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Googletest recently started replacing the term Test Case by Test Suite. From now on, the preferred API is TestSuite*; the older TestCase* API will be slowly deprecated. This CL moves WebRTC to the new set of APIs. More info in [1]. This CL has been generated with this script: declare -A items items[TYPED_TEST_CASE]=TYPED_TEST_SUITE items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P for i in "${!items[@]}" do git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g" done git cl format [1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature Bug: None Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f Reviewed-on: https://webrtc-review.googlesource.com/c/118701 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26494}
221 lines
7.3 KiB
C++
221 lines
7.3 KiB
C++
/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "absl/memory/memory.h"
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#include "api/test/simulated_network.h"
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#include "call/fake_network_pipe.h"
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#include "call/simulated_network.h"
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#include "system_wrappers/include/sleep.h"
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#include "test/call_test.h"
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#include "test/field_trial.h"
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#include "test/frame_generator.h"
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#include "test/gtest.h"
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#include "test/null_transport.h"
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namespace webrtc {
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class CallOperationEndToEndTest
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: public test::CallTest,
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public testing::WithParamInterface<std::string> {
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public:
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CallOperationEndToEndTest() : field_trial_(GetParam()) {}
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private:
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test::ScopedFieldTrials field_trial_;
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};
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INSTANTIATE_TEST_SUITE_P(
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FieldTrials,
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CallOperationEndToEndTest,
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::testing::Values("WebRTC-TaskQueueCongestionControl/Enabled/",
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"WebRTC-TaskQueueCongestionControl/Disabled/"));
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TEST_P(CallOperationEndToEndTest, ReceiverCanBeStartedTwice) {
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CreateCalls();
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test::NullTransport transport;
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CreateSendConfig(1, 0, 0, &transport);
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CreateMatchingReceiveConfigs(&transport);
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CreateVideoStreams();
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video_receive_streams_[0]->Start();
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video_receive_streams_[0]->Start();
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DestroyStreams();
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}
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TEST_P(CallOperationEndToEndTest, ReceiverCanBeStoppedTwice) {
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CreateCalls();
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test::NullTransport transport;
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CreateSendConfig(1, 0, 0, &transport);
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CreateMatchingReceiveConfigs(&transport);
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CreateVideoStreams();
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video_receive_streams_[0]->Stop();
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video_receive_streams_[0]->Stop();
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DestroyStreams();
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}
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TEST_P(CallOperationEndToEndTest, ReceiverCanBeStoppedAndRestarted) {
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CreateCalls();
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test::NullTransport transport;
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CreateSendConfig(1, 0, 0, &transport);
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CreateMatchingReceiveConfigs(&transport);
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CreateVideoStreams();
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video_receive_streams_[0]->Stop();
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video_receive_streams_[0]->Start();
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video_receive_streams_[0]->Stop();
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DestroyStreams();
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}
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TEST_P(CallOperationEndToEndTest, RendersSingleDelayedFrame) {
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static const int kWidth = 320;
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static const int kHeight = 240;
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// This constant is chosen to be higher than the timeout in the video_render
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// module. This makes sure that frames aren't dropped if there are no other
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// frames in the queue.
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static const int kRenderDelayMs = 1000;
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class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
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public:
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void OnFrame(const VideoFrame& video_frame) override {
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SleepMs(kRenderDelayMs);
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event_.Set();
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}
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bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
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rtc::Event event_;
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} renderer;
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test::FrameForwarder frame_forwarder;
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std::unique_ptr<test::DirectTransport> sender_transport;
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std::unique_ptr<test::DirectTransport> receiver_transport;
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task_queue_.SendTask([this, &renderer, &frame_forwarder, &sender_transport,
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&receiver_transport]() {
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CreateCalls();
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sender_transport = absl::make_unique<test::DirectTransport>(
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&task_queue_,
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absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
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absl::make_unique<SimulatedNetwork>(
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BuiltInNetworkBehaviorConfig())),
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sender_call_.get(), payload_type_map_);
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receiver_transport = absl::make_unique<test::DirectTransport>(
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&task_queue_,
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absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
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absl::make_unique<SimulatedNetwork>(
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BuiltInNetworkBehaviorConfig())),
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receiver_call_.get(), payload_type_map_);
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sender_transport->SetReceiver(receiver_call_->Receiver());
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receiver_transport->SetReceiver(sender_call_->Receiver());
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CreateSendConfig(1, 0, 0, sender_transport.get());
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CreateMatchingReceiveConfigs(receiver_transport.get());
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video_receive_configs_[0].renderer = &renderer;
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CreateVideoStreams();
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Start();
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// Create frames that are smaller than the send width/height, this is done
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// to check that the callbacks are done after processing video.
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std::unique_ptr<test::FrameGenerator> frame_generator(
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test::FrameGenerator::CreateSquareGenerator(
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kWidth, kHeight, absl::nullopt, absl::nullopt));
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GetVideoSendStream()->SetSource(&frame_forwarder,
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DegradationPreference::MAINTAIN_FRAMERATE);
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frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
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});
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EXPECT_TRUE(renderer.Wait())
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<< "Timed out while waiting for the frame to render.";
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task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
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Stop();
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DestroyStreams();
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sender_transport.reset();
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receiver_transport.reset();
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DestroyCalls();
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});
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}
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TEST_P(CallOperationEndToEndTest, TransmitsFirstFrame) {
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class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
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public:
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void OnFrame(const VideoFrame& video_frame) override { event_.Set(); }
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bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
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rtc::Event event_;
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} renderer;
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std::unique_ptr<test::FrameGenerator> frame_generator;
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test::FrameForwarder frame_forwarder;
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std::unique_ptr<test::DirectTransport> sender_transport;
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std::unique_ptr<test::DirectTransport> receiver_transport;
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task_queue_.SendTask([this, &renderer, &frame_generator, &frame_forwarder,
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&sender_transport, &receiver_transport]() {
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CreateCalls();
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sender_transport = absl::make_unique<test::DirectTransport>(
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&task_queue_,
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absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
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absl::make_unique<SimulatedNetwork>(
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BuiltInNetworkBehaviorConfig())),
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sender_call_.get(), payload_type_map_);
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receiver_transport = absl::make_unique<test::DirectTransport>(
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&task_queue_,
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absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
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absl::make_unique<SimulatedNetwork>(
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BuiltInNetworkBehaviorConfig())),
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receiver_call_.get(), payload_type_map_);
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sender_transport->SetReceiver(receiver_call_->Receiver());
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receiver_transport->SetReceiver(sender_call_->Receiver());
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CreateSendConfig(1, 0, 0, sender_transport.get());
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CreateMatchingReceiveConfigs(receiver_transport.get());
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video_receive_configs_[0].renderer = &renderer;
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CreateVideoStreams();
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Start();
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frame_generator = test::FrameGenerator::CreateSquareGenerator(
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kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt);
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GetVideoSendStream()->SetSource(&frame_forwarder,
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DegradationPreference::MAINTAIN_FRAMERATE);
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frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
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});
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EXPECT_TRUE(renderer.Wait())
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<< "Timed out while waiting for the frame to render.";
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task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
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Stop();
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DestroyStreams();
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sender_transport.reset();
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receiver_transport.reset();
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DestroyCalls();
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});
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}
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} // namespace webrtc
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