Unify Transport and newapi::Transport interfaces.

webrtc/modules/rtp_rtcp/source/rtp_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 717 matching lines @@
   if (!PrepareAndSendPacket(data_buffer, length, capture_time_ms,
                             (rtx & kRtxRetransmitted) > 0, true)) {
     return -1;
   }
   return static_cast<int32_t>(length);
 }
 
 bool RTPSender::SendPacketToNetwork(const uint8_t *packet, size_t size) {
   int bytes_sent = -1;
   if (transport_) {
-    bytes_sent = transport_->SendPacket(packet, size);
+    bytes_sent =
+        transport_->SendRtp(packet, size) ? static_cast<int>(size) : -1;
   }
   TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                        "RTPSender::SendPacketToNetwork", "size", size, "sent",
                        bytes_sent);
   // TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
   if (bytes_sent <= 0) {
     LOG(LS_WARNING) << "Transport failed to send packet";
     return false;
   }
   return true;
@@ skipping 1151 matching lines @@
   CriticalSectionScoped lock(send_critsect_.get());
 
   RtpState state;
   state.sequence_number = sequence_number_rtx_;
   state.start_timestamp = start_timestamp_;
 
   return state;
 }
 
 }  // namespace webrtc