Unify Transport and newapi::Transport interfaces.

webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc

Index: webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
diff --git a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
index 4f92e0670d20382304492d81fb7515db098d9f4e..d4700a963cadc9c0e43cc8aedf2c7f5fe5fe2f22 100644
--- a/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ b/webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
@@ -28,7 +28,7 @@ class ExtensionVerifyTransport : public webrtc::Transport {
         audio_level_id_(-1),
         absolute_sender_time_id_(-1) {}
 
-  int SendPacket(const void* data, size_t len) override {
+  bool SendRtp(const uint8_t* data, size_t len) override {
     webrtc::RTPHeader header;
     if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) {
       bool ok = true;
@@ -48,11 +48,11 @@ class ExtensionVerifyTransport : public webrtc::Transport {
     }
     // received_packets_ count all packets we receive.
     ++received_packets_;
-    return static_cast<int>(len);
+    return true;
   }
 
-  int SendRTCPPacket(const void* data, size_t len) override {
-    return static_cast<int>(len);
+  bool SendRtcp(const uint8_t* data, size_t len) override {
+    return true;
   }
 
   void SetAudioLevelId(int id) {