| // 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::kOpus; |
| audio_config_.use_external_encoder = false; |
| audio_config_.frequency = kDefaultAudioSamplingRate; |
| audio_config_.channels = 2; |
| audio_config_.bitrate = kDefaultAudioEncoderBitrate; |
| audio_config_.rtp_config.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 |