Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ )
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: https://chromium.googlesource.com/external/webrtc/+/37aa8ba61641962119071646175bfe3bc2bda063
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}
diff --git a/resources/voice_engine/audio_dtx16.wav.sha1 b/resources/voice_engine/audio_dtx16.wav.sha1
deleted file mode 100644
index 6a552c2..0000000
--- a/resources/voice_engine/audio_dtx16.wav.sha1
+++ /dev/null
@@ -1 +0,0 @@
-cafd7151d6b7b4313d0bd2a128a31fc83bca7aa3
\ No newline at end of file
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index 2821556..9280d54 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -496,7 +496,6 @@
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",
diff --git a/webrtc/audio/BUILD.gn b/webrtc/audio/BUILD.gn
index 414a7c6..1577316 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -113,8 +113,6 @@
"../test:fake_audio_device",
"../test:test_common",
"../test:test_main",
- "//testing/gmock",
- "//testing/gtest",
"//third_party/gflags",
]
if (is_android) {
@@ -124,7 +122,6 @@
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 @@
}
}
}
-
- 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" ]
- }
- }
}
diff --git a/webrtc/audio/test/audio_bwe_integration_test.cc b/webrtc/audio/test/audio_bwe_integration_test.cc
deleted file mode 100644
index e5c0c2f..0000000
--- a/webrtc/audio/test/audio_bwe_integration_test.cc
+++ /dev/null
@@ -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
diff --git a/webrtc/audio/test/audio_bwe_integration_test.h b/webrtc/audio/test/audio_bwe_integration_test.h
deleted file mode 100644
index 769603a..0000000
--- a/webrtc/audio/test/audio_bwe_integration_test.h
+++ /dev/null
@@ -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_
diff --git a/webrtc/modules/congestion_controller/BUILD.gn b/webrtc/modules/congestion_controller/BUILD.gn
index eade559..ef45297 100644
--- a/webrtc/modules/congestion_controller/BUILD.gn
+++ b/webrtc/modules/congestion_controller/BUILD.gn
@@ -10,10 +10,8 @@
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 @@
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 @@
":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",
diff --git a/webrtc/modules/congestion_controller/bitrate_estimator.cc b/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.cc
similarity index 75%
rename from webrtc/modules/congestion_controller/bitrate_estimator.cc
rename to webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.cc
index 831144e..3fa3ac9 100644
--- a/webrtc/modules/congestion_controller/bitrate_estimator.cc
+++ b/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.cc
@@ -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 {
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 @@
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 @@
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);
diff --git a/webrtc/modules/congestion_controller/bitrate_estimator.h b/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h
similarity index 67%
rename from webrtc/modules/congestion_controller/bitrate_estimator.h
rename to webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h
index 7d56208..7a9d669 100644
--- a/webrtc/modules/congestion_controller/bitrate_estimator.h
+++ b/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h
@@ -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 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
+#endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGE_BITRATE_ESTIMATOR_H_
diff --git a/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc
deleted file mode 100644
index bcccb43..0000000
--- a/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc
+++ /dev/null
@@ -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
diff --git a/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
deleted file mode 100644
index afed1dc..0000000
--- a/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
+++ /dev/null
@@ -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_
diff --git a/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator_unittest.cc b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator_unittest.cc
deleted file mode 100644
index 0c8522d..0000000
--- a/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator_unittest.cc
+++ /dev/null
@@ -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
diff --git a/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc b/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc
index 288886d..62c9f88 100644
--- a/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc
+++ b/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc
@@ -184,7 +184,7 @@
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 @@
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());
diff --git a/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h b/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h
index b7b7905..a12bee5 100644
--- a/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h
+++ b/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h
@@ -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"
diff --git a/webrtc/modules/congestion_controller/send_side_congestion_controller.cc b/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
index ce6b96b..d5f966a 100644
--- a/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
+++ b/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
@@ -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 @@
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 @@
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_);
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
index a6e857f..905659d 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
@@ -78,7 +78,7 @@
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)
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
index ed6a7cd..d2aa8de 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
@@ -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"