| /* |
| * 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 "webrtc/test/call_test.h" |
| |
| #include <algorithm> |
| |
| #include "webrtc/api/audio_codecs/builtin_audio_decoder_factory.h" |
| #include "webrtc/api/audio_codecs/builtin_audio_encoder_factory.h" |
| #include "webrtc/call/rtp_transport_controller_send.h" |
| #include "webrtc/config.h" |
| #include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
| #include "webrtc/rtc_base/checks.h" |
| #include "webrtc/rtc_base/event.h" |
| #include "webrtc/rtc_base/ptr_util.h" |
| #include "webrtc/test/testsupport/fileutils.h" |
| #include "webrtc/voice_engine/include/voe_base.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| namespace { |
| const int kVideoRotationRtpExtensionId = 4; |
| } |
| |
| CallTest::CallTest() |
| : clock_(Clock::GetRealTimeClock()), |
| event_log_(RtcEventLog::CreateNull()), |
| sender_call_transport_controller_(nullptr), |
| video_send_config_(nullptr), |
| video_send_stream_(nullptr), |
| audio_send_config_(nullptr), |
| audio_send_stream_(nullptr), |
| fake_encoder_(clock_), |
| num_video_streams_(1), |
| num_audio_streams_(0), |
| num_flexfec_streams_(0), |
| decoder_factory_(CreateBuiltinAudioDecoderFactory()), |
| encoder_factory_(CreateBuiltinAudioEncoderFactory()), |
| task_queue_("CallTestTaskQueue"), |
| fake_send_audio_device_(nullptr), |
| fake_recv_audio_device_(nullptr) {} |
| |
| CallTest::~CallTest() { |
| task_queue_.SendTask([this]() { |
| fake_send_audio_device_.reset(); |
| fake_recv_audio_device_.reset(); |
| frame_generator_capturer_.reset(); |
| }); |
| } |
| |
| void CallTest::RunBaseTest(BaseTest* test) { |
| task_queue_.SendTask([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); |
| Call::Config send_config(test->GetSenderCallConfig()); |
| if (num_audio_streams_ > 0) { |
| CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer()); |
| test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(), |
| fake_recv_audio_device_.get()); |
| apm_send_ = AudioProcessing::Create(); |
| apm_recv_ = AudioProcessing::Create(); |
| CreateVoiceEngines(); |
| AudioState::Config audio_state_config; |
| audio_state_config.voice_engine = voe_send_.voice_engine; |
| audio_state_config.audio_mixer = AudioMixerImpl::Create(); |
| audio_state_config.audio_processing = apm_send_; |
| send_config.audio_state = AudioState::Create(audio_state_config); |
| } |
| CreateSenderCall(send_config); |
| if (sender_call_transport_controller_ != nullptr) { |
| test->OnRtpTransportControllerSendCreated( |
| sender_call_transport_controller_); |
| } |
| if (test->ShouldCreateReceivers()) { |
| Call::Config recv_config(test->GetReceiverCallConfig()); |
| if (num_audio_streams_ > 0) { |
| AudioState::Config audio_state_config; |
| audio_state_config.voice_engine = voe_recv_.voice_engine; |
| audio_state_config.audio_mixer = AudioMixerImpl::Create(); |
| audio_state_config.audio_processing = apm_recv_; |
| recv_config.audio_state = AudioState::Create(audio_state_config); |
| } |
| CreateReceiverCall(recv_config); |
| } |
| test->OnCallsCreated(sender_call_.get(), receiver_call_.get()); |
| receive_transport_.reset(test->CreateReceiveTransport(&task_queue_)); |
| send_transport_.reset( |
| test->CreateSendTransport(&task_queue_, sender_call_.get())); |
| |
| if (test->ShouldCreateReceivers()) { |
| send_transport_->SetReceiver(receiver_call_->Receiver()); |
| receive_transport_->SetReceiver(sender_call_->Receiver()); |
| 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(receive_transport_.get()); |
| } |
| if (num_video_streams_ > 0) { |
| test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_, |
| &video_encoder_config_); |
| } |
| 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(video_send_stream_, video_receive_streams_); |
| } |
| if (num_audio_streams_ > 0) { |
| CreateAudioStreams(); |
| test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_); |
| } |
| |
| if (num_video_streams_ > 0) { |
