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Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )

Browse Source 
Reason for revert:
Resetting the estimate means that we need to start gathering data from scratch again. The combination of
1) DelayBasedEstimator not reacting to overuse unless there is a valid estimate of the acknowledged bitrate, and
2) AcknowledgedBitrateEstimator needing a significant amount of time/data to obtain an provide an estimate
causes poor performance in simulations/tests. It is not clear whether this will affect real networks negatively, but I suggest reverting this to be on the safe side.
See also https://bugs.chromium.org/p/webrtc/issues/detail?id=7884

Original issue's description:
> Test and fix for huge bwe drop after alr state.
>
> BUG=webrtc:7746
>
> Review-Url: https://codereview.webrtc.org/2931873002
> Cr-Commit-Position: refs/heads/master@{#18692}
> Committed: 37aa8ba616

TBR=solenberg@webrtc.org,kwiberg@webrtc.org,minyue@webrtc.org,holmer@chromium.org,philipel@webrtc.org,oprypin@webrtc.org,holmer@google.com,stefan@webrtc.org,tschumim@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7746

Review-Url: https://codereview.webrtc.org/2964213002
Cr-Commit-Position: refs/heads/master@{#18866}
This commit is contained in:
terelius
2017-06-30 08:11:44 -07:00
committed by Commit Bot
parent 2d54784d89
commit e75d96b5bd
16 changed files with 44 additions and 585 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
resources/voice_engine/audio_dtx16.wav.sha1
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@ -1 +0,0 @@
cafd7151d6b7b4313d0bd2a128a31fc83bca7aa3

1
webrtc/BUILD.gn
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@ -496,7 +496,6 @@ if (rtc_include_tests) {
configs += [ ":rtc_unittests_config" ]
deps = [
"audio:audio_perf_tests",
"call:call_perf_tests",
"modules/audio_coding:audio_coding_perf_tests",
"modules/audio_processing:audio_processing_perf_tests",

39
webrtc/audio/BUILD.gn
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@ -113,8 +113,6 @@ if (rtc_include_tests) {
"../test:fake_audio_device",
"../test:test_common",
"../test:test_main",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
if (is_android) {
@ -124,7 +122,6 @@ if (rtc_include_tests) {
data = [
"//resources/voice_engine/audio_tiny16.wav",
"//resources/voice_engine/audio_tiny48.wav",
"//resources/voice_engine/audio_dtx16.wav",
]
if (!build_with_chromium && is_clang) {
@ -133,40 +130,4 @@ if (rtc_include_tests) {
}
}
}
rtc_source_set("audio_perf_tests") {
testonly = true
# Skip restricting visibility on mobile platforms since the tests on those
# gets additional generated targets which would require many lines here to
# cover (which would be confusing to read and hard to maintain).
if (!is_android && !is_ios) {
visibility = [ "//webrtc:webrtc_perf_tests" ]
}
sources = [
"test/audio_bwe_integration_test.cc",
"test/audio_bwe_integration_test.h",
]
deps = [
"../base:rtc_base_approved",
"../common_audio",
"../system_wrappers",
"../test:fake_audio_device",
"../test:field_trial",
"../test:test_common",
"../test:test_main",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
data = [
"//resources/voice_engine/audio_dtx16.wav",
]
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" ]
}
}
}

