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To make testing easier all of PacketQueues functions have been made virtual, and PacketQueue2 now inherits PacketQueue. This change was made to minimize changes in PacedSender. Bug: webrtc:8287, webrtc:8288 Change-Id: I2593340e7cc7da617370b0a33e7b9deeb46d9487 Reviewed-on: https://webrtc-review.googlesource.com/9380 Reviewed-by: Stefan Holmer <stefan@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20385}
111 lines
3.4 KiB
C++
111 lines
3.4 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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#ifndef MODULES_PACING_PACKET_QUEUE2_H_
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#define MODULES_PACING_PACKET_QUEUE2_H_
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#include <map>
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#include <queue>
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#include <set>
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#include "modules/pacing/packet_queue.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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namespace webrtc {
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class PacketQueue2 : public PacketQueue {
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public:
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explicit PacketQueue2(const Clock* clock);
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~PacketQueue2() override;
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using Packet = PacketQueue::Packet;
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void Push(const Packet& packet) override;
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const Packet& BeginPop() override;
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void CancelPop(const Packet& packet) override;
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void FinalizePop(const Packet& packet) override;
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bool Empty() const override;
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size_t SizeInPackets() const override;
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uint64_t SizeInBytes() const override;
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int64_t OldestEnqueueTimeMs() const override;
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int64_t AverageQueueTimeMs() const override;
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void UpdateQueueTime(int64_t timestamp_ms) override;
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void SetPauseState(bool paused, int64_t timestamp_ms) override;
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struct StreamPrioKey {
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StreamPrioKey() = default;
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StreamPrioKey(RtpPacketSender::Priority priority, int64_t bytes)
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: priority(priority), bytes(bytes) {}
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bool operator<(const StreamPrioKey& other) const {
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if (priority != other.priority)
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return priority < other.priority;
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return bytes > other.bytes;
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}
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const RtpPacketSender::Priority priority;
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const size_t bytes;
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};
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struct Stream {
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Stream();
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virtual ~Stream();
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size_t bytes;
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uint32_t ssrc;
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std::priority_queue<Packet> packet_queue;
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// Whenever a packet is inserted for this stream we check if |priority_it|
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// points to an element in |stream_priorities_|, and if it does it means
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// this stream has already been scheduled, and if the scheduled priority is
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// lower than the priority of the incoming packet we reschedule this stream
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// with the higher priority.
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std::multimap<StreamPrioKey, uint32_t>::iterator priority_it;
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};
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private:
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static constexpr size_t kMaxLeadingBytes = 1400;
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Stream* GetHighestPriorityStream();
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// Just used to verify correctness.
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bool IsSsrcScheduled(uint32_t ssrc) const;
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const Clock* const clock_;
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int64_t time_last_updated_;
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rtc::Optional<Packet> pop_packet_;
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rtc::Optional<Stream*> pop_stream_;
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bool paused_ = false;
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size_t size_packets_ = 0;
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size_t size_bytes_ = 0;
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size_t max_bytes_ = kMaxLeadingBytes;
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int64_t queue_time_sum_ms_ = 0;
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int64_t pause_time_sum_ms_ = 0;
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// A map of streams used to prioritize from which stream to send next. We use
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// a multimap instead of a priority_queue since the priority of a stream can
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// change as a new packet is inserted, and a multimap allows us to remove and
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// then reinsert a StreamPrioKey if the priority has increased.
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std::multimap<StreamPrioKey, uint32_t> stream_priorities_;
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// A map of SSRCs to Streams.
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std::map<uint32_t, Stream> streams_;
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// The enqueue time of every packet currently in the queue. Used to figure out
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// the age of the oldest packet in the queue.
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std::multiset<int64_t> enqueue_times_;
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};
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} // namespace webrtc
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#endif // MODULES_PACING_PACKET_QUEUE2_H_
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