1d85771707d6a39a7cb306dd0836aab18dddd763
[platform/framework/web/crosswalk.git] / src / media / cast / audio_sender / audio_sender_unittest.cc
1 // Copyright 2013 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file.
4
5 #include "base/bind.h"
6 #include "base/bind_helpers.h"
7 #include "base/memory/scoped_ptr.h"
8 #include "base/test/simple_test_tick_clock.h"
9 #include "media/base/media.h"
10 #include "media/cast/audio_sender/audio_sender.h"
11 #include "media/cast/cast_config.h"
12 #include "media/cast/cast_environment.h"
13 #include "media/cast/rtcp/rtcp.h"
14 #include "media/cast/test/audio_utility.h"
15 #include "media/cast/test/fake_task_runner.h"
16 #include "media/cast/transport/cast_transport_config.h"
17 #include "media/cast/transport/cast_transport_sender_impl.h"
18 #include "testing/gmock/include/gmock/gmock.h"
19 #include "testing/gtest/include/gtest/gtest.h"
20
21 namespace media {
22 namespace cast {
23
24 static const int64 kStartMillisecond = GG_INT64_C(12345678900000);
25
26 using testing::_;
27 using testing::Exactly;
28
29 class TestPacketSender : public transport::PacketSender {
30  public:
31   TestPacketSender()
32       : number_of_rtp_packets_(0),
33         number_of_rtcp_packets_(0) {}
34
35   virtual bool SendPacket(const Packet& packet) OVERRIDE {
36     if (Rtcp::IsRtcpPacket(&packet[0], packet.size())) {
37       ++number_of_rtcp_packets_;
38     } else {
39       ++number_of_rtp_packets_;
40     }
41     return true;
42   }
43
44   int number_of_rtp_packets() const {
45     return number_of_rtp_packets_;
46   }
47
48   int number_of_rtcp_packets() const {
49     return number_of_rtcp_packets_;
50   }
51
52  private:
53   int number_of_rtp_packets_;
54   int number_of_rtcp_packets_;
55
56   DISALLOW_COPY_AND_ASSIGN(TestPacketSender);
57 };
58
59 class AudioSenderTest : public ::testing::Test {
60  public:
61   MOCK_METHOD0(InsertAudioCallback, void());
62
63  protected:
64   AudioSenderTest() {
65     InitializeMediaLibraryForTesting();
66     testing_clock_ = new base::SimpleTestTickClock();
67     testing_clock_->Advance(
68         base::TimeDelta::FromMilliseconds(kStartMillisecond));
69     task_runner_ = new test::FakeTaskRunner(testing_clock_);
70     cast_environment_ = new CastEnvironment(
71         scoped_ptr<base::TickClock>(testing_clock_).Pass(),
72         task_runner_, task_runner_, task_runner_, task_runner_,
73         task_runner_, task_runner_, GetDefaultCastSenderLoggingConfig());
74     audio_config_.codec = transport::kOpus;
75     audio_config_.use_external_encoder = false;
76     audio_config_.frequency = kDefaultAudioSamplingRate;
77     audio_config_.channels = 2;
78     audio_config_.bitrate = kDefaultAudioEncoderBitrate;
79     audio_config_.rtp_payload_type = 127;
80
81     transport::CastTransportConfig transport_config;
82     transport_config.audio_rtp_payload_type = 127;
83     transport_config.audio_channels = 2;
84     transport_sender_.reset(new transport::CastTransportSenderImpl(
85         testing_clock_,
86         transport_config,
87         base::Bind(&UpdateCastTransportStatus), task_runner_));
88     transport_sender_->InsertFakeTransportForTesting(&transport_);
89     audio_sender_.reset(new AudioSender(
90         cast_environment_, audio_config_, transport_sender_.get()));
91     task_runner_->RunTasks();
92   }
93
94   virtual ~AudioSenderTest() {}
95
96   static void UpdateCastTransportStatus(transport::CastTransportStatus status) {
97     EXPECT_EQ(status, transport::TRANSPORT_INITIALIZED);
98   }
99
100   base::SimpleTestTickClock* testing_clock_;  // Owned by CastEnvironment.
101   TestPacketSender transport_;
102   scoped_ptr<transport::CastTransportSenderImpl> transport_sender_;
103   scoped_refptr<test::FakeTaskRunner> task_runner_;
104   scoped_ptr<AudioSender> audio_sender_;
105   scoped_refptr<CastEnvironment> cast_environment_;
106   AudioSenderConfig audio_config_;
107 };
108
109 TEST_F(AudioSenderTest, Encode20ms) {
110   EXPECT_CALL(*this, InsertAudioCallback()).Times(Exactly(1));
111
112   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
113   scoped_ptr<AudioBus> bus(TestAudioBusFactory(
114       audio_config_.channels, audio_config_.frequency,
115       TestAudioBusFactory::kMiddleANoteFreq, 0.5f).NextAudioBus(kDuration));
116
117   base::TimeTicks recorded_time = base::TimeTicks::Now();
118   audio_sender_->InsertAudio(
119       bus.get(),
120       recorded_time,
121       base::Bind(
122           &AudioSenderTest::InsertAudioCallback,
123           base::Unretained(this)));
124   task_runner_->RunTasks();
125   EXPECT_GE(transport_.number_of_rtp_packets() +
126             transport_.number_of_rtcp_packets(), 1);
127 }
128
129 TEST_F(AudioSenderTest, RtcpTimer) {
130   EXPECT_CALL(*this, InsertAudioCallback()).Times(Exactly(1));
131
132   const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20);
133   scoped_ptr<AudioBus> bus(TestAudioBusFactory(
134       audio_config_.channels, audio_config_.frequency,
135       TestAudioBusFactory::kMiddleANoteFreq, 0.5f).NextAudioBus(kDuration));
136
137   base::TimeTicks recorded_time = base::TimeTicks::Now();
138   audio_sender_->InsertAudio(
139       bus.get(), recorded_time,
140       base::Bind(
141           &AudioSenderTest::InsertAudioCallback,
142           base::Unretained(this)));
143   task_runner_->RunTasks();
144
145   // Make sure that we send at least one RTCP packet.
146   base::TimeDelta max_rtcp_timeout =
147       base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2);
148   testing_clock_->Advance(max_rtcp_timeout);
149   task_runner_->RunTasks();
150   EXPECT_GE(transport_.number_of_rtp_packets(), 1);
151   EXPECT_EQ(transport_.number_of_rtcp_packets(), 1);
152 }
153
154 }  // namespace cast
155 }  // namespace media