| /* |
| * Copyright (c) 2014 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 "test/call_test.h" |
| |
| #include <algorithm> |
| #include <memory> |
| |
| #include "api/audio/audio_device.h" |
| #include "api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "api/environment/environment.h" |
| #include "api/environment/environment_factory.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/test/create_frame_generator.h" |
| #include "api/video/builtin_video_bitrate_allocator_factory.h" |
| #include "call/fake_network_pipe.h" |
| #include "call/packet_receiver.h" |
| #include "modules/audio_device/include/test_audio_device.h" |
| #include "modules/audio_mixer/audio_mixer_impl.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/event.h" |
| #include "rtc_base/task_queue_for_test.h" |
| #include "test/fake_encoder.h" |
| #include "test/network/simulated_network.h" |
| #include "test/rtp_rtcp_observer.h" |
| #include "test/testsupport/file_utils.h" |
| #include "test/video_test_constants.h" |
| #include "video/config/video_encoder_config.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| CallTest::CallTest() |
| : env_(CreateEnvironment(&field_trials_)), |
| send_env_(env_), |
| recv_env_(env_), |
| audio_send_config_(/*send_transport=*/nullptr), |
| audio_send_stream_(nullptr), |
| frame_generator_capturer_(nullptr), |
| fake_encoder_factory_( |
| [this](const Environment& env, const SdpVideoFormat& format) { |
| std::unique_ptr<FakeEncoder> fake_encoder; |
| if (video_encoder_configs_[0].codec_type == kVideoCodecVP8) { |
| fake_encoder = std::make_unique<FakeVp8Encoder>(env); |
| } else { |
| fake_encoder = std::make_unique<FakeEncoder>(env); |
| } |
| fake_encoder->SetMaxBitrate(fake_encoder_max_bitrate_); |
| return fake_encoder; |
| }), |
| fake_decoder_factory_([]() { return std::make_unique<FakeDecoder>(); }), |
| bitrate_allocator_factory_(CreateBuiltinVideoBitrateAllocatorFactory()), |
| num_video_streams_(1), |
| num_audio_streams_(0), |
| num_flexfec_streams_(0), |
| audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()), |
| audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()), |
| task_queue_(env_.task_queue_factory().CreateTaskQueue( |
| "CallTestTaskQueue", |
| TaskQueueFactory::Priority::NORMAL)) {} |
| |
| CallTest::~CallTest() = default; |
| |
| void CallTest::SetSendEventLog(std::unique_ptr<RtcEventLog> event_log) { |
| EnvironmentFactory f(env_); |
| f.Set(std::move(event_log)); |
| send_env_ = f.Create(); |
| } |
| |
| void CallTest::SetRecvEventLog(std::unique_ptr<RtcEventLog> event_log) { |
| EnvironmentFactory f(env_); |
| f.Set(std::move(event_log)); |
| recv_env_ = f.Create(); |
| } |
| |
| void CallTest::RegisterRtpExtension(const RtpExtension& extension) { |
| for (const RtpExtension& registered_extension : rtp_extensions_) { |
| if (registered_extension.id == extension.id) { |
| ASSERT_EQ(registered_extension.uri, extension.uri) |
| << "Different URIs associated with ID " << extension.id << "."; |
| ASSERT_EQ(registered_extension.encrypt, extension.encrypt) |
| << "Encryption mismatch associated with ID " << extension.id << "."; |
| return; |
| } else { // Different IDs. |
| // Different IDs referring to the same extension probably indicate |
| // a mistake in the test. |
| ASSERT_FALSE(registered_extension.uri == extension.uri && |
| registered_extension.encrypt == extension.encrypt) |
| << "URI " << extension.uri |
| << (extension.encrypt ? " with " : " without ") |
| << "encryption already registered with a different " |
| "ID (" |
| << extension.id << " vs. " << registered_extension.id << ")."; |
| } |
| } |
| rtp_extensions_.push_back(extension); |
| } |
| |
| void CallTest::RunBaseTest(BaseTest* test) { |
| SendTask(task_queue(), [this, test]() { |
| num_video_streams_ = test->GetNumVideoStreams(); |
| num_audio_streams_ = test->GetNumAudioStreams(); |
| num_flexfec_streams_ = test->GetNumFlexfecStreams(); |
| RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0); |
| CallConfig send_config = SendCallConfig(); |
