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diff --git a/chromium/third_party/webrtc/modules/pacing/packet_queue2.h b/chromium/third_party/webrtc/modules/pacing/packet_queue2.h
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+++ b/chromium/third_party/webrtc/modules/pacing/packet_queue2.h
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+/*
+ *  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 MODULES_PACING_PACKET_QUEUE2_H_
+#define MODULES_PACING_PACKET_QUEUE2_H_
+
+#include <map>
+#include <queue>
+#include <set>
+
+#include "modules/pacing/packet_queue.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class PacketQueue2 : public PacketQueue {
+ public:
+  explicit PacketQueue2(const Clock* clock);
+  ~PacketQueue2() override;
+
+  using Packet = PacketQueue::Packet;
+
+  void Push(const Packet& packet) override;
+  const Packet& BeginPop() override;
+  void CancelPop(const Packet& packet) override;
+  void FinalizePop(const Packet& packet) override;
+
+  bool Empty() const override;
+  size_t SizeInPackets() const override;
+  uint64_t SizeInBytes() const override;
+
+  int64_t OldestEnqueueTimeMs() const override;
+  int64_t AverageQueueTimeMs() const override;
+  void UpdateQueueTime(int64_t timestamp_ms) override;
+  void SetPauseState(bool paused, int64_t timestamp_ms) override;
+
+  struct StreamPrioKey {
+    StreamPrioKey() = default;
+    StreamPrioKey(RtpPacketSender::Priority priority, int64_t bytes)
+        : priority(priority), bytes(bytes) {}
+
+    bool operator<(const StreamPrioKey& other) const {
+      if (priority != other.priority)
+        return priority < other.priority;
+      return bytes > other.bytes;
+    }
+
+    const RtpPacketSender::Priority priority;
+    const size_t bytes;
+  };
+
+  struct Stream {
+    Stream();
+
+    virtual ~Stream();
+
+    size_t bytes;
+    uint32_t ssrc;
+    std::priority_queue<Packet> packet_queue;
+
+    // Whenever a packet is inserted for this stream we check if |priority_it|
+    // points to an element in |stream_priorities_|, and if it does it means
+    // this stream has already been scheduled, and if the scheduled priority is
+    // lower than the priority of the incoming packet we reschedule this stream
+    // with the higher priority.
+    std::multimap<StreamPrioKey, uint32_t>::iterator priority_it;
+  };
+
+ private:
+  static constexpr size_t kMaxLeadingBytes = 1400;
+
+  Stream* GetHighestPriorityStream();
+
+  // Just used to verify correctness.
+  bool IsSsrcScheduled(uint32_t ssrc) const;
+
+  const Clock* const clock_;
+  int64_t time_last_updated_;
+  rtc::Optional<Packet> pop_packet_;
+  rtc::Optional<Stream*> pop_stream_;
+
+  bool paused_ = false;
+  size_t size_packets_ = 0;
+  size_t size_bytes_ = 0;
+  size_t max_bytes_ = kMaxLeadingBytes;
+  int64_t queue_time_sum_ms_ = 0;
+  int64_t pause_time_sum_ms_ = 0;
+
+  // A map of streams used to prioritize from which stream to send next. We use
+  // a multimap instead of a priority_queue since the priority of a stream can
+  // change as a new packet is inserted, and a multimap allows us to remove and
+  // then reinsert a StreamPrioKey if the priority has increased.
+  std::multimap<StreamPrioKey, uint32_t> stream_priorities_;
+
+  // A map of SSRCs to Streams.
+  std::map<uint32_t, Stream> streams_;
+
+  // The enqueue time of every packet currently in the queue. Used to figure out
+  // the age of the oldest packet in the queue.
+  std::multiset<int64_t> enqueue_times_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_PACING_PACKET_QUEUE2_H_