Cast:Adding cast_transport_config and cleaning up

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 14 matching lines @@
  public:
   TestAudioEncoderCallback()
       : num_called_(0) {}
 
   void SetExpectedResult(uint8 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<EncodedAudioFrame> audio_frame,
-                                const base::TimeTicks& playout_time) {
+  void DeliverEncodedAudioFrame(
+      scoped_ptr<transport::EncodedAudioFrame> audio_frame,
+      const base::TimeTicks& playout_time) {
     EXPECT_EQ(expected_frame_id_, audio_frame->frame_id);
-    EXPECT_EQ(kPcm16, audio_frame->codec);
+    EXPECT_EQ(transport::kPcm16, audio_frame->codec);
     EXPECT_EQ(expected_playout_time_, playout_time);
     num_called_++;
   }
 
   int number_times_called() const { return num_called_;}
 
  protected:
   virtual ~TestAudioEncoderCallback() {}
 
  private:
@@ skipping 15 matching lines @@
   using AudioReceiver::IncomingParsedRtpPacket;
 };
 
 class AudioReceiverTest : public ::testing::Test {
  protected:
   AudioReceiverTest() {
     // Configure the audio receiver to use PCM16.
     audio_config_.rtp_payload_type = 127;
     audio_config_.frequency = 16000;
     audio_config_.channels = 1;
-    audio_config_.codec = kPcm16;
+    audio_config_.codec = transport::kPcm16;
     audio_config_.use_external_decoder = false;
     audio_config_.feedback_ssrc = 1234;
     testing_clock_.Advance(
         base::TimeDelta::FromMilliseconds(kStartMillisecond));
     task_runner_ = new test::FakeTaskRunner(&testing_clock_);
     cast_environment_ = new CastEnvironment(&testing_clock_, task_runner_,
         task_runner_, task_runner_, task_runner_, task_runner_,
-        GetDefaultCastLoggingConfig());
+        task_runner_, GetDefaultCastLoggingConfig());
     test_audio_encoder_callback_ = new TestAudioEncoderCallback();
   }
 
   void Configure(bool use_external_decoder) {
     audio_config_.use_external_decoder = use_external_decoder;
     receiver_.reset(new PeerAudioReceiver(cast_environment_, audio_config_,
                                           &mock_transport_));
   }
 
   virtual ~AudioReceiverTest() {}
 
   static void DummyDeletePacket(const uint8* packet) {};
 
   virtual void SetUp() {
-    payload_.assign(kIpPacketSize, 0);
+    payload_.assign(kMaxIpPacketSize, 0);
     rtp_header_.is_key_frame = true;
     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;
     rtp_header_.webrtc.header.timestamp = 0;
   }
 
   AudioReceiverConfig audio_config_;
   std::vector<uint8> payload_;
   RtpCastHeader rtp_header_;
   base::SimpleTestTickClock testing_clock_;
   transport::MockPacedPacketSender mock_transport_;
   scoped_refptr<test::FakeTaskRunner> task_runner_;
   scoped_ptr<PeerAudioReceiver> receiver_;
   scoped_refptr<CastEnvironment> cast_environment_;
   scoped_refptr<TestAudioEncoderCallback> test_audio_encoder_callback_;
 };
 
 TEST_F(AudioReceiverTest, GetOnePacketEncodedframe) {
   Configure(true);
   EXPECT_CALL(mock_transport_, SendRtcpPacket(testing::_)).Times(1);
 
   receiver_->IncomingParsedRtpPacket(payload_.data(),
       payload_.size(), rtp_header_);
-  EncodedAudioFrame audio_frame;
+  transport::EncodedAudioFrame audio_frame;
   base::TimeTicks playout_time;
   test_audio_encoder_callback_->SetExpectedResult(0, testing_clock_.NowTicks());
 
   AudioFrameEncodedCallback frame_encoded_callback =
       base::Bind(&TestAudioEncoderCallback::DeliverEncodedAudioFrame,
                  test_audio_encoder_callback_.get());
 
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
   task_runner_->RunTasks();
   EXPECT_EQ(1, test_audio_encoder_callback_->number_times_called());
 }
 
 TEST_F(AudioReceiverTest, MultiplePendingGetCalls) {
   Configure(true);
   EXPECT_CALL(mock_transport_, SendRtcpPacket(testing::_)).WillRepeatedly(
       testing::Return(true));
 
   AudioFrameEncodedCallback frame_encoded_callback =
       base::Bind(&TestAudioEncoderCallback::DeliverEncodedAudioFrame,
                  test_audio_encoder_callback_.get());
 
   receiver_->GetEncodedAudioFrame(frame_encoded_callback);
 
   receiver_->IncomingParsedRtpPacket(payload_.data(), payload_.size(),
                                      rtp_header_);
 
-  EncodedAudioFrame audio_frame;
+  transport::EncodedAudioFrame audio_frame;
   base::TimeTicks playout_time;
   test_audio_encoder_callback_->SetExpectedResult(0, testing_clock_.NowTicks());
 
   task_runner_->RunTasks();
   EXPECT_EQ(1, test_audio_encoder_callback_->number_times_called());
 
   TestRtcpPacketBuilder rtcp_packet;
 
   uint32 ntp_high;
   uint32 ntp_low;
@@ skipping 46 matching lines @@
   task_runner_->RunTasks();
   EXPECT_EQ(3, test_audio_encoder_callback_->number_times_called());
 }
 
 // TODO(mikhal): Add encoded frames.
 TEST_F(AudioReceiverTest, GetRawFrame) {
 }
 
 }  // namespace cast
 }  // namespace media