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Removes legacy bitrate controller.

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Bug: webrtc:9883
Change-Id: I66af2597059fc1f38c78682f6884361a4d16c4a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141408
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28232}
This commit is contained in:
Sebastian Jansson 2019-06-11 11:26:25 +02:00 committed by Commit Bot
parent 171bd2644d
commit 7742b21839
7 changed files with 1 additions and 1000 deletions
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1
PRESUBMIT.py
View file

@ -75,7 +75,6 @@ LEGACY_API_DIRS = (
'common_audio/include',
'modules/audio_coding/include',
'modules/audio_processing/include',
'modules/bitrate_controller/include',
'modules/congestion_controller/include',
'modules/include',
'modules/remote_bitrate_estimator/include',

19
modules/bitrate_controller/BUILD.gn
View file

@ -11,10 +11,6 @@ import("../../webrtc.gni")
rtc_static_library("bitrate_controller") {
visibility = [ "*" ]
sources = [
"bitrate_controller.cc",
"bitrate_controller_impl.cc",
"bitrate_controller_impl.h",
"include/bitrate_controller.h",
"loss_based_bandwidth_estimation.cc",
"loss_based_bandwidth_estimation.h",
"send_side_bandwidth_estimation.cc",
@ -28,7 +24,6 @@ rtc_static_library("bitrate_controller") {
}
deps = [
"..:module_api",
"../../api/transport:network_control",
"../../api/units:data_rate",
"../../api/units:time_delta",
@ -36,14 +31,10 @@ rtc_static_library("bitrate_controller") {
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../rtc_base:checks",
"../../rtc_base:deprecation",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:logging",
"../../rtc_base/experiments:field_trial_parser",
"../../system_wrappers",
"../../system_wrappers:field_trial",
"../../system_wrappers:metrics",
"../congestion_controller/goog_cc:delay_based_bwe",
"../pacing",
"../remote_bitrate_estimator",
"../rtp_rtcp",
"../rtp_rtcp:rtp_rtcp_format",
@ -57,7 +48,6 @@ if (rtc_include_tests) {
testonly = true
sources = [
"bitrate_controller_unittest.cc",
"send_side_bandwidth_estimation_unittest.cc",
]
deps = [
@ -65,14 +55,7 @@ if (rtc_include_tests) {
"../../logging:mocks",
"../../logging:rtc_event_bwe",
"../../logging:rtc_event_log_api",
"../../system_wrappers",
"../../test:field_trial",
"../../test:test_support",
"../congestion_controller/goog_cc:delay_based_bwe",
"../pacing",
"../pacing:mock_paced_sender",
"../remote_bitrate_estimator",
"../rtp_rtcp:rtp_rtcp_format",
]
}
}

23
modules/bitrate_controller/bitrate_controller.cc
View file

@ -1,23 +0,0 @@
/*
* 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.
*
* Usage: this class will register multiple RtcpBitrateObserver's one at each
* RTCP module. It will aggregate the results and run one bandwidth estimation
* and push the result to the encoders via BitrateObserver(s).
*/
#include "modules/bitrate_controller/include/bitrate_controller.h"
namespace webrtc {
size_t BitrateObserver::pacer_queue_size_in_bytes() {
return 0;
}
} // namespace webrtc

292
modules/bitrate_controller/bitrate_controller_impl.cc
View file

@ -1,292 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*
*/
#include "modules/bitrate_controller/bitrate_controller_impl.h"
#include <algorithm>
#include <utility>
#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
namespace webrtc {
namespace {
absl::optional<DataRate> ToOptionalDataRate(int send_bitrate_bps) {
if (send_bitrate_bps > 0)
return DataRate::bps(send_bitrate_bps);
return absl::nullopt;
}
DataRate MaxRate(int max_bitrate_bps) {
if (max_bitrate_bps == -1)
return DataRate::Infinity();
return DataRate::bps(max_bitrate_bps);
}
} // namespace
class BitrateControllerImpl::RtcpBandwidthObserverImpl
: public RtcpBandwidthObserver {
public:
explicit RtcpBandwidthObserverImpl(BitrateControllerImpl* owner)
: owner_(owner) {}
~RtcpBandwidthObserverImpl() override = default;
// Received RTCP REMB or TMMBR.
void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
owner_->OnReceivedEstimatedBitrate(bitrate);
}
// Received RTCP receiver block.
