| /* |
| * Copyright 2008 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #include "pc/channel_manager.h" |
| |
| #include <memory> |
| |
| #include "absl/memory/memory.h" |
| #include "api/media_transport_config.h" |
| #include "api/rtc_error.h" |
| #include "api/test/fake_media_transport.h" |
| #include "api/video/builtin_video_bitrate_allocator_factory.h" |
| #include "media/base/fake_media_engine.h" |
| #include "media/base/test_utils.h" |
| #include "media/engine/fake_webrtc_call.h" |
| #include "p2p/base/dtls_transport_internal.h" |
| #include "p2p/base/fake_dtls_transport.h" |
| #include "p2p/base/p2p_constants.h" |
| #include "p2p/base/packet_transport_internal.h" |
| #include "pc/dtls_srtp_transport.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/thread.h" |
| #include "test/gtest.h" |
| |
| namespace { |
| const bool kDefaultSrtpRequired = true; |
| } |
| |
| namespace cricket { |
| |
| static const AudioCodec kAudioCodecs[] = { |
| AudioCodec(97, "voice", 1, 2, 3), |
| AudioCodec(111, "OPUS", 48000, 32000, 2), |
| }; |
| |
| static const VideoCodec kVideoCodecs[] = { |
| VideoCodec(99, "H264"), |
| VideoCodec(100, "VP8"), |
| VideoCodec(96, "rtx"), |
| }; |
| |
| class ChannelManagerTest : public ::testing::Test { |
| protected: |
| ChannelManagerTest() |
| : network_(rtc::Thread::CreateWithSocketServer()), |
| worker_(rtc::Thread::Create()), |
| video_bitrate_allocator_factory_( |
| webrtc::CreateBuiltinVideoBitrateAllocatorFactory()), |
| fme_(new cricket::FakeMediaEngine()), |
| fdme_(new cricket::FakeDataEngine()), |
| cm_(new cricket::ChannelManager( |
| std::unique_ptr<MediaEngineInterface>(fme_), |
| std::unique_ptr<DataEngineInterface>(fdme_), |
| rtc::Thread::Current(), |
| rtc::Thread::Current())), |
| fake_call_() { |
| fme_->SetAudioCodecs(MAKE_VECTOR(kAudioCodecs)); |
| fme_->SetVideoCodecs(MAKE_VECTOR(kVideoCodecs)); |
| } |
| |
| std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() { |
| rtp_dtls_transport_ = absl::make_unique<FakeDtlsTransport>( |
| "fake_dtls_transport", cricket::ICE_CANDIDATE_COMPONENT_RTP); |
| auto dtls_srtp_transport = absl::make_unique<webrtc::DtlsSrtpTransport>( |
| /*rtcp_mux_required=*/true); |
| dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(), |
| /*rtcp_dtls_transport=*/nullptr); |
| return dtls_srtp_transport; |
| } |
| |
| std::unique_ptr<webrtc::MediaTransportInterface> CreateMediaTransport( |
| rtc::PacketTransportInternal* packet_transport) { |
| webrtc::MediaTransportSettings settings; |
| settings.is_caller = true; |
| auto media_transport_result = |
| fake_media_transport_factory_.CreateMediaTransport( |
| packet_transport, network_.get(), |
| /*is_caller=*/settings); |
| RTC_CHECK(media_transport_result.ok()); |
| return media_transport_result.MoveValue(); |
| } |
| |
| void TestCreateDestroyChannels( |
| webrtc::RtpTransportInternal* rtp_transport, |
| webrtc::MediaTransportConfig media_transport_config) { |
| cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel( |
| &fake_call_, cricket::MediaConfig(), rtp_transport, |
| media_transport_config, rtc::Thread::Current(), cricket::CN_AUDIO, |
| kDefaultSrtpRequired, webrtc::CryptoOptions(), &ssrc_generator_, |
| AudioOptions()); |
| EXPECT_TRUE(voice_channel != nullptr); |
| cricket::VideoChannel* video_channel = cm_->CreateVideoChannel( |
| &fake_call_, cricket::MediaConfig(), rtp_transport, |
| media_transport_config, rtc::Thread::Current(), cricket::CN_VIDEO, |
| kDefaultSrtpRequired, webrtc::CryptoOptions(), &ssrc_generator_, |
| VideoOptions(), video_bitrate_allocator_factory_.get()); |
| EXPECT_TRUE(video_channel != nullptr); |
| cricket::RtpDataChannel* rtp_data_channel = cm_->CreateRtpDataChannel( |
| cricket::MediaConfig(), rtp_transport, rtc::Thread::Current(), |
| cricket::CN_DATA, kDefaultSrtpRequired, webrtc::CryptoOptions(), |
| &ssrc_generator_); |
| EXPECT_TRUE(rtp_data_channel != nullptr); |
| cm_->DestroyVideoChannel(video_channel); |
