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Diffstat (limited to 'third_party/libwebrtc/video/end_to_end_tests/call_operation_tests.cc')
| -rw-r--r-- | third_party/libwebrtc/video/end_to_end_tests/call_operation_tests.cc | 195 |
1 files changed, 195 insertions, 0 deletions
diff --git a/third_party/libwebrtc/video/end_to_end_tests/call_operation_tests.cc b/third_party/libwebrtc/video/end_to_end_tests/call_operation_tests.cc new file mode 100644 index 0000000000..f5b32388b1 --- /dev/null +++ b/third_party/libwebrtc/video/end_to_end_tests/call_operation_tests.cc @@ -0,0 +1,195 @@ +/* + * Copyright 2018 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 <memory> + +#include "api/test/create_frame_generator.h" +#include "api/test/frame_generator_interface.h" +#include "api/test/simulated_network.h" +#include "call/fake_network_pipe.h" +#include "call/simulated_network.h" +#include "rtc_base/task_queue_for_test.h" +#include "system_wrappers/include/sleep.h" +#include "test/call_test.h" +#include "test/field_trial.h" +#include "test/frame_forwarder.h" +#include "test/gtest.h" +#include "test/null_transport.h" + +namespace webrtc { + +class CallOperationEndToEndTest : public test::CallTest {}; + +TEST_F(CallOperationEndToEndTest, ReceiverCanBeStartedTwice) { + CreateCalls(); + + test::NullTransport transport; + CreateSendConfig(1, 0, 0, &transport); + CreateMatchingReceiveConfigs(&transport); + + CreateVideoStreams(); + + video_receive_streams_[0]->Start(); + video_receive_streams_[0]->Start(); + + DestroyStreams(); +} + +TEST_F(CallOperationEndToEndTest, ReceiverCanBeStoppedTwice) { + CreateCalls(); + + test::NullTransport transport; + CreateSendConfig(1, 0, 0, &transport); + CreateMatchingReceiveConfigs(&transport); + + CreateVideoStreams(); + + video_receive_streams_[0]->Stop(); + video_receive_streams_[0]->Stop(); + + DestroyStreams(); +} + +TEST_F(CallOperationEndToEndTest, ReceiverCanBeStoppedAndRestarted) { + CreateCalls(); + + test::NullTransport transport; + CreateSendConfig(1, 0, 0, &transport); + CreateMatchingReceiveConfigs(&transport); + + CreateVideoStreams(); + + video_receive_streams_[0]->Stop(); + video_receive_streams_[0]->Start(); + video_receive_streams_[0]->Stop(); + + DestroyStreams(); +} + +TEST_F(CallOperationEndToEndTest, RendersSingleDelayedFrame) { + static const int kWidth = 320; + static const int kHeight = 240; + // This constant is chosen to be higher than the timeout in the video_render + // module. This makes sure that frames aren't dropped if there are no other + // frames in the queue. + static const int kRenderDelayMs = 1000; + + class Renderer : public rtc::VideoSinkInterface<VideoFrame> { + public: + void OnFrame(const VideoFrame& video_frame) override { + SleepMs(kRenderDelayMs); + event_.Set(); + } + + bool Wait() { return event_.Wait(kDefaultTimeout); } + + rtc::Event event_; + } renderer; + + test::FrameForwarder frame_forwarder; + + SendTask( + task_queue(), [this, &renderer, &frame_forwarder]() { + CreateCalls(); + CreateSendTransport(BuiltInNetworkBehaviorConfig(), + /*observer=*/nullptr); + + CreateReceiveTransport(BuiltInNetworkBehaviorConfig(), + /*observer=*/nullptr); + CreateSendConfig(1, 0, 0); + CreateMatchingReceiveConfigs(); + + video_receive_configs_[0].renderer = &renderer; + + CreateVideoStreams(); + Start(); + + // Create frames that are smaller than the send width/height, this is + // done to check that the callbacks are done after processing video. + std::unique_ptr<test::FrameGeneratorInterface> frame_generator( + test::CreateSquareFrameGenerator(kWidth, kHeight, absl::nullopt, + absl::nullopt)); + GetVideoSendStream()->SetSource( + &frame_forwarder, DegradationPreference::MAINTAIN_FRAMERATE); + + test::FrameGeneratorInterface::VideoFrameData frame_data = + frame_generator->NextFrame(); + VideoFrame frame = VideoFrame::Builder() + .set_video_frame_buffer(frame_data.buffer) + .set_update_rect(frame_data.update_rect) + .build(); + frame_forwarder.IncomingCapturedFrame(frame); + }); + + EXPECT_TRUE(renderer.Wait()) + << "Timed out while waiting for the frame to render."; + + SendTask(task_queue(), [this]() { + Stop(); + DestroyStreams(); + DestroyCalls(); + }); +} + +TEST_F(CallOperationEndToEndTest, TransmitsFirstFrame) { + class Renderer : public rtc::VideoSinkInterface<VideoFrame> { + public: + void OnFrame(const VideoFrame& video_frame) override { event_.Set(); } + + bool Wait() { return event_.Wait(kDefaultTimeout); } + + rtc::Event event_; + } renderer; + + std::unique_ptr<test::FrameGeneratorInterface> frame_generator; + test::FrameForwarder frame_forwarder; + + std::unique_ptr<test::DirectTransport> sender_transport; + std::unique_ptr<test::DirectTransport> receiver_transport; + + SendTask( + task_queue(), [this, &renderer, &frame_generator, &frame_forwarder]() { + CreateCalls(); + CreateSendTransport(BuiltInNetworkBehaviorConfig(), + /*observer=*/nullptr); + CreateReceiveTransport(BuiltInNetworkBehaviorConfig(), + /*observer=*/nullptr); + + CreateSendConfig(1, 0, 0); + CreateMatchingReceiveConfigs(); + video_receive_configs_[0].renderer = &renderer; + + CreateVideoStreams(); + Start(); + + frame_generator = test::CreateSquareFrameGenerator( + kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt); + GetVideoSendStream()->SetSource( + &frame_forwarder, DegradationPreference::MAINTAIN_FRAMERATE); + test::FrameGeneratorInterface::VideoFrameData frame_data = + frame_generator->NextFrame(); + VideoFrame frame = VideoFrame::Builder() + .set_video_frame_buffer(frame_data.buffer) + .set_update_rect(frame_data.update_rect) + .build(); + frame_forwarder.IncomingCapturedFrame(frame); + }); + + EXPECT_TRUE(renderer.Wait()) + << "Timed out while waiting for the frame to render."; + + SendTask(task_queue(), [this]() { + Stop(); + DestroyStreams(); + DestroyCalls(); + }); +} + +} // namespace webrtc |