void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
int64_t rtt,
int64_t now_ms) override {
owner_->OnReceivedRtcpReceiverReport(report_blocks, rtt, now_ms);
}
private:
BitrateControllerImpl* const owner_;
};
BitrateController* BitrateController::CreateBitrateController(
Clock* clock,
BitrateObserver* observer,
RtcEventLog* event_log) {
return new BitrateControllerImpl(clock, observer, event_log);
}
BitrateController* BitrateController::CreateBitrateController(
Clock* clock,
RtcEventLog* event_log) {
return CreateBitrateController(clock, nullptr, event_log);
}
BitrateControllerImpl::BitrateControllerImpl(Clock* clock,
BitrateObserver* observer,
RtcEventLog* event_log)
: clock_(clock),
observer_(observer),
last_bitrate_update_ms_(clock_->TimeInMilliseconds()),
event_log_(event_log),
bandwidth_estimation_(event_log),
last_bitrate_bps_(0),
last_fraction_loss_(0),
last_rtt_ms_(0) {
// This calls the observer_ if set, which means that the observer provided by
// the user must be ready to accept a bitrate update when it constructs the
// controller. We do this to avoid having to keep synchronized initial values
// in both the controller and the allocator.
MaybeTriggerOnNetworkChanged();
}
RtcpBandwidthObserver* BitrateControllerImpl::CreateRtcpBandwidthObserver() {
return new RtcpBandwidthObserverImpl(this);
}
void BitrateControllerImpl::SetStartBitrate(int start_bitrate_bps) {
{
rtc::CritScope cs(&critsect_);
bandwidth_estimation_.SetSendBitrate(
DataRate::bps(start_bitrate_bps),
Timestamp::ms(clock_->TimeInMilliseconds()));
}
MaybeTriggerOnNetworkChanged();
}
void BitrateControllerImpl::SetMinMaxBitrate(int min_bitrate_bps,
int max_bitrate_bps) {
{
rtc::CritScope cs(&critsect_);
bandwidth_estimation_.SetMinMaxBitrate(DataRate::bps(min_bitrate_bps),
DataRate::bps(max_bitrate_bps));
}
MaybeTriggerOnNetworkChanged();
}
void BitrateControllerImpl::SetBitrates(int start_bitrate_bps,
int min_bitrate_bps,
int max_bitrate_bps) {
{
rtc::CritScope cs(&critsect_);
bandwidth_estimation_.SetBitrates(
ToOptionalDataRate(start_bitrate_bps), DataRate::bps(min_bitrate_bps),
MaxRate(max_bitrate_bps), Timestamp::ms(clock_->TimeInMilliseconds()));
}
MaybeTriggerOnNetworkChanged();
}
void BitrateControllerImpl::ResetBitrates(int bitrate_bps,
int min_bitrate_bps,
int max_bitrate_bps) {
{
rtc::CritScope cs(&critsect_);
bandwidth_estimation_ = SendSideBandwidthEstimation(event_log_);
bandwidth_estimation_.SetBitrates(
ToOptionalDataRate(bitrate_bps), DataRate::bps(min_bitrate_bps),
MaxRate(max_bitrate_bps), Timestamp::ms(clock_->TimeInMilliseconds()));
}
MaybeTriggerOnNetworkChanged();
}
// This is called upon reception of REMB or TMMBR.
void BitrateControllerImpl::OnReceivedEstimatedBitrate(uint32_t bitrate) {
{
rtc::CritScope cs(&critsect_);
bandwidth_estimation_.UpdateReceiverEstimate(
Timestamp::ms(clock_->TimeInMilliseconds()), DataRate::bps(bitrate));
BWE_TEST_LOGGING_PLOT(1, "REMB_kbps", clock_->TimeInMilliseconds(),
bitrate / 1000);
}
MaybeTriggerOnNetworkChanged();
}
void BitrateControllerImpl::OnDelayBasedBweResult(
const DelayBasedBwe::Result& result) {
if (!result.updated)
return;
{
rtc::CritScope cs(&critsect_);
if (result.probe) {
bandwidth_estimation_.SetSendBitrate(
result.target_bitrate, Timestamp::ms(clock_->TimeInMilliseconds()));
}
// Since SetSendBitrate now resets the delay-based estimate, we have to call
// UpdateDelayBasedEstimate after SetSendBitrate.
