blob: 3bc628d6a504a243a548f3996e4ebc4f6d101cb2 [file] [log] [blame]
* 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.
#include <stddef.h>
#include <stdint.h>
#include <atomic>
#include <memory>
#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/packet_router.h"
#include "modules/pacing/round_robin_packet_queue.h"
#include "modules/rtp_rtcp/include/rtp_packet_pacer.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.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/experiments/field_trial_parser.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class Clock;
class RtcEventLog;
class PacedSender : public Module, public RtpPacketPacer {
static constexpr int64_t kNoCongestionWindow = -1;
// 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(Clock* clock,
PacketRouter* packet_router,
RtcEventLog* event_log,
const WebRtcKeyValueConfig* field_trials = nullptr);
~PacedSender() override;
virtual void CreateProbeCluster(int bitrate_bps, int cluster_id);
// Temporarily pause all sending.
void Pause();
// Resume sending packets.
void Resume();
void SetCongestionWindow(int64_t congestion_window_bytes);
void UpdateOutstandingData(int64_t outstanding_bytes);
// 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 pacing rates. Must be called once before packets can be sent.
void SetPacingRates(uint32_t pacing_rate_bps, uint32_t padding_rate_bps);
// Adds the packet information to the queue and calls 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;
// Adds the packet to the queue and calls PacketRouter::SendPacket() when
// it's time to send.
void EnqueuePacket(std::unique_ptr<RtpPacketToSend> packet) 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;
virtual int64_t QueueSizeBytes() 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 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 SetQueueTimeLimit(int limit_ms);
int64_t UpdateTimeAndGetElapsedMs(int64_t now_us)
bool ShouldSendKeepalive(int64_t at_time_us) const
// Updates the number of bytes that can be sent for the next time interval.
void UpdateBudgetWithElapsedTime(int64_t delta_time_in_ms)
void UpdateBudgetWithBytesSent(size_t bytes)
size_t PaddingBytesToAdd(absl::optional<size_t> recommended_probe_size,
size_t bytes_sent)
RoundRobinPacketQueue::QueuedPacket* GetPendingPacket(
const PacedPacketInfo& pacing_info)
void OnPacketSent(RoundRobinPacketQueue::QueuedPacket* packet)
void OnPaddingSent(size_t padding_sent)
bool Congested() const RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
int64_t TimeMilliseconds() const RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
Clock* const clock_;
PacketRouter* const packet_router_;
const std::unique_ptr<FieldTrialBasedConfig> fallback_field_trials_;
const WebRtcKeyValueConfig* field_trials_;
const bool drain_large_queues_;
const bool send_padding_if_silent_;
const bool pace_audio_;
FieldTrialParameter<int> min_packet_limit_ms_;
rtc::CriticalSection critsect_;
// TODO(webrtc:9716): Remove this when we are certain clocks are monotonic.
// The last millisecond timestamp returned by |clock_|.
mutable int64_t last_timestamp_ms_ RTC_GUARDED_BY(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.
IntervalBudget media_budget_ RTC_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.
IntervalBudget padding_budget_ RTC_GUARDED_BY(critsect_);
BitrateProber prober_ RTC_GUARDED_BY(critsect_);
bool probing_send_failure_ RTC_GUARDED_BY(critsect_);
uint32_t pacing_bitrate_kbps_ RTC_GUARDED_BY(critsect_);
int64_t time_last_process_us_ RTC_GUARDED_BY(critsect_);
int64_t last_send_time_us_ RTC_GUARDED_BY(critsect_);
int64_t first_sent_packet_ms_ RTC_GUARDED_BY(critsect_);
RoundRobinPacketQueue packets_ RTC_GUARDED_BY(critsect_);
uint64_t packet_counter_ RTC_GUARDED_BY(critsect_);
int64_t congestion_window_bytes_ RTC_GUARDED_BY(critsect_) =
int64_t outstanding_bytes_ RTC_GUARDED_BY(critsect_) = 0;
// 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_) = nullptr;
int64_t queue_time_limit RTC_GUARDED_BY(critsect_);
bool account_for_audio_ RTC_GUARDED_BY(critsect_);
// If true, PacedSender should only reference packets as in legacy mode.
// If false, PacedSender may have direct ownership of RtpPacketToSend objects.
// Defaults to true, will be changed to default false soon.
const bool legacy_packet_referencing_;
} // namespace webrtc