[fix] fix congestion control module

src/common/api/media_types.h

@@ -0,0 +1,42 @@
+/*
+ *  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_MEDIA_TYPES_H_
+#define API_MEDIA_TYPES_H_
+
+#include <string>
+
+// The cricket and webrtc have separate definitions for what a media type is.
+// They're not compatible. Watch out for this.
+
+namespace cricket {
+
+enum MediaType {
+  MEDIA_TYPE_AUDIO,
+  MEDIA_TYPE_VIDEO,
+  MEDIA_TYPE_DATA,
+  MEDIA_TYPE_UNSUPPORTED
+};
+
+extern const char kMediaTypeAudio[];
+extern const char kMediaTypeVideo[];
+extern const char kMediaTypeData[];
+
+std::string MediaTypeToString(MediaType type);
+
+}  // namespace cricket
+
+namespace webrtc {
+
+enum class MediaType { ANY, AUDIO, VIDEO, DATA };
+
+}  // namespace webrtc
+
+#endif  // API_MEDIA_TYPES_H_

src/common/api/network_state_predictor.h

@@ -0,0 +1,51 @@
+/*
+ *  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 API_NETWORK_STATE_PREDICTOR_H_
+#define API_NETWORK_STATE_PREDICTOR_H_
+
+#include <cstdint>
+#include <memory>
+
+#include "api/transport/bandwidth_usage.h"
+
+namespace webrtc {
+
+// TODO(yinwa): work in progress. API in class NetworkStatePredictor should not
+// be used by other users until this comment is removed.
+
+// NetworkStatePredictor predict network state based on current network metrics.
+// Usage:
+// Setup by calling Initialize.
+// For each update, call Update. Update returns network state
+// prediction.
+class NetworkStatePredictor {
+ public:
+  virtual ~NetworkStatePredictor() {}
+
+  // Returns current network state prediction.
+  // Inputs:  send_time_ms - packet send time.
+  //          arrival_time_ms - packet arrival time.
+  //          network_state - computed network state.
+  virtual BandwidthUsage Update(int64_t send_time_ms,
+                                int64_t arrival_time_ms,
+                                BandwidthUsage network_state) = 0;
+};
+
+class NetworkStatePredictorFactoryInterface {
+ public:
+  virtual std::unique_ptr<NetworkStatePredictor>
+  CreateNetworkStatePredictor() = 0;
+  virtual ~NetworkStatePredictorFactoryInterface() = default;
+};
+
+}  // namespace webrtc
+
+#endif  // API_NETWORK_STATE_PREDICTOR_H_

src/common/api/transport/bandwidth_usage.h

@@ -0,0 +1,17 @@
+/*
+ * @Author: DI JUNKUN
+ * @Date: 2025-01-15
+ * Copyright (c) 2025 by DI JUNKUN, All Rights Reserved.
+ */
+
+#ifndef _BANDWIDTH_USAGE_H_
+#define _BANDWIDTH_USAGE_H_
+
+enum class BandwidthUsage {
+  kBwNormal = 0,
+  kBwUnderusing = 1,
+  kBwOverusing = 2,
+  kLast
+};
+
+#endif

src/common/api/transport/network_control.h

@@ -0,0 +1,124 @@
+/*
+ *  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/rtc_base/bitrate_tracker.cc

@@ -0,0 +1,39 @@
+/*
+ *  Copyright (c) 2023 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/bitrate_tracker.h"
+
+#include <optional>
+
+#include "api/units/data_rate.h"
+#include "api/units/timestamp.h"
+#include "rtc_base/rate_statistics.h"
+
+namespace webrtc {
+
+BitrateTracker::BitrateTracker(TimeDelta max_window_size)
+    : impl_(max_window_size.ms(), RateStatistics::kBpsScale) {}
+
+std::optional<DataRate> BitrateTracker::Rate(Timestamp now) const {
+  if (std::optional<int64_t> rate = impl_.Rate(now.ms())) {
+    return DataRate::BitsPerSec(*rate);
+  }
+  return std::nullopt;
+}
+
+bool BitrateTracker::SetWindowSize(TimeDelta window_size, Timestamp now) {
+  return impl_.SetWindowSize(window_size.ms(), now.ms());
+}
+
+void BitrateTracker::Update(int64_t bytes, Timestamp now) {
+  impl_.Update(bytes, now.ms());
+}
+
+}  // namespace webrtc

