zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Reland "Moved congestion controller to goog_cc folder."

Browse Source 
This is a reland of e6cefdf9c572cdce55ff0497ad6e516c76132ee8.

Original change's description:
> Moved congestion controller to goog_cc folder.
> 
> Bug: webrtc:8415
> Change-Id: I2070da0cacf1dbfc4b6a89285af3e68fd03497ab
> Reviewed-on: https://webrtc-review.googlesource.com/43841
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21928}

Bug: webrtc:8415
Change-Id: Ib5cf8641466655d64ac80f720561817f4cab49a9
Reviewed-on: https://webrtc-review.googlesource.com/53062
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22244}
This commit is contained in:
Sebastian Jansson
2018-02-28 16:48:00 +01:00
committed by Commit Bot
parent cdff887238
commit fc7ec8e9f8
34 changed files with 239 additions and 219 deletions
Download Patch File Download Diff File Expand all files Collapse all files

164
modules/congestion_controller/goog_cc/BUILD.gn Normal file
View File

@ -0,0 +1,164 @@
# Copyright (c) 2018 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.
import("../../../webrtc.gni")
config("bwe_test_logging") {
if (rtc_enable_bwe_test_logging) {
defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
} else {
defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
}
}
rtc_static_library("goog_cc") {
configs += [ ":bwe_test_logging" ]
sources = [
"alr_detector.cc",
"alr_detector.h",
"goog_cc_factory.cc",
"goog_cc_network_control.cc",
"goog_cc_network_control.h",
"include/goog_cc_factory.h",
"probe_controller.cc",
"probe_controller.h",
]
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
":delay_based_bwe",
":estimators",
"../..:module_api",
"../../..:webrtc_common",
"../../../:typedefs",
"../../../api:optional",
"../../../logging:rtc_event_log_api",
"../../../logging:rtc_event_pacing",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base/experiments:alr_experiment",
"../../../system_wrappers",
"../../../system_wrappers:field_trial_api",
"../../../system_wrappers:metrics_api",
"../../bitrate_controller",
"../../pacing",
"../../remote_bitrate_estimator",
"../../rtp_rtcp:rtp_rtcp_format",
"../network_control",
]
}
rtc_source_set("estimators") {
configs += [ ":bwe_test_logging" ]
sources = [
"acknowledged_bitrate_estimator.cc",
"acknowledged_bitrate_estimator.h",
"bitrate_estimator.cc",
"bitrate_estimator.h",
"delay_increase_detector_interface.h",
"median_slope_estimator.cc",
"median_slope_estimator.h",
"probe_bitrate_estimator.cc",
"probe_bitrate_estimator.h",
"trendline_estimator.cc",
"trendline_estimator.h",
]
# TODO(jschuh): Bug 1348: fix this warning.
configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
"../../../api:optional",
"../../../logging:rtc_event_bwe",
"../../../logging:rtc_event_log_api",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:rtc_numerics",
"../../../system_wrappers:field_trial_api",
"../../../system_wrappers:metrics_api",
"../../remote_bitrate_estimator",
"../../rtp_rtcp:rtp_rtcp_format",
]
}
rtc_source_set("delay_based_bwe") {
configs += [ ":bwe_test_logging" ]
sources = [
"delay_based_bwe.cc",
"delay_based_bwe.h",
]
deps = [
":estimators",
"../../../:typedefs",
"../../../logging:rtc_event_bwe",
"../../../logging:rtc_event_log_api",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../system_wrappers:field_trial_api",
"../../../system_wrappers:metrics_api",
"../../pacing",
"../../remote_bitrate_estimator",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
if (rtc_include_tests) {
rtc_source_set("goog_cc_unittests") {
testonly = true
sources = [
"acknowledged_bitrate_estimator_unittest.cc",
"alr_detector_unittest.cc",
"delay_based_bwe_unittest.cc",
"delay_based_bwe_unittest_helper.cc",
"delay_based_bwe_unittest_helper.h",
"median_slope_estimator_unittest.cc",
"probe_bitrate_estimator_unittest.cc",
"probe_controller_unittest.cc",
"trendline_estimator_unittest.cc",
]
deps = [
":delay_based_bwe",
":estimators",
":goog_cc",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:rtc_base_tests_utils",
"../../../rtc_base/experiments:alr_experiment",
"../../../system_wrappers",
"../../../test:field_trial",
"../../../test:test_support",
"../../pacing",
"../../remote_bitrate_estimator",
"../../rtp_rtcp:rtp_rtcp_format",
"../network_control",
"//testing/gmock",
]
if (!build_with_chromium && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
}

69
modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc Normal file
View File

@ -0,0 +1,69 @@
/*
* 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 "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include <utility>
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/ptr_util.h"
namespace webrtc {
namespace webrtc_cc {
namespace {
bool IsInSendTimeHistory(const PacketFeedback& packet) {
return packet.send_time_ms != PacketFeedback::kNoSendTime;
}
} // namespace
AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator()
: AcknowledgedBitrateEstimator(rtc::MakeUnique<BitrateEstimator>()) {}
AcknowledgedBitrateEstimator::~AcknowledgedBitrateEstimator() {}
AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
std::unique_ptr<BitrateEstimator> bitrate_estimator)
: bitrate_estimator_(std::move(bitrate_estimator)) {}
void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector) {
RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
packet_feedback_vector.end(),
PacketFeedbackComparator()));
for (const auto& packet : packet_feedback_vector) {
if (IsInSendTimeHistory(packet)) {
MaybeExpectFastRateChange(packet.send_time_ms);
bitrate_estimator_->Update(packet.arrival_time_ms,
rtc::dchecked_cast<int>(packet.payload_size));
}
}
}
rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
return bitrate_estimator_->bitrate_bps();
}
void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
int64_t alr_ended_time_ms) {
alr_ended_time_ms_.emplace(alr_ended_time_ms);
}
void AcknowledgedBitrateEstimator::MaybeExpectFastRateChange(
int64_t packet_send_time_ms) {
if (alr_ended_time_ms_ && packet_send_time_ms > *alr_ended_time_ms_) {
bitrate_estimator_->ExpectFastRateChange();
alr_ended_time_ms_.reset();
}
}
} // namespace webrtc_cc
} // namespace webrtc

48
modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h Normal file
View File

@ -0,0 +1,48 @@
/*
* 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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
#include <memory>
#include <vector>
#include "api/optional.h"
#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
namespace webrtc {
struct PacketFeedback;
namespace webrtc_cc {
class AcknowledgedBitrateEstimator {
public:
explicit AcknowledgedBitrateEstimator(
std::unique_ptr<BitrateEstimator> bitrate_estimator);
AcknowledgedBitrateEstimator();
~AcknowledgedBitrateEstimator();
void IncomingPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector);
rtc::Optional<uint32_t> bitrate_bps() const;
void SetAlrEndedTimeMs(int64_t alr_ended_time_ms);
private:
void MaybeExpectFastRateChange(int64_t packet_arrival_time_ms);
rtc::Optional<int64_t> alr_ended_time_ms_;
std::unique_ptr<BitrateEstimator> bitrate_estimator_;
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_

136
modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc Normal file
View File

@ -0,0 +1,136 @@
/*
* 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 "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include <utility>
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/fakeclock.h"
#include "rtc_base/ptr_util.h"
#include "test/gmock.h"
#include "test/gtest.h"
using testing::_;
using testing::NiceMock;
using testing::InSequence;
using testing::Return;
namespace webrtc {
namespace webrtc_cc {
namespace {
constexpr int64_t kFirstArrivalTimeMs = 10;
constexpr int64_t kFirstSendTimeMs = 10;
constexpr uint16_t kSequenceNumber = 1;
constexpr size_t kPayloadSize = 10;
class MockBitrateEstimator : public BitrateEstimator {
public:
MOCK_METHOD2(Update, void(int64_t now_ms, int bytes));
MOCK_CONST_METHOD0(bitrate_bps, rtc::Optional<uint32_t>());
MOCK_METHOD0(ExpectFastRateChange, void());
};
struct AcknowledgedBitrateEstimatorTestStates {
std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator;
MockBitrateEstimator* mock_bitrate_estimator;
};
AcknowledgedBitrateEstimatorTestStates CreateTestStates() {
AcknowledgedBitrateEstimatorTestStates states;
auto mock_bitrate_estimator = rtc::MakeUnique<MockBitrateEstimator>();
states.mock_bitrate_estimator = mock_bitrate_estimator.get();
states.acknowledged_bitrate_estimator =
rtc::MakeUnique<AcknowledgedBitrateEstimator>(
std::move(mock_bitrate_estimator));
return states;
}
std::vector<PacketFeedback> CreateFeedbackVector() {
std::vector<PacketFeedback> packet_feedback_vector;
const PacedPacketInfo pacing_info;
packet_feedback_vector.push_back(
PacketFeedback(kFirstArrivalTimeMs, kFirstSendTimeMs, kSequenceNumber,
kPayloadSize, pacing_info));
packet_feedback_vector.push_back(
PacketFeedback(kFirstArrivalTimeMs + 10, kFirstSendTimeMs + 10,
kSequenceNumber, kPayloadSize + 10, pacing_info));
return packet_feedback_vector;
}
} // anonymous namespace
TEST(TestAcknowledgedBitrateEstimator, DontAddPacketsWhichAreNotInSendHistory) {
auto states = CreateTestStates();
std::vector<PacketFeedback> packet_feedback_vector;
packet_feedback_vector.push_back(
PacketFeedback(kFirstArrivalTimeMs, kSequenceNumber));
EXPECT_CALL(*states.mock_bitrate_estimator, Update(_, _)).Times(0);
states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
packet_feedback_vector);
}
TEST(TestAcknowledgedBitrateEstimator, UpdateBandwith) {
auto states = CreateTestStates();
auto packet_feedback_vector = CreateFeedbackVector();
{
InSequence dummy;
EXPECT_CALL(
*states.mock_bitrate_estimator,
Update(packet_feedback_vector[0].arrival_time_ms,
static_cast<int>(packet_feedback_vector[0].payload_size)))
.Times(1);
EXPECT_CALL(
*states.mock_bitrate_estimator,
Update(packet_feedback_vector[1].arrival_time_ms,
static_cast<int>(packet_feedback_vector[1].payload_size)))
.Times(1);
}
states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
packet_feedback_vector);
}
TEST(TestAcknowledgedBitrateEstimator, ExpectFastRateChangeWhenLeftAlr) {
auto states = CreateTestStates();
auto packet_feedback_vector = CreateFeedbackVector();
{
InSequence dummy;
EXPECT_CALL(
*states.mock_bitrate_estimator,
Update(packet_feedback_vector[0].arrival_time_ms,
static_cast<int>(packet_feedback_vector[0].payload_size)))
.Times(1);
EXPECT_CALL(*states.mock_bitrate_estimator, ExpectFastRateChange())
.Times(1);
EXPECT_CALL(
*states.mock_bitrate_estimator,
Update(packet_feedback_vector[1].arrival_time_ms,
static_cast<int>(packet_feedback_vector[1].payload_size)))
.Times(1);
}
states.acknowledged_bitrate_estimator->SetAlrEndedTimeMs(kFirstArrivalTimeMs +
1);
states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
packet_feedback_vector);
}
TEST(TestAcknowledgedBitrateEstimator, ReturnBitrate) {
auto states = CreateTestStates();
rtc::Optional<uint32_t> return_value(42);
EXPECT_CALL(*states.mock_bitrate_estimator, bitrate_bps())
.Times(1)
.WillOnce(Return(return_value));
EXPECT_EQ(return_value, states.acknowledged_bitrate_estimator->bitrate_bps());
}
} // namespace webrtc_cc
} // namespace webrtc*/

97
modules/congestion_controller/goog_cc/alr_detector.cc Normal file
View File

@ -0,0 +1,97 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/alr_detector.h"
#include <algorithm>
#include <cstdio>
#include <string>
#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/checks.h"
#include "rtc_base/experiments/alr_experiment.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace webrtc_cc {
AlrDetector::AlrDetector() : AlrDetector(nullptr) {}
AlrDetector::AlrDetector(RtcEventLog* event_log)
: bandwidth_usage_percent_(kDefaultAlrBandwidthUsagePercent),
alr_start_budget_level_percent_(kDefaultAlrStartBudgetLevelPercent),
alr_stop_budget_level_percent_(kDefaultAlrStopBudgetLevelPercent),
alr_budget_(0, true),
event_log_(event_log) {
RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
rtc::Optional<AlrExperimentSettings> experiment_settings =
AlrExperimentSettings::CreateFromFieldTrial(
AlrExperimentSettings::kScreenshareProbingBweExperimentName);
if (!experiment_settings) {
experiment_settings = AlrExperimentSettings::CreateFromFieldTrial(
AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
}
if (experiment_settings) {
alr_stop_budget_level_percent_ =
experiment_settings->alr_stop_budget_level_percent;
alr_start_budget_level_percent_ =
experiment_settings->alr_start_budget_level_percent;
bandwidth_usage_percent_ = experiment_settings->alr_bandwidth_usage_percent;
}
}
AlrDetector::~AlrDetector() {}
void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t send_time_ms) {
if (!last_send_time_ms_.has_value()) {
last_send_time_ms_ = send_time_ms;
// Since the duration for sending the bytes is unknwon, return without
// updating alr state.
return;
}
int64_t delta_time_ms = send_time_ms - *last_send_time_ms_;
last_send_time_ms_ = send_time_ms;
alr_budget_.UseBudget(bytes_sent);
alr_budget_.IncreaseBudget(delta_time_ms);
bool state_changed = false;
if (alr_budget_.budget_level_percent() > alr_start_budget_level_percent_ &&
!alr_started_time_ms_) {
alr_started_time_ms_.emplace(rtc::TimeMillis());
state_changed = true;
} else if (alr_budget_.budget_level_percent() <
alr_stop_budget_level_percent_ &&
alr_started_time_ms_) {
state_changed = true;
alr_started_time_ms_.reset();
}
if (event_log_ && state_changed) {
event_log_->Log(
rtc::MakeUnique<RtcEventAlrState>(alr_started_time_ms_.has_value()));
}
}
void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
RTC_DCHECK(bitrate_bps);
const auto target_rate_kbps = static_cast<int64_t>(bitrate_bps) *
bandwidth_usage_percent_ / (1000 * 100);
alr_budget_.set_target_rate_kbps(rtc::dchecked_cast<int>(target_rate_kbps));
}
rtc::Optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
const {
return alr_started_time_ms_;
}
} // namespace webrtc_cc
} // namespace webrtc

77
modules/congestion_controller/goog_cc/alr_detector.h Normal file
View File

@ -0,0 +1,77 @@
/*
* Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
#include "api/optional.h"
#include "common_types.h" // NOLINT(build/include)
#include "modules/pacing/interval_budget.h"
#include "modules/pacing/paced_sender.h"
#include "rtc_base/rate_statistics.h"
#include "typedefs.h" // NOLINT(build/include)
namespace webrtc {
class RtcEventLog;
namespace webrtc_cc {
// Application limited region detector is a class that utilizes signals of
// elapsed time and bytes sent to estimate whether network traffic is
// currently limited by the application's ability to generate traffic.
//
// AlrDetector provides a signal that can be utilized to adjust
// estimate bandwidth.
// Note: This class is not thread-safe.
class AlrDetector {
public:
AlrDetector();
explicit AlrDetector(RtcEventLog* event_log);
~AlrDetector();
void OnBytesSent(size_t bytes_sent, int64_t send_time_ms);
// Set current estimated bandwidth.
void SetEstimatedBitrate(int bitrate_bps);
// Returns time in milliseconds when the current application-limited region
// started or empty result if the sender is currently not application-limited.
rtc::Optional<int64_t> GetApplicationLimitedRegionStartTime() const;
// Sent traffic percentage as a function of network capacity used to determine
// application-limited region. ALR region start when bandwidth usage drops
// below kAlrStartUsagePercent and ends when it raises above
// kAlrEndUsagePercent. NOTE: This is intentionally conservative at the moment
// until BW adjustments of application limited region is fine tuned.
static constexpr int kDefaultAlrBandwidthUsagePercent = 65;
static constexpr int kDefaultAlrStartBudgetLevelPercent = 80;
static constexpr int kDefaultAlrStopBudgetLevelPercent = 50;
void UpdateBudgetWithElapsedTime(int64_t delta_time_ms);
void UpdateBudgetWithBytesSent(size_t bytes_sent);
private:
int bandwidth_usage_percent_;
int alr_start_budget_level_percent_;
int alr_stop_budget_level_percent_;
rtc::Optional<int64_t> last_send_time_ms_;
IntervalBudget alr_budget_;
rtc::Optional<int64_t> alr_started_time_ms_;
RtcEventLog* event_log_;
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_

177
modules/congestion_controller/goog_cc/alr_detector_unittest.cc Normal file
View File

@ -0,0 +1,177 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/alr_detector.h"
#include "rtc_base/experiments/alr_experiment.h"
#include "test/field_trial.h"
#include "test/gtest.h"
namespace {
constexpr int kEstimatedBitrateBps = 300000;
} // namespace
namespace webrtc {
namespace webrtc_cc {
namespace {
class SimulateOutgoingTrafficIn {
public:
explicit SimulateOutgoingTrafficIn(AlrDetector* alr_detector,
int64_t* timestamp_ms)
: alr_detector_(alr_detector), timestamp_ms_(timestamp_ms) {
RTC_CHECK(alr_detector_);
}
SimulateOutgoingTrafficIn& ForTimeMs(int time_ms) {
interval_ms_ = time_ms;
ProduceTraffic();
return *this;
}
SimulateOutgoingTrafficIn& AtPercentOfEstimatedBitrate(int usage_percentage) {
usage_percentage_.emplace(usage_percentage);
ProduceTraffic();
return *this;
}
private:
void ProduceTraffic() {
if (!interval_ms_ || !usage_percentage_)
return;
const int kTimeStepMs = 10;
for (int t = 0; t < *interval_ms_; t += kTimeStepMs) {
*timestamp_ms_ += kTimeStepMs;
alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
kTimeStepMs / (8 * 100 * 1000),
*timestamp_ms_);
}
int remainder_ms = *interval_ms_ % kTimeStepMs;
if (remainder_ms > 0) {
*timestamp_ms_ += kTimeStepMs;
alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
remainder_ms / (8 * 100 * 1000),
*timestamp_ms_);
}
}
AlrDetector* const alr_detector_;
int64_t* timestamp_ms_;
rtc::Optional<int> interval_ms_;
rtc::Optional<int> usage_percentage_;
};
} // namespace
class AlrDetectorTest : public testing::Test {
public:
void SetUp() override {
alr_detector_.SetEstimatedBitrate(kEstimatedBitrateBps);
}
protected:
AlrDetector alr_detector_;
int64_t timestamp_ms_ = 1000;
};
TEST_F(AlrDetectorTest, AlrDetection) {
// Start in non-ALR state.
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Stay in non-ALR state when usage is close to 100%.
SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
.ForTimeMs(1000)
.AtPercentOfEstimatedBitrate(90);
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Verify that we ALR starts when bitrate drops below 20%.
SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
.ForTimeMs(1500)
.AtPercentOfEstimatedBitrate(20);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Verify that ALR ends when usage is above 65%.
SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
.ForTimeMs(4000)
.AtPercentOfEstimatedBitrate(100);
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
}
TEST_F(AlrDetectorTest, ShortSpike) {
// Start in non-ALR state.
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Verify that we ALR starts when bitrate drops below 20%.
SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
.ForTimeMs(1000)
.AtPercentOfEstimatedBitrate(20);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Verify that we stay in ALR region even after a short bitrate spike.
SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
.ForTimeMs(100)
.AtPercentOfEstimatedBitrate(150);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
// ALR ends when usage is above 65%.
SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
.ForTimeMs(3000)
.AtPercentOfEstimatedBitrate(100);
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
}
TEST_F(AlrDetectorTest, BandwidthEstimateChanges) {
// Start in non-ALR state.
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
// ALR starts when bitrate drops below 20%.
SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
.ForTimeMs(1000)
.AtPercentOfEstimatedBitrate(20);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
// When bandwidth estimate drops the detector should stay in ALR mode and quit
// it shortly afterwards as the sender continues sending the same amount of
// traffic. This is necessary to ensure that ProbeController can still react
// to the BWE drop by initiating a new probe.
alr_detector_.SetEstimatedBitrate(kEstimatedBitrateBps / 5);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
.ForTimeMs(1000)
.AtPercentOfEstimatedBitrate(50);
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
}
TEST_F(AlrDetectorTest, ParseControlFieldTrial) {
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-ProbingScreenshareBwe/Control/");
rtc::Optional<AlrExperimentSettings> parsed_params =
AlrExperimentSettings::CreateFromFieldTrial(
"WebRTC-ProbingScreenshareBwe");
EXPECT_FALSE(static_cast<bool>(parsed_params));
}
TEST_F(AlrDetectorTest, ParseActiveFieldTrial) {
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/");
rtc::Optional<AlrExperimentSettings> parsed_params =
AlrExperimentSettings::CreateFromFieldTrial(
"WebRTC-ProbingScreenshareBwe");
ASSERT_TRUE(static_cast<bool>(parsed_params));
EXPECT_EQ(1.1f, parsed_params->pacing_factor);
EXPECT_EQ(2875, parsed_params->max_paced_queue_time);
EXPECT_EQ(85, parsed_params->alr_bandwidth_usage_percent);
EXPECT_EQ(20, parsed_params->alr_start_budget_level_percent);
EXPECT_EQ(-20, parsed_params->alr_stop_budget_level_percent);
EXPECT_EQ(1, parsed_params->group_id);
}
} // namespace webrtc_cc
} // namespace webrtc

