Final version of BBR, with tweaks made for WebRTC, major changes:

1) Entering PROBE_RTT when necessary.
2) Congestion window gain of 0.65 instead of constant 4 packets.
3) {1.1, 0.9} pair instead of {1.25, 0.75}
4) Recovery mode.
5) No reaction to losses due to Recovery mode's implementation.
6) Supports encoder.
7) A new test compiling most of the simulation tests.
8) Bucket for high gain phase, disabled by default.
9) Pacer specific to BBR.

BUG=webrtc:7713

Review-Url: https://codereview.webrtc.org/2999073002
Cr-Commit-Position: refs/heads/master@{#19418}

webrtc/modules/bitrate_controller/include/bitrate_controller.h

@@ -15,6 +15,8 @@
 #ifndef WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
 #define WEBRTC_MODULES_BITRATE_CONTROLLER_INCLUDE_BITRATE_CONTROLLER_H_
 
+#include <map>
+
 #include "webrtc/modules/congestion_controller/delay_based_bwe.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/pacing/paced_sender.h"
@@ -36,12 +38,18 @@ class BitrateObserver {
   virtual void OnNetworkChanged(uint32_t bitrate_bps,
                                 uint8_t fraction_loss,  // 0 - 255.
                                 int64_t rtt_ms) = 0;
-
+  // TODO(gnish): Merge these two into one function.
+  virtual void OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
+                                uint32_t bitrate_for_pacer_bps,
+                                bool in_probe_rtt,
+                                int64_t target_set_time,
+                                uint64_t congestion_window) {}
+  virtual void OnBytesAcked(size_t bytes) {}
+  virtual size_t pacer_queue_size_in_bytes() { return 0; }
   virtual ~BitrateObserver() {}
 };
 
-class BitrateController : public Module,
+class BitrateController : public Module, public RtcpBandwidthObserver {
-                          public RtcpBandwidthObserver {
   // This class collects feedback from all streams sent to a peer (via
   // RTCPBandwidthObservers). It does one  aggregated send side bandwidth
   // estimation and divide the available bitrate between all its registered

webrtc/modules/pacing/BUILD.gn

@@ -18,6 +18,7 @@ rtc_static_library("pacing") {
     "interval_budget.h",
     "paced_sender.cc",
     "paced_sender.h",
+    "pacer.h",
     "packet_router.cc",
     "packet_router.h",
   ]

webrtc/modules/pacing/paced_sender.cc

@@ -37,8 +37,8 @@ const int64_t kMaxIntervalTimeMs = 30;
 
 }  // namespace
 
-// TODO(sprang): Move at least PacketQueue out to separate
+// TODO(sprang): Move at least PacketQueue out to separate files, so that we can
-// files, so that we can more easily test them.
+// more easily test them.
 
 namespace webrtc {
 namespace paced_sender {

webrtc/modules/pacing/paced_sender.h

@@ -15,8 +15,7 @@
 #include <memory>
 #include <set>
 
-#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/pacing/pacer.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/rtc_base/criticalsection.h"
 #include "webrtc/rtc_base/optional.h"
 #include "webrtc/rtc_base/thread_annotations.h"
@@ -31,11 +30,12 @@ class RtcEventLog;
 class IntervalBudget;
 
 namespace paced_sender {
+class IntervalBudget;
 struct Packet;
 class PacketQueue;
 }  // namespace paced_sender
 
-class PacedSender : public Module, public RtpPacketSender {
+class PacedSender : public Pacer {
  public:
   class PacketSender {
    public:
@@ -93,7 +93,7 @@ class PacedSender : public Module, public RtpPacketSender {
   // |bitrate_bps| is our estimate of what we are allowed to send on average.
   // We will pace out bursts of packets at a bitrate of
   // |bitrate_bps| * kDefaultPaceMultiplier.
-  virtual void SetEstimatedBitrate(uint32_t bitrate_bps);
+  void SetEstimatedBitrate(uint32_t bitrate_bps) override;
 
   // Sets the minimum send bitrate and maximum padding bitrate requested by send
   // streams.
@@ -149,7 +149,6 @@ class PacedSender : public Module, public RtpPacketSender {
 
   // Called when the prober is associated with a process thread.
   void ProcessThreadAttached(ProcessThread* process_thread) override;
-
   void SetPacingFactor(float pacing_factor);
   void SetQueueTimeLimit(int limit_ms);
 

webrtc/modules/pacing/pacer.h

@@ -0,0 +1,37 @@
+/*
+ *  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_PACING_PACER_H_
+#define WEBRTC_MODULES_PACING_PACER_H_
+
+#include "webrtc/modules/include/module.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+
+namespace webrtc {
+class Pacer : public Module, public RtpPacketSender {
+ public:
+  virtual void SetEstimatedBitrate(uint32_t bitrate_bps) {}
+  virtual void SetEstimatedBitrateAndCongestionWindow(
+      uint32_t bitrate_bps,
+      bool in_probe_rtt,
+      uint64_t congestion_window) {}
+  virtual void OnBytesAcked(size_t bytes) {}
+  void InsertPacket(RtpPacketSender::Priority priority,
+                    uint32_t ssrc,
+                    uint16_t sequence_number,
+                    int64_t capture_time_ms,
+                    size_t bytes,
+                    bool retransmission) override = 0;
+  int64_t TimeUntilNextProcess() override = 0;
+  void Process() override = 0;
+  ~Pacer() override {}
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_PACING_PACER_H_

webrtc/modules/remote_bitrate_estimator/BUILD.gn

@@ -61,6 +61,8 @@ if (rtc_include_tests) {
   rtc_static_library("bwe_simulator_lib") {
     testonly = true
     sources = [
+      "test/bbr_paced_sender.cc",
+      "test/bbr_paced_sender.h",
       "test/bwe.cc",
       "test/bwe.h",
       "test/bwe_test.cc",

webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc

@@ -122,9 +122,64 @@ TEST_P(BweSimulation, Choke1000kbps500kbps1000kbps) {
   RunFor(60 * 1000);
 }
 