| int width = kDefaultWidth; |
| int height = kDefaultHeight; |
| int frame_rate = kDefaultFramerate; |
| test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate); |
| CreateFrameGeneratorCapturer(frame_rate, width, height); |
| test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get()); |
| } |
| |
| Start(); |
| }); |
| |
| test->PerformTest(); |
| |
| task_queue_.SendTask([this]() { |
| Stop(); |
| DestroyStreams(); |
| send_transport_.reset(); |
| receive_transport_.reset(); |
| DestroyCalls(); |
| if (num_audio_streams_ > 0) |
| DestroyVoiceEngines(); |
| }); |
| |
| test->OnTestFinished(); |
| } |
| |
| void CallTest::CreateCalls(const Call::Config& sender_config, |
| const Call::Config& receiver_config) { |
| CreateSenderCall(sender_config); |
| CreateReceiverCall(receiver_config); |
| } |
| |
| void CallTest::CreateSenderCall(const Call::Config& config) { |
| sender_call_transport_controller_ = new RtpTransportControllerSend( |
| Clock::GetRealTimeClock(), config.event_log); |
| |
| sender_call_.reset( |
| Call::Create(config, std::unique_ptr<RtpTransportControllerSend>( |
| sender_call_transport_controller_))); |
| } |
| |
| void CallTest::CreateReceiverCall(const Call::Config& config) { |
| receiver_call_.reset(Call::Create(config)); |
| } |
| |
| void CallTest::DestroyCalls() { |
| sender_call_.reset(); |
| receiver_call_.reset(); |
| } |
| |
| void CallTest::CreateSendConfig(size_t num_video_streams, |
| size_t num_audio_streams, |
| size_t num_flexfec_streams, |
| Transport* send_transport) { |
| RTC_DCHECK(num_video_streams <= kNumSsrcs); |
| RTC_DCHECK_LE(num_audio_streams, 1); |
| RTC_DCHECK_LE(num_flexfec_streams, 1); |
| RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0); |
| if (num_video_streams > 0) { |
| video_send_config_ = VideoSendStream::Config(send_transport); |
| video_send_config_.encoder_settings.encoder = &fake_encoder_; |
| video_send_config_.encoder_settings.payload_name = "FAKE"; |
| video_send_config_.encoder_settings.payload_type = |
| kFakeVideoSendPayloadType; |
| video_send_config_.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
| kTransportSequenceNumberExtensionId)); |
| video_send_config_.rtp.extensions.push_back(RtpExtension( |
| RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId)); |
| FillEncoderConfiguration(num_video_streams, &video_encoder_config_); |
| |
| for (size_t i = 0; i < num_video_streams; ++i) |
| video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]); |
| video_send_config_.rtp.extensions.push_back(RtpExtension( |
| RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId)); |
| } |
| |
| if (num_audio_streams > 0) { |
| audio_send_config_ = AudioSendStream::Config(send_transport); |
| audio_send_config_.voe_channel_id = voe_send_.channel_id; |
| audio_send_config_.rtp.ssrc = kAudioSendSsrc; |
| audio_send_config_.send_codec_spec = |
| rtc::Optional<AudioSendStream::Config::SendCodecSpec>( |
| {kAudioSendPayloadType, {"OPUS", 48000, 2, {{"stereo", "1"}}}}); |
| audio_send_config_.encoder_factory = encoder_factory_; |
| } |
| |
| // TODO(brandtr): Update this when we support multistream protection. |
| if (num_flexfec_streams > 0) { |
| video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType; |
| video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc; |
| video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]}; |
| } |
| } |
| |
| void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) { |
| video_receive_configs_.clear(); |
| allocated_decoders_.clear(); |
| if (num_video_streams_ > 0) { |
| RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty()); |
| VideoReceiveStream::Config video_config(rtcp_send_transport); |
| video_config.rtp.remb = false; |
| video_config.rtp.transport_cc = true; |
| video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc; |
| for (const RtpExtension& extension : video_send_config_.rtp.extensions) |
| video_config.rtp.extensions.push_back(extension); |
