketInfo {
+  PacketInfo();
+  PacketInfo(const PacketInfo& info);
+  ~PacketInfo();
+
+  bool included_in_feedback = false;
+  bool included_in_allocation = false;
+  // `is_media` is true if this is an audio or video packet, excluding
+  // retransmissions.
+  bool is_media = false;
+  PacketType packet_type = PacketType::kUnknown;
+  PacketInfoProtocolType protocol = PacketInfoProtocolType::kUnknown;
+  // A unique id assigned by the network manager, and std::nullopt if not set.
+  std::optional<uint16_t> network_id;
+  size_t packet_size_bytes = 0;
+  size_t turn_overhead_bytes = 0;
+  size_t ip_overhead_bytes = 0;
+};
+
+struct SentPacket {
+  SentPacket();
+  SentPacket(int64_t packet_id, int64_t send_time_ms);
+  SentPacket(int64_t packet_id, int64_t send_time_ms,
+             const rtc::PacketInfo& info);
+
+  int64_t packet_id = -1;
+  int64_t send_time_ms = -1;
+  rtc::PacketInfo info;
+};
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_NETWORK_SENT_PACKET_H_

src/common/rtc_base/network_constants.cc

@@ -0,0 +1,44 @@
+/*
+ *  Copyright 2020 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 "rtc_base/network_constants.h"
+
+namespace rtc {
+
+std::string AdapterTypeToString(AdapterType type) {
+  switch (type) {
+    case ADAPTER_TYPE_ANY:
+      return "Wildcard";
+    case ADAPTER_TYPE_UNKNOWN:
+      return "Unknown";
+    case ADAPTER_TYPE_ETHERNET:
+      return "Ethernet";
+    case ADAPTER_TYPE_WIFI:
+      return "Wifi";
+    case ADAPTER_TYPE_CELLULAR:
+      return "Cellular";
+    case ADAPTER_TYPE_CELLULAR_2G:
+      return "Cellular2G";
+    case ADAPTER_TYPE_CELLULAR_3G:
+      return "Cellular3G";
+    case ADAPTER_TYPE_CELLULAR_4G:
+      return "Cellular4G";
+    case ADAPTER_TYPE_CELLULAR_5G:
+      return "Cellular5G";
+    case ADAPTER_TYPE_VPN:
+      return "VPN";
+    case ADAPTER_TYPE_LOOPBACK:
+      return "Loopback";
+    default:
+      return std::string();
+  }
+}
+
+}  // namespace rtc

src/common/rtc_base/network_constants.h

@@ -0,0 +1,72 @@
+/*
+ *  Copyright 2004 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 RTC_BASE_NETWORK_CONSTANTS_H_
+#define RTC_BASE_NETWORK_CONSTANTS_H_
+
+#include <stdint.h>
+
+#include <string>
+
+namespace rtc {
+
+constexpr uint16_t kNetworkCostMax = 999;
+constexpr uint16_t kNetworkCostCellular2G = 980;
+constexpr uint16_t kNetworkCostCellular3G = 910;
+constexpr uint16_t kNetworkCostCellular = 900;
+constexpr uint16_t kNetworkCostCellular4G = 500;
+constexpr uint16_t kNetworkCostCellular5G = 250;
+constexpr uint16_t kNetworkCostUnknown = 50;
+constexpr uint16_t kNetworkCostLow = 10;
+constexpr uint16_t kNetworkCostMin = 0;
+
+// Add 1 to network cost of underlying network type
+// so that e.g a "plain" WIFI is prefered over a VPN over WIFI
+// everything else being equal.
+constexpr uint16_t kNetworkCostVpn = 1;
+
+// alias
+constexpr uint16_t kNetworkCostHigh = kNetworkCostCellular;
+
+enum AdapterType {
+  // This enum resembles the one in Chromium net::ConnectionType.
+  ADAPTER_TYPE_UNKNOWN = 0,
+  ADAPTER_TYPE_ETHERNET = 1 << 0,
+  ADAPTER_TYPE_WIFI = 1 << 1,
+  ADAPTER_TYPE_CELLULAR = 1 << 2,  // This is CELLULAR of unknown type.
+  ADAPTER_TYPE_VPN = 1 << 3,
+  ADAPTER_TYPE_LOOPBACK = 1 << 4,
+  // ADAPTER_TYPE_ANY is used for a network, which only contains a single "any
+  // address" IP address (INADDR_ANY for IPv4 or in6addr_any for IPv6), and can
+  // use any/all network interfaces. Whereas ADAPTER_TYPE_UNKNOWN is used
+  // when the network uses a specific interface/IP, but its interface type can
+  // not be determined or not fit in this enum.
+  ADAPTER_TYPE_ANY = 1 << 5,
+  ADAPTER_TYPE_CELLULAR_2G = 1 << 6,
+  ADAPTER_TYPE_CELLULAR_3G = 1 << 7,
+  ADAPTER_TYPE_CELLULAR_4G = 1 << 8,
+  ADAPTER_TYPE_CELLULAR_5G = 1 << 9
+};
+
+std::string AdapterTypeToString(AdapterType type);
+
+// Useful for testing!
+constexpr AdapterType kAllAdapterTypes[] = {
+    ADAPTER_TYPE_UNKNOWN,     ADAPTER_TYPE_ETHERNET,
+    ADAPTER_TYPE_WIFI,        ADAPTER_TYPE_CELLULAR,
+    ADAPTER_TYPE_VPN,         ADAPTER_TYPE_LOOPBACK,
+    ADAPTER_TYPE_ANY,         ADAPTER_TYPE_CELLULAR_2G,
+    ADAPTER_TYPE_CELLULAR_3G, ADAPTER_TYPE_CELLULAR_4G,
+    ADAPTER_TYPE_CELLULAR_5G,
+};
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_NETWORK_CONSTANTS_H_