| test->ModifySenderBitrateConfig(&send_config.bitrate_config); |
| if (num_audio_streams_ > 0) { |
| CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer()); |
| test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(), |
| fake_recv_audio_device_.get()); |
| apm_send_ = AudioProcessingBuilder().Create(); |
| apm_recv_ = AudioProcessingBuilder().Create(); |
| EXPECT_EQ(0, fake_send_audio_device_->Init()); |
| EXPECT_EQ(0, fake_recv_audio_device_->Init()); |
| AudioState::Config audio_state_config; |
| audio_state_config.audio_mixer = AudioMixerImpl::Create(); |
| audio_state_config.audio_processing = apm_send_; |
| audio_state_config.audio_device_module = fake_send_audio_device_; |
| send_config.audio_state = AudioState::Create(audio_state_config); |
| fake_send_audio_device_->RegisterAudioCallback( |
| send_config.audio_state->audio_transport()); |
| } |
| CreateSenderCall(std::move(send_config)); |
| if (test->ShouldCreateReceivers()) { |
| CallConfig recv_config = RecvCallConfig(); |
| test->ModifyReceiverBitrateConfig(&recv_config.bitrate_config); |
| if (num_audio_streams_ > 0) { |
| AudioState::Config audio_state_config; |
| audio_state_config.audio_mixer = AudioMixerImpl::Create(); |
| audio_state_config.audio_processing = apm_recv_; |
| audio_state_config.audio_device_module = fake_recv_audio_device_; |
| recv_config.audio_state = AudioState::Create(audio_state_config); |
| fake_recv_audio_device_->RegisterAudioCallback( |
| recv_config.audio_state->audio_transport()); |
| } |
| CreateReceiverCall(std::move(recv_config)); |
| } |
| test->OnCallsCreated(sender_call_.get(), receiver_call_.get()); |
| CreateReceiveTransport(test->GetReceiveTransportConfig(), test); |
| CreateSendTransport(test->GetSendTransportConfig(), test); |
| test->OnTransportCreated(send_transport_.get(), send_simulated_network_, |
| receive_transport_.get(), |
| receive_simulated_network_); |
| if (test->ShouldCreateReceivers()) { |
| if (num_video_streams_ > 0) |
| receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp); |
| if (num_audio_streams_ > 0) |
| receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp); |
| } else { |
| // Sender-only call delivers to itself. |
| send_transport_->SetReceiver(sender_call_->Receiver()); |
| receive_transport_->SetReceiver(nullptr); |
| } |
| |
| CreateSendConfig(num_video_streams_, num_audio_streams_, |
| num_flexfec_streams_, send_transport_.get()); |
| if (test->ShouldCreateReceivers()) { |
| CreateMatchingReceiveConfigs(); |
| } |
| if (num_video_streams_ > 0) { |
| test->ModifyVideoConfigs(GetVideoSendConfig(), &video_receive_configs_, |
| GetVideoEncoderConfig()); |
| } |
| if (num_audio_streams_ > 0) { |
| test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_); |
| } |
| if (num_flexfec_streams_ > 0) { |
| test->ModifyFlexfecConfigs(&flexfec_receive_configs_); |
| } |
| |
| if (num_flexfec_streams_ > 0) { |
| CreateFlexfecStreams(); |
| test->OnFlexfecStreamsCreated(flexfec_receive_streams_); |
| } |
| if (num_video_streams_ > 0) { |
| CreateVideoStreams(); |
| test->OnVideoStreamsCreated(GetVideoSendStream(), video_receive_streams_); |
| } |
| if (num_audio_streams_ > 0) { |
| CreateAudioStreams(); |
| test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_); |
| } |
| |
| if (num_video_streams_ > 0) { |
| int width = VideoTestConstants::kDefaultWidth; |
| int height = VideoTestConstants::kDefaultHeight; |
| int frame_rate = VideoTestConstants::kDefaultFramerate; |
| test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate); |
| test->ModifyVideoDegradationPreference(°radation_preference_); |
| CreateFrameGeneratorCapturer(frame_rate, width, height); |
| test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_); |
| } |
| |
| Start(); |
| }); |
| |
| test->PerformTest(); |
| |
| SendTask(task_queue(), [this, test]() { |
| Stop(); |
| test->OnStreamsStopped(); |
| DestroyStreams(); |
| send_transport_.reset(); |
| receive_transport_.reset(); |
| |
| frame_generator_capturer_ = nullptr; |
| DestroyCalls(); |
| |
| fake_send_audio_device_ = nullptr; |
