Move PacedSender ownership to RtpTransportControllerSend.

webrtc/call/rtp_transport_controller_send_interface.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_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
@@ skipping 13 matching lines @@
 // 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_