The goal of this work is to make it easier to experiment with the bandwidth estimation implementation. For this reason network control functionality is moved from SendSideCongestionController(SSCC), PacedSender and BitrateController to the newly created GoogCcNetworkController which implements the newly created NetworkControllerInterface. This allows the implementation to be replaced at runtime in the future. This is the first part of a split of a larger CL, see: https://webrtc-review.googlesource.com/c/src/+/39788/8 For further explanations. Bug: webrtc:8415 Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 Reviewed-on: https://webrtc-review.googlesource.com/43840 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21868}
151 lines
4.6 KiB
C++
151 lines
4.6 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/pacing/packet_queue.h"
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#include <algorithm>
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#include <list>
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#include <vector>
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#include "modules/include/module_common_types.h"
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#include "modules/pacing/bitrate_prober.h"
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#include "modules/pacing/interval_budget.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/field_trial.h"
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namespace webrtc {
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PacketQueue::Packet::Packet(RtpPacketSender::Priority priority,
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uint32_t ssrc,
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uint16_t seq_number,
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int64_t capture_time_ms,
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int64_t enqueue_time_ms,
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size_t length_in_bytes,
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bool retransmission,
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uint64_t enqueue_order)
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: priority(priority),
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ssrc(ssrc),
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sequence_number(seq_number),
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capture_time_ms(capture_time_ms),
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enqueue_time_ms(enqueue_time_ms),
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sum_paused_ms(0),
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bytes(length_in_bytes),
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retransmission(retransmission),
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enqueue_order(enqueue_order) {}
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PacketQueue::Packet::Packet(const Packet& other) = default;
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PacketQueue::Packet::~Packet() {}
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PacketQueue::PacketQueue(const Clock* clock)
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: bytes_(0),
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clock_(clock),
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queue_time_sum_(0),
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time_last_updated_(clock_->TimeInMilliseconds()),
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paused_(false) {}
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PacketQueue::~PacketQueue() {}
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void PacketQueue::Push(const Packet& packet) {
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UpdateQueueTime(packet.enqueue_time_ms);
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// Store packet in list, use pointers in priority queue for cheaper moves.
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// Packets have a handle to its own iterator in the list, for easy removal
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// when popping from queue.
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packet_list_.push_front(packet);
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std::list<Packet>::iterator it = packet_list_.begin();
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it->this_it = it; // Handle for direct removal from list.
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prio_queue_.push(&(*it)); // Pointer into list.
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bytes_ += packet.bytes;
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}
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const PacketQueue::Packet& PacketQueue::BeginPop() {
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const PacketQueue::Packet& packet = *prio_queue_.top();
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prio_queue_.pop();
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return packet;
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}
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void PacketQueue::CancelPop(const PacketQueue::Packet& packet) {
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prio_queue_.push(&(*packet.this_it));
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}
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void PacketQueue::FinalizePop(const PacketQueue::Packet& packet) {
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bytes_ -= packet.bytes;
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int64_t packet_queue_time_ms = time_last_updated_ - packet.enqueue_time_ms;
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RTC_DCHECK_LE(packet.sum_paused_ms, packet_queue_time_ms);
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packet_queue_time_ms -= packet.sum_paused_ms;
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RTC_DCHECK_LE(packet_queue_time_ms, queue_time_sum_);
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queue_time_sum_ -= packet_queue_time_ms;
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packet_list_.erase(packet.this_it);
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RTC_DCHECK_EQ(packet_list_.size(), prio_queue_.size());
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if (packet_list_.empty())
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RTC_DCHECK_EQ(0, queue_time_sum_);
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}
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bool PacketQueue::Empty() const {
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return prio_queue_.empty();
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}
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size_t PacketQueue::SizeInPackets() const {
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return prio_queue_.size();
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}
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uint64_t PacketQueue::SizeInBytes() const {
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return bytes_;
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}
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int64_t PacketQueue::OldestEnqueueTimeMs() const {
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auto it = packet_list_.rbegin();
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if (it == packet_list_.rend())
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return 0;
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return it->enqueue_time_ms;
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}
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void PacketQueue::UpdateQueueTime(int64_t timestamp_ms) {
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RTC_DCHECK_GE(timestamp_ms, time_last_updated_);
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if (timestamp_ms == time_last_updated_)
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return;
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int64_t delta_ms = timestamp_ms - time_last_updated_;
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if (paused_) {
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// Increase per-packet accumulators of time spent in queue while paused,
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// so that we can disregard that when subtracting main accumulator when
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// popping packet from the queue.
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for (auto& it : packet_list_) {
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it.sum_paused_ms += delta_ms;
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}
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} else {
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// Use packet packet_list_.size() not prio_queue_.size() here, as there
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// might be an outstanding element popped from prio_queue_ currently in
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// the SendPacket() call, while packet_list_ will always be correct.
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queue_time_sum_ += delta_ms * packet_list_.size();
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}
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time_last_updated_ = timestamp_ms;
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}
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void PacketQueue::SetPauseState(bool paused, int64_t timestamp_ms) {
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if (paused_ == paused)
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return;
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UpdateQueueTime(timestamp_ms);
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paused_ = paused;
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}
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int64_t PacketQueue::AverageQueueTimeMs() const {
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if (prio_queue_.empty())
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return 0;
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return queue_time_sum_ / packet_list_.size();
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}
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} // namespace webrtc
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