Define RtpTransportControllerSendInterface.

webrtc/call/rtp_transport_controller.h

+/*
+ *  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_CALL_RTP_TRANSPORT_CONTROLLER_H_

[danilchap, 2017/03/16 11:38:58]

may be rename this file to rtp_transport_controller_sender

[nisse-webrtc, 2017/03/23 09:16:17]

I'll rename it to rtp_transport_controller_send.h, analogous to
rtp_transport_controller_receive.h, introduced in cl 
https://codereview.webrtc.org/2709723003/

[nisse-webrtc, 2017/03/23 09:38:35]

Done.

+#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_H_
+
+namespace webrtc {
+
+class CongestionController;
+class Module;
+class PacketRouter;
+class RtpPacketSender;
+class TransportFeedbackObserver;
+class VieRemb;
+
+// An RtpTransportController should own everything related to the RTP
+// transport to/from a remote endpoint. We should have separate
+// interfaces for send and receive side, even if they are implemented
+// by the same class. This is an ongoing refactoring project. At some
+// point, this class should be promoted to a public api under
+// webrtc/api/rtp/.
+//
+// For a start, this object is just a collection of the objects needed
+// by the VideoSendStream constructor. The plan is to move ownership
+// of all RTP-related objects here, and add methods to create per-ssrc
+// objects which would then be passed to VideoSendStream. Eventually,
+// direct accessors like remb() and packet_router() should be removed.
+//
+// This should also have a reference to the underlying
+// webrtc::Transport(s). Currently, webrtc::Transport is implemented by
+// WebRtcVideoChannel2 and WebRtcVoiceMediaChannel, and owned by
+// WebrtcSession. Video and audio always uses different transport
+// objects, even in the common case where they are bundled over the
+// same underlying transport.
+//
+// Extracting the logic of the webrtc::Transport from BaseChannel and
+// subclasses into a separate class seems to be a prerequesite for
+// moving the transport here.
+class RtpTransportControllerSendInterface {
+ public:
+  virtual ~RtpTransportControllerSendInterface() {}
+  virtual VieRemb* remb() = 0;
+  virtual PacketRouter* packet_router() = 0;
+
+  // Currently returning the same pointer, but with different types.
+  virtual CongestionController* congestion_controller() = 0;
+  virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
+
+  virtual RtpPacketSender* packet_sender() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_H_