Add a PacketOptions struct to webrtc::Transport.

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 196 matching lines @@
 {
     CriticalSectionScoped cs(&_callbackCritSect);
     if (_rxVadObserverPtr)
     {
         _rxVadObserverPtr->OnRxVad(_channelId, vadDecision);
     }
 
     return 0;
 }
 
-bool
-Channel::SendRtp(const uint8_t *data, size_t len)
-{
+bool Channel::SendRtp(const uint8_t* data,
+                      size_t len,
+                      const PacketOptions& options) {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SendPacket(channel=%d, len=%" PRIuS ")", len);
 
     CriticalSectionScoped cs(&_callbackCritSect);
 
     if (_transportPtr == NULL)
     {
         WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
                      "Channel::SendPacket() failed to send RTP packet due to"
                      " invalid transport object");
         return false;
     }
 
     uint8_t* bufferToSendPtr = (uint8_t*)data;
     size_t bufferLength = len;
 
-    if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength)) {
+    if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) {
       std::string transport_name =
           _externalTransport ? "external transport" : "WebRtc sockets";
       WEBRTC_TRACE(kTraceError, kTraceVoice,
                    VoEId(_instanceId,_channelId),
                    "Channel::SendPacket() RTP transmission using %s failed",
                    transport_name.c_str());
       return false;
     }
     return true;
 }
@@ skipping 3707 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