bandwidth_estimation_.UpdateDelayBasedEstimate(
Timestamp::ms(clock_->TimeInMilliseconds()), result.target_bitrate);
}
MaybeTriggerOnNetworkChanged();
}
int64_t BitrateControllerImpl::TimeUntilNextProcess() {
const int64_t kBitrateControllerUpdateIntervalMs = 25;
rtc::CritScope cs(&critsect_);
int64_t time_since_update_ms =
clock_->TimeInMilliseconds() - last_bitrate_update_ms_;
return std::max<int64_t>(
kBitrateControllerUpdateIntervalMs - time_since_update_ms, 0);
}
void BitrateControllerImpl::Process() {
{
rtc::CritScope cs(&critsect_);
bandwidth_estimation_.UpdateEstimate(
Timestamp::ms(clock_->TimeInMilliseconds()));
}
MaybeTriggerOnNetworkChanged();
last_bitrate_update_ms_ = clock_->TimeInMilliseconds();
}
void BitrateControllerImpl::OnReceivedRtcpReceiverReport(
const ReportBlockList& report_blocks,
int64_t rtt,
int64_t now_ms) {
if (report_blocks.empty())
return;
{
rtc::CritScope cs(&critsect_);
int fraction_lost_aggregate = 0;
int total_number_of_packets = 0;
// Compute the a weighted average of the fraction loss from all report
// blocks.
for (const RTCPReportBlock& report_block : report_blocks) {
std::map<uint32_t, uint32_t>::iterator seq_num_it =
ssrc_to_last_received_extended_high_seq_num_.find(
report_block.source_ssrc);
int number_of_packets = 0;
if (seq_num_it != ssrc_to_last_received_extended_high_seq_num_.end()) {
number_of_packets =
report_block.extended_highest_sequence_number - seq_num_it->second;
}
fraction_lost_aggregate += number_of_packets * report_block.fraction_lost;
total_number_of_packets += number_of_packets;
// Update last received for this SSRC.
ssrc_to_last_received_extended_high_seq_num_[report_block.source_ssrc] =
report_block.extended_highest_sequence_number;
}
if (total_number_of_packets < 0) {
RTC_LOG(LS_WARNING)
<< "Received report block where extended high sequence "
"number goes backwards, ignoring.";
return;
}
if (total_number_of_packets == 0)
fraction_lost_aggregate = 0;
else
fraction_lost_aggregate =
(fraction_lost_aggregate + total_number_of_packets / 2) /
total_number_of_packets;
if (fraction_lost_aggregate > 255)
return;
RTC_DCHECK_GE(total_number_of_packets, 0);
bandwidth_estimation_.UpdateReceiverBlock(
fraction_lost_aggregate, TimeDelta::ms(rtt), total_number_of_packets,
Timestamp::ms(now_ms));
}
MaybeTriggerOnNetworkChanged();
}
void BitrateControllerImpl::MaybeTriggerOnNetworkChanged() {
if (!observer_)
return;
uint32_t bitrate_bps;
uint8_t fraction_loss;
int64_t rtt;
if (GetNetworkParameters(&bitrate_bps, &fraction_loss, &rtt))
observer_->OnNetworkChanged(bitrate_bps, fraction_loss, rtt);
}
bool BitrateControllerImpl::GetNetworkParameters(uint32_t* bitrate,
uint8_t* fraction_loss,
int64_t* rtt) {
rtc::CritScope cs(&critsect_);
int current_bitrate;
bandwidth_estimation_.CurrentEstimate(&current_bitrate, fraction_loss, rtt);
*bitrate = current_bitrate;
bool new_bitrate = false;
if (*bitrate != last_bitrate_bps_ || *fraction_loss != last_fraction_loss_ ||
*rtt != last_rtt_ms_) {
last_bitrate_bps_ = *bitrate;
last_fraction_loss_ = *fraction_loss;
last_rtt_ms_ = *rtt;
new_bitrate = true;
}
BWE_TEST_LOGGING_PLOT(1, "fraction_loss_%", clock_->TimeInMilliseconds(),
(last_fraction_loss_ * 100) / 256);
BWE_TEST_LOGGING_PLOT(1, "rtt_ms", clock_->TimeInMilliseconds(),
last_rtt_ms_);
BWE_TEST_LOGGING_PLOT(1, "Target_bitrate_kbps", clock_->TimeInMilliseconds(),
last_bitrate_bps_ / 1000);
return new_bitrate;
}
bool BitrateControllerImpl::AvailableBandwidth(uint32_t* bandwidth) const {
rtc::CritScope cs(&critsect_);
int bitrate;
uint8_t fraction_loss;
int64_t rtt;
bandwidth_estimation_.CurrentEstimate(&bitrate, &fraction_loss, &rtt);
if (bitrate > 0) {
*bandwidth = bitrate;
return true;
}
return false;
}
} // namespace webrtc

103
modules/bitrate_controller/bitrate_controller_impl.h
View file

@ -1,103 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*
* Usage: this class will register multiple RtcpBitrateObserver's one at each
* RTCP module. It will aggregate the results and run one bandwidth estimation
* and push the result to the encoder via VideoEncoderCallback.
*/
#ifndef MODULES_BITRATE_CONTROLLER_BITRATE_CONTROLLER_IMPL_H_
#define MODULES_BITRATE_CONTROLLER_BITRATE_CONTROLLER_IMPL_H_
#include "modules/bitrate_controller/include/bitrate_controller.h"
#include <list>
#include <map>
#include <utility>
#include <vector>
#include "modules/bitrate_controller/send_side_bandwidth_estimation.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
namespace webrtc {
class BitrateControllerImpl : public BitrateController {
public:
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
// |observer| is left for project that is not yet updated.
BitrateControllerImpl(Clock* clock,
BitrateObserver* observer,
RtcEventLog* event_log);
virtual ~BitrateControllerImpl() {}
bool AvailableBandwidth(uint32_t* bandwidth) const override;
RTC_DEPRECATED RtcpBandwidthObserver* CreateRtcpBandwidthObserver() override;
// Deprecated
void SetStartBitrate(int start_bitrate_bps) override;
// Deprecated
void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) override;
void SetBitrates(int start_bitrate_bps,
int min_bitrate_bps,
int max_bitrate_bps) override;
void ResetBitrates(int bitrate_bps,
int min_bitrate_bps,
int max_bitrate_bps) override;
// Returns true if the parameters have changed since the last call.
bool GetNetworkParameters(uint32_t* bitrate,
uint8_t* fraction_loss,
int64_t* rtt) override;
void OnDelayBasedBweResult(const DelayBasedBwe::Result& result) override;
int64_t TimeUntilNextProcess() override;
void Process() override;
private:
class RtcpBandwidthObserverImpl;
// Called by BitrateObserver's direct from the RTCP module.
// Implements RtcpBandwidthObserver.
void OnReceivedEstimatedBitrate(uint32_t bitrate) override;
void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
int64_t rtt,
int64_t now_ms) override;
// Deprecated
void MaybeTriggerOnNetworkChanged();
void OnNetworkChanged(uint32_t bitrate,
uint8_t fraction_loss, // 0 - 255.
int64_t rtt) RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
// Used by process thread.
Clock* const clock_;
BitrateObserver* const observer_;
int64_t last_bitrate_update_ms_;
RtcEventLog* const event_log_;
rtc::CriticalSection critsect_;
std::map<uint32_t, uint32_t> ssrc_to_last_received_extended_high_seq_num_
RTC_GUARDED_BY(critsect_);
SendSideBandwidthEstimation bandwidth_estimation_ RTC_GUARDED_BY(critsect_);
uint32_t last_bitrate_bps_ RTC_GUARDED_BY(critsect_);
uint8_t last_fraction_loss_ RTC_GUARDED_BY(critsect_);
int64_t last_rtt_ms_ RTC_GUARDED_BY(critsect_);
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(BitrateControllerImpl);
};
} // namespace webrtc
#endif // MODULES_BITRATE_CONTROLLER_BITRATE_CONTROLLER_IMPL_H_

461
modules/bitrate_controller/bitrate_controller_unittest.cc
View file

@ -1,461 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <stdint.h>
#include <memory>
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/bitrate_controller/include/bitrate_controller.h"
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "modules/pacing/paced_sender.h"
#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "test/gmock.h"
#include "test/gtest.h"
using ::testing::Exactly;
using ::testing::Return;
using webrtc::BitrateController;
using webrtc::BitrateObserver;
using webrtc::PacedSender;
using webrtc::RtcpBandwidthObserver;
uint8_t WeightedLoss(int num_packets1,
uint8_t fraction_loss1,
int num_packets2,
uint8_t fraction_loss2) {
int weighted_sum =
num_packets1 * fraction_loss1 + num_packets2 * fraction_loss2;
int total_num_packets = num_packets1 + num_packets2;
return (weighted_sum + total_num_packets / 2) / total_num_packets;
}
webrtc::RTCPReportBlock CreateReportBlock(
uint32_t remote_ssrc,
uint32_t source_ssrc,
uint8_t fraction_lost,
uint32_t extended_high_sequence_number) {
