/* * 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_ROUND_ROBIN_PACKET_QUEUE_H_ #define MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_ #include #include #include #include #include #include #include "absl/types/optional.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "system_wrappers/include/clock.h" namespace webrtc { class RoundRobinPacketQueue { public: explicit RoundRobinPacketQueue(int64_t start_time_us); ~RoundRobinPacketQueue(); struct Packet { Packet(RtpPacketSender::Priority priority, uint32_t ssrc, uint16_t seq_number, int64_t capture_time_ms, int64_t enqueue_time_ms, size_t length_in_bytes, bool retransmission, uint64_t enqueue_order); Packet(const Packet& other); virtual ~Packet(); bool operator<(const Packet& other) const; RtpPacketSender::Priority priority; uint32_t ssrc; uint16_t sequence_number; int64_t capture_time_ms; // Absolute time of frame capture. int64_t enqueue_time_ms; // Absolute time of pacer queue entry. int64_t sum_paused_ms; size_t bytes; bool retransmission; uint64_t enqueue_order; std::list::iterator this_it; std::multiset::iterator enqueue_time_it; }; void Push(const Packet& packet); const Packet& BeginPop(); void CancelPop(const Packet& packet); void FinalizePop(const Packet& packet); bool Empty() const; size_t SizeInPackets() const; uint64_t SizeInBytes() const; int64_t OldestEnqueueTimeMs() const; int64_t AverageQueueTimeMs() const; void UpdateQueueTime(int64_t timestamp_ms); void SetPauseState(bool paused, int64_t timestamp_ms); private: struct StreamPrioKey { 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(); Stream(const Stream&); virtual ~Stream(); size_t bytes; uint32_t ssrc; std::priority_queue 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::iterator priority_it; }; static constexpr size_t kMaxLeadingBytes = 1400; Stream* GetHighestPriorityStream(); // Just used to verify correctness. bool IsSsrcScheduled(uint32_t ssrc) const; int64_t time_last_updated_ms_; absl::optional pop_packet_; absl::optional 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 stream_priorities_; // A map of SSRCs to Streams. std::map 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 enqueue_times_; }; } // namespace webrtc #endif // MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_