Unify newapi::RtcpMode and RTCPMethod.

webrtc/modules/rtp_rtcp/source/rtcp_sender.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 121 matching lines @@
 };
 
 RTCPSender::RTCPSender(
     bool audio,
     Clock* clock,
     ReceiveStatistics* receive_statistics,
     RtcpPacketTypeCounterObserver* packet_type_counter_observer,
     Transport* outgoing_transport)
     : audio_(audio),
       clock_(clock),
-      method_(kRtcpOff),
+      method_(RtcpMode::kOff),
       transport_(outgoing_transport),
 
       critical_section_rtcp_sender_(
           CriticalSectionWrapper::CreateCriticalSection()),
       using_nack_(false),
       sending_(false),
       remb_enabled_(false),
       next_time_to_send_rtcp_(0),
       start_timestamp_(0),
       last_rtp_timestamp_(0),
@@ skipping 36 matching lines @@
   builders_[kRtcpNack] = &RTCPSender::BuildNACK;
   builders_[kRtcpXrVoipMetric] = &RTCPSender::BuildVoIPMetric;
   builders_[kRtcpXrReceiverReferenceTime] =
       &RTCPSender::BuildReceiverReferenceTime;
   builders_[kRtcpXrDlrrReportBlock] = &RTCPSender::BuildDlrr;
 }
 
 RTCPSender::~RTCPSender() {
 }
 
-RTCPMethod RTCPSender::Status() const {
+RtcpMode RTCPSender::Status() const {
   CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
   return method_;
 }
 
-void RTCPSender::SetRTCPStatus(RTCPMethod method) {
+void RTCPSender::SetRTCPStatus(RtcpMode method) {
   CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
   method_ = method;
 
-  if (method == kRtcpOff)
+  if (method == RtcpMode::kOff)
     return;
   next_time_to_send_rtcp_ =
       clock_->TimeInMilliseconds() +
       (audio_ ? RTCP_INTERVAL_AUDIO_MS / 2 : RTCP_INTERVAL_VIDEO_MS / 2);
 }
 
 bool RTCPSender::Sending() const {
   CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
   return sending_;
 }
 
 int32_t RTCPSender::SetSendingStatus(const FeedbackState& feedback_state,
                                      bool sending) {
   bool sendRTCPBye = false;
   {
     CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
 
-    if (method_ != kRtcpOff) {
+    if (method_ != RtcpMode::kOff) {
       if (sending == false && sending_ == true) {
         // Trigger RTCP bye
         sendRTCPBye = true;
       }
     }
     sending_ = sending;
   }
   if (sendRTCPBye)
     return SendRTCP(feedback_state, kRtcpBye);
   return 0;
@@ skipping 158 matching lines @@
 
    5. The resulting value of T is divided by e-3/2=1.21828 to compensate
       for the fact that the timer reconsideration algorithm converges to
       a value of the RTCP bandwidth below the intended average
 */
 
   int64_t now = clock_->TimeInMilliseconds();
 
   CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
 
-  if (method_ == kRtcpOff)
+  if (method_ == RtcpMode::kOff)
     return false;
 
   if (!audio_ && sendKeyframeBeforeRTP) {
     // for video key-frames we want to send the RTCP before the large key-frame
     // if we have a 100 ms margin
     now += RTCP_SEND_BEFORE_KEY_FRAME_MS;
   }
 
   if (now >= next_time_to_send_rtcp_) {
     return true;
@@ skipping 510 matching lines @@
 
 int32_t RTCPSender::SendCompoundRTCP(
     const FeedbackState& feedback_state,
     const std::set<RTCPPacketType>& packetTypes,
     int32_t nack_size,
     const uint16_t* nack_list,
     bool repeat,
     uint64_t pictureID) {
   {
     CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
-    if (method_ == kRtcpOff) {
+    if (method_ == RtcpMode::kOff) {
       LOG(LS_WARNING) << "Can't send rtcp if it is disabled.";
       return -1;
     }
   }
   uint8_t rtcp_buffer[IP_PACKET_SIZE];
   int rtcp_length =
       PrepareRTCP(feedback_state, packetTypes, nack_size, nack_list, repeat,
                   pictureID, rtcp_buffer, IP_PACKET_SIZE);
 
   // Sanity don't send empty packets.
@@ skipping 23 matching lines @@
   if (packet_type_counter_.first_packet_time_ms == -1)
     packet_type_counter_.first_packet_time_ms = clock_->TimeInMilliseconds();
 
   bool generate_report;
   if (IsFlagPresent(kRtcpSr) || IsFlagPresent(kRtcpRr)) {
     // Report type already explicitly set, don't automatically populate.
     generate_report = true;
     RTC_DCHECK(ConsumeFlag(kRtcpReport) == false);
   } else {
     generate_report =
-        (ConsumeFlag(kRtcpReport) && method_ == kRtcpNonCompound) ||
-        method_ == kRtcpCompound;
+        (ConsumeFlag(kRtcpReport) && method_ == RtcpMode::kReducedSize) ||
+        method_ == RtcpMode::kCompound;
     if (generate_report)
       SetFlag(sending_ ? kRtcpSr : kRtcpRr, true);
   }
 
   if (IsFlagPresent(kRtcpSr) || (IsFlagPresent(kRtcpRr) && !cname_.empty()))
     SetFlag(kRtcpSdes, true);
 
   // We need to send our NTP even if we haven't received any reports.
   clock_->CurrentNtp(context.ntp_sec, context.ntp_frac);
 
@@ skipping 222 matching lines @@
     Transport* const transport_;
     bool send_failure_;
   } sender(transport_);
 
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc