Reimplemeted "Test and fix for huge bwe drop after alr state"
BUG=webrtc:7746
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
patch from issue 2931873002 at patchset 320001 (http://crrev.com/2931873002#ps320001)
Review-Url: https://codereview.webrtc.org/2970653004
Cr-Commit-Position: refs/heads/master@{#19055}
diff --git a/resources/voice_engine/audio_dtx16.wav.sha1 b/resources/voice_engine/audio_dtx16.wav.sha1
new file mode 100644
index 0000000..6a552c2
--- /dev/null
+++ b/resources/voice_engine/audio_dtx16.wav.sha1
@@ -0,0 +1 @@
+cafd7151d6b7b4313d0bd2a128a31fc83bca7aa3
\ No newline at end of file
diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn
index e057ebe..a4cc0ee 100644
--- a/webrtc/BUILD.gn
+++ b/webrtc/BUILD.gn
@@ -496,6 +496,7 @@
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 5296f1f..4fc7337 100644
--- a/webrtc/audio/BUILD.gn
+++ b/webrtc/audio/BUILD.gn
@@ -113,6 +113,8 @@
"../test:fake_audio_device",
"../test:test_common",
"../test:test_main",
+ "//testing/gmock",
+ "//testing/gtest",
"//third_party/gflags",
]
if (is_android) {
@@ -122,6 +124,7 @@
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) {
@@ -130,4 +133,40 @@
}
}
}
+
+ 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
new file mode 100644
index 0000000..d3f7f0b
--- /dev/null
+++ b/webrtc/audio/test/audio_bwe_integration_test.cc
@@ -0,0 +1,151 @@
+/*
+ * 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/common_audio/wav_file.h"
+#include "webrtc/rtc_base/ptr_util.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 {
+
+namespace {
+// Wait a second between stopping sending and stopping receiving audio.
+constexpr int kExtraProcessTimeMs = 1000;
+} // namespace
+
+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 + kExtraProcessTimeMs);
+}
+
+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, 25000);
+ 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, 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
new file mode 100644
index 0000000..769603a
--- /dev/null
+++ b/webrtc/audio/test/audio_bwe_integration_test.h
@@ -0,0 +1,53 @@
+/*
+ * 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 dbfd56b..c3ee11f 100644
--- a/webrtc/modules/congestion_controller/BUILD.gn
+++ b/webrtc/modules/congestion_controller/BUILD.gn
@@ -10,8 +10,10 @@
rtc_static_library("congestion_controller") {
sources = [
- "acknowledge_bitrate_estimator.cc",
- "acknowledge_bitrate_estimator.h",
+ "acknowledged_bitrate_estimator.cc",
+ "acknowledged_bitrate_estimator.h",
+ "bitrate_estimator.cc",
+ "bitrate_estimator.h",
"congestion_controller.cc",
"delay_based_bwe.cc",
"delay_based_bwe.h",
@@ -73,6 +75,7 @@
visibility = [ "..:modules_unittests" ]
}
sources = [
+ "acknowledged_bitrate_estimator_unittest.cc",
"congestion_controller_unittest.cc",
"congestion_controller_unittests_helper.cc",
"congestion_controller_unittests_helper.h",
@@ -90,6 +93,7 @@
":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/acknowledged_bitrate_estimator.cc b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc
new file mode 100644
index 0000000..b150734
--- /dev/null
+++ b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc
@@ -0,0 +1,63 @@
+/*
+ * 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/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/rtc_base/ptr_util.h"
+
+namespace webrtc {
+
+namespace {
+bool IsInSendTimeHistory(const PacketFeedback& packet) {
+ return packet.send_time_ms >= 0;
+}
+} // namespace
+
+AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator()
+ : AcknowledgedBitrateEstimator(rtc::MakeUnique<BitrateEstimator>()) {}
+
+AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
+ std::unique_ptr<BitrateEstimator> bitrate_estimator)
+ : bitrate_estimator_(std::move(bitrate_estimator)) {}
+