146
webrtc/audio/test/audio_bwe_integration_test.cc
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@ -1,146 +0,0 @@
/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/audio/test/audio_bwe_integration_test.h"
#include "webrtc/base/ptr_util.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
namespace webrtc {
namespace test {
AudioBweTest::AudioBweTest() : EndToEndTest(CallTest::kDefaultTimeoutMs) {}
size_t AudioBweTest::GetNumVideoStreams() const {
return 0;
}
size_t AudioBweTest::GetNumAudioStreams() const {
return 1;
}
size_t AudioBweTest::GetNumFlexfecStreams() const {
return 0;
}
std::unique_ptr<test::FakeAudioDevice::Capturer>
AudioBweTest::CreateCapturer() {
return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
}
void AudioBweTest::OnFakeAudioDevicesCreated(
test::FakeAudioDevice* send_audio_device,
test::FakeAudioDevice* recv_audio_device) {
send_audio_device_ = send_audio_device;
}
test::PacketTransport* AudioBweTest::CreateSendTransport(Call* sender_call) {
return new test::PacketTransport(
sender_call, this, test::PacketTransport::kSender,
test::CallTest::payload_type_map_, GetNetworkPipeConfig());
}
test::PacketTransport* AudioBweTest::CreateReceiveTransport() {
return new test::PacketTransport(
nullptr, this, test::PacketTransport::kReceiver,
test::CallTest::payload_type_map_, GetNetworkPipeConfig());
}
void AudioBweTest::PerformTest() {
send_audio_device_->WaitForRecordingEnd();
SleepMs(GetNetworkPipeConfig().queue_delay_ms);
}
class StatsPollTask : public rtc::QueuedTask {
public:
explicit StatsPollTask(Call* sender_call) : sender_call_(sender_call) {}
private:
bool Run() override {
RTC_CHECK(sender_call_);
Call::Stats call_stats = sender_call_->GetStats();
EXPECT_GT(call_stats.send_bandwidth_bps, 30000);
rtc::TaskQueue::Current()->PostDelayedTask(
std::unique_ptr<QueuedTask>(this), 100);
return false;
}
Call* sender_call_;
};
class NoBandwidthDropAfterDtx : public AudioBweTest {
public:
NoBandwidthDropAfterDtx()
: sender_call_(nullptr), stats_poller_("stats poller task queue") {}
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->send_codec_spec =
rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
{test::CallTest::kAudioSendPayloadType,
{"OPUS",
48000,
2,
{{"ptime", "60"}, {"usedtx", "1"}, {"stereo", "1"}}}});
send_config->min_bitrate_bps = 6000;
send_config->max_bitrate_bps = 100000;
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri,
kTransportSequenceNumberExtensionId));
for (AudioReceiveStream::Config& recv_config : *receive_configs) {
recv_config.rtp.transport_cc = true;
recv_config.rtp.extensions = send_config->rtp.extensions;
recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
}
}
std::string AudioInputFile() override {
return test::ResourcePath("voice_engine/audio_dtx16", "wav");
}
FakeNetworkPipe::Config GetNetworkPipeConfig() override {
FakeNetworkPipe::Config pipe_config;
pipe_config.link_capacity_kbps = 50;
pipe_config.queue_length_packets = 1500;
pipe_config.queue_delay_ms = 300;
return pipe_config;
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
}
void PerformTest() override {
stats_poller_.PostDelayedTask(
std::unique_ptr<rtc::QueuedTask>(new StatsPollTask(sender_call_)), 100);
sender_call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO, 0);
AudioBweTest::PerformTest();
}
private:
Call* sender_call_;
rtc::TaskQueue stats_poller_;
};
using AudioBweIntegrationTest = CallTest;
TEST_F(AudioBweIntegrationTest, DISABLED_NoBandwidthDropAfterDtx) {
webrtc::test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-SendSideBwe/Enabled/"
"WebRTC-SendSideBwe-WithOverhead/Enabled/");
NoBandwidthDropAfterDtx test;
RunBaseTest(&test);
}
} // namespace test
} // namespace webrtc