109
modules/congestion_controller/goog_cc/bitrate_estimator.cc Normal file
View File

@ -0,0 +1,109 @@
/*
* 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 "modules/congestion_controller/goog_cc/bitrate_estimator.h"
#include <cmath>
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
namespace webrtc_cc {
namespace {
constexpr int kInitialRateWindowMs = 500;
constexpr int kRateWindowMs = 150;
} // namespace
BitrateEstimator::BitrateEstimator()
: sum_(0),
current_win_ms_(0),
prev_time_ms_(-1),
bitrate_estimate_(-1.0f),
bitrate_estimate_var_(50.0f) {}
BitrateEstimator::~BitrateEstimator() = default;
void BitrateEstimator::Update(int64_t now_ms, int bytes) {
int rate_window_ms = kRateWindowMs;
// We use a larger window at the beginning to get a more stable sample that
// we can use to initialize the estimate.
if (bitrate_estimate_ < 0.f)
rate_window_ms = kInitialRateWindowMs;
float bitrate_sample = UpdateWindow(now_ms, bytes, rate_window_ms);
if (bitrate_sample < 0.0f)
return;
if (bitrate_estimate_ < 0.0f) {
// This is the very first sample we get. Use it to initialize the estimate.
bitrate_estimate_ = bitrate_sample;
return;
}
// Define the sample uncertainty as a function of how far away it is from the
// current estimate.
float sample_uncertainty =
10.0f * std::abs(bitrate_estimate_ - bitrate_sample) / bitrate_estimate_;
float sample_var = sample_uncertainty * sample_uncertainty;
// Update a bayesian estimate of the rate, weighting it lower if the sample
// uncertainty is large.
// The bitrate estimate uncertainty is increased with each update to model
// that the bitrate changes over time.
float pred_bitrate_estimate_var = bitrate_estimate_var_ + 5.f;
bitrate_estimate_ = (sample_var * bitrate_estimate_ +
pred_bitrate_estimate_var * bitrate_sample) /
(sample_var + pred_bitrate_estimate_var);
bitrate_estimate_var_ = sample_var * pred_bitrate_estimate_var /
(sample_var + pred_bitrate_estimate_var);
BWE_TEST_LOGGING_PLOT(1, "acknowledged_bitrate", now_ms,
bitrate_estimate_ * 1000);
}
float BitrateEstimator::UpdateWindow(int64_t now_ms,
int bytes,
int rate_window_ms) {
// Reset if time moves backwards.
if (now_ms < prev_time_ms_) {
prev_time_ms_ = -1;
sum_ = 0;
current_win_ms_ = 0;
}
if (prev_time_ms_ >= 0) {
current_win_ms_ += now_ms - prev_time_ms_;
// Reset if nothing has been received for more than a full window.
if (now_ms - prev_time_ms_ > rate_window_ms) {
sum_ = 0;
current_win_ms_ %= rate_window_ms;
}
}
prev_time_ms_ = now_ms;
float bitrate_sample = -1.0f;
if (current_win_ms_ >= rate_window_ms) {
bitrate_sample = 8.0f * sum_ / static_cast<float>(rate_window_ms);
current_win_ms_ -= rate_window_ms;
sum_ = 0;
}
sum_ += bytes;
return bitrate_sample;
}
rtc::Optional<uint32_t> BitrateEstimator::bitrate_bps() const {
if (bitrate_estimate_ < 0.f)
return rtc::nullopt;
return bitrate_estimate_ * 1000;
}
void BitrateEstimator::ExpectFastRateChange() {
// By setting the bitrate-estimate variance to a higher value we allow the
// bitrate to change fast for the next few samples.
bitrate_estimate_var_ += 200;
}
} // namespace webrtc_cc
} // namespace webrtc

48
modules/congestion_controller/goog_cc/bitrate_estimator.h Normal file
View File

@ -0,0 +1,48 @@
/*
* 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 MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
#include <vector>
#include "api/optional.h"
namespace webrtc {
namespace webrtc_cc {
// Computes a bayesian estimate of the throughput given acks containing
// the arrival time and payload size. Samples which are far from the current
// estimate or are based on few packets are given a smaller weight, as they
// are considered to be more likely to have been caused by, e.g., delay spikes
// unrelated to congestion.
class BitrateEstimator {
public:
BitrateEstimator();
virtual ~BitrateEstimator();
virtual void Update(int64_t now_ms, int bytes);
virtual rtc::Optional<uint32_t> bitrate_bps() const;
virtual void ExpectFastRateChange();
private:
float UpdateWindow(int64_t now_ms, int bytes, int rate_window_ms);
int sum_;
int64_t current_win_ms_;
int64_t prev_time_ms_;
float bitrate_estimate_;
float bitrate_estimate_var_;
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_

331
modules/congestion_controller/goog_cc/delay_based_bwe.cc Normal file
View File

@ -0,0 +1,331 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include <algorithm>
#include <cmath>
#include <cstdio>
#include <string>
#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
#include "modules/pacing/paced_sender.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/thread_annotations.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "typedefs.h" // NOLINT(build/include)
namespace {
constexpr int kTimestampGroupLengthMs = 5;
constexpr int kAbsSendTimeFraction = 18;
constexpr int kAbsSendTimeInterArrivalUpshift = 8;
constexpr int kInterArrivalShift =
kAbsSendTimeFraction + kAbsSendTimeInterArrivalUpshift;
constexpr double kTimestampToMs =
1000.0 / static_cast<double>(1 << kInterArrivalShift);
// This ssrc is used to fulfill the current API but will be removed
// after the API has been changed.
constexpr uint32_t kFixedSsrc = 0;
// Parameters for linear least squares fit of regression line to noisy data.
constexpr size_t kDefaultTrendlineWindowSize = 20;
constexpr double kDefaultTrendlineSmoothingCoeff = 0.9;
constexpr double kDefaultTrendlineThresholdGain = 4.0;
constexpr int kMaxConsecutiveFailedLookups = 5;
const char kBweWindowSizeInPacketsExperiment[] =
"WebRTC-BweWindowSizeInPackets";
size_t ReadTrendlineFilterWindowSize() {
std::string experiment_string =
webrtc::field_trial::FindFullName(kBweWindowSizeInPacketsExperiment);
size_t window_size;
int parsed_values =
sscanf(experiment_string.c_str(), "Enabled-%zu", &window_size);
if (parsed_values == 1) {
if (window_size > 1)
return window_size;
RTC_LOG(WARNING) << "Window size must be greater than 1.";
}
RTC_LOG(LS_WARNING) << "Failed to parse parameters for BweTrendlineFilter "
"experiment from field trial string. Using default.";
return kDefaultTrendlineWindowSize;
}
} // namespace
namespace webrtc {
namespace webrtc_cc {
DelayBasedBwe::Result::Result()
: updated(false),
probe(false),
target_bitrate_bps(0),
recovered_from_overuse(false) {}
DelayBasedBwe::Result::Result(bool probe, uint32_t target_bitrate_bps)
: updated(true),
probe(probe),
target_bitrate_bps(target_bitrate_bps),
recovered_from_overuse(false) {}
DelayBasedBwe::Result::~Result() {}
DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log)
: event_log_(event_log),
inter_arrival_(),
delay_detector_(),
last_seen_packet_ms_(-1),
uma_recorded_(false),
probe_bitrate_estimator_(event_log),
trendline_window_size_(
webrtc::field_trial::IsEnabled(kBweWindowSizeInPacketsExperiment)
? ReadTrendlineFilterWindowSize()
: kDefaultTrendlineWindowSize),
trendline_smoothing_coeff_(kDefaultTrendlineSmoothingCoeff),
trendline_threshold_gain_(kDefaultTrendlineThresholdGain),
consecutive_delayed_feedbacks_(0),
prev_bitrate_(0),
prev_state_(BandwidthUsage::kBwNormal) {
RTC_LOG(LS_INFO)
<< "Using Trendline filter for delay change estimation with window size "
<< trendline_window_size_;
delay_detector_.reset(new TrendlineEstimator(trendline_window_size_,
trendline_smoothing_coeff_,
trendline_threshold_gain_));
}
DelayBasedBwe::~DelayBasedBwe() {}
DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector,
rtc::Optional<uint32_t> acked_bitrate_bps,
int64_t at_time_ms) {
RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
packet_feedback_vector.end(),
PacketFeedbackComparator()));
RTC_DCHECK_RUNS_SERIALIZED(&network_race_);
// TOOD(holmer): An empty feedback vector here likely means that
// all acks were too late and that the send time history had
// timed out. We should reduce the rate when this occurs.
if (packet_feedback_vector.empty()) {
RTC_LOG(LS_WARNING) << "Very late feedback received.";
return DelayBasedBwe::Result();
}
if (!uma_recorded_) {
RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
BweNames::kSendSideTransportSeqNum,
BweNames::kBweNamesMax);
uma_recorded_ = true;
}
bool delayed_feedback = true;
bool recovered_from_overuse = false;
BandwidthUsage prev_detector_state = delay_detector_->State();
for (const auto& packet_feedback : packet_feedback_vector) {
if (packet_feedback.send_time_ms < 0)
continue;
delayed_feedback = false;
IncomingPacketFeedback(packet_feedback, at_time_ms);
if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
delay_detector_->State() == BandwidthUsage::kBwNormal) {
recovered_from_overuse = true;
}
prev_detector_state = delay_detector_->State();
}
if (delayed_feedback) {
++consecutive_delayed_feedbacks_;
if (consecutive_delayed_feedbacks_ >= kMaxConsecutiveFailedLookups) {
consecutive_delayed_feedbacks_ = 0;
return OnLongFeedbackDelay(packet_feedback_vector.back().arrival_time_ms);
}
} else {
consecutive_delayed_feedbacks_ = 0;
return MaybeUpdateEstimate(acked_bitrate_bps, recovered_from_overuse,
at_time_ms);
}
return Result();
}
DelayBasedBwe::Result DelayBasedBwe::OnLongFeedbackDelay(
int64_t arrival_time_ms) {
// Estimate should always be valid since a start bitrate always is set in the
// Call constructor. An alternative would be to return an empty Result here,
// or to estimate the throughput based on the feedback we received.
RTC_DCHECK(rate_control_.ValidEstimate());
rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2,
arrival_time_ms);
Result result;
result.updated = true;
result.probe = false;
result.target_bitrate_bps = rate_control_.LatestEstimate();
RTC_LOG(LS_WARNING) << "Long feedback delay detected, reducing BWE to "
<< result.target_bitrate_bps;
return result;
}
void DelayBasedBwe::IncomingPacketFeedback(
const PacketFeedback& packet_feedback,
int64_t at_time_ms) {
int64_t now_ms = at_time_ms;
// Reset if the stream has timed out.
if (last_seen_packet_ms_ == -1 ||
now_ms - last_seen_packet_ms_ > kStreamTimeOutMs) {
inter_arrival_.reset(
new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
kTimestampToMs, true));
delay_detector_.reset(new TrendlineEstimator(trendline_window_size_,
trendline_smoothing_coeff_,
trendline_threshold_gain_));
}
last_seen_packet_ms_ = now_ms;
uint32_t send_time_24bits =
static_cast<uint32_t>(
((static_cast<uint64_t>(packet_feedback.send_time_ms)
<< kAbsSendTimeFraction) +
500) /
1000) &
0x00FFFFFF;
// Shift up send time to use the full 32 bits that inter_arrival works with,
// so wrapping works properly.
uint32_t timestamp = send_time_24bits << kAbsSendTimeInterArrivalUpshift;
uint32_t ts_delta = 0;
int64_t t_delta = 0;
int size_delta = 0;
if (inter_arrival_->ComputeDeltas(timestamp, packet_feedback.arrival_time_ms,
now_ms, packet_feedback.payload_size,
&ts_delta, &t_delta, &size_delta)) {
double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift);
delay_detector_->Update(t_delta, ts_delta_ms,
packet_feedback.arrival_time_ms);
}
if (packet_feedback.pacing_info.probe_cluster_id !=
PacedPacketInfo::kNotAProbe) {
probe_bitrate_estimator_.HandleProbeAndEstimateBitrate(packet_feedback);
}
}
DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
rtc::Optional<uint32_t> acked_bitrate_bps,
bool recovered_from_overuse,
int64_t at_time_ms) {
Result result;
int64_t now_ms = at_time_ms;
rtc::Optional<int> probe_bitrate_bps =
probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
// Currently overusing the bandwidth.
if (delay_detector_->State() == BandwidthUsage::kBwOverusing) {
if (acked_bitrate_bps &&
rate_control_.TimeToReduceFurther(now_ms, *acked_bitrate_bps)) {
result.updated =
UpdateEstimate(now_ms, acked_bitrate_bps, &result.target_bitrate_bps);
} else if (!acked_bitrate_bps && rate_control_.ValidEstimate() &&
rate_control_.TimeToReduceFurther(
now_ms, rate_control_.LatestEstimate() / 2 - 1)) {
// Overusing before we have a measured acknowledged bitrate. We check
// TimeToReduceFurther (with a fake acknowledged bitrate) to avoid
// reducing too often.
// TODO(tschumim): Improve this and/or the acknowledged bitrate estimator
// so that we (almost) always have a bitrate estimate.
rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, now_ms);
result.updated = true;
result.probe = false;
result.target_bitrate_bps = rate_control_.LatestEstimate();
}
} else {
if (probe_bitrate_bps) {
result.probe = true;
result.updated = true;
result.target_bitrate_bps = *probe_bitrate_bps;
rate_control_.SetEstimate(*probe_bitrate_bps, now_ms);
} else {
result.updated =
UpdateEstimate(now_ms, acked_bitrate_bps, &result.target_bitrate_bps);
result.recovered_from_overuse = recovered_from_overuse;
}
}
BandwidthUsage detector_state = delay_detector_->State();
if ((result.updated && prev_bitrate_ != result.target_bitrate_bps) ||
detector_state != prev_state_) {
uint32_t bitrate_bps =
result.updated ? result.target_bitrate_bps : prev_bitrate_;
BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", now_ms, bitrate_bps);
if (event_log_) {
event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateDelayBased>(
bitrate_bps, detector_state));
}
prev_bitrate_ = bitrate_bps;
prev_state_ = detector_state;
}
return result;
}
bool DelayBasedBwe::UpdateEstimate(int64_t now_ms,
rtc::Optional<uint32_t> acked_bitrate_bps,
uint32_t* target_bitrate_bps) {
// TODO(terelius): RateControlInput::noise_var is deprecated and will be
// removed. In the meantime, we set it to zero.
const RateControlInput input(delay_detector_->State(), acked_bitrate_bps, 0);
*target_bitrate_bps = rate_control_.Update(&input, now_ms);
return rate_control_.ValidEstimate();
}
void DelayBasedBwe::OnRttUpdate(int64_t avg_rtt_ms) {
rate_control_.SetRtt(avg_rtt_ms);
}
bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
uint32_t* bitrate_bps) const {
// Currently accessed from both the process thread (see
// ModuleRtpRtcpImpl::Process()) and the configuration thread (see
// Call::GetStats()). Should in the future only be accessed from a single
// thread.
RTC_DCHECK(ssrcs);
RTC_DCHECK(bitrate_bps);
if (!rate_control_.ValidEstimate())
return false;
*ssrcs = {kFixedSsrc};
*bitrate_bps = rate_control_.LatestEstimate();
return true;
}
void DelayBasedBwe::SetStartBitrate(int start_bitrate_bps) {
RTC_LOG(LS_WARNING) << "BWE Setting start bitrate to: " << start_bitrate_bps;
rate_control_.SetStartBitrate(start_bitrate_bps);
}
void DelayBasedBwe::SetMinBitrate(int min_bitrate_bps) {
// Called from both the configuration thread and the network thread. Shouldn't
// be called from the network thread in the future.
rate_control_.SetMinBitrate(min_bitrate_bps);
}
int64_t DelayBasedBwe::GetExpectedBwePeriodMs() const {
return rate_control_.GetExpectedBandwidthPeriodMs();
}
} // namespace webrtc_cc
} // namespace webrtc

94
modules/congestion_controller/goog_cc/delay_based_bwe.h Normal file
View File

@ -0,0 +1,94 @@
/*
* Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
#include <memory>
#include <utility>
#include <vector>
#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "modules/remote_bitrate_estimator/inter_arrival.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/race_checker.h"
namespace webrtc {
class RtcEventLog;
namespace webrtc_cc {
class DelayBasedBwe {
public:
static const int64_t kStreamTimeOutMs = 2000;
struct Result {
Result();
Result(bool probe, uint32_t target_bitrate_bps);
~Result();
bool updated;
bool probe;
uint32_t target_bitrate_bps;
bool recovered_from_overuse;
};
explicit DelayBasedBwe(RtcEventLog* event_log);
virtual ~DelayBasedBwe();
Result IncomingPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector,
rtc::Optional<uint32_t> acked_bitrate_bps,
int64_t at_time_ms);
void OnRttUpdate(int64_t avg_rtt_ms);
bool LatestEstimate(std::vector<uint32_t>* ssrcs,
uint32_t* bitrate_bps) const;
void SetStartBitrate(int start_bitrate_bps);
void SetMinBitrate(int min_bitrate_bps);
int64_t GetExpectedBwePeriodMs() const;
private:
void IncomingPacketFeedback(const PacketFeedback& packet_feedback,
int64_t at_time_ms);
Result OnLongFeedbackDelay(int64_t arrival_time_ms);
Result MaybeUpdateEstimate(rtc::Optional<uint32_t> acked_bitrate_bps,
bool request_probe,
int64_t at_time_ms);
// Updates the current remote rate estimate and returns true if a valid
// estimate exists.
bool UpdateEstimate(int64_t now_ms,
rtc::Optional<uint32_t> acked_bitrate_bps,
uint32_t* target_bitrate_bps);
rtc::RaceChecker network_race_;
RtcEventLog* const event_log_;
std::unique_ptr<InterArrival> inter_arrival_;
std::unique_ptr<DelayIncreaseDetectorInterface> delay_detector_;
int64_t last_seen_packet_ms_;
bool uma_recorded_;
AimdRateControl rate_control_;
ProbeBitrateEstimator probe_bitrate_estimator_;
size_t trendline_window_size_;
double trendline_smoothing_coeff_;
double trendline_threshold_gain_;
int consecutive_delayed_feedbacks_;
uint32_t prev_bitrate_;
BandwidthUsage prev_state_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DelayBasedBwe);
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_