src/common/rtc_base/bitrate_tracker.h

@@ -0,0 +1,64 @@
+/*
+ *  Copyright (c) 2023 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_BITRATE_TRACKER_H_
+#define RTC_BASE_BITRATE_TRACKER_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <optional>
+
+#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/rate_statistics.h"
+
+namespace webrtc {
+// Class to estimate bitrates over running window.
+// Timestamps used in Update(), Rate() and SetWindowSize() must never
+// decrease for two consecutive calls.
+// This class is thread unsafe.
+class BitrateTracker {
+ public:
+  // max_window_sizes = Maximum window size for the rate estimation.
+  //                    Initial window size is set to this, but may be changed
+  //                    to something lower by calling SetWindowSize().
+  explicit BitrateTracker(TimeDelta max_window_size);
+
+  BitrateTracker(const BitrateTracker&) = default;
+  BitrateTracker(BitrateTracker&&) = default;
+  BitrateTracker& operator=(const BitrateTracker&) = delete;
+  BitrateTracker& operator=(BitrateTracker&&) = delete;
+
+  ~BitrateTracker() = default;
+
+  // Resets instance to original state.
+  void Reset() { impl_.Reset(); }
+
+  // Updates bitrate with a new data point, moving averaging window as needed.
+  void Update(int64_t bytes, Timestamp now);
+  void Update(DataSize size, Timestamp now) { Update(size.bytes(), now); }
+
+  // Returns bitrate, moving averaging window as needed.
+  // Returns nullopt when bitrate can't be measured.
+  std::optional<DataRate> Rate(Timestamp now) const;
+
+  // Update the size of the averaging window. The maximum allowed value for
+  // `window_size` is `max_window_size` as supplied in the constructor.
+  bool SetWindowSize(TimeDelta window_size, Timestamp now);
+
+ private:
+  RateStatistics impl_;
+};
+}  // namespace webrtc
+
+#endif  // RTC_BASE_BITRATE_TRACKER_H_