| fake_recv_audio_device_ = nullptr; |
| }); |
| } |
| |
| CallConfig CallTest::SendCallConfig() const { |
| CallConfig sender_config(send_env_); |
| sender_config.network_state_predictor_factory = |
| network_state_predictor_factory_.get(); |
| sender_config.network_controller_factory = network_controller_factory_.get(); |
| return sender_config; |
| } |
| |
| CallConfig CallTest::RecvCallConfig() const { |
| return CallConfig(recv_env_); |
| } |
| |
| void CallTest::CreateCalls() { |
| CreateCalls(SendCallConfig(), RecvCallConfig()); |
| } |
| |
| void CallTest::CreateCalls(CallConfig sender_config, |
| CallConfig receiver_config) { |
| CreateSenderCall(std::move(sender_config)); |
| CreateReceiverCall(std::move(receiver_config)); |
| } |
| |
| void CallTest::CreateSenderCall() { |
| CreateSenderCall(SendCallConfig()); |
| } |
| |
| void CallTest::CreateSenderCall(CallConfig config) { |
| sender_call_ = Call::Create(std::move(config)); |
| } |
| |
| void CallTest::CreateReceiverCall(CallConfig config) { |
| receiver_call_ = Call::Create(std::move(config)); |
| } |
| |
| void CallTest::DestroyCalls() { |
| send_transport_.reset(); |
| receive_transport_.reset(); |
| sender_call_.reset(); |
| receiver_call_.reset(); |
| } |
| |
| void CallTest::CreateVideoSendConfig(VideoSendStream::Config* video_config, |
| size_t num_video_streams, |
| size_t num_used_ssrcs, |
| Transport* send_transport) { |
| RTC_DCHECK_LE(num_video_streams + num_used_ssrcs, |
| VideoTestConstants::kNumSsrcs); |
| *video_config = VideoSendStream::Config(send_transport); |
| video_config->encoder_settings.encoder_factory = &fake_encoder_factory_; |
| video_config->encoder_settings.bitrate_allocator_factory = |
| bitrate_allocator_factory_.get(); |
| video_config->rtp.payload_name = "FAKE"; |
| video_config->rtp.payload_type = |
| VideoTestConstants::kFakeVideoSendPayloadType; |
| video_config->rtp.extmap_allow_mixed = true; |
| AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri, |
| &video_config->rtp.extensions); |
| AddRtpExtensionByUri(RtpExtension::kAbsSendTimeUri, |
| &video_config->rtp.extensions); |
| AddRtpExtensionByUri(RtpExtension::kTimestampOffsetUri, |
| &video_config->rtp.extensions); |
| AddRtpExtensionByUri(RtpExtension::kVideoContentTypeUri, |
| &video_config->rtp.extensions); |
| AddRtpExtensionByUri(RtpExtension::kGenericFrameDescriptorUri00, |
| &video_config->rtp.extensions); |
| AddRtpExtensionByUri(RtpExtension::kDependencyDescriptorUri, |
| &video_config->rtp.extensions); |
| if (video_encoder_configs_.empty()) { |
| video_encoder_configs_.emplace_back(); |
| FillEncoderConfiguration(kVideoCodecGeneric, num_video_streams, |
| &video_encoder_configs_.back()); |
| } |
| for (size_t i = 0; i < num_video_streams; ++i) |
| video_config->rtp.ssrcs.push_back( |
| VideoTestConstants::kVideoSendSsrcs[num_used_ssrcs + i]); |
| AddRtpExtensionByUri(RtpExtension::kVideoRotationUri, |
| &video_config->rtp.extensions); |
| AddRtpExtensionByUri(RtpExtension::kColorSpaceUri, |
| &video_config->rtp.extensions); |
| } |
| |
| void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams, |
| size_t num_flexfec_streams, |
| Transport* send_transport) { |
| RTC_DCHECK_LE(num_audio_streams, 1); |
| RTC_DCHECK_LE(num_flexfec_streams, 1); |
| if (num_audio_streams > 0) { |
| AudioSendStream::Config audio_send_config(send_transport); |
| audio_send_config.rtp.ssrc = VideoTestConstants::kAudioSendSsrc; |
| AddRtpExtensionByUri(RtpExtension::kTransportSequenceNumberUri, |
| &audio_send_config.rtp.extensions); |
| |
| audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec( |
| VideoTestConstants::kAudioSendPayloadType, |
| {"opus", 48000, 2, {{"stereo", "1"}}}); |
| audio_send_config.min_bitrate_bps = 6000; |
| audio_send_config.max_bitrate_bps = 60000; |
| audio_send_config.encoder_factory = audio_encoder_factory_; |
| SetAudioConfig(audio_send_config); |
| } |
| |
| // TODO(brandtr): Update this when we support multistream protection. |
| if (num_flexfec_streams > 0) { |
| SetSendFecConfig({VideoTestConstants::kVideoSendSsrcs[0]}); |
