Final version of BBR, with tweaks made for WebRTC, major changes:

1) Entering PROBE_RTT when necessary.
2) Congestion window gain of 0.65 instead of constant 4 packets.
3) {1.1, 0.9} pair instead of {1.25, 0.75}
4) Recovery mode.
5) No reaction to losses due to Recovery mode's implementation.
6) Supports encoder.
7) A new test compiling most of the simulation tests.
8) Bucket for high gain phase, disabled by default.
9) Pacer specific to BBR.

BUG=webrtc:7713

Review-Url: https://codereview.webrtc.org/2999073002
Cr-Commit-Position: refs/heads/master@{#19418}
This commit is contained in:
gnish
2017-08-20 09:19:58 -07:00
committed by Commit Bot
parent 801830691a
commit a36165c77b
22 changed files with 545 additions and 131 deletions

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@ -15,6 +15,8 @@
#ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
#define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
#include <map>
#include "webrtc/modules/congestion_controller/delay_based_bwe.h"
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/pacing/paced_sender.h"
@ -36,12 +38,18 @@ class BitrateObserver {
virtual void OnNetworkChanged(uint32_t bitrate_bps,
uint8_t fraction_loss, // 0 - 255.
int64_t rtt_ms) = 0;
// TODO(gnish): Merge these two into one function.
virtual void OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
uint32_t bitrate_for_pacer_bps,
bool in_probe_rtt,
int64_t target_set_time,
uint64_t congestion_window) {}
virtual void OnBytesAcked(size_t bytes) {}
virtual size_t pacer_queue_size_in_bytes() { return 0; }
virtual ~BitrateObserver() {}
};
class BitrateController : public Module,
public RtcpBandwidthObserver {
class BitrateController : public Module, public RtcpBandwidthObserver {
// This class collects feedback from all streams sent to a peer (via
// RTCPBandwidthObservers). It does one aggregated send side bandwidth
// estimation and divide the available bitrate between all its registered

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@ -18,6 +18,7 @@ rtc_static_library("pacing") {
"interval_budget.h",
"paced_sender.cc",
"paced_sender.h",
"pacer.h",
"packet_router.cc",
"packet_router.h",
]

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@ -37,8 +37,8 @@ const int64_t kMaxIntervalTimeMs = 30;
} // namespace
// TODO(sprang): Move at least PacketQueue out to separate
// files, so that we can more easily test them.
// TODO(sprang): Move at least PacketQueue out to separate files, so that we can
// more easily test them.
namespace webrtc {
namespace paced_sender {

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@ -15,8 +15,7 @@
#include <memory>
#include <set>
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/pacing/pacer.h"
#include "webrtc/rtc_base/criticalsection.h"
#include "webrtc/rtc_base/optional.h"
#include "webrtc/rtc_base/thread_annotations.h"
@ -31,11 +30,12 @@ class RtcEventLog;
class IntervalBudget;
namespace paced_sender {
class IntervalBudget;
struct Packet;
class PacketQueue;
} // namespace paced_sender
class PacedSender : public Module, public RtpPacketSender {
class PacedSender : public Pacer {
public:
class PacketSender {
public:
@ -93,7 +93,7 @@ class PacedSender : public Module, public RtpPacketSender {
// |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);
void SetEstimatedBitrate(uint32_t bitrate_bps) override;
// Sets the minimum send bitrate and maximum padding bitrate requested by send
// streams.
@ -149,7 +149,6 @@ class PacedSender : public Module, public RtpPacketSender {
// Called when the prober is associated with a process thread.
void ProcessThreadAttached(ProcessThread* process_thread) override;
void SetPacingFactor(float pacing_factor);
void SetQueueTimeLimit(int limit_ms);

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@ -0,0 +1,37 @@
/*
* Copyright (c) 2017 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_PACER_H_
#define WEBRTC_MODULES_PACING_PACER_H_
#include "webrtc/modules/include/module.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
class Pacer : public Module, public RtpPacketSender {
public:
virtual void SetEstimatedBitrate(uint32_t bitrate_bps) {}
virtual void SetEstimatedBitrateAndCongestionWindow(
uint32_t bitrate_bps,
bool in_probe_rtt,
uint64_t congestion_window) {}
virtual void OnBytesAcked(size_t bytes) {}
void InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) override = 0;
int64_t TimeUntilNextProcess() override = 0;
void Process() override = 0;
~Pacer() override {}
};
} // namespace webrtc
#endif // WEBRTC_MODULES_PACING_PACER_H_

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@ -61,6 +61,8 @@ if (rtc_include_tests) {
rtc_static_library("bwe_simulator_lib") {
testonly = true
sources = [
"test/bbr_paced_sender.cc",
"test/bbr_paced_sender.h",
"test/bwe.cc",
"test/bwe.h",
"test/bwe_test.cc",

