[Cast] EncodedAudioFrame+EncodedVideoFrame+reference_time --> EncodedFrame

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 19 matching lines @@
   FakeAudioClient() : num_called_(0) {}
   virtual ~FakeAudioClient() {}
 
   void SetNextExpectedResult(uint32 expected_frame_id,
                              const base::TimeTicks& expected_playout_time) {
     expected_frame_id_ = expected_frame_id;
     expected_playout_time_ = expected_playout_time;
   }
 
   void DeliverEncodedAudioFrame(
-      scoped_ptr<transport::EncodedAudioFrame> audio_frame,
-      const base::TimeTicks& playout_time) {
+      scoped_ptr<transport::EncodedFrame> audio_frame) {
     ASSERT_FALSE(!audio_frame)
         << "If at shutdown: There were unsatisfied requests enqueued.";
     EXPECT_EQ(expected_frame_id_, audio_frame->frame_id);
-    EXPECT_EQ(transport::kPcm16, audio_frame->codec);
-    EXPECT_EQ(expected_playout_time_, playout_time);
+    EXPECT_EQ(expected_playout_time_, audio_frame->reference_time);
     num_called_++;
   }
 
   int number_times_called() const { return num_called_; }
 
  private:
   int num_called_;
   uint32 expected_frame_id_;
   base::TimeTicks expected_playout_time_;
 
@@ skipping 28 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_.reference_frame_id = 0;
+    rtp_header_.reference_frame_id = rtp_header_.frame_id;
     rtp_header_.rtp_timestamp = 0;
   }
 
   void FeedOneFrameIntoReceiver() {
     receiver_->OnReceivedPayloadData(
         payload_.data(), payload_.size(), rtp_header_);
   }
 
   AudioReceiverConfig audio_config_;
   std::vector<uint8> payload_;
@@ skipping 41 matching lines @@
   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 =
+  const FrameEncodedCallback 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();
@@ skipping 56 matching lines @@
 
   // 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