| } |
| } |
| |
| void CallTest::SetAudioConfig(const AudioSendStream::Config& config) { |
| audio_send_config_ = config; |
| } |
| |
| void CallTest::SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs) { |
| GetVideoSendConfig()->rtp.flexfec.payload_type = |
| VideoTestConstants::kFlexfecPayloadType; |
| GetVideoSendConfig()->rtp.flexfec.ssrc = VideoTestConstants::kFlexfecSendSsrc; |
| GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs = video_send_ssrcs; |
| } |
| |
| void CallTest::SetSendUlpFecConfig(VideoSendStream::Config* send_config) { |
| send_config->rtp.ulpfec.red_payload_type = |
| VideoTestConstants::kRedPayloadType; |
| send_config->rtp.ulpfec.ulpfec_payload_type = |
| VideoTestConstants::kUlpfecPayloadType; |
| send_config->rtp.ulpfec.red_rtx_payload_type = |
| VideoTestConstants::kRtxRedPayloadType; |
| } |
| |
| void CallTest::SetReceiveUlpFecConfig( |
| VideoReceiveStreamInterface::Config* receive_config) { |
| receive_config->rtp.red_payload_type = VideoTestConstants::kRedPayloadType; |
| receive_config->rtp.ulpfec_payload_type = |
| VideoTestConstants::kUlpfecPayloadType; |
| receive_config->rtp |
| .rtx_associated_payload_types[VideoTestConstants::kRtxRedPayloadType] = |
| VideoTestConstants::kRedPayloadType; |
| } |
| |
| void CallTest::CreateSendConfig(size_t num_video_streams, |
| size_t num_audio_streams, |
| size_t num_flexfec_streams, |
| Transport* send_transport) { |
| if (num_video_streams > 0) { |
| video_send_configs_.clear(); |
| video_send_configs_.emplace_back(nullptr); |
| CreateVideoSendConfig(&video_send_configs_.back(), num_video_streams, 0, |
| send_transport); |
| } |
| CreateAudioAndFecSendConfigs(num_audio_streams, num_flexfec_streams, |
| send_transport); |
| } |
| |
| void CallTest::CreateMatchingVideoReceiveConfigs( |
| const VideoSendStream::Config& video_send_config, |
| Transport* rtcp_send_transport) { |
| CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport, |
| &fake_decoder_factory_, absl::nullopt, |
| false, 0); |
| } |
| |
| void CallTest::CreateMatchingVideoReceiveConfigs( |
| const VideoSendStream::Config& video_send_config, |
| Transport* rtcp_send_transport, |
| VideoDecoderFactory* decoder_factory, |
| absl::optional<size_t> decode_sub_stream, |
| bool receiver_reference_time_report, |
| int rtp_history_ms) { |
| AddMatchingVideoReceiveConfigs( |
| &video_receive_configs_, video_send_config, rtcp_send_transport, |
| decoder_factory, decode_sub_stream, receiver_reference_time_report, |
| rtp_history_ms); |
| } |
| |
| void CallTest::AddMatchingVideoReceiveConfigs( |
| std::vector<VideoReceiveStreamInterface::Config>* receive_configs, |
| const VideoSendStream::Config& video_send_config, |
| Transport* rtcp_send_transport, |
| VideoDecoderFactory* decoder_factory, |
| absl::optional<size_t> decode_sub_stream, |
| bool receiver_reference_time_report, |
| int rtp_history_ms) { |
| RTC_DCHECK(!video_send_config.rtp.ssrcs.empty()); |
| VideoReceiveStreamInterface::Config default_config(rtcp_send_transport); |
| default_config.rtp.local_ssrc = VideoTestConstants::kReceiverLocalVideoSsrc; |
| default_config.rtp.nack.rtp_history_ms = rtp_history_ms; |
| // Enable RTT calculation so NTP time estimator will work. |
| default_config.rtp.rtcp_xr.receiver_reference_time_report = |
| receiver_reference_time_report; |
| default_config.renderer = &fake_renderer_; |
| |
| for (size_t i = 0; i < video_send_config.rtp.ssrcs.size(); ++i) { |
| VideoReceiveStreamInterface::Config video_recv_config( |
| default_config.Copy()); |
| video_recv_config.decoders.clear(); |
| if (!video_send_config.rtp.rtx.ssrcs.empty()) { |
| video_recv_config.rtp.rtx_ssrc = video_send_config.rtp.rtx.ssrcs[i]; |
| video_recv_config.rtp.rtx_associated_payload_types |
| [VideoTestConstants::kSendRtxPayloadType] = |
| video_send_config.rtp.payload_type; |
| } |
| video_recv_config.rtp.remote_ssrc = video_send_config.rtp.ssrcs[i]; |
| VideoReceiveStreamInterface::Decoder decoder; |
| |
| decoder.payload_type = video_send_config.rtp.payload_type; |
