Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

call/rtp_transport_controller_send_interface.h

@@ -0,0 +1,73 @@
+/*
+ *  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 PacedSender;
+class PacketRouter;
+class RtpPacketSender;
+struct RtpKeepAliveConfig;
+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
+// WebRtcVideoChannel 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;
+  virtual PacedSender* pacer() = 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;
+  virtual const RtpKeepAliveConfig& keepalive_config() const = 0;
+
+  // SetAllocatedSendBitrateLimits sets bitrates limits imposed by send codec
+  // settings.
+  // |min_send_bitrate_bps| is the total minimum send bitrate required by all
+  // sending streams.  This is the minimum bitrate the PacedSender will use.
+  // Note that SendSideCongestionController::OnNetworkChanged can still be
+  // called with a lower bitrate estimate. |max_padding_bitrate_bps| is the max
+  // bitrate the send streams request for padding. This can be higher than the
+  // current network estimate and tells the PacedSender how much it should max
+  // pad unless there is real packets to send.
+  virtual void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps,
+                                             int max_padding_bitrate_bps) = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_