Unify Transport and newapi::Transport interfaces.

webrtc/voice_engine/test/auto_test/fakes/conference_transport.cc

 /*
  *  Copyright (c) 2015 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 90 matching lines @@
   remote_base_->Release();
 
   local_rtp_rtcp_->Release();
   local_network_->Release();
   local_base_->Release();
 
   EXPECT_TRUE(webrtc::VoiceEngine::Delete(remote_voe_));
   EXPECT_TRUE(webrtc::VoiceEngine::Delete(local_voe_));
 }
 
-int ConferenceTransport::SendPacket(const void* data, size_t len) {
+bool ConferenceTransport::SendRtp(const uint8_t* data, size_t len) {
   StorePacket(Packet::Rtp, data, len);
-  return static_cast<int>(len);
+  return true;
 }
 
-int ConferenceTransport::SendRTCPPacket(const void* data, size_t len) {
+bool ConferenceTransport::SendRtcp(const uint8_t* data, size_t len) {
   StorePacket(Packet::Rtcp, data, len);
-  return static_cast<int>(len);
+  return true;
 }
 
 int ConferenceTransport::GetReceiverChannelForSsrc(unsigned int sender_ssrc)
     const {
   webrtc::CriticalSectionScoped lock(stream_crit_.get());
   auto it = streams_.find(sender_ssrc);
   if (it != streams_.end()) {
     return it->second.second;
   }
   return -1;
@@ skipping 150 matching lines @@
 bool ConferenceTransport::GetReceiverStatistics(unsigned int id,
                                                 webrtc::CallStatistics* stats) {
   int dst = GetReceiverChannelForSsrc(id);
   if (dst == -1) {
     return false;
   }
   EXPECT_EQ(0, local_rtp_rtcp_->GetRTCPStatistics(dst, *stats));
   return true;
 }
 }  // namespace voetest