+TEST_P(BweSimulation, SimulationsCompiled) {
+  AdaptiveVideoSource source(0, 30, 300, 0, 0);
+  PacedVideoSender sender(&uplink_, &source, GetParam());
+  int zero = 0;
+  // CreateFlowIds() doesn't support passing int as a flow id, so we pass
+  // pointer instead.
+  DelayFilter delay(&uplink_, CreateFlowIds(&zero, 1));
+  delay.SetOneWayDelayMs(100);
+  ChokeFilter filter(&uplink_, 0);
+  RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
+  PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
+  filter.set_max_delay_ms(500);
+  filter.set_capacity_kbps(1000);
+  RunFor(60 * 1000);
+  filter.set_capacity_kbps(500);
+  RunFor(50 * 1000);
+  filter.set_capacity_kbps(1000);
+  RunFor(60 * 1000);
+  filter.set_capacity_kbps(200);
+  RunFor(60 * 1000);
+  filter.set_capacity_kbps(50);
+  RunFor(60 * 1000);
+  filter.set_capacity_kbps(200);
+  RunFor(60 * 1000);
+  filter.set_capacity_kbps(500);
+  RunFor(60 * 1000);
+  filter.set_capacity_kbps(300);
+  RunFor(60 * 1000);
+  filter.set_capacity_kbps(1000);
+  RunFor(60 * 1000);
+  const int kStartingCapacityKbps = 150;
+  const int kEndingCapacityKbps = 1500;
+  const int kStepKbps = 5;
+  const int kStepTimeMs = 1000;
+  for (int i = kStartingCapacityKbps; i <= kEndingCapacityKbps;
+       i += kStepKbps) {
+    filter.set_capacity_kbps(i);
+    RunFor(kStepTimeMs);
+  }
+  for (int i = kEndingCapacityKbps; i >= kStartingCapacityKbps;
+       i -= kStepKbps) {
+    filter.set_capacity_kbps(i);
+    RunFor(kStepTimeMs);
+  }
+  filter.set_capacity_kbps(150);
+  RunFor(120 * 1000);
+  filter.set_capacity_kbps(500);
+  RunFor(60 * 1000);
+}
+
 TEST_P(BweSimulation, PacerChoke1000kbps500kbps1000kbps) {
   AdaptiveVideoSource source(0, 30, 300, 0, 0);
   PacedVideoSender sender(&uplink_, &source, GetParam());
+  const int kFlowId = 0;
+  // CreateFlowIds() doesn't support passing int as a flow id, so we pass
+  // pointer instead.
+  DelayFilter delay(&uplink_, CreateFlowIds(&kFlowId, 1));
+  delay.SetOneWayDelayMs(100);
   ChokeFilter filter(&uplink_, 0);
   RateCounterFilter counter(&uplink_, 0, "Receiver", bwe_names[GetParam()]);
   PacketReceiver receiver(&uplink_, 0, GetParam(), true, true);
@@ -262,6 +317,11 @@ TEST_P(BweSimulation, LinearDecreasingCapacity) {
 TEST_P(BweSimulation, PacerGoogleWifiTrace3Mbps) {
   PeriodicKeyFrameSource source(0, 30, 300, 0, 0, 1000);
   PacedVideoSender sender(&uplink_, &source, GetParam());
+  int kFlowId = 0;
+  // CreateFlowIds() doesn't support passing int as a flow id, so we pass
+  // pointer instead.
+  DelayFilter delay(&uplink_, CreateFlowIds(&kFlowId, 1));
+  delay.SetOneWayDelayMs(100);
   RateCounterFilter counter1(&uplink_, 0, "sender_output",
                              bwe_names[GetParam()]);
   TraceBasedDeliveryFilter filter(&uplink_, 0, "link_capacity");

webrtc/modules/remote_bitrate_estimator/test/bbr_paced_sender.cc

@@ -0,0 +1,140 @@
+/*
+ *  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/remote_bitrate_estimator/test/bbr_paced_sender.h"
+
+#include <algorithm>
+#include <queue>
+#include <set>
+#include <vector>
+
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h"
+#include "webrtc/system_wrappers/include/clock.h"
+
+namespace webrtc {
+
+BbrPacedSender::BbrPacedSender(const Clock* clock,
+                               PacedSender::PacketSender* packet_sender,
+                               RtcEventLog* event_log)
+    : clock_(clock),
+      packet_sender_(packet_sender),
+      estimated_bitrate_bps_(100000),
+      min_send_bitrate_kbps_(0),
+      pacing_bitrate_kbps_(0),
+      time_last_update_us_(clock->TimeInMicroseconds()),
+      time_last_update_ms_(clock->TimeInMilliseconds()),
+      next_packet_send_time_(clock_->TimeInMilliseconds()),
+      rounding_error_time_ms_(0.0f),
+      packets_(),
+      max_data_inflight_bytes_(10000),
+      congestion_window_(new testing::bwe::CongestionWindow()) {}
+BbrPacedSender::~BbrPacedSender() {}
+
+void BbrPacedSender::SetEstimatedBitrateAndCongestionWindow(
+    uint32_t bitrate_bps,
+    bool in_probe_rtt,
+    uint64_t congestion_window) {
+  estimated_bitrate_bps_ = bitrate_bps;
+  max_data_inflight_bytes_ = congestion_window;
+}
+
+void BbrPacedSender::SetMinBitrate(int min_send_bitrate_bps) {
+  min_send_bitrate_kbps_ = min_send_bitrate_bps / 1000;
+  pacing_bitrate_kbps_ =
+      std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000);
+}
+
+void BbrPacedSender::InsertPacket(RtpPacketSender::Priority priority,
+                                  uint32_t ssrc,
+                                  uint16_t sequence_number,
+                                  int64_t capture_time_ms,
+                                  size_t bytes,
+                                  bool retransmission) {
+  int64_t now_ms = clock_->TimeInMilliseconds();
+  if (capture_time_ms < 0)
+    capture_time_ms = now_ms;
+  packets_.push_back(new Packet(priority, ssrc, sequence_number,
+                                capture_time_ms, now_ms, bytes,
+                                retransmission));
+}
+
+int64_t BbrPacedSender::TimeUntilNextProcess() {
+  // Once errors absolute value hits 1 millisecond, add compensating term to
+  // the |next_packet_send_time_|, so that we can send packet earlier or later,
+  // depending on the error.
+  rounding_error_time_ms_ = std::min(rounding_error_time_ms_, 1.0f);
+  if (rounding_error_time_ms_ < -0.9f)
+    rounding_error_time_ms_ = -1.0f;
+  int64_t result =
+      std::max<int64_t>(next_packet_send_time_ + time_last_update_ms_ -
+                            clock_->TimeInMilliseconds(),
+                        0);
+  if (rounding_error_time_ms_ == 1.0f || rounding_error_time_ms_ == -1.0f) {
+    next_packet_send_time_ -= rounding_error_time_ms_;
+    result = std::max<int64_t>(next_packet_send_time_ + time_last_update_ms_ -
+                                   clock_->TimeInMilliseconds(),
+                               0);
+    rounding_error_time_ms_ = 0;
+  }
+  return result;
+}
+
+void BbrPacedSender::OnBytesAcked(size_t bytes) {
+  congestion_window_->AckReceived(bytes);
+}
+
+void BbrPacedSender::Process() {
+  pacing_bitrate_kbps_ =
+      std::max(min_send_bitrate_kbps_, estimated_bitrate_bps_ / 1000);
+  // If we have nothing to send, try sending again in 1 millisecond.
+  if (packets_.empty()) {
+    next_packet_send_time_ = 1;
+    return;
+  }
+  // If congestion window doesn't allow sending, try again in 1 millisecond.
+  if (packets_.front()->size_in_bytes + congestion_window_->data_inflight() >
+      max_data_inflight_bytes_) {
+    next_packet_send_time_ = 1;
+    return;
+  }
+  bool sent = TryToSendPacket(packets_.front());
+  if (sent) {
+    congestion_window_->PacketSent(packets_.front()->size_in_bytes);
+    delete packets_.front();
+    packets_.pop_front();
+    time_last_update_ms_ = clock_->TimeInMilliseconds();
+    if (!packets_.empty()) {
+      // Calculate in what time we should send current packet.
+      next_packet_send_time_ = (packets_.front()->size_in_bytes * 8000 +
+                                estimated_bitrate_bps_ / 2) /
+                               estimated_bitrate_bps_;
+      // As rounding errors may happen, |rounding_error_time_ms_| could be
+      // positive or negative depending on packet was sent earlier or later,
+      // after it hits certain threshold we will send a packet earlier or later
+      // depending on error we had so far.
+      rounding_error_time_ms_ +=
+          (next_packet_send_time_ - packets_.front()->size_in_bytes * 8000.0f /
+                                        estimated_bitrate_bps_ * 1.0f);
+    } else {
+      // If sending was unsuccessful try again in 1 millisecond.
+      next_packet_send_time_ = 1;
+    }
+  }
+}
+
+bool BbrPacedSender::TryToSendPacket(Packet* packet) {
+  PacedPacketInfo pacing_info;
+  return packet_sender_->TimeToSendPacket(packet->ssrc, packet->sequence_number,
+                                          packet->capture_time_ms,
+                                          packet->retransmission, pacing_info);
+}
+
+}  // namespace webrtc