53
webrtc/audio/test/audio_bwe_integration_test.h
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@ -1,53 +0,0 @@
/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
#define WEBRTC_AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_
#include <memory>
#include <string>
#include "webrtc/test/call_test.h"
#include "webrtc/test/fake_audio_device.h"
namespace webrtc {
namespace test {
class AudioBweTest : public test::EndToEndTest {
public:
AudioBweTest();
protected:
virtual std::string AudioInputFile() = 0;
virtual FakeNetworkPipe::Config GetNetworkPipeConfig() = 0;
size_t GetNumVideoStreams() const override;
size_t GetNumAudioStreams() const override;
size_t GetNumFlexfecStreams() const override;
std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override;
void OnFakeAudioDevicesCreated(
test::FakeAudioDevice* send_audio_device,
test::FakeAudioDevice* recv_audio_device) override;
test::PacketTransport* CreateSendTransport(Call* sender_call) override;
test::PacketTransport* CreateReceiveTransport() override;
void PerformTest() override;
private:
test::FakeAudioDevice* send_audio_device_;
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_AUDIO_TEST_AUDIO_BWE_INTEGRATION_TEST_H_

8
webrtc/modules/congestion_controller/BUILD.gn
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@ -10,10 +10,8 @@ import("../../webrtc.gni")
rtc_static_library("congestion_controller") {
sources = [
"acknowledged_bitrate_estimator.cc",
"acknowledged_bitrate_estimator.h",
"bitrate_estimator.cc",
"bitrate_estimator.h",
"acknowledge_bitrate_estimator.cc",
"acknowledge_bitrate_estimator.h",
"congestion_controller.cc",
"delay_based_bwe.cc",
"delay_based_bwe.h",
@ -75,7 +73,6 @@ if (rtc_include_tests) {
visibility = [ "//webrtc/modules:modules_unittests" ]
}
sources = [
"acknowledged_bitrate_estimator_unittest.cc",
"congestion_controller_unittest.cc",
"congestion_controller_unittests_helper.cc",
"congestion_controller_unittests_helper.h",
@ -93,7 +90,6 @@ if (rtc_include_tests) {
":mock_congestion_controller",
"../../base:rtc_base",
"../../base:rtc_base_approved",
"../../base:rtc_base_tests_utils",
"../../system_wrappers:system_wrappers",
"../../test:field_trial",
"../../test:test_support",

30
webrtc/modules/congestion_controller/bitrate_estimator.cc → webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.cc
View File

@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/congestion_controller/bitrate_estimator.h"
#include "webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h"
#include <cmath>
@ -20,18 +20,32 @@ namespace webrtc {
namespace {
constexpr int kInitialRateWindowMs = 500;
constexpr int kRateWindowMs = 150;
bool IsInSendTimeHistory(const PacketFeedback& packet) {
return packet.send_time_ms >= 0;
}
} // namespace
BitrateEstimator::BitrateEstimator()
AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator()
: sum_(0),
current_win_ms_(0),
prev_time_ms_(-1),
bitrate_estimate_(-1.0f),
bitrate_estimate_var_(50.0f) {}
BitrateEstimator::~BitrateEstimator() = default;
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))
Update(packet.arrival_time_ms, packet.payload_size);
}
}
void BitrateEstimator::Update(int64_t now_ms, int bytes) {
void AcknowledgedBitrateEstimator::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.
@ -64,9 +78,9 @@ void BitrateEstimator::Update(int64_t now_ms, int bytes) {
bitrate_estimate_ * 1000);
}
float BitrateEstimator::UpdateWindow(int64_t now_ms,
int bytes,
int rate_window_ms) {
float AcknowledgedBitrateEstimator::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;
@ -92,7 +106,7 @@ float BitrateEstimator::UpdateWindow(int64_t now_ms,
return bitrate_sample;
}
rtc::Optional<uint32_t> BitrateEstimator::bitrate_bps() const {
rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
if (bitrate_estimate_ < 0.f)
return rtc::Optional<uint32_t>();
return rtc::Optional<uint32_t>(bitrate_estimate_ * 1000);

20
webrtc/modules/congestion_controller/bitrate_estimator.h → webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h
View File

