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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 WEBRTC_MODULES_PACING_PACED_SENDER_H_
#define WEBRTC_MODULES_PACING_PACED_SENDER_H_
#include <list>
#include <memory>
#include <set>
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class BitrateProber;
class Clock;
class CriticalSectionWrapper;
namespace paced_sender {
class IntervalBudget;
struct Packet;
class PacketQueue;
} // namespace paced_sender
class PacedSender : public Module, public RtpPacketSender {
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,
int probe_cluster_id) = 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, int probe_cluster_id) = 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;
static const size_t kMinProbePacketSize = 200;
PacedSender(Clock* clock,
PacketSender* packet_sender);
virtual ~PacedSender();
// 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.
virtual void SetEstimatedBitrate(uint32_t bitrate_bps);
// Sets the bitrate that has been allocated for encoders.
// |allocated_bitrate| might be higher that the estimated available network
// bitrate and if so, the pacer will send with |allocated_bitrate|.
// Padding packets will be utilized to reach |padding_bitrate| unless enough
// media packets are available.
void SetAllocatedSendBitrate(int allocated_bitrate_bps,
int 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;
// Returns the time since the oldest queued packet was enqueued.
virtual int64_t QueueInMs() const;
virtual size_t QueueSizePackets() 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 average time since being enqueued, in milliseconds, for all
// packets currently in the pacer queue, or 0 if queue is empty.
virtual int64_t AverageQueueTimeMs();
// 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;
private:
// Updates the number of bytes that can be sent for the next time interval.
void UpdateBytesPerInterval(int64_t delta_time_in_ms)
EXCLUSIVE_LOCKS_REQUIRED(critsect_);
bool SendPacket(const paced_sender::Packet& packet, int probe_cluster_id)
EXCLUSIVE_LOCKS_REQUIRED(critsect_);
void SendPadding(size_t padding_needed, int probe_cluster_id)
EXCLUSIVE_LOCKS_REQUIRED(critsect_);
Clock* const clock_;
PacketSender* const packet_sender_;
std::unique_ptr<CriticalSectionWrapper> critsect_;
bool paused_ GUARDED_BY(critsect_);
bool probing_enabled_;
// This is the media budget, keeping track of how many bits of media
// we can pace out during the current interval.
std::unique_ptr<paced_sender::IntervalBudget> media_budget_
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.
std::unique_ptr<paced_sender::IntervalBudget> padding_budget_
GUARDED_BY(critsect_);
std::unique_ptr<BitrateProber> prober_ GUARDED_BY(critsect_);
// Actual configured bitrates (media_budget_ may temporarily be higher in
// order to meet pace time constraint).
uint32_t estimated_bitrate_bps_ GUARDED_BY(critsect_);
uint32_t min_send_bitrate_kbps_ GUARDED_BY(critsect_);
uint32_t pacing_bitrate_kbps_ GUARDED_BY(critsect_);
int64_t time_last_update_us_ GUARDED_BY(critsect_);
std::unique_ptr<paced_sender::PacketQueue> packets_ GUARDED_BY(critsect_);
uint64_t packet_counter_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_PACING_PACED_SENDER_H_