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}
diff --git a/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/webrtc/modules/bitrate_controller/include/bitrate_controller.h
index c6695e9..f479e52 100644
--- a/webrtc/modules/bitrate_controller/include/bitrate_controller.h
+++ b/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 @@
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,
- public RtcpBandwidthObserver {
+class BitrateController : public Module, 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
diff --git a/webrtc/modules/pacing/BUILD.gn b/webrtc/modules/pacing/BUILD.gn
index b22c178..bc9d144 100644
--- a/webrtc/modules/pacing/BUILD.gn
+++ b/webrtc/modules/pacing/BUILD.gn
@@ -18,6 +18,7 @@
"interval_budget.h",
"paced_sender.cc",
"paced_sender.h",
+ "pacer.h",
"packet_router.cc",
"packet_router.h",
]
diff --git a/webrtc/modules/pacing/paced_sender.cc b/webrtc/modules/pacing/paced_sender.cc
index 74a885a..d0a8864 100644
--- a/webrtc/modules/pacing/paced_sender.cc
+++ b/webrtc/modules/pacing/paced_sender.cc
@@ -37,8 +37,8 @@
} // namespace
-// TODO(sprang): Move at least PacketQueue out to separate
-// files, so that we can more easily test them.
+// TODO(sprang): Move at least PacketQueue out to separate files, so that we can
+// more easily test them.
namespace webrtc {
namespace paced_sender {
diff --git a/webrtc/modules/pacing/paced_sender.h b/webrtc/modules/pacing/paced_sender.h
index a1f7ebe..74b8780 100644
--- a/webrtc/modules/pacing/paced_sender.h
+++ b/webrtc/modules/pacing/paced_sender.h
@@ -15,8 +15,7 @@
#include <memory>
#include <set>
-#include "webrtc/modules/include/module.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "webrtc/modules/pacing/pacer.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 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 @@
// |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 @@
// 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);
diff --git a/webrtc/modules/pacing/pacer.h b/webrtc/modules/pacing/pacer.h
new file mode 100644
index 0000000..b5ac2ec
--- /dev/null
+++ b/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_
diff --git a/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
index 8a48077..71487a5 100644
--- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn
+++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
@@ -61,6 +61,8 @@
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",
diff --git a/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc b/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
index b3cb98d..92d5a55 100644
--- a/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
+++ b/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
@@ -122,9 +122,64 @@
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, 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");
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bbr_paced_sender.cc b/webrtc/modules/remote_bitrate_estimator/test/bbr_paced_sender.cc
new file mode 100644
index 0000000..3df97aa
--- /dev/null
+++ b/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
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bbr_paced_sender.h b/webrtc/modules/remote_bitrate_estimator/test/bbr_paced_sender.h
new file mode 100644
index 0000000..f5ddaef
--- /dev/null
+++ b/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_
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
index 755e59d..abc301c 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
@@ -95,7 +95,7 @@
case kNadaEstimator:
return new NadaBweSender(kbps, observer, clock);
case kBbrEstimator:
- return new BbrBweSender(clock);
+ return new BbrBweSender(observer, clock);
case kTcpEstimator:
FALLTHROUGH();
case kNullEstimator:
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
index edaeffc..51c1f55 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
@@ -157,7 +157,7 @@
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 @@
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);
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.cc
index a45700b..64e6bf2 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.cc
@@ -34,7 +34,7 @@
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 @@
// 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 @@
// 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 @@
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 @@
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 @@
bytes_acked_ - packet->payload_size_bytes / 2;
high_gain_over_ = true;
}
- // Notify pacer that an ack was received, to adjust data inflight.
- // TODO(gnish): Add implementation for BitrateObserver class, to notify
- // pacer about incoming acks.
+ observer_->OnBytesAcked(packet->payload_size_bytes);
congestion_window_->AckReceived(packet->payload_size_bytes);
HandleLoss(last_packet_acked_sequence_number_, packet->sequence_number);
last_packet_acked_sequence_number_ = packet->sequence_number;
@@ -196,9 +212,8 @@
round_trip_end_ = last_packet_sent_sequence_number_;
}
}
- bool min_rtt_expired = false;
- min_rtt_expired =
- UpdateBandwidthAndMinRtt(now_ms, feedback_vector, bytes_acked_);
+ TryEnteringProbeRtt(now_ms);
+ UpdateBandwidthAndMinRtt(now_ms, feedback_vector, bytes_acked_);
if (new_round_started && !full_bandwidth_reached_) {
full_bandwidth_reached_ = max_bandwidth_filter_->FullBandwidthReached(
kStartupGrowthTarget, kMaxRoundsWithoutGrowth);
@@ -221,19 +236,37 @@
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
- // rate for the pacer and the encoder.
+ int64_t rate_for_pacer_bps = pacing_rate_bps_;
+ 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 @@
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 @@
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 @@
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)
- return false;
- min_rtt_filter_->AddRttSample(*min_rtt_sample_ms, now_ms);
- bool min_rtt_expired = min_rtt_filter_->MinRttExpired(now_ms);
- return min_rtt_expired;
+ // We only feed RTT samples into the min_rtt filter which were not produced
+ // 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);
}
void BbrBweSender::EnterStartup() {
@@ -365,8 +400,9 @@
// 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() <
- TargetCongestionWindow(pacing_gain_))
+ if (pacing_gain_ > 1.0 &&
+ 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 @@
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::TryExitingProbeRtt(int64_t now_ms, int64_t round) {
if (!minimum_congestion_window_start_time_ms_) {
if (congestion_window_->data_inflight() <=
- CongestionWindow::kMinimumCongestionWindowBytes) {
- *minimum_congestion_window_start_time_ms_ = now_ms;
+ TargetCongestionWindow(kProbeRttCongestionWindowGain)) {
+ 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 @@
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::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
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h
index 63dc78b..6f4d831 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h
@@ -30,7 +30,7 @@
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 @@
private:
void EnterStartup();
- bool UpdateBandwidthAndMinRtt(int64_t now_ms,
- const std::vector<uint64_t>& feedback_vector,
+ void UpdateBandwidthAndMinRtt(int64_t now_ms,
+ const std::vector<uint16_t>& feedback_vector,
int64_t bytes_acked);
void TryExitingStartup();
void TryExitingDrain(int64_t now_ms);
@@ -145,6 +145,7 @@
// 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 @@
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
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc
index 3d35770..5c01e89 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc
@@ -27,8 +27,6 @@
const int kStartingCongestionWindowBytes = 6000;
} // namespace
-const int CongestionWindow::kMinimumCongestionWindowBytes;
-
CongestionWindow::CongestionWindow() : data_inflight_bytes_(0) {}
CongestionWindow::~CongestionWindow() {}
@@ -37,8 +35,6 @@
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::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 @@
// 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,
- CongestionWindow::kMinimumCongestionWindowBytes);
+ return congestion_window;
}
} // namespace bwe
} // namespace testing
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
index b9b2300..105a748 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
@@ -21,10 +21,6 @@
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,
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window_unittest.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window_unittest.cc
index 0c6d59c..2415c13 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window_unittest.cc
@@ -17,9 +17,8 @@
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 @@
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
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h b/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
index b4932e5..f45437e 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
+++ b/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 testing {
namespace bwe {
-// Expiration time for min_rtt sample, which is set to 10 seconds according to
-// BBR design doc.
-const int64_t kMinRttFilterSizeMs = 10000;
+// Average rtt for past |kRttFilterSize| packets should grow by
+// |kRttIncreaseThresholdForExpiry| in order to enter PROBE_RTT mode and expire.
+// 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 @@
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
- // milliseconds.
+ // Checks whether or not last RTT values for past |kRttFilterSize| packets
+ // 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
diff --git a/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter_unittest.cc b/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter_unittest.cc
index a8e76e1..5cf08d4 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter_unittest.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter_unittest.cc
@@ -18,28 +18,24 @@
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.discovery_time(), 5);
+ EXPECT_EQ(*min_rtt_filter.min_rtt_ms(), 120);
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(), 140);
+ EXPECT_EQ(*min_rtt_filter.min_rtt_ms(), 119);
}
TEST(MinRttFilterTest, MinRttExpired) {
MinRttFilter min_rtt_filter;
- min_rtt_filter.AddRttSample(120, 5);
- EXPECT_EQ(min_rtt_filter.MinRttExpired(10006), true);
- EXPECT_EQ(min_rtt_filter.MinRttExpired(10), false);
+ for (int i = 1; i <= 25; i++)
+ min_rtt_filter.AddRttSample(i, i);
+ EXPECT_EQ(min_rtt_filter.MinRttExpired(25), true);
+ EXPECT_EQ(min_rtt_filter.MinRttExpired(24), false);
}
} // namespace bwe
} // namespace testing
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet.h b/webrtc/modules/remote_bitrate_estimator/test/packet.h
index 525397c..05e1267 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet.h
@@ -114,15 +114,15 @@
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 {
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
index de4d93d..33209ca 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
+++ b/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 @@
PacedVideoSender::PacedVideoSender(PacketProcessorListener* listener,
VideoSource* source,
BandwidthEstimatorType estimator)
- : VideoSender(listener, source, estimator), pacer_(&clock_, this, nullptr) {
- modules_.push_back(&pacer_);
- pacer_.SetEstimatedBitrate(source->bits_per_second());
+ : VideoSender(listener, source, estimator),
+ // Ugly hack to use BBR's pacer.
+ // 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 @@
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 @@
// 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 @@
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();
diff --git a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
index 0df61b0..86ad0e8 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
+++ b/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
@@ -81,7 +81,6 @@
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 @@
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_;
diff --git a/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py b/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py
index 51aca1c..02b2052 100755
--- a/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py
+++ b/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py
@@ -78,7 +78,6 @@
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
@@ -111,10 +110,12 @@
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)
- plt.plot(x, y, label=l, linewidth=2.0)
+ if l == 'MaxThroughput_':
+ plt.plot(x, y, label=l, linewidth=4.0)
+ else:
+ plt.plot(x, y, label=l, linewidth=2.0)
count += 1
plt.grid(True)
@@ -148,8 +149,12 @@
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: