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..0b566b89e993f5c7d33a44ef8305308db8830c9e 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;
@@ -51,7 +51,7 @@ class ExtensionVerifyTransport : public webrtc::Transport {
     return static_cast<int>(len);
   }
 
-  int SendRTCPPacket(const void* data, size_t len) override {
+  bool SendRtcp(const uint8_t* data, size_t len) override {
     return static_cast<int>(len);
   }