src/common/rtc_base/rate_statistics.cc

@@ -0,0 +1,160 @@
+/*
+ *  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 "rtc_base/rate_statistics.h"
+
+#include <algorithm>
+#include <limits>
+#include <memory>
+
+#include "log.h"
+#include "rtc_base/numerics/safe_conversions.h"
+
+namespace webrtc {
+
+RateStatistics::Bucket::Bucket(int64_t timestamp)
+    : sum(0), num_samples(0), timestamp(timestamp) {}
+
+RateStatistics::RateStatistics(int64_t window_size_ms, float scale)
+    : accumulated_count_(0),
+      first_timestamp_(-1),
+      num_samples_(0),
+      scale_(scale),
+      max_window_size_ms_(window_size_ms),
+      current_window_size_ms_(max_window_size_ms_) {}
+
+RateStatistics::RateStatistics(const RateStatistics& other)
+    : buckets_(other.buckets_),
+      accumulated_count_(other.accumulated_count_),
+      first_timestamp_(other.first_timestamp_),
+      overflow_(other.overflow_),
+      num_samples_(other.num_samples_),
+      scale_(other.scale_),
+      max_window_size_ms_(other.max_window_size_ms_),
+      current_window_size_ms_(other.current_window_size_ms_) {}
+
+RateStatistics::RateStatistics(RateStatistics&& other) = default;
+
+RateStatistics::~RateStatistics() {}
+
+void RateStatistics::Reset() {
+  accumulated_count_ = 0;
+  overflow_ = false;
+  num_samples_ = 0;
+  first_timestamp_ = -1;
+  current_window_size_ms_ = max_window_size_ms_;
+  buckets_.clear();
+}
+
+void RateStatistics::Update(int64_t count, int64_t now_ms) {
+  // Don't reset `first_timestamp_` if the last sample removed by EraseOld() was
+  // recent. This ensures that the window maintains its intended duration even
+  // when samples are received near the boundary. Use a margin of 50% of the
+  // current window size.
+  const int64_t recent_sample_time_margin = 1.5 * current_window_size_ms_;
+  bool last_sample_is_recent =
+      !buckets_.empty() &&
+      buckets_.back().timestamp > now_ms - recent_sample_time_margin;
+
+  EraseOld(now_ms);
+  if (first_timestamp_ == -1 || (num_samples_ == 0 && !last_sample_is_recent)) {
+    first_timestamp_ = now_ms;
+  }
+
+  if (buckets_.empty() || now_ms != buckets_.back().timestamp) {
+    if (!buckets_.empty() && now_ms < buckets_.back().timestamp) {
+      LOG_WARN(
+          "Timestamp {} is before the last added timestamp in the rate window: "
+          "{}, aligning to that.",
+          now_ms, buckets_.back().timestamp);
+      now_ms = buckets_.back().timestamp;
+    }
+    buckets_.emplace_back(now_ms);
+  }
+  Bucket& last_bucket = buckets_.back();
+  last_bucket.sum += count;
+  ++last_bucket.num_samples;
+
+  if (std::numeric_limits<int64_t>::max() - accumulated_count_ > count) {
+    accumulated_count_ += count;
+  } else {
+    overflow_ = true;
+  }
+  ++num_samples_;
+}
+
+std::optional<int64_t> RateStatistics::Rate(int64_t now_ms) const {
+  // Yeah, this const_cast ain't pretty, but the alternative is to declare most
+  // of the members as mutable...
+  const_cast<RateStatistics*>(this)->EraseOld(now_ms);
+
+  int active_window_size = 0;
+  if (first_timestamp_ != -1) {
+    if (first_timestamp_ <= now_ms - current_window_size_ms_) {
+      // Count window as full even if no data points currently in view, if the
+      // data stream started before the window.
+      active_window_size = current_window_size_ms_;
+    } else {
+      // Size of a single bucket is 1ms, so even if now_ms == first_timestmap_
+      // the window size should be 1.
+      active_window_size = now_ms - first_timestamp_ + 1;
+    }
+  }
+
+  // If window is a single bucket or there is only one sample in a data set that
+  // has not grown to the full window size, or if the accumulator has
+  // overflowed, treat this as rate unavailable.
+  if (num_samples_ == 0 || active_window_size <= 1 ||
+      (num_samples_ <= 1 &&
+       rtc::SafeLt(active_window_size, current_window_size_ms_)) ||
+      overflow_) {
+    return std::nullopt;
+  }
+
+  float scale = static_cast<float>(scale_) / active_window_size;
+  float result = accumulated_count_ * scale + 0.5f;
+
+  // Better return unavailable rate than garbage value (undefined behavior).
+  if (result > static_cast<float>(std::numeric_limits<int64_t>::max())) {
+    return std::nullopt;
+  }
+  return rtc::dchecked_cast<int64_t>(result);
+}
+
+void RateStatistics::EraseOld(int64_t now_ms) {
+  // New oldest time that is included in data set.
+  const int64_t new_oldest_time = now_ms - current_window_size_ms_ + 1;
+
+  // Loop over buckets and remove too old data points.
+  while (!buckets_.empty() && buckets_.front().timestamp < new_oldest_time) {
+    const Bucket& oldest_bucket = buckets_.front();
+    accumulated_count_ -= oldest_bucket.sum;
+    num_samples_ -= oldest_bucket.num_samples;
+    buckets_.pop_front();
+    // This does not clear overflow_ even when counter is empty.
+    // TODO(https://bugs.webrtc.org/11247): Consider if overflow_ can be reset.
+  }
+}
+
+bool RateStatistics::SetWindowSize(int64_t window_size_ms, int64_t now_ms) {
+  if (window_size_ms <= 0 || window_size_ms > max_window_size_ms_) return false;
+  if (first_timestamp_ != -1) {
+    // If the window changes (e.g. decreases - removing data point, then
+    // increases again) we need to update the first timestamp mark as
+    // otherwise it indicates the window coveres a region of zeros, suddenly
+    // under-estimating the rate.
+    first_timestamp_ = std::max(first_timestamp_, now_ms - window_size_ms + 1);
+  }
+  current_window_size_ms_ = window_size_ms;
+  EraseOld(now_ms);
+  return true;
+}
+
+}  // namespace webrtc

