Add accessor functions for protected member variables of ModuleRtpRtcpImpl.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h

 /*
  *  Copyright (c) 2012 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.
  */
 
@@ skipping 294 matching lines @@
       const std::vector<uint16_t>& nack_sequence_numbers) override;
   void OnReceivedRtcpReportBlocks(
       const ReportBlockList& report_blocks) override;
   void OnRequestSendReport() override;
 
   void SetVideoBitrateAllocation(const BitrateAllocation& bitrate) override;
 
  protected:
   bool UpdateRTCPReceiveInformationTimers();
 
+  RTPSender* rtp_sender() { return &rtp_sender_; }
+  const RTPSender* rtp_sender() const { return &rtp_sender_; }
+
+  RTCPSender* rtcp_sender() { return &rtcp_sender_; }
+  const RTCPSender* rtcp_sender() const { return &rtcp_sender_; }
+
+  RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; }
+  const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; }
+
+  Clock* clock() { return clock_; }

[danilchap, 2017/03/13 12:57:19]

since Clock interface has only const functions, it might be better to make this
accessor const:
const Clock* clock() const { return clock_; }

[nisse-webrtc, 2017/03/13 14:05:46]

Done. Made the |clock_| member const const as well.

+
+  // TODO(nisse): Demote all member variables to private, as soon as
+  // downstream code is updated to use the above accessor methods.
   RTPSender rtp_sender_;
 
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 
   Clock* clock_;
 
  private:
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
@@ skipping 26 matching lines @@
   PacketLossStats receive_loss_stats_;
 
   // The processed RTT from RtcpRttStats.
   rtc::CriticalSection critical_section_rtt_;
   int64_t rtt_ms_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_