Add a PacketOptions struct to webrtc::Transport.

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 d4700a963cadc9c0e43cc8aedf2c7f5fe5fe2f22..bc6ce56cb45ab57ed4a6b39d04dca87a4f23d707 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 +28,9 @@ class ExtensionVerifyTransport : public webrtc::Transport {
         audio_level_id_(-1),
         absolute_sender_time_id_(-1) {}
 
-  bool SendRtp(const uint8_t* data, size_t len) override {
+  bool SendRtp(const uint8_t* data,
+               size_t len,
+               const webrtc::PacketOptions& options) override {
     webrtc::RTPHeader header;
     if (parser_->Parse(reinterpret_cast<const uint8_t*>(data), len, &header)) {
       bool ok = true;