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@ -122,9 +122,64 @@ TEST_P(BweSimulation, Choke1000kbps500kbps1000kbps) {
RunFor(60 * 1000);
}
TEST_P(BweSimulation, SimulationsCompiled) {
AdaptiveVideoSource source(0, 30, 300, 0, 0);
PacedVideoSender sender(&uplink_, &source, GetParam());
int zero = 0;
// CreateFlowIds() doesn't support passing int as a flow id, so we pass
// pointer instead.
DelayFilter delay(&uplink_, CreateFlowIds(&zero, 1));
delay.SetOneWayDelayMs(100);
ChokeFilter filter(&uplink_, 0);
RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
filter.set_max_delay_ms(500);
filter.set_capacity_kbps(1000);
RunFor(60 * 1000);
filter.set_capacity_kbps(500);
RunFor(50 * 1000);
filter.set_capacity_kbps(1000);
RunFor(60 * 1000);
filter.set_capacity_kbps(200);
RunFor(60 * 1000);
filter.set_capacity_kbps(50);
RunFor(60 * 1000);
filter.set_capacity_kbps(200);
RunFor(60 * 1000);
filter.set_capacity_kbps(500);
RunFor(60 * 1000);
filter.set_capacity_kbps(300);
RunFor(60 * 1000);
filter.set_capacity_kbps(1000);
RunFor(60 * 1000);
const int kStartingCapacityKbps = 150;
const int kEndingCapacityKbps = 1500;
const int kStepKbps = 5;
const int kStepTimeMs = 1000;
for (int i = kStartingCapacityKbps; i <= kEndingCapacityKbps;
i += kStepKbps) {
filter.set_capacity_kbps(i);
RunFor(kStepTimeMs);
}
for (int i = kEndingCapacityKbps; i >= kStartingCapacityKbps;
i -= kStepKbps) {
filter.set_capacity_kbps(i);
RunFor(kStepTimeMs);
}
filter.set_capacity_kbps(150);
RunFor(120 * 1000);
filter.set_capacity_kbps(500);
RunFor(60 * 1000);
}
TEST_P(BweSimulation, PacerChoke1000kbps500kbps1000kbps) {
AdaptiveVideoSource source(0, 30, 300, 0, 0);
PacedVideoSender sender(&uplink_, &source, GetParam());
const int kFlowId = 0;
// CreateFlowIds() doesn't support passing int as a flow id, so we pass
// pointer instead.
DelayFilter delay(&uplink_, CreateFlowIds(&kFlowId, 1));
delay.SetOneWayDelayMs(100);
ChokeFilter filter(&uplink_, 0);
RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
@ -262,6 +317,11 @@ TEST_P(BweSimulation, LinearDecreasingCapacity) {
TEST_P(BweSimulation, PacerGoogleWifiTrace3Mbps) {
PeriodicKeyFrameSource source(0, 30, 300, 0, 0, 1000);
PacedVideoSender sender(&uplink_, &source, GetParam());
int kFlowId = 0;
// CreateFlowIds() doesn't support passing int as a flow id, so we pass
// pointer instead.
DelayFilter delay(&uplink_, CreateFlowIds(&kFlowId, 1));
delay.SetOneWayDelayMs(100);
RateCounterFilter counter1(&uplink_, 0, "sender_output",
bwe_names[GetParam()]);
TraceBasedDeliveryFilter filter(&uplink_, 0, "link_capacity");

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@ -0,0 +1,140 @@
/*
* Copyright (c) 2017 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 "webrtc/modules/remote_bitrate_estimator/test/bbr_paced_sender.h"
#include <algorithm>
#include <queue>
#include <set>
#include <vector>
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
BbrPacedSender::BbrPacedSender(const Clock* clock,
PacedSender::PacketSender* packet_sender,
RtcEventLog* event_log)
: clock_(clock),
packet_sender_(packet_sender),
estimated_bitrate_bps_(100000),
min_send_bitrate_kbps_(0),
pacing_bitrate_kbps_(0),
time_last_update_us_(clock->TimeInMicroseconds()),
time_last_update_ms_(clock->TimeInMilliseconds()),
next_packet_send_time_(clock_->TimeInMilliseconds()),
rounding_error_time_ms_(0.0f),
packets_(),
max_data_inflight_bytes_(10000),
congestion_window_(new testing::bwe::CongestionWindow()) {}
BbrPacedSender::~BbrPacedSender() {}
void BbrPacedSender::SetEstimatedBitrateAndCongestionWindow(
uint32_t bitrate_bps,
bool in_probe_rtt,
uint64_t congestion_window) {
estimated_bitrate_bps_ = bitrate_bps;
max_data_inflight_bytes_ = congestion_window;
}
void BbrPacedSender::SetMinBitrate(int min_send_bitrate_bps) {
min_send_bitrate_kbps_ = min_send_bitrate_bps / 1000;
pacing_bitrate_kbps_ =
std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000);
}
void BbrPacedSender::InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) {
int64_t now_ms = clock_->TimeInMilliseconds();
if (capture_time_ms < 0)
capture_time_ms = now_ms;
packets_.push_back(new Packet(priority, ssrc, sequence_number,
capture_time_ms, now_ms, bytes,
retransmission));
}
int64_t BbrPacedSender::TimeUntilNextProcess() {
// Once errors absolute value hits 1 millisecond, add compensating term to
// the |next_packet_send_time_|, so that we can send packet earlier or later,
// depending on the error.
rounding_error_time_ms_ = std::min(rounding_error_time_ms_, 1.0f);
if (rounding_error_time_ms_ < -0.9f)
rounding_error_time_ms_ = -1.0f;
int64_t result =
std::max<int64_t>(next_packet_send_time_ + time_last_update_ms_ -
clock_->TimeInMilliseconds(),
0);
if (rounding_error_time_ms_ == 1.0f || rounding_error_time_ms_ == -1.0f) {
next_packet_send_time_ -= rounding_error_time_ms_;
result = std::max<int64_t>(next_packet_send_time_ + time_last_update_ms_ -
clock_->TimeInMilliseconds(),
0);
rounding_error_time_ms_ = 0;
}
return result;
}
void BbrPacedSender::OnBytesAcked(size_t bytes) {
congestion_window_->AckReceived(bytes);
}
void BbrPacedSender::Process() {
pacing_bitrate_kbps_ =
std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000);
// If we have nothing to send, try sending again in 1 millisecond.
if (packets_.empty()) {
next_packet_send_time_ = 1;
return;
}
// If congestion window doesn't allow sending, try again in 1 millisecond.
if (packets_.front()->size_in_bytes + congestion_window_->data_inflight() >
max_data_inflight_bytes_) {
next_packet_send_time_ = 1;
return;
}
bool sent = TryToSendPacket(packets_.front());
if (sent) {
congestion_window_->PacketSent(packets_.front()->size_in_bytes);
delete packets_.front();
packets_.pop_front();
time_last_update_ms_ = clock_->TimeInMilliseconds();
if (!packets_.empty()) {
// Calculate in what time we should send current packet.
next_packet_send_time_ = (packets_.front()->size_in_bytes * 8000 +
estimated_bitrate_bps_ / 2) /
estimated_bitrate_bps_;
// As rounding errors may happen, |rounding_error_time_ms_| could be
// positive or negative depending on packet was sent earlier or later,
// after it hits certain threshold we will send a packet earlier or later
// depending on error we had so far.
rounding_error_time_ms_ +=
(next_packet_send_time_ - packets_.front()->size_in_bytes * 8000.0f /
estimated_bitrate_bps_ * 1.0f);
} else {
// If sending was unsuccessful try again in 1 millisecond.
next_packet_send_time_ = 1;
}
}
}
bool BbrPacedSender::TryToSendPacket(Packet* packet) {
PacedPacketInfo pacing_info;
return packet_sender_->TimeToSendPacket(packet->ssrc, packet->sequence_number,
packet->capture_time_ms,
packet->retransmission, pacing_info);
}
} // namespace webrtc