src/common/rtc_base/network_route.cc

@@ -0,0 +1,27 @@
+/*
+ *  Copyright 2020 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 "rtc_base/network_route.h"
+
+namespace rtc {
+
+bool RouteEndpoint::operator==(const RouteEndpoint& other) const {
+  return adapter_type_ == other.adapter_type_ &&
+         adapter_id_ == other.adapter_id_ && network_id_ == other.network_id_ &&
+         uses_turn_ == other.uses_turn_;
+}
+
+bool NetworkRoute::operator==(const NetworkRoute& other) const {
+  return connected == other.connected && local == other.local &&
+         remote == other.remote && packet_overhead == other.packet_overhead &&
+         last_sent_packet_id == other.last_sent_packet_id;
+}
+
+}  // namespace rtc

src/common/rtc_base/network_route.h

@@ -0,0 +1,93 @@
+/*
+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
+ *
+ *  Use of this source code is governed by a BSD-style license
+ *  that can be found in the LICENSE file in the root of the source
+ *  tree. An additional intellectual property rights grant can be found
+ *  in the file PATENTS.  All contributing project authors may
+ *  be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef RTC_BASE_NETWORK_ROUTE_H_
+#define RTC_BASE_NETWORK_ROUTE_H_
+
+#include <stdint.h>
+
+#include <sstream>
+#include <string>
+
+#include "rtc_base/network_constants.h"
+
+// TODO(honghaiz): Make a directory that describes the interfaces and structs
+// the media code can rely on and the network code can implement, and both can
+// depend on that, but not depend on each other. Then, move this file to that
+// directory.
+namespace rtc {
+
+class RouteEndpoint {
+ public:
+  RouteEndpoint() {}  // Used by tests.
+  RouteEndpoint(AdapterType adapter_type, uint16_t adapter_id,
+                uint16_t network_id, bool uses_turn)
+      : adapter_type_(adapter_type),
+        adapter_id_(adapter_id),
+        network_id_(network_id),
+        uses_turn_(uses_turn) {}
+
+  RouteEndpoint(const RouteEndpoint&) = default;
+  RouteEndpoint& operator=(const RouteEndpoint&) = default;
+
+  // Used by tests.
+  static RouteEndpoint CreateWithNetworkId(uint16_t network_id) {
+    return RouteEndpoint(ADAPTER_TYPE_UNKNOWN,
+                         /* adapter_id = */ 0, network_id,
+                         /* uses_turn = */ false);
+  }
+  RouteEndpoint CreateWithTurn(bool uses_turn) const {
+    return RouteEndpoint(adapter_type_, adapter_id_, network_id_, uses_turn);
+  }
+
+  AdapterType adapter_type() const { return adapter_type_; }
+  uint16_t adapter_id() const { return adapter_id_; }
+  uint16_t network_id() const { return network_id_; }
+  bool uses_turn() const { return uses_turn_; }
+
+  bool operator==(const RouteEndpoint& other) const;
+
+ private:
+  AdapterType adapter_type_ = ADAPTER_TYPE_UNKNOWN;
+  uint16_t adapter_id_ = 0;
+  uint16_t network_id_ = 0;
+  bool uses_turn_ = false;
+};
+
+struct NetworkRoute {
+  bool connected = false;
+  RouteEndpoint local;
+  RouteEndpoint remote;
+  // Last packet id sent on the PREVIOUS route.
+  int last_sent_packet_id = -1;
+  // The overhead in bytes from IP layer and above.
+  // This is the maximum of any part of the route.
+  int packet_overhead = 0;
+
+  std::string DebugString() const {
+    std::ostringstream oss;
+    oss << "[ connected: " << connected << " local: [ " << local.adapter_id()
+        << "/" << local.network_id() << " "
+        << AdapterTypeToString(local.adapter_type())
+        << " turn: " << local.uses_turn() << " ] remote: [ "
+        << remote.adapter_id() << "/" << remote.network_id() << " "
+        << AdapterTypeToString(remote.adapter_type())
+        << " turn: " << remote.uses_turn()
+        << " ] packet_overhead_bytes: " << packet_overhead << " ]";
+    return oss.str();
+  }
+
+  bool operator==(const NetworkRoute& other) const;
+  bool operator!=(const NetworkRoute& other) { return !operator==(other); }
+};
+
+}  // namespace rtc
+
+#endif  // RTC_BASE_NETWORK_ROUTE_H_