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/*
* 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 <stddef.h>
#include <stdint.h>
#include <list>
#include <map>
#include <queue>
#include <set>
#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<Packet>::iterator this_it;
std::multiset<int64_t>::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> 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<StreamPrioKey, uint32_t>::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<Packet> pop_packet_;
absl::optional<Stream*> 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<StreamPrioKey, uint32_t> stream_priorities_;
// A map of SSRCs to Streams.
std::map<uint32_t, Stream> 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<int64_t> enqueue_times_;
};
} // namespace webrtc
#endif // MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_