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This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
138 lines
4.2 KiB
C++
138 lines
4.2 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_ROUND_ROBIN_PACKET_QUEUE_H_
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#define MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <list>
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#include <map>
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#include <queue>
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#include <set>
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#include "absl/types/optional.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "system_wrappers/include/clock.h"
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namespace webrtc {
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class RoundRobinPacketQueue {
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public:
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explicit RoundRobinPacketQueue(int64_t start_time_us);
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~RoundRobinPacketQueue();
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struct Packet {
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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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Packet(const Packet& other);
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virtual ~Packet();
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bool operator<(const Packet& other) const;
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RtpPacketSender::Priority priority;
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uint32_t ssrc;
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uint16_t sequence_number;
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int64_t capture_time_ms; // Absolute time of frame capture.
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int64_t enqueue_time_ms; // Absolute time of pacer queue entry.
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int64_t sum_paused_ms;
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size_t bytes;
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bool retransmission;
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uint64_t enqueue_order;
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std::list<Packet>::iterator this_it;
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std::multiset<int64_t>::iterator enqueue_time_it;
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};
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void Push(const Packet& packet);
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const Packet& BeginPop();
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void CancelPop(const Packet& packet);
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void FinalizePop(const Packet& packet);
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bool Empty() const;
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size_t SizeInPackets() const;
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uint64_t SizeInBytes() const;
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int64_t OldestEnqueueTimeMs() const;
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int64_t AverageQueueTimeMs() const;
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void UpdateQueueTime(int64_t timestamp_ms);
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void SetPauseState(bool paused, int64_t timestamp_ms);
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private:
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struct StreamPrioKey {
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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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Stream(const 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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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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int64_t time_last_updated_ms_;
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absl::optional<Packet> pop_packet_;
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absl::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_ROUND_ROBIN_PACKET_QUEUE_H_
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