1 files changed, 209 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/pacing/packet_queue2.cc b/chromium/third_party/webrtc/modules/pacing/packet_queue2.cc
new file mode 100644
index 00000000000..6aee807af36
--- /dev/null
+++ b/chromium/third_party/webrtc/modules/pacing/packet_queue2.cc
@@ -0,0 +1,209 @@
+/*
+ *  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 "modules/pacing/packet_queue2.h"
+
+#include <algorithm>
+
+#include "rtc_base/checks.h"
+#include "system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+PacketQueue2::Stream::Stream() : bytes(0) {}
+PacketQueue2::Stream::~Stream() {}
+
+PacketQueue2::PacketQueue2(const Clock* clock)
+    : PacketQueue(clock),
+      clock_(clock),
+      time_last_updated_(clock_->TimeInMilliseconds()) {}
+
+PacketQueue2::~PacketQueue2() {}
+
+void PacketQueue2::Push(const Packet& packet_to_insert) {
+  Packet packet(packet_to_insert);
+
+  auto stream_info_it = streams_.find(packet.ssrc);
+  if (stream_info_it == streams_.end()) {
+    stream_info_it = streams_.emplace(packet.ssrc, Stream()).first;
+    stream_info_it->second.priority_it = stream_priorities_.end();
+    stream_info_it->second.ssrc = packet.ssrc;
+  }
+
+  Stream* streams_ = &stream_info_it->second;
+
+  if (streams_->priority_it == stream_priorities_.end()) {
+    // If the SSRC is not currently scheduled, add it to |stream_priorities_|.
+    RTC_CHECK(!IsSsrcScheduled(streams_->ssrc));
+    streams_->priority_it = stream_priorities_.emplace(
+        StreamPrioKey(packet.priority, streams_->bytes), packet.ssrc);
+  } else if (packet.priority < streams_->priority_it->first.priority) {
+    // If the priority of this SSRC increased, remove the outdated StreamPrioKey
+    // and insert a new one with the new priority. Note that
+    // RtpPacketSender::Priority uses lower ordinal for higher priority.
+    stream_priorities_.erase(streams_->priority_it);
+    streams_->priority_it = stream_priorities_.emplace(
+        StreamPrioKey(packet.priority, streams_->bytes), packet.ssrc);
+  }
+  RTC_CHECK(streams_->priority_it != stream_priorities_.end());
+
+  packet.enqueue_time_it = enqueue_times_.insert(packet.enqueue_time_ms);
+
+  // In order to figure out how much time a packet has spent in the queue while
+  // not in a paused state, we subtract the total amount of time the queue has
+  // been paused so far, and when the packet is poped we subtract the total
+  // amount of time the queue has been paused at that moment. This way we
+  // subtract the total amount of time the packet has spent in the queue while
+  // in a paused state.
+  UpdateQueueTime(packet.enqueue_time_ms);
+  packet.enqueue_time_ms -= pause_time_sum_ms_;
+  streams_->packet_queue.push(packet);
+
+  size_packets_ += 1;
+  size_bytes_ += packet.bytes;
+}
+
+const PacketQueue2::Packet& PacketQueue2::BeginPop() {
+  RTC_CHECK(!pop_packet_ && !pop_stream_);
+
+  Stream* stream = GetHighestPriorityStream();
+  pop_stream_.emplace(stream);
+  pop_packet_.emplace(stream->packet_queue.top());
+  stream->packet_queue.pop();
+
+  return *pop_packet_;
+}
+
+void PacketQueue2::CancelPop(const Packet& packet) {
+  RTC_CHECK(pop_packet_ && pop_stream_);
+  (*pop_stream_)->packet_queue.push(*pop_packet_);
+  pop_packet_.reset();
+  pop_stream_.reset();
+}
+
+void PacketQueue2::FinalizePop(const Packet& packet) {
+  RTC_CHECK(!paused_);
+  if (!Empty()) {
+    RTC_CHECK(pop_packet_ && pop_stream_);
+    Stream* stream = *pop_stream_;
+    stream_priorities_.erase(stream->priority_it);
+    const Packet& packet = *pop_packet_;
+
+    // Calculate the total amount of time spent by this packet in the queue
+    // while in a non-paused state. Note that the |pause_time_sum_ms_| was
+    // subtracted from |packet.enqueue_time_ms| when the packet was pushed, and
+    // by subtracting it now we effectively remove the time spent in in the
+    // queue while in a paused state.
