Style fixes for cast_unittest to build

media/cast/framer/framer_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/framer/framer.h"
 #include "media/cast/rtp_common/mock_rtp_payload_feedback.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 namespace cast {
 
 static const int64 kFrameTimeMillisecond = 33;
 
 class FramerTest : public ::testing::Test {
  protected:
   FramerTest()
       : mock_rtp_payload_feedback_(),
         framer_(&mock_rtp_payload_feedback_, 0, true, 0) {
     framer_.set_clock(&testing_clock_);
   }
 
-  ~FramerTest() {}
+  virtual ~FramerTest() {}
 
-  void SetUp() {
+  virtual void SetUp() {
     // Build a default one packet frame - populate webrtc header.
     rtp_header_.is_key_frame = false;
     rtp_header_.frame_id = 0;
     rtp_header_.packet_id = 0;
     rtp_header_.max_packet_id = 0;
     rtp_header_.is_reference = false;
     rtp_header_.reference_frame_id = 0;
     payload_.assign(kIpPacketSize, 0);
 
     EXPECT_CALL(mock_rtp_payload_feedback_,
@@ skipping 306 matching lines @@
   framer_.ReleaseFrame(frame.frame_id);
   EXPECT_TRUE(framer_.GetEncodedAudioFrame(timeout, &frame, &rtp_timestamp,
                                            &next_frame));
   EXPECT_TRUE(next_frame);
   EXPECT_EQ(0, frame.frame_id);
   framer_.ReleaseFrame(frame.frame_id);
 }
 
 }  // namespace cast
 }  // namespace media