239
modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc Normal file
View File

@ -0,0 +1,239 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
#include "modules/pacing/paced_sender.h"
#include "rtc_base/constructormagic.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "test/gtest.h"
namespace webrtc {
namespace webrtc_cc {
namespace {
constexpr int kNumProbesCluster0 = 5;
constexpr int kNumProbesCluster1 = 8;
const PacedPacketInfo kPacingInfo0(0, kNumProbesCluster0, 2000);
const PacedPacketInfo kPacingInfo1(1, kNumProbesCluster1, 4000);
constexpr float kTargetUtilizationFraction = 0.95f;
constexpr int64_t kDummyTimestamp = 1000;
} // namespace
TEST_F(DelayBasedBweTest, NoCrashEmptyFeedback) {
std::vector<PacketFeedback> packet_feedback_vector;
bitrate_estimator_->IncomingPacketFeedbackVector(
packet_feedback_vector, rtc::nullopt, kDummyTimestamp);
}
TEST_F(DelayBasedBweTest, NoCrashOnlyLostFeedback) {
std::vector<PacketFeedback> packet_feedback_vector;
packet_feedback_vector.push_back(PacketFeedback(PacketFeedback::kNotReceived,
PacketFeedback::kNoSendTime,
0, 1500, PacedPacketInfo()));
packet_feedback_vector.push_back(PacketFeedback(PacketFeedback::kNotReceived,
PacketFeedback::kNoSendTime,
1, 1500, PacedPacketInfo()));
bitrate_estimator_->IncomingPacketFeedbackVector(
packet_feedback_vector, rtc::nullopt, kDummyTimestamp);
}
TEST_F(DelayBasedBweTest, ProbeDetection) {
int64_t now_ms = clock_.TimeInMilliseconds();
uint16_t seq_num = 0;
// First burst sent at 8 * 1000 / 10 = 800 kbps.
for (int i = 0; i < kNumProbesCluster0; ++i) {
clock_.AdvanceTimeMilliseconds(10);
now_ms = clock_.TimeInMilliseconds();
IncomingFeedback(now_ms, now_ms, seq_num++, 1000, kPacingInfo0);
}
EXPECT_TRUE(bitrate_observer_.updated());
// Second burst sent at 8 * 1000 / 5 = 1600 kbps.
for (int i = 0; i < kNumProbesCluster1; ++i) {
clock_.AdvanceTimeMilliseconds(5);
now_ms = clock_.TimeInMilliseconds();
IncomingFeedback(now_ms, now_ms, seq_num++, 1000, kPacingInfo1);
}
EXPECT_TRUE(bitrate_observer_.updated());
EXPECT_GT(bitrate_observer_.latest_bitrate(), 1500000u);
}
TEST_F(DelayBasedBweTest, ProbeDetectionNonPacedPackets) {
int64_t now_ms = clock_.TimeInMilliseconds();
uint16_t seq_num = 0;
// First burst sent at 8 * 1000 / 10 = 800 kbps, but with every other packet
// not being paced which could mess things up.
for (int i = 0; i < kNumProbesCluster0; ++i) {
clock_.AdvanceTimeMilliseconds(5);
now_ms = clock_.TimeInMilliseconds();
IncomingFeedback(now_ms, now_ms, seq_num++, 1000, kPacingInfo0);
// Non-paced packet, arriving 5 ms after.
clock_.AdvanceTimeMilliseconds(5);
IncomingFeedback(now_ms, now_ms, seq_num++, 100, PacedPacketInfo());
}
EXPECT_TRUE(bitrate_observer_.updated());
EXPECT_GT(bitrate_observer_.latest_bitrate(), 800000u);
}
TEST_F(DelayBasedBweTest, ProbeDetectionFasterArrival) {
int64_t now_ms = clock_.TimeInMilliseconds();
uint16_t seq_num = 0;
// First burst sent at 8 * 1000 / 10 = 800 kbps.
// Arriving at 8 * 1000 / 5 = 1600 kbps.
int64_t send_time_ms = 0;
for (int i = 0; i < kNumProbesCluster0; ++i) {
clock_.AdvanceTimeMilliseconds(1);
send_time_ms += 10;
now_ms = clock_.TimeInMilliseconds();
IncomingFeedback(now_ms, send_time_ms, seq_num++, 1000, kPacingInfo0);
}
EXPECT_FALSE(bitrate_observer_.updated());
}
TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrival) {
int64_t now_ms = clock_.TimeInMilliseconds();
uint16_t seq_num = 0;
// First burst sent at 8 * 1000 / 5 = 1600 kbps.
// Arriving at 8 * 1000 / 7 = 1142 kbps.
// Since the receive rate is significantly below the send rate, we expect to
// use 95% of the estimated capacity.
int64_t send_time_ms = 0;
for (int i = 0; i < kNumProbesCluster1; ++i) {
clock_.AdvanceTimeMilliseconds(7);
send_time_ms += 5;
now_ms = clock_.TimeInMilliseconds();
IncomingFeedback(now_ms, send_time_ms, seq_num++, 1000, kPacingInfo1);
}
EXPECT_TRUE(bitrate_observer_.updated());
EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
kTargetUtilizationFraction * 1140000u, 10000u);
}
TEST_F(DelayBasedBweTest, ProbeDetectionSlowerArrivalHighBitrate) {
int64_t now_ms = clock_.TimeInMilliseconds();
uint16_t seq_num = 0;
// Burst sent at 8 * 1000 / 1 = 8000 kbps.
// Arriving at 8 * 1000 / 2 = 4000 kbps.
// Since the receive rate is significantly below the send rate, we expect to
// use 95% of the estimated capacity.
int64_t send_time_ms = 0;
for (int i = 0; i < kNumProbesCluster1; ++i) {
clock_.AdvanceTimeMilliseconds(2);
send_time_ms += 1;
now_ms = clock_.TimeInMilliseconds();
IncomingFeedback(now_ms, send_time_ms, seq_num++, 1000, kPacingInfo1);
}
EXPECT_TRUE(bitrate_observer_.updated());
EXPECT_NEAR(bitrate_observer_.latest_bitrate(),
kTargetUtilizationFraction * 4000000u, 10000u);
}
TEST_F(DelayBasedBweTest, GetExpectedBwePeriodMs) {
int64_t default_interval_ms = bitrate_estimator_->GetExpectedBwePeriodMs();
EXPECT_GT(default_interval_ms, 0);
CapacityDropTestHelper(1, true, 333, 0);
int64_t interval_ms = bitrate_estimator_->GetExpectedBwePeriodMs();
EXPECT_GT(interval_ms, 0);
EXPECT_NE(interval_ms, default_interval_ms);
}
TEST_F(DelayBasedBweTest, InitialBehavior) {
InitialBehaviorTestHelper(730000);
}
TEST_F(DelayBasedBweTest, RateIncreaseReordering) {
RateIncreaseReorderingTestHelper(730000);
}
TEST_F(DelayBasedBweTest, RateIncreaseRtpTimestamps) {
RateIncreaseRtpTimestampsTestHelper(627);
}
TEST_F(DelayBasedBweTest, CapacityDropOneStream) {
CapacityDropTestHelper(1, false, 300, 0);
}
TEST_F(DelayBasedBweTest, CapacityDropPosOffsetChange) {
CapacityDropTestHelper(1, false, 867, 30000);
}
TEST_F(DelayBasedBweTest, CapacityDropNegOffsetChange) {
CapacityDropTestHelper(1, false, 933, -30000);
}
TEST_F(DelayBasedBweTest, CapacityDropOneStreamWrap) {
CapacityDropTestHelper(1, true, 333, 0);
}
TEST_F(DelayBasedBweTest, TestTimestampGrouping) {
TestTimestampGroupingTestHelper();
}
TEST_F(DelayBasedBweTest, TestShortTimeoutAndWrap) {
// Simulate a client leaving and rejoining the call after 35 seconds. This
// will make abs send time wrap, so if streams aren't timed out properly
// the next 30 seconds of packets will be out of order.
TestWrappingHelper(35);
}
TEST_F(DelayBasedBweTest, TestLongTimeoutAndWrap) {
// Simulate a client leaving and rejoining the call after some multiple of
// 64 seconds later. This will cause a zero difference in abs send times due
// to the wrap, but a big difference in arrival time, if streams aren't
// properly timed out.
TestWrappingHelper(10 * 64);
}
TEST_F(DelayBasedBweTest, TestInitialOveruse) {
const uint32_t kStartBitrate = 300e3;
const uint32_t kInitialCapacityBps = 200e3;
const uint32_t kDummySsrc = 0;
// High FPS to ensure that we send a lot of packets in a short time.
const int kFps = 90;
stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate));
stream_generator_->set_capacity_bps(kInitialCapacityBps);
// Needed to initialize the AimdRateControl.
bitrate_estimator_->SetStartBitrate(kStartBitrate);
// Produce 30 frames (in 1/3 second) and give them to the estimator.
uint32_t bitrate_bps = kStartBitrate;
bool seen_overuse = false;
for (int i = 0; i < 30; ++i) {
bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
// The purpose of this test is to ensure that we back down even if we don't
// have any acknowledged bitrate estimate yet. Hence, if the test works
// as expected, we should not have a measured bitrate yet.
EXPECT_FALSE(acknowledged_bitrate_estimator_->bitrate_bps().has_value());
if (overuse) {
EXPECT_TRUE(bitrate_observer_.updated());
EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate / 2, 15000);
bitrate_bps = bitrate_observer_.latest_bitrate();
seen_overuse = true;
break;
} else if (bitrate_observer_.updated()) {
bitrate_bps = bitrate_observer_.latest_bitrate();
bitrate_observer_.Reset();
}
}
EXPECT_TRUE(seen_overuse);
EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate / 2, 15000);
}
} // namespace webrtc_cc
} // namespace webrtc

517
modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc Normal file
View File