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@ -0,0 +1,92 @@
/*
* Copyright (c) 2017 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_REMOTE_BITRATE_ESTIMATOR_TEST_BBR_PACED_SENDER_H_
#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BBR_PACED_SENDER_H_
#include <list>
#include <memory>
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/pacing/pacer.h"
namespace webrtc {
namespace testing {
namespace bwe {
class CongestionWindow;
}
} // namespace testing
struct Packet {
Packet(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t seq_number,
int64_t capture_time_ms,
int64_t enqueue_time_ms,
size_t size_in_bytes,
bool retransmission)
: priority(priority),
ssrc(ssrc),
sequence_number(seq_number),
capture_time_ms(capture_time_ms),
enqueue_time_ms(enqueue_time_ms),
size_in_bytes(size_in_bytes),
retransmission(retransmission) {}
RtpPacketSender::Priority priority;
uint32_t ssrc;
uint16_t sequence_number;
int64_t capture_time_ms;
int64_t enqueue_time_ms;
size_t size_in_bytes;
bool retransmission;
};
class Clock;
class RtcEventLog;
struct Packet;
class BbrPacedSender : public Pacer {
public:
BbrPacedSender(const Clock* clock,
PacedSender::PacketSender* packet_sender,
RtcEventLog* event_log);
~BbrPacedSender() override;
void SetEstimatedBitrateAndCongestionWindow(
uint32_t bitrate_bps,
bool in_probe_rtt,
uint64_t congestion_window) override;
void SetMinBitrate(int min_send_bitrate_bps);
void InsertPacket(RtpPacketSender::Priority priority,
uint32_t ssrc,
uint16_t sequence_number,
int64_t capture_time_ms,
size_t bytes,
bool retransmission) override;
int64_t TimeUntilNextProcess() override;
void OnBytesAcked(size_t bytes) override;
void Process() override;
bool TryToSendPacket(Packet* packet);
private:
const Clock* const clock_;
PacedSender::PacketSender* const packet_sender_;
uint32_t estimated_bitrate_bps_;
uint32_t min_send_bitrate_kbps_;
uint32_t pacing_bitrate_kbps_;
int64_t time_last_update_us_;
int64_t time_last_update_ms_;
int64_t next_packet_send_time_;
float rounding_error_time_ms_;
std::list<Packet*> packets_;
// TODO(gnish): integrate |max_data_inflight| into congestion window class.
size_t max_data_inflight_bytes_;
std::unique_ptr<testing::bwe::CongestionWindow> congestion_window_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BBR_PACED_SENDER_H_

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@ -95,7 +95,7 @@ BweSender* CreateBweSender(BandwidthEstimatorType estimator,
case kNadaEstimator:
return new NadaBweSender(kbps, observer, clock);
case kBbrEstimator:
return new BbrBweSender(clock);
return new BbrBweSender(observer, clock);
case kTcpEstimator:
FALLTHROUGH();
case kNullEstimator:

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@ -157,7 +157,7 @@ BbrBweFeedback::BbrBweFeedback(
int flow_id,
int64_t send_time_us,
int64_t latest_send_time_ms,
const std::vector<uint64_t>& packet_feedback_vector)
const std::vector<uint16_t>& packet_feedback_vector)
: FeedbackPacket(flow_id, send_time_us, latest_send_time_ms),
packet_feedback_vector_(packet_feedback_vector) {}
@ -518,12 +518,12 @@ void ChokeFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
for (PacketsIt it = in_out->begin(); it != in_out->end(); ) {
int64_t earliest_send_time_us =
std::max(last_send_time_us_, (*it)->send_time_us());
int64_t new_send_time_us =
earliest_send_time_us +
((*it)->payload_size() * 8 * 1000 + capacity_kbps_ / 2) /
capacity_kbps_;
BWE_TEST_LOGGING_PLOT(0, "MaxThroughput_", new_send_time_us / 1000,
capacity_kbps_);
if (delay_cap_helper_->ShouldSendPacket(new_send_time_us,
(*it)->send_time_us())) {
(*it)->set_send_time_us(new_send_time_us);