@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
#define WEBRTC_MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
#ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGE_BITRATE_ESTIMATOR_H_
#define WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGE_BITRATE_ESTIMATOR_H_
#include <vector>
@ -17,21 +17,25 @@
namespace webrtc {
struct PacketFeedback;
// 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 {
class AcknowledgedBitrateEstimator {
public:
BitrateEstimator();
virtual ~BitrateEstimator();
virtual void Update(int64_t now_ms, int bytes);
AcknowledgedBitrateEstimator();
virtual rtc::Optional<uint32_t> bitrate_bps() const;
void IncomingPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector);
rtc::Optional<uint32_t> bitrate_bps() const;
private:
void Update(int64_t now_ms, int bytes);
float UpdateWindow(int64_t now_ms, int bytes, int rate_window_ms);
int sum_;
int64_t current_win_ms_;
int64_t prev_time_ms_;
@ -41,4 +45,4 @@ class BitrateEstimator {
} // namespace webrtc
#endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
#endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGE_BITRATE_ESTIMATOR_H_

81
webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc
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@ -1,81 +0,0 @@
/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h"
#include <utility>
#include "webrtc/base/ptr_util.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
namespace webrtc {
namespace {
bool IsInSendTimeHistory(const PacketFeedback& packet) {
return packet.send_time_ms >= 0;
}
} // namespace
std::unique_ptr<BitrateEstimator> BitrateEstimatorCreator::Create() {
return rtc::MakeUnique<BitrateEstimator>();
}
AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator()
: AcknowledgedBitrateEstimator(rtc::MakeUnique<BitrateEstimatorCreator>()) {
}
AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
std::unique_ptr<BitrateEstimatorCreator> bitrate_estimator_creator)
: was_in_alr_(false),
bitrate_estimator_creator_(std::move(bitrate_estimator_creator)),
bitrate_estimator_(bitrate_estimator_creator_->Create()) {}
void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector,
bool currently_in_alr) {
RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
packet_feedback_vector.end(),
PacketFeedbackComparator()));
MaybeResetBitrateEstimator(currently_in_alr);
for (const auto& packet : packet_feedback_vector) {
if (IsInSendTimeHistory(packet) && !SentBeforeAlrEnded(packet))
bitrate_estimator_->Update(packet.arrival_time_ms, packet.payload_size);
}
}
rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
return bitrate_estimator_->bitrate_bps();
}
bool AcknowledgedBitrateEstimator::SentBeforeAlrEnded(
const PacketFeedback& packet) {
if (alr_ended_time_ms_) {
if (*alr_ended_time_ms_ > packet.send_time_ms) {
return true;
}
}
return false;
}
bool AcknowledgedBitrateEstimator::AlrEnded(bool currently_in_alr) const {
return was_in_alr_ && !currently_in_alr;
}
void AcknowledgedBitrateEstimator::MaybeResetBitrateEstimator(
bool currently_in_alr) {
if (AlrEnded(currently_in_alr)) {
bitrate_estimator_ = bitrate_estimator_creator_->Create();
alr_ended_time_ms_ = rtc::Optional<int64_t>(rtc::TimeMillis());
}
was_in_alr_ = currently_in_alr;
}
} // namespace webrtc

55
webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
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@ -1,55 +0,0 @@
/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
#define WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
#include <memory>
#include <vector>
#include "webrtc/base/optional.h"
#include "webrtc/modules/congestion_controller/bitrate_estimator.h"
namespace webrtc {
struct PacketFeedback;
class BitrateEstimatorCreator {
public:
virtual ~BitrateEstimatorCreator() = default;
virtual std::unique_ptr<BitrateEstimator> Create();
};
class AcknowledgedBitrateEstimator {
public:
explicit AcknowledgedBitrateEstimator(
std::unique_ptr<BitrateEstimatorCreator> bitrate_estimator_creator);
AcknowledgedBitrateEstimator();
void IncomingPacketFeedbackVector(
const std::vector<PacketFeedback>& packet_feedback_vector,
bool currently_in_alr);
rtc::Optional<uint32_t> bitrate_bps() const;
private:
bool SentBeforeAlrEnded(const PacketFeedback& packet);
bool AlrEnded(bool currently_in_alr) const;
void MaybeResetBitrateEstimator(bool currently_in_alr);
bool was_in_alr_;
rtc::Optional<int64_t> alr_ended_time_ms_;
const std::unique_ptr<BitrateEstimatorCreator> bitrate_estimator_creator_;
std::unique_ptr<BitrateEstimator> bitrate_estimator_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_

