Backport changes to congestion control from libwebrtc

node/src/Transport.ts

@@ -56,7 +56,40 @@ export type TransportTuple =
 /**
  * Valid types for 'trace' event.
  */
-export type TransportTraceEventType = 'probation' | 'bwe';
+export type TransportTraceEventType = 'probation' | 'bwe' | 'bweStats';
+
+/**
+ * bweStats typings for trace event.
+ */
+export type TransportTraceEventBweStatsInfo = {
+	estimatedBitrate: number;
+	delay: {
+		slope: number;
+		rSquared: number;
+		threshold: number;
+		rtt: number;
+		rateControlState: number;
+		delayDetectorState: number;
+	};
+	probe: {
+		estimatedBitrate: number;
+	};
+	loss: {
+		inherent: number;
+		avg: number;
+		estimatedBitrate: number;
+		sendingRate: number;
+	};
+	alr: boolean;
+	ackBitrate: number;
+	desiredBitrate: number;
+	effectiveDesiredBitrate: number;
+	minBitrate: number;
+	maxBitrate: number;
+	startBitrate: number;
+	maxPaddingBitrate: number;
+	sendingRate: number;
+};
 
 /**
  * 'trace' event data.
@@ -81,7 +114,7 @@ export type TransportTraceEventData =
 	/**
 	 * Per type information.
 	 */
-	info: any;
+	info: TransportTraceEventBweStatsInfo | any;
 };
 
 export type SctpState = 'new' | 'connecting' | 'connected' | 'failed' | 'closed';

node/src/Worker.ts

@@ -63,16 +63,17 @@ export type WorkerSettings =
 	 * certificate is dynamically created.
 	 */
 	dtlsPrivateKeyFile?: string;
-
+	/* eslint-disable max-len */
 	/**
 	 * Field trials for libwebrtc.
 	 * @private
 	 *
 	 * NOTE: For advanced users only. An invalid value will make the worker crash.
 	 * Default value is
-	 * "WebRTC-Bwe-AlrLimitedBackoff/Enabled/".
+	 * "WebRTC-Bwe-AlrLimitedBackoff/Enabled/WebRTC-Bwe-LossBasedBweV2/BwRampupUpperBoundFactor:1.1,TrendlineIntegrationEnabled:true/WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled/"
 	 */
 	libwebrtcFieldTrials?: string;
+	/* eslint-enable max-len */
 
 	/**
 	 * Custom application data.

rust/src/worker.rs

@@ -186,7 +186,7 @@ pub struct WorkerSettings {
     ///
     /// NOTE: For advanced users only. An invalid value will make the worker crash.
     /// Default value is
-    /// "WebRTC-Bwe-AlrLimitedBackoff/Enabled/".
+    /// "WebRTC-Bwe-AlrLimitedBackoff/Enabled/WebRTC-Bwe-LossBasedBweV2/BwRampupUpperBoundFactor:1.1,TrendlineIntegrationEnabled:true/WebRTC-Bwe-RobustThroughputEstimatorSettings/enabled/"
     #[doc(hidden)]
     pub libwebrtc_field_trials: Option<String>,
     /// Function that will be called under worker thread before worker starts, can be used for

worker/deps/libwebrtc/libwebrtc.gyp

@@ -52,7 +52,6 @@
       'libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.cc',
       'libwebrtc/modules/congestion_controller/goog_cc/alr_detector.cc',
       'libwebrtc/modules/congestion_controller/goog_cc/probe_controller.cc',
-      'libwebrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc',
       'libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc',
       'libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc',
       'libwebrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc',
@@ -110,7 +109,6 @@
       'libwebrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h',
       'libwebrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h',
       'libwebrtc/modules/congestion_controller/goog_cc/link_capacity_estimator.h',
-      'libwebrtc/modules/congestion_controller/goog_cc/median_slope_estimator.h',
       'libwebrtc/modules/congestion_controller/goog_cc/probe_controller.h',
       'libwebrtc/modules/congestion_controller/goog_cc/trendline_estimator.h',
       'libwebrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h',

worker/deps/libwebrtc/libwebrtc/api/transport/network_control.h

@@ -32,6 +32,13 @@ class TargetTransferRateObserver {
   virtual void OnStartRateUpdate(DataRate) {}
 };
 
