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/*
* 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 <memory>
#include "api/optional.h"
#include "modules/pacing/pacer.h"
#include "modules/pacing/packet_queue2.h"
#include "rtc_base/criticalsection.h"
#include "rtc_base/thread_annotations.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
class AlrDetector;
class BitrateProber;
class Clock;
class ProbeClusterCreatedObserver;
class RtcEventLog;
class IntervalBudget;
class PacedSender : public Pacer {
public:
class PacketSender {
public:
// Note: packets sent as a result of a callback should not pass by this
// module again.
// Called when it's time to send a queued packet.
// Returns false if packet cannot be sent.
virtual bool TimeToSendPacket(uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
bool retransmission,
const PacedPacketInfo& cluster_info) = 0;
// Called when it's a good time to send a padding data.
// Returns the number of bytes sent.
virtual size_t TimeToSendPadding(size_t bytes,
const PacedPacketInfo& cluster_info) = 0;
protected:
virtual ~PacketSender() {}
};
// Expected max pacer delay in ms. If ExpectedQueueTimeMs() 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;
// 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(const Clock* clock,
PacketSender* packet_sender,
RtcEventLog* event_log);
PacedSender(const Clock* clock,
PacketSender* packet_sender,
RtcEventLog* event_log,
std::unique_ptr<PacketQueue> packets);
~PacedSender() override;
virtual void CreateProbeCluster(int bitrate_bps);
// Temporarily pause all sending.
void Pause();
// Resume sending packets.
void Resume();
// Enable bitrate probing. Enabled by default, mostly here to simplify
// testing. Must be called before any packets are being sent to have an
// effect.
void SetProbingEnabled(bool enabled);
// Sets the estimated capacity of the network. Must be called once before
// packets can be sent.
// |bitrate_bps| is our estimate of what we are allowed to send on average.
// We will pace out bursts of packets at a bitrate of
// |bitrate_bps| * kDefaultPaceMultiplier.
void SetEstimatedBitrate(uint32_t bitrate_bps) override;
// Sets the minimum send bitrate and maximum padding bitrate requested by send
// streams.
// |min_send_bitrate_bps| might be higher that the estimated available network
// bitrate and if so, the pacer will send with |min_send_bitrate_bps|.
// |max_padding_bitrate_bps| might be higher than the estimate available
// network bitrate and if so, the pacer will send padding packets to reach
// the min of the estimated available bitrate and |max_padding_bitrate_bps|.
void SetSendBitrateLimits(int min_send_bitrate_bps,
int max_padding_bitrate_bps);
// Returns true if we send the packet now, else it will add the packet
// information to the queue and call TimeToSendPacket when it's time to send.
void InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) 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.
virtual int64_t QueueInMs() const;
virtual size_t QueueSizePackets() const;
// Returns the time when the first packet was sent, or -1 if no packet is
// sent.
virtual int64_t FirstSentPacketTimeMs() const;
// Returns the number of milliseconds it will take to send the current
// packets in the queue, given the current size and bitrate, ignoring prio.
virtual int64_t ExpectedQueueTimeMs() const;
// Returns time in milliseconds when the current application-limited region
// started or empty result if the sender is currently not application-limited.
//
// Application Limited Region (ALR) refers to operating in a state where the
// traffic on network is limited due to application not having enough
// traffic to meet the current channel capacity.
virtual rtc::Optional<int64_t> GetApplicationLimitedRegionStartTime() const;
// 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;
void SetPacingFactor(float pacing_factor);
float GetPacingFactor() const;
void SetQueueTimeLimit(int limit_ms);
private:
// Updates the number of bytes that can be sent for the next time interval.
void UpdateBudgetWithElapsedTime(int64_t delta_time_in_ms)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
void UpdateBudgetWithBytesSent(size_t bytes)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
bool SendPacket(const PacketQueue::Packet& packet,
const PacedPacketInfo& cluster_info)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
size_t SendPadding(size_t padding_needed, const PacedPacketInfo& cluster_info)
RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
const Clock* const clock_;
PacketSender* const packet_sender_;
const std::unique_ptr<AlrDetector> alr_detector_ RTC_PT_GUARDED_BY(critsect_);
rtc::CriticalSection critsect_;
bool paused_ RTC_GUARDED_BY(critsect_);
// This is the media budget, keeping track of how many bits of media
// we can pace out during the current interval.
const std::unique_ptr<IntervalBudget> media_budget_
RTC_PT_GUARDED_BY(critsect_);
// This is the padding budget, keeping track of how many bits of padding we're
// allowed to send out during the current interval. This budget will be
// utilized when there's no media to send.
const std::unique_ptr<IntervalBudget> padding_budget_
RTC_PT_GUARDED_BY(critsect_);
const std::unique_ptr<BitrateProber> prober_ RTC_PT_GUARDED_BY(critsect_);
bool probing_send_failure_ RTC_GUARDED_BY(critsect_);
// Actual configured bitrates (media_budget_ may temporarily be higher in
// order to meet pace time constraint).
uint32_t estimated_bitrate_bps_ RTC_GUARDED_BY(critsect_);
uint32_t min_send_bitrate_kbps_ RTC_GUARDED_BY(critsect_);
uint32_t max_padding_bitrate_kbps_ RTC_GUARDED_BY(critsect_);
uint32_t pacing_bitrate_kbps_ RTC_GUARDED_BY(critsect_);
int64_t time_last_update_us_ RTC_GUARDED_BY(critsect_);
int64_t first_sent_packet_ms_ RTC_GUARDED_BY(critsect_);
const std::unique_ptr<PacketQueue> packets_ RTC_PT_GUARDED_BY(critsect_);
uint64_t packet_counter_ RTC_GUARDED_BY(critsect_);
ProcessThread* process_thread_ = nullptr;
float pacing_factor_ RTC_GUARDED_BY(critsect_);
int64_t queue_time_limit RTC_GUARDED_BY(critsect_);
bool account_for_audio_ RTC_GUARDED_BY(critsect_);
};
} // namespace webrtc
#endif // MODULES_PACING_PACED_SENDER_H_