Move Ownership of RtpModules to VideoSendStream.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc

 /*
  *  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 283 matching lines @@
 
 uint16_t ModuleRtpRtcpImpl::SequenceNumber() const {
   return rtp_sender_.SequenceNumber();
 }
 
 // Set SequenceNumber, default is a random number.
 void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
   rtp_sender_.SetSequenceNumber(seq_num);
 }
 
-bool ModuleRtpRtcpImpl::SetRtpStateForSsrc(uint32_t ssrc,
-                                           const RtpState& rtp_state) {
-  if (rtp_sender_.SSRC() == ssrc) {
-    SetStartTimestamp(rtp_state.start_timestamp);
-    rtp_sender_.SetRtpState(rtp_state);
-    return true;
-  }
-  if (rtp_sender_.RtxSsrc() == ssrc) {
-    rtp_sender_.SetRtxRtpState(rtp_state);
-    return true;
-  }
-  return false;
+void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) {
+  SetStartTimestamp(rtp_state.start_timestamp);
+  rtp_sender_.SetRtpState(rtp_state);
 }
 
-bool ModuleRtpRtcpImpl::GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) {
-  if (rtp_sender_.SSRC() == ssrc) {
-    *rtp_state = rtp_sender_.GetRtpState();
-    return true;
-  }
-  if (rtp_sender_.RtxSsrc() == ssrc) {
-    *rtp_state = rtp_sender_.GetRtxRtpState();
-    return true;
-  }
-  return false;
+void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) {
+  rtp_sender_.SetRtxRtpState(rtp_state);
+}
+
+RtpState ModuleRtpRtcpImpl::GetRtpState() const {
+  return rtp_sender_.GetRtpState();
+}
+
+RtpState ModuleRtpRtcpImpl::GetRtxState() const {
+  return rtp_sender_.GetRtxRtpState();
 }
 
 uint32_t ModuleRtpRtcpImpl::SSRC() const {
   return rtp_sender_.SSRC();
 }
 
 // Configure SSRC, default is a random number.
 void ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) {
   rtp_sender_.SetSSRC(ssrc);
   rtcp_sender_.SetSSRC(ssrc);
@@ skipping 656 matching lines @@
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
   rtp_sender_.RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
   return rtp_sender_.GetRtpStatisticsCallback();
 }
 }  // namespace webrtc