Cast: Reshuffle files under media/cast

media/cast/sender/audio_sender_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 <stdint.h>
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/base/media.h"
-#include "media/cast/audio_sender/audio_sender.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
-#include "media/cast/rtcp/rtcp.h"
+#include "media/cast/net/cast_transport_config.h"
+#include "media/cast/net/cast_transport_sender_impl.h"
+#include "media/cast/net/rtcp/rtcp.h"
+#include "media/cast/sender/audio_sender.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/utility/audio_utility.h"
-#include "media/cast/transport/cast_transport_config.h"
-#include "media/cast/transport/cast_transport_sender_impl.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 namespace cast {
 
-class TestPacketSender : public transport::PacketSender {
+class TestPacketSender : public PacketSender {
  public:
   TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
 
-  virtual bool SendPacket(transport::PacketRef packet,
+  virtual bool SendPacket(PacketRef packet,
                           const base::Closure& cb) OVERRIDE {
     if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) {
       ++number_of_rtcp_packets_;
     } else {
       // Check that at least one RTCP packet was sent before the first RTP
       // packet.  This confirms that the receiver will have the necessary lip
       // sync info before it has to calculate the playout time of the first
       // frame.
       if (number_of_rtp_packets_ == 0)
         EXPECT_LE(1, number_of_rtcp_packets_);
@@ skipping 18 matching lines @@
   AudioSenderTest() {
     InitializeMediaLibraryForTesting();
     testing_clock_ = new base::SimpleTestTickClock();
     testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks());
     task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_);
     cast_environment_ =
         new CastEnvironment(scoped_ptr<base::TickClock>(testing_clock_).Pass(),
                             task_runner_,
                             task_runner_,
                             task_runner_);
-    audio_config_.codec = transport::CODEC_AUDIO_OPUS;
+    audio_config_.codec = CODEC_AUDIO_OPUS;
     audio_config_.use_external_encoder = false;
     audio_config_.frequency = kDefaultAudioSamplingRate;
     audio_config_.channels = 2;
     audio_config_.bitrate = kDefaultAudioEncoderBitrate;
     audio_config_.rtp_payload_type = 127;
 
     net::IPEndPoint dummy_endpoint;
 
-    transport_sender_.reset(new transport::CastTransportSenderImpl(
+    transport_sender_.reset(new CastTransportSenderImpl(
         NULL,
         testing_clock_,
         dummy_endpoint,
         base::Bind(&UpdateCastTransportStatus),
-        transport::BulkRawEventsCallback(),
+        BulkRawEventsCallback(),
         base::TimeDelta(),
         task_runner_,
         &transport_));
     audio_sender_.reset(new AudioSender(
         cast_environment_, audio_config_, transport_sender_.get()));
     task_runner_->RunTasks();
   }
 
   virtual ~AudioSenderTest() {}
 
-  static void UpdateCastTransportStatus(transport::CastTransportStatus status) {
-    EXPECT_EQ(transport::TRANSPORT_AUDIO_INITIALIZED, status);
+  static void UpdateCastTransportStatus(CastTransportStatus status) {
+    EXPECT_EQ(TRANSPORT_AUDIO_INITIALIZED, status);
   }
 
   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
   TestPacketSender transport_;
-  scoped_ptr<transport::CastTransportSenderImpl> transport_sender_;
+  scoped_ptr<CastTransportSenderImpl> transport_sender_;
   scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_;
   scoped_ptr<AudioSender> audio_sender_;
   scoped_refptr<CastEnvironment> cast_environment_;
   AudioSenderConfig audio_config_;
 };
 
 TEST_F(AudioSenderTest, Encode20ms) {
   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
   scoped_ptr<AudioBus> bus(
       TestAudioBusFactory(audio_config_.channels,
@@ skipping 22 matching lines @@
   base::TimeDelta max_rtcp_timeout =
       base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2);
   testing_clock_->Advance(max_rtcp_timeout);
   task_runner_->RunTasks();
   EXPECT_LE(1, transport_.number_of_rtp_packets());
   EXPECT_LE(1, transport_.number_of_rtcp_packets());
 }
 
 }  // namespace cast
 }  // namespace media