Unify newapi::RtcpMode and RTCPMethod.

webrtc/video_engine/vie_channel.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 454 matching lines @@
 }
 
 uint32_t ViEChannel::DiscardedPackets() const {
   return vcm_->DiscardedPackets();
 }
 
 int ViEChannel::ReceiveDelay() const {
   return vcm_->Delay();
 }
 
-void ViEChannel::SetRTCPMode(const RTCPMethod rtcp_mode) {
+void ViEChannel::SetRTCPMode(const RtcpMode rtcp_mode) {
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
     rtp_rtcp->SetRTCPStatus(rtcp_mode);
 }
 
 void ViEChannel::SetProtectionMode(bool enable_nack,
                                    bool enable_fec,
                                    int payload_type_red,
                                    int payload_type_fec) {
   // Validate payload types.
   if (enable_fec) {
@@ skipping 25 matching lines @@
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetGenericFECStatus(enable_fec,
                                   static_cast<uint8_t>(payload_type_red),
                                   static_cast<uint8_t>(payload_type_fec));
   }
 }
 
 void ViEChannel::ProcessNACKRequest(const bool enable) {
   if (enable) {
     // Turn on NACK.
-    if (rtp_rtcp_modules_[0]->RTCP() == kRtcpOff)
+    if (rtp_rtcp_modules_[0]->RTCP() == RtcpMode::kOff)
       return;
     vie_receiver_.SetNackStatus(true, max_nack_reordering_threshold_);
 
     for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
       rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_);
 
     vcm_->RegisterPacketRequestCallback(this);
     // Don't introduce errors when NACK is enabled.
     vcm_->SetDecodeErrorMode(kNoErrors);
   } else {
@@ skipping 630 matching lines @@
   configuration.send_frame_count_observer = send_frame_count_observer;
   configuration.send_side_delay_observer = send_side_delay_observer;
   configuration.bandwidth_callback = bandwidth_callback;
   configuration.transport_feedback_callback = transport_feedback_callback;
 
   std::vector<RtpRtcp*> modules;
   for (size_t i = 0; i < num_modules; ++i) {
     RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
     rtp_rtcp->SetSendingStatus(false);
     rtp_rtcp->SetSendingMediaStatus(false);
-    rtp_rtcp->SetRTCPStatus(kRtcpCompound);
+    rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
     modules.push_back(rtp_rtcp);
     // Receive statistics and remote bitrate estimator should only be set for
     // the primary (first) module.
     configuration.receive_statistics = null_receive_statistics;
     configuration.remote_bitrate_estimator = nullptr;
   }
   return modules;
 }
 
 void ViEChannel::StartDecodeThread() {
@@ skipping 70 matching lines @@
   CriticalSectionScoped cs(crit_.get());
   receive_stats_callback_ = receive_statistics_proxy;
 }
 
 void ViEChannel::SetIncomingVideoStream(
     IncomingVideoStream* incoming_video_stream) {
   CriticalSectionScoped cs(crit_.get());
   incoming_video_stream_ = incoming_video_stream;
 }
 }  // namespace webrtc