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@ -34,7 +34,7 @@ const float kDrainGain = 1 / kHighGain;
const float kStartupGrowthTarget = 1.25f;
const int kMaxRoundsWithoutGrowth = 3;
// Pacing gain values for Probe Bandwidth mode.
const float kPacingGain[] = {1.25, 0.75, 1, 1, 1, 1, 1, 1};
const float kPacingGain[] = {1.1, 0.9, 1, 1, 1, 1, 1, 1};
const size_t kGainCycleLength = sizeof(kPacingGain) / sizeof(kPacingGain[0]);
// Least number of rounds PROBE_RTT should last.
const int kProbeRttDurationRounds = 1;
@ -46,7 +46,7 @@ const float kTargetCongestionWindowGain = 1;
// equal to 1, but in practice because of delayed acks and the way networks
// work, it is nice to have some extra room in congestion window for full link
// utilization. Value chosen by observations on different tests.
const float kCruisingCongestionWindowGain = 1.5f;
const float kCruisingCongestionWindowGain = 2;
// Pacing gain specific for Recovery mode. Chosen by experiments in simulation
// tool.
const float kRecoveryPacingGain = 0.5f;
@ -61,10 +61,29 @@ const float kRttIncreaseThreshold = 3;
// Threshold to assume average RTT has decreased for a round. Chosen by
// experiments in simulation tool.
const float kRttDecreaseThreshold = 1.5f;
// If |kCongestionWindowThreshold| of the congestion window is filled up, tell
// encoder to stop, to avoid building sender side queues.
const float kCongestionWindowThreshold = 0.69f;
// Duration we send at |kDefaultRatebps| in order to ensure BBR has data to work
// with.
const int64_t kDefaultDurationMs = 200;
const int64_t kDefaultRatebps = 300000;
// Congestion window gain for PROBE_RTT mode.
const float kProbeRttCongestionWindowGain = 0.65f;
// We need to be sure that data inflight has increased by at least
// |kTargetCongestionWindowGainForHighGain| compared to the congestion window in
// PROBE_BW's high gain phase, to make ramp-up quicker. As high gain value has
// been decreased from 1.25 to 1.1 we need to make
// |kTargetCongestionWindowGainForHighGain| slightly higher than the actual high
// gain value.
const float kTargetCongestionWindowGainForHighGain = 1.15f;
// Encoder rate gain value for PROBE_RTT mode.
const float kEncoderRateGainForProbeRtt = 0.1f;
} // namespace
BbrBweSender::BbrBweSender(Clock* clock)
BbrBweSender::BbrBweSender(BitrateObserver* observer, Clock* clock)
: BweSender(0),
observer_(observer),
clock_(clock),
mode_(STARTUP),
max_bandwidth_filter_(new MaxBandwidthFilter()),
@ -113,14 +132,13 @@ void BbrBweSender::CalculatePacingRate() {
max_bandwidth_filter_->max_bandwidth_estimate_bps() * pacing_gain_;
}
// Declare lost packets as acked.
void BbrBweSender::HandleLoss(uint64_t last_acked_packet,
uint64_t recently_acked_packet) {
// Logic specific to wrapping sequence numbers.
if (!last_acked_packet)
return;
for (uint16_t i = last_acked_packet + 1;
AheadOrAt<uint16_t>(recently_acked_packet - 1, i); i++) {
congestion_window_->AckReceived(packet_stats_[i].payload_size_bytes);
observer_->OnBytesAcked(packet_stats_[i].payload_size_bytes);
}
}
@ -148,7 +166,7 @@ void BbrBweSender::GiveFeedback(const FeedbackPacket& feedback) {
last_packet_ack_time_ = now_ms;
const BbrBweFeedback& fb = static_cast<const BbrBweFeedback&>(feedback);
// feedback_vector holds values of acknowledged packets' sequence numbers.
const std::vector<uint64_t>& feedback_vector = fb.packet_feedback_vector();
const std::vector<uint16_t>& feedback_vector = fb.packet_feedback_vector();
// Go through all the packets acked, update variables/containers accordingly.
for (uint16_t sequence_number : feedback_vector) {
// Completing packet information with a recently received ack.
@ -175,9 +193,7 @@ void BbrBweSender::GiveFeedback(const FeedbackPacket& feedback) {
bytes_acked_ - packet->payload_size_bytes / 2;
high_gain_over_ = true;
}
// Notify pacer that an ack was received, to adjust data inflight.
// TODO(gnish): Add implementation for BitrateObserver class, to notify
// pacer about incoming acks.
observer_->OnBytesAcked(packet->payload_size_bytes);
congestion_window_->AckReceived(packet->payload_size_bytes);
HandleLoss(last_packet_acked_sequence_number_, packet->sequence_number);
last_packet_acked_sequence_number_ = packet->sequence_number;
@ -196,9 +212,8 @@ void BbrBweSender::GiveFeedback(const FeedbackPacket& feedback) {
round_trip_end_ = last_packet_sent_sequence_number_;
}
}
bool min_rtt_expired = false;
min_rtt_expired =
UpdateBandwidthAndMinRtt(now_ms, feedback_vector, bytes_acked_);
TryEnteringProbeRtt(now_ms);
UpdateBandwidthAndMinRtt(now_ms, feedback_vector, bytes_acked_);
if (new_round_started && !full_bandwidth_reached_) {
full_bandwidth_reached_ = max_bandwidth_filter_->FullBandwidthReached(
kStartupGrowthTarget, kMaxRoundsWithoutGrowth);
@ -221,19 +236,37 @@ void BbrBweSender::GiveFeedback(const FeedbackPacket& feedback) {
TryExitingRecovery(new_round_started);
break;
}
TryEnteringProbeRtt(now_ms);
TryEnteringRecovery(new_round_started); // Comment this line to disable
// entering Recovery mode.
for (uint64_t f : feedback_vector)
for (uint16_t f : feedback_vector)
AddToPastRtts(packet_stats_[f].ack_time_ms - packet_stats_[f].send_time_ms);
CalculatePacingRate();
size_t cwnd = congestion_window_->GetCongestionWindow(
mode_, max_bandwidth_filter_->max_bandwidth_estimate_bps(),
min_rtt_filter_->min_rtt_ms(), congestion_window_gain_);
// Make sure we don't get stuck when pacing_rate is 0, because of simulation
// tool specifics.
if (!pacing_rate_bps_)
pacing_rate_bps_ = 100;
BWE_TEST_LOGGING_PLOT(1, "SendRate", now_ms, pacing_rate_bps_ / 1000);
// TODO(gnish): Add implementation for BitrateObserver class to update pacing
// rate for the pacer and the encoder.
int64_t rate_for_pacer_bps = pacing_rate_bps_;
int64_t rate_for_encoder_bps = pacing_rate_bps_;
if (congestion_window_->data_inflight() >= cwnd * kCongestionWindowThreshold)
rate_for_encoder_bps = 0;
// We dont completely stop sending during PROBE_RTT, so we need encoder to
// produce something, another way of doing this would be telling encoder to
// stop and send padding instead of actual data.
if (mode_ == PROBE_RTT)
rate_for_encoder_bps = rate_for_pacer_bps * kEncoderRateGainForProbeRtt;
// Send for 300 kbps for first 200 ms, so that BBR has data to work with.
if (now_ms <= kDefaultDurationMs)
observer_->OnNetworkChanged(
kDefaultRatebps, kDefaultRatebps, false,
clock_->TimeInMicroseconds() + kFeedbackIntervalsMs * 1000, cwnd);
else
observer_->OnNetworkChanged(
rate_for_encoder_bps, rate_for_pacer_bps, mode_ == PROBE_RTT,
clock_->TimeInMicroseconds() + kFeedbackIntervalsMs * 1000, cwnd);
}