src/common/rtc_base/rate_statistics.h

@@ -0,0 +1,103 @@
+/*
+ *  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.
+ */
+
+#ifndef RTC_BASE_RATE_STATISTICS_H_
+#define RTC_BASE_RATE_STATISTICS_H_
+
+#include <stddef.h>
+#include <stdint.h>
+
+#include <deque>
+#include <memory>
+#include <optional>
+
+namespace webrtc {
+
+// Class to estimate rates based on counts in a sequence of 1-millisecond
+// intervals.
+
+// This class uses int64 for all its numbers because some rates can be very
+// high; for instance, a 20 Mbit/sec video stream can wrap a 32-bit byte
+// counter in 14 minutes.
+
+// Note that timestamps used in Update(), Rate() and SetWindowSize() must never
+// decrease for two consecutive calls.
+// TODO(bugs.webrtc.org/11600): Migrate from int64_t to Timestamp.
+
+class RateStatistics {
+ public:
+  static constexpr float kBpsScale = 8000.0f;
+
+  // max_window_size_ms = Maximum window size in ms for the rate estimation.
+  //                      Initial window size is set to this, but may be changed
+  //                      to something lower by calling SetWindowSize().
+  // scale = coefficient to convert counts/ms to desired unit
+  //         ex: kBpsScale (8000) for bits/s if count represents bytes.
+  RateStatistics(int64_t max_window_size_ms, float scale);
+
+  RateStatistics(const RateStatistics& other);
+
+  RateStatistics(RateStatistics&& other);
+
+  ~RateStatistics();
+
+  // Reset instance to original state.
+  void Reset();
+
+  // Update rate with a new data point, moving averaging window as needed.
+  void Update(int64_t count, int64_t now_ms);
+
+  // Note that despite this being a const method, it still updates the internal
+  // state (moves averaging window), but it doesn't make any alterations that
+  // are observable from the other methods, as long as supplied timestamps are
+  // from a monotonic clock. Ie, it doesn't matter if this call moves the
+  // window, since any subsequent call to Update or Rate would still have moved
+  // the window as much or more.
+  std::optional<int64_t> Rate(int64_t now_ms) const;
+
+  // Update the size of the averaging window. The maximum allowed value for
+  // window_size_ms is max_window_size_ms as supplied in the constructor.
+  bool SetWindowSize(int64_t window_size_ms, int64_t now_ms);
+
+ private:
+  void EraseOld(int64_t now_ms);
+
+  struct Bucket {
+    explicit Bucket(int64_t timestamp);
+    int64_t sum;              // Sum of all samples in this bucket.
+    int num_samples;          // Number of samples in this bucket.
+    const int64_t timestamp;  // Timestamp this bucket corresponds to.
+  };
+  // All buckets within the time window, ordered by time.
+  std::deque<Bucket> buckets_;
+
+  // Total count recorded in all buckets.
+  int64_t accumulated_count_;
+
+  // Timestamp of the first data point seen, or -1 of none seen.
+  int64_t first_timestamp_;
+
+  // True if accumulated_count_ has ever grown too large to be
+  // contained in its integer type.
+  bool overflow_ = false;
+
+  // The total number of samples in the buckets.
+  int num_samples_;
+
+  // To convert counts/ms to desired units
+  const float scale_;
+
+  // The window sizes, in ms, over which the rate is calculated.
+  const int64_t max_window_size_ms_;
+  int64_t current_window_size_ms_;
+};
+}  // namespace webrtc
+
+#endif  // RTC_BASE_RATE_STATISTICS_H_