| decoder.video_format = SdpVideoFormat(video_send_config.rtp.payload_name); |
| // Force fake decoders on non-selected simulcast streams. |
| if (!decode_sub_stream || i == *decode_sub_stream) { |
| video_recv_config.decoder_factory = decoder_factory; |
| } else { |
| video_recv_config.decoder_factory = &fake_decoder_factory_; |
| } |
| video_recv_config.decoders.push_back(decoder); |
| receive_configs->emplace_back(std::move(video_recv_config)); |
| } |
| } |
| |
| void CallTest::CreateMatchingAudioAndFecConfigs( |
| Transport* rtcp_send_transport) { |
| RTC_DCHECK_GE(1, num_audio_streams_); |
| if (num_audio_streams_ == 1) { |
| CreateMatchingAudioConfigs(rtcp_send_transport, ""); |
| } |
| |
| // TODO(brandtr): Update this when we support multistream protection. |
| RTC_DCHECK(num_flexfec_streams_ <= 1); |
| if (num_flexfec_streams_ == 1) { |
| CreateMatchingFecConfig(rtcp_send_transport, *GetVideoSendConfig()); |
| } |
| } |
| |
| void CallTest::CreateMatchingAudioConfigs(Transport* transport, |
| std::string sync_group) { |
| audio_receive_configs_.push_back(CreateMatchingAudioConfig( |
| audio_send_config_, audio_decoder_factory_, transport, sync_group)); |
| } |
| |
| AudioReceiveStreamInterface::Config CallTest::CreateMatchingAudioConfig( |
| const AudioSendStream::Config& send_config, |
| rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory, |
| Transport* transport, |
| std::string sync_group) { |
| AudioReceiveStreamInterface::Config audio_config; |
| audio_config.rtp.local_ssrc = VideoTestConstants::kReceiverLocalAudioSsrc; |
| audio_config.rtcp_send_transport = transport; |
| audio_config.rtp.remote_ssrc = send_config.rtp.ssrc; |
| audio_config.decoder_factory = audio_decoder_factory; |
| audio_config.decoder_map = { |
| {VideoTestConstants::kAudioSendPayloadType, {"opus", 48000, 2}}}; |
| audio_config.sync_group = sync_group; |
| return audio_config; |
| } |
| |
| void CallTest::CreateMatchingFecConfig( |
| Transport* transport, |
| const VideoSendStream::Config& send_config) { |
| FlexfecReceiveStream::Config config(transport); |
| config.payload_type = send_config.rtp.flexfec.payload_type; |
| config.rtp.remote_ssrc = send_config.rtp.flexfec.ssrc; |
| config.protected_media_ssrcs = send_config.rtp.flexfec.protected_media_ssrcs; |
| config.rtp.local_ssrc = VideoTestConstants::kReceiverLocalVideoSsrc; |
| if (!video_receive_configs_.empty()) { |
| video_receive_configs_[0].rtp.protected_by_flexfec = true; |
| video_receive_configs_[0].rtp.packet_sink_ = this; |
| } |
| flexfec_receive_configs_.push_back(config); |
| } |
| |
| void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) { |
| video_receive_configs_.clear(); |
| for (VideoSendStream::Config& video_send_config : video_send_configs_) { |
| CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport); |
| } |
| CreateMatchingAudioAndFecConfigs(rtcp_send_transport); |
| } |
| |
| void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock, |
| float speed, |
| int framerate, |
| int width, |
| int height) { |
| video_sources_.clear(); |
| auto frame_generator_capturer = |
| std::make_unique<test::FrameGeneratorCapturer>( |
| clock, |
| test::CreateSquareFrameGenerator(width, height, absl::nullopt, |
| absl::nullopt), |
| framerate * speed, env_.task_queue_factory()); |
| frame_generator_capturer_ = frame_generator_capturer.get(); |
| frame_generator_capturer->Init(); |
| video_sources_.push_back(std::move(frame_generator_capturer)); |
| ConnectVideoSourcesToStreams(); |
| } |
| |
| void CallTest::CreateFrameGeneratorCapturer(int framerate, |
| int width, |
| int height) { |
| video_sources_.clear(); |
| auto frame_generator_capturer = |
| std::make_unique<test::FrameGeneratorCapturer>( |
| &env_.clock(), |
| test::CreateSquareFrameGenerator(width, height, absl::nullopt, |
| absl::nullopt), |
| framerate, env_.task_queue_factory()); |
| frame_generator_capturer_ = frame_generator_capturer.get(); |
| frame_generator_capturer->Init(); |
| video_sources_.push_back(std::move(frame_generator_capturer)); |
| ConnectVideoSourcesToStreams(); |
