6673437775
When congestion window is used, two different mechanisms can currently update the outstanding data state in the pacer: * OnPacketSent() withing the pacer itself, when a packet is sent * UpdateOutstandingData(), when RtpTransportControllerSend either: a. Receives an OnPacketSent() callback (increase outstanding data) b. Receives transport feedback (decrease outstanding data) This creates a lot of calls to UpdateOutstandingData(), more than one per sent packet. Each requires locking and/or thread jumps. To avoid that, this CL moves the congestion window state to RtpTransportController send - and we only post a congested flag down the the pacer when the state is changed. The only benefit I can see is of the old way is we prevent sending new packets immedately when the window is full, rather than in some edge cases queue extra packets on the network task queue before the congestion signal is received. That should be rare and benign. I think this simplified logic, which is easier to read and more performant, is a better tradeoff. Bug: webrtc:13417 Change-Id: I326dd88db86dc0d6dc685c61920654ac024e57ef Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255600 Auto-Submit: Erik Språng <sprang@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36220}
72 lines
2.6 KiB
C++
72 lines
2.6 KiB
C++
/*
|
|
* Copyright (c) 2019 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_RTP_PACKET_PACER_H_
|
|
#define MODULES_PACING_RTP_PACKET_PACER_H_
|
|
|
|
#include <stdint.h>
|
|
|
|
#include "absl/types/optional.h"
|
|
#include "api/units/data_rate.h"
|
|
#include "api/units/data_size.h"
|
|
#include "api/units/time_delta.h"
|
|
#include "api/units/timestamp.h"
|
|
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
|
|
|
|
namespace webrtc {
|
|
|
|
class RtpPacketPacer {
|
|
public:
|
|
virtual ~RtpPacketPacer() = default;
|
|
|
|
virtual void CreateProbeCluster(DataRate bitrate, int cluster_id) = 0;
|
|
|
|
// Temporarily pause all sending.
|
|
virtual void Pause() = 0;
|
|
|
|
// Resume sending packets.
|
|
virtual void Resume() = 0;
|
|
|
|
virtual void SetCongested(bool congested) = 0;
|
|
|
|
// Sets the pacing rates. Must be called once before packets can be sent.
|
|
virtual void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) = 0;
|
|
|
|
// Time since the oldest packet currently in the queue was added.
|
|
virtual TimeDelta OldestPacketWaitTime() const = 0;
|
|
|
|
// Sum of payload + padding bytes of all packets currently in the pacer queue.
|
|
virtual DataSize QueueSizeData() const = 0;
|
|
|
|
// Returns the time when the first packet was sent.
|
|
virtual absl::optional<Timestamp> FirstSentPacketTime() const = 0;
|
|
|
|
// Returns the expected number of milliseconds it will take to send the
|
|
// current packets in the queue, given the current size and bitrate, ignoring
|
|
// priority.
|
|
virtual TimeDelta ExpectedQueueTime() const = 0;
|
|
|
|
// Set the average upper bound on pacer queuing delay. The pacer may send at
|
|
// a higher rate than what was configured via SetPacingRates() in order to
|
|
// keep ExpectedQueueTimeMs() below `limit_ms` on average.
|
|
virtual void SetQueueTimeLimit(TimeDelta limit) = 0;
|
|
|
|
// Currently audio traffic is not accounted by pacer and passed through.
|
|
// With the introduction of audio BWE audio traffic will be accounted for
|
|
// the pacer budget calculation. The audio traffic still will be injected
|
|
// at high priority.
|
|
virtual void SetAccountForAudioPackets(bool account_for_audio) = 0;
|
|
virtual void SetIncludeOverhead() = 0;
|
|
virtual void SetTransportOverhead(DataSize overhead_per_packet) = 0;
|
|
};
|
|
|
|
} // namespace webrtc
|
|
#endif // MODULES_PACING_RTP_PACKET_PACER_H_
|