| video_config.renderer = &fake_renderer_; |
| for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) { |
| VideoReceiveStream::Decoder decoder = |
| test::CreateMatchingDecoder(video_send_config_.encoder_settings); |
| allocated_decoders_.push_back( |
| std::unique_ptr<VideoDecoder>(decoder.decoder)); |
| video_config.decoders.clear(); |
| video_config.decoders.push_back(decoder); |
| video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i]; |
| video_receive_configs_.push_back(video_config.Copy()); |
| } |
| } |
| |
| RTC_DCHECK_GE(1, num_audio_streams_); |
| if (num_audio_streams_ == 1) { |
| RTC_DCHECK_LE(0, voe_send_.channel_id); |
| AudioReceiveStream::Config audio_config; |
| audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc; |
| audio_config.rtcp_send_transport = rtcp_send_transport; |
| audio_config.voe_channel_id = voe_recv_.channel_id; |
| audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc; |
| audio_config.decoder_factory = decoder_factory_; |
| audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}}; |
| audio_receive_configs_.push_back(audio_config); |
| } |
| |
| // TODO(brandtr): Update this when we support multistream protection. |
| RTC_DCHECK(num_flexfec_streams_ <= 1); |
| if (num_flexfec_streams_ == 1) { |
| FlexfecReceiveStream::Config config(rtcp_send_transport); |
| config.payload_type = kFlexfecPayloadType; |
| config.remote_ssrc = kFlexfecSendSsrc; |
| config.protected_media_ssrcs = {kVideoSendSsrcs[0]}; |
| config.local_ssrc = kReceiverLocalVideoSsrc; |
| for (const RtpExtension& extension : video_send_config_.rtp.extensions) |
| config.rtp_header_extensions.push_back(extension); |
| flexfec_receive_configs_.push_back(config); |
| } |
| } |
| |
| void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock, |
| float speed, |
| int framerate, |
| int width, |
| int height) { |
| frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create( |
| width, height, framerate * speed, clock)); |
| video_send_stream_->SetSource( |
| frame_generator_capturer_.get(), |
| VideoSendStream::DegradationPreference::kMaintainFramerate); |
| } |
| |
| void CallTest::CreateFrameGeneratorCapturer(int framerate, |
| int width, |
| int height) { |
| frame_generator_capturer_.reset( |
| test::FrameGeneratorCapturer::Create(width, height, framerate, clock_)); |
| video_send_stream_->SetSource( |
| frame_generator_capturer_.get(), |
| VideoSendStream::DegradationPreference::kMaintainFramerate); |
| } |
| |
| void CallTest::CreateFakeAudioDevices( |
| std::unique_ptr<FakeAudioDevice::Capturer> capturer, |
| std::unique_ptr<FakeAudioDevice::Renderer> renderer) { |
| fake_send_audio_device_.reset(new FakeAudioDevice( |
| std::move(capturer), nullptr, 1.f)); |
| fake_recv_audio_device_.reset(new FakeAudioDevice( |
| nullptr, std::move(renderer), 1.f)); |
| } |
| |
| void CallTest::CreateVideoStreams() { |
| RTC_DCHECK(video_send_stream_ == nullptr); |
| RTC_DCHECK(video_receive_streams_.empty()); |
| RTC_DCHECK(audio_send_stream_ == nullptr); |
| RTC_DCHECK(audio_receive_streams_.empty()); |
| |
| video_send_stream_ = sender_call_->CreateVideoSendStream( |
| video_send_config_.Copy(), video_encoder_config_.Copy()); |
| for (size_t i = 0; i < video_receive_configs_.size(); ++i) { |
| video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream( |
| video_receive_configs_[i].Copy())); |
| } |
| |
| AssociateFlexfecStreamsWithVideoStreams(); |
| } |
| |
| void CallTest::CreateAudioStreams() { |
| 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])); |
| } |
| |
| AssociateFlexfecStreamsWithVideoStreams(); |
| } |
| |
| void CallTest::AssociateFlexfecStreamsWithVideoStreams() { |
| // All FlexFEC streams protect all of the video streams. |
| for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) { |
| for (VideoReceiveStream* video_recv_stream : video_receive_streams_) { |
| video_recv_stream->AddSecondarySink(flexfec_recv_stream); |
| } |
| } |
| } |
| |
