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

media/cast/audio_receiver/audio_receiver_unittest.cc

Index: media/cast/audio_receiver/audio_receiver_unittest.cc
diff --git a/media/cast/audio_receiver/audio_receiver_unittest.cc b/media/cast/audio_receiver/audio_receiver_unittest.cc
index c9f3ebac9222f48cbd4eea303a7001ae33121778..106c5979f2335feee28f9e929f4ef853f6768a15 100644
--- a/media/cast/audio_receiver/audio_receiver_unittest.cc
+++ b/media/cast/audio_receiver/audio_receiver_unittest.cc
@@ -92,9 +92,8 @@ class AudioReceiverTest : public ::testing::Test {
     rtp_header_.frame_id = kFirstFrameId;
     rtp_header_.packet_id = 0;
     rtp_header_.max_packet_id = 0;
-    rtp_header_.is_reference = false;
     rtp_header_.reference_frame_id = 0;
-    rtp_header_.webrtc.header.timestamp = 0;
+    rtp_header_.rtp_timestamp = 0;
   }
 
   void FeedOneFrameIntoReceiver() {
@@ -144,8 +143,7 @@ TEST_F(AudioReceiverTest, GetOnePacketEncodedFrame) {
   ASSERT_TRUE(!frame_events.empty());
   EXPECT_EQ(kAudioAckSent, frame_events.begin()->type);
   EXPECT_EQ(rtp_header_.frame_id, frame_events.begin()->frame_id);
-  EXPECT_EQ(rtp_header_.webrtc.header.timestamp,
-            frame_events.begin()->rtp_timestamp);
+  EXPECT_EQ(rtp_header_.rtp_timestamp, frame_events.begin()->rtp_timestamp);
 
   cast_environment_->Logging()->RemoveRawEventSubscriber(&event_subscriber);
 }
@@ -175,7 +173,7 @@ TEST_F(AudioReceiverTest, MultiplePendingGetCalls) {
   uint32 ntp_low;
   ConvertTimeTicksToNtp(testing_clock_->NowTicks(), &ntp_high, &ntp_low);
   rtcp_packet.AddSrWithNtp(audio_config_.feedback_ssrc, ntp_high, ntp_low,
-                           rtp_header_.webrtc.header.timestamp);
+                           rtp_header_.rtp_timestamp);
 
   testing_clock_->Advance(base::TimeDelta::FromMilliseconds(20));
 
@@ -191,9 +189,8 @@ TEST_F(AudioReceiverTest, MultiplePendingGetCalls) {
   // and that the RTP timestamp represents a time in the future.
   rtp_header_.is_key_frame = false;
   rtp_header_.frame_id = kFirstFrameId + 2;
-  rtp_header_.is_reference = true;
   rtp_header_.reference_frame_id = 0;
-  rtp_header_.webrtc.header.timestamp = 960;
+  rtp_header_.rtp_timestamp = 960;
   fake_audio_client_.SetNextExpectedResult(
       kFirstFrameId + 2,
       testing_clock_->NowTicks() + base::TimeDelta::FromMilliseconds(100));
@@ -216,9 +213,8 @@ TEST_F(AudioReceiverTest, MultiplePendingGetCalls) {
 
   // Receive Frame 3 and expect it to fulfill the third request immediately.
   rtp_header_.frame_id = kFirstFrameId + 3;
-  rtp_header_.is_reference = false;
-  rtp_header_.reference_frame_id = 0;
-  rtp_header_.webrtc.header.timestamp = 1280;
+  rtp_header_.reference_frame_id = rtp_header_.frame_id - 1;
+  rtp_header_.rtp_timestamp = 1280;
   fake_audio_client_.SetNextExpectedResult(kFirstFrameId + 3,
                                            testing_clock_->NowTicks());
   FeedOneFrameIntoReceiver();