blob: 9446a8e174ae6ce29e9a6e4682356fb0643ae05d [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 "api/transport/webrtc_key_value_config.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.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 {
RoundRobinPacketQueue(Timestamp start_time,
const WebRtcKeyValueConfig* field_trials);
void Push(int priority,
Timestamp enqueue_time,
uint64_t enqueue_order,
std::unique_ptr<RtpPacketToSend> packet);
std::unique_ptr<RtpPacketToSend> Pop();
bool Empty() const;
size_t SizeInPackets() const;
DataSize Size() const;
// If the next packet, that would be returned by Pop() if called
// now, is an audio packet this method returns the enqueue time
// of that packet. If queue is empty or top packet is not audio,
// returns nullopt.
absl::optional<Timestamp> LeadingAudioPacketEnqueueTime() const;
Timestamp OldestEnqueueTime() const;
TimeDelta AverageQueueTime() const;
void UpdateQueueTime(Timestamp now);
void SetPauseState(bool paused, Timestamp now);
void SetIncludeOverhead();
void SetTransportOverhead(DataSize overhead_per_packet);
struct QueuedPacket {
QueuedPacket(int priority,
Timestamp enqueue_time,
uint64_t enqueue_order,
std::multiset<Timestamp>::iterator enqueue_time_it,
std::unique_ptr<RtpPacketToSend> packet);
QueuedPacket(const QueuedPacket& rhs);
bool operator<(const QueuedPacket& other) const;
int Priority() const;
RtpPacketMediaType Type() const;
uint32_t Ssrc() const;
Timestamp EnqueueTime() const;
bool IsRetransmission() const;
uint64_t EnqueueOrder() const;
RtpPacketToSend* RtpPacket() const;
std::multiset<Timestamp>::iterator EnqueueTimeIterator() const;
void UpdateEnqueueTimeIterator(std::multiset<Timestamp>::iterator it);
void SubtractPauseTime(TimeDelta pause_time_sum);
int priority_;
Timestamp enqueue_time_; // Absolute time of pacer queue entry.
uint64_t enqueue_order_;
bool is_retransmission_; // Cached for performance.
std::multiset<Timestamp>::iterator enqueue_time_it_;
// Raw pointer since priority_queue doesn't allow for moving
// out of the container.
RtpPacketToSend* owned_packet_;
class PriorityPacketQueue : public std::priority_queue<QueuedPacket> {
using const_iterator = container_type::const_iterator;
const_iterator begin() const;
const_iterator end() const;
struct StreamPrioKey {
StreamPrioKey(int priority, DataSize size)
: priority(priority), size(size) {}
bool operator<(const StreamPrioKey& other) const {
if (priority != other.priority)
return priority < other.priority;
return size < other.size;
const int priority;
const DataSize size;
struct Stream {
Stream(const Stream&);
virtual ~Stream();
DataSize size;
uint32_t ssrc;
PriorityPacketQueue 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;
void Push(QueuedPacket packet);
DataSize PacketSize(const QueuedPacket& packet) const;
void MaybePromoteSinglePacketToNormalQueue();
Stream* GetHighestPriorityStream();
// Just used to verify correctness.
bool IsSsrcScheduled(uint32_t ssrc) const;
DataSize transport_overhead_per_packet_;
Timestamp time_last_updated_;
bool paused_;
size_t size_packets_;
DataSize size_;
DataSize max_size_;
TimeDelta queue_time_sum_;
TimeDelta pause_time_sum_;
// 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<Timestamp> enqueue_times_;
absl::optional<QueuedPacket> single_packet_queue_;
bool include_overhead_;
} // namespace webrtc