webrtc/modules/remote_bitrate_estimator/test/bbr_paced_sender.h

@@ -0,0 +1,92 @@
+/*
+ *  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_REMOTE_BITRATE_ESTIMATOR_TEST_BBR_PACED_SENDER_H_
+#define WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BBR_PACED_SENDER_H_
+
+#include <list>
+#include <memory>
+
+#include "webrtc/modules/pacing/paced_sender.h"
+#include "webrtc/modules/pacing/pacer.h"
+
+namespace webrtc {
+namespace testing {
+namespace bwe {
+class CongestionWindow;
+}
+}  // namespace testing
+
+struct Packet {
+  Packet(RtpPacketSender::Priority priority,
+         uint32_t ssrc,
+         uint16_t seq_number,
+         int64_t capture_time_ms,
+         int64_t enqueue_time_ms,
+         size_t size_in_bytes,
+         bool retransmission)
+      : priority(priority),
+        ssrc(ssrc),
+        sequence_number(seq_number),
+        capture_time_ms(capture_time_ms),
+        enqueue_time_ms(enqueue_time_ms),
+        size_in_bytes(size_in_bytes),
+        retransmission(retransmission) {}
+  RtpPacketSender::Priority priority;
+  uint32_t ssrc;
+  uint16_t sequence_number;
+  int64_t capture_time_ms;
+  int64_t enqueue_time_ms;
+  size_t size_in_bytes;
+  bool retransmission;
+};
+
+class Clock;
+class RtcEventLog;
+struct Packet;
+class BbrPacedSender : public Pacer {
+ public:
+  BbrPacedSender(const Clock* clock,
+                 PacedSender::PacketSender* packet_sender,
+                 RtcEventLog* event_log);
+  ~BbrPacedSender() override;
+  void SetEstimatedBitrateAndCongestionWindow(
+      uint32_t bitrate_bps,
+      bool in_probe_rtt,
+      uint64_t congestion_window) override;
+  void SetMinBitrate(int min_send_bitrate_bps);
+  void InsertPacket(RtpPacketSender::Priority priority,
+                    uint32_t ssrc,
+                    uint16_t sequence_number,
+                    int64_t capture_time_ms,
+                    size_t bytes,
+                    bool retransmission) override;
+  int64_t TimeUntilNextProcess() override;
+  void OnBytesAcked(size_t bytes) override;
+  void Process() override;
+  bool TryToSendPacket(Packet* packet);
+
+ private:
+  const Clock* const clock_;
+  PacedSender::PacketSender* const packet_sender_;
+  uint32_t estimated_bitrate_bps_;
+  uint32_t min_send_bitrate_kbps_;
+  uint32_t pacing_bitrate_kbps_;
+  int64_t time_last_update_us_;
+  int64_t time_last_update_ms_;
+  int64_t next_packet_send_time_;
+  float rounding_error_time_ms_;
+  std::list<Packet*> packets_;
+  // TODO(gnish): integrate |max_data_inflight| into congestion window class.
+  size_t max_data_inflight_bytes_;
+  std::unique_ptr<testing::bwe::CongestionWindow> congestion_window_;
+};
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_BBR_PACED_SENDER_H_

webrtc/modules/remote_bitrate_estimator/test/bwe.cc

@@ -95,7 +95,7 @@ BweSender* CreateBweSender(BandwidthEstimatorType estimator,
     case kNadaEstimator:
       return new NadaBweSender(kbps, observer, clock);
     case kBbrEstimator:
-      return new BbrBweSender(clock);
+      return new BbrBweSender(observer, clock);
     case kTcpEstimator:
       FALLTHROUGH();
     case kNullEstimator:

webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc

@@ -157,7 +157,7 @@ BbrBweFeedback::BbrBweFeedback(
     int flow_id,
     int64_t send_time_us,
     int64_t latest_send_time_ms,
-    const std::vector<uint64_t>& packet_feedback_vector)
+    const std::vector<uint16_t>& packet_feedback_vector)
     : FeedbackPacket(flow_id, send_time_us, latest_send_time_ms),
       packet_feedback_vector_(packet_feedback_vector) {}
 
@@ -518,12 +518,12 @@ void ChokeFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
   for (PacketsIt it = in_out->begin(); it != in_out->end(); ) {
     int64_t earliest_send_time_us =
         std::max(last_send_time_us_, (*it)->send_time_us());
-
     int64_t new_send_time_us =
         earliest_send_time_us +
         ((*it)->payload_size() * 8 * 1000 + capacity_kbps_ / 2) /
             capacity_kbps_;
-
+    BWE_TEST_LOGGING_PLOT(0, "MaxThroughput_", new_send_time_us / 1000,
+                          capacity_kbps_);
     if (delay_cap_helper_->ShouldSendPacket(new_send_time_us,
                                             (*it)->send_time_us())) {
       (*it)->set_send_time_us(new_send_time_us);

webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.cc

@@ -34,7 +34,7 @@ const float kDrainGain = 1 / kHighGain;
 const float kStartupGrowthTarget = 1.25f;
 const int kMaxRoundsWithoutGrowth = 3;
 // Pacing gain values for Probe Bandwidth mode.
-const float kPacingGain[] = {1.25, 0.75, 1, 1, 1, 1, 1, 1};
+const float kPacingGain[] = {1.1, 0.9, 1, 1, 1, 1, 1, 1};
 const size_t kGainCycleLength = sizeof(kPacingGain) / sizeof(kPacingGain[0]);
 // Least number of rounds PROBE_RTT should last.
 const int kProbeRttDurationRounds = 1;
@@ -46,7 +46,7 @@ const float kTargetCongestionWindowGain = 1;
 // equal to 1, but in practice because of delayed acks and the way networks
 // work, it is nice to have some extra room in congestion window for full link
 // utilization. Value chosen by observations on different tests.
-const float kCruisingCongestionWindowGain = 1.5f;
+const float kCruisingCongestionWindowGain = 2;
 // Pacing gain specific for Recovery mode. Chosen by experiments in simulation
 // tool.
 const float kRecoveryPacingGain = 0.5f;
@@ -61,10 +61,29 @@ const float kRttIncreaseThreshold = 3;
 // Threshold to assume average RTT has decreased for a round. Chosen by
 // experiments in simulation tool.
 const float kRttDecreaseThreshold = 1.5f;
+// If |kCongestionWindowThreshold| of the congestion window is filled up, tell
+// encoder to stop, to avoid building sender side queues.
+const float kCongestionWindowThreshold = 0.69f;
+// Duration we send at |kDefaultRatebps| in order to ensure BBR has data to work
+// with.
+const int64_t kDefaultDurationMs = 200;
+const int64_t kDefaultRatebps = 300000;
+// Congestion window gain for PROBE_RTT mode.
+const float kProbeRttCongestionWindowGain = 0.65f;
+// We need to be sure that data inflight has increased by at least
+// |kTargetCongestionWindowGainForHighGain| compared to the congestion window in
+// PROBE_BW's high gain phase, to make ramp-up quicker. As high gain value has
+// been decreased from 1.25 to 1.1 we need to make
+// |kTargetCongestionWindowGainForHighGain| slightly higher than the actual high
+// gain value.
+const float kTargetCongestionWindowGainForHighGain = 1.15f;
+// Encoder rate gain value for PROBE_RTT mode.
+const float kEncoderRateGainForProbeRtt = 0.1f;
 }  // namespace
 
-BbrBweSender::BbrBweSender(Clock* clock)
+BbrBweSender::BbrBweSender(BitrateObserver* observer, Clock* clock)
     : BweSender(0),
+      observer_(observer),
       clock_(clock),
       mode_(STARTUP),
       max_bandwidth_filter_(new MaxBandwidthFilter()),
@@ -113,14 +132,13 @@ void BbrBweSender::CalculatePacingRate() {
       max_bandwidth_filter_->max_bandwidth_estimate_bps() * pacing_gain_;
 }
 
+// Declare lost packets as acked.
 void BbrBweSender::HandleLoss(uint64_t last_acked_packet,
                               uint64_t recently_acked_packet) {
-  // Logic specific to wrapping sequence numbers.
-  if (!last_acked_packet)
-    return;
   for (uint16_t i = last_acked_packet + 1;
        AheadOrAt<uint16_t>(recently_acked_packet - 1, i); i++) {
     congestion_window_->AckReceived(packet_stats_[i].payload_size_bytes);
+    observer_->OnBytesAcked(packet_stats_[i].payload_size_bytes);
   }
 }
 
@@ -148,7 +166,7 @@ void BbrBweSender::GiveFeedback(const FeedbackPacket& feedback) {
   last_packet_ack_time_ = now_ms;
   const BbrBweFeedback& fb = static_cast<const BbrBweFeedback&>(feedback);
   // feedback_vector holds values of acknowledged packets' sequence numbers.
-  const std::vector<uint64_t>& feedback_vector = fb.packet_feedback_vector();
+  const std::vector<uint16_t>& feedback_vector = fb.packet_feedback_vector();
   // Go through all the packets acked, update variables/containers accordingly.
   for (uint16_t sequence_number : feedback_vector) {
     // Completing packet information with a recently received ack.
@@ -175,9 +193,7 @@ void BbrBweSender::GiveFeedback(const FeedbackPacket& feedback) {
           bytes_acked_ - packet->payload_size_bytes / 2;
       high_gain_over_ = true;
     }
-    // Notify pacer that an ack was received, to adjust data inflight.
+    observer_->OnBytesAcked(packet->payload_size_bytes);
-    // TODO(gnish): Add implementation for BitrateObserver class, to notify
-    // pacer about incoming acks.
     congestion_window_->AckReceived(packet->payload_size_bytes);
     HandleLoss(last_packet_acked_sequence_number_, packet->sequence_number);
     last_packet_acked_sequence_number_ = packet->sequence_number;
@@ -196,8 +212,7 @@ void BbrBweSender::GiveFeedback(const FeedbackPacket& feedback) {
       round_trip_end_ = last_packet_sent_sequence_number_;
     }
   }
-  bool min_rtt_expired = false;
+  TryEnteringProbeRtt(now_ms);
-  min_rtt_expired =
   UpdateBandwidthAndMinRtt(now_ms, feedback_vector, bytes_acked_);
   if (new_round_started && !full_bandwidth_reached_) {
     full_bandwidth_reached_ = max_bandwidth_filter_->FullBandwidthReached(
@@ -221,19 +236,37 @@ void BbrBweSender::GiveFeedback(const FeedbackPacket& feedback) {
       TryExitingRecovery(new_round_started);
       break;
   }
-  TryEnteringProbeRtt(now_ms);
   TryEnteringRecovery(new_round_started);  // Comment this line to disable
                                            // entering Recovery mode.
-  for (uint64_t f : feedback_vector)
+  for (uint16_t f : feedback_vector)
     AddToPastRtts(packet_stats_[f].ack_time_ms - packet_stats_[f].send_time_ms);
   CalculatePacingRate();
+  size_t cwnd = congestion_window_->GetCongestionWindow(
+      mode_, max_bandwidth_filter_->max_bandwidth_estimate_bps(),
+      min_rtt_filter_->min_rtt_ms(), congestion_window_gain_);
   // Make sure we don't get stuck when pacing_rate is 0, because of simulation
   // tool specifics.
   if (!pacing_rate_bps_)
     pacing_rate_bps_ = 100;
   BWE_TEST_LOGGING_PLOT(1, "SendRate", now_ms, pacing_rate_bps_ / 1000);
-  // TODO(gnish): Add implementation for BitrateObserver class to update pacing
+  int64_t rate_for_pacer_bps = pacing_rate_bps_;
-  // rate for the pacer and the encoder.
+  int64_t rate_for_encoder_bps = pacing_rate_bps_;
+  if (congestion_window_->data_inflight() >= cwnd * kCongestionWindowThreshold)
+    rate_for_encoder_bps = 0;
+  // We dont completely stop sending during PROBE_RTT, so we need encoder to
+  // produce something, another way of doing this would be telling encoder to
+  // stop and send padding instead of actual data.
+  if (mode_ == PROBE_RTT)
+    rate_for_encoder_bps = rate_for_pacer_bps * kEncoderRateGainForProbeRtt;
+  // Send for 300 kbps for first 200 ms, so that BBR has data to work with.
+  if (now_ms <= kDefaultDurationMs)
+    observer_->OnNetworkChanged(
+        kDefaultRatebps, kDefaultRatebps, false,
+        clock_->TimeInMicroseconds() + kFeedbackIntervalsMs * 1000, cwnd);
+  else
+    observer_->OnNetworkChanged(
+        rate_for_encoder_bps, rate_for_pacer_bps, mode_ == PROBE_RTT,
+        clock_->TimeInMicroseconds() + kFeedbackIntervalsMs * 1000, cwnd);
 }
 