| void CallTest::DissociateFlexfecStreamsFromVideoStreams() { |
| for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) { |
| for (VideoReceiveStream* video_recv_stream : video_receive_streams_) { |
| video_recv_stream->RemoveSecondarySink(flexfec_recv_stream); |
| } |
| } |
| } |
| |
| void CallTest::Start() { |
| if (video_send_stream_) |
| video_send_stream_->Start(); |
| for (VideoReceiveStream* video_recv_stream : video_receive_streams_) |
| video_recv_stream->Start(); |
| if (audio_send_stream_) { |
| audio_send_stream_->Start(); |
| } |
| for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) |
| audio_recv_stream->Start(); |
| if (frame_generator_capturer_.get() != NULL) |
| frame_generator_capturer_->Start(); |
| } |
| |
| void CallTest::Stop() { |
| if (frame_generator_capturer_.get() != NULL) |
| frame_generator_capturer_->Stop(); |
| for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) |
| audio_recv_stream->Stop(); |
| if (audio_send_stream_) { |
| audio_send_stream_->Stop(); |
| } |
| for (VideoReceiveStream* video_recv_stream : video_receive_streams_) |
| video_recv_stream->Stop(); |
| if (video_send_stream_) |
| video_send_stream_->Stop(); |
| } |
| |
| void CallTest::DestroyStreams() { |
| DissociateFlexfecStreamsFromVideoStreams(); |
| |
| if (audio_send_stream_) |
| sender_call_->DestroyAudioSendStream(audio_send_stream_); |
| audio_send_stream_ = nullptr; |
| for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_) |
| receiver_call_->DestroyAudioReceiveStream(audio_recv_stream); |
| |
| if (video_send_stream_) |
| sender_call_->DestroyVideoSendStream(video_send_stream_); |
| video_send_stream_ = nullptr; |
| |
| for (VideoReceiveStream* 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(); |
| allocated_decoders_.clear(); |
| } |
| |
| void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) { |
| frame_generator_capturer_->SetFakeRotation(rotation); |
| } |
| |
| void CallTest::CreateVoiceEngines() { |
| voe_send_.voice_engine = VoiceEngine::Create(); |
| voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine); |
| EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), |
| apm_send_.get(), decoder_factory_)); |
| VoEBase::ChannelConfig config; |
| config.enable_voice_pacing = true; |
| voe_send_.channel_id = voe_send_.base->CreateChannel(config); |
| EXPECT_GE(voe_send_.channel_id, 0); |
| |
| voe_recv_.voice_engine = VoiceEngine::Create(); |
| voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine); |
| EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), |
| apm_recv_.get(), decoder_factory_)); |
| voe_recv_.channel_id = voe_recv_.base->CreateChannel(); |
| EXPECT_GE(voe_recv_.channel_id, 0); |
| } |
| |
| void CallTest::DestroyVoiceEngines() { |
| voe_recv_.base->DeleteChannel(voe_recv_.channel_id); |
| voe_recv_.channel_id = -1; |
| voe_recv_.base->Release(); |
| voe_recv_.base = nullptr; |
| |
| voe_send_.base->DeleteChannel(voe_send_.channel_id); |
| voe_send_.channel_id = -1; |
| voe_send_.base->Release(); |
| voe_send_.base = nullptr; |
| |
| VoiceEngine::Delete(voe_send_.voice_engine); |
| voe_send_.voice_engine = nullptr; |
| VoiceEngine::Delete(voe_recv_.voice_engine); |
| voe_recv_.voice_engine = nullptr; |
| } |
| |
| const int CallTest::kDefaultWidth; |
| const int CallTest::kDefaultHeight; |
| const int CallTest::kDefaultFramerate; |
| const int CallTest::kDefaultTimeoutMs = 30 * 1000; |
| const int CallTest::kLongTimeoutMs = 120 * 1000; |
| const uint8_t CallTest::kVideoSendPayloadType = 100; |
| const uint8_t CallTest::kFakeVideoSendPayloadType = 125; |
| const uint8_t CallTest::kSendRtxPayloadType = 98; |
| const uint8_t CallTest::kRedPayloadType = 118; |
| const uint8_t CallTest::kRtxRedPayloadType = 99; |
| const uint8_t CallTest::kUlpfecPayloadType = 119; |
| const uint8_t CallTest::kFlexfecPayloadType = 120; |
| const uint8_t CallTest::kAudioSendPayloadType = 103; |
| const uint8_t CallTest::kPayloadTypeH264 = 122; |