177
webrtc/modules/congestion_controller/acknowledged_bitrate_estimator_unittest.cc
View File

@ -1,177 +0,0 @@
/*
* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h"
#include <utility>
#include "webrtc/base/fakeclock.h"
#include "webrtc/base/ptr_util.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
using testing::_;
using testing::NiceMock;
using testing::InSequence;
using testing::Return;
namespace webrtc {
namespace {
constexpr int64_t first_arrival_time_ms = 10;
constexpr int64_t first_send_time_ms = 10;
constexpr uint16_t sequence_number = 1;
constexpr size_t payload_size = 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>());
};
class MockBitrateEstimatorCreator : public BitrateEstimatorCreator {
public:
MockBitrateEstimatorCreator()
: mock_bitrate_estimator_(nullptr), num_created_bitrate_estimators_(0) {}
std::unique_ptr<BitrateEstimator> Create() override {
auto bitrate_estimator = rtc::MakeUnique<NiceMock<MockBitrateEstimator>>();
mock_bitrate_estimator_ = bitrate_estimator.get();
num_created_bitrate_estimators_++;
return bitrate_estimator;
}
int num_created_bitrate_estimators() {
return num_created_bitrate_estimators_;
}
MockBitrateEstimator* get_mock_bitrate_estimator() {
RTC_CHECK(mock_bitrate_estimator_);
return mock_bitrate_estimator_;
}
private:
MockBitrateEstimator* mock_bitrate_estimator_;
int num_created_bitrate_estimators_;
};
struct AcknowledgedBitrateEstimatorTestStates {
std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator;
MockBitrateEstimatorCreator* mock_bitrate_estimator_creator;
};
AcknowledgedBitrateEstimatorTestStates CreateTestStates() {
AcknowledgedBitrateEstimatorTestStates states;
auto mock_bitrate_estimator_creator =
rtc::MakeUnique<MockBitrateEstimatorCreator>();
states.mock_bitrate_estimator_creator = mock_bitrate_estimator_creator.get();
states.acknowledged_bitrate_estimator =
rtc::MakeUnique<AcknowledgedBitrateEstimator>(
std::move(mock_bitrate_estimator_creator));
return states;
}
std::vector<PacketFeedback> CreateFeedbackVector() {
std::vector<PacketFeedback> packet_feedback_vector;
const PacedPacketInfo pacing_info;
packet_feedback_vector.push_back(
PacketFeedback(first_arrival_time_ms, first_send_time_ms, sequence_number,
payload_size, pacing_info));
packet_feedback_vector.push_back(
PacketFeedback(first_arrival_time_ms + 10, first_send_time_ms + 10,
sequence_number, payload_size + 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(first_arrival_time_ms, sequence_number));
EXPECT_CALL(
*states.mock_bitrate_estimator_creator->get_mock_bitrate_estimator(),
Update(_, _))
.Times(0);
states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
packet_feedback_vector, false);
}
TEST(TestAcknowledgedBitrateEstimator, UpdateBandwith) {
auto states = CreateTestStates();
auto packet_feedback_vector = CreateFeedbackVector();
{
InSequence dummy;
EXPECT_CALL(
*states.mock_bitrate_estimator_creator->get_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_creator->get_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, false);
}
TEST(TestAcknowledgedBitrateEstimator,
ResetAfterLeafAlrStateAndDontAddOldPackets) {
auto states = CreateTestStates();
auto packet_feedback_vector = CreateFeedbackVector();
states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
packet_feedback_vector, true);
rtc::ScopedFakeClock fake_clock;
fake_clock.AdvanceTime(
rtc::TimeDelta::FromMilliseconds(first_arrival_time_ms + 1));
EXPECT_EQ(
1,
states.mock_bitrate_estimator_creator->num_created_bitrate_estimators());
states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
std::vector<PacketFeedback>(), false);
EXPECT_EQ(
2,
states.mock_bitrate_estimator_creator->num_created_bitrate_estimators());
{
InSequence dummy;
EXPECT_CALL(
*states.mock_bitrate_estimator_creator->get_mock_bitrate_estimator(),
Update(packet_feedback_vector[0].arrival_time_ms,
static_cast<int>(packet_feedback_vector[0].payload_size)))
.Times(0);
EXPECT_CALL(
*states.mock_bitrate_estimator_creator->get_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, false);
}
TEST(TestAcknowledgedBitrateEstimator, ReturnBitrate) {
auto states = CreateTestStates();
rtc::Optional<uint32_t> return_value(42);
EXPECT_CALL(
*states.mock_bitrate_estimator_creator->get_mock_bitrate_estimator(),
bitrate_bps())
.Times(1)
.WillOnce(Return(return_value));
EXPECT_EQ(return_value, states.acknowledged_bitrate_estimator->bitrate_bps());
}
} // namespace webrtc

