Add datagram_transport and congestion_control interface

This change introduces experimental datagram_transport interface and congestion_control interfaces. The goal is to integrate support for datagram transport in DTLS transport and set it up in a similar way we currently setup media_transport. Datagram transport will be injected in peer connection factory the same way media_transport is injected (we might even keep using the same factory which creates both media and datagram transports for now until we decided what to do next).

Bug: webrtc:9719
Change-Id: I80e70ce8d3827664ac5f5f7e55b706fe2dd2fbef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136782
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27943}

api/congestion_control_interface.h

@@ -0,0 +1,67 @@
+/* Copyright 2018 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.
+ */
+
+// This is EXPERIMENTAL interface for media and datagram transports.
+
+#ifndef API_CONGESTION_CONTROL_INTERFACE_H_
+#define API_CONGESTION_CONTROL_INTERFACE_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "api/media_transport_interface.h"
+#include "api/units/data_rate.h"
+
+namespace webrtc {
+
+// Defines congestion control feedback interface for media and datagram
+// transports.
+class CongestionControlInterface {
+ public:
+  virtual ~CongestionControlInterface() = default;
+
+  // Updates allocation limits.
+  virtual void SetAllocatedBitrateLimits(
+      const MediaTransportAllocatedBitrateLimits& limits) = 0;
+
+  // Sets starting rate.
+  virtual void SetTargetBitrateLimits(
+      const MediaTransportTargetRateConstraints& target_rate_constraints) = 0;
+
+  // Intended for receive side. AddRttObserver registers an observer to be
+  // called for each RTT measurement, typically once per ACK. Before media
+  // transport is destructed the observer must be unregistered.
+  //
+  // TODO(sukhanov): Looks like AddRttObserver and RemoveRttObserver were
+  // never implemented for media transport, so keeping noop implementation.
+  virtual void AddRttObserver(MediaTransportRttObserver* observer) {}
+  virtual void RemoveRttObserver(MediaTransportRttObserver* observer) {}
+
+  // Adds a target bitrate observer. Before media transport is destructed
+  // the observer must be unregistered (by calling
+  // RemoveTargetTransferRateObserver).
+  // A newly registered observer will be called back with the latest recorded
+  // target rate, if available.
+  virtual void AddTargetTransferRateObserver(
+      TargetTransferRateObserver* observer) = 0;
+
+  // Removes an existing |observer| from observers. If observer was never
+  // registered, an error is logged and method does nothing.
+  virtual void RemoveTargetTransferRateObserver(
+      TargetTransferRateObserver* observer) = 0;
+
+  // Returns the last known target transfer rate as reported to the above
+  // observers.
+  virtual absl::optional<TargetTransferRate> GetLatestTargetTransferRate() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // API_CONGESTION_CONTROL_INTERFACE_H_