src/common/rtc_base/thread_annotations.h

@@ -0,0 +1,98 @@
+//
+// 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.
+//
+// Borrowed from
+// https://code.google.com/p/gperftools/source/browse/src/base/thread_annotations.h
+// but adapted for clang attributes instead of the gcc.
+//
+// This header file contains the macro definitions for thread safety
+// annotations that allow the developers to document the locking policies
+// of their multi-threaded code. The annotations can also help program
+// analysis tools to identify potential thread safety issues.
+
+#ifndef RTC_BASE_THREAD_ANNOTATIONS_H_
+#define RTC_BASE_THREAD_ANNOTATIONS_H_
+
+#if defined(__clang__) && (!defined(SWIG))
+#define RTC_THREAD_ANNOTATION_ATTRIBUTE__(x) __attribute__((x))
+#else
+#define RTC_THREAD_ANNOTATION_ATTRIBUTE__(x)  // no-op
+#endif
+
+// Document if a shared variable/field needs to be protected by a lock.
+// GUARDED_BY allows the user to specify a particular lock that should be
+// held when accessing the annotated variable.
+#define RTC_GUARDED_BY(x) RTC_THREAD_ANNOTATION_ATTRIBUTE__(guarded_by(x))
+
+// Document if the memory location pointed to by a pointer should be guarded
+// by a lock when dereferencing the pointer. Note that a pointer variable to a
+// shared memory location could itself be a shared variable. For example, if a
+// shared global pointer q, which is guarded by mu1, points to a shared memory
+// location that is guarded by mu2, q should be annotated as follows:
+//     int *q GUARDED_BY(mu1) PT_GUARDED_BY(mu2);
+#define RTC_PT_GUARDED_BY(x) RTC_THREAD_ANNOTATION_ATTRIBUTE__(pt_guarded_by(x))
+
+// Document the acquisition order between locks that can be held
+// simultaneously by a thread. For any two locks that need to be annotated
+// to establish an acquisition order, only one of them needs the annotation.
+// (i.e. You don't have to annotate both locks with both ACQUIRED_AFTER
+// and ACQUIRED_BEFORE.)
+#define RTC_ACQUIRED_AFTER(x) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(acquired_after(x))
+#define RTC_ACQUIRED_BEFORE(x) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(acquired_before(x))
+
+// The following three annotations document the lock requirements for
+// functions/methods.
+
+// Document if a function expects certain locks to be held before it is called
+#define RTC_EXCLUSIVE_LOCKS_REQUIRED(...) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(exclusive_locks_required(__VA_ARGS__))
+#define RTC_SHARED_LOCKS_REQUIRED(...) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(shared_locks_required(__VA_ARGS__))
+
+// Document the locks acquired in the body of the function. These locks
+// cannot be held when calling this function (as google3's Mutex locks are
+// non-reentrant).
+#define RTC_LOCKS_EXCLUDED(...) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(locks_excluded(__VA_ARGS__))
+
+// Document the lock the annotated function returns without acquiring it.
+#define RTC_LOCK_RETURNED(x) RTC_THREAD_ANNOTATION_ATTRIBUTE__(lock_returned(x))
+
+// Document if a class/type is a lockable type (such as the Mutex class).
+#define RTC_LOCKABLE RTC_THREAD_ANNOTATION_ATTRIBUTE__(lockable)
+
+// Document if a class is a scoped lockable type (such as the MutexLock class).
+#define RTC_SCOPED_LOCKABLE RTC_THREAD_ANNOTATION_ATTRIBUTE__(scoped_lockable)
+
+// The following annotations specify lock and unlock primitives.
+#define RTC_EXCLUSIVE_LOCK_FUNCTION(...) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(exclusive_lock_function(__VA_ARGS__))
+
+#define RTC_SHARED_LOCK_FUNCTION(...) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(shared_lock_function(__VA_ARGS__))
+
+#define RTC_EXCLUSIVE_TRYLOCK_FUNCTION(...) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(exclusive_trylock_function(__VA_ARGS__))
+
+#define RTC_SHARED_TRYLOCK_FUNCTION(...) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(shared_trylock_function(__VA_ARGS__))
+
+#define RTC_UNLOCK_FUNCTION(...) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(unlock_function(__VA_ARGS__))
+
+#define RTC_ASSERT_EXCLUSIVE_LOCK(...) \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(assert_exclusive_lock(__VA_ARGS__))
+
+// An escape hatch for thread safety analysis to ignore the annotated function.
+#define RTC_NO_THREAD_SAFETY_ANALYSIS \
+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(no_thread_safety_analysis)
+
+#endif  // RTC_BASE_THREAD_ANNOTATIONS_H_