4
webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc
View File

@ -184,7 +184,7 @@ void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
sequence_number, payload_size, pacing_info);
std::vector<PacketFeedback> packets;
packets.push_back(packet);
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets, false);
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
DelayBasedBwe::Result result =
bitrate_estimator_->IncomingPacketFeedbackVector(
packets, acknowledged_bitrate_estimator_->bitrate_bps());
@ -219,7 +219,7 @@ bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
packet.arrival_time_ms += arrival_time_offset_ms_;
}
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets, false);
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
DelayBasedBwe::Result result =
bitrate_estimator_->IncomingPacketFeedbackVector(
packets, acknowledged_bitrate_estimator_->bitrate_bps());

2
webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h
View File

@ -18,7 +18,7 @@
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h"
#include "webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h"
#include "webrtc/modules/congestion_controller/delay_based_bwe.h"
#include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
#include "webrtc/system_wrappers/include/clock.h"

8
webrtc/modules/congestion_controller/send_side_congestion_controller.cc
View File

@ -20,7 +20,7 @@
#include "webrtc/base/rate_limiter.h"
#include "webrtc/base/socket.h"
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h"
#include "webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h"
#include "webrtc/modules/congestion_controller/probe_controller.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
@ -89,8 +89,7 @@ SendSideCongestionController::SendSideCongestionController(
bitrate_controller_(
BitrateController::CreateBitrateController(clock_, event_log)),
acknowledged_bitrate_estimator_(
rtc::MakeUnique<AcknowledgedBitrateEstimator>(
rtc::MakeUnique<BitrateEstimatorCreator>())),
rtc::MakeUnique<AcknowledgedBitrateEstimator>()),
probe_controller_(new ProbeController(pacer_.get(), clock_)),
retransmission_rate_limiter_(
new RateLimiter(clock, kRetransmitWindowSizeMs)),
@ -279,8 +278,7 @@ void SendSideCongestionController::OnTransportFeedback(
transport_feedback_adapter_.GetTransportFeedbackVector());
SortPacketFeedbackVector(&feedback_vector);
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
feedback_vector,
static_cast<bool>(pacer_->GetApplicationLimitedRegionStartTime()));
feedback_vector);
DelayBasedBwe::Result result;
{
rtc::CritScope cs(&bwe_lock_);

2
webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
View File

@ -78,7 +78,7 @@ void SendSideBweSender::GiveFeedback(const FeedbackPacket& feedback) {
std::sort(packet_feedback_vector.begin(), packet_feedback_vector.end(),
PacketFeedbackComparator());
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
packet_feedback_vector, false);
packet_feedback_vector);
DelayBasedBwe::Result result = bwe_->IncomingPacketFeedbackVector(
packet_feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps());
if (result.updated)

2
webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
View File

@ -15,7 +15,7 @@
#include <vector>
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h"
#include "webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"