|  | /* | 
|  | *  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/builtin_audio_processing_builder.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(); | 
|  | CallConfig recv_config = RecvCallConfig(); | 
|  | 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_ = BuiltinAudioProcessingBuilder().Build(send_config.env); | 
|  | apm_recv_ = BuiltinAudioProcessingBuilder().Build(recv_config.env); | 
|  | 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()) { | 
|  | 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_, std::nullopt, false, | 
|  | 0); | 
|  | } | 
|  |  | 
|  | void CallTest::CreateMatchingVideoReceiveConfigs( | 
|  | const VideoSendStream::Config& video_send_config, | 
|  | Transport* rtcp_send_transport, | 
|  | VideoDecoderFactory* decoder_factory, | 
|  | std::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, | 
|  | std::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, | 
|  | 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, std::nullopt, | 
|  | std::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, std::nullopt, | 
|  | std::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); | 
|  | } | 
|  |  | 
|  | std::optional<RtpExtension> CallTest::GetRtpExtensionByUri( | 
|  | const std::string& uri) const { | 
|  | for (const auto& extension : rtp_extensions_) { | 
|  | if (extension.uri == uri) { | 
|  | return extension; | 
|  | } | 
|  | } | 
|  | return std::nullopt; | 
|  | } | 
|  |  | 
|  | void CallTest::AddRtpExtensionByUri( | 
|  | const std::string& uri, | 
|  | std::vector<RtpExtension>* extensions) const { | 
|  | const std::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 |