Unify Transport and newapi::Transport interfaces.

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 1099 matching lines @@
     delaySinceLastReceivedSR = now-receiveTime;
   }
   report_block->delaySinceLastSR = delaySinceLastReceivedSR;
   report_block->lastSR = feedback_state.remote_sr;
   return true;
 }
 
 int32_t RTCPSender::SendToNetwork(const uint8_t* dataBuffer, size_t length) {
   CriticalSectionScoped lock(critical_section_transport_.get());
   if (cbTransport_) {
-    if (cbTransport_->SendRTCPPacket(dataBuffer, length) > 0)
+    if (cbTransport_->SendRtcp(dataBuffer, length))
       return 0;
   }
   return -1;
 }
 
 void RTCPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
   assert(csrcs.size() <= kRtpCsrcSize);
   CriticalSectionScoped lock(critical_section_rtcp_sender_.get());
   csrcs_ = csrcs;
 }
@@ skipping 82 matching lines @@
   CriticalSectionScoped lock(critical_section_transport_.get());
   if (!cbTransport_)
     return false;
 
   class Sender : public rtcp::RtcpPacket::PacketReadyCallback {
    public:
     Sender(Transport* transport)
         : transport_(transport), send_failure_(false) {}
 
     void OnPacketReady(uint8_t* data, size_t length) override {
-      if (transport_->SendRTCPPacket(data, length) <= 0)
+      if (!transport_->SendRtcp(data, length))
         send_failure_ = true;
     }
 
     Transport* const transport_;
     bool send_failure_;
   } sender(cbTransport_);
 
   uint8_t buffer[IP_PACKET_SIZE];
   return packet.BuildExternalBuffer(buffer, IP_PACKET_SIZE, &sender) &&
          !sender.send_failure_;
 }
 
 }  // namespace webrtc