+class BweStatsTracer {
+public:
+	virtual ~BweStatsTracer() = default;
+	virtual void OnBweStats(BweStats) = 0;
+};
+
+
 // Configuration sent to factory create function. The parameters here are
 // optional to use for a network controller implementation.
 struct NetworkControllerConfig {
@@ -72,7 +79,11 @@ class NetworkControllerInterface {
   // Called round trip time has been calculated by protocol specific mechanisms.
   virtual NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate) = 0;
   // Called when a packet is sent on the network.
-  virtual NetworkControlUpdate OnSentPacket(SentPacket) = 0;
+  virtual NetworkControlUpdate OnSentPacket(
+      SentPacket) = 0;
+  // Called when a packet is received from the remote client.
+  virtual NetworkControlUpdate OnReceivedPacket(
+      ReceivedPacket) = 0;
   // Called when the stream specific configuration has been updated.
   virtual NetworkControlUpdate OnStreamsConfig(StreamsConfig) = 0;
   // Called when target transfer rate constraints has been changed.
@@ -85,6 +96,7 @@ class NetworkControllerInterface {
       TransportPacketsFeedback) = 0;
   // Called with network state estimate updates.
   virtual NetworkControlUpdate OnNetworkStateEstimate(NetworkStateEstimate) = 0;
+	virtual BweStats GetBweStats() = 0;
 };
 
 // NetworkControllerFactoryInterface is an interface for creating a network
@@ -108,7 +120,11 @@ class NetworkStateEstimator {
   // Gets the current best estimate according to the estimator.
   virtual absl::optional<NetworkStateEstimate> GetCurrentEstimate() = 0;
   // Called with per packet feedback regarding receive time.
+  // Used when the NetworkStateEstimator runs in the sending endpoint.
   virtual void OnTransportPacketsFeedback(const TransportPacketsFeedback&) = 0;
+  // Called with per packet feedback regarding receive time.
+  // Used when the NetworkStateEstimator runs in the receiving endpoint.
+  virtual void OnReceivedPacket(const PacketResult&) {}
   // Called when the receiving or sending endpoint changes address.
   virtual void OnRouteChange(const NetworkRouteChange&) = 0;
   virtual ~NetworkStateEstimator() = default;

worker/deps/libwebrtc/libwebrtc/api/transport/network_types.cc

@@ -13,11 +13,14 @@
 #include <algorithm>
 
 namespace webrtc {
-// TODO(srte): Revert to using default after removing union member.
-StreamsConfig::StreamsConfig() {}
+StreamsConfig::StreamsConfig() = default;
 StreamsConfig::StreamsConfig(const StreamsConfig&) = default;
 StreamsConfig::~StreamsConfig() = default;
 
+BweStats::BweStats() = default;
+BweStats::BweStats(const BweStats&) = default;
+BweStats::~BweStats() = default;
+
 TargetRateConstraints::TargetRateConstraints() = default;
 TargetRateConstraints::TargetRateConstraints(const TargetRateConstraints&) =
     default;
@@ -49,7 +52,7 @@ std::vector<PacketResult> TransportPacketsFeedback::ReceivedWithSendInfo()
     const {
   std::vector<PacketResult> res;
   for (const PacketResult& fb : packet_feedbacks) {
-    if (fb.receive_time.IsFinite()) {
+    if (fb.IsReceived()) {
       res.push_back(fb);
     }
   }
@@ -59,7 +62,7 @@ std::vector<PacketResult> TransportPacketsFeedback::ReceivedWithSendInfo()
 std::vector<PacketResult> TransportPacketsFeedback::LostWithSendInfo() const {
   std::vector<PacketResult> res;
   for (const PacketResult& fb : packet_feedbacks) {
-    if (fb.receive_time.IsPlusInfinity()) {
+    if (!fb.IsReceived()) {
       res.push_back(fb);
     }
   }
@@ -75,7 +78,7 @@ std::vector<PacketResult> TransportPacketsFeedback::SortedByReceiveTime()
     const {
   std::vector<PacketResult> res;
   for (const PacketResult& fb : packet_feedbacks) {
-    if (fb.receive_time.IsFinite()) {
+    if (fb.IsReceived()) {
       res.push_back(fb);
     }
   }

worker/deps/libwebrtc/libwebrtc/api/transport/network_types.h

@@ -15,6 +15,8 @@
 #include "api/units/data_size.h"
 #include "api/units/time_delta.h"
 #include "api/units/timestamp.h"
+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
 
 #include <absl/types/optional.h>
 #include <stdint.h>
@@ -24,6 +26,50 @@ namespace webrtc {
 
 // Configuration
 
+// Represents constraints and rates related to the currently enabled streams.
+// This is used as input to the congestion controller via the StreamsConfig
+// struct.
+struct BitrateAllocationLimits {
+  // The total minimum send bitrate required by all sending streams.
+  DataRate min_allocatable_rate = DataRate::Zero();
+  // The total maximum allocatable bitrate for all currently available streams.
+  DataRate max_allocatable_rate = DataRate::Zero();
+  // The max bitrate to use for padding. The sum of the per-stream max padding
+  // rate.
+  DataRate max_padding_rate = DataRate::Zero();
+};
+
+struct LossEstimatorState {
+	~LossEstimatorState() = default;
+	absl::optional<DataRate> bandwidth_estimate = DataRate::Zero();
+	double inherent_loss;
+	double avg_loss;
+	absl::optional<DataRate> sending_rate = DataRate::Zero();
+};
+
+struct DelayBasedBweState {
+	RateControlState rate_control_state;
+	DelayIncreaseDetectorInterface::RegressionResult trend;
+	BandwidthUsage delay_detector_state;
+	double threshold;
+};
+
+struct BweStats {
+	BweStats();
+	BweStats(const BweStats&);
+	~BweStats();
+	Timestamp time = Timestamp::PlusInfinity();
+	absl::optional<DataRate> estimated_bitrate;
+	absl::optional<DataRate> acknowledged_bitrate;
+	DelayBasedBweState delay;
+	TimeDelta rtt = TimeDelta::ms(0);
+	bool in_alr = false;
+	LossEstimatorState loss_estimator_state;
+	absl::optional<DataRate> probe_bitrate;
+};
+
+
+
 // Use StreamsConfig for information about streams that is required for specific
 // adjustments to the algorithms in network controllers. Especially useful
 // for experiments.
@@ -34,9 +80,9 @@ struct StreamsConfig {
   Timestamp at_time = Timestamp::PlusInfinity();
   absl::optional<bool> requests_alr_probing;
   absl::optional<double> pacing_factor;
-  union {
-    absl::optional<DataRate> min_total_allocated_bitrate = absl::nullopt;
-  };
+
+  // TODO: (srte) Use BitrateAllocationLimits here.
+  absl::optional<DataRate> min_total_allocated_bitrate;
   absl::optional<DataRate> max_padding_rate;
   absl::optional<DataRate> max_total_allocated_bitrate;
 };
@@ -84,13 +130,20 @@ struct PacedPacketInfo {
   int probe_cluster_id = kNotAProbe;
   int probe_cluster_min_probes = -1;
   int probe_cluster_min_bytes = -1;
+  int probe_cluster_bytes_sent = 0;
 };
 
 struct SentPacket {
   Timestamp send_time = Timestamp::PlusInfinity();
+  // Size of packet with overhead up to IP layer.
   DataSize size = DataSize::Zero();
+  // Size of preceeding packets that are not part of feedback.
   DataSize prior_unacked_data = DataSize::Zero();
+  // Probe cluster id and parameters including bitrate, number of packets and
+  // number of bytes.
   PacedPacketInfo pacing_info;
+  // True if the packet is an audio packet, false for video, padding, RTX etc.
+  bool audio = false;
   // Transport independent sequence number, any tracked packet should have a
   // sequence number that is unique over the whole call and increasing by 1 for
   // each packet.
@@ -138,6 +191,8 @@ struct PacketResult {
   PacketResult(const PacketResult&);
   ~PacketResult();
 
+  inline bool IsReceived() const { return !receive_time.IsPlusInfinity(); }
+
   SentPacket sent_packet;
   Timestamp receive_time = Timestamp::PlusInfinity();
 };
@@ -166,6 +221,7 @@ struct TransportPacketsFeedback {
 
 struct NetworkEstimate {
   Timestamp at_time = Timestamp::PlusInfinity();
+  // Deprecated, use TargetTransferRate::target_rate instead.
   DataRate bandwidth = DataRate::Infinity();
   TimeDelta round_trip_time = TimeDelta::PlusInfinity();
   TimeDelta bwe_period = TimeDelta::PlusInfinity();
@@ -199,6 +255,8 @@ struct TargetTransferRate {
   // The estimate on which the target rate is based on.
   NetworkEstimate network_estimate;
   DataRate target_rate = DataRate::Zero();
+  DataRate stable_target_rate = DataRate::Zero();
+  double cwnd_reduce_ratio = 0;
 };
 
 // Contains updates of network controller comand state. Using optionals to

worker/deps/libwebrtc/libwebrtc/call/rtp_transport_controller_send.cc

@@ -61,6 +61,7 @@ RtpTransportControllerSend::RtpTransportControllerSend(
     : packet_router_(packet_router),
       pacer_(packet_router_),
       observer_(nullptr),
+	    stats_tracer_(nullptr),
       controller_factory_override_(controller_factory),
       process_interval_(controller_factory_override_->GetProcessInterval()),
       last_report_block_time_(Timestamp::ms(DepLibUV::GetTimeMsInt64())),
@@ -143,6 +144,12 @@ void RtpTransportControllerSend::RegisterTargetTransferRateObserver(
     MaybeCreateControllers();
 }
 
+void RtpTransportControllerSend::RegisterBweStatsTracer(webrtc::BweStatsTracer* tracer) {
+	MS_ASSERT(stats_tracer_ == nullptr, "stats_tracer already set");
+
+	stats_tracer_ = tracer;
+}
+
 void RtpTransportControllerSend::OnNetworkAvailability(bool network_available) {
   MS_DEBUG_DEV("<<<<< network_available:%s", network_available ? "true" : "false");
 
@@ -242,6 +249,7 @@ void RtpTransportControllerSend::OnTransportFeedback(
     PostUpdates(controller_->OnTransportPacketsFeedback(*feedback_msg));
   pacer_.UpdateOutstandingData(
       transport_feedback_adapter_.GetOutstandingData().bytes());
+	stats_tracer_->OnBweStats(controller_->GetBweStats());
 }
 
 void RtpTransportControllerSend::OnRemoteNetworkEstimate(

worker/deps/libwebrtc/libwebrtc/call/rtp_transport_controller_send.h

@@ -61,6 +61,8 @@ class RtpTransportControllerSend final
   void SetPacingFactor(float pacing_factor) override;
   void RegisterTargetTransferRateObserver(
       TargetTransferRateObserver* observer) override;
+	void RegisterBweStatsTracer(
+		BweStatsTracer* tracer) override;
   void OnNetworkAvailability(bool network_available) override;
   RtcpBandwidthObserver* GetBandwidthObserver() override;
   void EnablePeriodicAlrProbing(bool enable) override;
@@ -102,6 +104,8 @@ class RtpTransportControllerSend final
 
   TargetTransferRateObserver* observer_;
 
+	BweStatsTracer* stats_tracer_;
+
   NetworkControllerFactoryInterface* const controller_factory_override_;
 
   TransportFeedbackAdapter transport_feedback_adapter_;

worker/deps/libwebrtc/libwebrtc/call/rtp_transport_controller_send_interface.h

@@ -36,6 +36,7 @@ struct NetworkRoute;
 namespace webrtc {
 
 class TargetTransferRateObserver;
+class BweStatsTracer;
 class Transport;
 class PacedSender;
 class PacketFeedbackObserver;
@@ -99,6 +100,8 @@ class RtpTransportControllerSendInterface {
       // PacketFeedbackObserver* observer) = 0;
   virtual void RegisterTargetTransferRateObserver(
       TargetTransferRateObserver* observer) = 0;
+	virtual void RegisterBweStatsTracer(
+		BweStatsTracer* tracer) = 0;
   virtual void OnNetworkAvailability(bool network_available) = 0;
   virtual RtcpBandwidthObserver* GetBandwidthObserver() = 0;
   virtual void EnablePeriodicAlrProbing(bool enable) = 0;