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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 <stdint.h> | |
6 | |
7 #include "base/bind.h" | |
8 #include "base/bind_helpers.h" | |
9 #include "base/memory/scoped_ptr.h" | |
10 #include "base/test/simple_test_tick_clock.h" | |
11 #include "media/base/media.h" | |
12 #include "media/cast/audio_sender/audio_sender.h" | |
13 #include "media/cast/cast_config.h" | |
14 #include "media/cast/cast_environment.h" | |
15 #include "media/cast/rtcp/rtcp.h" | |
16 #include "media/cast/test/fake_single_thread_task_runner.h" | |
17 #include "media/cast/test/utility/audio_utility.h" | |
18 #include "media/cast/transport/cast_transport_config.h" | |
19 #include "media/cast/transport/cast_transport_sender_impl.h" | |
20 #include "testing/gtest/include/gtest/gtest.h" | |
21 | |
22 namespace media { | |
23 namespace cast { | |
24 | |
25 class TestPacketSender : public transport::PacketSender { | |
26 public: | |
27 TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {} | |
28 | |
29 virtual bool SendPacket(transport::PacketRef packet, | |
30 const base::Closure& cb) OVERRIDE { | |
31 if (Rtcp::IsRtcpPacket(&packet->data[0], packet->data.size())) { | |
32 ++number_of_rtcp_packets_; | |
33 } else { | |
34 // Check that at least one RTCP packet was sent before the first RTP | |
35 // packet. This confirms that the receiver will have the necessary lip | |
36 // sync info before it has to calculate the playout time of the first | |
37 // frame. | |
38 if (number_of_rtp_packets_ == 0) | |
39 EXPECT_LE(1, number_of_rtcp_packets_); | |
40 ++number_of_rtp_packets_; | |
41 } | |
42 return true; | |
43 } | |
44 | |
45 int number_of_rtp_packets() const { return number_of_rtp_packets_; } | |
46 | |
47 int number_of_rtcp_packets() const { return number_of_rtcp_packets_; } | |
48 | |
49 private: | |
50 int number_of_rtp_packets_; | |
51 int number_of_rtcp_packets_; | |
52 | |
53 DISALLOW_COPY_AND_ASSIGN(TestPacketSender); | |
54 }; | |
55 | |
56 class AudioSenderTest : public ::testing::Test { | |
57 protected: | |
58 AudioSenderTest() { | |
59 InitializeMediaLibraryForTesting(); | |
60 testing_clock_ = new base::SimpleTestTickClock(); | |
61 testing_clock_->Advance(base::TimeTicks::Now() - base::TimeTicks()); | |
62 task_runner_ = new test::FakeSingleThreadTaskRunner(testing_clock_); | |
63 cast_environment_ = | |
64 new CastEnvironment(scoped_ptr<base::TickClock>(testing_clock_).Pass(), | |
65 task_runner_, | |
66 task_runner_, | |
67 task_runner_); | |
68 audio_config_.codec = transport::CODEC_AUDIO_OPUS; | |
69 audio_config_.use_external_encoder = false; | |
70 audio_config_.frequency = kDefaultAudioSamplingRate; | |
71 audio_config_.channels = 2; | |
72 audio_config_.bitrate = kDefaultAudioEncoderBitrate; | |
73 audio_config_.rtp_payload_type = 127; | |
74 | |
75 net::IPEndPoint dummy_endpoint; | |
76 | |
77 transport_sender_.reset(new transport::CastTransportSenderImpl( | |
78 NULL, | |
79 testing_clock_, | |
80 dummy_endpoint, | |
81 base::Bind(&UpdateCastTransportStatus), | |
82 transport::BulkRawEventsCallback(), | |
83 base::TimeDelta(), | |
84 task_runner_, | |
85 &transport_)); | |
86 audio_sender_.reset(new AudioSender( | |
87 cast_environment_, audio_config_, transport_sender_.get())); | |
88 task_runner_->RunTasks(); | |
89 } | |
90 | |
91 virtual ~AudioSenderTest() {} | |
92 | |
93 static void UpdateCastTransportStatus(transport::CastTransportStatus status) { | |
94 EXPECT_EQ(transport::TRANSPORT_AUDIO_INITIALIZED, status); | |
95 } | |
96 | |
97 base::SimpleTestTickClock* testing_clock_; // Owned by CastEnvironment. | |
98 TestPacketSender transport_; | |
99 scoped_ptr<transport::CastTransportSenderImpl> transport_sender_; | |
100 scoped_refptr<test::FakeSingleThreadTaskRunner> task_runner_; | |
101 scoped_ptr<AudioSender> audio_sender_; | |
102 scoped_refptr<CastEnvironment> cast_environment_; | |
103 AudioSenderConfig audio_config_; | |
104 }; | |
105 | |
106 TEST_F(AudioSenderTest, Encode20ms) { | |
107 const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20); | |
108 scoped_ptr<AudioBus> bus( | |
109 TestAudioBusFactory(audio_config_.channels, | |
110 audio_config_.frequency, | |
111 TestAudioBusFactory::kMiddleANoteFreq, | |
112 0.5f).NextAudioBus(kDuration)); | |
113 | |
114 audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks()); | |
115 task_runner_->RunTasks(); | |
116 EXPECT_LE(1, transport_.number_of_rtp_packets()); | |
117 EXPECT_LE(1, transport_.number_of_rtcp_packets()); | |
118 } | |
119 | |
120 TEST_F(AudioSenderTest, RtcpTimer) { | |
121 const base::TimeDelta kDuration = base::TimeDelta::FromMilliseconds(20); | |
122 scoped_ptr<AudioBus> bus( | |
123 TestAudioBusFactory(audio_config_.channels, | |
124 audio_config_.frequency, | |
125 TestAudioBusFactory::kMiddleANoteFreq, | |
126 0.5f).NextAudioBus(kDuration)); | |
127 | |
128 audio_sender_->InsertAudio(bus.Pass(), testing_clock_->NowTicks()); | |
129 task_runner_->RunTasks(); | |
130 | |
131 // Make sure that we send at least one RTCP packet. | |
132 base::TimeDelta max_rtcp_timeout = | |
133 base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2); | |
134 testing_clock_->Advance(max_rtcp_timeout); | |
135 task_runner_->RunTasks(); | |
136 EXPECT_LE(1, transport_.number_of_rtp_packets()); | |
137 EXPECT_LE(1, transport_.number_of_rtcp_packets()); | |
138 } | |
139 | |
140 } // namespace cast | |
141 } // namespace media | |
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