webrtc_m130/modules/pacing/paced_sender.h

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

149 lines
5.6 KiB
C
Raw Normal View History

/*
* Copyright (c) 2012 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_PACED_SENDER_H_
#define MODULES_PACING_PACED_SENDER_H_
#include <stddef.h>
#include <stdint.h>
#include <atomic>
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/function_view.h"
#include "api/transport/field_trial_based_config.h"
#include "api/transport/network_types.h"
#include "api/transport/webrtc_key_value_config.h"
#include "modules/include/module.h"
#include "modules/pacing/bitrate_prober.h"
#include "modules/pacing/interval_budget.h"
#include "modules/pacing/pacing_controller.h"
#include "modules/pacing/packet_router.h"
#include "modules/pacing/rtp_packet_pacer.h"
#include "modules/rtp_rtcp/include/rtp_packet_sender.h"
#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
#include "modules/utility/include/process_thread.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class Clock;
class RtcEventLog;
class PacedSender : public Module,
public RtpPacketPacer,
public RtpPacketSender,
private PacingController::PacketSender {
public:
// Expected max pacer delay in ms. If ExpectedQueueTime() is higher than
// this value, the packet producers should wait (eg drop frames rather than
// encoding them). Bitrate sent may temporarily exceed target set by
// UpdateBitrate() so that this limit will be upheld.
static const int64_t kMaxQueueLengthMs;
Revert "Revert "Revert "Reland "Moved congestion controller to task queue."""" This reverts commit 65792c5a5c542201f7b9feefded505842692e6ed. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Revert "Reland "Moved congestion controller to task queue.""" > > This reverts commit 4e849f6925b2ac44b0957a228d7131fc391fca54. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Reland "Moved congestion controller to task queue."" > > > > This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793. > > > > Reason for revert: Cause increased congestion and deadlocks in downstream project > > > > Original change's description: > > > Reland "Moved congestion controller to task queue." > > > > > > This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9. > > > > > > Original change's description: > > > > Moved congestion controller to task queue. > > > > > > > > The goal of this work is to make it easier to experiment with the > > > > bandwidth estimation implementation. For this reason network control > > > > functionality is moved from SendSideCongestionController(SSCC), > > > > PacedSender and BitrateController to the newly created > > > > GoogCcNetworkController which implements the newly created > > > > NetworkControllerInterface. This allows the implementation to be > > > > replaced at runtime in the future. > > > > > > > > This is the first part of a split of a larger CL, see: > > > > https://webrtc-review.googlesource.com/c/src/+/39788/8 > > > > For further explanations. > > > > > > > > Bug: webrtc:8415 > > > > Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 > > > > Reviewed-on: https://webrtc-review.googlesource.com/43840 > > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > > Reviewed-by: Björn Terelius <terelius@webrtc.org> > > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > > Cr-Commit-Position: refs/heads/master@{#21868} > > > > > > Bug: webrtc:8415 > > > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da > > > Reviewed-on: https://webrtc-review.googlesource.com/48000 > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21899} > > > > TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:8415 > > Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83 > > Reviewed-on: https://webrtc-review.googlesource.com/52980 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22017} > > TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8415 > Reviewed-on: https://webrtc-review.googlesource.com/53262 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22023} TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Id68ad986ee51142b7be3381d0793709b4392fe2c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8415 Reviewed-on: https://webrtc-review.googlesource.com/53360 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22024}
2018-02-14 16:53:38 +00:00
// Pacing-rate relative to our target send rate.
// Multiplicative factor that is applied to the target bitrate to calculate
// the number of bytes that can be transmitted per interval.
// Increasing this factor will result in lower delays in cases of bitrate
// overshoots from the encoder.
static const float kDefaultPaceMultiplier;
PacedSender(Clock* clock,
PacketRouter* packet_router,
RtcEventLog* event_log,
const WebRtcKeyValueConfig* field_trials = nullptr);
~PacedSender() override;
// Methods implementing RtpPacketSender.
// Adds the packet to the queue and calls PacketRouter::SendPacket() when
// it's time to send.
void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) override;
// Methods implementing RtpPacketPacer:
void CreateProbeCluster(DataRate bitrate, int cluster_id) override;
// Temporarily pause all sending.
void Pause() override;
// Resume sending packets.
void Resume() override;
void SetCongestionWindow(DataSize congestion_window_size) override;
void UpdateOutstandingData(DataSize outstanding_data) override;
// Sets the pacing rates. Must be called once before packets can be sent.
void SetPacingRates(DataRate pacing_rate, DataRate padding_rate) override;
// 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.
void SetAccountForAudioPackets(bool account_for_audio) override;
// Returns the time since the oldest queued packet was enqueued.
TimeDelta OldestPacketWaitTime() const override;
size_t QueueSizePackets() const override;
DataSize QueueSizeData() const override;
// Returns the time when the first packet was sent;
absl::optional<Timestamp> FirstSentPacketTime() const override;
// Returns the number of milliseconds it will take to send the current
// packets in the queue, given the current size and bitrate, ignoring prio.
TimeDelta ExpectedQueueTime() const override;
void SetQueueTimeLimit(TimeDelta limit) override;
// Below are methods specific to this implementation, such as things related
// to module processing thread specifics or methods exposed for test.
// Methods implementing Module.
// Returns the number of milliseconds until the module want a worker thread
// to call Process.
int64_t TimeUntilNextProcess() override;
// Process any pending packets in the queue(s).
void Process() override;
// Called when the prober is associated with a process thread.
void ProcessThreadAttached(ProcessThread* process_thread) override;
private:
// Methods implementing PacedSenderController:PacketSender.
void SendRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
const PacedPacketInfo& cluster_info) override
RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
DataSize size) override RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
rtc::CriticalSection critsect_;
PacingController pacing_controller_ RTC_GUARDED_BY(critsect_);
PacketRouter* const packet_router_;
// Lock to avoid race when attaching process thread. This can happen due to
// the Call class setting network state on RtpTransportControllerSend, which
// in turn calls Pause/Resume on Pacedsender, before actually starting the
// pacer process thread. If RtpTransportControllerSend is running on a task
// queue separate from the thread used by Call, this causes a race.
rtc::CriticalSection process_thread_lock_;
ProcessThread* process_thread_ RTC_GUARDED_BY(process_thread_lock_);
};
} // namespace webrtc
#endif // MODULES_PACING_PACED_SENDER_H_