Move Ownership of RtpModules to VideoSendStream.

webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.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 67 matching lines @@
                void(int payload_type, const char* payload_name));
   MOCK_METHOD1(DeRegisterSendPayload,
       int32_t(const int8_t payloadType));
   MOCK_METHOD2(RegisterSendRtpHeaderExtension,
       int32_t(const RTPExtensionType type, const uint8_t id));
   MOCK_METHOD1(DeregisterSendRtpHeaderExtension,
       int32_t(const RTPExtensionType type));
   MOCK_CONST_METHOD0(StartTimestamp,
       uint32_t());
   MOCK_METHOD1(SetStartTimestamp, void(const uint32_t timestamp));
-  MOCK_CONST_METHOD0(SequenceNumber,
-      uint16_t());
+  MOCK_CONST_METHOD0(SequenceNumber, uint16_t());
   MOCK_METHOD1(SetSequenceNumber, void(const uint16_t seq));
-  MOCK_METHOD2(SetRtpStateForSsrc,
-               bool(uint32_t ssrc, const RtpState& rtp_state));
-  MOCK_METHOD2(GetRtpStateForSsrc, bool(uint32_t ssrc, RtpState* rtp_state));
+  MOCK_METHOD1(SetRtpState, void(const RtpState& rtp_state));
+  MOCK_METHOD1(SetRtxState, void(const RtpState& rtp_state));
+  MOCK_CONST_METHOD0(GetRtpState, RtpState());
+  MOCK_CONST_METHOD0(GetRtxState, RtpState());
   MOCK_CONST_METHOD0(SSRC,
       uint32_t());
   MOCK_METHOD1(SetSSRC,
       void(const uint32_t ssrc));
   MOCK_CONST_METHOD1(CSRCs,
       int32_t(uint32_t arrOfCSRC[kRtpCsrcSize]));
   MOCK_METHOD1(SetCsrcs, void(const std::vector<uint32_t>& csrcs));
   MOCK_METHOD1(SetCSRCStatus,
       int32_t(const bool include));
   MOCK_METHOD1(SetRtxSendStatus, void(int modes));
@@ skipping 155 matching lines @@
       void(StreamDataCountersCallback*));
   MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
       StreamDataCountersCallback*(void));
   // Members.
   unsigned int remote_ssrc_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_