blob: 4bab4fab99fe7fbde300eb6ed23f4fec3cb539b4 [file] [log] [blame]
* 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.
#include <stddef.h>
#include <stdint.h>
#include <list>
#include <map>
#include <memory>
#include <queue>
#include <set>
#include "absl/types/optional.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "system_wrappers/include/clock.h"
namespace webrtc {
class RoundRobinPacketQueue {
explicit RoundRobinPacketQueue(int64_t start_time_us);
struct QueuedPacket {
int priority,
RtpPacketToSend::Type type,
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,
std::multiset<int64_t>::iterator enqueue_time_it,
QueuedPacket(const QueuedPacket& rhs);
bool operator<(const QueuedPacket& other) const;
int priority() const { return priority_; }
RtpPacketToSend::Type type() const { return type_; }
uint32_t ssrc() const { return ssrc_; }
uint16_t sequence_number() const { return sequence_number_; }
int64_t capture_time_ms() const { return capture_time_ms_; }
int64_t enqueue_time_ms() const { return enqueue_time_ms_; }
size_t size_in_bytes() const { return bytes_; }
bool is_retransmission() const { return retransmission_; }
uint64_t enqueue_order() const { return enqueue_order_; }
std::unique_ptr<RtpPacketToSend> ReleasePacket();
// For internal use.
PacketIterator() const {
return packet_it_;
std::multiset<int64_t>::iterator EnqueueTimeIterator() const {
return enqueue_time_it_;
void SubtractPauseTimeMs(int64_t pause_time_sum_ms);
RtpPacketToSend::Type type_;
int 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.
size_t bytes_;
bool retransmission_;
uint64_t enqueue_order_;
std::multiset<int64_t>::iterator enqueue_time_it_;
// Iterator into |rtp_packets_| where the memory for RtpPacket is owned,
// if applicable.
void Push(int priority,
RtpPacketToSend::Type type,
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);
void Push(int priority,
int64_t enqueue_time_ms,
uint64_t enqueue_order,
std::unique_ptr<RtpPacketToSend> packet);
QueuedPacket* BeginPop();
void CancelPop();
void FinalizePop();
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);
struct StreamPrioKey {
StreamPrioKey(int 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 int priority;
const size_t bytes;
struct Stream {
Stream(const Stream&);
virtual ~Stream();
size_t bytes;
uint32_t ssrc;
std::priority_queue<QueuedPacket> 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;
void Push(QueuedPacket packet);
Stream* GetHighestPriorityStream();
// Just used to verify correctness.
bool IsSsrcScheduled(uint32_t ssrc) const;
int64_t time_last_updated_ms_;
absl::optional<QueuedPacket> 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_;
// List of RTP packets to be sent, not necessarily in the order they will be
// sent. PacketInfo.packet_it will point to an entry in this list, or the
// end iterator of this list if queue does not have direct ownership of the
// packet.
std::list<std::unique_ptr<RtpPacketToSend>> rtp_packets_;
} // namespace webrtc