@ -0,0 +1,517 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
#include <algorithm>
#include <limits>
#include <utility>
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "rtc_base/checks.h"
#include "rtc_base/ptr_util.h"
namespace webrtc {
namespace webrtc_cc {
constexpr size_t kMtu = 1200;
constexpr uint32_t kAcceptedBitrateErrorBps = 50000;
// Number of packets needed before we have a valid estimate.
constexpr int kNumInitialPackets = 2;
constexpr int kInitialProbingPackets = 5;
namespace test {
void TestBitrateObserver::OnReceiveBitrateChanged(
const std::vector<uint32_t>& ssrcs,
uint32_t bitrate) {
latest_bitrate_ = bitrate;
updated_ = true;
}
RtpStream::RtpStream(int fps, int bitrate_bps)
: fps_(fps),
bitrate_bps_(bitrate_bps),
next_rtp_time_(0),
sequence_number_(0) {
RTC_CHECK_GT(fps_, 0);
}
// Generates a new frame for this stream. If called too soon after the
// previous frame, no frame will be generated. The frame is split into
// packets.
int64_t RtpStream::GenerateFrame(int64_t time_now_us,
std::vector<PacketFeedback>* packets) {
if (time_now_us < next_rtp_time_) {
return next_rtp_time_;
}
RTC_CHECK(packets != NULL);
size_t bits_per_frame = (bitrate_bps_ + fps_ / 2) / fps_;
size_t n_packets =
std::max<size_t>((bits_per_frame + 4 * kMtu) / (8 * kMtu), 1u);
size_t payload_size = (bits_per_frame + 4 * n_packets) / (8 * n_packets);
for (size_t i = 0; i < n_packets; ++i) {
PacketFeedback packet(-1, sequence_number_++);
packet.send_time_ms = (time_now_us + kSendSideOffsetUs) / 1000;
packet.payload_size = payload_size;
packets->push_back(packet);
}
next_rtp_time_ = time_now_us + (1000000 + fps_ / 2) / fps_;
return next_rtp_time_;
}
// The send-side time when the next frame can be generated.
int64_t RtpStream::next_rtp_time() const {
return next_rtp_time_;
}
void RtpStream::set_bitrate_bps(int bitrate_bps) {
ASSERT_GE(bitrate_bps, 0);
bitrate_bps_ = bitrate_bps;
}
int RtpStream::bitrate_bps() const {
return bitrate_bps_;
}
bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs,
const std::unique_ptr<RtpStream>& rhs) {
return lhs->next_rtp_time_ < rhs->next_rtp_time_;
}
StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
: capacity_(capacity), prev_arrival_time_us_(time_now) {}
// Add a new stream.
void StreamGenerator::AddStream(RtpStream* stream) {
streams_.push_back(std::unique_ptr<RtpStream>(stream));
}
// Set the link capacity.
void StreamGenerator::set_capacity_bps(int capacity_bps) {
ASSERT_GT(capacity_bps, 0);
capacity_ = capacity_bps;
}
// Divides |bitrate_bps| among all streams. The allocated bitrate per stream
// is decided by the current allocation ratios.
void StreamGenerator::SetBitrateBps(int bitrate_bps) {
ASSERT_GE(streams_.size(), 0u);
int total_bitrate_before = 0;
for (const auto& stream : streams_) {
total_bitrate_before += stream->bitrate_bps();
}
int64_t bitrate_before = 0;
int total_bitrate_after = 0;
for (const auto& stream : streams_) {
bitrate_before += stream->bitrate_bps();
int64_t bitrate_after =
(bitrate_before * bitrate_bps + total_bitrate_before / 2) /
total_bitrate_before;
stream->set_bitrate_bps(bitrate_after - total_bitrate_after);
total_bitrate_after += stream->bitrate_bps();
}
ASSERT_EQ(bitrate_before, total_bitrate_before);
EXPECT_EQ(total_bitrate_after, bitrate_bps);
}
// TODO(holmer): Break out the channel simulation part from this class to make
// it possible to simulate different types of channels.
int64_t StreamGenerator::GenerateFrame(std::vector<PacketFeedback>* packets,
int64_t time_now_us) {
RTC_CHECK(packets != NULL);
RTC_CHECK(packets->empty());
RTC_CHECK_GT(capacity_, 0);
auto it =
std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
(*it)->GenerateFrame(time_now_us, packets);
int i = 0;
for (PacketFeedback& packet : *packets) {
int capacity_bpus = capacity_ / 1000;
int64_t required_network_time_us =
(8 * 1000 * packet.payload_size + capacity_bpus / 2) / capacity_bpus;
prev_arrival_time_us_ =
std::max(time_now_us + required_network_time_us,
prev_arrival_time_us_ + required_network_time_us);
packet.arrival_time_ms = prev_arrival_time_us_ / 1000;
++i;
}
it = std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
return std::max((*it)->next_rtp_time(), time_now_us);
}
} // namespace test
DelayBasedBweTest::DelayBasedBweTest()
: clock_(100000000),
acknowledged_bitrate_estimator_(
rtc::MakeUnique<AcknowledgedBitrateEstimator>()),
bitrate_estimator_(new DelayBasedBwe(nullptr)),
stream_generator_(new test::StreamGenerator(1e6, // Capacity.
clock_.TimeInMicroseconds())),
arrival_time_offset_ms_(0),
first_update_(true) {}
DelayBasedBweTest::~DelayBasedBweTest() {}
void DelayBasedBweTest::AddDefaultStream() {
stream_generator_->AddStream(new test::RtpStream(30, 3e5));
}
const uint32_t DelayBasedBweTest::kDefaultSsrc = 0;
void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
int64_t send_time_ms,
uint16_t sequence_number,
size_t payload_size) {
IncomingFeedback(arrival_time_ms, send_time_ms, sequence_number, payload_size,
PacedPacketInfo());
}
void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
int64_t send_time_ms,
uint16_t sequence_number,
size_t payload_size,
const PacedPacketInfo& pacing_info) {
RTC_CHECK_GE(arrival_time_ms + arrival_time_offset_ms_, 0);
PacketFeedback packet(arrival_time_ms + arrival_time_offset_ms_, send_time_ms,
sequence_number, payload_size, pacing_info);
std::vector<PacketFeedback> packets;
packets.push_back(packet);
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
DelayBasedBwe::Result result =
bitrate_estimator_->IncomingPacketFeedbackVector(
packets, acknowledged_bitrate_estimator_->bitrate_bps(),
clock_.TimeInMilliseconds());
const uint32_t kDummySsrc = 0;
if (result.updated) {
bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
result.target_bitrate_bps);
}
}
// Generates a frame of packets belonging to a stream at a given bitrate and
// with a given ssrc. The stream is pushed through a very simple simulated
// network, and is then given to the receive-side bandwidth estimator.
// Returns true if an over-use was seen, false otherwise.
// The StreamGenerator::updated() should be used to check for any changes in
// target bitrate after the call to this function.
bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
uint32_t bitrate_bps) {
stream_generator_->SetBitrateBps(bitrate_bps);
std::vector<PacketFeedback> packets;
int64_t next_time_us =
stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
if (packets.empty())
return false;
bool overuse = false;
bitrate_observer_.Reset();
clock_.AdvanceTimeMicroseconds(1000 * packets.back().arrival_time_ms -
clock_.TimeInMicroseconds());
for (auto& packet : packets) {
RTC_CHECK_GE(packet.arrival_time_ms + arrival_time_offset_ms_, 0);
packet.arrival_time_ms += arrival_time_offset_ms_;
}
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
DelayBasedBwe::Result result =
bitrate_estimator_->IncomingPacketFeedbackVector(
packets, acknowledged_bitrate_estimator_->bitrate_bps(),
clock_.TimeInMilliseconds());
const uint32_t kDummySsrc = 0;
if (result.updated) {
bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
result.target_bitrate_bps);
if (!first_update_ && result.target_bitrate_bps < bitrate_bps)
overuse = true;
first_update_ = false;
}
clock_.AdvanceTimeMicroseconds(next_time_us - clock_.TimeInMicroseconds());
return overuse;
}
// Run the bandwidth estimator with a stream of |number_of_frames| frames, or
// until it reaches |target_bitrate|.
// Can for instance be used to run the estimator for some time to get it
// into a steady state.
uint32_t DelayBasedBweTest::SteadyStateRun(uint32_t ssrc,
int max_number_of_frames,
uint32_t start_bitrate,
uint32_t min_bitrate,
uint32_t max_bitrate,
uint32_t target_bitrate) {
uint32_t bitrate_bps = start_bitrate;
bool bitrate_update_seen = false;
// Produce |number_of_frames| frames and give them to the estimator.
for (int i = 0; i < max_number_of_frames; ++i) {
bool overuse = GenerateAndProcessFrame(ssrc, bitrate_bps);
if (overuse) {
EXPECT_LT(bitrate_observer_.latest_bitrate(), max_bitrate);
EXPECT_GT(bitrate_observer_.latest_bitrate(), min_bitrate);
bitrate_bps = bitrate_observer_.latest_bitrate();
bitrate_update_seen = true;
} else if (bitrate_observer_.updated()) {
bitrate_bps = bitrate_observer_.latest_bitrate();
bitrate_observer_.Reset();
}
if (bitrate_update_seen && bitrate_bps > target_bitrate) {
break;
}
}
EXPECT_TRUE(bitrate_update_seen);
return bitrate_bps;
}
void DelayBasedBweTest::InitialBehaviorTestHelper(
uint32_t expected_converge_bitrate) {
const int kFramerate = 50; // 50 fps to avoid rounding errors.
const int kFrameIntervalMs = 1000 / kFramerate;
const PacedPacketInfo kPacingInfo(0, 5, 5000);
uint32_t bitrate_bps = 0;
int64_t send_time_ms = 0;
uint16_t sequence_number = 0;
std::vector<uint32_t> ssrcs;
EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
EXPECT_EQ(0u, ssrcs.size());
clock_.AdvanceTimeMilliseconds(1000);
EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
EXPECT_FALSE(bitrate_observer_.updated());
bitrate_observer_.Reset();
clock_.AdvanceTimeMilliseconds(1000);
// Inserting packets for 5 seconds to get a valid estimate.
for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
// NOTE!!! If the following line is moved under the if case then this test
// wont work on windows realease bots.
PacedPacketInfo pacing_info =
i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
if (i == kNumInitialPackets) {
EXPECT_FALSE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
EXPECT_EQ(0u, ssrcs.size());
EXPECT_FALSE(bitrate_observer_.updated());
bitrate_observer_.Reset();
}
IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
sequence_number++, kMtu, pacing_info);
clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
send_time_ms += kFrameIntervalMs;
}
EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
ASSERT_EQ(1u, ssrcs.size());
EXPECT_EQ(kDefaultSsrc, ssrcs.front());
EXPECT_NEAR(expected_converge_bitrate, bitrate_bps, kAcceptedBitrateErrorBps);
EXPECT_TRUE(bitrate_observer_.updated());
bitrate_observer_.Reset();
EXPECT_EQ(bitrate_observer_.latest_bitrate(), bitrate_bps);
}
void DelayBasedBweTest::RateIncreaseReorderingTestHelper(
uint32_t expected_bitrate_bps) {
const int kFramerate = 50; // 50 fps to avoid rounding errors.
const int kFrameIntervalMs = 1000 / kFramerate;
const PacedPacketInfo kPacingInfo(0, 5, 5000);
int64_t send_time_ms = 0;
uint16_t sequence_number = 0;
// Inserting packets for five seconds to get a valid estimate.
for (int i = 0; i < 5 * kFramerate + 1 + kNumInitialPackets; ++i) {
// NOTE!!! If the following line is moved under the if case then this test
// wont work on windows realease bots.
PacedPacketInfo pacing_info =
i < kInitialProbingPackets ? kPacingInfo : PacedPacketInfo();
// TODO(sprang): Remove this hack once the single stream estimator is gone,
// as it doesn't do anything in Process().
if (i == kNumInitialPackets) {
// Process after we have enough frames to get a valid input rate estimate.
EXPECT_FALSE(bitrate_observer_.updated()); // No valid estimate.
}
IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
sequence_number++, kMtu, pacing_info);
clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
send_time_ms += kFrameIntervalMs;
}
EXPECT_TRUE(bitrate_observer_.updated());
EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
kAcceptedBitrateErrorBps);
for (int i = 0; i < 10; ++i) {
clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
send_time_ms += 2 * kFrameIntervalMs;
IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
sequence_number + 2, 1000);
IncomingFeedback(clock_.TimeInMilliseconds(),
send_time_ms - kFrameIntervalMs, sequence_number + 1,
1000);
sequence_number += 2;
}
EXPECT_TRUE(bitrate_observer_.updated());
EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_.latest_bitrate(),
kAcceptedBitrateErrorBps);
}
// Make sure we initially increase the bitrate as expected.
void DelayBasedBweTest::RateIncreaseRtpTimestampsTestHelper(
int expected_iterations) {
// This threshold corresponds approximately to increasing linearly with
// bitrate(i) = 1.04 * bitrate(i-1) + 1000
// until bitrate(i) > 500000, with bitrate(1) ~= 30000.
uint32_t bitrate_bps = 30000;
int iterations = 0;
AddDefaultStream();
// Feed the estimator with a stream of packets and verify that it reaches
// 500 kbps at the expected time.
while (bitrate_bps < 5e5) {
bool overuse = GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
if (overuse) {
EXPECT_GT(bitrate_observer_.latest_bitrate(), bitrate_bps);
bitrate_bps = bitrate_observer_.latest_bitrate();
bitrate_observer_.Reset();
} else if (bitrate_observer_.updated()) {
bitrate_bps = bitrate_observer_.latest_bitrate();
bitrate_observer_.Reset();
}
++iterations;
}
ASSERT_EQ(expected_iterations, iterations);
}
void DelayBasedBweTest::CapacityDropTestHelper(
int number_of_streams,
bool wrap_time_stamp,
uint32_t expected_bitrate_drop_delta,
int64_t receiver_clock_offset_change_ms) {
const int kFramerate = 30;
const int kStartBitrate = 900e3;
const int kMinExpectedBitrate = 800e3;
const int kMaxExpectedBitrate = 1100e3;
const uint32_t kInitialCapacityBps = 1000e3;
const uint32_t kReducedCapacityBps = 500e3;
int steady_state_time = 0;
if (number_of_streams <= 1) {
steady_state_time = 10;
AddDefaultStream();
} else {
steady_state_time = 10 * number_of_streams;
int bitrate_sum = 0;
int kBitrateDenom = number_of_streams * (number_of_streams - 1);
for (int i = 0; i < number_of_streams; i++) {
// First stream gets half available bitrate, while the rest share the
// remaining half i.e.: 1/2 = Sum[n/(N*(N-1))] for n=1..N-1 (rounded up)
int bitrate = kStartBitrate / 2;
if (i > 0) {
bitrate = (kStartBitrate * i + kBitrateDenom / 2) / kBitrateDenom;
}
stream_generator_->AddStream(new test::RtpStream(kFramerate, bitrate));
bitrate_sum += bitrate;
}
ASSERT_EQ(bitrate_sum, kStartBitrate);
}
// Run in steady state to make the estimator converge.
stream_generator_->set_capacity_bps(kInitialCapacityBps);
uint32_t bitrate_bps = SteadyStateRun(
kDefaultSsrc, steady_state_time * kFramerate, kStartBitrate,
kMinExpectedBitrate, kMaxExpectedBitrate, kInitialCapacityBps);
EXPECT_NEAR(kInitialCapacityBps, bitrate_bps, 180000u);
bitrate_observer_.Reset();
// Add an offset to make sure the BWE can handle it.
arrival_time_offset_ms_ += receiver_clock_offset_change_ms;
// Reduce the capacity and verify the decrease time.
stream_generator_->set_capacity_bps(kReducedCapacityBps);
int64_t overuse_start_time = clock_.TimeInMilliseconds();
int64_t bitrate_drop_time = -1;
for (int i = 0; i < 100 * number_of_streams; ++i) {
GenerateAndProcessFrame(kDefaultSsrc, bitrate_bps);
if (bitrate_drop_time == -1 &&
bitrate_observer_.latest_bitrate() <= kReducedCapacityBps) {
bitrate_drop_time = clock_.TimeInMilliseconds();
}
if (bitrate_observer_.updated())
bitrate_bps = bitrate_observer_.latest_bitrate();
}
EXPECT_NEAR(expected_bitrate_drop_delta,
bitrate_drop_time - overuse_start_time, 33);
}
void DelayBasedBweTest::TestTimestampGroupingTestHelper() {
const int kFramerate = 50; // 50 fps to avoid rounding errors.
const int kFrameIntervalMs = 1000 / kFramerate;
int64_t send_time_ms = 0;
uint16_t sequence_number = 0;
// Initial set of frames to increase the bitrate. 6 seconds to have enough
// time for the first estimate to be generated and for Process() to be called.
for (int i = 0; i <= 6 * kFramerate; ++i) {
IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
sequence_number++, 1000);
clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
send_time_ms += kFrameIntervalMs;
}
EXPECT_TRUE(bitrate_observer_.updated());
EXPECT_GE(bitrate_observer_.latest_bitrate(), 400000u);
// Insert batches of frames which were sent very close in time. Also simulate
// capacity over-use to see that we back off correctly.
const int kTimestampGroupLength = 15;
for (int i = 0; i < 100; ++i) {
for (int j = 0; j < kTimestampGroupLength; ++j) {
// Insert |kTimestampGroupLength| frames with just 1 timestamp ticks in
// between. Should be treated as part of the same group by the estimator.
IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
sequence_number++, 100);
clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
send_time_ms += 1;
}
// Increase time until next batch to simulate over-use.
clock_.AdvanceTimeMilliseconds(10);
send_time_ms += kFrameIntervalMs - kTimestampGroupLength;
}
EXPECT_TRUE(bitrate_observer_.updated());
// Should have reduced the estimate.
EXPECT_LT(bitrate_observer_.latest_bitrate(), 400000u);
}
void DelayBasedBweTest::TestWrappingHelper(int silence_time_s) {
const int kFramerate = 100;
const int kFrameIntervalMs = 1000 / kFramerate;
int64_t send_time_ms = 0;
uint16_t sequence_number = 0;
for (size_t i = 0; i < 3000; ++i) {
IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
sequence_number++, 1000);
clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
send_time_ms += kFrameIntervalMs;
}
uint32_t bitrate_before = 0;
std::vector<uint32_t> ssrcs;
bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before);
clock_.AdvanceTimeMilliseconds(silence_time_s * 1000);
send_time_ms += silence_time_s * 1000;
for (size_t i = 0; i < 24; ++i) {
IncomingFeedback(clock_.TimeInMilliseconds(), send_time_ms,
sequence_number++, 1000);
clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
send_time_ms += kFrameIntervalMs;
}
uint32_t bitrate_after = 0;
bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_after);
EXPECT_LT(bitrate_after, bitrate_before);
}
} // namespace webrtc_cc
} // namespace webrtc

180
modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h Normal file
View File

@ -0,0 +1,180 @@
/*
* Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
#include <list>
#include <map>
#include <memory>
#include <utility>
#include <vector>
#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "rtc_base/constructormagic.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
namespace webrtc {
namespace webrtc_cc {
namespace test {
class TestBitrateObserver : public RemoteBitrateObserver {
public:
TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
virtual ~TestBitrateObserver() {}
void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
uint32_t bitrate) override;
void Reset() { updated_ = false; }
bool updated() const { return updated_; }
uint32_t latest_bitrate() const { return latest_bitrate_; }
private:
bool updated_;
uint32_t latest_bitrate_;
};
class RtpStream {
public:
enum { kSendSideOffsetUs = 1000000 };
RtpStream(int fps, int bitrate_bps);
// Generates a new frame for this stream. If called too soon after the
// previous frame, no frame will be generated. The frame is split into
// packets.
int64_t GenerateFrame(int64_t time_now_us,
std::vector<PacketFeedback>* packets);
// The send-side time when the next frame can be generated.
int64_t next_rtp_time() const;
void set_bitrate_bps(int bitrate_bps);
int bitrate_bps() const;
static bool Compare(const std::unique_ptr<RtpStream>& lhs,
const std::unique_ptr<RtpStream>& rhs);
private:
int fps_;
int bitrate_bps_;
int64_t next_rtp_time_;
uint16_t sequence_number_;
RTC_DISALLOW_COPY_AND_ASSIGN(RtpStream);
};
class StreamGenerator {
public:
StreamGenerator(int capacity, int64_t time_now);
// Add a new stream.
void AddStream(RtpStream* stream);
// Set the link capacity.
void set_capacity_bps(int capacity_bps);
// Divides |bitrate_bps| among all streams. The allocated bitrate per stream
// is decided by the initial allocation ratios.
void SetBitrateBps(int bitrate_bps);
// Set the RTP timestamp offset for the stream identified by |ssrc|.
void set_rtp_timestamp_offset(uint32_t ssrc, uint32_t offset);
// TODO(holmer): Break out the channel simulation part from this class to make
// it possible to simulate different types of channels.
int64_t GenerateFrame(std::vector<PacketFeedback>* packets,
int64_t time_now_us);
private:
// Capacity of the simulated channel in bits per second.
int capacity_;
// The time when the last packet arrived.
int64_t prev_arrival_time_us_;
// All streams being transmitted on this simulated channel.
std::vector<std::unique_ptr<RtpStream>> streams_;
RTC_DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
};
} // namespace test
class DelayBasedBweTest : public ::testing::Test {
public:
DelayBasedBweTest();
virtual ~DelayBasedBweTest();
protected:
void AddDefaultStream();
// Helpers to insert a single packet into the delay-based BWE.
void IncomingFeedback(int64_t arrival_time_ms,
int64_t send_time_ms,
uint16_t sequence_number,
size_t payload_size);
void IncomingFeedback(int64_t arrival_time_ms,
int64_t send_time_ms,
uint16_t sequence_number,
size_t payload_size,
const PacedPacketInfo& pacing_info);
// Generates a frame of packets belonging to a stream at a given bitrate and
// with a given ssrc. The stream is pushed through a very simple simulated
// network, and is then given to the receive-side bandwidth estimator.
// Returns true if an over-use was seen, false otherwise.
// The StreamGenerator::updated() should be used to check for any changes in
// target bitrate after the call to this function.
bool GenerateAndProcessFrame(uint32_t ssrc, uint32_t bitrate_bps);
// Run the bandwidth estimator with a stream of |number_of_frames| frames, or
// until it reaches |target_bitrate|.
// Can for instance be used to run the estimator for some time to get it
// into a steady state.
uint32_t SteadyStateRun(uint32_t ssrc,
int number_of_frames,
uint32_t start_bitrate,
uint32_t min_bitrate,
uint32_t max_bitrate,
uint32_t target_bitrate);
void TestTimestampGroupingTestHelper();
void TestWrappingHelper(int silence_time_s);
void InitialBehaviorTestHelper(uint32_t expected_converge_bitrate);
void RateIncreaseReorderingTestHelper(uint32_t expected_bitrate);
void RateIncreaseRtpTimestampsTestHelper(int expected_iterations);
void CapacityDropTestHelper(int number_of_streams,
bool wrap_time_stamp,
uint32_t expected_bitrate_drop_delta,
int64_t receiver_clock_offset_change_ms);
static const uint32_t kDefaultSsrc;
SimulatedClock clock_; // Time at the receiver.
test::TestBitrateObserver bitrate_observer_;
std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator_;
std::unique_ptr<DelayBasedBwe> bitrate_estimator_;
std::unique_ptr<test::StreamGenerator> stream_generator_;
int64_t arrival_time_offset_ms_;
bool first_update_;
RTC_DISALLOW_COPY_AND_ASSIGN(DelayBasedBweTest);
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_

39
modules/congestion_controller/goog_cc/delay_increase_detector_interface.h Normal file
View File

@ -0,0 +1,39 @@
/*
* Copyright (c) 2018 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_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
#include <stdint.h>
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
namespace webrtc_cc {
class DelayIncreaseDetectorInterface {
public:
DelayIncreaseDetectorInterface() {}
virtual ~DelayIncreaseDetectorInterface() {}
// Update the detector with a new sample. The deltas should represent deltas
// between timestamp groups as defined by the InterArrival class.
virtual void Update(double recv_delta_ms,
double send_delta_ms,
int64_t arrival_time_ms) = 0;
virtual BandwidthUsage State() const = 0;
RTC_DISALLOW_COPY_AND_ASSIGN(DelayIncreaseDetectorInterface);
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_

30
modules/congestion_controller/goog_cc/goog_cc_factory.cc Normal file
View File

@ -0,0 +1,30 @@
/*
* Copyright (c) 2018 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 "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
#include "rtc_base/ptr_util.h"
namespace webrtc {
GoogCcNetworkControllerFactory::GoogCcNetworkControllerFactory(
RtcEventLog* event_log)
: event_log_(event_log) {}
NetworkControllerInterface::uptr GoogCcNetworkControllerFactory::Create(
NetworkControllerObserver* observer) {
return rtc::MakeUnique<webrtc_cc::GoogCcNetworkController>(event_log_,
observer);
}
TimeDelta GoogCcNetworkControllerFactory::GetProcessInterval() const {
const int64_t kUpdateIntervalMs = 25;
return TimeDelta::ms(kUpdateIntervalMs);
}
} // namespace webrtc

413
modules/congestion_controller/goog_cc/goog_cc_network_control.cc Normal file
View File

