Move RtpTransportControllerSend to a new file.

Also move RtpTransportControllerSendInterface to its own header file.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2808043002
Cr-Commit-Position: refs/heads/master@{#17840}

webrtc/call/rtp_transport_controller_send_interface.h

diff --git a/webrtc/call/rtp_transport_controller_send_interface.h b/webrtc/call/rtp_transport_controller_send_interface.h
new file mode 100644
index 0000000..31f0746
--- /dev/null
+++ b/webrtc/call/rtp_transport_controller_send_interface.h
@@ -0,0 +1,57 @@
+/*
+ *  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_SEND_INTERFACE_H_
+#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
+
+namespace webrtc {
+
+class PacketRouter;
+class RtpPacketSender;
+class SendSideCongestionController;
+class TransportFeedbackObserver;
+
+// 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 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 PacketRouter* packet_router() = 0;
+  // Currently returning the same pointer, but with different types.
+  virtual SendSideCongestionController* send_side_cc() = 0;
+  virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
+
+  virtual RtpPacketSender* packet_sender() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_