webrtc_m130/video/end_to_end_tests/call_operation_tests.cc
Mirko Bonadei 317a1f09ed Use std::make_unique instead of absl::make_unique.
WebRTC is now using C++14 so there is no need to use the Abseil version
of std::make_unique.

This CL has been created with the following steps:

git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt
git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt
git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt

diff --new-line-format="" --unchanged-line-format="" \
  /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \
  uniq > /tmp/only_make_unique.txt
diff --new-line-format="" --unchanged-line-format="" \
  /tmp/only_make_unique.txt /tmp/memory.txt | \
  xargs grep -l "absl/memory" > /tmp/add-memory.txt

git grep -l "\babsl::make_unique\b" | \
  xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g"

git checkout PRESUBMIT.py abseil-in-webrtc.md

cat /tmp/add-memory.txt | \
  xargs sed -i \
  's/#include "absl\/memory\/memory.h"/#include <memory>/g'
git cl format
# Manual fix order of the new inserted #include <memory>

cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \
  xargs sed -i '/#include "absl\/memory\/memory.h"/d'

git ls-files | grep BUILD.gn | \
  xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d'

python tools_webrtc/gn_check_autofix.py \
  -m tryserver.webrtc -b linux_rel

# Repead the gn_check_autofix step for other platforms

git ls-files | grep BUILD.gn | \
  xargs sed -i 's/absl\/memory:memory/absl\/memory/g'
git cl format

Bug: webrtc:10945
Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29209}
2019-09-17 15:47:29 +00:00

208 lines
6.7 KiB
C++

/*
* 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/simulated_network.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "system_wrappers/include/sleep.h"
#include "test/call_test.h"
#include "test/field_trial.h"
#include "test/frame_generator.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(kDefaultTimeoutMs); }
rtc::Event event_;
} renderer;
test::FrameForwarder frame_forwarder;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
task_queue_.SendTask([this, &renderer, &frame_forwarder, &sender_transport,
&receiver_transport]() {
CreateCalls();
sender_transport = std::make_unique<test::DirectTransport>(
&task_queue_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
sender_call_.get(), payload_type_map_);
receiver_transport = std::make_unique<test::DirectTransport>(
&task_queue_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
receiver_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
receiver_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(receiver_transport.get());
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::FrameGenerator> frame_generator(
test::FrameGenerator::CreateSquareGenerator(
kWidth, kHeight, absl::nullopt, absl::nullopt));
GetVideoSendStream()->SetSource(&frame_forwarder,
DegradationPreference::MAINTAIN_FRAMERATE);
frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
});
EXPECT_TRUE(renderer.Wait())
<< "Timed out while waiting for the frame to render.";
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
receiver_transport.reset();
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(kDefaultTimeoutMs); }
rtc::Event event_;
} renderer;
std::unique_ptr<test::FrameGenerator> frame_generator;
test::FrameForwarder frame_forwarder;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
task_queue_.SendTask([this, &renderer, &frame_generator, &frame_forwarder,
&sender_transport, &receiver_transport]() {
CreateCalls();
sender_transport = std::make_unique<test::DirectTransport>(
&task_queue_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
sender_call_.get(), payload_type_map_);
receiver_transport = std::make_unique<test::DirectTransport>(
&task_queue_,
std::make_unique<FakeNetworkPipe>(
Clock::GetRealTimeClock(),
std::make_unique<SimulatedNetwork>(BuiltInNetworkBehaviorConfig())),
receiver_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
receiver_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(receiver_transport.get());
video_receive_configs_[0].renderer = &renderer;
CreateVideoStreams();
Start();
frame_generator = test::FrameGenerator::CreateSquareGenerator(
kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt);
GetVideoSendStream()->SetSource(&frame_forwarder,
DegradationPreference::MAINTAIN_FRAMERATE);
frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
});
EXPECT_TRUE(renderer.Wait())
<< "Timed out while waiting for the frame to render.";
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
receiver_transport.reset();
DestroyCalls();
});
}
} // namespace webrtc