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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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