@ -0,0 +1,413 @@
/*
* Copyright (c) 2018 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 "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
#include <algorithm>
#include <functional>
#include <limits>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/alr_detector.h"
#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
#include "modules/congestion_controller/goog_cc/probe_controller.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/checks.h"
#include "rtc_base/format_macros.h"
#include "rtc_base/logging.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace webrtc_cc {
namespace {
const char kCwndExperiment[] = "WebRTC-CwndExperiment";
const int64_t kDefaultAcceptedQueueMs = 250;
// 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.
const float kDefaultPaceMultiplier = 2.5f;
bool CwndExperimentEnabled() {
std::string experiment_string =
webrtc::field_trial::FindFullName(kCwndExperiment);
// The experiment is enabled iff the field trial string begins with "Enabled".
return experiment_string.find("Enabled") == 0;
}
bool ReadCwndExperimentParameter(int64_t* accepted_queue_ms) {
RTC_DCHECK(accepted_queue_ms);
std::string experiment_string =
webrtc::field_trial::FindFullName(kCwndExperiment);
int parsed_values =
sscanf(experiment_string.c_str(), "Enabled-%" PRId64, accepted_queue_ms);
if (parsed_values == 1) {
RTC_CHECK_GE(*accepted_queue_ms, 0)
<< "Accepted must be greater than or equal to 0.";
return true;
}
return false;
}
// Makes sure that the bitrate and the min, max values are in valid range.
static void ClampBitrates(int64_t* bitrate_bps,
int64_t* min_bitrate_bps,
int64_t* max_bitrate_bps) {
// TODO(holmer): We should make sure the default bitrates are set to 10 kbps,
// and that we don't try to set the min bitrate to 0 from any applications.
// The congestion controller should allow a min bitrate of 0.
if (*min_bitrate_bps < congestion_controller::GetMinBitrateBps())
*min_bitrate_bps = congestion_controller::GetMinBitrateBps();
if (*max_bitrate_bps > 0)
*max_bitrate_bps = std::max(*min_bitrate_bps, *max_bitrate_bps);
if (*bitrate_bps > 0)
*bitrate_bps = std::max(*min_bitrate_bps, *bitrate_bps);
}
std::vector<PacketFeedback> ReceivedPacketsFeedbackAsRtp(
const TransportPacketsFeedback report) {
std::vector<PacketFeedback> packet_feedback_vector;
for (auto& fb : report.PacketsWithFeedback()) {
if (fb.receive_time.IsFinite()) {
PacketFeedback pf(fb.receive_time.ms(), 0);
pf.creation_time_ms = report.feedback_time.ms();
if (fb.sent_packet.has_value()) {
pf.payload_size = fb.sent_packet->size.bytes();
pf.pacing_info = fb.sent_packet->pacing_info;
pf.send_time_ms = fb.sent_packet->send_time.ms();
} else {
pf.send_time_ms = PacketFeedback::kNoSendTime;
}
packet_feedback_vector.push_back(pf);
}
}
return packet_feedback_vector;
}
} // namespace
GoogCcNetworkController::GoogCcNetworkController(
RtcEventLog* event_log,
NetworkControllerObserver* observer)
: event_log_(event_log),
observer_(observer),
probe_controller_(new ProbeController(observer_)),
bandwidth_estimation_(
rtc::MakeUnique<SendSideBandwidthEstimation>(event_log_)),
alr_detector_(rtc::MakeUnique<AlrDetector>()),
delay_based_bwe_(new DelayBasedBwe(event_log_)),
acknowledged_bitrate_estimator_(
rtc::MakeUnique<AcknowledgedBitrateEstimator>()),
pacing_factor_(kDefaultPaceMultiplier),
min_pacing_rate_(DataRate::Zero()),
max_padding_rate_(DataRate::Zero()),
in_cwnd_experiment_(CwndExperimentEnabled()),
accepted_queue_ms_(kDefaultAcceptedQueueMs) {
delay_based_bwe_->SetMinBitrate(congestion_controller::GetMinBitrateBps());
if (in_cwnd_experiment_ &&
!ReadCwndExperimentParameter(&accepted_queue_ms_)) {
RTC_LOG(LS_WARNING) << "Failed to parse parameters for CwndExperiment "
"from field trial string. Experiment disabled.";
in_cwnd_experiment_ = false;
}
}
GoogCcNetworkController::~GoogCcNetworkController() {}
void GoogCcNetworkController::OnNetworkAvailability(NetworkAvailability msg) {
probe_controller_->OnNetworkAvailability(msg);
}
void GoogCcNetworkController::OnNetworkRouteChange(NetworkRouteChange msg) {
int64_t min_bitrate_bps = msg.constraints.min_data_rate.bps();
int64_t max_bitrate_bps = -1;
int64_t start_bitrate_bps = -1;
if (msg.constraints.max_data_rate.IsFinite())
max_bitrate_bps = msg.constraints.max_data_rate.bps();
if (msg.constraints.starting_rate.IsFinite())
start_bitrate_bps = msg.constraints.starting_rate.bps();
ClampBitrates(&start_bitrate_bps, &min_bitrate_bps, &max_bitrate_bps);
bandwidth_estimation_ =
rtc::MakeUnique<SendSideBandwidthEstimation>(event_log_);
bandwidth_estimation_->SetBitrates(start_bitrate_bps, min_bitrate_bps,
max_bitrate_bps);
delay_based_bwe_.reset(new DelayBasedBwe(event_log_));
acknowledged_bitrate_estimator_.reset(new AcknowledgedBitrateEstimator());
delay_based_bwe_->SetStartBitrate(start_bitrate_bps);
delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
probe_controller_->Reset(msg.at_time.ms());
probe_controller_->SetBitrates(min_bitrate_bps, start_bitrate_bps,
max_bitrate_bps, msg.at_time.ms());
MaybeTriggerOnNetworkChanged(msg.at_time);
}
void GoogCcNetworkController::OnProcessInterval(ProcessInterval msg) {
bandwidth_estimation_->UpdateEstimate(msg.at_time.ms());
rtc::Optional<int64_t> start_time_ms =
alr_detector_->GetApplicationLimitedRegionStartTime();
probe_controller_->SetAlrStartTimeMs(start_time_ms);
probe_controller_->Process(msg.at_time.ms());
MaybeTriggerOnNetworkChanged(msg.at_time);
}
void GoogCcNetworkController::OnRemoteBitrateReport(RemoteBitrateReport msg) {
bandwidth_estimation_->UpdateReceiverEstimate(msg.receive_time.ms(),
msg.bandwidth.bps());
BWE_TEST_LOGGING_PLOT(1, "REMB_kbps", msg.receive_time.ms(),
msg.bandwidth.bps() / 1000);
}
void GoogCcNetworkController::OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) {
if (msg.smoothed) {
delay_based_bwe_->OnRttUpdate(msg.round_trip_time.ms());
} else {
bandwidth_estimation_->UpdateRtt(msg.round_trip_time.ms(),
msg.receive_time.ms());
}
}
void GoogCcNetworkController::OnSentPacket(SentPacket sent_packet) {
alr_detector_->OnBytesSent(sent_packet.size.bytes(),
sent_packet.send_time.ms());
}
void GoogCcNetworkController::OnStreamsConfig(StreamsConfig msg) {
probe_controller_->EnablePeriodicAlrProbing(msg.requests_alr_probing);
bool pacing_changed = false;
if (msg.pacing_factor && *msg.pacing_factor != pacing_factor_) {
pacing_factor_ = *msg.pacing_factor;
pacing_changed = true;
}
if (msg.min_pacing_rate && *msg.min_pacing_rate != min_pacing_rate_) {
min_pacing_rate_ = *msg.min_pacing_rate;
pacing_changed = true;
}
if (msg.max_padding_rate && *msg.max_padding_rate != max_padding_rate_) {
max_padding_rate_ = *msg.max_padding_rate;
pacing_changed = true;
}
if (pacing_changed)
UpdatePacingRates(msg.at_time);
}
void GoogCcNetworkController::OnTargetRateConstraints(
TargetRateConstraints constraints) {
int64_t min_bitrate_bps = constraints.min_data_rate.bps();
int64_t max_bitrate_bps = -1;
int64_t start_bitrate_bps = -1;
if (constraints.max_data_rate.IsFinite())
max_bitrate_bps = constraints.max_data_rate.bps();
if (constraints.starting_rate.IsFinite())
start_bitrate_bps = constraints.starting_rate.bps();
ClampBitrates(&start_bitrate_bps, &min_bitrate_bps, &max_bitrate_bps);
probe_controller_->SetBitrates(min_bitrate_bps, start_bitrate_bps,
max_bitrate_bps, constraints.at_time.ms());
bandwidth_estimation_->SetBitrates(start_bitrate_bps, min_bitrate_bps,
max_bitrate_bps);
if (start_bitrate_bps > 0)
delay_based_bwe_->SetStartBitrate(start_bitrate_bps);
delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
MaybeTriggerOnNetworkChanged(constraints.at_time);
}
void GoogCcNetworkController::OnTransportLossReport(TransportLossReport msg) {
int64_t total_packets_delta =
msg.packets_received_delta + msg.packets_lost_delta;
bandwidth_estimation_->UpdatePacketsLost(
msg.packets_lost_delta, total_packets_delta, msg.receive_time.ms());
}
void GoogCcNetworkController::OnTransportPacketsFeedback(
TransportPacketsFeedback report) {
int64_t feedback_rtt = -1;
for (const auto& packet_feedback : report.PacketsWithFeedback()) {
if (packet_feedback.sent_packet.has_value() &&
packet_feedback.receive_time.IsFinite()) {
int64_t rtt = report.feedback_time.ms() -
packet_feedback.sent_packet->send_time.ms();
// max() is used to account for feedback being delayed by the
// receiver.
feedback_rtt = std::max(rtt, feedback_rtt);
}
}
if (feedback_rtt > -1) {
feedback_rtts_.push_back(feedback_rtt);
const size_t kFeedbackRttWindow = 32;
if (feedback_rtts_.size() > kFeedbackRttWindow)
feedback_rtts_.pop_front();
min_feedback_rtt_ms_.emplace(
*std::min_element(feedback_rtts_.begin(), feedback_rtts_.end()));
}
std::vector<PacketFeedback> received_feedback_vector =
ReceivedPacketsFeedbackAsRtp(report);
rtc::Optional<int64_t> alr_start_time =
alr_detector_->GetApplicationLimitedRegionStartTime();
if (previously_in_alr && !alr_start_time.has_value()) {
int64_t now_ms = report.feedback_time.ms();
acknowledged_bitrate_estimator_->SetAlrEndedTimeMs(now_ms);
probe_controller_->SetAlrEndedTimeMs(now_ms);
}
previously_in_alr = alr_start_time.has_value();
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
received_feedback_vector);
DelayBasedBwe::Result result;
result = delay_based_bwe_->IncomingPacketFeedbackVector(
received_feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps(),
report.feedback_time.ms());
if (result.updated) {
if (result.probe) {
bandwidth_estimation_->SetSendBitrate(result.target_bitrate_bps);
}
// Since SetSendBitrate now resets the delay-based estimate, we have to call
// UpdateDelayBasedEstimate after SetSendBitrate.
bandwidth_estimation_->UpdateDelayBasedEstimate(report.feedback_time.ms(),
result.target_bitrate_bps);
// Update the estimate in the ProbeController, in case we want to probe.
MaybeTriggerOnNetworkChanged(report.feedback_time);
}
if (result.recovered_from_overuse) {
probe_controller_->SetAlrStartTimeMs(alr_start_time);
probe_controller_->RequestProbe(report.feedback_time.ms());
}
MaybeUpdateCongestionWindow();
}
void GoogCcNetworkController::MaybeUpdateCongestionWindow() {
if (!in_cwnd_experiment_)
return;
// No valid RTT. Could be because send-side BWE isn't used, in which case
// we don't try to limit the outstanding packets.
if (!min_feedback_rtt_ms_)
return;
if (!last_estimate_.has_value())
return;
const DataSize kMinCwnd = DataSize::bytes(2 * 1500);
TimeDelta time_window =
TimeDelta::ms(*min_feedback_rtt_ms_ + accepted_queue_ms_);
DataSize data_window = last_estimate_->bandwidth * time_window;
CongestionWindow msg;
msg.enabled = true;
msg.data_window = std::max(kMinCwnd, data_window);
observer_->OnCongestionWindow(msg);
RTC_LOG(LS_INFO) << "Feedback rtt: " << *min_feedback_rtt_ms_
<< " Bitrate: " << last_estimate_->bandwidth.bps();
}
void GoogCcNetworkController::MaybeTriggerOnNetworkChanged(Timestamp at_time) {
int32_t estimated_bitrate_bps;
uint8_t fraction_loss;
int64_t rtt_ms;
bool estimate_changed = GetNetworkParameters(
&estimated_bitrate_bps, &fraction_loss, &rtt_ms, at_time);
if (estimate_changed) {
TimeDelta bwe_period =
TimeDelta::ms(delay_based_bwe_->GetExpectedBwePeriodMs());
NetworkEstimate new_estimate;
new_estimate.at_time = at_time;
new_estimate.round_trip_time = TimeDelta::ms(rtt_ms);
new_estimate.bandwidth = DataRate::bps(estimated_bitrate_bps);
new_estimate.loss_rate_ratio = fraction_loss / 255.0f;
new_estimate.bwe_period = bwe_period;
new_estimate.changed = true;
last_estimate_ = new_estimate;
OnNetworkEstimate(new_estimate);
}
}
bool GoogCcNetworkController::GetNetworkParameters(
int32_t* estimated_bitrate_bps,
uint8_t* fraction_loss,
int64_t* rtt_ms,
Timestamp at_time) {
bandwidth_estimation_->CurrentEstimate(estimated_bitrate_bps, fraction_loss,
rtt_ms);
*estimated_bitrate_bps = std::max<int32_t>(
*estimated_bitrate_bps, bandwidth_estimation_->GetMinBitrate());
bool estimate_changed = false;
if ((*estimated_bitrate_bps != last_estimated_bitrate_bps_) ||
(*fraction_loss != last_estimated_fraction_loss_) ||
(*rtt_ms != last_estimated_rtt_ms_)) {
last_estimated_bitrate_bps_ = *estimated_bitrate_bps;
last_estimated_fraction_loss_ = *fraction_loss;
last_estimated_rtt_ms_ = *rtt_ms;
estimate_changed = true;
}
BWE_TEST_LOGGING_PLOT(1, "fraction_loss_%", at_time.ms(),
(*fraction_loss * 100) / 256);
BWE_TEST_LOGGING_PLOT(1, "rtt_ms", at_time.ms(), *rtt_ms);
BWE_TEST_LOGGING_PLOT(1, "Target_bitrate_kbps", at_time.ms(),
*estimated_bitrate_bps / 1000);
return estimate_changed;
}
void GoogCcNetworkController::OnNetworkEstimate(NetworkEstimate estimate) {
if (!estimate.changed)
return;
UpdatePacingRates(estimate.at_time);
alr_detector_->SetEstimatedBitrate(estimate.bandwidth.bps());
probe_controller_->SetEstimatedBitrate(estimate.bandwidth.bps(),
estimate.at_time.ms());
TargetTransferRate target_rate;
target_rate.at_time = estimate.at_time;
// Set the target rate to the full estimated bandwidth since the estimation
// for legacy reasons includes target rate constraints.
target_rate.target_rate = estimate.bandwidth;
target_rate.network_estimate = estimate;
observer_->OnTargetTransferRate(target_rate);
}
void GoogCcNetworkController::UpdatePacingRates(Timestamp at_time) {
if (!last_estimate_)
return;
DataRate pacing_rate =
std::max(min_pacing_rate_, last_estimate_->bandwidth) * pacing_factor_;
DataRate padding_rate =
std::min(max_padding_rate_, last_estimate_->bandwidth);
PacerConfig msg;
msg.at_time = at_time;
msg.time_window = TimeDelta::s(1);
msg.data_window = pacing_rate * msg.time_window;
msg.pad_window = padding_rate * msg.time_window;
observer_->OnPacerConfig(msg);
}
} // namespace webrtc_cc
} // namespace webrtc

94
modules/congestion_controller/goog_cc/goog_cc_network_control.h Normal file
View File

@ -0,0 +1,94 @@
/*
* Copyright (c) 2018 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_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
#include <stdint.h>
#include <deque>
#include <memory>
#include <vector>
#include "api/optional.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/bitrate_controller/send_side_bandwidth_estimation.h"
#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
#include "modules/congestion_controller/goog_cc/alr_detector.h"
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "modules/congestion_controller/goog_cc/probe_controller.h"
#include "modules/congestion_controller/network_control/include/network_control.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
namespace webrtc_cc {
class GoogCcNetworkController : public NetworkControllerInterface {
public:
GoogCcNetworkController(RtcEventLog* event_log,
NetworkControllerObserver* observer);
~GoogCcNetworkController() override;
// NetworkControllerInterface
void OnNetworkAvailability(NetworkAvailability msg) override;
void OnNetworkRouteChange(NetworkRouteChange msg) override;
void OnProcessInterval(ProcessInterval msg) override;
void OnRemoteBitrateReport(RemoteBitrateReport msg) override;
void OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override;
void OnSentPacket(SentPacket msg) override;
void OnStreamsConfig(StreamsConfig msg) override;
void OnTargetRateConstraints(TargetRateConstraints msg) override;
void OnTransportLossReport(TransportLossReport msg) override;
void OnTransportPacketsFeedback(TransportPacketsFeedback msg) override;
private:
void MaybeUpdateCongestionWindow();
void MaybeTriggerOnNetworkChanged(Timestamp at_time);
bool GetNetworkParameters(int32_t* estimated_bitrate_bps,
uint8_t* fraction_loss,
int64_t* rtt_ms,
Timestamp at_time);
void OnNetworkEstimate(NetworkEstimate msg);
void UpdatePacingRates(Timestamp at_time);
RtcEventLog* const event_log_;
NetworkControllerObserver* const observer_;
const std::unique_ptr<ProbeController> probe_controller_;
std::unique_ptr<SendSideBandwidthEstimation> bandwidth_estimation_;
std::unique_ptr<AlrDetector> alr_detector_;
std::unique_ptr<DelayBasedBwe> delay_based_bwe_;
std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator_;
std::deque<int64_t> feedback_rtts_;
rtc::Optional<int64_t> min_feedback_rtt_ms_;
rtc::Optional<NetworkEstimate> last_estimate_;
rtc::Optional<TargetTransferRate> last_target_rate_;
int32_t last_estimated_bitrate_bps_ = 0;
uint8_t last_estimated_fraction_loss_ = 0;
int64_t last_estimated_rtt_ms_ = 0;
double pacing_factor_;
DataRate min_pacing_rate_;
DataRate max_padding_rate_;
bool in_cwnd_experiment_;
int64_t accepted_queue_ms_;
bool previously_in_alr = false;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GoogCcNetworkController);
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_

32
modules/congestion_controller/goog_cc/include/goog_cc_factory.h Normal file
View File

@ -0,0 +1,32 @@
/*
* Copyright (c) 2018 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_CONGESTION_CONTROLLER_GOOG_CC_INCLUDE_GOOG_CC_FACTORY_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_INCLUDE_GOOG_CC_FACTORY_H_
#include "modules/congestion_controller/network_control/include/network_control.h"
namespace webrtc {
class Clock;
class RtcEventLog;
class GoogCcNetworkControllerFactory
: public NetworkControllerFactoryInterface {
public:
explicit GoogCcNetworkControllerFactory(RtcEventLog*);
NetworkControllerInterface::uptr Create(
NetworkControllerObserver* observer) override;
TimeDelta GetProcessInterval() const override;
private:
RtcEventLog* const event_log_;
};
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_INCLUDE_GOOG_CC_FACTORY_H_

95
modules/congestion_controller/goog_cc/median_slope_estimator.cc Normal file
View File

@ -0,0 +1,95 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/median_slope_estimator.h"
#include <algorithm>
#include <vector>
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace webrtc_cc {
constexpr unsigned int kDeltaCounterMax = 1000;
MedianSlopeEstimator::MedianSlopeEstimator(size_t window_size,
double threshold_gain)
: window_size_(window_size),
threshold_gain_(threshold_gain),
num_of_deltas_(0),
accumulated_delay_(0),
delay_hist_(),
median_filter_(0.5),
trendline_(0) {}
MedianSlopeEstimator::~MedianSlopeEstimator() {}
MedianSlopeEstimator::DelayInfo::DelayInfo(int64_t time,
double delay,
size_t slope_count)
: time(time), delay(delay) {
slopes.reserve(slope_count);
}
MedianSlopeEstimator::DelayInfo::~DelayInfo() = default;
void MedianSlopeEstimator::Update(double recv_delta_ms,
double send_delta_ms,
int64_t arrival_time_ms) {
const double delta_ms = recv_delta_ms - send_delta_ms;
++num_of_deltas_;
if (num_of_deltas_ > kDeltaCounterMax)
num_of_deltas_ = kDeltaCounterMax;
accumulated_delay_ += delta_ms;
BWE_TEST_LOGGING_PLOT(1, "accumulated_delay_ms", arrival_time_ms,
accumulated_delay_);
// If the window is full, remove the |window_size_| - 1 slopes that belong to
// the oldest point.
if (delay_hist_.size() == window_size_) {
for (double slope : delay_hist_.front().slopes) {
const bool success = median_filter_.Erase(slope);
RTC_CHECK(success);
}
delay_hist_.pop_front();
}
// Add |window_size_| - 1 new slopes.
for (auto& old_delay : delay_hist_) {
if (arrival_time_ms - old_delay.time != 0) {
// The C99 standard explicitly states that casts and assignments must
// perform the associated conversions. This means that |slope| will be
// a 64-bit double even if the division is computed using, e.g., 80-bit
// extended precision. I believe this also holds in C++ even though the
// C++11 standard isn't as explicit. Furthermore, there are good reasons
// to believe that compilers couldn't perform optimizations that break
// this assumption even if they wanted to.
double slope = (accumulated_delay_ - old_delay.delay) /
static_cast<double>(arrival_time_ms - old_delay.time);
median_filter_.Insert(slope);
// We want to avoid issues with different rounding mode / precision
// which we might get if we recomputed the slope when we remove it.
old_delay.slopes.push_back(slope);
}
}
delay_hist_.emplace_back(arrival_time_ms, accumulated_delay_,
window_size_ - 1);
// Recompute the median slope.
if (delay_hist_.size() == window_size_)
trendline_ = median_filter_.GetPercentileValue();
BWE_TEST_LOGGING_PLOT(1, "trendline_slope", arrival_time_ms, trendline_);
}
} // namespace webrtc_cc
} // namespace webrtc

74
modules/congestion_controller/goog_cc/median_slope_estimator.h Normal file
View File

@ -0,0 +1,74 @@
/*
* Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_MEDIAN_SLOPE_ESTIMATOR_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_MEDIAN_SLOPE_ESTIMATOR_H_
#include <stddef.h>
#include <stdint.h>
#include <deque>
#include <vector>
#include "rtc_base/constructormagic.h"
#include "rtc_base/numerics/percentile_filter.h"
namespace webrtc {
namespace webrtc_cc {
class MedianSlopeEstimator {
public:
// |window_size| is the number of points required to compute a trend line.
// |threshold_gain| is used to scale the trendline slope for comparison to
// the old threshold. Once the old estimator has been removed (or the
// thresholds been merged into the estimators), we can just set the
// threshold instead of setting a gain.
MedianSlopeEstimator(size_t window_size, double threshold_gain);
~MedianSlopeEstimator();
// Update the estimator with a new sample. The deltas should represent deltas
// between timestamp groups as defined by the InterArrival class.
void Update(double recv_delta_ms,
double send_delta_ms,
int64_t arrival_time_ms);
// Returns the estimated trend k multiplied by some gain.
// 0 < k < 1 -> the delay increases, queues are filling up
// k == 0 -> the delay does not change
// k < 0 -> the delay decreases, queues are being emptied
double trendline_slope() const { return trendline_ * threshold_gain_; }
// Returns the number of deltas which the current estimator state is based on.
unsigned int num_of_deltas() const { return num_of_deltas_; }
private:
struct DelayInfo {
DelayInfo(int64_t time, double delay, size_t slope_count);
~DelayInfo();
int64_t time;
double delay;
std::vector<double> slopes;
};
// Parameters.
const size_t window_size_;
const double threshold_gain_;
// Used by the existing threshold.
unsigned int num_of_deltas_;
// Theil-Sen robust line fitting
double accumulated_delay_;
std::deque<DelayInfo> delay_hist_;
PercentileFilter<double> median_filter_;
double trendline_;
RTC_DISALLOW_COPY_AND_ASSIGN(MedianSlopeEstimator);
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_MEDIAN_SLOPE_ESTIMATOR_H_

74
modules/congestion_controller/goog_cc/median_slope_estimator_unittest.cc Normal file
View File

@ -0,0 +1,74 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/median_slope_estimator.h"
#include "rtc_base/random.h"
#include "test/gtest.h"
namespace webrtc {
namespace webrtc_cc {
namespace {
constexpr size_t kWindowSize = 20;
constexpr double kGain = 1;
constexpr int64_t kAvgTimeBetweenPackets = 10;
constexpr size_t kPacketCount = 2 * kWindowSize + 1;
void TestEstimator(double slope, double jitter_stddev, double tolerance) {
MedianSlopeEstimator estimator(kWindowSize, kGain);
Random random(0x1234567);
int64_t send_times[kPacketCount];
int64_t recv_times[kPacketCount];
int64_t send_start_time = random.Rand(1000000);
int64_t recv_start_time = random.Rand(1000000);
for (size_t i = 0; i < kPacketCount; ++i) {
send_times[i] = send_start_time + i * kAvgTimeBetweenPackets;
double latency = i * kAvgTimeBetweenPackets / (1 - slope);
double jitter = random.Gaussian(0, jitter_stddev);
recv_times[i] = recv_start_time + latency + jitter;
}
for (size_t i = 1; i < kPacketCount; ++i) {
double recv_delta = recv_times[i] - recv_times[i - 1];
double send_delta = send_times[i] - send_times[i - 1];
estimator.Update(recv_delta, send_delta, recv_times[i]);
if (i < kWindowSize)
EXPECT_NEAR(estimator.trendline_slope(), 0, 0.001);
else
EXPECT_NEAR(estimator.trendline_slope(), slope, tolerance);
}
}
} // namespace
TEST(MedianSlopeEstimator, PerfectLineSlopeOneHalf) {
TestEstimator(0.5, 0, 0.001);
}
TEST(MedianSlopeEstimator, PerfectLineSlopeMinusOne) {
TestEstimator(-1, 0, 0.001);
}
TEST(MedianSlopeEstimator, PerfectLineSlopeZero) {
TestEstimator(0, 0, 0.001);
}
TEST(MedianSlopeEstimator, JitteryLineSlopeOneHalf) {
TestEstimator(0.5, kAvgTimeBetweenPackets / 3.0, 0.01);
}
TEST(MedianSlopeEstimator, JitteryLineSlopeMinusOne) {
TestEstimator(-1, kAvgTimeBetweenPackets / 3.0, 0.05);
}
TEST(MedianSlopeEstimator, JitteryLineSlopeZero) {
TestEstimator(0, kAvgTimeBetweenPackets / 3.0, 0.02);
}
} // namespace webrtc_cc
} // namespace webrtc

191
modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc Normal file
View File

@ -0,0 +1,191 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
#include <algorithm>
#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/ptr_util.h"
namespace {
// The minumum number of probes we need to receive feedback about in percent
// in order to have a valid estimate.
constexpr int kMinReceivedProbesPercent = 80;
// The minumum number of bytes we need to receive feedback about in percent
// in order to have a valid estimate.
constexpr int kMinReceivedBytesPercent = 80;
// The maximum |receive rate| / |send rate| ratio for a valid estimate.
constexpr float kMaxValidRatio = 2.0f;
// The minimum |receive rate| / |send rate| ratio assuming that the link is
// not saturated, i.e. we assume that we will receive at least
// kMinRatioForUnsaturatedLink * |send rate| if |send rate| is less than the
// link capacity.
constexpr float kMinRatioForUnsaturatedLink = 0.9f;
// The target utilization of the link. If we know true link capacity
// we'd like to send at 95% of that rate.
constexpr float kTargetUtilizationFraction = 0.95f;
// The maximum time period over which the cluster history is retained.
// This is also the maximum time period beyond which a probing burst is not
// expected to last.
constexpr int kMaxClusterHistoryMs = 1000;
// The maximum time interval between first and the last probe on a cluster
// on the sender side as well as the receive side.
constexpr int kMaxProbeIntervalMs = 1000;
} // namespace
namespace webrtc {
namespace webrtc_cc {
ProbeBitrateEstimator::ProbeBitrateEstimator(RtcEventLog* event_log)
: event_log_(event_log) {}
ProbeBitrateEstimator::~ProbeBitrateEstimator() = default;
int ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
const PacketFeedback& packet_feedback) {
int cluster_id = packet_feedback.pacing_info.probe_cluster_id;
RTC_DCHECK_NE(cluster_id, PacedPacketInfo::kNotAProbe);
EraseOldClusters(packet_feedback.arrival_time_ms - kMaxClusterHistoryMs);
int payload_size_bits =
rtc::dchecked_cast<int>(packet_feedback.payload_size * 8);
AggregatedCluster* cluster = &clusters_[cluster_id];
if (packet_feedback.send_time_ms < cluster->first_send_ms) {
cluster->first_send_ms = packet_feedback.send_time_ms;
}
if (packet_feedback.send_time_ms > cluster->last_send_ms) {
cluster->last_send_ms = packet_feedback.send_time_ms;
cluster->size_last_send = payload_size_bits;
}
if (packet_feedback.arrival_time_ms < cluster->first_receive_ms) {
cluster->first_receive_ms = packet_feedback.arrival_time_ms;
cluster->size_first_receive = payload_size_bits;
}
if (packet_feedback.arrival_time_ms > cluster->last_receive_ms) {
cluster->last_receive_ms = packet_feedback.arrival_time_ms;
}
cluster->size_total += payload_size_bits;
cluster->num_probes += 1;
RTC_DCHECK_GT(packet_feedback.pacing_info.probe_cluster_min_probes, 0);
RTC_DCHECK_GT(packet_feedback.pacing_info.probe_cluster_min_bytes, 0);
int min_probes = packet_feedback.pacing_info.probe_cluster_min_probes *
kMinReceivedProbesPercent / 100;
int min_bytes = packet_feedback.pacing_info.probe_cluster_min_bytes *
kMinReceivedBytesPercent / 100;
if (cluster->num_probes < min_probes || cluster->size_total < min_bytes * 8)
return -1;
float send_interval_ms = cluster->last_send_ms - cluster->first_send_ms;
float receive_interval_ms =
cluster->last_receive_ms - cluster->first_receive_ms;
if (send_interval_ms <= 0 || send_interval_ms > kMaxProbeIntervalMs ||
receive_interval_ms <= 0 || receive_interval_ms > kMaxProbeIntervalMs) {
RTC_LOG(LS_INFO) << "Probing unsuccessful, invalid send/receive interval"
<< " [cluster id: " << cluster_id
<< "] [send interval: " << send_interval_ms << " ms]"
<< " [receive interval: " << receive_interval_ms << " ms]";
if (event_log_) {
event_log_->Log(rtc::MakeUnique<RtcEventProbeResultFailure>(
cluster_id, ProbeFailureReason::kInvalidSendReceiveInterval));
}
return -1;
}
// Since the |send_interval_ms| does not include the time it takes to actually
// send the last packet the size of the last sent packet should not be
// included when calculating the send bitrate.
RTC_DCHECK_GT(cluster->size_total, cluster->size_last_send);
float send_size = cluster->size_total - cluster->size_last_send;
float send_bps = send_size / send_interval_ms * 1000;
// Since the |receive_interval_ms| does not include the time it takes to
// actually receive the first packet the size of the first received packet
// should not be included when calculating the receive bitrate.
RTC_DCHECK_GT(cluster->size_total, cluster->size_first_receive);
float receive_size = cluster->size_total - cluster->size_first_receive;
float receive_bps = receive_size / receive_interval_ms * 1000;
float ratio = receive_bps / send_bps;
if (ratio > kMaxValidRatio) {
RTC_LOG(LS_INFO) << "Probing unsuccessful, receive/send ratio too high"
<< " [cluster id: " << cluster_id
<< "] [send: " << send_size << " bytes / "
<< send_interval_ms << " ms = " << send_bps / 1000
<< " kb/s]"
<< " [receive: " << receive_size << " bytes / "
<< receive_interval_ms << " ms = " << receive_bps / 1000
<< " kb/s]"
<< " [ratio: " << receive_bps / 1000 << " / "
<< send_bps / 1000 << " = " << ratio
<< " > kMaxValidRatio (" << kMaxValidRatio << ")]";
if (event_log_) {
event_log_->Log(rtc::MakeUnique<RtcEventProbeResultFailure>(
cluster_id, ProbeFailureReason::kInvalidSendReceiveRatio));
}
return -1;
}
RTC_LOG(LS_INFO) << "Probing successful"
<< " [cluster id: " << cluster_id << "] [send: " << send_size
<< " bytes / " << send_interval_ms
<< " ms = " << send_bps / 1000 << " kb/s]"
<< " [receive: " << receive_size << " bytes / "
<< receive_interval_ms << " ms = " << receive_bps / 1000
<< " kb/s]";
float res = std::min(send_bps, receive_bps);
// If we're receiving at significantly lower bitrate than we were sending at,
// it suggests that we've found the true capacity of the link. In this case,
// set the target bitrate slightly lower to not immediately overuse.
if (receive_bps < kMinRatioForUnsaturatedLink * send_bps) {
RTC_DCHECK_GT(send_bps, receive_bps);
res = kTargetUtilizationFraction * receive_bps;
}
if (event_log_) {
event_log_->Log(
rtc::MakeUnique<RtcEventProbeResultSuccess>(cluster_id, res));
}
estimated_bitrate_bps_ = res;
return *estimated_bitrate_bps_;
}
rtc::Optional<int>
ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrateBps() {
rtc::Optional<int> estimated_bitrate_bps = estimated_bitrate_bps_;
estimated_bitrate_bps_.reset();
return estimated_bitrate_bps;
}
void ProbeBitrateEstimator::EraseOldClusters(int64_t timestamp_ms) {
for (auto it = clusters_.begin(); it != clusters_.end();) {
if (it->second.last_receive_ms < timestamp_ms) {
it = clusters_.erase(it);
} else {
++it;
}
}
}
} // namespace webrtc_cc
} // namespace webrtc

57
modules/congestion_controller/goog_cc/probe_bitrate_estimator.h Normal file
View File

@ -0,0 +1,57 @@
/*
* Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
#include <limits>
#include <map>
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
class RtcEventLog;
namespace webrtc_cc {
class ProbeBitrateEstimator {
public:
explicit ProbeBitrateEstimator(RtcEventLog* event_log);
~ProbeBitrateEstimator();
// Should be called for every probe packet we receive feedback about.
// Returns the estimated bitrate if the probe completes a valid cluster.
int HandleProbeAndEstimateBitrate(const PacketFeedback& packet_feedback);
rtc::Optional<int> FetchAndResetLastEstimatedBitrateBps();
private:
struct AggregatedCluster {
int num_probes = 0;
int64_t first_send_ms = std::numeric_limits<int64_t>::max();
int64_t last_send_ms = 0;
int64_t first_receive_ms = std::numeric_limits<int64_t>::max();
int64_t last_receive_ms = 0;
int size_last_send = 0;
int size_first_receive = 0;
int size_total = 0;
};
// Erases old cluster data that was seen before |timestamp_ms|.
void EraseOldClusters(int64_t timestamp_ms);
std::map<int, AggregatedCluster> clusters_;
RtcEventLog* const event_log_;
rtc::Optional<int> estimated_bitrate_bps_;
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_

223
modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc Normal file
View File

@ -0,0 +1,223 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
#include <utility>
#include <vector>
#include "modules/remote_bitrate_estimator/aimd_rate_control.h"
#include "test/gmock.h"
#include "test/gtest.h"
namespace webrtc {
namespace webrtc_cc {
namespace {
constexpr int kInvalidBitrate = -1;
constexpr int kDefaultMinProbes = 5;
constexpr int kDefaultMinBytes = 5000;
constexpr float kTargetUtilizationFraction = 0.95f;
} // anonymous namespace
class TestProbeBitrateEstimator : public ::testing::Test {
public:
TestProbeBitrateEstimator() : probe_bitrate_estimator_(nullptr) {}
// TODO(philipel): Use PacedPacketInfo when ProbeBitrateEstimator is rewritten
// to use that information.
void AddPacketFeedback(int probe_cluster_id,
size_t size_bytes,
int64_t send_time_ms,
int64_t arrival_time_ms,
int min_probes = kDefaultMinProbes,
int min_bytes = kDefaultMinBytes) {
PacedPacketInfo pacing_info(probe_cluster_id, min_probes, min_bytes);
PacketFeedback packet_feedback(arrival_time_ms, send_time_ms, 0, size_bytes,
pacing_info);
measured_bps_ =
probe_bitrate_estimator_.HandleProbeAndEstimateBitrate(packet_feedback);
}
protected:
int measured_bps_ = kInvalidBitrate;
ProbeBitrateEstimator probe_bitrate_estimator_;
};
TEST_F(TestProbeBitrateEstimator, OneCluster) {
AddPacketFeedback(0, 1000, 0, 10);
AddPacketFeedback(0, 1000, 10, 20);
AddPacketFeedback(0, 1000, 20, 30);
AddPacketFeedback(0, 1000, 30, 40);
EXPECT_NEAR(measured_bps_, 800000, 10);
}
TEST_F(TestProbeBitrateEstimator, OneClusterTooFewProbes) {
AddPacketFeedback(0, 2000, 0, 10);
AddPacketFeedback(0, 2000, 10, 20);
AddPacketFeedback(0, 2000, 20, 30);
EXPECT_EQ(kInvalidBitrate, measured_bps_);
}
TEST_F(TestProbeBitrateEstimator, OneClusterTooFewBytes) {
const int kMinBytes = 6000;
AddPacketFeedback(0, 800, 0, 10, kDefaultMinProbes, kMinBytes);
AddPacketFeedback(0, 800, 10, 20, kDefaultMinProbes, kMinBytes);
AddPacketFeedback(0, 800, 20, 30, kDefaultMinProbes, kMinBytes);
AddPacketFeedback(0, 800, 30, 40, kDefaultMinProbes, kMinBytes);
AddPacketFeedback(0, 800, 40, 50, kDefaultMinProbes, kMinBytes);
EXPECT_EQ(kInvalidBitrate, measured_bps_);
}
TEST_F(TestProbeBitrateEstimator, SmallCluster) {
const int kMinBytes = 1000;
AddPacketFeedback(0, 150, 0, 10, kDefaultMinProbes, kMinBytes);
AddPacketFeedback(0, 150, 10, 20, kDefaultMinProbes, kMinBytes);
AddPacketFeedback(0, 150, 20, 30, kDefaultMinProbes, kMinBytes);
AddPacketFeedback(0, 150, 30, 40, kDefaultMinProbes, kMinBytes);
AddPacketFeedback(0, 150, 40, 50, kDefaultMinProbes, kMinBytes);
AddPacketFeedback(0, 150, 50, 60, kDefaultMinProbes, kMinBytes);
EXPECT_NEAR(measured_bps_, 120000, 10);
}
TEST_F(TestProbeBitrateEstimator, LargeCluster) {
const int kMinProbes = 30;
const int kMinBytes = 312500;
int64_t send_time = 0;
int64_t receive_time = 5;
for (int i = 0; i < 25; ++i) {
AddPacketFeedback(0, 12500, send_time, receive_time, kMinProbes, kMinBytes);
++send_time;
++receive_time;
}
EXPECT_NEAR(measured_bps_, 100000000, 10);
}
TEST_F(TestProbeBitrateEstimator, FastReceive) {
AddPacketFeedback(0, 1000, 0, 15);
AddPacketFeedback(0, 1000, 10, 30);
AddPacketFeedback(0, 1000, 20, 35);
AddPacketFeedback(0, 1000, 30, 40);
EXPECT_NEAR(measured_bps_, 800000, 10);
}
TEST_F(TestProbeBitrateEstimator, TooFastReceive) {
AddPacketFeedback(0, 1000, 0, 19);
AddPacketFeedback(0, 1000, 10, 22);
AddPacketFeedback(0, 1000, 20, 25);
AddPacketFeedback(0, 1000, 40, 27);
EXPECT_EQ(measured_bps_, kInvalidBitrate);
}
TEST_F(TestProbeBitrateEstimator, SlowReceive) {
AddPacketFeedback(0, 1000, 0, 10);
AddPacketFeedback(0, 1000, 10, 40);
AddPacketFeedback(0, 1000, 20, 70);
AddPacketFeedback(0, 1000, 30, 85);
// Expected send rate = 800 kbps, expected receive rate = 320 kbps.
EXPECT_NEAR(measured_bps_, kTargetUtilizationFraction * 320000, 10);
}
TEST_F(TestProbeBitrateEstimator, BurstReceive) {
AddPacketFeedback(0, 1000, 0, 50);
AddPacketFeedback(0, 1000, 10, 50);
AddPacketFeedback(0, 1000, 20, 50);
AddPacketFeedback(0, 1000, 40, 50);
EXPECT_EQ(measured_bps_, kInvalidBitrate);
}
TEST_F(TestProbeBitrateEstimator, MultipleClusters) {
AddPacketFeedback(0, 1000, 0, 10);
AddPacketFeedback(0, 1000, 10, 20);
AddPacketFeedback(0, 1000, 20, 30);
AddPacketFeedback(0, 1000, 40, 60);
// Expected send rate = 600 kbps, expected receive rate = 480 kbps.
EXPECT_NEAR(measured_bps_, kTargetUtilizationFraction * 480000, 10);
AddPacketFeedback(0, 1000, 50, 60);
// Expected send rate = 640 kbps, expected receive rate = 640 kbps.
EXPECT_NEAR(measured_bps_, 640000, 10);
AddPacketFeedback(1, 1000, 60, 70);
AddPacketFeedback(1, 1000, 65, 77);
AddPacketFeedback(1, 1000, 70, 84);
AddPacketFeedback(1, 1000, 75, 90);
// Expected send rate = 1600 kbps, expected receive rate = 1200 kbps.
EXPECT_NEAR(measured_bps_, kTargetUtilizationFraction * 1200000, 10);
}
TEST_F(TestProbeBitrateEstimator, IgnoreOldClusters) {
AddPacketFeedback(0, 1000, 0, 10);
AddPacketFeedback(0, 1000, 10, 20);
AddPacketFeedback(0, 1000, 20, 30);
AddPacketFeedback(1, 1000, 60, 70);
AddPacketFeedback(1, 1000, 65, 77);
AddPacketFeedback(1, 1000, 70, 84);
AddPacketFeedback(1, 1000, 75, 90);
// Expected send rate = 1600 kbps, expected receive rate = 1200 kbps.
EXPECT_NEAR(measured_bps_, kTargetUtilizationFraction * 1200000, 10);
// Coming in 6s later
AddPacketFeedback(0, 1000, 40 + 6000, 60 + 6000);
EXPECT_EQ(measured_bps_, kInvalidBitrate);
}
TEST_F(TestProbeBitrateEstimator, IgnoreSizeLastSendPacket) {
AddPacketFeedback(0, 1000, 0, 10);
AddPacketFeedback(0, 1000, 10, 20);
AddPacketFeedback(0, 1000, 20, 30);
AddPacketFeedback(0, 1000, 30, 40);
AddPacketFeedback(0, 1500, 40, 50);
// Expected send rate = 800 kbps, expected receive rate = 900 kbps.
EXPECT_NEAR(measured_bps_, 800000, 10);
}
TEST_F(TestProbeBitrateEstimator, IgnoreSizeFirstReceivePacket) {
AddPacketFeedback(0, 1500, 0, 10);
AddPacketFeedback(0, 1000, 10, 20);
AddPacketFeedback(0, 1000, 20, 30);
AddPacketFeedback(0, 1000, 30, 40);
// Expected send rate = 933 kbps, expected receive rate = 800 kbps.
EXPECT_NEAR(measured_bps_, kTargetUtilizationFraction * 800000, 10);
}
TEST_F(TestProbeBitrateEstimator, NoLastEstimatedBitrateBps) {
EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps());
}
TEST_F(TestProbeBitrateEstimator, FetchLastEstimatedBitrateBps) {
AddPacketFeedback(0, 1000, 0, 10);
AddPacketFeedback(0, 1000, 10, 20);
AddPacketFeedback(0, 1000, 20, 30);
AddPacketFeedback(0, 1000, 30, 40);
auto estimated_bitrate_bps =
probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
EXPECT_TRUE(estimated_bitrate_bps);
EXPECT_NEAR(*estimated_bitrate_bps, 800000, 10);
EXPECT_FALSE(probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps());
}
} // namespace webrtc_cc
} // namespace webrtc