+    int64_t time_in_non_paused_state_ms =
+        time_last_updated_ - packet.enqueue_time_ms - pause_time_sum_ms_;
+    queue_time_sum_ms_ -= time_in_non_paused_state_ms;
+
+    RTC_CHECK(packet.enqueue_time_it != enqueue_times_.end());
+    enqueue_times_.erase(packet.enqueue_time_it);
+
+    // Update |bytes| of this stream. The general idea is that the stream that
+    // has sent the least amount of bytes should have the highest priority.
+    // The problem with that is if streams send with different rates, in which
+    // case a "budget" will be built up for the stream sending at the lower
+    // rate. To avoid building a too large budget we limit |bytes| to be within
+    // kMaxLeading bytes of the stream that has sent the most amount of bytes.
+    stream->bytes =
+        std::max(stream->bytes + packet.bytes, max_bytes_ - kMaxLeadingBytes);
+    max_bytes_ = std::max(max_bytes_, stream->bytes);
+
+    size_bytes_ -= packet.bytes;
+    size_packets_ -= 1;
+    RTC_CHECK(size_packets_ > 0 || queue_time_sum_ms_ == 0);
+
+    // If there are packets left to be sent, schedule the stream again.
+    RTC_CHECK(!IsSsrcScheduled(stream->ssrc));
+    if (stream->packet_queue.empty()) {
+      stream->priority_it = stream_priorities_.end();
+    } else {
+      RtpPacketSender::Priority priority = stream->packet_queue.top().priority;
+      stream->priority_it = stream_priorities_.emplace(
+          StreamPrioKey(priority, stream->bytes), stream->ssrc);
+    }
+
+    pop_packet_.reset();
+    pop_stream_.reset();
+  }
+}
+
+bool PacketQueue2::Empty() const {
+  RTC_CHECK((!stream_priorities_.empty() && size_packets_ > 0) ||
+            (stream_priorities_.empty() && size_packets_ == 0));
+  return stream_priorities_.empty();
+}
+
+size_t PacketQueue2::SizeInPackets() const {
+  return size_packets_;
+}
+
+uint64_t PacketQueue2::SizeInBytes() const {
+  return size_bytes_;
+}
+
+int64_t PacketQueue2::OldestEnqueueTimeMs() const {
+  if (Empty())
+    return 0;
+  RTC_CHECK(!enqueue_times_.empty());
+  return *enqueue_times_.begin();
+}
+
+void PacketQueue2::UpdateQueueTime(int64_t timestamp_ms) {
+  RTC_CHECK_GE(timestamp_ms, time_last_updated_);
+  if (timestamp_ms == time_last_updated_)
+    return;
+
+  int64_t delta_ms = timestamp_ms - time_last_updated_;
+
+  if (paused_) {
+    pause_time_sum_ms_ += delta_ms;
+  } else {
+    queue_time_sum_ms_ += delta_ms * size_packets_;
+  }
+
+  time_last_updated_ = timestamp_ms;
+}
+
+void PacketQueue2::SetPauseState(bool paused, int64_t timestamp_ms) {
+  if (paused_ == paused)
+    return;
+  UpdateQueueTime(timestamp_ms);
+  paused_ = paused;
+}
+
+int64_t PacketQueue2::AverageQueueTimeMs() const {
+  if (Empty())
+    return 0;
+  return queue_time_sum_ms_ / size_packets_;
+}
+
+PacketQueue2::Stream* PacketQueue2::GetHighestPriorityStream() {
+  RTC_CHECK(!stream_priorities_.empty());
+  uint32_t ssrc = stream_priorities_.begin()->second;
+
+  auto stream_info_it = streams_.find(ssrc);
+  RTC_CHECK(stream_info_it != streams_.end());
+  RTC_CHECK(stream_info_it->second.priority_it == stream_priorities_.begin());
+  RTC_CHECK(!stream_info_it->second.packet_queue.empty());
+  return &stream_info_it->second;
+}
+
+bool PacketQueue2::IsSsrcScheduled(uint32_t ssrc) const {
+  for (const auto& scheduled_stream : stream_priorities_) {
+    if (scheduled_stream.second == ssrc)
+      return true;
+  }
+  return false;
+}
+
+}  // namespace webrtc