return webrtc::RTCPReportBlock(remote_ssrc, source_ssrc, fraction_lost, 0,
extended_high_sequence_number, 0, 0, 0);
}
class TestBitrateObserver : public BitrateObserver {
public:
TestBitrateObserver()
: last_bitrate_(0), last_fraction_loss_(0), last_rtt_(0) {}
virtual void OnNetworkChanged(uint32_t bitrate,
uint8_t fraction_loss,
int64_t rtt) {
last_bitrate_ = static_cast<int>(bitrate);
last_fraction_loss_ = fraction_loss;
last_rtt_ = rtt;
}
int last_bitrate_;
uint8_t last_fraction_loss_;
int64_t last_rtt_;
};
class BitrateControllerTest : public ::testing::Test {
protected:
BitrateControllerTest() : clock_(0) {}
~BitrateControllerTest() {}
virtual void SetUp() {
controller_.reset(BitrateController::CreateBitrateController(
&clock_, &bitrate_observer_, &event_log_));
controller_->SetStartBitrate(kStartBitrateBps);
EXPECT_EQ(kStartBitrateBps, bitrate_observer_.last_bitrate_);
controller_->SetMinMaxBitrate(kMinBitrateBps, kMaxBitrateBps);
EXPECT_EQ(kStartBitrateBps, bitrate_observer_.last_bitrate_);
bandwidth_observer_ = controller_.get();
}
virtual void TearDown() {}
const int kMinBitrateBps = 100000;
const int kStartBitrateBps = 200000;
const int kMaxBitrateBps = 300000;
const int kDefaultMinBitrateBps = 10000;
const int kDefaultMaxBitrateBps = 1000000000;
webrtc::SimulatedClock clock_;
TestBitrateObserver bitrate_observer_;
std::unique_ptr<BitrateController> controller_;
RtcpBandwidthObserver* bandwidth_observer_;
::testing::NiceMock<webrtc::MockRtcEventLog> event_log_;
};
TEST_F(BitrateControllerTest, DefaultMinMaxBitrate) {
// Receive successively lower REMBs, verify the reserved bitrate is deducted.
controller_->SetMinMaxBitrate(0, 0);
EXPECT_EQ(kStartBitrateBps, bitrate_observer_.last_bitrate_);
bandwidth_observer_->OnReceivedEstimatedBitrate(kDefaultMinBitrateBps / 2);
EXPECT_EQ(webrtc::congestion_controller::GetMinBitrateBps(),
bitrate_observer_.last_bitrate_);
bandwidth_observer_->OnReceivedEstimatedBitrate(2 * kDefaultMaxBitrateBps);
clock_.AdvanceTimeMilliseconds(1000);
controller_->Process();
EXPECT_EQ(kDefaultMaxBitrateBps, bitrate_observer_.last_bitrate_);
}
TEST_F(BitrateControllerTest, OneBitrateObserverOneRtcpObserver) {
// First REMB applies immediately.
int64_t time_ms = 1001;
webrtc::ReportBlockList report_blocks;
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
bandwidth_observer_->OnReceivedEstimatedBitrate(200000);
EXPECT_EQ(200000, bitrate_observer_.last_bitrate_);
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(0, bitrate_observer_.last_rtt_);
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
report_blocks.clear();
time_ms += 2000;
// Receive a high remb, test bitrate inc.
bandwidth_observer_->OnReceivedEstimatedBitrate(400000);
// Test bitrate increase 8% per second.
report_blocks.push_back(CreateReportBlock(1, 2, 0, 21));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
time_ms += 1000;
report_blocks.clear();
report_blocks.push_back(CreateReportBlock(1, 2, 0, 41));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_EQ(235360, bitrate_observer_.last_bitrate_);
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
time_ms += 1000;
report_blocks.clear();
report_blocks.push_back(CreateReportBlock(1, 2, 0, 61));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_EQ(255189, bitrate_observer_.last_bitrate_);
time_ms += 1000;
report_blocks.clear();
report_blocks.push_back(CreateReportBlock(1, 2, 0, 81));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_EQ(276604, bitrate_observer_.last_bitrate_);
time_ms += 1000;
report_blocks.clear();
report_blocks.push_back(CreateReportBlock(1, 2, 0, 101));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_EQ(299732, bitrate_observer_.last_bitrate_);
time_ms += 1000;
// Reach max cap.