+void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
+ const std::vector<PacketFeedback>& packet_feedback_vector) {
+ RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
+ packet_feedback_vector.end(),
+ PacketFeedbackComparator()));
+ for (const auto& packet : packet_feedback_vector) {
+ if (IsInSendTimeHistory(packet)) {
+ MaybeExpectFastRateChange(packet.send_time_ms);
+ bitrate_estimator_->Update(packet.arrival_time_ms, packet.payload_size);
+ }
+ }
+}
+
+rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
+ return bitrate_estimator_->bitrate_bps();
+}
+
+void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
+ int64_t alr_ended_time_ms) {
+ alr_ended_time_ms_.emplace(alr_ended_time_ms);
+}
+
+void AcknowledgedBitrateEstimator::MaybeExpectFastRateChange(
+ int64_t packet_send_time_ms) {
+ if (alr_ended_time_ms_ && packet_send_time_ms > *alr_ended_time_ms_) {
+ bitrate_estimator_->ExpectFastRateChange();
+ alr_ended_time_ms_.reset();
+ }
+}
+
+} // namespace webrtc
diff --git a/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
new file mode 100644
index 0000000..bab5d93
--- /dev/null
+++ b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
@@ -0,0 +1,44 @@
+/*
+ * 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/modules/congestion_controller/bitrate_estimator.h"
+#include "webrtc/rtc_base/optional.h"
+
+namespace webrtc {
+
+struct PacketFeedback;
+
+class AcknowledgedBitrateEstimator {
+ public:
+ explicit AcknowledgedBitrateEstimator(
+ std::unique_ptr<BitrateEstimator> bitrate_estimator);
+
+ AcknowledgedBitrateEstimator();
+
+ void IncomingPacketFeedbackVector(
+ const std::vector<PacketFeedback>& packet_feedback_vector);
+ rtc::Optional<uint32_t> bitrate_bps() const;
+ void SetAlrEndedTimeMs(int64_t alr_ended_time_ms);
+
+ private:
+ void MaybeExpectFastRateChange(int64_t packet_arrival_time_ms);
+ rtc::Optional<int64_t> alr_ended_time_ms_;
+ std::unique_ptr<BitrateEstimator> bitrate_estimator_;
+};
+
+} // namespace webrtc
+
+#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
new file mode 100644
index 0000000..70ce0c1
--- /dev/null
+++ b/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator_unittest.cc
@@ -0,0 +1,134 @@
+/*
+ * 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/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/rtc_base/fakeclock.h"
+#include "webrtc/rtc_base/ptr_util.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 kFirstArrivalTimeMs = 10;
+constexpr int64_t kFirstSendTimeMs = 10;
+constexpr uint16_t kSequenceNumber = 1;
+constexpr size_t kPayloadSize = 10;
+
+class MockBitrateEstimator : public BitrateEstimator {
+ public:
+ MOCK_METHOD2(Update, void(int64_t now_ms, int bytes));
+ MOCK_CONST_METHOD0(bitrate_bps, rtc::Optional<uint32_t>());
+ MOCK_METHOD0(ExpectFastRateChange, void());
+};
+
+struct AcknowledgedBitrateEstimatorTestStates {
+ std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator;
+ MockBitrateEstimator* mock_bitrate_estimator;
+};
+
+AcknowledgedBitrateEstimatorTestStates CreateTestStates() {
+ AcknowledgedBitrateEstimatorTestStates states;
+ auto mock_bitrate_estimator = rtc::MakeUnique<MockBitrateEstimator>();
+ states.mock_bitrate_estimator = mock_bitrate_estimator.get();
+ states.acknowledged_bitrate_estimator =
+ rtc::MakeUnique<AcknowledgedBitrateEstimator>(
+ std::move(mock_bitrate_estimator));
+ return states;
+}
+
+std::vector<PacketFeedback> CreateFeedbackVector() {
+ std::vector<PacketFeedback> packet_feedback_vector;
+ const PacedPacketInfo pacing_info;
+ packet_feedback_vector.push_back(
+ PacketFeedback(kFirstArrivalTimeMs, kFirstSendTimeMs, kSequenceNumber,
+ kPayloadSize, pacing_info));
+ packet_feedback_vector.push_back(
+ PacketFeedback(kFirstArrivalTimeMs + 10, kFirstSendTimeMs + 10,
