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_
+#define WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_H_
+
+#include "webrtc/modules/congestion_controller/include/congestion_controller.h"

[danilchap, 2017/02/09 09:04:04]

why include congestion controller and forward declare it?

[nisse-webrtc, 2017/02/09 12:56:43]

Good catch. Dropping forward declaration (include is needed for
voice_engine/channel.cc).

+
+namespace webrtc {
+
+class VieRemb;
+class PacketRouter;
+class CongestionController;
+
+// An RtpTransportController should own everything related to the RTP
+// transport to/from a remote endpoint. We should have separate
+// interfaces for send and receive sice, even if they are implemented
+// by the same class. This is on 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

[danilchap, 2017/02/09 09:04:04]

probably per-(media)-stream instead of per-ssrc:
Do you want separate object for rtx or not?
May be you want separate object for flexfec (since it can protect multipe
media-streams and thus doesn't belong to any)
In quic there will be no ssrc (most likely), but we might want to use same
interface.
 

[nisse-webrtc, 2017/02/09 12:56:43]

The plan is that VideoSendStream should create one RtpTransportSender per ssrc
it uses.   

+// objects which would then be passed to VideoSendStream.
+//
+// This should also have a reference to the underlying
+// webrtc::Transport. Currently, webrtc::Transport is implemented by
+// WebRtcVideoChannel2 and WebRtcVoiceMediaChannel, and owned by
+// WebrtcSession. It's unclear why video and audio uses different
+// transports, possibly because it is implemented by BaseChannel and
+// there are other reasons for BaseChannel subclasses specific for
+// video and audio.
+//
+// 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. As an interim step, using this class for
+// video only, we could consider passing the WebRtcVideoChannel2
+// transport here.
+class RtpTransportControllerSenderInterface {
+ public:
+  virtual ~RtpTransportControllerSenderInterface() {}
+  virtual VieRemb* remb() = 0;
+  virtual PacketRouter* packet_router() = 0;
+  virtual CongestionController* congestion_controller() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_H_