report_blocks.clear();
report_blocks.push_back(CreateReportBlock(1, 2, 0, 121));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_EQ(300000, bitrate_observer_.last_bitrate_);
time_ms += 1000;
report_blocks.clear();
report_blocks.push_back(CreateReportBlock(1, 2, 0, 141));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_EQ(300000, bitrate_observer_.last_bitrate_);
// Test that a low delay-based estimate limits the combined estimate.
webrtc::DelayBasedBwe::Result result(false, webrtc::DataRate::kbps(280));
controller_->OnDelayBasedBweResult(result);
EXPECT_EQ(280000, bitrate_observer_.last_bitrate_);
// Test that a low REMB limits the combined estimate.
bandwidth_observer_->OnReceivedEstimatedBitrate(250000);
EXPECT_EQ(250000, bitrate_observer_.last_bitrate_);
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
bandwidth_observer_->OnReceivedEstimatedBitrate(1000);
EXPECT_EQ(100000, bitrate_observer_.last_bitrate_);
}
TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) {
const uint32_t kSenderSsrc1 = 1;
const uint32_t kSenderSsrc2 = 2;
const uint32_t kMediaSsrc1 = 3;
const uint32_t kMediaSsrc2 = 4;
int64_t time_ms = 1;
webrtc::ReportBlockList report_blocks;
report_blocks = {CreateReportBlock(kSenderSsrc1, kMediaSsrc1, 0, 1)};
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
time_ms += 500;
RtcpBandwidthObserver* second_bandwidth_observer = controller_.get();
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 21)};
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
time_ms);
// Test start bitrate.
EXPECT_EQ(200000, bitrate_observer_.last_bitrate_);
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(100, bitrate_observer_.last_rtt_);
time_ms += 500;
// Test bitrate increase 8% per second.
report_blocks = {CreateReportBlock(kSenderSsrc1, kMediaSsrc1, 0, 21)};
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
time_ms += 500;
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 21)};
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
time_ms);
EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(100, bitrate_observer_.last_rtt_);
time_ms += 500;
// Extra report should not change estimate.
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 31)};
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
time_ms);
EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
time_ms += 500;
report_blocks = {CreateReportBlock(kSenderSsrc1, kMediaSsrc1, 0, 41)};
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_EQ(235360, bitrate_observer_.last_bitrate_);
// Second report should not change estimate.
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 41)};
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
time_ms);
EXPECT_EQ(235360, bitrate_observer_.last_bitrate_);
time_ms += 1000;
// Reports from only one bandwidth observer is ok.
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 61)};
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
time_ms);
EXPECT_EQ(255189, bitrate_observer_.last_bitrate_);
time_ms += 1000;
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 81)};
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
time_ms);
EXPECT_EQ(276604, bitrate_observer_.last_bitrate_);
time_ms += 1000;
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 121)};
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
time_ms);
EXPECT_EQ(299732, bitrate_observer_.last_bitrate_);
time_ms += 1000;
// Reach max cap.
report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 141)};
second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
time_ms);
EXPECT_EQ(300000, bitrate_observer_.last_bitrate_);
// Test that a low REMB trigger immediately.
// We don't care which bandwidth observer that delivers the REMB.
second_bandwidth_observer->OnReceivedEstimatedBitrate(250000);
EXPECT_EQ(250000, bitrate_observer_.last_bitrate_);
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
// Min cap.
bandwidth_observer_->OnReceivedEstimatedBitrate(1000);
EXPECT_EQ(100000, bitrate_observer_.last_bitrate_);
}
TEST_F(BitrateControllerTest, OneBitrateObserverMultipleReportBlocks) {
uint32_t sequence_number[2] = {0, 0xFF00};
const int kStartBitrate = 200000;
const int kMinBitrate = 100000;
const int kMaxBitrate = 300000;
controller_->SetStartBitrate(kStartBitrate);
controller_->SetMinMaxBitrate(kMinBitrate, kMaxBitrate);
// REMBs during the first 2 seconds apply immediately.
int64_t time_ms = 1001;
webrtc::ReportBlockList report_blocks;
report_blocks.push_back(CreateReportBlock(1, 2, 0, sequence_number[0]));
bandwidth_observer_->OnReceivedEstimatedBitrate(kStartBitrate);
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
report_blocks.clear();
time_ms += 2000;
// Receive a high REMB, test bitrate increase.
bandwidth_observer_->OnReceivedEstimatedBitrate(400000);
int last_bitrate = 0;
// Ramp up to max bitrate.
for (int i = 0; i < 7; ++i) {
report_blocks.push_back(CreateReportBlock(1, 2, 0, sequence_number[0]));
report_blocks.push_back(CreateReportBlock(1, 3, 0, sequence_number[1]));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50,
time_ms);
EXPECT_GT(bitrate_observer_.last_bitrate_, last_bitrate);
EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
last_bitrate = bitrate_observer_.last_bitrate_;
time_ms += 1000;
sequence_number[0] += 20;
sequence_number[1] += 1;
report_blocks.clear();
}
EXPECT_EQ(kMaxBitrate, bitrate_observer_.last_bitrate_);
// Packet loss on the first stream. Verify that bitrate decreases.
report_blocks.push_back(CreateReportBlock(1, 2, 50, sequence_number[0]));
report_blocks.push_back(CreateReportBlock(1, 3, 0, sequence_number[1]));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_LT(bitrate_observer_.last_bitrate_, last_bitrate);
EXPECT_EQ(WeightedLoss(20, 50, 1, 0), bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
last_bitrate = bitrate_observer_.last_bitrate_;
sequence_number[0] += 20;
sequence_number[1] += 20;
time_ms += 1000;
report_blocks.clear();
// Packet loss on the second stream. Verify that bitrate decreases.
report_blocks.push_back(CreateReportBlock(1, 2, 0, sequence_number[0]));
report_blocks.push_back(CreateReportBlock(1, 3, 75, sequence_number[1]));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_LT(bitrate_observer_.last_bitrate_, last_bitrate);
EXPECT_EQ(WeightedLoss(20, 0, 20, 75), bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
last_bitrate = bitrate_observer_.last_bitrate_;
sequence_number[0] += 20;
sequence_number[1] += 1;
time_ms += 1000;
report_blocks.clear();
// All packets lost on stream with few packets, no back-off.
report_blocks.push_back(CreateReportBlock(1, 2, 0, sequence_number[0]));
report_blocks.push_back(CreateReportBlock(1, 3, 255, sequence_number[1]));
bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
EXPECT_EQ(bitrate_observer_.last_bitrate_, last_bitrate);
EXPECT_EQ(WeightedLoss(20, 0, 1, 255), bitrate_observer_.last_fraction_loss_);
EXPECT_EQ(50, bitrate_observer_.last_rtt_);
last_bitrate = bitrate_observer_.last_bitrate_;
sequence_number[0] += 20;
sequence_number[1] += 1;
report_blocks.clear();
}
TEST_F(BitrateControllerTest, TimeoutsWithoutFeedback) {
{
webrtc::test::ScopedFieldTrials override_field_trials(
"WebRTC-FeedbackTimeout/Enabled/");
SetUp();
int expected_bitrate_bps = 300000;
controller_->SetBitrates(300000, kDefaultMinBitrateBps,
kDefaultMaxBitrateBps);
webrtc::ReportBlockList report_blocks;
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
bandwidth_observer_->OnReceivedRtcpReceiverReport(
report_blocks, 50, clock_.TimeInMilliseconds());
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(500);
report_blocks.push_back(CreateReportBlock(1, 2, 0, 21));
bandwidth_observer_->OnReceivedRtcpReceiverReport(
report_blocks, 50, clock_.TimeInMilliseconds());
report_blocks.clear();
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(1500);
report_blocks.push_back(CreateReportBlock(1, 2, 0, 41));
bandwidth_observer_->OnReceivedRtcpReceiverReport(
report_blocks, 50, clock_.TimeInMilliseconds());
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(4000);
// 4 seconds since feedback, expect increase.
controller_->Process();
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(2000);
// 6 seconds since feedback, expect no increase.
controller_->Process();
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(9001);
// More than 15 seconds since feedback, expect decrease.
controller_->Process();
expected_bitrate_bps *= 0.8;
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(500);
// Only one timeout every second.
controller_->Process();
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(501);
// New timeout allowed.
controller_->Process();
expected_bitrate_bps *= 0.8;
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
}
}
TEST_F(BitrateControllerTest, StopIncreaseWithoutPacketReports) {
int expected_bitrate_bps = 300000;
controller_->SetBitrates(300000, kDefaultMinBitrateBps,
kDefaultMaxBitrateBps);
webrtc::ReportBlockList report_blocks;
report_blocks.push_back(CreateReportBlock(1, 2, 0, 1));
bandwidth_observer_->OnReceivedRtcpReceiverReport(
report_blocks, 50, clock_.TimeInMilliseconds());
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(500);
report_blocks.push_back(CreateReportBlock(1, 2, 0, 21));
bandwidth_observer_->OnReceivedRtcpReceiverReport(
report_blocks, 50, clock_.TimeInMilliseconds());
report_blocks.clear();
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(1500);
// 1.2 seconds without packets reported as received, no increase.