| const uint8_t CallTest::kPayloadTypeVP8 = 123; |
| const uint8_t CallTest::kPayloadTypeVP9 = 124; |
| const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE, |
| 0xBADCAFF}; |
| const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE, |
| 0xC0FFEF}; |
| const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF; |
| const uint32_t CallTest::kFlexfecSendSsrc = 0xBADBEEF; |
| const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456; |
| const uint32_t CallTest::kReceiverLocalAudioSsrc = 0x1234567; |
| const int CallTest::kNackRtpHistoryMs = 1000; |
| |
| const uint8_t CallTest::kDefaultKeepalivePayloadType = |
| RtpKeepAliveConfig().payload_type; |
| |
| const std::map<uint8_t, MediaType> CallTest::payload_type_map_ = { |
| {CallTest::kVideoSendPayloadType, MediaType::VIDEO}, |
| {CallTest::kFakeVideoSendPayloadType, MediaType::VIDEO}, |
| {CallTest::kSendRtxPayloadType, MediaType::VIDEO}, |
| {CallTest::kRedPayloadType, MediaType::VIDEO}, |
| {CallTest::kRtxRedPayloadType, MediaType::VIDEO}, |
| {CallTest::kUlpfecPayloadType, MediaType::VIDEO}, |
| {CallTest::kFlexfecPayloadType, MediaType::VIDEO}, |
| {CallTest::kAudioSendPayloadType, MediaType::AUDIO}, |
| {CallTest::kDefaultKeepalivePayloadType, MediaType::ANY}}; |
| |
| BaseTest::BaseTest() : event_log_(RtcEventLog::CreateNull()) {} |
| |
| BaseTest::BaseTest(unsigned int timeout_ms) |
| : RtpRtcpObserver(timeout_ms), event_log_(RtcEventLog::CreateNull()) {} |
| |
| BaseTest::~BaseTest() { |
| } |
| |
| std::unique_ptr<FakeAudioDevice::Capturer> BaseTest::CreateCapturer() { |
| return FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000); |
| } |
| |
| std::unique_ptr<FakeAudioDevice::Renderer> BaseTest::CreateRenderer() { |
| return FakeAudioDevice::CreateDiscardRenderer(48000); |
| } |
| |
| void BaseTest::OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device, |
| FakeAudioDevice* recv_audio_device) { |
| } |
| |
| Call::Config BaseTest::GetSenderCallConfig() { |
| return Call::Config(event_log_.get()); |
| } |
| |
| Call::Config BaseTest::GetReceiverCallConfig() { |
| return Call::Config(event_log_.get()); |
| } |
| |
| void BaseTest::OnRtpTransportControllerSendCreated( |
| RtpTransportControllerSend* controller) {} |
| |
| void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) { |
| } |
| |
| test::PacketTransport* BaseTest::CreateSendTransport( |
| SingleThreadedTaskQueueForTesting* task_queue, |
| Call* sender_call) { |
| return new PacketTransport( |
| task_queue, sender_call, this, test::PacketTransport::kSender, |
| CallTest::payload_type_map_, FakeNetworkPipe::Config()); |
| } |
| |
| test::PacketTransport* BaseTest::CreateReceiveTransport( |
| SingleThreadedTaskQueueForTesting* task_queue) { |
| return new PacketTransport( |
| task_queue, nullptr, this, test::PacketTransport::kReceiver, |
| CallTest::payload_type_map_, FakeNetworkPipe::Config()); |
| } |
| |
| 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<VideoReceiveStream::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) {} |
| |
| void BaseTest::ModifyVideoCaptureStartResolution(int* width, |
| int* heigt, |
| int* frame_rate) {} |
| |
| void BaseTest::OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) {} |
| |
| void BaseTest::ModifyAudioConfigs( |
| AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStream::Config>* receive_configs) {} |
| |
| void BaseTest::OnAudioStreamsCreated( |
| AudioSendStream* send_stream, |
| const std::vector<AudioReceiveStream*>& 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::OnTestFinished() { |
| } |
| |
| SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
| } |
| |
| bool SendTest::ShouldCreateReceivers() const { |
| return false; |
| } |
| |
| EndToEndTest::EndToEndTest() {} |
| |
| EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) { |
| } |
| |
| bool EndToEndTest::ShouldCreateReceivers() const { |
| return true; |
| } |
| |
| } // namespace test |
| } // namespace webrtc |