+ kSequenceNumber, kPayloadSize + 10, pacing_info));
+ return packet_feedback_vector;
+}
+
+} // anonymous namespace
+
+TEST(TestAcknowledgedBitrateEstimator, DontAddPacketsWhichAreNotInSendHistory) {
+ auto states = CreateTestStates();
+ std::vector<PacketFeedback> packet_feedback_vector;
+ packet_feedback_vector.push_back(
+ PacketFeedback(kFirstArrivalTimeMs, kSequenceNumber));
+ EXPECT_CALL(*states.mock_bitrate_estimator, Update(_, _)).Times(0);
+ states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+ packet_feedback_vector);
+}
+
+TEST(TestAcknowledgedBitrateEstimator, UpdateBandwith) {
+ auto states = CreateTestStates();
+ auto packet_feedback_vector = CreateFeedbackVector();
+ {
+ InSequence dummy;
+ EXPECT_CALL(
+ *states.mock_bitrate_estimator,
+ Update(packet_feedback_vector[0].arrival_time_ms,
+ static_cast<int>(packet_feedback_vector[0].payload_size)))
+ .Times(1);
+ EXPECT_CALL(
+ *states.mock_bitrate_estimator,
+ Update(packet_feedback_vector[1].arrival_time_ms,
+ static_cast<int>(packet_feedback_vector[1].payload_size)))
+ .Times(1);
+ }
+ states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+ packet_feedback_vector);
+}
+
+TEST(TestAcknowledgedBitrateEstimator, ExpectFastRateChangeWhenLeftAlr) {
+ auto states = CreateTestStates();
+ auto packet_feedback_vector = CreateFeedbackVector();
+ {
+ InSequence dummy;
+ EXPECT_CALL(
+ *states.mock_bitrate_estimator,
+ Update(packet_feedback_vector[0].arrival_time_ms,
+ static_cast<int>(packet_feedback_vector[0].payload_size)))
+ .Times(1);
+ EXPECT_CALL(*states.mock_bitrate_estimator, ExpectFastRateChange())
+ .Times(1);
+ EXPECT_CALL(
+ *states.mock_bitrate_estimator,
+ Update(packet_feedback_vector[1].arrival_time_ms,
+ static_cast<int>(packet_feedback_vector[1].payload_size)))
+ .Times(1);
+ }
+ states.acknowledged_bitrate_estimator->SetAlrEndedTimeMs(kFirstArrivalTimeMs +
+ 1);
+ states.acknowledged_bitrate_estimator->IncomingPacketFeedbackVector(
+ packet_feedback_vector);
+}
+
+TEST(TestAcknowledgedBitrateEstimator, ReturnBitrate) {
+ auto states = CreateTestStates();
+ rtc::Optional<uint32_t> return_value(42);
+ EXPECT_CALL(*states.mock_bitrate_estimator, bitrate_bps())
+ .Times(1)
+ .WillOnce(Return(return_value));
+ EXPECT_EQ(return_value, states.acknowledged_bitrate_estimator->bitrate_bps());
+}
+
+} // namespace webrtc*/
diff --git a/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.cc b/webrtc/modules/congestion_controller/bitrate_estimator.cc
similarity index 75%
rename from webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.cc
rename to webrtc/modules/congestion_controller/bitrate_estimator.cc
index 3fa3ac9..1627bf8 100644
--- a/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.cc
+++ b/webrtc/modules/congestion_controller/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/acknowledge_bitrate_estimator.h"
+#include "webrtc/modules/congestion_controller/bitrate_estimator.h"
#include <cmath>
@@ -20,32 +20,18 @@
namespace {
constexpr int kInitialRateWindowMs = 500;
constexpr int kRateWindowMs = 150;
-
-bool IsInSendTimeHistory(const PacketFeedback& packet) {
- return packet.send_time_ms >= 0;
-}
-
} // namespace
-AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator()
+BitrateEstimator::BitrateEstimator()
: sum_(0),
current_win_ms_(0),
prev_time_ms_(-1),
bitrate_estimate_(-1.0f),
bitrate_estimate_var_(50.0f) {}
-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);
- }
-}
+BitrateEstimator::~BitrateEstimator() = default;
-void AcknowledgedBitrateEstimator::Update(int64_t now_ms, int bytes) {
+void BitrateEstimator::Update(int64_t now_ms, int bytes) {
int rate_window_ms = kRateWindowMs;
// We use a larger window at the beginning to get a more stable sample that
// we can use to initialize the estimate.