report_blocks.push_back(CreateReportBlock(1, 2, 0, 21));
bandwidth_observer_->OnReceivedRtcpReceiverReport(
report_blocks, 50, clock_.TimeInMilliseconds());
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(1000);
// 5 packets reported as received since last, too few, no increase.
report_blocks.push_back(CreateReportBlock(1, 2, 0, 26));
bandwidth_observer_->OnReceivedRtcpReceiverReport(
report_blocks, 50, clock_.TimeInMilliseconds());
report_blocks.clear();
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(100);
// 15 packets reported as received since last, enough to increase.
report_blocks.push_back(CreateReportBlock(1, 2, 0, 41));
bandwidth_observer_->OnReceivedRtcpReceiverReport(
report_blocks, 50, clock_.TimeInMilliseconds());
expected_bitrate_bps = expected_bitrate_bps * 1.08 + 1000;
EXPECT_EQ(expected_bitrate_bps, bitrate_observer_.last_bitrate_);
clock_.AdvanceTimeMilliseconds(1000);
}

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modules/bitrate_controller/include/bitrate_controller.h
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@ -1,102 +0,0 @@
/*
* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*
* Usage: this class will register multiple RtcpBitrateObserver's one at each
* RTCP module. It will aggregate the results and run one bandwidth estimation
* and push the result to the encoders via BitrateObserver(s).
*/
#ifndef MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
#define MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
#include <stddef.h>
#include <stdint.h>
#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
#include "modules/include/module.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "rtc_base/deprecation.h"
namespace webrtc {
class Clock;
class RtcEventLog;
// Deprecated
// TODO(perkj): Remove BitrateObserver when no implementations use it.
class BitrateObserver {
// Observer class for bitrate changes announced due to change in bandwidth
// estimate or due to bitrate allocation changes. Fraction loss and rtt is
// also part of this callback to allow the obsevrer to optimize its settings
// for different types of network environments. The bitrate does not include
// packet headers and is measured in bits per second.
public:
virtual void OnNetworkChanged(uint32_t bitrate_bps,
uint8_t fraction_loss, // 0 - 255.
int64_t rtt_ms) = 0;
// TODO(gnish): Merge these two into one function.
virtual void OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
uint32_t bitrate_for_pacer_bps,
bool in_probe_rtt,
int64_t target_set_time,
uint64_t congestion_window) {}
virtual void OnBytesAcked(size_t bytes) {}
virtual size_t pacer_queue_size_in_bytes();
virtual ~BitrateObserver() {}
};
class BitrateController : public Module, public RtcpBandwidthObserver {
// This class collects feedback from all streams sent to a peer (via
// RTCPBandwidthObservers). It does one aggregated send side bandwidth
// estimation and divide the available bitrate between all its registered
// BitrateObservers.
public:
static const int kDefaultStartBitratebps = 300000;
// Deprecated:
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
// Remove this method once other other projects does not use it.
static BitrateController* CreateBitrateController(Clock* clock,
BitrateObserver* observer,
RtcEventLog* event_log);
static BitrateController* CreateBitrateController(Clock* clock,
RtcEventLog* event_log);
~BitrateController() override {}
// Deprecated, use raw pointer to BitrateController instance instead.
// Creates RtcpBandwidthObserver caller responsible to delete.
RTC_DEPRECATED virtual RtcpBandwidthObserver*
CreateRtcpBandwidthObserver() = 0;
// Deprecated
virtual void SetStartBitrate(int start_bitrate_bps) = 0;
// Deprecated
virtual void SetMinMaxBitrate(int min_bitrate_bps, int max_bitrate_bps) = 0;
virtual void SetBitrates(int start_bitrate_bps,
int min_bitrate_bps,
int max_bitrate_bps) = 0;
virtual void ResetBitrates(int bitrate_bps,
int min_bitrate_bps,
int max_bitrate_bps) = 0;
virtual void OnDelayBasedBweResult(const DelayBasedBwe::Result& result) = 0;
// Gets the available payload bandwidth in bits per second. Note that
// this bandwidth excludes packet headers.
virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
virtual bool GetNetworkParameters(uint32_t* bitrate,
uint8_t* fraction_loss,
int64_t* rtt) = 0;
};
} // namespace webrtc
#endif // MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_