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: 37aa8ba616

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}

resources/voice_engine/audio_dtx16.wav.sha1

@@ -0,0 +1 @@
+cafd7151d6b7b4313d0bd2a128a31fc83bca7aa3

webrtc/BUILD.gn

@@ -496,6 +496,7 @@ 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",

webrtc/audio/BUILD.gn

@@ -113,6 +113,8 @@ if (rtc_include_tests) {
         "../test:fake_audio_device",
         "../test:test_common",
         "../test:test_main",
+        "//testing/gmock",
+        "//testing/gtest",
         "//third_party/gflags",
       ]
       if (is_android) {
@@ -122,6 +124,7 @@ 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) {
@@ -130,4 +133,40 @@ 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" ]
+    }
+  }
 }

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

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_

webrtc/modules/congestion_controller/BUILD.gn

@@ -10,8 +10,10 @@ import("../../webrtc.gni")
 
 rtc_static_library("congestion_controller") {
   sources = [
-    "acknowledge_bitrate_estimator.cc",
+    "acknowledged_bitrate_estimator.cc",
-    "acknowledge_bitrate_estimator.h",
+    "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 @@ if (rtc_include_tests) {
       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 @@ 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",

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

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_

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*/

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 webrtc {
 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(
+BitrateEstimator::~BitrateEstimator() = default;
-    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 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 @@ void AcknowledgedBitrateEstimator::Update(int64_t now_ms, int bytes) {
                         bitrate_estimate_ * 1000);
 }
 
-float AcknowledgedBitrateEstimator::UpdateWindow(int64_t now_ms,
+float BitrateEstimator::UpdateWindow(int64_t now_ms,
-                                                 int bytes,
+                                     int bytes,
-                                                 int rate_window_ms) {
+                                     int rate_window_ms) {
   // Reset if time moves backwards.
   if (now_ms < prev_time_ms_) {
     prev_time_ms_ = -1;
@@ -106,10 +92,16 @@ float AcknowledgedBitrateEstimator::UpdateWindow(int64_t now_ms,
   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

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_
+#ifndef WEBRTC_MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
-#define WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGE_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(
+  virtual rtc::Optional<uint32_t> bitrate_bps() const;
-      const std::vector<PacketFeedback>& packet_feedback_vector);
+
-  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 @@ class AcknowledgedBitrateEstimator {
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGE_BITRATE_ESTIMATOR_H_
+#endif  // WEBRTC_MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_

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"

webrtc/modules/congestion_controller/include/send_side_congestion_controller.h

@@ -154,6 +154,7 @@ class SendSideCongestionController : public CallStatsObserver,
   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_;
 

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 @@ SendSideCongestionController::SendSideCongestionController(
       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 @@ void SendSideCongestionController::OnTransportFeedback(
   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;

webrtc/modules/pacing/alr_detector.h

@@ -27,7 +27,6 @@ namespace webrtc {
 // 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 @@ class AlrDetector {
   // 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 kDefaultAlrStartBudgetLevelPercent = 80;
-  static constexpr int kDefaultAlrStopBudgetLevelPercent = -20;
+  static constexpr int kDefaultAlrStopBudgetLevelPercent = 50;
   static const char* kScreenshareProbingBweExperimentName;
 
   void UpdateBudgetWithElapsedTime(int64_t delta_time_ms);

webrtc/modules/pacing/alr_detector_unittest.cc

@@ -86,7 +86,7 @@ TEST_F(AlrDetectorTest, AlrDetection) {
 
   // 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 @@ TEST_F(AlrDetectorTest, ShortSpike) {
 
   // 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());
 

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"