size_t BbrBweSender::TargetCongestionWindow(float gain) {
@ -251,16 +284,16 @@ rtc::Optional<int64_t> BbrBweSender::CalculateBandwidthSample(
int64_t ack_time_delta_ms) {
rtc::Optional<int64_t> bandwidth_sample;
if (send_time_delta_ms > 0)
*bandwidth_sample = data_sent_bytes * 8000 / send_time_delta_ms;
bandwidth_sample.emplace(data_sent_bytes * 8000 / send_time_delta_ms);
rtc::Optional<int64_t> ack_rate;
if (ack_time_delta_ms > 0)
*ack_rate = data_acked_bytes * 8000 / ack_time_delta_ms;
ack_rate.emplace(data_acked_bytes * 8000 / ack_time_delta_ms);
// If send rate couldn't be calculated automaticaly set |bandwidth_sample| to
// ack_rate.
if (!bandwidth_sample)
bandwidth_sample = ack_rate;
if (bandwidth_sample && ack_rate)
*bandwidth_sample = std::min(*bandwidth_sample, *ack_rate);
bandwidth_sample.emplace(std::min(*bandwidth_sample, *ack_rate));
return bandwidth_sample;
}
@ -285,12 +318,12 @@ void BbrBweSender::AddSampleForHighGain() {
first_packet_send_time_during_high_gain_ms_.reset();
}
bool BbrBweSender::UpdateBandwidthAndMinRtt(
void BbrBweSender::UpdateBandwidthAndMinRtt(
int64_t now_ms,
const std::vector<uint64_t>& feedback_vector,
const std::vector<uint16_t>& feedback_vector,
int64_t bytes_acked) {
rtc::Optional<int64_t> min_rtt_sample_ms;
for (uint64_t f : feedback_vector) {
for (uint16_t f : feedback_vector) {
PacketStats packet = packet_stats_[f];
size_t data_sent_bytes =
packet.data_sent_bytes - packet.data_sent_before_last_sent_packet_bytes;
@ -306,20 +339,22 @@ bool BbrBweSender::UpdateBandwidthAndMinRtt(
if (bandwidth_sample)
max_bandwidth_filter_->AddBandwidthSample(*bandwidth_sample,
round_count_);
AddSampleForHighGain(); // Comment to disable bucket for high gain.
// AddSampleForHighGain(); // Comment to disable bucket for high gain.
if (!min_rtt_sample_ms)
*min_rtt_sample_ms = packet.ack_time_ms - packet.send_time_ms;
min_rtt_sample_ms.emplace(packet.ack_time_ms - packet.send_time_ms);
else
*min_rtt_sample_ms = std::min(*min_rtt_sample_ms,
packet.ack_time_ms - packet.send_time_ms);
BWE_TEST_LOGGING_PLOT(1, "MinRtt", now_ms,
packet.ack_time_ms - packet.send_time_ms);
}
if (!min_rtt_sample_ms)
return false;
min_rtt_filter_->AddRttSample(*min_rtt_sample_ms, now_ms);
bool min_rtt_expired = min_rtt_filter_->MinRttExpired(now_ms);
return min_rtt_expired;
// We only feed RTT samples into the min_rtt filter which were not produced
// during 1.1 gain phase, to ensure they contain no queueing delay. But if the
// rtt sample from 1.1 gain phase improves the current estimate then we should
// make it as a new best estimate.
if (pacing_gain_ <= 1.0f || !min_rtt_filter_->min_rtt_ms() ||
*min_rtt_filter_->min_rtt_ms() >= *min_rtt_sample_ms)
min_rtt_filter_->AddRttSample(*min_rtt_sample_ms, now_ms);
}
void BbrBweSender::EnterStartup() {
@ -365,8 +400,9 @@ void BbrBweSender::TryUpdatingCyclePhase(int64_t now_ms) {
// If BBR was probing and it couldn't increase data inflight sufficiently in
// one min_rtt time, continue probing. BBR design doc isn't clear about this,
// but condition helps in quicker ramp-up and performs better.
if (pacing_gain_ > 1.0 && congestion_window_->data_inflight() <
TargetCongestionWindow(pacing_gain_))
if (pacing_gain_ > 1.0 &&
congestion_window_->data_inflight() <
TargetCongestionWindow(kTargetCongestionWindowGainForHighGain))
advance_cycle_phase = false;
// If BBR has already drained queues there is no point in continuing draining
// phase.
@ -385,6 +421,7 @@ void BbrBweSender::TryEnteringProbeRtt(int64_t now_ms) {
if (min_rtt_filter_->MinRttExpired(now_ms) && mode_ != PROBE_RTT) {
mode_ = PROBE_RTT;
pacing_gain_ = 1;
congestion_window_gain_ = kProbeRttCongestionWindowGain;
probe_rtt_start_time_ms_ = now_ms;
minimum_congestion_window_start_time_ms_.reset();
}
@ -397,22 +434,23 @@ void BbrBweSender::TryEnteringProbeRtt(int64_t now_ms) {
void BbrBweSender::TryExitingProbeRtt(int64_t now_ms, int64_t round) {
if (!minimum_congestion_window_start_time_ms_) {
if (congestion_window_->data_inflight() <=
CongestionWindow::kMinimumCongestionWindowBytes) {
*minimum_congestion_window_start_time_ms_ = now_ms;
TargetCongestionWindow(kProbeRttCongestionWindowGain)) {
minimum_congestion_window_start_time_ms_.emplace(now_ms);
minimum_congestion_window_start_round_ = round;
}
} else {
if (now_ms - *minimum_congestion_window_start_time_ms_ >=
kProbeRttDurationMs &&
round - minimum_congestion_window_start_round_ >=
kProbeRttDurationRounds)
kProbeRttDurationRounds) {
EnterProbeBw(now_ms);
}
}
}
void BbrBweSender::TryEnteringRecovery(bool new_round_started) {
// If we are already in Recovery don't try to enter.
if (mode_ == RECOVERY || !new_round_started || !full_bandwidth_reached_)
if (mode_ == RECOVERY || !new_round_started || !full_bandwidth_reached_ ||
!min_rtt_filter_->min_rtt_ms())
return;
uint64_t increased_rtt_round_counter = 0;
// If average RTT for past |kPastRttsFilterSize| rounds has been more than
@ -460,7 +498,8 @@ void BbrBweSender::OnPacketsSent(const Packets& packets) {
last_packet_sent_sequence_number_ = media_packet->sequence_number();
// If this is the first packet sent for high gain phase, save data for it.
if (!first_packet_send_time_during_high_gain_ms_ && pacing_gain_ > 1) {
*first_packet_send_time_during_high_gain_ms_ = last_packet_send_time_;
first_packet_send_time_during_high_gain_ms_.emplace(
last_packet_send_time_);
data_sent_before_high_gain_started_bytes_ =
bytes_sent_ - media_packet->payload_size() / 2;
first_packet_seq_num_during_high_gain_ =
@ -483,15 +522,31 @@ void BbrBweSender::OnPacketsSent(const Packets& packets) {
void BbrBweSender::Process() {}
BbrBweReceiver::BbrBweReceiver(int flow_id)
: BweReceiver(flow_id, kReceivingRateTimeWindowMs), clock_(0) {}
: BweReceiver(flow_id, kReceivingRateTimeWindowMs),
clock_(0),
packet_feedbacks_(),
last_feedback_ms_(0) {}
BbrBweReceiver::~BbrBweReceiver() {}
void BbrBweReceiver::ReceivePacket(int64_t arrival_time_ms,
const MediaPacket& media_packet) {}
const MediaPacket& media_packet) {
packet_feedbacks_.push_back(media_packet.sequence_number());
BweReceiver::ReceivePacket(arrival_time_ms, media_packet);
}
FeedbackPacket* BbrBweReceiver::GetFeedback(int64_t now_ms) {
return nullptr;
last_feedback_ms_ = now_ms;
int64_t corrected_send_time_ms = 0L;
if (!received_packets_.empty()) {
PacketIdentifierNode* latest = *(received_packets_.begin());
corrected_send_time_ms =
latest->send_time_ms + now_ms - latest->arrival_time_ms;
}
FeedbackPacket* fb = new BbrBweFeedback(
flow_id_, now_ms * 1000, corrected_send_time_ms, packet_feedbacks_);
packet_feedbacks_.clear();
return fb;
}
} // namespace bwe
} // namespace testing