| } |
| |
| void CallTest::CreateFakeAudioDevices( |
| std::unique_ptr<TestAudioDeviceModule::Capturer> capturer, |
| std::unique_ptr<TestAudioDeviceModule::Renderer> renderer) { |
| fake_send_audio_device_ = TestAudioDeviceModule::Create( |
| &env_.task_queue_factory(), std::move(capturer), nullptr, 1.f); |
| fake_recv_audio_device_ = TestAudioDeviceModule::Create( |
| &env_.task_queue_factory(), nullptr, std::move(renderer), 1.f); |
| } |
| |
| void CallTest::CreateVideoStreams() { |
| RTC_DCHECK(video_receive_streams_.empty()); |
| CreateVideoSendStreams(); |
| for (size_t i = 0; i < video_receive_configs_.size(); ++i) { |
| video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream( |
| video_receive_configs_[i].Copy())); |
| } |
| } |
| |
| void CallTest::CreateVideoSendStreams() { |
| RTC_DCHECK(video_send_streams_.empty()); |
| |
| // We currently only support testing external fec controllers with a single |
| // VideoSendStream. |
| if (fec_controller_factory_.get()) { |
| RTC_DCHECK_LE(video_send_configs_.size(), 1); |
| } |
| |
| // TODO(http://crbug/818127): |
| // Remove this workaround when ALR is not screenshare-specific. |
| std::list<size_t> streams_creation_order; |
| for (size_t i = 0; i < video_send_configs_.size(); ++i) { |
| // If dual streams are created, add the screenshare stream last. |
| if (video_encoder_configs_[i].content_type == |
| VideoEncoderConfig::ContentType::kScreen) { |
| streams_creation_order.push_back(i); |
| } else { |
| streams_creation_order.push_front(i); |
| } |
| } |
| |
| video_send_streams_.resize(video_send_configs_.size(), nullptr); |
| |
| for (size_t i : streams_creation_order) { |
| if (fec_controller_factory_.get()) { |
| video_send_streams_[i] = sender_call_->CreateVideoSendStream( |
| video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy(), |
| fec_controller_factory_->CreateFecController(send_env_)); |
| } else { |
| video_send_streams_[i] = sender_call_->CreateVideoSendStream( |
| video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy()); |
| } |
| } |
| } |
| |
| void CallTest::CreateVideoSendStream(const VideoEncoderConfig& encoder_config) { |
| RTC_DCHECK(video_send_streams_.empty()); |
| video_send_streams_.push_back(sender_call_->CreateVideoSendStream( |
| GetVideoSendConfig()->Copy(), encoder_config.Copy())); |
| } |
| |
| void CallTest::CreateAudioStreams() { |
| RTC_DCHECK(audio_send_stream_ == nullptr); |
| RTC_DCHECK(audio_receive_streams_.empty()); |
| audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_); |
| for (size_t i = 0; i < audio_receive_configs_.size(); ++i) { |
| audio_receive_streams_.push_back( |
| receiver_call_->CreateAudioReceiveStream(audio_receive_configs_[i])); |
| } |
| } |
| |
| void CallTest::CreateFlexfecStreams() { |
| for (size_t i = 0; i < flexfec_receive_configs_.size(); ++i) { |
| flexfec_receive_streams_.push_back( |
| receiver_call_->CreateFlexfecReceiveStream( |
| flexfec_receive_configs_[i])); |
| } |
| } |
| |
| void CallTest::CreateSendTransport(const BuiltInNetworkBehaviorConfig& config, |
| RtpRtcpObserver* observer) { |
| PacketReceiver* receiver = |
| receiver_call_ ? receiver_call_->Receiver() : nullptr; |
| |
| auto network = std::make_unique<SimulatedNetwork>(config); |
| send_simulated_network_ = network.get(); |
| send_transport_ = std::make_unique<PacketTransport>( |
| task_queue(), sender_call_.get(), observer, |
| test::PacketTransport::kSender, payload_type_map_, |
| std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(), |
| std::move(network), receiver), |
| rtp_extensions_, rtp_extensions_); |
| } |
| |
| void CallTest::CreateReceiveTransport( |
| const BuiltInNetworkBehaviorConfig& config, |
| RtpRtcpObserver* observer) { |
| auto network = std::make_unique<SimulatedNetwork>(config); |
| receive_simulated_network_ = network.get(); |
| receive_transport_ = std::make_unique<PacketTransport>( |
| task_queue(), nullptr, observer, test::PacketTransport::kReceiver, |
| payload_type_map_, |
| std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(), |