 size_t BbrBweSender::TargetCongestionWindow(float gain) {
@@ -251,16 +284,16 @@ rtc::Optional<int64_t> BbrBweSender::CalculateBandwidthSample(
     int64_t ack_time_delta_ms) {
   rtc::Optional<int64_t> bandwidth_sample;
   if (send_time_delta_ms > 0)
-    *bandwidth_sample = data_sent_bytes * 8000 / send_time_delta_ms;
+    bandwidth_sample.emplace(data_sent_bytes * 8000 / send_time_delta_ms);
   rtc::Optional<int64_t> ack_rate;
   if (ack_time_delta_ms > 0)
-    *ack_rate = data_acked_bytes * 8000 / ack_time_delta_ms;
+    ack_rate.emplace(data_acked_bytes * 8000 / ack_time_delta_ms);
   // If send rate couldn't be calculated automaticaly set |bandwidth_sample| to
   // ack_rate.
   if (!bandwidth_sample)
     bandwidth_sample = ack_rate;
   if (bandwidth_sample && ack_rate)
-    *bandwidth_sample = std::min(*bandwidth_sample, *ack_rate);
+    bandwidth_sample.emplace(std::min(*bandwidth_sample, *ack_rate));
   return bandwidth_sample;
 }
 
@@ -285,12 +318,12 @@ void BbrBweSender::AddSampleForHighGain() {
   first_packet_send_time_during_high_gain_ms_.reset();
 }
 
-bool BbrBweSender::UpdateBandwidthAndMinRtt(
+void BbrBweSender::UpdateBandwidthAndMinRtt(
     int64_t now_ms,
-    const std::vector<uint64_t>& feedback_vector,
+    const std::vector<uint16_t>& feedback_vector,
     int64_t bytes_acked) {
   rtc::Optional<int64_t> min_rtt_sample_ms;
-  for (uint64_t f : feedback_vector) {
+  for (uint16_t f : feedback_vector) {
     PacketStats packet = packet_stats_[f];
     size_t data_sent_bytes =
         packet.data_sent_bytes - packet.data_sent_before_last_sent_packet_bytes;
@@ -306,20 +339,22 @@ bool BbrBweSender::UpdateBandwidthAndMinRtt(
     if (bandwidth_sample)
       max_bandwidth_filter_->AddBandwidthSample(*bandwidth_sample,
                                                 round_count_);
-    AddSampleForHighGain();  // Comment to disable bucket for high gain.
+    // AddSampleForHighGain();  // Comment to disable bucket for high gain.
     if (!min_rtt_sample_ms)
-      *min_rtt_sample_ms = packet.ack_time_ms - packet.send_time_ms;
+      min_rtt_sample_ms.emplace(packet.ack_time_ms - packet.send_time_ms);
     else
       *min_rtt_sample_ms = std::min(*min_rtt_sample_ms,
                                     packet.ack_time_ms - packet.send_time_ms);
     BWE_TEST_LOGGING_PLOT(1, "MinRtt", now_ms,
                           packet.ack_time_ms - packet.send_time_ms);
   }
-  if (!min_rtt_sample_ms)
+  // We only feed RTT samples into the min_rtt filter which were not produced
-    return false;
+  // during 1.1 gain phase, to ensure they contain no queueing delay. But if the
+  // rtt sample from 1.1 gain phase improves the current estimate then we should
+  // make it as a new best estimate.
+  if (pacing_gain_ <= 1.0f || !min_rtt_filter_->min_rtt_ms() ||
+      *min_rtt_filter_->min_rtt_ms() >= *min_rtt_sample_ms)
     min_rtt_filter_->AddRttSample(*min_rtt_sample_ms, now_ms);
-  bool min_rtt_expired = min_rtt_filter_->MinRttExpired(now_ms);
-  return min_rtt_expired;
 }
 
 void BbrBweSender::EnterStartup() {
@@ -365,8 +400,9 @@ void BbrBweSender::TryUpdatingCyclePhase(int64_t now_ms) {
   // If BBR was probing and it couldn't increase data inflight sufficiently in
   // one min_rtt time, continue probing. BBR design doc isn't clear about this,
   // but condition helps in quicker ramp-up and performs better.
-  if (pacing_gain_ > 1.0 && congestion_window_->data_inflight() <
+  if (pacing_gain_ > 1.0 &&
-                                TargetCongestionWindow(pacing_gain_))
+      congestion_window_->data_inflight() <
+          TargetCongestionWindow(kTargetCongestionWindowGainForHighGain))
     advance_cycle_phase = false;
   // If BBR has already drained queues there is no point in continuing draining
   // phase.
@@ -385,6 +421,7 @@ void BbrBweSender::TryEnteringProbeRtt(int64_t now_ms) {
   if (min_rtt_filter_->MinRttExpired(now_ms) && mode_ != PROBE_RTT) {
     mode_ = PROBE_RTT;
     pacing_gain_ = 1;
+    congestion_window_gain_ = kProbeRttCongestionWindowGain;
     probe_rtt_start_time_ms_ = now_ms;
     minimum_congestion_window_start_time_ms_.reset();
   }
@@ -397,22 +434,23 @@ void BbrBweSender::TryEnteringProbeRtt(int64_t now_ms) {
 void BbrBweSender::TryExitingProbeRtt(int64_t now_ms, int64_t round) {
   if (!minimum_congestion_window_start_time_ms_) {
     if (congestion_window_->data_inflight() <=
-        CongestionWindow::kMinimumCongestionWindowBytes) {
+        TargetCongestionWindow(kProbeRttCongestionWindowGain)) {
-      *minimum_congestion_window_start_time_ms_ = now_ms;
+      minimum_congestion_window_start_time_ms_.emplace(now_ms);
       minimum_congestion_window_start_round_ = round;
     }
   } else {
     if (now_ms - *minimum_congestion_window_start_time_ms_ >=
             kProbeRttDurationMs &&
         round - minimum_congestion_window_start_round_ >=
-            kProbeRttDurationRounds)
+            kProbeRttDurationRounds) {
       EnterProbeBw(now_ms);
     }
   }
+}
 
 void BbrBweSender::TryEnteringRecovery(bool new_round_started) {
-  // If we are already in Recovery don't try to enter.
+  if (mode_ == RECOVERY || !new_round_started || !full_bandwidth_reached_ ||
-  if (mode_ == RECOVERY || !new_round_started || !full_bandwidth_reached_)
+      !min_rtt_filter_->min_rtt_ms())
     return;
   uint64_t increased_rtt_round_counter = 0;
   // If average RTT for past |kPastRttsFilterSize| rounds has been more than
@@ -460,7 +498,8 @@ void BbrBweSender::OnPacketsSent(const Packets& packets) {
       last_packet_sent_sequence_number_ = media_packet->sequence_number();
       // If this is the first packet sent for high gain phase, save data for it.
       if (!first_packet_send_time_during_high_gain_ms_ && pacing_gain_ > 1) {
-        *first_packet_send_time_during_high_gain_ms_ = last_packet_send_time_;
+        first_packet_send_time_during_high_gain_ms_.emplace(
+            last_packet_send_time_);
         data_sent_before_high_gain_started_bytes_ =
             bytes_sent_ - media_packet->payload_size() / 2;
         first_packet_seq_num_during_high_gain_ =
@@ -483,15 +522,31 @@ void BbrBweSender::OnPacketsSent(const Packets& packets) {
 void BbrBweSender::Process() {}
 
 BbrBweReceiver::BbrBweReceiver(int flow_id)
-    : BweReceiver(flow_id, kReceivingRateTimeWindowMs), clock_(0) {}
+    : BweReceiver(flow_id, kReceivingRateTimeWindowMs),
+      clock_(0),
+      packet_feedbacks_(),
+      last_feedback_ms_(0) {}
 
 BbrBweReceiver::~BbrBweReceiver() {}
 
 void BbrBweReceiver::ReceivePacket(int64_t arrival_time_ms,
-                                   const MediaPacket& media_packet) {}
+                                   const MediaPacket& media_packet) {
+  packet_feedbacks_.push_back(media_packet.sequence_number());
+  BweReceiver::ReceivePacket(arrival_time_ms, media_packet);
+}
 
 FeedbackPacket* BbrBweReceiver::GetFeedback(int64_t now_ms) {
-  return nullptr;
+  last_feedback_ms_ = now_ms;
+  int64_t corrected_send_time_ms = 0L;
+  if (!received_packets_.empty()) {
+    PacketIdentifierNode* latest = *(received_packets_.begin());
+    corrected_send_time_ms =
+        latest->send_time_ms + now_ms - latest->arrival_time_ms;
+  }
+  FeedbackPacket* fb = new BbrBweFeedback(
+      flow_id_, now_ms * 1000, corrected_send_time_ms, packet_feedbacks_);
+  packet_feedbacks_.clear();
+  return fb;
 }
 }  // namespace bwe
 }  // namespace testing

webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h

@@ -30,7 +30,7 @@ class MinRttFilter;
 class CongestionWindow;
 class BbrBweSender : public BweSender {
  public:
-  explicit BbrBweSender(Clock* clock);
+  explicit BbrBweSender(BitrateObserver* observer, Clock* clock);
   virtual ~BbrBweSender();
   enum Mode {
     // Startup phase.
@@ -113,8 +113,8 @@ class BbrBweSender : public BweSender {
 
  private:
   void EnterStartup();
-  bool UpdateBandwidthAndMinRtt(int64_t now_ms,
+  void UpdateBandwidthAndMinRtt(int64_t now_ms,
-                                const std::vector<uint64_t>& feedback_vector,
+                                const std::vector<uint16_t>& feedback_vector,
                                 int64_t bytes_acked);
   void TryExitingStartup();
   void TryExitingDrain(int64_t now_ms);
@@ -145,6 +145,7 @@ class BbrBweSender : public BweSender {
   // declare those packets as lost immediately.
   void HandleLoss(uint64_t last_acked_packet, uint64_t recently_acked_packet);
   void AddToPastRtts(int64_t rtt_sample_ms);
+  BitrateObserver* observer_;
   Clock* const clock_;
   Mode mode_;
   std::unique_ptr<MaxBandwidthFilter> max_bandwidth_filter_;
@@ -229,10 +230,10 @@ class BbrBweReceiver : public BweReceiver {
   void ReceivePacket(int64_t arrival_time_ms,
                      const MediaPacket& media_packet) override;
   FeedbackPacket* GetFeedback(int64_t now_ms) override;
-
  private:
   SimulatedClock clock_;
-  std::vector<uint64_t> packet_feedbacks_;
+  std::vector<uint16_t> packet_feedbacks_;
+  int64_t last_feedback_ms_;
 };
 }  // namespace bwe
 }  // namespace testing

webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc

@@ -27,8 +27,6 @@ namespace {
 const int kStartingCongestionWindowBytes = 6000;
 }  // namespace
 
-const int CongestionWindow::kMinimumCongestionWindowBytes;
-
 CongestionWindow::CongestionWindow() : data_inflight_bytes_(0) {}
 
 CongestionWindow::~CongestionWindow() {}
@@ -37,8 +35,6 @@ int CongestionWindow::GetCongestionWindow(BbrBweSender::Mode mode,
                                           int64_t bandwidth_estimate_bps,
                                           rtc::Optional<int64_t> min_rtt_ms,
                                           float gain) {
-  if (mode == BbrBweSender::PROBE_RTT)
-    return CongestionWindow::kMinimumCongestionWindowBytes;
   return GetTargetCongestionWindow(bandwidth_estimate_bps, min_rtt_ms, gain);
 }
 
@@ -47,6 +43,7 @@ void CongestionWindow::PacketSent(size_t sent_packet_size_bytes) {
 }
 
 void CongestionWindow::AckReceived(size_t received_packet_size_bytes) {
+  RTC_DCHECK_GE(data_inflight_bytes_ >= received_packet_size_bytes, true);
   data_inflight_bytes_ -= received_packet_size_bytes;
 }
 
@@ -57,14 +54,13 @@ int CongestionWindow::GetTargetCongestionWindow(
   // If we have no rtt sample yet, return the starting congestion window size.
   if (!min_rtt_ms)
     return gain * kStartingCongestionWindowBytes;
-  int bdp = *min_rtt_ms * bandwidth_estimate_bps;
+  int bdp = *min_rtt_ms * bandwidth_estimate_bps / 8000;
   int congestion_window = bdp * gain;
   // Congestion window could be zero in rare cases, when either no bandwidth
   // estimate is available, or path's min_rtt value is zero.
   if (!congestion_window)
     congestion_window = gain * kStartingCongestionWindowBytes;
-  return std::max(congestion_window,
+  return congestion_window;
-                  CongestionWindow::kMinimumCongestionWindowBytes);
 }
 }  // namespace bwe
 }  // namespace testing

webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h

@@ -21,10 +21,6 @@ namespace testing {
 namespace bwe {
 class CongestionWindow {
  public:
-  // Size of congestion window while in PROBE_RTT mode, suggested by BBR's
-  // source code of QUIC's implementation.
-  static const int kMinimumCongestionWindowBytes = 4000;
-
   CongestionWindow();
   ~CongestionWindow();
   int GetCongestionWindow(BbrBweSender::Mode mode,

webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window_unittest.cc

@@ -17,9 +17,8 @@ namespace webrtc {
 namespace testing {
 namespace bwe {
 namespace {
-// These are the same values used in CongestionWindow class.
+// Same value used in CongestionWindow class.
 const int64_t kStartingCongestionWindow = 6000;
-const int64_t kMinimumCongestionWindow = 4000;
 }  // namespace
 
 TEST(CongestionWindowTest, InitializationCheck) {
@@ -51,33 +50,13 @@ TEST(CongestionWindowTest, ZeroBandwidthDelayProduct) {
   EXPECT_EQ(target_congestion_window, 2.885f * kStartingCongestionWindow);
 }
 