310
modules/congestion_controller/goog_cc/probe_controller.cc Normal file
View File

@ -0,0 +1,310 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/probe_controller.h"
#include <algorithm>
#include <initializer_list>
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
namespace webrtc {
namespace webrtc_cc {
namespace {
// The minimum number probing packets used.
constexpr int kMinProbePacketsSent = 5;
// The minimum probing duration in ms.
constexpr int kMinProbeDurationMs = 15;
// Maximum waiting time from the time of initiating probing to getting
// the measured results back.
constexpr int64_t kMaxWaitingTimeForProbingResultMs = 1000;
// Value of |min_bitrate_to_probe_further_bps_| that indicates
// further probing is disabled.
constexpr int kExponentialProbingDisabled = 0;
// Default probing bitrate limit. Applied only when the application didn't
// specify max bitrate.
constexpr int64_t kDefaultMaxProbingBitrateBps = 5000000;
// Interval between probes when ALR periodic probing is enabled.
constexpr int64_t kAlrPeriodicProbingIntervalMs = 5000;
// Minimum probe bitrate percentage to probe further for repeated probes,
// relative to the previous probe. For example, if 1Mbps probe results in
// 80kbps, then we'll probe again at 1.6Mbps. In that case second probe won't be
// sent if we get 600kbps from the first one.
constexpr int kRepeatedProbeMinPercentage = 70;
// If the bitrate drops to a factor |kBitrateDropThreshold| or lower
// and we recover within |kBitrateDropTimeoutMs|, then we'll send
// a probe at a fraction |kProbeFractionAfterDrop| of the original bitrate.
constexpr double kBitrateDropThreshold = 0.66;
constexpr int kBitrateDropTimeoutMs = 5000;
constexpr double kProbeFractionAfterDrop = 0.85;
// Timeout for probing after leaving ALR. If the bitrate drops significantly,
// (as determined by the delay based estimator) and we leave ALR, then we will
// send a probe if we recover within |kLeftAlrTimeoutMs| ms.
constexpr int kAlrEndedTimeoutMs = 3000;
// The expected uncertainty of probe result (as a fraction of the target probe
// This is a limit on how often probing can be done when there is a BW
// drop detected in ALR.
constexpr int64_t kMinTimeBetweenAlrProbesMs = 5000;
// bitrate). Used to avoid probing if the probe bitrate is close to our current
// estimate.
constexpr double kProbeUncertainty = 0.05;
// Use probing to recover faster after large bitrate estimate drops.
constexpr char kBweRapidRecoveryExperiment[] =
"WebRTC-BweRapidRecoveryExperiment";
} // namespace
ProbeController::ProbeController(NetworkControllerObserver* observer)
: observer_(observer), enable_periodic_alr_probing_(false) {
Reset(0);
in_rapid_recovery_experiment_ = webrtc::field_trial::FindFullName(
kBweRapidRecoveryExperiment) == "Enabled";
}
ProbeController::~ProbeController() {}
void ProbeController::SetBitrates(int64_t min_bitrate_bps,
int64_t start_bitrate_bps,
int64_t max_bitrate_bps,
int64_t at_time_ms) {
if (start_bitrate_bps > 0) {
start_bitrate_bps_ = start_bitrate_bps;
estimated_bitrate_bps_ = start_bitrate_bps;
} else if (start_bitrate_bps_ == 0) {
start_bitrate_bps_ = min_bitrate_bps;
}
// The reason we use the variable |old_max_bitrate_pbs| is because we
// need to set |max_bitrate_bps_| before we call InitiateProbing.
int64_t old_max_bitrate_bps = max_bitrate_bps_;
max_bitrate_bps_ = max_bitrate_bps;
switch (state_) {
case State::kInit:
if (network_available_)
InitiateExponentialProbing(at_time_ms);
break;
case State::kWaitingForProbingResult:
break;
case State::kProbingComplete:
// If the new max bitrate is higher than the old max bitrate and the
// estimate is lower than the new max bitrate then initiate probing.
if (estimated_bitrate_bps_ != 0 &&
old_max_bitrate_bps < max_bitrate_bps_ &&
estimated_bitrate_bps_ < max_bitrate_bps_) {
// The assumption is that if we jump more than 20% in the bandwidth
// estimate or if the bandwidth estimate is within 90% of the new
// max bitrate then the probing attempt was successful.
mid_call_probing_succcess_threshold_ =
std::min(estimated_bitrate_bps_ * 1.2, max_bitrate_bps_ * 0.9);
mid_call_probing_waiting_for_result_ = true;
mid_call_probing_bitrate_bps_ = max_bitrate_bps_;
RTC_HISTOGRAM_COUNTS_10000("WebRTC.BWE.MidCallProbing.Initiated",
max_bitrate_bps_ / 1000);
InitiateProbing(at_time_ms, {max_bitrate_bps}, false);
}
break;
}
}
void ProbeController::OnNetworkAvailability(NetworkAvailability msg) {
network_available_ = msg.network_available;
if (network_available_ && state_ == State::kInit && start_bitrate_bps_ > 0)
InitiateExponentialProbing(msg.at_time.ms());
}
void ProbeController::InitiateExponentialProbing(int64_t at_time_ms) {
RTC_DCHECK(network_available_);
RTC_DCHECK(state_ == State::kInit);
RTC_DCHECK_GT(start_bitrate_bps_, 0);
// When probing at 1.8 Mbps ( 6x 300), this represents a threshold of
// 1.2 Mbps to continue probing.
InitiateProbing(at_time_ms, {3 * start_bitrate_bps_, 6 * start_bitrate_bps_},
true);
}
void ProbeController::SetEstimatedBitrate(int64_t bitrate_bps,
int64_t at_time_ms) {
int64_t now_ms = at_time_ms;
if (mid_call_probing_waiting_for_result_ &&
bitrate_bps >= mid_call_probing_succcess_threshold_) {
RTC_HISTOGRAM_COUNTS_10000("WebRTC.BWE.MidCallProbing.Success",
mid_call_probing_bitrate_bps_ / 1000);
RTC_HISTOGRAM_COUNTS_10000("WebRTC.BWE.MidCallProbing.ProbedKbps",
bitrate_bps / 1000);
mid_call_probing_waiting_for_result_ = false;
}
if (state_ == State::kWaitingForProbingResult) {
// Continue probing if probing results indicate channel has greater
// capacity.
RTC_LOG(LS_INFO) << "Measured bitrate: " << bitrate_bps
<< " Minimum to probe further: "
<< min_bitrate_to_probe_further_bps_;
if (min_bitrate_to_probe_further_bps_ != kExponentialProbingDisabled &&
bitrate_bps > min_bitrate_to_probe_further_bps_) {
// Double the probing bitrate.
InitiateProbing(now_ms, {2 * bitrate_bps}, true);
}
}
if (bitrate_bps < kBitrateDropThreshold * estimated_bitrate_bps_) {
time_of_last_large_drop_ms_ = now_ms;
bitrate_before_last_large_drop_bps_ = estimated_bitrate_bps_;
}
estimated_bitrate_bps_ = bitrate_bps;
}
void ProbeController::EnablePeriodicAlrProbing(bool enable) {
enable_periodic_alr_probing_ = enable;
}
void ProbeController::SetAlrStartTimeMs(
rtc::Optional<int64_t> alr_start_time_ms) {
alr_start_time_ms_ = alr_start_time_ms;
}
void ProbeController::SetAlrEndedTimeMs(int64_t alr_end_time_ms) {
alr_end_time_ms_.emplace(alr_end_time_ms);
}
void ProbeController::RequestProbe(int64_t at_time_ms) {
// Called once we have returned to normal state after a large drop in
// estimated bandwidth. The current response is to initiate a single probe
// session (if not already probing) at the previous bitrate.
//
// If the probe session fails, the assumption is that this drop was a
// real one from a competing flow or a network change.
bool in_alr = alr_start_time_ms_.has_value();
bool alr_ended_recently =
(alr_end_time_ms_.has_value() &&
at_time_ms - alr_end_time_ms_.value() < kAlrEndedTimeoutMs);
if (in_alr || alr_ended_recently || in_rapid_recovery_experiment_) {
if (state_ == State::kProbingComplete) {
uint32_t suggested_probe_bps =
kProbeFractionAfterDrop * bitrate_before_last_large_drop_bps_;
uint32_t min_expected_probe_result_bps =
(1 - kProbeUncertainty) * suggested_probe_bps;
int64_t time_since_drop_ms = at_time_ms - time_of_last_large_drop_ms_;
int64_t time_since_probe_ms = at_time_ms - last_bwe_drop_probing_time_ms_;
if (min_expected_probe_result_bps > estimated_bitrate_bps_ &&
time_since_drop_ms < kBitrateDropTimeoutMs &&
time_since_probe_ms > kMinTimeBetweenAlrProbesMs) {
RTC_LOG(LS_INFO) << "Detected big bandwidth drop, start probing.";
// Track how often we probe in response to bandwidth drop in ALR.
RTC_HISTOGRAM_COUNTS_10000(
"WebRTC.BWE.BweDropProbingIntervalInS",
(at_time_ms - last_bwe_drop_probing_time_ms_) / 1000);
InitiateProbing(at_time_ms, {suggested_probe_bps}, false);
last_bwe_drop_probing_time_ms_ = at_time_ms;
}
}
}
}
void ProbeController::Reset(int64_t at_time_ms) {
network_available_ = true;
state_ = State::kInit;
min_bitrate_to_probe_further_bps_ = kExponentialProbingDisabled;
time_last_probing_initiated_ms_ = 0;
estimated_bitrate_bps_ = 0;
start_bitrate_bps_ = 0;
max_bitrate_bps_ = 0;
int64_t now_ms = at_time_ms;
last_bwe_drop_probing_time_ms_ = now_ms;
alr_end_time_ms_.reset();
mid_call_probing_waiting_for_result_ = false;
time_of_last_large_drop_ms_ = now_ms;
bitrate_before_last_large_drop_bps_ = 0;
}
void ProbeController::Process(int64_t at_time_ms) {
int64_t now_ms = at_time_ms;
if (now_ms - time_last_probing_initiated_ms_ >
kMaxWaitingTimeForProbingResultMs) {
mid_call_probing_waiting_for_result_ = false;
if (state_ == State::kWaitingForProbingResult) {
RTC_LOG(LS_INFO) << "kWaitingForProbingResult: timeout";
state_ = State::kProbingComplete;
min_bitrate_to_probe_further_bps_ = kExponentialProbingDisabled;
}
}
if (state_ != State::kProbingComplete || !enable_periodic_alr_probing_)
return;
// Probe bandwidth periodically when in ALR state.
if (alr_start_time_ms_ && estimated_bitrate_bps_ > 0) {
int64_t next_probe_time_ms =
std::max(*alr_start_time_ms_, time_last_probing_initiated_ms_) +
kAlrPeriodicProbingIntervalMs;
if (now_ms >= next_probe_time_ms) {
InitiateProbing(now_ms, {estimated_bitrate_bps_ * 2}, true);
}
}
}
void ProbeController::InitiateProbing(
int64_t now_ms,
std::initializer_list<int64_t> bitrates_to_probe,
bool probe_further) {
for (int64_t bitrate : bitrates_to_probe) {
RTC_DCHECK_GT(bitrate, 0);
int64_t max_probe_bitrate_bps =
max_bitrate_bps_ > 0 ? max_bitrate_bps_ : kDefaultMaxProbingBitrateBps;
if (bitrate > max_probe_bitrate_bps) {
bitrate = max_probe_bitrate_bps;
probe_further = false;
}
ProbeClusterConfig config;
config.at_time = Timestamp::ms(now_ms);
config.target_data_rate = DataRate::bps(rtc::dchecked_cast<int>(bitrate));
config.target_duration = TimeDelta::ms(kMinProbeDurationMs);
config.target_probe_count = kMinProbePacketsSent;
observer_->OnProbeClusterConfig(config);
}
time_last_probing_initiated_ms_ = now_ms;
if (probe_further) {
state_ = State::kWaitingForProbingResult;
min_bitrate_to_probe_further_bps_ =
(*(bitrates_to_probe.end() - 1)) * kRepeatedProbeMinPercentage / 100;
} else {
state_ = State::kProbingComplete;
min_bitrate_to_probe_further_bps_ = kExponentialProbingDisabled;
}
}
} // namespace webrtc_cc
} // namespace webrtc

100
modules/congestion_controller/goog_cc/probe_controller.h Normal file
View File

@ -0,0 +1,100 @@
/*
* Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
#include <stdint.h>
#include <initializer_list>
#include "api/optional.h"
#include "modules/congestion_controller/network_control/include/network_control.h"
namespace webrtc {
class Clock;
namespace webrtc_cc {
// This class controls initiation of probing to estimate initial channel
// capacity. There is also support for probing during a session when max
// bitrate is adjusted by an application.
class ProbeController {
public:
explicit ProbeController(NetworkControllerObserver* observer);
~ProbeController();
void SetBitrates(int64_t min_bitrate_bps,
int64_t start_bitrate_bps,
int64_t max_bitrate_bps,
int64_t at_time_ms);
void OnNetworkAvailability(NetworkAvailability msg);
void SetEstimatedBitrate(int64_t bitrate_bps, int64_t at_time_ms);
void EnablePeriodicAlrProbing(bool enable);
void SetAlrStartTimeMs(rtc::Optional<int64_t> alr_start_time);
void SetAlrEndedTimeMs(int64_t alr_end_time);
void RequestProbe(int64_t at_time_ms);
// Resets the ProbeController to a state equivalent to as if it was just
// created EXCEPT for |enable_periodic_alr_probing_|.
void Reset(int64_t at_time_ms);
void Process(int64_t at_time_ms);
private:
enum class State {
// Initial state where no probing has been triggered yet.
kInit,
// Waiting for probing results to continue further probing.
kWaitingForProbingResult,
// Probing is complete.
kProbingComplete,
};
void InitiateExponentialProbing(int64_t at_time_ms);
void InitiateProbing(int64_t now_ms,
std::initializer_list<int64_t> bitrates_to_probe,
bool probe_further);
NetworkControllerObserver* const observer_;
bool network_available_;
State state_;
int64_t min_bitrate_to_probe_further_bps_;
int64_t time_last_probing_initiated_ms_;
int64_t estimated_bitrate_bps_;
int64_t start_bitrate_bps_;
int64_t max_bitrate_bps_;
int64_t last_bwe_drop_probing_time_ms_;
rtc::Optional<int64_t> alr_start_time_ms_;
rtc::Optional<int64_t> alr_end_time_ms_;
bool enable_periodic_alr_probing_;
int64_t time_of_last_large_drop_ms_;
int64_t bitrate_before_last_large_drop_bps_;
bool in_rapid_recovery_experiment_;
// For WebRTC.BWE.MidCallProbing.* metric.
bool mid_call_probing_waiting_for_result_;
int64_t mid_call_probing_bitrate_bps_;
int64_t mid_call_probing_succcess_threshold_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ProbeController);
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_

292
modules/congestion_controller/goog_cc/probe_controller_unittest.cc Normal file
View File