| std::move(network), |
| sender_call_->Receiver()), |
| rtp_extensions_, rtp_extensions_); |
| } |
| |
| void CallTest::ConnectVideoSourcesToStreams() { |
| for (size_t i = 0; i < video_sources_.size(); ++i) |
| video_send_streams_[i]->SetSource(video_sources_[i].get(), |
| degradation_preference_); |
| } |
| |
| void CallTest::Start() { |
| StartVideoStreams(); |
| if (audio_send_stream_) { |
| audio_send_stream_->Start(); |
| } |
| for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_) |
| audio_recv_stream->Start(); |
| } |
| |
| void CallTest::StartVideoSources() { |
| for (size_t i = 0; i < video_sources_.size(); ++i) { |
| video_sources_[i]->Start(); |
| } |
| } |
| |
| void CallTest::StartVideoStreams() { |
| StartVideoSources(); |
| for (size_t i = 0; i < video_send_streams_.size(); ++i) { |
| video_send_streams_[i]->Start(); |
| } |
| for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_) |
| video_recv_stream->Start(); |
| } |
| |
| void CallTest::Stop() { |
| for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_) |
| audio_recv_stream->Stop(); |
| if (audio_send_stream_) { |
| audio_send_stream_->Stop(); |
| } |
| StopVideoStreams(); |
| } |
| |
| void CallTest::StopVideoStreams() { |
| for (VideoSendStream* video_send_stream : video_send_streams_) |
| video_send_stream->Stop(); |
| for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_) |
| video_recv_stream->Stop(); |
| } |
| |
| void CallTest::DestroyStreams() { |
| if (audio_send_stream_) |
| sender_call_->DestroyAudioSendStream(audio_send_stream_); |
| audio_send_stream_ = nullptr; |
| for (AudioReceiveStreamInterface* audio_recv_stream : audio_receive_streams_) |
| receiver_call_->DestroyAudioReceiveStream(audio_recv_stream); |
| |
| DestroyVideoSendStreams(); |
| |
| for (VideoReceiveStreamInterface* video_recv_stream : video_receive_streams_) |
| receiver_call_->DestroyVideoReceiveStream(video_recv_stream); |
| |
| for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) |
| receiver_call_->DestroyFlexfecReceiveStream(flexfec_recv_stream); |
| |
| video_receive_streams_.clear(); |
| video_sources_.clear(); |
| } |
| |
| void CallTest::DestroyVideoSendStreams() { |
| for (VideoSendStream* video_send_stream : video_send_streams_) |
| sender_call_->DestroyVideoSendStream(video_send_stream); |
| video_send_streams_.clear(); |
| } |
| |
| void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) { |
| frame_generator_capturer_->SetFakeRotation(rotation); |
| } |
| |
| void CallTest::SetVideoDegradation(DegradationPreference preference) { |
| GetVideoSendStream()->SetSource(frame_generator_capturer_, preference); |
| } |
| |
| VideoSendStream::Config* CallTest::GetVideoSendConfig() { |
| return &video_send_configs_[0]; |
| } |
| |
| void CallTest::SetVideoSendConfig(const VideoSendStream::Config& config) { |
| video_send_configs_.clear(); |
| video_send_configs_.push_back(config.Copy()); |
| } |
| |
| VideoEncoderConfig* CallTest::GetVideoEncoderConfig() { |
| return &video_encoder_configs_[0]; |
| } |
| |
| void CallTest::SetVideoEncoderConfig(const VideoEncoderConfig& config) { |
| video_encoder_configs_.clear(); |
| video_encoder_configs_.push_back(config.Copy()); |
| } |
| |
| VideoSendStream* CallTest::GetVideoSendStream() { |
| return video_send_streams_[0]; |
| } |
| FlexfecReceiveStream::Config* CallTest::GetFlexFecConfig() { |
| return &flexfec_receive_configs_[0]; |
| } |
| |
| void CallTest::OnRtpPacket(const RtpPacketReceived& packet) { |
| // All FlexFEC streams protect all of the video streams. |
| for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) |
| flexfec_recv_stream->OnRtpPacket(packet); |
| } |
| |
| absl::optional<RtpExtension> CallTest::GetRtpExtensionByUri( |
| const std::string& uri) const { |
| for (const auto& extension : rtp_extensions_) { |
| if (extension.uri == uri) { |
| return extension; |
| } |
| } |
| return absl::nullopt; |
| } |
| |
| void CallTest::AddRtpExtensionByUri( |
| const std::string& uri, |
| std::vector<RtpExtension>* extensions) const { |
| const absl::optional<RtpExtension> extension = GetRtpExtensionByUri(uri); |
| if (extension) { |
| extensions->push_back(*extension); |
| } |
| } |
| |
| const std::map<uint8_t, MediaType> CallTest::payload_type_map_ = { |
| {VideoTestConstants::kVideoSendPayloadType, MediaType::VIDEO}, |
| {VideoTestConstants::kFakeVideoSendPayloadType, MediaType::VIDEO}, |
| {VideoTestConstants::kSendRtxPayloadType, MediaType::VIDEO}, |
| {VideoTestConstants::kPayloadTypeVP8, MediaType::VIDEO}, |
| {VideoTestConstants::kPayloadTypeVP9, MediaType::VIDEO}, |
| {VideoTestConstants::kPayloadTypeH264, MediaType::VIDEO}, |
| {VideoTestConstants::kPayloadTypeGeneric, MediaType::VIDEO}, |
| {VideoTestConstants::kRedPayloadType, MediaType::VIDEO}, |
| {VideoTestConstants::kRtxRedPayloadType, MediaType::VIDEO}, |
| {VideoTestConstants::kUlpfecPayloadType, MediaType::VIDEO}, |
| {VideoTestConstants::kFlexfecPayloadType, MediaType::VIDEO}, |
| {VideoTestConstants::kAudioSendPayloadType, MediaType::AUDIO}}; |
| |
| BaseTest::BaseTest() {} |
| |
| BaseTest::BaseTest(TimeDelta timeout) : RtpRtcpObserver(timeout) {} |
| |
| BaseTest::~BaseTest() {} |
| |
| std::unique_ptr<TestAudioDeviceModule::Capturer> BaseTest::CreateCapturer() { |
| return TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000); |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::Renderer> BaseTest::CreateRenderer() { |
| return TestAudioDeviceModule::CreateDiscardRenderer(48000); |
| } |
| |
| void BaseTest::OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device, |
| AudioDeviceModule* recv_audio_device) { |
| } |
| |
| void BaseTest::ModifySenderBitrateConfig(BitrateConstraints* bitrate_config) {} |
| |
| void BaseTest::ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) { |
| } |
| |
| void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {} |
| |
| void BaseTest::OnTransportCreated(PacketTransport* to_receiver, |
| SimulatedNetworkInterface* sender_network, |
| PacketTransport* to_sender, |
| SimulatedNetworkInterface* receiver_network) { |
| } |
| |
| BuiltInNetworkBehaviorConfig BaseTest::GetSendTransportConfig() const { |
| return BuiltInNetworkBehaviorConfig(); |
| } |
| BuiltInNetworkBehaviorConfig BaseTest::GetReceiveTransportConfig() const { |
| return BuiltInNetworkBehaviorConfig(); |
| } |
| size_t BaseTest::GetNumVideoStreams() const { |
| return 1; |
| } |
| |
| size_t BaseTest::GetNumAudioStreams() const { |
| return 0; |
| } |
| |
| size_t BaseTest::GetNumFlexfecStreams() const { |
| return 0; |
| } |
| |
| void BaseTest::ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStreamInterface::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) {} |
| |
| void BaseTest::ModifyVideoCaptureStartResolution(int* width, |
| int* heigt, |
| int* frame_rate) {} |
| |
| void BaseTest::ModifyVideoDegradationPreference( |
| DegradationPreference* degradation_preference) {} |
| |
| void BaseTest::OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStreamInterface*>& receive_streams) {} |
| |
| void BaseTest::ModifyAudioConfigs( |
| AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStreamInterface::Config>* receive_configs) {} |
| |
| void BaseTest::OnAudioStreamsCreated( |
| AudioSendStream* send_stream, |
| const std::vector<AudioReceiveStreamInterface*>& receive_streams) {} |
| |
| void BaseTest::ModifyFlexfecConfigs( |
| std::vector<FlexfecReceiveStream::Config>* receive_configs) {} |
| |
| void BaseTest::OnFlexfecStreamsCreated( |
| const std::vector<FlexfecReceiveStream*>& receive_streams) {} |
| |
| void BaseTest::OnFrameGeneratorCapturerCreated( |
| FrameGeneratorCapturer* frame_generator_capturer) {} |
| |
| void BaseTest::OnStreamsStopped() {} |
| |
| SendTest::SendTest(TimeDelta timeout) : BaseTest(timeout) {} |
| |
| bool SendTest::ShouldCreateReceivers() const { |
| return false; |
| } |
| |
| EndToEndTest::EndToEndTest() {} |
| |
| EndToEndTest::EndToEndTest(TimeDelta timeout) : BaseTest(timeout) {} |
| |
| bool EndToEndTest::ShouldCreateReceivers() const { |
| return true; |
| } |
| |
| } // namespace test |
| } // namespace webrtc |