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

media/cast/audio_receiver/audio_receiver_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/bind.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/audio_receiver/audio_receiver.h"
 #include "media/cast/cast_defines.h"
@@ skipping 74 matching lines @@
   }
 
   virtual ~AudioReceiverTest() {}
 
   virtual void SetUp() {
     payload_.assign(kMaxIpPacketSize, 0);
     rtp_header_.is_key_frame = true;
     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() {
     receiver_->OnReceivedPayloadData(
         payload_.data(), payload_.size(), rtp_header_);
   }
 
   AudioReceiverConfig audio_config_;
   std::vector<uint8> payload_;
   RtpCastHeader rtp_header_;
@@ skipping 29 matching lines @@
   FeedOneFrameIntoReceiver();
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
   std::vector<FrameEvent> frame_events;
   event_subscriber.GetFrameEventsAndReset(&frame_events);
 
   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);
 }
 
 TEST_F(AudioReceiverTest, MultiplePendingGetCalls) {
   EXPECT_CALL(mock_transport_, SendRtcpPacket(_))
       .WillRepeatedly(testing::Return(true));
 
   // Enqueue a request for an audio frame.
   const AudioFrameEncodedCallback frame_encoded_callback =
       base::Bind(&FakeAudioClient::DeliverEncodedAudioFrame,
                  base::Unretained(&fake_audio_client_));
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
   EXPECT_EQ(0, fake_audio_client_.number_times_called());
 
   // Receive one audio frame and expect to see the first request satisfied.
   fake_audio_client_.SetNextExpectedResult(kFirstFrameId,
                                            testing_clock_->NowTicks());
   FeedOneFrameIntoReceiver();
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
   TestRtcpPacketBuilder rtcp_packet;
 
   uint32 ntp_high;
   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));
 
   receiver_->IncomingPacket(rtcp_packet.GetPacket().Pass());
 
   // Enqueue a second request for an audio frame, but it should not be
   // fulfilled yet.
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
   // Receive one audio frame out-of-order: Make sure that we are not continuous
   // 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));
   FeedOneFrameIntoReceiver();
 
   // Frame 2 should not come out at this point in time.
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
   // Enqueue a third request for an audio frame.
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
   EXPECT_EQ(1, fake_audio_client_.number_times_called());
 
   // After 100 ms has elapsed, Frame 2 is emitted (to satisfy the second
   // request) because a decision was made to skip over the no-show Frame 1.
   testing_clock_->Advance(base::TimeDelta::FromMilliseconds(100));
   task_runner_->RunTasks();
   EXPECT_EQ(2, fake_audio_client_.number_times_called());
 
   // 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();
   task_runner_->RunTasks();
   EXPECT_EQ(3, fake_audio_client_.number_times_called());
 
   // Move forward another 100 ms and run any pending tasks (there should be
   // none).  Expect no additional frames where emitted.
   testing_clock_->Advance(base::TimeDelta::FromMilliseconds(100));
   task_runner_->RunTasks();
   EXPECT_EQ(3, fake_audio_client_.number_times_called());
 }
 
 }  // namespace cast
 }  // namespace media