Cast: Reshuffle files under media/cast

media/cast/audio_sender/audio_sender_unittest.cc

Index: media/cast/audio_sender/audio_sender_unittest.cc
diff --git a/media/cast/audio_sender/audio_sender_unittest.cc b/media/cast/audio_sender/audio_sender_unittest.cc
deleted file mode 100644
index 69b6d85b83dfbe11f605fe4c304751e32f2d2634..0000000000000000000000000000000000000000
--- a/media/cast/audio_sender/audio_sender_unittest.cc
+++ /dev/null
@@ -1,141 +0,0 @@
-// 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/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 {
- public:
-  TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
-
-  virtual bool SendPacket(transport::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_);
-      ++number_of_rtp_packets_;
-    }
-    return true;
-  }
-
-  int number_of_rtp_packets() const { return number_of_rtp_packets_; }
-
-  int number_of_rtcp_packets() const { return number_of_rtcp_packets_; }
-
- private:
-  int number_of_rtp_packets_;
-  int number_of_rtcp_packets_;
-
-  DISALLOW_COPY_AND_ASSIGN(TestPacketSender);
-};
-
-class AudioSenderTest : public ::testing::Test {
- protected:
-  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_.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(
-        NULL,
-        testing_clock_,
-        dummy_endpoint,
-        base::Bind(&UpdateCastTransportStatus),
-        transport::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);
-  }
-
-  base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
-  TestPacketSender transport_;
-  scoped_ptr<transport::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,
-                          audio_config_.frequency,
-                          TestAudioBusFactory::kMiddleANoteFreq,
-                          0.5f).NextAudioBus(kDuration));
-
-  audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks());
-  task_runner_->RunTasks();
-  EXPECT_LE(1, transport_.number_of_rtp_packets());
-  EXPECT_LE(1, transport_.number_of_rtcp_packets());
-}
-
-TEST_F(AudioSenderTest, RtcpTimer) {
-  const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
-  scoped_ptr<AudioBus> bus(
-      TestAudioBusFactory(audio_config_.channels,
-                          audio_config_.frequency,
-                          TestAudioBusFactory::kMiddleANoteFreq,
-                          0.5f).NextAudioBus(kDuration));
-
-  audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks());
-  task_runner_->RunTasks();
-
-  // Make sure that we send at least one RTCP packet.
-  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