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 903 matching lines @@
 
     // --- RTP/RTCP module initialization
 
     // Ensure that RTCP is enabled by default for the created channel.
     // Note that, the module will keep generating RTCP until it is explicitly
     // disabled by the user.
     // After StopListen (when no sockets exists), RTCP packets will no longer
     // be transmitted since the Transport object will then be invalid.
     telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
     // RTCP is enabled by default.
-    _rtpRtcpModule->SetRTCPStatus(kRtcpCompound);
+    _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
     // --- Register all permanent callbacks
     const bool fail =
         (audio_coding_->RegisterTransportCallback(this) == -1) ||
         (audio_coding_->RegisterVADCallback(this) == -1);
 
     if (fail)
     {
         _engineStatisticsPtr->SetLastError(
             VE_CANNOT_INIT_CHANNEL, kTraceError,
             "Channel::Init() callbacks not registered");
@@ skipping 1841 matching lines @@
   if (enable && !rtp_header_parser_->RegisterRtpHeaderExtension(
       kRtpExtensionAbsoluteSendTime, id)) {
     return -1;
   }
   return 0;
 }
 
 void Channel::SetRTCPStatus(bool enable) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetRTCPStatus()");
-  _rtpRtcpModule->SetRTCPStatus(enable ? kRtcpCompound : kRtcpOff);
+  _rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
 }
 
 int
 Channel::GetRTCPStatus(bool& enabled)
 {
-    RTCPMethod method = _rtpRtcpModule->RTCP();
-    enabled = (method != kRtcpOff);
+  RtcpMode method = _rtpRtcpModule->RTCP();
+  enabled = (method != RtcpMode::kOff);
     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 125 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();
-    if (status == kRtcpOff)
-    {
+    RtcpMode status = _rtpRtcpModule->RTCP();
+    if (status == RtcpMode::kOff) {
         _engineStatisticsPtr->SetLastError(
             VE_RTCP_ERROR, kTraceError,
             "SendApplicationDefinedRTCPPacket() RTCP is disabled");
         return -1;
     }
 
     // Create and schedule the RTCP APP packet for transmission
     if (_rtpRtcpModule->SetRTCPApplicationSpecificData(
         subType,
         name,
         (const unsigned char*) data,
         dataLengthInBytes) != 0)
     {
         _engineStatisticsPtr->SetLastError(
             VE_SEND_ERROR, kTraceError,
             "SendApplicationDefinedRTCPPacket() failed to send RTCP packet");
         return -1;
     }
     return 0;
 }
 
 int
 Channel::GetRTPStatistics(
         unsigned int& averageJitterMs,
         unsigned int& maxJitterMs,
         unsigned int& discardedPackets)
 {
     // The jitter statistics is updated for each received RTP packet and is
     // based on received packets.
-    if (_rtpRtcpModule->RTCP() == kRtcpOff) {
+    if (_rtpRtcpModule->RTCP() == RtcpMode::kOff) {
       // If RTCP is off, there is no timed thread in the RTCP module regularly
       // generating new stats, trigger the update manually here instead.
       StreamStatistician* statistician =
           rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC());
       if (statistician) {
         // Don't use returned statistics, use data from proxy instead so that
         // max jitter can be fetched atomically.
         RtcpStatistics s;
         statistician->GetStatistics(&s, true);
       }
@@ skipping 50 matching lines @@
 int
 Channel::GetRTPStatistics(CallStatistics& stats)
 {
     // --- RtcpStatistics
 
     // The jitter statistics is updated for each received RTP packet and is
     // based on received packets.
     RtcpStatistics statistics;
     StreamStatistician* statistician =
         rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC());
-    if (!statistician || !statistician->GetStatistics(
-        &statistics, _rtpRtcpModule->RTCP() == kRtcpOff)) {
+    if (!statistician ||
+        !statistician->GetStatistics(
+            &statistics, _rtpRtcpModule->RTCP() == RtcpMode::kOff)) {
       _engineStatisticsPtr->SetLastError(
           VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
           "GetRTPStatistics() failed to read RTP statistics from the "
           "RTP/RTCP module");
     }
 
     stats.fractionLost = statistics.fraction_lost;
     stats.cumulativeLost = statistics.cumulative_lost;
     stats.extendedMax = statistics.extended_max_sequence_number;
     stats.jitterSamples = statistics.jitter;
@@ skipping 850 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();
-  if (method == kRtcpOff) {
+  RtcpMode method = _rtpRtcpModule->RTCP();
+  if (method == RtcpMode::kOff) {
     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());
       Channel* channel = associate_send_channel_.channel();
@@ skipping 28 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