Don't allocate any RTPSender object for a receive only RtpRtcp module

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.
  */
 
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
 
+#include <memory>
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/gtest_prod_util.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/packet_loss_stats.h"
@@ skipping 26 matching lines @@
 
   int32_t RegisterSendPayload(const CodecInst& voice_codec) override;
 
   int32_t RegisterSendPayload(const VideoCodec& video_codec) override;
 
   void RegisterVideoSendPayload(int payload_type,
                                 const char* payload_name) override;
 
   int32_t DeRegisterSendPayload(int8_t payload_type) override;
 
-  int8_t SendPayloadType() const;

[danilchap, 2017/03/10 15:48:12]

may be delete this method in another, tiny CL.

[nisse-webrtc, 2017/03/13 08:20:10]

I can do that, to reduce the risk of downstream breakage from this cl.

-
   // Register RTP header extension.
   int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
                                          uint8_t id) override;
 
   int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) override;
 
   bool HasBweExtensions() const override;
 
   // Get start timestamp.
   uint32_t StartTimestamp() const override;
@@ skipping 235 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_;
+  std::unique_ptr<RTPSender> rtp_sender_;
 
   RTCPSender rtcp_sender_;
   RTCPReceiver rtcp_receiver_;
 
   Clock* clock_;
 
  private:
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt);
   FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly);
   int64_t RtcpReportInterval();
@@ skipping 25 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_