-TEST(CongestionWindowTest, BelowMinimumTargetCongestionWindow) {
-  CongestionWindow congestion_window;
-  int64_t target_congestion_window =
-      congestion_window.GetTargetCongestionWindow(
-          100, rtc::Optional<int64_t>(2), 2.885f);
-  EXPECT_EQ(target_congestion_window, kMinimumCongestionWindow);
-}
-
-TEST(CongestionWindowTest, AboveMinimumTargetCongestionWindow) {
-  CongestionWindow congestion_window;
-  int64_t target_congestion_window =
-      congestion_window.GetTargetCongestionWindow(
-          100000, rtc::Optional<int64_t>(2), 2.885f);
-  EXPECT_EQ(target_congestion_window, 577000);
-}
-
-TEST(CongestionWindowTest, MinimumCongestionWindow) {
-  CongestionWindow congestion_window;
-  int64_t cwnd = congestion_window.GetCongestionWindow(
-      BbrBweSender::PROBE_RTT, 100, rtc::Optional<int64_t>(100), 2.885f);
-  EXPECT_EQ(cwnd, kMinimumCongestionWindow);
-}
-
 TEST(CongestionWindowTest, CalculateCongestionWindow) {
   CongestionWindow congestion_window;
   int64_t cwnd = congestion_window.GetCongestionWindow(
-      BbrBweSender::STARTUP, 100, rtc::Optional<int64_t>(100l), 2.885f);
+      BbrBweSender::STARTUP, 800000, rtc::Optional<int64_t>(100l), 2.885f);
+  EXPECT_EQ(cwnd, 28850);
+  cwnd = congestion_window.GetCongestionWindow(
+      BbrBweSender::STARTUP, 400000, rtc::Optional<int64_t>(200l), 2.885f);
   EXPECT_EQ(cwnd, 28850);
 }
 }  // namespace bwe

webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h

@@ -14,6 +14,7 @@
 
 #include <cstdint>
 #include <limits>
+#include <list>
 
 #include "webrtc/rtc_base/optional.h"
 
@@ -21,12 +22,17 @@ namespace webrtc {
 namespace testing {
 namespace bwe {
 
-// Expiration time for min_rtt sample, which is set to 10 seconds according to
+// Average rtt for past |kRttFilterSize| packets should grow by
-// BBR design doc.
+// |kRttIncreaseThresholdForExpiry| in order to enter PROBE_RTT mode and expire.
-const int64_t kMinRttFilterSizeMs = 10000;
+// old min_rtt estimate.
+const float kRttIncreaseThresholdForExpiry = 2.3f;
+const size_t kRttFilterSize = 25;
 
 class MinRttFilter {
  public:
+  // This class implements a simple filter to ensure that PROBE_RTT is only
+  // entered when RTTs start to increase, instead of fixed 10 second window as
+  // in orginal BBR design doc, to avoid unnecessary freezes in stream.
   MinRttFilter() {}
   ~MinRttFilter() {}
 
@@ -34,20 +40,31 @@ class MinRttFilter {
   void AddRttSample(int64_t rtt_ms, int64_t now_ms) {
     if (!min_rtt_ms_ || rtt_ms <= *min_rtt_ms_ || MinRttExpired(now_ms)) {
       min_rtt_ms_.emplace(rtt_ms);
-      discovery_time_ms_ = now_ms;
     }
+    rtt_samples_.push_back(rtt_ms);
+    if (rtt_samples_.size() > kRttFilterSize)
+      rtt_samples_.pop_front();
   }
-  int64_t discovery_time() { return discovery_time_ms_; }
 
-  // Checks whether or not last discovered min_rtt value is older than x
+  // Checks whether or not last RTT values for past |kRttFilterSize| packets
-  // milliseconds.
+  // started to increase, meaning we have to update min_rtt estimate.
   bool MinRttExpired(int64_t now_ms) {
-    return now_ms - discovery_time_ms_ >= kMinRttFilterSizeMs;
+    if (rtt_samples_.size() < kRttFilterSize || !min_rtt_ms_)
+      return false;
+    int64_t sum_of_rtts_ms = 0;
+    for (int64_t i : rtt_samples_)
+      sum_of_rtts_ms += i;
+    if (sum_of_rtts_ms >=
+        *min_rtt_ms_ * kRttIncreaseThresholdForExpiry * kRttFilterSize) {
+      rtt_samples_.clear();
+      return true;
+    }
+    return false;
   }
 
  private:
   rtc::Optional<int64_t> min_rtt_ms_;
-  int64_t discovery_time_ms_ = 0;
+  std::list<int64_t> rtt_samples_;
 };
 }  // namespace bwe
 }  // namespace testing

webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter_unittest.cc

@@ -18,28 +18,24 @@ namespace bwe {
 TEST(MinRttFilterTest, InitializationCheck) {
   MinRttFilter min_rtt_filter;
   EXPECT_FALSE(min_rtt_filter.min_rtt_ms());
-  EXPECT_EQ(min_rtt_filter.discovery_time(), 0);
 }
 
 TEST(MinRttFilterTest, AddRttSample) {
   MinRttFilter min_rtt_filter;
   min_rtt_filter.AddRttSample(120, 5);
-  EXPECT_EQ(min_rtt_filter.min_rtt_ms(), 120);
+  EXPECT_EQ(*min_rtt_filter.min_rtt_ms(), 120);
-  EXPECT_EQ(min_rtt_filter.discovery_time(), 5);
   min_rtt_filter.AddRttSample(121, 6);
-  EXPECT_EQ(min_rtt_filter.discovery_time(), 5);
   min_rtt_filter.AddRttSample(119, 7);
-  EXPECT_EQ(min_rtt_filter.discovery_time(), 7);
   min_rtt_filter.AddRttSample(140, 10007);
-  EXPECT_EQ(min_rtt_filter.discovery_time(), 10007);
+  EXPECT_EQ(*min_rtt_filter.min_rtt_ms(), 119);
-  EXPECT_EQ(min_rtt_filter.min_rtt_ms(), 140);
 }
 
 TEST(MinRttFilterTest, MinRttExpired) {
   MinRttFilter min_rtt_filter;
-  min_rtt_filter.AddRttSample(120, 5);
+  for (int i = 1; i <= 25; i++)
-  EXPECT_EQ(min_rtt_filter.MinRttExpired(10006), true);
+    min_rtt_filter.AddRttSample(i, i);
-  EXPECT_EQ(min_rtt_filter.MinRttExpired(10), false);
+  EXPECT_EQ(min_rtt_filter.MinRttExpired(25), true);
+  EXPECT_EQ(min_rtt_filter.MinRttExpired(24), false);
 }
 }  // namespace bwe
 }  // namespace testing

webrtc/modules/remote_bitrate_estimator/test/packet.h

@@ -114,15 +114,15 @@ class BbrBweFeedback : public FeedbackPacket {
   BbrBweFeedback(int flow_id,
                  int64_t send_time_us,
                  int64_t latest_send_time_ms,
-                 const std::vector<uint64_t>& packet_feedback_vector);
+                 const std::vector<uint16_t>& packet_feedback_vector);
   virtual ~BbrBweFeedback() {}
 