View File

@ -30,7 +30,7 @@ class MinRttFilter;
class CongestionWindow;
class BbrBweSender : public BweSender {
public:
explicit BbrBweSender(Clock* clock);
explicit BbrBweSender(BitrateObserver* observer, Clock* clock);
virtual ~BbrBweSender();
enum Mode {
// Startup phase.
@ -113,8 +113,8 @@ class BbrBweSender : public BweSender {
private:
void EnterStartup();
bool UpdateBandwidthAndMinRtt(int64_t now_ms,
const std::vector<uint64_t>& feedback_vector,
void UpdateBandwidthAndMinRtt(int64_t now_ms,
const std::vector<uint16_t>& feedback_vector,
int64_t bytes_acked);
void TryExitingStartup();
void TryExitingDrain(int64_t now_ms);
@ -145,6 +145,7 @@ class BbrBweSender : public BweSender {
// declare those packets as lost immediately.
void HandleLoss(uint64_t last_acked_packet, uint64_t recently_acked_packet);
void AddToPastRtts(int64_t rtt_sample_ms);
BitrateObserver* observer_;
Clock* const clock_;
Mode mode_;
std::unique_ptr<MaxBandwidthFilter> max_bandwidth_filter_;
@ -229,10 +230,10 @@ class BbrBweReceiver : public BweReceiver {
void ReceivePacket(int64_t arrival_time_ms,
const MediaPacket& media_packet) override;
FeedbackPacket* GetFeedback(int64_t now_ms) override;
private:
SimulatedClock clock_;
std::vector<uint64_t> packet_feedbacks_;
std::vector<uint16_t> packet_feedbacks_;
int64_t last_feedback_ms_;
};
} // namespace bwe
} // namespace testing

View File

@ -27,8 +27,6 @@ namespace {
const int kStartingCongestionWindowBytes = 6000;
} // namespace
const int CongestionWindow::kMinimumCongestionWindowBytes;
CongestionWindow::CongestionWindow() : data_inflight_bytes_(0) {}
CongestionWindow::~CongestionWindow() {}
@ -37,8 +35,6 @@ int CongestionWindow::GetCongestionWindow(BbrBweSender::Mode mode,
int64_t bandwidth_estimate_bps,
rtc::Optional<int64_t> min_rtt_ms,
float gain) {
if (mode == BbrBweSender::PROBE_RTT)
return CongestionWindow::kMinimumCongestionWindowBytes;
return GetTargetCongestionWindow(bandwidth_estimate_bps, min_rtt_ms, gain);
}
@ -47,6 +43,7 @@ void CongestionWindow::PacketSent(size_t sent_packet_size_bytes) {
}
void CongestionWindow::AckReceived(size_t received_packet_size_bytes) {
RTC_DCHECK_GE(data_inflight_bytes_ >= received_packet_size_bytes, true);
data_inflight_bytes_ -= received_packet_size_bytes;
}
@ -57,14 +54,13 @@ int CongestionWindow::GetTargetCongestionWindow(
// If we have no rtt sample yet, return the starting congestion window size.
if (!min_rtt_ms)
return gain * kStartingCongestionWindowBytes;
int bdp = *min_rtt_ms * bandwidth_estimate_bps;
int bdp = *min_rtt_ms * bandwidth_estimate_bps / 8000;
int congestion_window = bdp * gain;
// Congestion window could be zero in rare cases, when either no bandwidth
// estimate is available, or path's min_rtt value is zero.
if (!congestion_window)
congestion_window = gain * kStartingCongestionWindowBytes;
return std::max(congestion_window,
CongestionWindow::kMinimumCongestionWindowBytes);
return congestion_window;
}
} // namespace bwe
} // namespace testing

View File

@ -21,10 +21,6 @@ namespace testing {
namespace bwe {
class CongestionWindow {
public:
// Size of congestion window while in PROBE_RTT mode, suggested by BBR's
// source code of QUIC's implementation.
static const int kMinimumCongestionWindowBytes = 4000;
CongestionWindow();
~CongestionWindow();
int GetCongestionWindow(BbrBweSender::Mode mode,

View File

@ -17,9 +17,8 @@ namespace webrtc {
namespace testing {
namespace bwe {
namespace {
// These are the same values used in CongestionWindow class.
// Same value used in CongestionWindow class.
const int64_t kStartingCongestionWindow = 6000;
const int64_t kMinimumCongestionWindow = 4000;
} // namespace
TEST(CongestionWindowTest, InitializationCheck) {
@ -51,33 +50,13 @@ TEST(CongestionWindowTest, ZeroBandwidthDelayProduct) {
EXPECT_EQ(target_congestion_window, 2.885f * kStartingCongestionWindow);
}
TEST(CongestionWindowTest, BelowMinimumTargetCongestionWindow) {
CongestionWindow congestion_window;
int64_t target_congestion_window =
congestion_window.GetTargetCongestionWindow(
100, rtc::Optional<int64_t>(2), 2.885f);
EXPECT_EQ(target_congestion_window, kMinimumCongestionWindow);
}
TEST(CongestionWindowTest, AboveMinimumTargetCongestionWindow) {
CongestionWindow congestion_window;
int64_t target_congestion_window =
congestion_window.GetTargetCongestionWindow(
100000, rtc::Optional<int64_t>(2), 2.885f);
EXPECT_EQ(target_congestion_window, 577000);
}
TEST(CongestionWindowTest, MinimumCongestionWindow) {
CongestionWindow congestion_window;
int64_t cwnd = congestion_window.GetCongestionWindow(
BbrBweSender::PROBE_RTT, 100, rtc::Optional<int64_t>(100), 2.885f);
EXPECT_EQ(cwnd, kMinimumCongestionWindow);
}
TEST(CongestionWindowTest, CalculateCongestionWindow) {
CongestionWindow congestion_window;
int64_t cwnd = congestion_window.GetCongestionWindow(
BbrBweSender::STARTUP, 100, rtc::Optional<int64_t>(100l), 2.885f);
BbrBweSender::STARTUP, 800000, rtc::Optional<int64_t>(100l), 2.885f);
EXPECT_EQ(cwnd, 28850);
cwnd = congestion_window.GetCongestionWindow(
BbrBweSender::STARTUP, 400000, rtc::Optional<int64_t>(200l), 2.885f);
EXPECT_EQ(cwnd, 28850);
}
} // namespace bwe

View File

@ -14,6 +14,7 @@
#include <cstdint>
#include <limits>
#include <list>
#include "webrtc/rtc_base/optional.h"
@ -21,12 +22,17 @@ namespace webrtc {
namespace testing {
namespace bwe {
// Expiration time for min_rtt sample, which is set to 10 seconds according to
// BBR design doc.
const int64_t kMinRttFilterSizeMs = 10000;
// Average rtt for past |kRttFilterSize| packets should grow by
// |kRttIncreaseThresholdForExpiry| in order to enter PROBE_RTT mode and expire.
// old min_rtt estimate.
const float kRttIncreaseThresholdForExpiry = 2.3f;
const size_t kRttFilterSize = 25;
class MinRttFilter {
public:
// This class implements a simple filter to ensure that PROBE_RTT is only
// entered when RTTs start to increase, instead of fixed 10 second window as
// in orginal BBR design doc, to avoid unnecessary freezes in stream.
MinRttFilter() {}
~MinRttFilter() {}
@ -34,20 +40,31 @@ class MinRttFilter {
void AddRttSample(int64_t rtt_ms, int64_t now_ms) {
if (!min_rtt_ms_ || rtt_ms <= *min_rtt_ms_ || MinRttExpired(now_ms)) {
min_rtt_ms_.emplace(rtt_ms);
discovery_time_ms_ = now_ms;
}
rtt_samples_.push_back(rtt_ms);
if (rtt_samples_.size() > kRttFilterSize)
rtt_samples_.pop_front();
}
int64_t discovery_time() { return discovery_time_ms_; }
// Checks whether or not last discovered min_rtt value is older than x
// milliseconds.
// Checks whether or not last RTT values for past |kRttFilterSize| packets
// started to increase, meaning we have to update min_rtt estimate.
bool MinRttExpired(int64_t now_ms) {
return now_ms - discovery_time_ms_ >= kMinRttFilterSizeMs;
if (rtt_samples_.size() < kRttFilterSize || !min_rtt_ms_)
return false;
int64_t sum_of_rtts_ms = 0;
for (int64_t i : rtt_samples_)
sum_of_rtts_ms += i;
if (sum_of_rtts_ms >=
*min_rtt_ms_ * kRttIncreaseThresholdForExpiry * kRttFilterSize) {
rtt_samples_.clear();
return true;
}
return false;
}
private:
rtc::Optional<int64_t> min_rtt_ms_;
int64_t discovery_time_ms_ = 0;
std::list<int64_t> rtt_samples_;
};
} // namespace bwe
} // namespace testing

View File

@ -18,28 +18,24 @@ namespace bwe {
TEST(MinRttFilterTest, InitializationCheck) {
MinRttFilter min_rtt_filter;
EXPECT_FALSE(min_rtt_filter.min_rtt_ms());
EXPECT_EQ(min_rtt_filter.discovery_time(), 0);
}
TEST(MinRttFilterTest, AddRttSample) {
MinRttFilter min_rtt_filter;
min_rtt_filter.AddRttSample(120, 5);
EXPECT_EQ(min_rtt_filter.min_rtt_ms(), 120);
EXPECT_EQ(min_rtt_filter.discovery_time(), 5);
EXPECT_EQ(*min_rtt_filter.min_rtt_ms(), 120);
min_rtt_filter.AddRttSample(121, 6);
EXPECT_EQ(min_rtt_filter.discovery_time(), 5);
min_rtt_filter.AddRttSample(119, 7);
EXPECT_EQ(min_rtt_filter.discovery_time(), 7);
min_rtt_filter.AddRttSample(140, 10007);
EXPECT_EQ(min_rtt_filter.discovery_time(), 10007);
EXPECT_EQ(min_rtt_filter.min_rtt_ms(), 140);
EXPECT_EQ(*min_rtt_filter.min_rtt_ms(), 119);
}
TEST(MinRttFilterTest, MinRttExpired) {
MinRttFilter min_rtt_filter;
min_rtt_filter.AddRttSample(120, 5);
EXPECT_EQ(min_rtt_filter.MinRttExpired(10006), true);
EXPECT_EQ(min_rtt_filter.MinRttExpired(10), false);
for (int i = 1; i <= 25; i++)
min_rtt_filter.AddRttSample(i, i);
EXPECT_EQ(min_rtt_filter.MinRttExpired(25), true);
EXPECT_EQ(min_rtt_filter.MinRttExpired(24), false);
}
} // namespace bwe
} // namespace testing

View File

@ -114,15 +114,15 @@ class BbrBweFeedback : public FeedbackPacket {
BbrBweFeedback(int flow_id,
int64_t send_time_us,
int64_t latest_send_time_ms,
const std::vector<uint64_t>& packet_feedback_vector);
const std::vector<uint16_t>& packet_feedback_vector);
virtual ~BbrBweFeedback() {}
const std::vector<uint64_t>& packet_feedback_vector() const {
const std::vector<uint16_t>& packet_feedback_vector() const {
return packet_feedback_vector_;
}
private:
const std::vector<uint64_t> packet_feedback_vector_;
const std::vector<uint16_t> packet_feedback_vector_;
};
class RembFeedback : public FeedbackPacket {

View File

@ -15,6 +15,8 @@
#include <sstream>
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/pacing/pacer.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bbr_paced_sender.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
#include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
#include "webrtc/rtc_base/checks.h"
@ -156,9 +158,12 @@ uint32_t VideoSender::TargetBitrateKbps() {
PacedVideoSender::PacedVideoSender(PacketProcessorListener* listener,
VideoSource* source,
BandwidthEstimatorType estimator)
: VideoSender(listener, source, estimator), pacer_(&clock_, this, nullptr) {
modules_.push_back(&pacer_);
pacer_.SetEstimatedBitrate(source->bits_per_second());
: VideoSender(listener, source, estimator),
// Ugly hack to use BBR's pacer.
// TODO(gnish): Make pacer choice dependant on the algorithm being used.
pacer_(new BbrPacedSender(&clock_, this, nullptr)) {
modules_.push_back(pacer_.get());
pacer_->SetEstimatedBitrate(source->bits_per_second());
}
PacedVideoSender::~PacedVideoSender() {
@ -204,10 +209,11 @@ void PacedVideoSender::RunFor(int64_t time_ms, Packets* in_out) {
if (!generated_packets.empty()) {
for (Packet* packet : generated_packets) {
MediaPacket* media_packet = static_cast<MediaPacket*>(packet);
pacer_.InsertPacket(
pacer_->InsertPacket(
PacedSender::kNormalPriority, media_packet->header().ssrc,
media_packet->header().sequenceNumber, media_packet->send_time_ms(),
media_packet->payload_size(), false);
pacer_queue_size_in_bytes_ += media_packet->payload_size();
pacer_queue_.push_back(packet);
assert(pacer_queue_.size() < 10000);
}
@ -284,11 +290,11 @@ bool PacedVideoSender::TimeToSendPacket(uint32_t ssrc,
// Make sure a packet is never paced out earlier than when it was put into
// the pacer.
assert(pace_out_time_ms >= media_packet->send_time_ms());
media_packet->SetAbsSendTimeMs(pace_out_time_ms);
media_packet->set_send_time_us(1000 * pace_out_time_ms);
media_packet->set_sender_timestamp_us(1000 * pace_out_time_ms);
queue_.push_back(media_packet);
pacer_queue_size_in_bytes_ -= media_packet->payload_size();
pacer_queue_.erase(it);
return true;
}
@ -305,7 +311,21 @@ void PacedVideoSender::OnNetworkChanged(uint32_t target_bitrate_bps,
uint8_t fraction_lost,
int64_t rtt) {
VideoSender::OnNetworkChanged(target_bitrate_bps, fraction_lost, rtt);
pacer_.SetEstimatedBitrate(target_bitrate_bps);
pacer_->SetEstimatedBitrate(target_bitrate_bps);
}
void PacedVideoSender::OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
uint32_t bitrate_for_pacer_bps,
bool in_probe_rtt,
int64_t target_set_time,
uint64_t congestion_window) {
VideoSender::OnNetworkChanged(bitrate_for_encoder_bps, 0u, 0u);
pacer_->SetEstimatedBitrateAndCongestionWindow(
bitrate_for_pacer_bps, in_probe_rtt, congestion_window);
}
void PacedVideoSender::OnBytesAcked(size_t bytes) {
pacer_->OnBytesAcked(bytes);
}
const int kNoLimit = std::numeric_limits<int>::max();

View File

@ -81,7 +81,6 @@ class VideoSender : public PacketSender, public BitrateObserver {
void OnNetworkChanged(uint32_t target_bitrate_bps,
uint8_t fraction_lost,
int64_t rtt) override;
void Pause() override;
void Resume(int64_t paused_time_ms) override;
@ -123,12 +122,23 @@ class PacedVideoSender : public VideoSender, public PacedSender::PacketSender {
uint8_t fraction_lost,
int64_t rtt) override;
void OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
uint32_t bitrate_for_pacer_bps,
bool in_probe_rtt,
int64_t rtt,
uint64_t congestion_window) override;
size_t pacer_queue_size_in_bytes() override {
return pacer_queue_size_in_bytes_;
}
void OnBytesAcked(size_t bytes) override;
private:
int64_t TimeUntilNextProcess(const std::list<Module*>& modules);
void CallProcess(const std::list<Module*>& modules);
void QueuePackets(Packets* batch, int64_t end_of_batch_time_us);
PacedSender pacer_;
size_t pacer_queue_size_in_bytes_ = 0;
std::unique_ptr<Pacer> pacer_;
Packets queue_;
Packets pacer_queue_;

View File

@ -78,7 +78,6 @@ class Figure(object):
axis = fig.add_subplot(n, 1, i+1)
self.subplots[i].PlotSubplot(axis)
class Subplot(object):
def __init__(self, var_names, xlabel, ylabel):
self.xlabel = xlabel
@ -111,10 +110,12 @@ class Subplot(object):
y = [sample[1] for sample in self.samples[alg_name][ssrc][var_name]]
x = numpy.array(x)
y = numpy.array(y)
ssrc_count = len(self.samples[alg_name].keys())
l = GenerateLabel(var_name, ssrc, ssrc_count, alg_name)
plt.plot(x, y, label=l, linewidth=2.0)
if l == 'MaxThroughput_':
plt.plot(x, y, label=l, linewidth=4.0)
else:
plt.plot(x, y, label=l, linewidth=2.0)
count += 1
plt.grid(True)
@ -148,8 +149,12 @@ def main():
target_bitrate.AddSubplot(['target_bitrate_bps', 'acknowledged_bitrate'],
"Time (s)", "Bitrate (bps)")
min_rtt_state = Figure("MinRttState")
min_rtt_state.AddSubplot(['MinRtt'], "Time (s)", "Time (ms)")
# Select which figures to plot here.
figures = [receiver, detector_state, trendline_state, target_bitrate]
figures = [receiver, detector_state, trendline_state, target_bitrate,
min_rtt_state]
# Add samples to the figures.
for line in sys.stdin: