[Cast] Clean-up RtpCastHeader and RtpParser, removing the last WebRTC dependency.

media/cast/framer/frame_buffer_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 "media/cast/framer/frame_buffer.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 namespace cast {
 
 class FrameBufferTest : public ::testing::Test {
  protected:
   FrameBufferTest() {
     payload_.assign(kMaxIpPacketSize, 0);
-    // 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;
   }
 
   virtual ~FrameBufferTest() {}
 
   FrameBuffer buffer_;
   std::vector<uint8> payload_;
   RtpCastHeader rtp_header_;
 
   DISALLOW_COPY_AND_ASSIGN(FrameBufferTest);
 };
 
 TEST_F(FrameBufferTest, OnePacketInsertSanity) {
-  rtp_header_.webrtc.header.timestamp = 3000u;
+  rtp_header_.rtp_timestamp = 3000;
   rtp_header_.is_key_frame = true;
   rtp_header_.frame_id = 5;
+  rtp_header_.reference_frame_id = 5;
   buffer_.InsertPacket(payload_.data(), payload_.size(), rtp_header_);
   transport::EncodedVideoFrame frame;
   EXPECT_TRUE(buffer_.GetEncodedVideoFrame(&frame));
   EXPECT_EQ(5u, frame.frame_id);
   EXPECT_TRUE(frame.key_frame);
   EXPECT_EQ(3000u, frame.rtp_timestamp);
 }
 
 TEST_F(FrameBufferTest, EmptyBuffer) {
   EXPECT_FALSE(buffer_.Complete());
@@ skipping 40 matching lines @@
   buffer_.InsertPacket(payload_.data(), payload_.size(), rtp_header_);
   EXPECT_FALSE(buffer_.Complete());
   // Insert missing packet.
   rtp_header_.packet_id = 2;
   buffer_.InsertPacket(payload_.data(), payload_.size(), rtp_header_);
   EXPECT_TRUE(buffer_.Complete());
 }
 
 }  // namespace media
 }  // namespace cast