-  const std::vector<uint64_t>& packet_feedback_vector() const {
+  const std::vector<uint16_t>& packet_feedback_vector() const {
     return packet_feedback_vector_;
   }
 
  private:
-  const std::vector<uint64_t> packet_feedback_vector_;
+  const std::vector<uint16_t> packet_feedback_vector_;
 };
 
 class RembFeedback : public FeedbackPacket {

webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc

@@ -15,6 +15,8 @@
 #include <sstream>
 
 #include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/modules/pacing/pacer.h"
+#include "webrtc/modules/remote_bitrate_estimator/test/bbr_paced_sender.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/bwe.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h"
 #include "webrtc/rtc_base/checks.h"
@@ -156,9 +158,12 @@ uint32_t VideoSender::TargetBitrateKbps() {
 PacedVideoSender::PacedVideoSender(PacketProcessorListener* listener,
                                    VideoSource* source,
                                    BandwidthEstimatorType estimator)
-    : VideoSender(listener, source, estimator), pacer_(&clock_, this, nullptr) {
+    : VideoSender(listener, source, estimator),
-  modules_.push_back(&pacer_);
+      // Ugly hack to use BBR's pacer.
-  pacer_.SetEstimatedBitrate(source->bits_per_second());
+      // TODO(gnish): Make pacer choice dependant on the algorithm being used.
+      pacer_(new BbrPacedSender(&clock_, this, nullptr)) {
+  modules_.push_back(pacer_.get());
+  pacer_->SetEstimatedBitrate(source->bits_per_second());
 }
 
 PacedVideoSender::~PacedVideoSender() {
@@ -204,10 +209,11 @@ void PacedVideoSender::RunFor(int64_t time_ms, Packets* in_out) {
     if (!generated_packets.empty()) {
       for (Packet* packet : generated_packets) {
         MediaPacket* media_packet = static_cast<MediaPacket*>(packet);
-        pacer_.InsertPacket(
+        pacer_->InsertPacket(
             PacedSender::kNormalPriority, media_packet->header().ssrc,
             media_packet->header().sequenceNumber, media_packet->send_time_ms(),
             media_packet->payload_size(), false);
+        pacer_queue_size_in_bytes_ += media_packet->payload_size();
         pacer_queue_.push_back(packet);
         assert(pacer_queue_.size() < 10000);
       }
@@ -284,11 +290,11 @@ bool PacedVideoSender::TimeToSendPacket(uint32_t ssrc,
       // Make sure a packet is never paced out earlier than when it was put into
       // the pacer.
       assert(pace_out_time_ms >= media_packet->send_time_ms());
-
       media_packet->SetAbsSendTimeMs(pace_out_time_ms);
       media_packet->set_send_time_us(1000 * pace_out_time_ms);
       media_packet->set_sender_timestamp_us(1000 * pace_out_time_ms);
       queue_.push_back(media_packet);
+      pacer_queue_size_in_bytes_ -= media_packet->payload_size();
       pacer_queue_.erase(it);
       return true;
     }
@@ -305,7 +311,21 @@ void PacedVideoSender::OnNetworkChanged(uint32_t target_bitrate_bps,
                                         uint8_t fraction_lost,
                                         int64_t rtt) {
   VideoSender::OnNetworkChanged(target_bitrate_bps, fraction_lost, rtt);
-  pacer_.SetEstimatedBitrate(target_bitrate_bps);
+  pacer_->SetEstimatedBitrate(target_bitrate_bps);
+}
+
+void PacedVideoSender::OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
+                                        uint32_t bitrate_for_pacer_bps,
+                                        bool in_probe_rtt,
+                                        int64_t target_set_time,
+                                        uint64_t congestion_window) {
+  VideoSender::OnNetworkChanged(bitrate_for_encoder_bps, 0u, 0u);
+  pacer_->SetEstimatedBitrateAndCongestionWindow(
+      bitrate_for_pacer_bps, in_probe_rtt, congestion_window);
+}
+
+void PacedVideoSender::OnBytesAcked(size_t bytes) {
+  pacer_->OnBytesAcked(bytes);
 }
 
 const int kNoLimit = std::numeric_limits<int>::max();

webrtc/modules/remote_bitrate_estimator/test/packet_sender.h

@@ -81,7 +81,6 @@ class VideoSender : public PacketSender, public BitrateObserver {
   void OnNetworkChanged(uint32_t target_bitrate_bps,
                         uint8_t fraction_lost,
                         int64_t rtt) override;
-
   void Pause() override;
   void Resume(int64_t paused_time_ms) override;
 
@@ -123,12 +122,23 @@ class PacedVideoSender : public VideoSender, public PacedSender::PacketSender {
                         uint8_t fraction_lost,
                         int64_t rtt) override;
 
+  void OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
+                        uint32_t bitrate_for_pacer_bps,
+                        bool in_probe_rtt,
+                        int64_t rtt,
+                        uint64_t congestion_window) override;
+  size_t pacer_queue_size_in_bytes() override {
+    return pacer_queue_size_in_bytes_;
+  }
+  void OnBytesAcked(size_t bytes) override;
+
  private:
   int64_t TimeUntilNextProcess(const std::list<Module*>& modules);
   void CallProcess(const std::list<Module*>& modules);
   void QueuePackets(Packets* batch, int64_t end_of_batch_time_us);
 
-  PacedSender pacer_;
+  size_t pacer_queue_size_in_bytes_ = 0;
+  std::unique_ptr<Pacer> pacer_;
   Packets queue_;
   Packets pacer_queue_;
 

webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py

@@ -78,7 +78,6 @@ class Figure(object):
       axis = fig.add_subplot(n, 1, i+1)
       self.subplots[i].PlotSubplot(axis)
 
-
 class Subplot(object):
   def __init__(self, var_names, xlabel, ylabel):
     self.xlabel = xlabel
@@ -111,9 +110,11 @@ class Subplot(object):
           y = [sample[1] for sample in self.samples[alg_name][ssrc][var_name]]
           x = numpy.array(x)
           y = numpy.array(y)
-
           ssrc_count = len(self.samples[alg_name].keys())
           l = GenerateLabel(var_name, ssrc, ssrc_count, alg_name)
+          if l == 'MaxThroughput_':
+            plt.plot(x, y, label=l, linewidth=4.0)
+          else:
             plt.plot(x, y, label=l, linewidth=2.0)
           count += 1
 
@@ -148,8 +149,12 @@ def main():
   target_bitrate.AddSubplot(['target_bitrate_bps', 'acknowledged_bitrate'],
                             "Time (s)", "Bitrate (bps)")
 
+  min_rtt_state = Figure("MinRttState")
+  min_rtt_state.AddSubplot(['MinRtt'], "Time (s)", "Time (ms)")
+
   # Select which figures to plot here.
-  figures = [receiver, detector_state, trendline_state, target_bitrate]
+  figures = [receiver, detector_state, trendline_state, target_bitrate,
+    min_rtt_state]
 
   # Add samples to the figures.
   for line in sys.stdin: