Move Ownership of RtpModules to VideoSendStream.

webrtc/video/vie_receiver.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 106 matching lines @@
 }
 
 uint32_t ViEReceiver::GetRemoteSsrc() const {
   return rtp_receiver_->SSRC();
 }
 
 int ViEReceiver::GetCsrcs(uint32_t* csrcs) const {
   return rtp_receiver_->CSRCs(csrcs);
 }
 
-void ViEReceiver::Init(const std::vector<RtpRtcp*>& modules) {
-  rtp_rtcp_ = modules;
+void ViEReceiver::Init(RtpRtcp* rtp_rtcp) {
+  rtp_rtcp_ = rtp_rtcp;
 }
 
 RtpReceiver* ViEReceiver::GetRtpReceiver() const {
   return rtp_receiver_.get();
 }
 
 void ViEReceiver::EnableReceiveRtpHeaderExtension(const std::string& extension,
                                                   int id) {
   RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
   RTC_CHECK(rtp_header_parser_->RegisterRtpHeaderExtension(
@@ skipping 172 matching lines @@
   rtp_header.type.Video.rotation = kVideoRotation_0;
   if (header.extension.hasVideoRotation) {
     rtp_header.type.Video.rotation =
         ConvertCVOByteToVideoRotation(header.extension.videoRotation);
   }
   OnReceivedPayloadData(nullptr, 0, &rtp_header);
 }
 
 bool ViEReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
                               size_t rtcp_packet_length) {
-  // Should be set by owner at construction time.
-  RTC_DCHECK(!rtp_rtcp_.empty());
   {
     rtc::CritScope lock(&receive_cs_);
     if (!receiving_) {
       return false;
     }
   }
 
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_)
-    rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
+  rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
 
   int64_t rtt = 0;
-  rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
+  rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
   if (rtt == 0) {
     // Waiting for valid rtt.
     return true;
   }
   uint32_t ntp_secs = 0;
   uint32_t ntp_frac = 0;
   uint32_t rtp_timestamp = 0;
-  if (rtp_rtcp_[0]->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
-                              &rtp_timestamp) != 0) {
+  if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
+                           &rtp_timestamp) != 0) {
     // Waiting for RTCP.
     return true;
   }
   ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
 
   return true;
 }
 
 void ViEReceiver::StartReceive() {
   rtc::CritScope lock(&receive_cs_);
@@ skipping 21 matching lines @@
                                         bool in_order) const {
   // Retransmissions are handled separately if RTX is enabled.
   if (rtp_payload_registry_.RtxEnabled())
     return false;
   StreamStatistician* statistician =
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   // Check if this is a retransmission.
   int64_t min_rtt = 0;
-  rtp_rtcp_[0]->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr);
+  rtp_rtcp_->RTT(rtp_receiver_->SSRC(), nullptr, nullptr, &min_rtt, nullptr);
   return !in_order &&
       statistician->IsRetransmitOfOldPacket(header, min_rtt);
 }
 }  // namespace webrtc