@@ -78,9 +64,9 @@
bitrate_estimate_ * 1000);
}
-float AcknowledgedBitrateEstimator::UpdateWindow(int64_t now_ms,
- int bytes,
- int rate_window_ms) {
+float BitrateEstimator::UpdateWindow(int64_t now_ms,
+ int bytes,
+ int rate_window_ms) {
// Reset if time moves backwards.
if (now_ms < prev_time_ms_) {
prev_time_ms_ = -1;
@@ -106,10 +92,16 @@
return bitrate_sample;
}
-rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
+rtc::Optional<uint32_t> BitrateEstimator::bitrate_bps() const {
if (bitrate_estimate_ < 0.f)
return rtc::Optional<uint32_t>();
return rtc::Optional<uint32_t>(bitrate_estimate_ * 1000);
}
+void BitrateEstimator::ExpectFastRateChange() {
+ // By setting the bitrate-estimate variance to a higher value we allow the
+ // bitrate to change fast for the next few samples.
+ bitrate_estimate_var_ += 200;
+}
+
} // namespace webrtc
diff --git a/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h b/webrtc/modules/congestion_controller/bitrate_estimator.h
similarity index 67%
rename from webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h
rename to webrtc/modules/congestion_controller/bitrate_estimator.h
index 0bc2dac..f27d063 100644
--- a/webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h
+++ b/webrtc/modules/congestion_controller/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_ACKNOWLEDGE_BITRATE_ESTIMATOR_H_
-#define WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGE_BITRATE_ESTIMATOR_H_
+#ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
+#define WEBRTC_MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
#include <vector>
@@ -17,25 +17,23 @@
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 AcknowledgedBitrateEstimator {
+class BitrateEstimator {
public:
- AcknowledgedBitrateEstimator();
+ BitrateEstimator();
+ virtual ~BitrateEstimator();
+ virtual void Update(int64_t now_ms, int bytes);
- void IncomingPacketFeedbackVector(
- const std::vector<PacketFeedback>& packet_feedback_vector);
- rtc::Optional<uint32_t> bitrate_bps() const;
+ virtual rtc::Optional<uint32_t> bitrate_bps() const;
+
+ virtual void ExpectFastRateChange();
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_;
@@ -45,4 +43,4 @@
} // namespace webrtc
-#endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGE_BITRATE_ESTIMATOR_H_
+#endif // WEBRTC_MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
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 4e9c21c..2210a2e 100644
--- a/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h
+++ b/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h
@@ -17,7 +17,7 @@
#include <utility>
#include <vector>
-#include "webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h"
+#include "webrtc/modules/congestion_controller/acknowledged_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/rtc_base/constructormagic.h"
diff --git a/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h b/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
index 7a8f362..6bebf23 100644
--- a/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
+++ b/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
@@ -154,6 +154,7 @@
rtc::CriticalSection bwe_lock_;
int min_bitrate_bps_ GUARDED_BY(bwe_lock_);
std::unique_ptr<DelayBasedBwe> delay_based_bwe_ GUARDED_BY(bwe_lock_);
+ bool was_in_alr_;
rtc::RaceChecker worker_race_;
diff --git a/webrtc/modules/congestion_controller/send_side_congestion_controller.cc b/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
index 60e692c..5328161 100644
--- a/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
+++ b/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
@@ -15,14 +15,16 @@
#include <vector>
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
-#include "webrtc/modules/congestion_controller/acknowledge_bitrate_estimator.h"
+#include "webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h"
#include "webrtc/modules/congestion_controller/probe_controller.h"
+#include "webrtc/modules/pacing/alr_detector.h"
#include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/logging.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/rate_limiter.h"
#include "webrtc/rtc_base/socket.h"
+#include "webrtc/rtc_base/timeutils.h"
namespace webrtc {
namespace {
@@ -99,7 +101,8 @@
last_reported_rtt_(0),
network_state_(kNetworkUp),
min_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
- delay_based_bwe_(new DelayBasedBwe(event_log_, clock_)) {
+ delay_based_bwe_(new DelayBasedBwe(event_log_, clock_)),
+ was_in_alr_(0) {
delay_based_bwe_->SetMinBitrate(min_bitrate_bps_);
}
@@ -277,6 +280,14 @@
std::vector<PacketFeedback> feedback_vector = ReceivedPacketFeedbackVector(
transport_feedback_adapter_.GetTransportFeedbackVector());
SortPacketFeedbackVector(&feedback_vector);
+
+ bool currently_in_alr =
+ pacer_->GetApplicationLimitedRegionStartTime().has_value();
+ if (!currently_in_alr && was_in_alr_) {
+ acknowledged_bitrate_estimator_->SetAlrEndedTimeMs(rtc::TimeMillis());
+ }
+ was_in_alr_ = currently_in_alr;
+
acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
feedback_vector);
DelayBasedBwe::Result result;
diff --git a/webrtc/modules/pacing/alr_detector.h b/webrtc/modules/pacing/alr_detector.h
index 11eecea..7e63097 100644
--- a/webrtc/modules/pacing/alr_detector.h
+++ b/webrtc/modules/pacing/alr_detector.h
@@ -27,7 +27,6 @@
// AlrDetector provides a signal that can be utilized to adjust
// estimate bandwidth.
// Note: This class is not thread-safe.
-
class AlrDetector {
public:
AlrDetector();
@@ -57,8 +56,8 @@
// kAlrEndUsagePercent. NOTE: This is intentionally conservative at the moment
// until BW adjustments of application limited region is fine tuned.
static constexpr int kDefaultAlrBandwidthUsagePercent = 65;
- static constexpr int kDefaultAlrStartBudgetLevelPercent = 20;
- static constexpr int kDefaultAlrStopBudgetLevelPercent = -20;
+ static constexpr int kDefaultAlrStartBudgetLevelPercent = 80;
+ static constexpr int kDefaultAlrStopBudgetLevelPercent = 50;
static const char* kScreenshareProbingBweExperimentName;
void UpdateBudgetWithElapsedTime(int64_t delta_time_ms);
diff --git a/webrtc/modules/pacing/alr_detector_unittest.cc b/webrtc/modules/pacing/alr_detector_unittest.cc
index e1a7810..9f1998b 100644
--- a/webrtc/modules/pacing/alr_detector_unittest.cc
+++ b/webrtc/modules/pacing/alr_detector_unittest.cc
@@ -86,7 +86,7 @@
// Verify that we ALR starts when bitrate drops below 20%.
SimulateOutgoingTrafficIn(&alr_detector_)
- .ForTimeMs(1000)
+ .ForTimeMs(1500)
.AtPercentOfEstimatedBitrate(20);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
@@ -109,7 +109,7 @@
// Verify that we stay in ALR region even after a short bitrate spike.
SimulateOutgoingTrafficIn(&alr_detector_)
- .ForTimeMs(200)
+ .ForTimeMs(100)
.AtPercentOfEstimatedBitrate(150);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
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 d2aa8de..ed6a7cd 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/acknowledge_bitrate_estimator.h"
+#include "webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h"
#include "webrtc/modules/remote_bitrate_estimator/include/send_time_history.h"
#include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"