blob: f142f5517330e24bd5d451f5bbdd7eccae3a9362 [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 <list>
#include <set>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/interface/module.h"
#include "webrtc/modules/rtp_rtcp/interface/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 {
class Callback {
// 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) = 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) = 0;
virtual ~Callback() {}
static const int64_t kDefaultMaxQueueLengthMs = 2000;
// Pace in kbits/s until we receive first estimate.
static const int kDefaultInitialPaceKbps = 2000;
// 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,
Callback* callback,
int bitrate_kbps,
int max_bitrate_kbps,
int min_bitrate_kbps);
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);
// Set target bitrates for the pacer.
// We will pace out bursts of packets at a bitrate of |max_bitrate_kbps|.
// |bitrate_kbps| is our estimate of what we are allowed to send on average.
// Padding packets will be utilized to reach |min_bitrate| unless enough media
// packets are available.
void UpdateBitrate(int bitrate_kbps,
int max_bitrate_kbps,
int min_bitrate_kbps);
// 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 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).
int32_t Process() override;
// Updates the number of bytes that can be sent for the next time interval.
void UpdateBytesPerInterval(int64_t delta_time_in_ms)
bool SendPacket(const paced_sender::Packet& packet)
void SendPadding(size_t padding_needed) EXCLUSIVE_LOCKS_REQUIRED(critsect_);
Clock* const clock_;
Callback* const callback_;
rtc::scoped_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.
rtc::scoped_ptr<paced_sender::IntervalBudget> media_budget_
// 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.
rtc::scoped_ptr<paced_sender::IntervalBudget> padding_budget_
rtc::scoped_ptr<BitrateProber> prober_ GUARDED_BY(critsect_);
int bitrate_bps_ GUARDED_BY(critsect_);
int64_t time_last_update_us_ GUARDED_BY(critsect_);
rtc::scoped_ptr<paced_sender::PacketQueue> packets_ GUARDED_BY(critsect_);
uint64_t packet_counter_;
} // namespace webrtc