| cm_->DestroyVoiceChannel(voice_channel); |
| cm_->DestroyRtpDataChannel(rtp_data_channel); |
| cm_->Terminate(); |
| } |
| |
| std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport_; |
| std::unique_ptr<rtc::Thread> network_; |
| std::unique_ptr<rtc::Thread> worker_; |
| std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> |
| video_bitrate_allocator_factory_; |
| // |fme_| and |fdme_| are actually owned by |cm_|. |
| cricket::FakeMediaEngine* fme_; |
| cricket::FakeDataEngine* fdme_; |
| std::unique_ptr<cricket::ChannelManager> cm_; |
| cricket::FakeCall fake_call_; |
| webrtc::FakeMediaTransportFactory fake_media_transport_factory_; |
| rtc::UniqueRandomIdGenerator ssrc_generator_; |
| }; |
| |
| // Test that we startup/shutdown properly. |
| TEST_F(ChannelManagerTest, StartupShutdown) { |
| EXPECT_FALSE(cm_->initialized()); |
| EXPECT_EQ(rtc::Thread::Current(), cm_->worker_thread()); |
| EXPECT_TRUE(cm_->Init()); |
| EXPECT_TRUE(cm_->initialized()); |
| cm_->Terminate(); |
| EXPECT_FALSE(cm_->initialized()); |
| } |
| |
| // Test that we startup/shutdown properly with a worker thread. |
| TEST_F(ChannelManagerTest, StartupShutdownOnThread) { |
| network_->Start(); |
| worker_->Start(); |
| EXPECT_FALSE(cm_->initialized()); |
| EXPECT_EQ(rtc::Thread::Current(), cm_->worker_thread()); |
| EXPECT_TRUE(cm_->set_network_thread(network_.get())); |
| EXPECT_EQ(network_.get(), cm_->network_thread()); |
| EXPECT_TRUE(cm_->set_worker_thread(worker_.get())); |
| EXPECT_EQ(worker_.get(), cm_->worker_thread()); |
| EXPECT_TRUE(cm_->Init()); |
| EXPECT_TRUE(cm_->initialized()); |
| // Setting the network or worker thread while initialized should fail. |
| EXPECT_FALSE(cm_->set_network_thread(rtc::Thread::Current())); |
| EXPECT_FALSE(cm_->set_worker_thread(rtc::Thread::Current())); |
| cm_->Terminate(); |
| EXPECT_FALSE(cm_->initialized()); |
| } |
| |
| TEST_F(ChannelManagerTest, SetVideoRtxEnabled) { |
| std::vector<VideoCodec> codecs; |
| const VideoCodec rtx_codec(96, "rtx"); |
| |
| // By default RTX is disabled. |
| cm_->GetSupportedVideoCodecs(&codecs); |
| EXPECT_FALSE(ContainsMatchingCodec(codecs, rtx_codec)); |
| |
| // Enable and check. |
| EXPECT_TRUE(cm_->SetVideoRtxEnabled(true)); |
| cm_->GetSupportedVideoCodecs(&codecs); |
| EXPECT_TRUE(ContainsMatchingCodec(codecs, rtx_codec)); |
| |
| // Disable and check. |
| EXPECT_TRUE(cm_->SetVideoRtxEnabled(false)); |
| cm_->GetSupportedVideoCodecs(&codecs); |
| EXPECT_FALSE(ContainsMatchingCodec(codecs, rtx_codec)); |
| |
| // Cannot toggle rtx after initialization. |
| EXPECT_TRUE(cm_->Init()); |
| EXPECT_FALSE(cm_->SetVideoRtxEnabled(true)); |
| EXPECT_FALSE(cm_->SetVideoRtxEnabled(false)); |
| |
| // Can set again after terminate. |
| cm_->Terminate(); |
| EXPECT_TRUE(cm_->SetVideoRtxEnabled(true)); |
| cm_->GetSupportedVideoCodecs(&codecs); |
| EXPECT_TRUE(ContainsMatchingCodec(codecs, rtx_codec)); |
| } |
| |
| TEST_F(ChannelManagerTest, CreateDestroyChannels) { |
| EXPECT_TRUE(cm_->Init()); |
| auto rtp_transport = CreateDtlsSrtpTransport(); |
| TestCreateDestroyChannels(rtp_transport.get(), |
| webrtc::MediaTransportConfig()); |
| } |
| |
| TEST_F(ChannelManagerTest, CreateDestroyChannelsWithMediaTransport) { |
| EXPECT_TRUE(cm_->Init()); |
| auto rtp_transport = CreateDtlsSrtpTransport(); |
| auto media_transport = CreateMediaTransport(rtp_dtls_transport_.get()); |
| TestCreateDestroyChannels( |
| rtp_transport.get(), webrtc::MediaTransportConfig(media_transport.get())); |
| } |
| |
| TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) { |
| network_->Start(); |
| worker_->Start(); |
| EXPECT_TRUE(cm_->set_worker_thread(worker_.get())); |
| EXPECT_TRUE(cm_->set_network_thread(network_.get())); |
| EXPECT_TRUE(cm_->Init()); |
| auto rtp_transport = CreateDtlsSrtpTransport(); |
| TestCreateDestroyChannels(rtp_transport.get(), |
| webrtc::MediaTransportConfig()); |
| } |
| |
| } // namespace cricket |