@ -0,0 +1,292 @@
/*
* Copyright (c) 2016 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 <memory>
#include "modules/congestion_controller/goog_cc/probe_controller.h"
#include "modules/congestion_controller/network_control/include/network_types.h"
#include "rtc_base/logging.h"
#include "system_wrappers/include/clock.h"
#include "test/gmock.h"
#include "test/gtest.h"
using testing::_;
using testing::AtLeast;
using testing::Field;
using testing::Matcher;
using testing::NiceMock;
using testing::Return;
namespace webrtc {
namespace webrtc_cc {
namespace test {
namespace {
constexpr int kMinBitrateBps = 100;
constexpr int kStartBitrateBps = 300;
constexpr int kMaxBitrateBps = 10000;
constexpr int kExponentialProbingTimeoutMs = 5000;
constexpr int kAlrProbeInterval = 5000;
constexpr int kAlrEndedTimeoutMs = 3000;
constexpr int kBitrateDropTimeoutMs = 5000;
inline Matcher<ProbeClusterConfig> DataRateEqBps(int bps) {
return Field(&ProbeClusterConfig::target_data_rate, DataRate::bps(bps));
}
class MockNetworkControllerObserver : public NetworkControllerObserver {
public:
MOCK_METHOD1(OnCongestionWindow, void(CongestionWindow));
MOCK_METHOD1(OnPacerConfig, void(PacerConfig));
MOCK_METHOD1(OnProbeClusterConfig, void(ProbeClusterConfig));
MOCK_METHOD1(OnTargetTransferRate, void(TargetTransferRate));
};
} // namespace
class ProbeControllerTest : public ::testing::Test {
protected:
ProbeControllerTest() : clock_(100000000L) {
probe_controller_.reset(new ProbeController(&cluster_handler_));
}
~ProbeControllerTest() override {}
void SetNetworkAvailable(bool available) {
NetworkAvailability msg;
msg.at_time = Timestamp::ms(NowMs());
msg.network_available = available;
probe_controller_->OnNetworkAvailability(msg);
}
int64_t NowMs() { return clock_.TimeInMilliseconds(); }
SimulatedClock clock_;
NiceMock<MockNetworkControllerObserver> cluster_handler_;
std::unique_ptr<ProbeController> probe_controller_;
};
TEST_F(ProbeControllerTest, InitiatesProbingAtStart) {
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(AtLeast(2));
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
}
TEST_F(ProbeControllerTest, ProbeOnlyWhenNetworkIsUp) {
SetNetworkAvailable(false);
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(0);
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(AtLeast(2));
SetNetworkAvailable(true);
}
TEST_F(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncrease) {
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(AtLeast(2));
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
// Long enough to time out exponential probing.
clock_.AdvanceTimeMilliseconds(kExponentialProbingTimeoutMs);
probe_controller_->SetEstimatedBitrate(kStartBitrateBps, NowMs());
probe_controller_->Process(NowMs());
EXPECT_CALL(cluster_handler_,
OnProbeClusterConfig(DataRateEqBps(kMaxBitrateBps + 100)));
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps + 100, NowMs());
}
TEST_F(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncreaseAtMaxBitrate) {
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(AtLeast(2));
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
// Long enough to time out exponential probing.
clock_.AdvanceTimeMilliseconds(kExponentialProbingTimeoutMs);
probe_controller_->SetEstimatedBitrate(kStartBitrateBps, NowMs());
probe_controller_->Process(NowMs());
probe_controller_->SetEstimatedBitrate(kMaxBitrateBps, NowMs());
EXPECT_CALL(cluster_handler_,
OnProbeClusterConfig(DataRateEqBps(kMaxBitrateBps + 100)));
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps + 100, NowMs());
}
TEST_F(ProbeControllerTest, TestExponentialProbing) {
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
// Repeated probe should only be sent when estimated bitrate climbs above
// 0.7 * 6 * kStartBitrateBps = 1260.
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(0);
probe_controller_->SetEstimatedBitrate(1000, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(DataRateEqBps(2 * 1800)));
probe_controller_->SetEstimatedBitrate(1800, NowMs());
}
TEST_F(ProbeControllerTest, TestExponentialProbingTimeout) {
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
// Advance far enough to cause a time out in waiting for probing result.
clock_.AdvanceTimeMilliseconds(kExponentialProbingTimeoutMs);
probe_controller_->Process(NowMs());
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(0);
probe_controller_->SetEstimatedBitrate(1800, NowMs());
}
TEST_F(ProbeControllerTest, RequestProbeInAlr) {
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(2);
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
probe_controller_->SetEstimatedBitrate(500, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(DataRateEqBps(0.85 * 500)))
.Times(1);
probe_controller_->SetAlrStartTimeMs(clock_.TimeInMilliseconds());
clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
probe_controller_->Process(NowMs());
probe_controller_->SetEstimatedBitrate(250, NowMs());
probe_controller_->RequestProbe(NowMs());
}
TEST_F(ProbeControllerTest, RequestProbeWhenAlrEndedRecently) {
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(2);
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
probe_controller_->SetEstimatedBitrate(500, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(DataRateEqBps(0.85 * 500)))
.Times(1);
probe_controller_->SetAlrStartTimeMs(rtc::nullopt);
clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
probe_controller_->Process(NowMs());
probe_controller_->SetEstimatedBitrate(250, NowMs());
probe_controller_->SetAlrEndedTimeMs(clock_.TimeInMilliseconds());
clock_.AdvanceTimeMilliseconds(kAlrEndedTimeoutMs - 1);
probe_controller_->RequestProbe(NowMs());
}
TEST_F(ProbeControllerTest, RequestProbeWhenAlrNotEndedRecently) {
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(2);
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
probe_controller_->SetEstimatedBitrate(500, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(0);
probe_controller_->SetAlrStartTimeMs(rtc::nullopt);
clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
probe_controller_->Process(NowMs());
probe_controller_->SetEstimatedBitrate(250, NowMs());
probe_controller_->SetAlrEndedTimeMs(clock_.TimeInMilliseconds());
clock_.AdvanceTimeMilliseconds(kAlrEndedTimeoutMs + 1);
probe_controller_->RequestProbe(NowMs());
}
TEST_F(ProbeControllerTest, RequestProbeWhenBweDropNotRecent) {
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(2);
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
probe_controller_->SetEstimatedBitrate(500, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(0);
probe_controller_->SetAlrStartTimeMs(clock_.TimeInMilliseconds());
clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
probe_controller_->Process(NowMs());
probe_controller_->SetEstimatedBitrate(250, NowMs());
clock_.AdvanceTimeMilliseconds(kBitrateDropTimeoutMs + 1);
probe_controller_->RequestProbe(NowMs());
}
TEST_F(ProbeControllerTest, PeriodicProbing) {
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(2);
probe_controller_->EnablePeriodicAlrProbing(true);
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
probe_controller_->SetEstimatedBitrate(500, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
int64_t start_time = clock_.TimeInMilliseconds();
// Expect the controller to send a new probe after 5s has passed.
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(DataRateEqBps(1000)))
.Times(1);
probe_controller_->SetAlrStartTimeMs(start_time);
clock_.AdvanceTimeMilliseconds(5000);
probe_controller_->Process(NowMs());
probe_controller_->SetEstimatedBitrate(500, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
// The following probe should be sent at 10s into ALR.
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(0);
probe_controller_->SetAlrStartTimeMs(start_time);
clock_.AdvanceTimeMilliseconds(4000);
probe_controller_->Process(NowMs());
probe_controller_->SetEstimatedBitrate(500, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(1);
probe_controller_->SetAlrStartTimeMs(start_time);
clock_.AdvanceTimeMilliseconds(1000);
probe_controller_->Process(NowMs());
probe_controller_->SetEstimatedBitrate(500, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
}
TEST_F(ProbeControllerTest, PeriodicProbingAfterReset) {
NiceMock<MockNetworkControllerObserver> local_handler;
probe_controller_.reset(new ProbeController(&local_handler));
int64_t alr_start_time = clock_.TimeInMilliseconds();
probe_controller_->SetAlrStartTimeMs(alr_start_time);
EXPECT_CALL(local_handler, OnProbeClusterConfig(_)).Times(2);
probe_controller_->EnablePeriodicAlrProbing(true);
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
probe_controller_->Reset(NowMs());
clock_.AdvanceTimeMilliseconds(10000);
probe_controller_->Process(NowMs());
EXPECT_CALL(local_handler, OnProbeClusterConfig(_)).Times(2);
probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
kMaxBitrateBps, NowMs());
// Make sure we use |kStartBitrateBps| as the estimated bitrate
// until SetEstimatedBitrate is called with an updated estimate.
clock_.AdvanceTimeMilliseconds(10000);
EXPECT_CALL(local_handler,
OnProbeClusterConfig(DataRateEqBps(kStartBitrateBps * 2)));
probe_controller_->Process(NowMs());
}
TEST_F(ProbeControllerTest, TestExponentialProbingOverflow) {
const int64_t kMbpsMultiplier = 1000000;
probe_controller_->SetBitrates(kMinBitrateBps, 10 * kMbpsMultiplier,
100 * kMbpsMultiplier, NowMs());
// Verify that probe bitrate is capped at the specified max bitrate.
EXPECT_CALL(cluster_handler_,
OnProbeClusterConfig(DataRateEqBps(100 * kMbpsMultiplier)));
probe_controller_->SetEstimatedBitrate(60 * kMbpsMultiplier, NowMs());
testing::Mock::VerifyAndClearExpectations(&cluster_handler_);
// Verify that repeated probes aren't sent.
EXPECT_CALL(cluster_handler_, OnProbeClusterConfig(_)).Times(0);
probe_controller_->SetEstimatedBitrate(100 * kMbpsMultiplier, NowMs());
}
} // namespace test
} // namespace webrtc_cc
} // namespace webrtc

185
modules/congestion_controller/goog_cc/trendline_estimator.cc Normal file
View File

@ -0,0 +1,185 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/trendline_estimator.h"
#include <math.h>
#include <algorithm>
#include "api/optional.h"
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
namespace webrtc_cc {
namespace {
rtc::Optional<double> LinearFitSlope(
const std::deque<std::pair<double, double>>& points) {
RTC_DCHECK(points.size() >= 2);
// Compute the "center of mass".
double sum_x = 0;
double sum_y = 0;
for (const auto& point : points) {
sum_x += point.first;
sum_y += point.second;
}
double x_avg = sum_x / points.size();
double y_avg = sum_y / points.size();
// Compute the slope k = \sum (x_i-x_avg)(y_i-y_avg) / \sum (x_i-x_avg)^2
double numerator = 0;
double denominator = 0;
for (const auto& point : points) {
numerator += (point.first - x_avg) * (point.second - y_avg);
denominator += (point.first - x_avg) * (point.first - x_avg);
}
if (denominator == 0)
return rtc::nullopt;
return numerator / denominator;
}
constexpr double kMaxAdaptOffsetMs = 15.0;
constexpr double kOverUsingTimeThreshold = 10;
constexpr int kMinNumDeltas = 60;
} // namespace
enum { kDeltaCounterMax = 1000 };
TrendlineEstimator::TrendlineEstimator(size_t window_size,
double smoothing_coef,
double threshold_gain)
: window_size_(window_size),
smoothing_coef_(smoothing_coef),
threshold_gain_(threshold_gain),
num_of_deltas_(0),
first_arrival_time_ms_(-1),
accumulated_delay_(0),
smoothed_delay_(0),
delay_hist_(),
trendline_(0),
k_up_(0.0087),
k_down_(0.039),
overusing_time_threshold_(kOverUsingTimeThreshold),
threshold_(12.5),
last_update_ms_(-1),
prev_offset_(0.0),
time_over_using_(-1),
overuse_counter_(0),
hypothesis_(BandwidthUsage::kBwNormal) {}
TrendlineEstimator::~TrendlineEstimator() {}
void TrendlineEstimator::Update(double recv_delta_ms,
double send_delta_ms,
int64_t arrival_time_ms) {
const double delta_ms = recv_delta_ms - send_delta_ms;
++num_of_deltas_;
if (num_of_deltas_ > kDeltaCounterMax)
num_of_deltas_ = kDeltaCounterMax;
if (first_arrival_time_ms_ == -1)
first_arrival_time_ms_ = arrival_time_ms;
// Exponential backoff filter.
accumulated_delay_ += delta_ms;
BWE_TEST_LOGGING_PLOT(1, "accumulated_delay_ms", arrival_time_ms,
accumulated_delay_);
smoothed_delay_ = smoothing_coef_ * smoothed_delay_ +
(1 - smoothing_coef_) * accumulated_delay_;
BWE_TEST_LOGGING_PLOT(1, "smoothed_delay_ms", arrival_time_ms,
smoothed_delay_);
// Simple linear regression.
delay_hist_.push_back(std::make_pair(
static_cast<double>(arrival_time_ms - first_arrival_time_ms_),
smoothed_delay_));
if (delay_hist_.size() > window_size_)
delay_hist_.pop_front();
if (delay_hist_.size() == window_size_) {
// Only update trendline_ if it is possible to fit a line to the data.
trendline_ = LinearFitSlope(delay_hist_).value_or(trendline_);
}
BWE_TEST_LOGGING_PLOT(1, "trendline_slope", arrival_time_ms, trendline_);
Detect(trendline_slope(), send_delta_ms, num_of_deltas(), arrival_time_ms);
}
BandwidthUsage TrendlineEstimator::State() const {
return hypothesis_;
}
void TrendlineEstimator::Detect(double offset,
double ts_delta,
int num_of_deltas,
int64_t now_ms) {
if (num_of_deltas < 2) {
hypothesis_ = BandwidthUsage::kBwNormal;
return;
}
const double T = std::min(num_of_deltas, kMinNumDeltas) * offset;
BWE_TEST_LOGGING_PLOT(1, "T", now_ms, T);
BWE_TEST_LOGGING_PLOT(1, "threshold", now_ms, threshold_);
if (T > threshold_) {
if (time_over_using_ == -1) {
// Initialize the timer. Assume that we've been
// over-using half of the time since the previous
// sample.
time_over_using_ = ts_delta / 2;
} else {
// Increment timer
time_over_using_ += ts_delta;
}
overuse_counter_++;
if (time_over_using_ > overusing_time_threshold_ && overuse_counter_ > 1) {
if (offset >= prev_offset_) {
time_over_using_ = 0;
overuse_counter_ = 0;
hypothesis_ = BandwidthUsage::kBwOverusing;
}
}
} else if (T < -threshold_) {
time_over_using_ = -1;
overuse_counter_ = 0;
hypothesis_ = BandwidthUsage::kBwUnderusing;
} else {
time_over_using_ = -1;
overuse_counter_ = 0;
hypothesis_ = BandwidthUsage::kBwNormal;
}
prev_offset_ = offset;
UpdateThreshold(T, now_ms);
}
void TrendlineEstimator::UpdateThreshold(double modified_offset,
int64_t now_ms) {
if (last_update_ms_ == -1)
last_update_ms_ = now_ms;
if (fabs(modified_offset) > threshold_ + kMaxAdaptOffsetMs) {
// Avoid adapting the threshold to big latency spikes, caused e.g.,
// by a sudden capacity drop.
last_update_ms_ = now_ms;
return;
}
const double k = fabs(modified_offset) < threshold_ ? k_down_ : k_up_;
const int64_t kMaxTimeDeltaMs = 100;
int64_t time_delta_ms = std::min(now_ms - last_update_ms_, kMaxTimeDeltaMs);
threshold_ += k * (fabs(modified_offset) - threshold_) * time_delta_ms;
threshold_ = rtc::SafeClamp(threshold_, 6.f, 600.f);
last_update_ms_ = now_ms;
}
} // namespace webrtc_cc
} // namespace webrtc

94
modules/congestion_controller/goog_cc/trendline_estimator.h Normal file
View File

@ -0,0 +1,94 @@
/*
* Copyright (c) 2016 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_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
#include <stddef.h>
#include <stdint.h>
#include <deque>
#include <utility>
#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
namespace webrtc_cc {
class TrendlineEstimator : public DelayIncreaseDetectorInterface {
public:
// |window_size| is the number of points required to compute a trend line.
// |smoothing_coef| controls how much we smooth out the delay before fitting
// the trend line. |threshold_gain| is used to scale the trendline slope for
// comparison to the old threshold. Once the old estimator has been removed
// (or the thresholds been merged into the estimators), we can just set the
// threshold instead of setting a gain.
TrendlineEstimator(size_t window_size,
double smoothing_coef,
double threshold_gain);
~TrendlineEstimator() override;
// Update the estimator with a new sample. The deltas should represent deltas
// between timestamp groups as defined by the InterArrival class.
void Update(double recv_delta_ms,
double send_delta_ms,
int64_t arrival_time_ms) override;
BandwidthUsage State() const override;
// Returns the estimated trend k multiplied by some gain.
// 0 < k < 1 -> the delay increases, queues are filling up
// k == 0 -> the delay does not change
// k < 0 -> the delay decreases, queues are being emptied
double trendline_slope() const { return trendline_ * threshold_gain_; }
// Returns the number of deltas which the current estimator state is based on.
unsigned int num_of_deltas() const { return num_of_deltas_; }
private:
void Detect(double offset,
double ts_delta,
int num_of_deltas,
int64_t now_ms);
void UpdateThreshold(double modified_offset, int64_t now_ms);
// Parameters.
const size_t window_size_;
const double smoothing_coef_;
const double threshold_gain_;
// Used by the existing threshold.
unsigned int num_of_deltas_;
// Keep the arrival times small by using the change from the first packet.
int64_t first_arrival_time_ms_;
// Exponential backoff filtering.
double accumulated_delay_;
double smoothed_delay_;
// Linear least squares regression.
std::deque<std::pair<double, double>> delay_hist_;
double trendline_;
const double k_up_;
const double k_down_;
double overusing_time_threshold_;
double threshold_;
int64_t last_update_ms_;
double prev_offset_;
double time_over_using_;
int overuse_counter_;
BandwidthUsage hypothesis_;
RTC_DISALLOW_COPY_AND_ASSIGN(TrendlineEstimator);
};
} // namespace webrtc_cc
} // namespace webrtc
#endif // MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_

75
modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc Normal file
View File

@ -0,0 +1,75 @@
/*
* Copyright (c) 2016 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 "modules/congestion_controller/goog_cc/trendline_estimator.h"
#include "rtc_base/random.h"
#include "test/gtest.h"
namespace webrtc {
namespace webrtc_cc {
namespace {
constexpr size_t kWindowSize = 20;
constexpr double kSmoothing = 0.0;
constexpr double kGain = 1;
constexpr int64_t kAvgTimeBetweenPackets = 10;
constexpr size_t kPacketCount = 2 * kWindowSize + 1;
void TestEstimator(double slope, double jitter_stddev, double tolerance) {
TrendlineEstimator estimator(kWindowSize, kSmoothing, kGain);
Random random(0x1234567);
int64_t send_times[kPacketCount];
int64_t recv_times[kPacketCount];
int64_t send_start_time = random.Rand(1000000);
int64_t recv_start_time = random.Rand(1000000);
for (size_t i = 0; i < kPacketCount; ++i) {
send_times[i] = send_start_time + i * kAvgTimeBetweenPackets;
double latency = i * kAvgTimeBetweenPackets / (1 - slope);
double jitter = random.Gaussian(0, jitter_stddev);
recv_times[i] = recv_start_time + latency + jitter;
}
for (size_t i = 1; i < kPacketCount; ++i) {
double recv_delta = recv_times[i] - recv_times[i - 1];
double send_delta = send_times[i] - send_times[i - 1];
estimator.Update(recv_delta, send_delta, recv_times[i]);
if (i < kWindowSize)
EXPECT_NEAR(estimator.trendline_slope(), 0, 0.001);
else
EXPECT_NEAR(estimator.trendline_slope(), slope, tolerance);
}
}
} // namespace
TEST(TrendlineEstimator, PerfectLineSlopeOneHalf) {
TestEstimator(0.5, 0, 0.001);
}
TEST(TrendlineEstimator, PerfectLineSlopeMinusOne) {
TestEstimator(-1, 0, 0.001);
}
TEST(TrendlineEstimator, PerfectLineSlopeZero) {
TestEstimator(0, 0, 0.001);
}
TEST(TrendlineEstimator, JitteryLineSlopeOneHalf) {
TestEstimator(0.5, kAvgTimeBetweenPackets / 3.0, 0.01);
}
TEST(TrendlineEstimator, JitteryLineSlopeMinusOne) {
TestEstimator(-1, kAvgTimeBetweenPackets / 3.0, 0.075);
}
TEST(TrendlineEstimator, JitteryLineSlopeZero) {
TestEstimator(0, kAvgTimeBetweenPackets / 3.0, 0.02);
}
} // namespace webrtc_cc
} // namespace webrtc