Unify newapi::RtcpMode and RTCPMethod.

webrtc/voice_engine/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 2771 matching lines @@
 
 void Channel::SetRTCPStatus(bool enable) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetRTCPStatus()");
   _rtpRtcpModule->SetRTCPStatus(enable ? kRtcpCompound : kRtcpOff);
 }
 
 int
 Channel::GetRTCPStatus(bool& enabled)
 {
-    RTCPMethod method = _rtpRtcpModule->RTCP();
+    RtcpMode method = _rtpRtcpModule->RTCP();
     enabled = (method != kRtcpOff);
     return 0;
 }
 
 int
 Channel::SetRTCP_CNAME(const char cName[256])
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::SetRTCP_CNAME()");
     if (_rtpRtcpModule->SetCNAME(cName) != 0)
@@ skipping 126 matching lines @@
             "SendApplicationDefinedRTCPPacket() invalid data value");
         return -1;
     }
     if (dataLengthInBytes % 4 != 0)
     {
         _engineStatisticsPtr->SetLastError(
             VE_INVALID_ARGUMENT, kTraceError,
             "SendApplicationDefinedRTCPPacket() invalid length value");
         return -1;
     }
-    RTCPMethod status = _rtpRtcpModule->RTCP();
+    RtcpMode status = _rtpRtcpModule->RTCP();
     if (status == kRtcpOff)
     {
         _engineStatisticsPtr->SetLastError(
             VE_RTCP_ERROR, kTraceError,
             "SendApplicationDefinedRTCPPacket() RTCP is disabled");
         return -1;
     }
 
     // Create and schedule the RTCP APP packet for transmission
     if (_rtpRtcpModule->SetRTCPApplicationSpecificData(
@@ skipping 954 matching lines @@
       // DSP routines can operate at 48,000 Hz, but the RTP clock
       // rate for the Opus payload format is standardized to 48,000 Hz,
       // because that is the maximum supported decoding sampling rate.
       playout_frequency = 48000;
     }
   }
   return playout_frequency;
 }
 
 int64_t Channel::GetRTT(bool allow_associate_channel) const {
-  RTCPMethod method = _rtpRtcpModule->RTCP();
+  RtcpMode method = _rtpRtcpModule->RTCP();
   if (method == kRtcpOff) {
     return 0;
   }
   std::vector<RTCPReportBlock> report_blocks;
   _rtpRtcpModule->RemoteRTCPStat(&report_blocks);
 
   int64_t rtt = 0;
   if (report_blocks.empty()) {
     if (allow_associate_channel) {
       CriticalSectionScoped lock(assoc_send_channel_lock_.get());
@@ skipping 29 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc